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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  int c = coeffs[0];
53  int 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] += s0 >> qlevel;
63  s1 += c*d;
64  decoded[j + 1] += s1 >> qlevel;
65  }
66  if (i < len) {
67  int sum = 0;
68  for (j = 0; j < pred_order; j++)
69  sum += coeffs[j] * decoded[j];
70  decoded[j] += 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)
127  ff_flacdsp_init_arm(c, fmt, channels, bps);
128  if (ARCH_X86)
129  ff_flacdsp_init_x86(c, fmt, channels, bps);
130 }