X-Git-Url: https://git.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=b4aa2bacda401b35d7ca3c5420b210545a9f5450;hb=8414829693900c3d6362a4f15d677bb7e1462c3e;hp=8d3b0e1288ea5b53627a9cfb82619b0ccc125c7e;hpb=4ba8772aef261da209bbb882325fd61a8b479fd7;p=dcpomatic.git diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc index 8d3b0e128..b4aa2bacd 100644 --- a/src/lib/audio_decoder.cc +++ b/src/lib/audio_decoder.cc @@ -1,89 +1,167 @@ /* - Copyright (C) 2012 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 "exceptions.h" +#include "audio_content.h" #include "log.h" #include "resampler.h" -#include "util.h" -#include "film.h" +#include "compose.hpp" +#include +#include #include "i18n.h" -using std::stringstream; -using std::list; -using std::pair; +#define LOG_GENERAL(...) _log->log (String::compose (__VA_ARGS__), LogEntry::TYPE_GENERAL); + using std::cout; -using boost::optional; +using std::map; +using std::pair; using boost::shared_ptr; +using boost::optional; -AudioDecoder::AudioDecoder (shared_ptr film, shared_ptr content) - : Decoder (film) - , _audio_content (content) +AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr content, shared_ptr log, bool fast) + : DecoderPart (parent, log) + , _content (content) + , _fast (fast) { - if (content->output_audio_frame_rate() != content->content_audio_frame_rate() && content->audio_channels ()) { - _resampler.reset (new Resampler (content->content_audio_frame_rate(), content->output_audio_frame_rate(), content->audio_channels ())); + /* Set up _positions so that we have one for each stream */ + BOOST_FOREACH (AudioStreamPtr i, content->streams ()) { + _positions[i] = 0; } } -/** Audio timestamping is made hard by many factors, but the final nail in the coffin is resampling. - * We have to assume that we are feeding continuous data into the resampler, and so we get continuous - * data out. Hence we do the timestamping here, post-resampler, just by counting samples. - * - * The time is passed in here so that after a seek we can set up our _audio_position. The - * time is ignored once this has been done. - */ void -AudioDecoder::audio (shared_ptr data, ContentTime time) +AudioDecoder::emit (AudioStreamPtr stream, shared_ptr data, ContentTime time) { - if (_resampler) { - data = _resampler->run (data); + if (ignore ()) { + return; } - if (!_audio_position) { - shared_ptr film = _film.lock (); - assert (film); - FrameRateChange frc = film->active_frame_rate_change (_audio_content->position ()); - _audio_position = (double (time) / frc.speed_up) * film->audio_frame_rate() / TIME_HZ; + 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, as it seems that ContentTimes are unreliable from + FFmpegDecoder (not quite continuous; perhaps due to some rounding error). + */ + if (_content->delay() > 0) { + /* Insert silence to give the delay */ + silence (_content->delay ()); + } + time += ContentTime::from_seconds (_content->delay() / 1000.0); + _positions[stream] = time.frames_round (_content->resampled_frame_rate ()); } - _pending.push_back (shared_ptr (new DecodedAudio (data, _audio_position.get ()))); - _audio_position = _audio_position.get() + data->frames (); + shared_ptr resampler; + ResamplerMap::iterator i = _resamplers.find(stream); + if (i != _resamplers.end ()) { + resampler = i->second; + } else { + if (stream->frame_rate() != _content->resampled_frame_rate()) { + LOG_GENERAL ( + "Creating new resampler from %1 to %2 with %3 channels", + stream->frame_rate(), + _content->resampled_frame_rate(), + stream->channels() + ); + + resampler.reset (new Resampler (stream->frame_rate(), _content->resampled_frame_rate(), stream->channels())); + if (_fast) { + resampler->set_fast (); + } + _resamplers[stream] = resampler; + } + } + + if (resampler) { + shared_ptr ro = resampler->run (data); + if (ro->frames() == 0) { + return; + } + data = ro; + } + + 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 (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()); +} + +ContentTime +AudioDecoder::position () const +{ + optional p; + for (PositionMap::const_iterator i = _positions.begin(); i != _positions.end(); ++i) { + ContentTime const ct = stream_position (i->first); + if (!p || ct < *p) { + p = ct; + } + } + + return p.get_value_or(ContentTime()); +} + +void +AudioDecoder::seek () +{ + 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::flush () { - if (!_resampler) { - return; + 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(); + } } - shared_ptr b = _resampler->flush (); - if (b) { - _pending.push_back (shared_ptr (new DecodedAudio (b, _audio_position.get ()))); - _audio_position = _audio_position.get() + b->frames (); + if (_content->delay() < 0) { + /* Finish off with the gap caused by the delay */ + silence (-_content->delay ()); } } void -AudioDecoder::seek (ContentTime, bool) +AudioDecoder::silence (int milliseconds) { - _audio_position.reset (); + 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])); + } }