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);
70 AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
72 shared_ptr<ContentAudio> dec;
74 _log->log (String::compose ("-> ADS has request for %1 %2", frame, length), LogEntry::TYPE_DEBUG_DECODE);
76 Frame const from = frame;
77 Frame const to = from + length;
78 Frame const have_from = _decoded.frame;
79 Frame const have_to = _decoded.frame + _decoded.audio->frames();
81 optional<Frame> missing;
82 if (have_from > from || have_to < to) {
83 /* We need something */
84 if (have_from < from && from < have_to) {
92 _audio_decoder->maybe_seek (ContentTime::from_frames (*missing, _content->resampled_frame_rate()), accurate);
95 /* Offset of the data that we want from the start of _decoded.audio
96 (to be set up shortly)
98 Frame decoded_offset = 0;
100 /* Now enough pass() calls will either:
101 * (a) give us what we want, or
102 * (b) hit the end of the decoder.
104 * If we are being accurate, we want the right frames,
105 * otherwise any frames will do.
108 /* Keep stuffing data into _decoded until we have enough data, or the subclass does not want to give us any more */
110 (_decoded.frame > frame || (_decoded.frame + _decoded.audio->frames()) <= to) &&
111 !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
115 decoded_offset = frame - _decoded.frame;
118 String::compose ("Accurate ADS::get has offset %1 from request %2 and available %3", decoded_offset, frame, have_from),
119 LogEntry::TYPE_DEBUG_DECODE
123 _decoded.audio->frames() < length &&
124 !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
128 /* Use decoded_offset of 0, as we don't really care what frames we return */
131 /* The amount of data available in _decoded.audio starting from `frame'. This could be -ve
132 if pass() returned true before we got enough data.
134 Frame const available = _decoded.audio->frames() - decoded_offset;
136 /* We will return either that, or the requested amount, whichever is smaller */
137 Frame const to_return = max ((Frame) 0, min (available, length));
139 /* Copy our data to the output */
140 shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded.audio->channels(), to_return));
141 out->copy_from (_decoded.audio.get(), to_return, decoded_offset, 0);
143 Frame const remaining = max ((Frame) 0, available - to_return);
145 /* Clean up decoded; first, move the data after what we just returned to the start of the buffer */
146 _decoded.audio->move (decoded_offset + to_return, 0, remaining);
147 /* And set up the number of frames we have left */
148 _decoded.audio->set_frames (remaining);
149 /* Also bump where those frames are in terms of the content */
150 _decoded.frame += decoded_offset + to_return;
152 return ContentAudio (out, frame);
155 /** Audio timestamping is made hard by many factors, but perhaps the most entertaining is resampling.
156 * We have to assume that we are feeding continuous data into the resampler, and so we get continuous
157 * data out. Hence we do the timestamping here, post-resampler, just by counting samples.
159 * The time is passed in here so that after a seek we can set up our _position. The
160 * time is ignored once this has been done.
163 AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
165 _log->log (String::compose ("ADS receives %1 %2", to_string(time), data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
168 data = _resampler->run (data);
171 Frame const frame_rate = _content->resampled_frame_rate ();
173 if (_seek_reference) {
174 /* We've had an accurate seek and now we're seeing some data */
175 ContentTime const delta = time - _seek_reference.get ();
176 Frame const delta_frames = delta.frames_round (frame_rate);
177 if (delta_frames > 0) {
178 /* This data comes after the seek time. Pad the data with some silence. */
179 shared_ptr<AudioBuffers> padded (new AudioBuffers (data->channels(), data->frames() + delta_frames));
180 padded->make_silent ();
181 padded->copy_from (data.get(), data->frames(), 0, delta_frames);
185 _seek_reference = optional<ContentTime> ();
189 _position = time.frames_round (frame_rate);
192 DCPOMATIC_ASSERT (_position.get() >= (_decoded.frame + _decoded.audio->frames()));
198 AudioDecoderStream::add (shared_ptr<const AudioBuffers> data)
201 /* This should only happen when there is a seek followed by a flush, but
202 we need to cope with it.
207 /* Resize _decoded to fit the new data */
209 if (_decoded.audio->frames() == 0) {
210 /* There's nothing in there, so just store the new data */
211 new_size = data->frames ();
212 _decoded.frame = _position.get ();
214 /* Otherwise we need to extend _decoded to include the new stuff */
215 new_size = _position.get() + data->frames() - _decoded.frame;
218 _decoded.audio->ensure_size (new_size);
219 _decoded.audio->set_frames (new_size);
221 /* Copy new data in */
222 _decoded.audio->copy_from (data.get(), data->frames(), 0, _position.get() - _decoded.frame);
223 _position = _position.get() + data->frames ();
225 /* Limit the amount of data we keep in case nobody is asking for it */
226 int const max_frames = _content->resampled_frame_rate () * 10;
227 if (_decoded.audio->frames() > max_frames) {
228 int const to_remove = _decoded.audio->frames() - max_frames;
229 _decoded.frame += to_remove;
230 _decoded.audio->move (to_remove, 0, max_frames);
231 _decoded.audio->set_frames (max_frames);
236 AudioDecoderStream::flush ()
242 shared_ptr<const AudioBuffers> b = _resampler->flush ();
249 AudioDecoderStream::seek (ContentTime t, bool accurate)
259 AudioDecoderStream::set_fast ()
262 _resampler->set_fast ();