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"
31 #include <boost/make_shared.hpp>
41 using boost::optional;
42 using boost::shared_ptr;
43 using boost::make_shared;
45 AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, Decoder* decoder, bool fast, shared_ptr<Log> log)
51 if (content->resampled_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
52 _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_frame_rate(), _stream->channels (), fast));
59 AudioDecoderStream::reset_decoded ()
61 _decoded = ContentAudio (make_shared<AudioBuffers> (_stream->channels(), 0), 0);
65 AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
67 shared_ptr<ContentAudio> dec;
69 _log->log (String::compose ("-> ADS has request for %1 %2", frame, length), LogEntry::TYPE_DEBUG_DECODE);
71 Frame const end = frame + length - 1;
73 if (frame < _decoded.frame || end > (_decoded.frame + length * 4)) {
74 /* Either we have no decoded data, or what we do have is a long way from what we want: seek */
75 _decoder->seek (ContentTime::from_frames (frame, _content->resampled_frame_rate()), accurate);
78 /* Offset of the data that we want from the start of _decoded.audio
79 (to be set up shortly)
81 Frame decoded_offset = 0;
83 /* Now enough pass() calls will either:
84 * (a) give us what we want, or
85 * (b) hit the end of the decoder.
87 * If we are being accurate, we want the right frames,
88 * otherwise any frames will do.
91 /* Keep stuffing data into _decoded until we have enough data, or the subclass does not want to give us any more */
93 (_decoded.frame > frame || (_decoded.frame + _decoded.audio->frames()) < end) &&
94 !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
98 decoded_offset = frame - _decoded.frame;
101 String::compose ("Accurate ADS::get has offset %1 from request %2 and available %3", decoded_offset, frame, _decoded.frame),
102 LogEntry::TYPE_DEBUG_DECODE
106 _decoded.audio->frames() < length &&
107 !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
111 /* Use decoded_offset of 0, as we don't really care what frames we return */
114 /* The amount of data available in _decoded.audio starting from `frame'. This could be -ve
115 if pass() returned true before we got enough data.
117 Frame const available = _decoded.audio->frames() - decoded_offset;
119 /* We will return either that, or the requested amount, whichever is smaller */
120 Frame const to_return = max ((Frame) 0, min (available, length));
122 /* Copy our data to the output */
123 shared_ptr<AudioBuffers> out = make_shared<AudioBuffers> (_decoded.audio->channels(), to_return);
124 out->copy_from (_decoded.audio.get(), to_return, decoded_offset, 0);
126 Frame const remaining = max ((Frame) 0, available - to_return);
128 /* Clean up decoded; first, move the data after what we just returned to the start of the buffer */
129 _decoded.audio->move (decoded_offset + to_return, 0, remaining);
130 /* And set up the number of frames we have left */
131 _decoded.audio->set_frames (remaining);
132 /* Also bump where those frames are in terms of the content */
133 _decoded.frame += decoded_offset + to_return;
135 return ContentAudio (out, frame);
138 /** Audio timestamping is made hard by many factors, but perhaps the most entertaining is resampling.
139 * We have to assume that we are feeding continuous data into the resampler, and so we get continuous
140 * data out. Hence we do the timestamping here, post-resampler, just by counting samples.
142 * The time is passed in here so that after a seek we can set up our _position. The
143 * time is ignored once this has been done.
146 AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
148 _log->log (String::compose ("ADS receives %1 %2", time, data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
151 data = _resampler->run (data);
154 Frame const frame_rate = _content->resampled_frame_rate ();
156 if (_seek_reference) {
157 /* We've had an accurate seek and now we're seeing some data */
158 ContentTime const delta = time - _seek_reference.get ();
159 Frame const delta_frames = delta.frames_round (frame_rate);
160 if (delta_frames > 0) {
161 /* This data comes after the seek time. Pad the data with some silence. */
162 shared_ptr<AudioBuffers> padded = make_shared<AudioBuffers> (data->channels(), data->frames() + delta_frames);
163 padded->make_silent ();
164 padded->copy_from (data.get(), data->frames(), 0, delta_frames);
167 } else if (delta_frames < 0) {
168 /* This data comes before the seek time. Throw some data away */
169 Frame const to_discard = min (-delta_frames, static_cast<Frame> (data->frames()));
170 Frame const to_keep = data->frames() - to_discard;
172 /* We have to throw all this data away, so keep _seek_reference and
173 try again next time some data arrives.
177 shared_ptr<AudioBuffers> trimmed = make_shared<AudioBuffers> (data->channels(), to_keep);
178 trimmed->copy_from (data.get(), to_keep, to_discard, 0);
180 time += ContentTime::from_frames (to_discard, frame_rate);
182 _seek_reference = optional<ContentTime> ();
186 _position = time.frames_round (frame_rate);
189 DCPOMATIC_ASSERT (_position.get() >= (_decoded.frame + _decoded.audio->frames()));
195 AudioDecoderStream::add (shared_ptr<const AudioBuffers> data)
198 /* This should only happen when there is a seek followed by a flush, but
199 we need to cope with it.
204 /* Resize _decoded to fit the new data */
206 if (_decoded.audio->frames() == 0) {
207 /* There's nothing in there, so just store the new data */
208 new_size = data->frames ();
209 _decoded.frame = _position.get ();
211 /* Otherwise we need to extend _decoded to include the new stuff */
212 new_size = _position.get() + data->frames() - _decoded.frame;
215 _decoded.audio->ensure_size (new_size);
216 _decoded.audio->set_frames (new_size);
218 /* Copy new data in */
219 _decoded.audio->copy_from (data.get(), data->frames(), 0, _position.get() - _decoded.frame);
220 _position = _position.get() + data->frames ();
222 /* Limit the amount of data we keep in case nobody is asking for it */
223 int const max_frames = _content->resampled_frame_rate () * 10;
224 if (_decoded.audio->frames() > max_frames) {
225 int const to_remove = _decoded.audio->frames() - max_frames;
226 _decoded.frame += to_remove;
227 _decoded.audio->move (to_remove, 0, max_frames);
228 _decoded.audio->set_frames (max_frames);
233 AudioDecoderStream::flush ()
239 shared_ptr<const AudioBuffers> b = _resampler->flush ();
246 AudioDecoderStream::seek (ContentTime t, bool accurate)