X-Git-Url: https://main.carlh.net/gitweb/?a=blobdiff_plain;f=src%2Flib%2Faudio_decoder.cc;h=31f713fda3a3b743ec42b90133563af4d1192b38;hb=58dce923b9d438a27ce1cd7e3125370f74d46e3a;hp=2c0388fc39318851242b96a7a672b014ba5fca27;hpb=f1d30fb114b3b2c6ccd8fdf5823e7cd6b26c1eef;p=dcpomatic.git diff --git a/src/lib/audio_decoder.cc b/src/lib/audio_decoder.cc index 2c0388fc3..31f713fda 100644 --- a/src/lib/audio_decoder.cc +++ b/src/lib/audio_decoder.cc @@ -1,175 +1,79 @@ /* - Copyright (C) 2012-2014 Carl Hetherington + Copyright (C) 2012-2016 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; using std::cout; -using std::min; -using std::max; -using boost::optional; +using std::map; using boost::shared_ptr; +using boost::optional; -AudioDecoder::AudioDecoder (shared_ptr content) - : _audio_content (content) +AudioDecoder::AudioDecoder (Decoder* parent, shared_ptr content, shared_ptr log) + : DecoderPart (parent, log) { - if (content->resampled_audio_frame_rate() != content->audio_frame_rate() && content->audio_channels ()) { - _resampler.reset (new Resampler (content->audio_frame_rate(), content->resampled_audio_frame_rate(), content->audio_channels ())); + BOOST_FOREACH (AudioStreamPtr i, content->streams ()) { + _positions[i] = 0; } - - reset_decoded_audio (); } void -AudioDecoder::reset_decoded_audio () -{ - _decoded_audio = ContentAudio (shared_ptr (new AudioBuffers (_audio_content->audio_channels(), 0)), 0); -} - -shared_ptr -AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate) +AudioDecoder::emit (AudioStreamPtr stream, shared_ptr data, ContentTime time) { - shared_ptr dec; - - AudioFrame const end = frame + length - 1; - - if (frame < _decoded_audio.frame || end > (_decoded_audio.frame + length * 4)) { - /* Either we have no decoded data, or what we do have is a long way from what we want: seek */ - seek (ContentTime::from_frames (frame, _audio_content->audio_frame_rate()), accurate); + if (ignore ()) { + return; } - /* Offset of the data that we want from the start of _decoded_audio.audio - (to be set up shortly) - */ - AudioFrame decoded_offset = 0; - - /* Now enough pass() calls will either: - * (a) give us what we want, or - * (b) hit the end of the decoder. - * - * If we are being accurate, we want the right frames, - * otherwise any frames will do. - */ - if (accurate) { - /* Keep stuffing data into _decoded_audio until we have enough data, or the subclass does not want to give us any more */ - while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {} - decoded_offset = frame - _decoded_audio.frame; - } else { - while (!pass() && _decoded_audio.audio->frames() < length) {} - /* Use decoded_offset of 0, as we don't really care what frames we return */ + if (_positions[stream] == 0) { + _positions[stream] = time.frames_round (stream->frame_rate ()); } - /* The amount of data available in _decoded_audio.audio starting from `frame'. This could be -ve - if pass() returned true before we got enough data. - */ - AudioFrame const available = _decoded_audio.audio->frames() - decoded_offset; - - /* We will return either that, or the requested amount, whichever is smaller */ - AudioFrame const to_return = max ((AudioFrame) 0, min (available, length)); - - /* Copy our data to the output */ - shared_ptr out (new AudioBuffers (_decoded_audio.audio->channels(), to_return)); - out->copy_from (_decoded_audio.audio.get(), to_return, decoded_offset, 0); - - AudioFrame const remaining = max ((AudioFrame) 0, available - to_return); - - /* Clean up decoded; first, move the data after what we just returned to the start of the buffer */ - _decoded_audio.audio->move (decoded_offset + to_return, 0, remaining); - /* And set up the number of frames we have left */ - _decoded_audio.audio->set_frames (remaining); - /* Also bump where those frames are in terms of the content */ - _decoded_audio.frame += decoded_offset + to_return; - - return shared_ptr (new ContentAudio (out, frame)); + Data (stream, ContentAudio (data, _positions[stream])); + _positions[stream] += data->frames(); } -/** Called by subclasses when audio data is ready. - * - * Audio timestamping is made hard by many factors, but perhaps the most entertaining 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) +ContentTime +AudioDecoder::position () const { - if (_resampler) { - data = _resampler->run (data); - } - - if (!_audio_position) { - _audio_position = time.frames (_audio_content->resampled_audio_frame_rate ()); + optional p; + for (map::const_iterator i = _positions.begin(); i != _positions.end(); ++i) { + ContentTime const ct = ContentTime::from_frames (i->second, i->first->frame_rate ()); + if (!p || ct < *p) { + p = ct; + } } - assert (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames())); - - /* Resize _decoded_audio to fit the new data */ - int new_size = 0; - if (_decoded_audio.audio->frames() == 0) { - /* There's nothing in there, so just store the new data */ - new_size = data->frames (); - _decoded_audio.frame = _audio_position.get (); - } else { - /* Otherwise we need to extend _decoded_audio to include the new stuff */ - new_size = _audio_position.get() + data->frames() - _decoded_audio.frame; - } - - _decoded_audio.audio->ensure_size (new_size); - _decoded_audio.audio->set_frames (new_size); - - /* Copy new data in */ - _decoded_audio.audio->copy_from (data.get(), data->frames(), 0, _audio_position.get() - _decoded_audio.frame); - _audio_position = _audio_position.get() + data->frames (); + return p.get_value_or(ContentTime()); } -/* XXX: called? */ void -AudioDecoder::flush () +AudioDecoder::seek () { - if (!_resampler) { - return; + for (map::iterator i = _positions.begin(); i != _positions.end(); ++i) { + i->second = 0; } - - /* - 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 (); - } - */ -} - -void -AudioDecoder::seek (ContentTime, bool) -{ - _audio_position.reset (); - reset_decoded_audio (); }