FFmpeg
flacdsp.c
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1 /*
2  * Copyright (c) 2012 Mans Rullgard <mans@mansr.com>
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "libavutil/attributes.h"
22 #include "libavutil/samplefmt.h"
23 #include "flacdsp.h"
24 #include "config.h"
25 
26 #define SAMPLE_SIZE 16
27 #define PLANAR 0
28 #include "flacdsp_template.c"
29 #include "flacdsp_lpc_template.c"
30 
31 #undef PLANAR
32 #define PLANAR 1
33 #include "flacdsp_template.c"
34 
35 #undef SAMPLE_SIZE
36 #undef PLANAR
37 #define SAMPLE_SIZE 32
38 #define PLANAR 0
39 #include "flacdsp_template.c"
40 #include "flacdsp_lpc_template.c"
41 
42 #undef PLANAR
43 #define PLANAR 1
44 #include "flacdsp_template.c"
45 
46 static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32],
47  int pred_order, int qlevel, int len)
48 {
49  int i, j;
50 
51  for (i = pred_order; i < len - 1; i += 2, decoded += 2) {
52  SUINT c = coeffs[0];
53  SUINT d = decoded[0];
54  int s0 = 0, s1 = 0;
55  for (j = 1; j < pred_order; j++) {
56  s0 += c*d;
57  d = decoded[j];
58  s1 += c*d;
59  c = coeffs[j];
60  }
61  s0 += c*d;
62  d = decoded[j] += (SUINT)(s0 >> qlevel);
63  s1 += c*d;
64  decoded[j + 1] += (SUINT)(s1 >> qlevel);
65  }
66  if (i < len) {
67  int sum = 0;
68  for (j = 0; j < pred_order; j++)
69  sum += coeffs[j] * (SUINT)decoded[j];
70  decoded[j] = decoded[j] + (unsigned)(sum >> qlevel);
71  }
72 }
73 
74 static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32],
75  int pred_order, int qlevel, int len)
76 {
77  int i, j;
78 
79  for (i = pred_order; i < len; i++, decoded++) {
80  int64_t sum = 0;
81  for (j = 0; j < pred_order; j++)
82  sum += (int64_t)coeffs[j] * decoded[j];
83  decoded[j] += sum >> qlevel;
84  }
85 
86 }
87 
89  int bps)
90 {
91  c->lpc16 = flac_lpc_16_c;
92  c->lpc32 = flac_lpc_32_c;
93  c->lpc16_encode = flac_lpc_encode_c_16;
94  c->lpc32_encode = flac_lpc_encode_c_32;
95 
96  switch (fmt) {
97  case AV_SAMPLE_FMT_S32:
98  c->decorrelate[0] = flac_decorrelate_indep_c_32;
99  c->decorrelate[1] = flac_decorrelate_ls_c_32;
100  c->decorrelate[2] = flac_decorrelate_rs_c_32;
101  c->decorrelate[3] = flac_decorrelate_ms_c_32;
102  break;
103 
104  case AV_SAMPLE_FMT_S32P:
105  c->decorrelate[0] = flac_decorrelate_indep_c_32p;
106  c->decorrelate[1] = flac_decorrelate_ls_c_32p;
107  c->decorrelate[2] = flac_decorrelate_rs_c_32p;
108  c->decorrelate[3] = flac_decorrelate_ms_c_32p;
109  break;
110 
111  case AV_SAMPLE_FMT_S16:
112  c->decorrelate[0] = flac_decorrelate_indep_c_16;
113  c->decorrelate[1] = flac_decorrelate_ls_c_16;
114  c->decorrelate[2] = flac_decorrelate_rs_c_16;
115  c->decorrelate[3] = flac_decorrelate_ms_c_16;
116  break;
117 
118  case AV_SAMPLE_FMT_S16P:
119  c->decorrelate[0] = flac_decorrelate_indep_c_16p;
120  c->decorrelate[1] = flac_decorrelate_ls_c_16p;
121  c->decorrelate[2] = flac_decorrelate_rs_c_16p;
122  c->decorrelate[3] = flac_decorrelate_ms_c_16p;
123  break;
124  }
125 
126  if (ARCH_ARM)
128  if (ARCH_X86)
130 }
ff_flacdsp_init
av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
Definition: flacdsp.c:88
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
flacdsp.h
samplefmt.h
av_cold
#define av_cold
Definition: attributes.h:90
s1
#define s1
Definition: regdef.h:38
channels
channels
Definition: aptx.h:33
ff_flacdsp_init_x86
void ff_flacdsp_init_x86(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
Definition: flacdsp_init.c:54
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
flac_lpc_32_c
static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32], int pred_order, int qlevel, int len)
Definition: flacdsp.c:74
ff_flacdsp_init_arm
av_cold void ff_flacdsp_init_arm(FLACDSPContext *c, enum AVSampleFormat fmt, int channels, int bps)
Definition: flacdsp_init_arm.c:28
bps
unsigned bps
Definition: movenc.c:1595
attributes.h
AV_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
i
int i
Definition: input.c:407
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
SUINT
#define SUINT
Definition: dct32_template.c:30
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
len
int len
Definition: vorbis_enc_data.h:452
flac_lpc_16_c
static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32], int pred_order, int qlevel, int len)
Definition: flacdsp.c:46
flacdsp_lpc_template.c
flacdsp_template.c
s0
#define s0
Definition: regdef.h:37
d
d
Definition: ffmpeg_filter.c:156
int32_t
int32_t
Definition: audioconvert.c:56
AV_SAMPLE_FMT_S32
@ AV_SAMPLE_FMT_S32
signed 32 bits
Definition: samplefmt.h:62
FLACDSPContext
Definition: flacdsp.h:26