2 Copyright (C) 2012-2021 Carl Hetherington <cth@carlh.net>
4 This file is part of DCP-o-matic.
6 DCP-o-matic is free software; you can redistribute it and/or modify
7 it under the terms of the GNU General Public License as published by
8 the Free Software Foundation; either version 2 of the License, or
9 (at your option) any later version.
11 DCP-o-matic is distributed in the hope that it will be useful,
12 but WITHOUT ANY WARRANTY; without even the implied warranty of
13 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14 GNU General Public License for more details.
16 You should have received a copy of the GNU General Public License
17 along with DCP-o-matic. If not, see <http://www.gnu.org/licenses/>.
22 #include "audio_decoder.h"
23 #include "audio_buffers.h"
24 #include "audio_content.h"
25 #include "dcpomatic_log.h"
27 #include "resampler.h"
28 #include "compose.hpp"
35 using std::shared_ptr;
36 using std::make_shared;
37 using boost::optional;
38 using namespace dcpomatic;
41 AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr<const AudioContent> content, bool fast)
42 : DecoderPart (parent)
46 /* Set up _positions so that we have one for each stream */
47 for (auto i: content->streams ()) {
53 /** @param time_already_delayed true if the delay should not be added to time */
55 AudioDecoder::emit(shared_ptr<const Film> film, AudioStreamPtr stream, shared_ptr<const AudioBuffers> data, ContentTime time, bool flushing)
61 int const resampled_rate = _content->resampled_frame_rate(film);
63 time += ContentTime::from_seconds (_content->delay() / 1000.0);
66 /* Amount of error we will tolerate on audio timestamps; see comment below.
67 * We'll use 1 24fps video frame as this seems to be roughly how ffplay does it.
69 Frame const slack_frames = resampled_rate / 24;
71 /* first_since_seek is set to true if this is the first data we have
72 received since initialisation or seek. We'll set the position based
73 on the ContentTime that was given. After this first time we just
74 count samples unless the timestamp is more than slack_frames away
75 from where we think it should be. This is because ContentTimes seem
76 to be slightly unreliable from FFmpegDecoder (i.e. not sample
77 accurate), but we still need to obey them sometimes otherwise we get
78 sync problems such as #1833.
81 auto const first_since_seek = _positions[stream] == 0;
82 auto const need_reset = !first_since_seek && (std::abs(_positions[stream] - time.frames_round(resampled_rate)) > slack_frames);
86 "Reset audio position: was %1, new data at %2, slack: %3 frames",
88 time.frames_round(resampled_rate),
89 std::abs(_positions[stream] - time.frames_round(resampled_rate))
93 if (first_since_seek || need_reset) {
94 _positions[stream] = time.frames_round (resampled_rate);
97 if (first_since_seek && _content->delay() > 0) {
98 silence (stream, _content->delay());
101 shared_ptr<Resampler> resampler;
102 auto i = _resamplers.find(stream);
103 if (i != _resamplers.end()) {
104 resampler = i->second;
106 if (stream->frame_rate() != resampled_rate) {
108 "Creating new resampler from %1 to %2 with %3 channels",
109 stream->frame_rate(),
114 resampler = make_shared<Resampler>(stream->frame_rate(), resampled_rate, stream->channels());
116 resampler->set_fast ();
118 _resamplers[stream] = resampler;
122 if (resampler && !flushing) {
123 /* It can be the the data here has a different number of channels than the stream
124 * it comes from (e.g. the files decoded by FFmpegDecoder sometimes have a random
125 * frame, often at the end, with more channels). Insert silence or discard channels
128 if (resampler->channels() != data->channels()) {
129 LOG_WARNING("Received audio data with an unexpected channel count of %1 instead of %2", data->channels(), resampler->channels());
130 auto data_copy = data->clone();
131 data_copy->set_channels(resampler->channels());
132 data = resampler->run(data_copy);
134 data = resampler->run(data);
137 if (data->frames() == 0) {
142 Data(stream, ContentAudio (data, _positions[stream]));
143 _positions[stream] += data->frames();
147 /** @return Time just after the last thing that was emitted from a given stream */
149 AudioDecoder::stream_position (shared_ptr<const Film> film, AudioStreamPtr stream) const
151 auto i = _positions.find (stream);
152 DCPOMATIC_ASSERT (i != _positions.end ());
153 return ContentTime::from_frames (i->second, _content->resampled_frame_rate(film));
157 boost::optional<ContentTime>
158 AudioDecoder::position (shared_ptr<const Film> film) const
160 optional<ContentTime> p;
161 for (auto i: _positions) {
162 auto const ct = stream_position (film, i.first);
173 AudioDecoder::seek ()
175 for (auto i: _resamplers) {
180 for (auto& i: _positions) {
187 AudioDecoder::flush ()
189 for (auto const& i: _resamplers) {
190 auto ro = i.second->flush ();
191 if (ro->frames() > 0) {
192 Data (i.first, ContentAudio (ro, _positions[i.first]));
193 _positions[i.first] += ro->frames();
197 if (_content->delay() < 0) {
198 /* Finish off with the gap caused by the delay */
199 for (auto stream: _content->streams()) {
200 silence (stream, -_content->delay());
207 AudioDecoder::silence (AudioStreamPtr stream, int milliseconds)
209 int const samples = ContentTime::from_seconds(milliseconds / 1000.0).frames_round(stream->frame_rate());
210 auto silence = make_shared<AudioBuffers>(stream->channels(), samples);
211 silence->make_silent ();
212 Data (stream, ContentAudio(silence, _positions[stream]));