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
|
/*
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 <iostream>
#include <sndfile.h>
#include "sndfile_content.h"
#include "sndfile_decoder.h"
#include "exceptions.h"
#include "audio_buffers.h"
#include "i18n.h"
using std::vector;
using std::string;
using std::min;
using std::cout;
using boost::shared_ptr;
SndfileDecoder::SndfileDecoder (shared_ptr<const SndfileContent> c)
: Sndfile (c)
, AudioDecoder (c)
, _done (0)
, _remaining (_info.frames)
, _deinterleave_buffer (0)
{
}
SndfileDecoder::~SndfileDecoder ()
{
delete[] _deinterleave_buffer;
}
bool
SndfileDecoder::pass ()
{
if (_remaining == 0) {
return true;
}
/* Do things in half second blocks as I think there may be limits
to what FFmpeg (and in particular the resampler) can cope with.
*/
sf_count_t const block = _sndfile_content->audio_stream()->frame_rate() / 2;
sf_count_t const this_time = min (block, _remaining);
int const channels = _sndfile_content->audio_stream()->channels ();
shared_ptr<AudioBuffers> data (new AudioBuffers (channels, this_time));
if (_sndfile_content->audio_stream()->channels() == 1) {
/* No de-interleaving required */
sf_read_float (_sndfile, data->data(0), this_time);
} else {
/* Deinterleave */
if (!_deinterleave_buffer) {
_deinterleave_buffer = new float[block * channels];
}
sf_readf_float (_sndfile, _deinterleave_buffer, this_time);
vector<float*> out_ptr (channels);
for (int i = 0; i < channels; ++i) {
out_ptr[i] = data->data(i);
}
float* in_ptr = _deinterleave_buffer;
for (int i = 0; i < this_time; ++i) {
for (int j = 0; j < channels; ++j) {
*out_ptr[j]++ = *in_ptr++;
}
}
}
data->set_frames (this_time);
audio (_sndfile_content->audio_stream (), data, ContentTime::from_frames (_done, _info.samplerate));
_done += this_time;
_remaining -= this_time;
return _remaining == 0;
}
void
SndfileDecoder::seek (ContentTime t, bool accurate)
{
AudioDecoder::seek (t, accurate);
_done = t.frames (_info.samplerate);
_remaining = _info.frames - _done;
}
|
