X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=2d02043b5948d1071d0de18982e14abfcf03c12e;hb=ac34066d5e448d1984d11a180be74e31b6e13b5c;hp=bbd4ced6c1a2c114f337729f8fe609f71e1eb353;hpb=883d885dc8690519d205c8baa275385af8a39f4b;p=dcpomatic.git diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc index bbd4ced6c..2d02043b5 100644 --- a/src/lib/audio_decoder.cc +++ b/src/lib/audio_decoder.cc @@ -1,167 +1,200 @@ /* - Copyright (C) 2012 Carl Hetherington + Copyright (C) 2012-2021 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 "dcpomatic_log.h" #include "log.h" +#include "resampler.h" +#include "compose.hpp" +#include #include "i18n.h" -using std::stringstream; -using std::list; -using std::pair; + using std::cout; +using std::shared_ptr; +using std::make_shared; using boost::optional; -using boost::shared_ptr; +using namespace dcpomatic; -AudioDecoder::AudioDecoder (shared_ptr f, shared_ptr c) - : Decoder (f) - , _next_audio (0) - , _audio_content (c) + +AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr content, bool fast) + : DecoderPart (parent) + , _content (content) + , _fast (fast) { - if (_audio_content->content_audio_frame_rate() != _audio_content->output_audio_frame_rate()) { + /* Set up _positions so that we have one for each stream */ + for (auto i: content->streams ()) { + _positions[i] = 0; + } +} - shared_ptr film = _film.lock (); - assert (film); - stringstream s; - s << String::compose ( - "Will resample audio from %1 to %2", - _audio_content->content_audio_frame_rate(), _audio_content->output_audio_frame_rate() - ); - - film->log()->log (s.str ()); - - /* We will be using planar float data when we call the - resampler. As far as I can see, the audio channel - layout is not necessary for our purposes; it seems - only to be used get the number of channels and - decide if rematrixing is needed. It won't be, since - input and output layouts are the same. - */ +/** @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) +{ + if (ignore ()) { + 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; - _swr_context = swr_alloc_set_opts ( - 0, - av_get_default_channel_layout (MAX_AUDIO_CHANNELS), - AV_SAMPLE_FMT_FLTP, - _audio_content->output_audio_frame_rate(), - av_get_default_channel_layout (MAX_AUDIO_CHANNELS), - AV_SAMPLE_FMT_FLTP, - _audio_content->content_audio_frame_rate(), - 0, 0 + int const resampled_rate = _content->resampled_frame_rate(film); + if (!time_already_delayed) { + time += ContentTime::from_seconds (_content->delay() / 1000.0); + } + + 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)) ); - - swr_init (_swr_context); + } + + if (reset) { + _positions[stream] = time.frames_round (resampled_rate); + } + + shared_ptr resampler; + auto i = _resamplers.find(stream); + if (i != _resamplers.end()) { + resampler = i->second; } else { - _swr_context = 0; + 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; + } } -} -AudioDecoder::~AudioDecoder () -{ - if (_swr_context) { - swr_free (&_swr_context); + if (resampler) { + auto ro = resampler->run (data); + if (ro->frames() == 0) { + return; + } + data = ro; } + + Data(stream, ContentAudio (data, _positions[stream])); + _positions[stream] += data->frames(); } - -#if 0 -void -AudioDecoder::process_end () + +/** @return Time just after the last thing that was emitted from a given stream */ +ContentTime +AudioDecoder::stream_position (shared_ptr film, AudioStreamPtr stream) const { - if (_swr_context) { - - shared_ptr film = _film.lock (); - assert (film); - - shared_ptr out (new AudioBuffers (film->audio_mapping().dcp_channels(), 256)); - - while (1) { - int const frames = swr_convert (_swr_context, (uint8_t **) out->data(), 256, 0, 0); - - if (frames < 0) { - throw EncodeError (_("could not run sample-rate converter")); - } + auto i = _positions.find (stream); + DCPOMATIC_ASSERT (i != _positions.end ()); + return ContentTime::from_frames (i->second, _content->resampled_frame_rate(film)); +} - if (frames == 0) { - break; - } - out->set_frames (frames); - _writer->write (out); +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; } -#endif + void -AudioDecoder::audio (shared_ptr data, Time time) +AudioDecoder::seek () { - /* Maybe resample */ - if (_swr_context) { + for (auto i: _resamplers) { + i.second->flush (); + i.second->reset (); + } - /* Compute the resampled frames count and add 32 for luck */ - int const max_resampled_frames = ceil ( - (int64_t) data->frames() * _audio_content->output_audio_frame_rate() / _audio_content->content_audio_frame_rate() - ) + 32; + for (auto& i: _positions) { + i.second = 0; + } +} - shared_ptr resampled (new AudioBuffers (data->channels(), max_resampled_frames)); - /* Resample audio */ - int const resampled_frames = swr_convert ( - _swr_context, (uint8_t **) resampled->data(), max_resampled_frames, (uint8_t const **) data->data(), data->frames() - ); - - if (resampled_frames < 0) { - throw EncodeError (_("could not run sample-rate converter")); +void +AudioDecoder::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(); } - - resampled->set_frames (resampled_frames); - - /* And point our variables at the resampled audio */ - data = resampled; } - shared_ptr film = _film.lock (); - assert (film); - - /* Remap channels */ - shared_ptr dcp_mapped (new AudioBuffers (film->dcp_audio_channels(), data->frames())); - dcp_mapped->make_silent (); - list > map = _audio_content->audio_mapping().content_to_dcp (); - for (list >::iterator i = map.begin(); i != map.end(); ++i) { - dcp_mapped->accumulate_channel (data.get(), i->first, i->second); + if (_content->delay() < 0) { + /* Finish off with the gap caused by the delay */ + silence (-_content->delay ()); } - - Audio (dcp_mapped, time); - cout << "bumping n.a. by " << data->frames() << " ie " << film->audio_frames_to_time(data->frames()) << "\n"; - _next_audio = time + film->audio_frames_to_time (data->frames()); } -bool -AudioDecoder::audio_done () const + +void +AudioDecoder::silence (int milliseconds) { - shared_ptr film = _film.lock (); - assert (film); - - return (_audio_content->length() - _next_audio) < film->audio_frames_to_time (1); + 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])); + } } -