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
|
/*
Copyright (C) 2012-2014 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.h"
#include "audio_buffers.h"
#include "exceptions.h"
#include "log.h"
#include "resampler.h"
#include "util.h"
#include "film.h"
#include "i18n.h"
using std::stringstream;
using std::list;
using std::pair;
using std::cout;
using std::min;
using boost::optional;
using boost::shared_ptr;
AudioDecoder::AudioDecoder (shared_ptr<const AudioContent> content)
: _audio_content (content)
{
if (content->output_audio_frame_rate() != content->content_audio_frame_rate() && content->audio_channels ()) {
_resampler.reset (new Resampler (content->content_audio_frame_rate(), content->output_audio_frame_rate(), content->audio_channels ()));
}
reset_decoded_audio ();
}
void
AudioDecoder::reset_decoded_audio ()
{
_decoded_audio = ContentAudio (shared_ptr<AudioBuffers> (new AudioBuffers (_audio_content->audio_channels(), 0)), 0);
}
shared_ptr<ContentAudio>
AudioDecoder::get_audio (AudioFrame frame, AudioFrame length, bool accurate)
{
shared_ptr<ContentAudio> dec;
AudioFrame const end = frame + length - 1;
if (frame < _decoded_audio.frame || end > (_decoded_audio.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, _audio_content->content_audio_frame_rate()), accurate);
}
AudioFrame 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) {
while (!pass() && _decoded_audio.audio->frames() < length) {}
/* Use decoded_offset of 0, as we don't really care what frames we return */
} else {
while (!pass() && (_decoded_audio.frame > frame || (_decoded_audio.frame + _decoded_audio.audio->frames()) < end)) {}
decoded_offset = frame - _decoded_audio.frame;
}
AudioFrame const amount_left = _decoded_audio.audio->frames() - decoded_offset;
AudioFrame const to_return = min (amount_left, length);
shared_ptr<AudioBuffers> out (new AudioBuffers (_decoded_audio.audio->channels(), to_return));
out->copy_from (_decoded_audio.audio.get(), to_return, decoded_offset, 0);
/* Clean up decoded */
_decoded_audio.audio->move (decoded_offset + to_return, 0, amount_left - to_return);
_decoded_audio.audio->set_frames (amount_left - to_return);
return shared_ptr<ContentAudio> (new ContentAudio (out, frame));
}
/** Called by subclasses when audio data is ready.
*
* 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 _audio_position. The
* time is ignored once this has been done.
*/
void
AudioDecoder::audio (shared_ptr<const AudioBuffers> data, ContentTime time)
{
if (_resampler) {
data = _resampler->run (data);
}
if (!_audio_position) {
_audio_position = time.frames (_audio_content->output_audio_frame_rate ());
}
assert (_audio_position.get() >= (_decoded_audio.frame + _decoded_audio.audio->frames()));
/* Resize _decoded_audio to fit the new data */
int const new_size = _audio_position.get() + data->frames() - _decoded_audio.frame;
_decoded_audio.audio->ensure_size (new_size);
_decoded_audio.audio->set_frames (new_size);
/* Copy new data in */
_decoded_audio.audio->copy_from (data.get(), data->frames(), 0, _audio_position.get() - _decoded_audio.frame);
_audio_position = _audio_position.get() + data->frames ();
}
/* XXX: called? */
void
AudioDecoder::flush ()
{
if (!_resampler) {
return;
}
/*
shared_ptr<const AudioBuffers> b = _resampler->flush ();
if (b) {
_pending.push_back (shared_ptr<DecodedAudio> (new DecodedAudio (b, _audio_position.get ())));
_audio_position = _audio_position.get() + b->frames ();
}
*/
}
void
AudioDecoder::seek (ContentTime, bool)
{
_audio_position.reset ();
reset_decoded_audio ();
}
|
