X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=5334dfa345071e887fa6e9f7a04d34701d23054b;hb=dd9be86db6cde0afa5da0d1d1ac43b42e05dca26;hp=7ceb9680bf95b3be487ed6d3dd42e7ce4ece58b3;hpb=504c63b3d62038bc486ca8a09e77fbb403907edd;p=dcpomatic.git diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc index 7ceb9680b..5334dfa34 100644 --- a/src/lib/audio_decoder.cc +++ b/src/lib/audio_decoder.cc @@ -1,26 +1,30 @@ /* - Copyright (C) 2012-2016 Carl Hetherington + Copyright (C) 2012-2018 Carl Hetherington - This program is free software; you can redistribute it and/or modify + This file is part of DCP-o-matic. + + DCP-o-matic is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. - This program is distributed in the hope that it will be useful, + DCP-o-matic is distributed in the hope that it will be useful, but WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for more details. You should have received a copy of the GNU General Public License - along with this program; if not, write to the Free Software - Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + along with DCP-o-matic. If not, see . */ #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 @@ -28,76 +32,162 @@ using std::cout; using std::map; -using boost::shared_ptr; - -AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr content, bool fast, shared_ptr log) - : _audio_content (content) - , _ignore_audio (false) +using std::pair; +using std::shared_ptr; +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 ()) { - _streams[i] = shared_ptr (new AudioDecoderStream (_audio_content, i, parent, log)); + _positions[i] = 0; } } -ContentAudio -AudioDecoder::get_audio (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::audio (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_audio) { + 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. + 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; + 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; + ResamplerMap::iterator 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.reset (new Resampler(stream->frame_rate(), resampled_rate, stream->channels())); + if (_fast) { + resampler->set_fast (); + } + _resamplers[stream] = resampler; + } + } - return; + if (resampler) { + shared_ptr 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 +{ + PositionMap::const_iterator 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); + if (!p || ct < *p) { + p = ct; + } + } + + return p; } void -AudioDecoder::flush () +AudioDecoder::seek () { - for (map >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) { + for (ResamplerMap::iterator i = _resamplers.begin(); i != _resamplers.end(); ++i) { i->second->flush (); + i->second->reset (); + } + + for (PositionMap::iterator i = _positions.begin(); i != _positions.end(); ++i) { + i->second = 0; } } void -AudioDecoder::seek (ContentTime t, bool accurate) +AudioDecoder::flush () { - for (map >::const_iterator i = _streams.begin(); i != _streams.end(); ++i) { - i->second->seek (t, accurate); + for (ResamplerMap::iterator i = _resamplers.begin(); i != _resamplers.end(); ++i) { + shared_ptr 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 ()); } } -/** Set this player never to produce any audio data */ void -AudioDecoder::set_ignore_audio () +AudioDecoder::silence (int milliseconds) { - _ignore_audio = true; + 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])); + } }