2 Copyright (C) 2012-2016 Carl Hetherington <cth@carlh.net>
4 This file is part of DCP-o-matic.
6 DCP-o-matic is free software; you can redistribute it and/or modify
7 it under the terms of the GNU General Public License as published by
8 the Free Software Foundation; either version 2 of the License, or
9 (at your option) any later version.
11 DCP-o-matic is distributed in the hope that it will be useful,
12 but WITHOUT ANY WARRANTY; without even the implied warranty of
13 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14 GNU General Public License for more details.
16 You should have received a copy of the GNU General Public License
17 along with DCP-o-matic. If not, see <http://www.gnu.org/licenses/>.
21 #include "audio_decoder_stream.h"
22 #include "audio_buffers.h"
23 #include "audio_processor.h"
24 #include "audio_decoder.h"
25 #include "resampler.h"
29 #include "audio_content.h"
30 #include "compose.hpp"
40 using boost::optional;
41 using boost::shared_ptr;
43 AudioDecoderStream::AudioDecoderStream (
44 shared_ptr<const AudioContent> content, AudioStreamPtr stream, Decoder* decoder, AudioDecoder* audio_decoder, shared_ptr<Log> log
49 , _audio_decoder (audio_decoder)
51 /* We effectively start having done a seek to zero; this allows silence-padding of the first
52 data that comes out of our decoder.
54 , _seek_reference (ContentTime ())
56 if (content->resampled_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
57 _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_frame_rate(), _stream->channels ()));
64 AudioDecoderStream::reset_decoded ()
66 _decoded = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_stream->channels(), 0)), 0);
69 /** Audio timestamping is made hard by many factors, but perhaps the most entertaining is resampling.
70 * We have to assume that we are feeding continuous data into the resampler, and so we get continuous
71 * data out. Hence we do the timestamping here, post-resampler, just by counting samples.
73 * The time is passed in here so that after a seek we can set up our _position. The
74 * time is ignored once this has been done.
77 AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
79 _log->log (String::compose ("ADS receives %1 %2", to_string(time), data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
82 data = _resampler->run (data);
85 Frame const frame_rate = _content->resampled_frame_rate ();
87 if (_seek_reference) {
88 /* We've had an accurate seek and now we're seeing some data */
89 ContentTime const delta = time - _seek_reference.get ();
90 Frame const delta_frames = delta.frames_round (frame_rate);
91 if (delta_frames > 0) {
92 /* This data comes after the seek time. Pad the data with some silence. */
93 shared_ptr<AudioBuffers> padded (new AudioBuffers (data->channels(), data->frames() + delta_frames));
94 padded->make_silent ();
95 padded->copy_from (data.get(), data->frames(), 0, delta_frames);
99 _seek_reference = optional<ContentTime> ();
103 _position = time.frames_round (frame_rate);
106 DCPOMATIC_ASSERT (_position.get() >= (_decoded.frame + _decoded.audio->frames()));
112 AudioDecoderStream::add (shared_ptr<const AudioBuffers> data)
115 /* This should only happen when there is a seek followed by a flush, but
116 we need to cope with it.
121 /* Resize _decoded to fit the new data */
123 if (_decoded.audio->frames() == 0) {
124 /* There's nothing in there, so just store the new data */
125 new_size = data->frames ();
126 _decoded.frame = _position.get ();
128 /* Otherwise we need to extend _decoded to include the new stuff */
129 new_size = _position.get() + data->frames() - _decoded.frame;
132 _decoded.audio->ensure_size (new_size);
133 _decoded.audio->set_frames (new_size);
135 /* Copy new data in */
136 _decoded.audio->copy_from (data.get(), data->frames(), 0, _position.get() - _decoded.frame);
137 _position = _position.get() + data->frames ();
139 /* Limit the amount of data we keep in case nobody is asking for it */
140 int const max_frames = _content->resampled_frame_rate () * 10;
141 if (_decoded.audio->frames() > max_frames) {
142 int const to_remove = _decoded.audio->frames() - max_frames;
143 _decoded.frame += to_remove;
144 _decoded.audio->move (to_remove, 0, max_frames);
145 _decoded.audio->set_frames (max_frames);
150 AudioDecoderStream::flush ()
156 shared_ptr<const AudioBuffers> b = _resampler->flush ();
163 AudioDecoderStream::set_fast ()
166 _resampler->set_fast ();
170 optional<ContentTime>
171 AudioDecoderStream::position () const
174 return optional<ContentTime> ();
177 return ContentTime::from_frames (_position.get(), _content->resampled_frame_rate());