/*
- Copyright (C) 2012-2016 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_decoder_stream.h"
#include "audio_content.h"
+#include "dcpomatic_log.h"
#include "log.h"
+#include "resampler.h"
#include "compose.hpp"
-#include <boost/foreach.hpp>
#include <iostream>
#include "i18n.h"
+
using std::cout;
-using std::map;
-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)
- : DecoderPart (parent, log)
+AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr<const AudioContent> content, bool fast)
+ : DecoderPart (parent)
+ , _content (content)
+ , _fast (fast)
{
- BOOST_FOREACH (AudioStreamPtr i, content->streams ()) {
- _streams[i] = shared_ptr<AudioDecoderStream> (new AudioDecoderStream (content, i, parent, this, log));
+ /* 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 (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 flushing)
{
if (ignore ()) {
return;
}
- if (_streams.find (stream) == _streams.end ()) {
+ int const resampled_rate = _content->resampled_frame_rate(film);
+ if (!flushing) {
+ time += ContentTime::from_seconds (_content->delay() / 1000.0);
+ }
- /* This method can be called with an unknown stream during the following sequence:
- - Add KDM to some DCP content.
- - Content gets re-examined.
- - SingleStreamAudioContent::take_from_audio_examiner creates a new stream.
- - Some content property change signal is delivered so Player::Changed is emitted.
- - Film viewer to re-gets the frame.
- - Player calls DCPDecoder pass which calls this method on the new stream.
+ /* Amount of error we will tolerate on audio timestamps; see comment below.
+ * We'll use 1 24fps video frame as this seems to be roughly how ffplay does it.
+ */
+ Frame const slack_frames = resampled_rate / 24;
+
+ /* first_since_seek is set to true if this is the first data we have
+ received since initialisation or seek. We'll 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.
+ */
+
+ auto const first_since_seek = _positions[stream] == 0;
+ auto const need_reset = !first_since_seek && (std::abs(_positions[stream] - time.frames_round(resampled_rate)) > slack_frames);
+
+ if (need_reset) {
+ 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 (first_since_seek || need_reset) {
+ _positions[stream] = time.frames_round (resampled_rate);
+ }
- At this point the AudioDecoder does not know about the new stream.
+ if (first_since_seek && _content->delay() > 0) {
+ silence (stream, _content->delay());
+ }
- Then
- - Some other property change signal is delivered which marks the player's pieces invalid.
- - Film viewer re-gets again.
- - Everything is OK.
+ shared_ptr<Resampler> resampler;
+ 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;
+ }
+ }
- In this situation it is fine for us to silently drop the audio.
- */
+ if (resampler && !flushing) {
+ /* It can be the the data here has a different number of channels than the stream
+ * it comes from (e.g. the files decoded by FFmpegDecoder sometimes have a random
+ * frame, often at the end, with more channels). Insert silence or discard channels
+ * here.
+ */
+ if (resampler->channels() != data->channels()) {
+ LOG_WARNING("Received audio data with an unexpected channel count of %1 instead of %2", data->channels(), resampler->channels());
+ auto data_copy = data->clone();
+ data_copy->set_channels(resampler->channels());
+ data = resampler->run(data_copy);
+ } else {
+ data = resampler->run(data);
+ }
- return;
+ if (data->frames() == 0) {
+ return;
+ }
}
- _streams[stream]->audio (data, time);
+ Data(stream, ContentAudio (data, _positions[stream]));
+ _positions[stream] += data->frames();
}
-void
-AudioDecoder::flush ()
+
+/** @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
{
- for (StreamMap::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
- i->second->flush ();
+ 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::set_fast ()
+AudioDecoder::seek ()
{
- for (StreamMap::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
- i->second->set_fast ();
+ for (auto i: _resamplers) {
+ i.second->flush ();
+ i.second->reset ();
+ }
+
+ for (auto& i: _positions) {
+ i.second = 0;
}
}
-optional<ContentTime>
-AudioDecoder::position () const
+
+void
+AudioDecoder::flush ()
{
- optional<ContentTime> p;
- for (map<AudioStreamPtr, ContentTime>::const_iterator i = _positions.begin(); i != _positions.end(); ++i) {
- if (!p || i->second < *p) {
- p = i->second;
+ for (auto const& i: _resamplers) {
+ auto ro = i.second->flush ();
+ if (ro->frames() > 0) {
+ Data (i.first, ContentAudio (ro, _positions[i.first]));
+ _positions[i.first] += ro->frames();
}
}
- return p;
+ if (_content->delay() < 0) {
+ /* Finish off with the gap caused by the delay */
+ for (auto stream: _content->streams()) {
+ silence (stream, -_content->delay());
+ }
+ }
}
+
void
-AudioDecoder::set_position (AudioStreamPtr stream, ContentTime time)
+AudioDecoder::silence (AudioStreamPtr stream, int milliseconds)
{
- _positions[stream] = time;
+ int const samples = ContentTime::from_seconds(milliseconds / 1000.0).frames_round(stream->frame_rate());
+ auto silence = make_shared<AudioBuffers>(stream->channels(), samples);
+ silence->make_silent ();
+ Data (stream, ContentAudio(silence, _positions[stream]));
}
projects / dcpomatic.git / blobdiff