X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=ca1faa0100492b5e76363cde68b0aa4dbed1e9ab;hb=f49a724918ad3d1082384576960b1098d7f15822;hp=5334dfa345071e887fa6e9f7a04d34701d23054b;hpb=dd9be86db6cde0afa5da0d1d1ac43b42e05dca26;p=dcpomatic.git diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc index 5334dfa34..ca1faa010 100644 --- a/src/lib/audio_decoder.cc +++ b/src/lib/audio_decoder.cc @@ -1,5 +1,5 @@ /* - Copyright (C) 2012-2018 Carl Hetherington + Copyright (C) 2012-2021 Carl Hetherington This file is part of DCP-o-matic. @@ -18,6 +18,7 @@ */ + #include "audio_decoder.h" #include "audio_buffers.h" #include "audio_content.h" @@ -25,29 +26,30 @@ #include "log.h" #include "resampler.h" #include "compose.hpp" -#include #include #include "i18n.h" + using std::cout; -using std::map; -using std::pair; using std::shared_ptr; +using std::make_shared; using boost::optional; using namespace dcpomatic; + AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr 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 ()) { + 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 (shared_ptr film, AudioStreamPtr stream, shared_ptr data, ContentTime time, bool time_already_delayed) @@ -56,33 +58,30 @@ AudioDecoder::emit (shared_ptr film, AudioStreamPtr stream, shared_p 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 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; + /* 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], @@ -91,13 +90,17 @@ AudioDecoder::emit (shared_ptr film, AudioStreamPtr stream, shared_p ); } - if (reset) { + if (first_since_seek || need_reset) { _positions[stream] = time.frames_round (resampled_rate); } + if (first_since_seek && _content->delay() > 0) { + silence (stream, _content->delay()); + } + shared_ptr resampler; - ResamplerMap::iterator i = _resamplers.find(stream); - if (i != _resamplers.end ()) { + auto i = _resamplers.find(stream); + if (i != _resamplers.end()) { resampler = i->second; } else { if (stream->frame_rate() != resampled_rate) { @@ -108,7 +111,7 @@ AudioDecoder::emit (shared_ptr film, AudioStreamPtr stream, shared_p stream->channels() ); - resampler.reset (new Resampler(stream->frame_rate(), resampled_rate, stream->channels())); + resampler = make_shared(stream->frame_rate(), resampled_rate, stream->channels()); if (_fast) { resampler->set_fast (); } @@ -117,7 +120,7 @@ AudioDecoder::emit (shared_ptr film, AudioStreamPtr stream, shared_p } if (resampler) { - shared_ptr ro = resampler->run (data); + auto ro = resampler->run (data); if (ro->frames() == 0) { return; } @@ -128,21 +131,23 @@ AudioDecoder::emit (shared_ptr film, AudioStreamPtr stream, shared_p _positions[stream] += data->frames(); } + /** @return Time just after the last thing that was emitted from a given stream */ ContentTime AudioDecoder::stream_position (shared_ptr film, AudioStreamPtr stream) const { - PositionMap::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(film)); } + boost::optional AudioDecoder::position (shared_ptr film) const { optional p; - for (PositionMap::const_iterator i = _positions.begin(); i != _positions.end(); ++i) { - ContentTime const ct = stream_position (film, i->first); + for (auto i: _positions) { + auto const ct = stream_position (film, i.first); if (!p || ct < *p) { p = ct; } @@ -151,43 +156,46 @@ AudioDecoder::position (shared_ptr film) const return p; } + void AudioDecoder::seek () { - for (ResamplerMap::iterator i = _resamplers.begin(); i != _resamplers.end(); ++i) { - i->second->flush (); - i->second->reset (); + for (auto i: _resamplers) { + i.second->flush (); + i.second->reset (); } - for (PositionMap::iterator i = _positions.begin(); i != _positions.end(); ++i) { - i->second = 0; + for (auto& i: _positions) { + i.second = 0; } } + void AudioDecoder::flush () { - for (ResamplerMap::iterator i = _resamplers.begin(); i != _resamplers.end(); ++i) { - shared_ptr ro = i->second->flush (); + 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(); + 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 ()); + for (auto stream: _content->streams()) { + silence (stream, -_content->delay()); + } } } + void -AudioDecoder::silence (int milliseconds) +AudioDecoder::silence (AudioStreamPtr stream, int milliseconds) { - BOOST_FOREACH (AudioStreamPtr i, _content->streams ()) { - int const samples = ContentTime::from_seconds(milliseconds / 1000.0).frames_round(i->frame_rate()); - shared_ptr silence (new AudioBuffers (i->channels(), samples)); - silence->make_silent (); - Data (i, ContentAudio (silence, _positions[i])); - } + int const samples = ContentTime::from_seconds(milliseconds / 1000.0).frames_round(stream->frame_rate()); + auto silence = make_shared(stream->channels(), samples); + silence->make_silent (); + Data (stream, ContentAudio(silence, _positions[stream])); }