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
|
/*
* Copyright (c) 2012 Mans Rullgard <mans@mansr.com>
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/attributes.h"
#include "libavutil/samplefmt.h"
#include "flacdsp.h"
#include "config.h"
#define SAMPLE_SIZE 16
#define PLANAR 0
#include "flacdsp_template.c"
#include "flacdsp_lpc_template.c"
#undef PLANAR
#define PLANAR 1
#include "flacdsp_template.c"
#undef SAMPLE_SIZE
#undef PLANAR
#define SAMPLE_SIZE 32
#define PLANAR 0
#include "flacdsp_template.c"
#include "flacdsp_lpc_template.c"
#undef PLANAR
#define PLANAR 1
#include "flacdsp_template.c"
static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32],
int pred_order, int qlevel, int len)
{
int i, j;
for (i = pred_order; i < len - 1; i += 2, decoded += 2) {
int c = coeffs[0];
int d = decoded[0];
int s0 = 0, s1 = 0;
for (j = 1; j < pred_order; j++) {
s0 += c*d;
d = decoded[j];
s1 += c*d;
c = coeffs[j];
}
s0 += c*d;
d = decoded[j] += s0 >> qlevel;
s1 += c*d;
decoded[j + 1] += s1 >> qlevel;
}
if (i < len) {
int sum = 0;
for (j = 0; j < pred_order; j++)
sum += coeffs[j] * decoded[j];
decoded[j] += sum >> qlevel;
}
}
static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32],
int pred_order, int qlevel, int len)
{
int i, j;
for (i = pred_order; i < len; i++, decoded++) {
int64_t sum = 0;
for (j = 0; j < pred_order; j++)
sum += (int64_t)coeffs[j] * decoded[j];
decoded[j] += sum >> qlevel;
}
}
av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt,
int bps)
{
if (bps > 16) {
c->lpc = flac_lpc_32_c;
c->lpc_encode = flac_lpc_encode_c_32;
} else {
c->lpc = flac_lpc_16_c;
c->lpc_encode = flac_lpc_encode_c_16;
}
switch (fmt) {
case AV_SAMPLE_FMT_S32:
c->decorrelate[0] = flac_decorrelate_indep_c_32;
c->decorrelate[1] = flac_decorrelate_ls_c_32;
c->decorrelate[2] = flac_decorrelate_rs_c_32;
c->decorrelate[3] = flac_decorrelate_ms_c_32;
break;
case AV_SAMPLE_FMT_S32P:
c->decorrelate[0] = flac_decorrelate_indep_c_32p;
c->decorrelate[1] = flac_decorrelate_ls_c_32p;
c->decorrelate[2] = flac_decorrelate_rs_c_32p;
c->decorrelate[3] = flac_decorrelate_ms_c_32p;
break;
case AV_SAMPLE_FMT_S16:
c->decorrelate[0] = flac_decorrelate_indep_c_16;
c->decorrelate[1] = flac_decorrelate_ls_c_16;
c->decorrelate[2] = flac_decorrelate_rs_c_16;
c->decorrelate[3] = flac_decorrelate_ms_c_16;
break;
case AV_SAMPLE_FMT_S16P:
c->decorrelate[0] = flac_decorrelate_indep_c_16p;
c->decorrelate[1] = flac_decorrelate_ls_c_16p;
c->decorrelate[2] = flac_decorrelate_rs_c_16p;
c->decorrelate[3] = flac_decorrelate_ms_c_16p;
break;
}
if (ARCH_ARM)
ff_flacdsp_init_arm(c, fmt, bps);
}
|
