/*
- 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 <boost/foreach.hpp>
+#include "dcpomatic_log.h"
+#include "log.h"
+#include "resampler.h"
+#include "compose.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;
}
}
-ContentAudio
-AudioDecoder::get (AudioStreamPtr stream, Frame frame, Frame length, bool accurate)
-{
- return _streams[stream]->get (frame, length, accurate);
-}
+/** @param time_already_delayed true if the delay should not be added to time */
void
-AudioDecoder::give (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;
}
- if (_streams.find (stream) == _streams.end ()) {
+ /* 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;
- /* 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.
+ int const resampled_rate = _content->resampled_frame_rate(film);
+ if (!time_already_delayed) {
+ time += ContentTime::from_seconds (_content->delay() / 1000.0);
+ }
- At this point the AudioDecoder does not know about the new stream.
+ 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))
+ );
+ }
- Then
- - Some other property change signal is delivered which marks the player's pieces invalid.
- - Film viewer re-gets again.
- - Everything is OK.
+ if (reset) {
+ _positions[stream] = time.frames_round (resampled_rate);
+ }
- In this situation it is fine for us to silently drop the audio.
- */
+ 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;
+ }
+ }
- return;
+ if (resampler) {
+ auto ro = resampler->run (data);
+ if (ro->frames() == 0) {
+ return;
+ }
+ data = ro;
}
- _streams[stream]->audio (data, time);
+ 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::flush ()
+AudioDecoder::seek ()
{
- for (map<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
- i->second->flush ();
+ for (auto i: _resamplers) {
+ i.second->flush ();
+ i.second->reset ();
+ }
+
+ for (auto& i: _positions) {
+ i.second = 0;
}
}
+
void
-AudioDecoder::seek (ContentTime t, bool accurate)
+AudioDecoder::flush ()
{
- for (map<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
- i->second->seek (t, accurate);
+ 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();
+ }
+ }
+
+ if (_content->delay() < 0) {
+ /* Finish off with the gap caused by the delay */
+ silence (-_content->delay ());
}
}
+
void
-AudioDecoder::set_fast ()
+AudioDecoder::silence (int milliseconds)
{
- for (map<AudioStreamPtr, shared_ptr<AudioDecoderStream> >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) {
- i->second->set_fast ();
+ 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]));
}
}