X-Git-Url: https://main.carlh.net/gitweb/?p=dcpomatic.git;a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=77c9b0695a3d7b6f4c7d3b57df84193e017863eb;hp=fd0835596924c8d2e19abde7d742396740d9d71b;hb=da44da6f31f97d39ca91c35955e573e76371f2c2;hpb=f02d4c995a312e7ccf2eac7a2cb71d8e68d77189 diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc index fd0835596..77c9b0695 100644 --- a/src/lib/audio_decoder.cc +++ b/src/lib/audio_decoder.cc @@ -1,5 +1,5 @@ /* - Copyright (C) 2012-2016 Carl Hetherington + Copyright (C) 2012-2021 Carl Hetherington This file is part of DCP-o-matic. @@ -18,89 +18,185 @@ */ + #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 #include #include "i18n.h" + using std::cout; using std::map; -using boost::shared_ptr; +using std::pair; +using std::shared_ptr; +using std::make_shared; +using boost::optional; +using namespace dcpomatic; -AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr content, shared_ptr log) - : DecoderPart (parent, log) + +AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr content, bool fast) + : DecoderPart (parent) + , _content (content) + , _fast (fast) { - BOOST_FOREACH (AudioStreamPtr i, content->streams ()) { - _streams[i] = shared_ptr (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 data, ContentTime time) +AudioDecoder::emit (shared_ptr film, AudioStreamPtr stream, shared_ptr 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; + 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(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 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 +AudioDecoder::position (shared_ptr film) const +{ + optional 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 >::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 () { - _log->log (String::compose ("AD seek to %1", to_string(t)), LogEntry::TYPE_DEBUG_DECODE); - for (map >::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 >::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(i->channels(), samples); + silence->make_silent (); + Data (i, ContentAudio (silence, _positions[i])); } }