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 (shared_ptr<const AudioContent> content, AudioStreamPtr stream, Decoder* decoder, shared_ptr<Log> log)
49 if (content->resampled_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
50 _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_frame_rate(), _stream->channels ()));
57 AudioDecoderStream::reset_decoded ()
59 _decoded = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_stream->channels(), 0)), 0);
63 AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
65 shared_ptr<ContentAudio> dec;
67 _log->log (String::compose ("-> ADS has request for %1 %2", frame, length), LogEntry::TYPE_DEBUG_DECODE);
69 Frame const end = frame + length - 1;
71 if (frame < _decoded.frame || end > (_decoded.frame + length * 4)) {
72 /* Either we have no decoded data, or what we do have is a long way from what we want: seek */
73 _decoder->seek (ContentTime::from_frames (frame, _content->resampled_frame_rate()), accurate);
76 /* Offset of the data that we want from the start of _decoded.audio
77 (to be set up shortly)
79 Frame decoded_offset = 0;
81 /* Now enough pass() calls will either:
82 * (a) give us what we want, or
83 * (b) hit the end of the decoder.
85 * If we are being accurate, we want the right frames,
86 * otherwise any frames will do.
89 /* Keep stuffing data into _decoded until we have enough data, or the subclass does not want to give us any more */
91 (_decoded.frame > frame || (_decoded.frame + _decoded.audio->frames()) < end) &&
92 !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
96 decoded_offset = frame - _decoded.frame;
99 String::compose ("Accurate ADS::get has offset %1 from request %2 and available %3", decoded_offset, frame, _decoded.frame),
100 LogEntry::TYPE_DEBUG_DECODE
104 _decoded.audio->frames() < length &&
105 !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
109 /* Use decoded_offset of 0, as we don't really care what frames we return */
112 /* The amount of data available in _decoded.audio starting from `frame'. This could be -ve
113 if pass() returned true before we got enough data.
115 Frame const available = _decoded.audio->frames() - decoded_offset;
117 /* We will return either that, or the requested amount, whichever is smaller */
118 Frame const to_return = max ((Frame) 0, min (available, length));
120 /* Copy our data to the output */
121 shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded.audio->channels(), to_return));
122 out->copy_from (_decoded.audio.get(), to_return, decoded_offset, 0);
124 Frame const remaining = max ((Frame) 0, available - to_return);
126 /* Clean up decoded; first, move the data after what we just returned to the start of the buffer */
127 _decoded.audio->move (decoded_offset + to_return, 0, remaining);
128 /* And set up the number of frames we have left */
129 _decoded.audio->set_frames (remaining);
130 /* Also bump where those frames are in terms of the content */
131 _decoded.frame += decoded_offset + to_return;
133 return ContentAudio (out, frame);
136 /** Audio timestamping is made hard by many factors, but perhaps the most entertaining is resampling.
137 * We have to assume that we are feeding continuous data into the resampler, and so we get continuous
138 * data out. Hence we do the timestamping here, post-resampler, just by counting samples.
140 * The time is passed in here so that after a seek we can set up our _position. The
141 * time is ignored once this has been done.
144 AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
146 _log->log (String::compose ("ADS receives %1 %2", time, data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
149 data = _resampler->run (data);
152 Frame const frame_rate = _content->resampled_frame_rate ();
154 if (_seek_reference) {
155 /* We've had an accurate seek and now we're seeing some data */
156 ContentTime const delta = time - _seek_reference.get ();
157 Frame const delta_frames = delta.frames_round (frame_rate);
158 if (delta_frames > 0) {
159 /* This data comes after the seek time. Pad the data with some silence. */
160 shared_ptr<AudioBuffers> padded (new AudioBuffers (data->channels(), data->frames() + delta_frames));
161 padded->make_silent ();
162 padded->copy_from (data.get(), data->frames(), 0, delta_frames);
165 } else if (delta_frames < 0) {
166 /* This data comes before the seek time. Throw some data away */
167 Frame const to_discard = min (-delta_frames, static_cast<Frame> (data->frames()));
168 Frame const to_keep = data->frames() - to_discard;
170 /* We have to throw all this data away, so keep _seek_reference and
171 try again next time some data arrives.
175 shared_ptr<AudioBuffers> trimmed (new AudioBuffers (data->channels(), to_keep));
176 trimmed->copy_from (data.get(), to_keep, to_discard, 0);
178 time += ContentTime::from_frames (to_discard, frame_rate);
180 _seek_reference = optional<ContentTime> ();
184 _position = time.frames_round (frame_rate);
187 DCPOMATIC_ASSERT (_position.get() >= (_decoded.frame + _decoded.audio->frames()));
193 AudioDecoderStream::add (shared_ptr<const AudioBuffers> data)
196 /* This should only happen when there is a seek followed by a flush, but
197 we need to cope with it.
202 /* Resize _decoded to fit the new data */
204 if (_decoded.audio->frames() == 0) {
205 /* There's nothing in there, so just store the new data */
206 new_size = data->frames ();
207 _decoded.frame = _position.get ();
209 /* Otherwise we need to extend _decoded to include the new stuff */
210 new_size = _position.get() + data->frames() - _decoded.frame;
213 _decoded.audio->ensure_size (new_size);
214 _decoded.audio->set_frames (new_size);
216 /* Copy new data in */
217 _decoded.audio->copy_from (data.get(), data->frames(), 0, _position.get() - _decoded.frame);
218 _position = _position.get() + data->frames ();
220 /* Limit the amount of data we keep in case nobody is asking for it */
221 int const max_frames = _content->resampled_frame_rate () * 10;
222 if (_decoded.audio->frames() > max_frames) {
223 int const to_remove = _decoded.audio->frames() - max_frames;
224 _decoded.frame += to_remove;
225 _decoded.audio->move (to_remove, 0, max_frames);
226 _decoded.audio->set_frames (max_frames);
231 AudioDecoderStream::flush ()
237 shared_ptr<const AudioBuffers> b = _resampler->flush ();
244 AudioDecoderStream::seek (ContentTime t, bool accurate)
254 AudioDecoderStream::set_fast ()
257 _resampler->set_fast ();