X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=a5e86f22b8e352d529fa39e69149ad10cca9fec4;hb=ce17803bf356f3e796dccde43b4cc3656609e7fc;hp=fbc66c335bf21ec149335e28e029c94e09f03de1;hpb=596441a4e8cf03a88113646ca6da2f90e721a38b;p=dcpomatic.git diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc index fbc66c335..a5e86f22b 100644 --- a/src/lib/audio_decoder.cc +++ b/src/lib/audio_decoder.cc @@ -1,136 +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 "dcpomatic_log.h" #include "log.h" +#include "resampler.h" +#include "compose.hpp" +#include +#include #include "i18n.h" -using std::stringstream; -using boost::optional; +using std::cout; +using std::map; +using std::pair; using boost::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 ()) { + _positions[i] = 0; + } +} -AudioDecoder::AudioDecoder (shared_ptr f, shared_ptr c) - : Decoder (f) - , _next_audio (0) - , _audio_content (c) - , _output_audio_frame_rate (_audio_content->output_audio_frame_rate (f)) +void +AudioDecoder::emit (shared_ptr film, AudioStreamPtr stream, shared_ptr data, ContentTime time) { - if (_audio_content->content_audio_frame_rate() != _output_audio_frame_rate) { - - stringstream s; - s << String::compose ("Will resample audio from %1 to %2", _audio_content->content_audio_frame_rate(), _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. + if (ignore ()) { + return; + } + + 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(film)); + } - _swr_context = swr_alloc_set_opts ( - 0, - av_get_default_channel_layout (MAX_AUDIO_CHANNELS), - AV_SAMPLE_FMT_FLTP, - _output_audio_frame_rate, - av_get_default_channel_layout (MAX_AUDIO_CHANNELS), - AV_SAMPLE_FMT_FLTP, - _audio_content->content_audio_frame_rate(), - 0, 0 - ); - - swr_init (_swr_context); + shared_ptr resampler; + ResamplerMap::iterator i = _resamplers.find(stream); + if (i != _resamplers.end ()) { + resampler = i->second; } else { - _swr_context = 0; + if (stream->frame_rate() != _content->resampled_frame_rate(film)) { + LOG_GENERAL ( + "Creating new resampler from %1 to %2 with %3 channels", + stream->frame_rate(), + _content->resampled_frame_rate(film), + stream->channels() + ); + + resampler.reset (new Resampler (stream->frame_rate(), _content->resampled_frame_rate(film), stream->channels())); + if (_fast) { + resampler->set_fast (); + } + _resamplers[stream] = resampler; + } } -} -AudioDecoder::~AudioDecoder () -{ - if (_swr_context) { - swr_free (&_swr_context); + 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(); } - -#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 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); + PositionMap::const_iterator i = _positions.find (stream); + DCPOMATIC_ASSERT (i != _positions.end ()); + return ContentTime::from_frames (i->second, _content->resampled_frame_rate(film)); +} - if (frames < 0) { - throw EncodeError (_("could not run sample-rate converter")); - } +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; + } + } - if (frames == 0) { - break; - } + return p; +} - out->set_frames (frames); - _writer->write (out); - } +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; } } -#endif void -AudioDecoder::emit_audio (shared_ptr data, Time time) +AudioDecoder::flush () { - /* XXX: map audio to 5.1 */ - - /* Maybe resample */ - if (_swr_context) { - - /* Compute the resampled frames count and add 32 for luck */ - int const max_resampled_frames = ceil ((int64_t) data->frames() * _output_audio_frame_rate / _audio_content->content_audio_frame_rate()) + 32; - - shared_ptr resampled (new AudioBuffers (MAX_AUDIO_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")); + 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(); } - - resampled->set_frames (resampled_frames); - - /* And point our variables at the resampled audio */ - data = resampled; } - Audio (data, time); - - _next_audio = time + _film->audio_frames_to_time (data->frames()); + if (_content->delay() < 0) { + /* Finish off with the gap caused by the delay */ + silence (-_content->delay ()); + } } - +void +AudioDecoder::silence (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])); + } +}