2 Copyright (C) 2012-2016 Carl Hetherington <cth@carlh.net>
4 This program is free software; you can redistribute it and/or modify
5 it under the terms of the GNU General Public License as published by
6 the Free Software Foundation; either version 2 of the License, or
7 (at your option) any later version.
9 This program is distributed in the hope that it will be useful,
10 but WITHOUT ANY WARRANTY; without even the implied warranty of
11 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
12 GNU General Public License for more details.
14 You should have received a copy of the GNU General Public License
15 along with this program; if not, write to the Free Software
16 Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
20 #include "audio_decoder_stream.h"
21 #include "audio_buffers.h"
22 #include "audio_processor.h"
23 #include "audio_decoder.h"
24 #include "resampler.h"
28 #include "audio_content.h"
29 #include "compose.hpp"
39 using boost::optional;
40 using boost::shared_ptr;
42 AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, Decoder* decoder, bool fast, shared_ptr<Log> log)
48 if (content->resampled_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
49 _resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_frame_rate(), _stream->channels (), fast));
56 AudioDecoderStream::reset_decoded ()
58 _decoded = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_stream->channels(), 0)), 0);
62 AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
64 shared_ptr<ContentAudio> dec;
66 _log->log (String::compose ("-> ADS has request for %1 %2", frame, length), LogEntry::TYPE_DEBUG_DECODE);
68 Frame const end = frame + length - 1;
70 if (frame < _decoded.frame || end > (_decoded.frame + length * 4)) {
71 /* Either we have no decoded data, or what we do have is a long way from what we want: seek */
72 _decoder->seek (ContentTime::from_frames (frame, _content->resampled_frame_rate()), accurate);
75 /* Offset of the data that we want from the start of _decoded.audio
76 (to be set up shortly)
78 Frame decoded_offset = 0;
80 /* Now enough pass() calls will either:
81 * (a) give us what we want, or
82 * (b) hit the end of the decoder.
84 * If we are being accurate, we want the right frames,
85 * otherwise any frames will do.
88 /* Keep stuffing data into _decoded until we have enough data, or the subclass does not want to give us any more */
90 (_decoded.frame > frame || (_decoded.frame + _decoded.audio->frames()) < end) &&
91 !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
95 decoded_offset = frame - _decoded.frame;
98 String::compose ("Accurate ADS::get has offset %1 from request %2 and available %3", decoded_offset, frame, _decoded.frame),
99 LogEntry::TYPE_DEBUG_DECODE
103 _decoded.audio->frames() < length &&
104 !_decoder->pass (Decoder::PASS_REASON_AUDIO, accurate)
108 /* Use decoded_offset of 0, as we don't really care what frames we return */
111 /* The amount of data available in _decoded.audio starting from `frame'. This could be -ve
112 if pass() returned true before we got enough data.
114 Frame const available = _decoded.audio->frames() - decoded_offset;
116 /* We will return either that, or the requested amount, whichever is smaller */
117 Frame const to_return = max ((Frame) 0, min (available, length));
119 /* Copy our data to the output */
120 shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded.audio->channels(), to_return));
121 out->copy_from (_decoded.audio.get(), to_return, decoded_offset, 0);
123 Frame const remaining = max ((Frame) 0, available - to_return);
125 /* Clean up decoded; first, move the data after what we just returned to the start of the buffer */
126 _decoded.audio->move (decoded_offset + to_return, 0, remaining);
127 /* And set up the number of frames we have left */
128 _decoded.audio->set_frames (remaining);
129 /* Also bump where those frames are in terms of the content */
130 _decoded.frame += decoded_offset + to_return;
132 return ContentAudio (out, frame);
135 /** Audio timestamping is made hard by many factors, but perhaps the most entertaining is resampling.
136 * We have to assume that we are feeding continuous data into the resampler, and so we get continuous
137 * data out. Hence we do the timestamping here, post-resampler, just by counting samples.
139 * The time is passed in here so that after a seek we can set up our _position. The
140 * time is ignored once this has been done.
143 AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
145 _log->log (String::compose ("ADS receives %1 %2", time, data->frames ()), LogEntry::TYPE_DEBUG_DECODE);
148 data = _resampler->run (data);
151 Frame const frame_rate = _content->resampled_frame_rate ();
153 if (_seek_reference) {
154 /* We've had an accurate seek and now we're seeing some data */
155 ContentTime const delta = time - _seek_reference.get ();
156 Frame const delta_frames = delta.frames_round (frame_rate);
157 if (delta_frames > 0) {
158 /* This data comes after the seek time. Pad the data with some silence. */
159 shared_ptr<AudioBuffers> padded (new AudioBuffers (data->channels(), data->frames() + delta_frames));
160 padded->make_silent ();
161 padded->copy_from (data.get(), data->frames(), 0, delta_frames);
164 } else if (delta_frames < 0) {
165 /* This data comes before the seek time. Throw some data away */
166 Frame const to_discard = min (-delta_frames, static_cast<Frame> (data->frames()));
167 Frame const to_keep = data->frames() - to_discard;
169 /* We have to throw all this data away, so keep _seek_reference and
170 try again next time some data arrives.
174 shared_ptr<AudioBuffers> trimmed (new AudioBuffers (data->channels(), to_keep));
175 trimmed->copy_from (data.get(), to_keep, to_discard, 0);
177 time += ContentTime::from_frames (to_discard, frame_rate);
179 _seek_reference = optional<ContentTime> ();
183 _position = time.frames_round (frame_rate);
186 DCPOMATIC_ASSERT (_position.get() >= (_decoded.frame + _decoded.audio->frames()));
192 AudioDecoderStream::add (shared_ptr<const AudioBuffers> data)
195 /* This should only happen when there is a seek followed by a flush, but
196 we need to cope with it.
201 /* Resize _decoded to fit the new data */
203 if (_decoded.audio->frames() == 0) {
204 /* There's nothing in there, so just store the new data */
205 new_size = data->frames ();
206 _decoded.frame = _position.get ();
208 /* Otherwise we need to extend _decoded to include the new stuff */
209 new_size = _position.get() + data->frames() - _decoded.frame;
212 _decoded.audio->ensure_size (new_size);
213 _decoded.audio->set_frames (new_size);
215 /* Copy new data in */
216 _decoded.audio->copy_from (data.get(), data->frames(), 0, _position.get() - _decoded.frame);
217 _position = _position.get() + data->frames ();
219 /* Limit the amount of data we keep in case nobody is asking for it */
220 int const max_frames = _content->resampled_frame_rate () * 10;
221 if (_decoded.audio->frames() > max_frames) {
222 int const to_remove = _decoded.audio->frames() - max_frames;
223 _decoded.frame += to_remove;
224 _decoded.audio->move (to_remove, 0, max_frames);
225 _decoded.audio->set_frames (max_frames);
230 AudioDecoderStream::flush ()
236 shared_ptr<const AudioBuffers> b = _resampler->flush ();
243 AudioDecoderStream::seek (ContentTime t, bool accurate)