1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
|
/*
Copyright (C) 2012-2015 Carl Hetherington <cth@carlh.net>
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*/
#include "audio_decoder_stream.h"
#include "audio_buffers.h"
#include "audio_processor.h"
#include "audio_decoder.h"
#include "resampler.h"
#include "util.h"
#include "film.h"
#include "log.h"
#include "audio_content.h"
#include "compose.hpp"
#include <iostream>
#include "i18n.h"
using std::list;
using std::pair;
using std::cout;
using std::min;
using std::max;
using boost::optional;
using boost::shared_ptr;
AudioDecoderStream::AudioDecoderStream (shared_ptr<const AudioContent> content, AudioStreamPtr stream, AudioDecoder* decoder)
: _content (content)
, _stream (stream)
, _decoder (decoder)
{
if (content->resampled_audio_frame_rate() != _stream->frame_rate() && _stream->channels() > 0) {
_resampler.reset (new Resampler (_stream->frame_rate(), content->resampled_audio_frame_rate(), _stream->channels (), decoder->fast ()));
}
reset_decoded ();
}
void
AudioDecoderStream::reset_decoded ()
{
_decoded = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_stream->channels(), 0)), 0);
}
ContentAudio
AudioDecoderStream::get (Frame frame, Frame length, bool accurate)
{
shared_ptr<ContentAudio> dec;
_content->film()->log()->log (String::compose ("-> ADS has request for %1 %2", frame, length), Log::TYPE_DEBUG_DECODE);
Frame const end = frame + length - 1;
if (frame < _decoded.frame || end > (_decoded.frame + length * 4)) {
/* Either we have no decoded data, or what we do have is a long way from what we want: seek */
seek (ContentTime::from_frames (frame, _content->resampled_audio_frame_rate()), accurate);
}
/* Offset of the data that we want from the start of _decoded.audio
(to be set up shortly)
*/
Frame decoded_offset = 0;
/* Now enough pass() calls will either:
* (a) give us what we want, or
* (b) hit the end of the decoder.
*
* If we are being accurate, we want the right frames,
* otherwise any frames will do.
*/
if (accurate) {
/* Keep stuffing data into _decoded until we have enough data, or the subclass does not want to give us any more */
while (
(_decoded.frame > frame || (_decoded.frame + _decoded.audio->frames()) < end) &&
!_decoder->pass ()
)
{}
decoded_offset = frame - _decoded.frame;
} else {
while (
_decoded.audio->frames() < length &&
!_decoder->pass ()
)
{}
/* Use decoded_offset of 0, as we don't really care what frames we return */
}
/* The amount of data available in _decoded.audio starting from `frame'. This could be -ve
if pass() returned true before we got enough data.
*/
Frame const available = _decoded.audio->frames() - decoded_offset;
/* We will return either that, or the requested amount, whichever is smaller */
Frame const to_return = max ((Frame) 0, min (available, length));
/* Copy our data to the output */
shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded.audio->channels(), to_return));
out->copy_from (_decoded.audio.get(), to_return, decoded_offset, 0);
Frame const remaining = max ((Frame) 0, available - to_return);
/* Clean up decoded; first, move the data after what we just returned to the start of the buffer */
_decoded.audio->move (decoded_offset + to_return, 0, remaining);
/* And set up the number of frames we have left */
_decoded.audio->set_frames (remaining);
/* Also bump where those frames are in terms of the content */
_decoded.frame += decoded_offset + to_return;
return ContentAudio (out, frame);
}
/** Audio timestamping is made hard by many factors, but perhaps the most entertaining is resampling.
* We have to assume that we are feeding continuous data into the resampler, and so we get continuous
* data out. Hence we do the timestamping here, post-resampler, just by counting samples.
*
* The time is passed in here so that after a seek we can set up our _position. The
* time is ignored once this has been done.
*/
void
AudioDecoderStream::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
{
_content->film()->log()->log (String::compose ("ADS receives %1 %2", time, data->frames ()), Log::TYPE_DEBUG_DECODE);
if (_resampler) {
data = _resampler->run (data);
}
Frame const frame_rate = _content->resampled_audio_frame_rate ();
if (_seek_reference) {
/* We've had an accurate seek and now we're seeing some data */
ContentTime const delta = time - _seek_reference.get ();
Frame const delta_frames = delta.frames_round (frame_rate);
if (delta_frames > 0) {
/* This data comes after the seek time. Pad the data with some silence. */
shared_ptr<AudioBuffers> padded (new AudioBuffers (data->channels(), data->frames() + delta_frames));
padded->make_silent ();
padded->copy_from (data.get(), data->frames(), 0, delta_frames);
data = padded;
time -= delta;
} else if (delta_frames < 0) {
/* This data comes before the seek time. Throw some data away */
Frame const to_discard = min (-delta_frames, static_cast<Frame> (data->frames()));
Frame const to_keep = data->frames() - to_discard;
if (to_keep == 0) {
/* We have to throw all this data away, so keep _seek_reference and
try again next time some data arrives.
*/
return;
}
shared_ptr<AudioBuffers> trimmed (new AudioBuffers (data->channels(), to_keep));
trimmed->copy_from (data.get(), to_keep, to_discard, 0);
data = trimmed;
time += ContentTime::from_frames (to_discard, frame_rate);
}
_seek_reference = optional<ContentTime> ();
}
if (!_position) {
_position = time.frames_round (frame_rate);
}
DCPOMATIC_ASSERT (_position.get() >= (_decoded.frame + _decoded.audio->frames()));
add (data);
}
void
AudioDecoderStream::add (shared_ptr<const AudioBuffers> data)
{
if (!_position) {
/* This should only happen when there is a seek followed by a flush, but
we need to cope with it.
*/
return;
}
/* Resize _decoded to fit the new data */
int new_size = 0;
if (_decoded.audio->frames() == 0) {
/* There's nothing in there, so just store the new data */
new_size = data->frames ();
_decoded.frame = _position.get ();
} else {
/* Otherwise we need to extend _decoded to include the new stuff */
new_size = _position.get() + data->frames() - _decoded.frame;
}
_decoded.audio->ensure_size (new_size);
_decoded.audio->set_frames (new_size);
/* Copy new data in */
_decoded.audio->copy_from (data.get(), data->frames(), 0, _position.get() - _decoded.frame);
_position = _position.get() + data->frames ();
/* Limit the amount of data we keep in case nobody is asking for it */
int const max_frames = _content->resampled_audio_frame_rate () * 10;
if (_decoded.audio->frames() > max_frames) {
int const to_remove = _decoded.audio->frames() - max_frames;
_decoded.frame += to_remove;
_decoded.audio->move (to_remove, 0, max_frames);
_decoded.audio->set_frames (max_frames);
}
}
void
AudioDecoderStream::flush ()
{
if (!_resampler) {
return;
}
shared_ptr<const AudioBuffers> b = _resampler->flush ();
if (b) {
add (b);
}
}
void
AudioDecoderStream::seek (ContentTime t, bool accurate)
{
_position.reset ();
reset_decoded ();
if (accurate) {
_seek_reference = t;
}
}
|
