Imported Debian version 2.5.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / flacdsp.c
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
88 av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt, int channels,
89 int bps)
90 {
91 if (bps > 16) {
92 c->lpc = flac_lpc_32_c;
93 c->lpc_encode = flac_lpc_encode_c_32;
94 } else {
95 c->lpc = flac_lpc_16_c;
96 c->lpc_encode = flac_lpc_encode_c_16;
97 }
98
99 switch (fmt) {
100 case AV_SAMPLE_FMT_S32:
101 c->decorrelate[0] = flac_decorrelate_indep_c_32;
102 c->decorrelate[1] = flac_decorrelate_ls_c_32;
103 c->decorrelate[2] = flac_decorrelate_rs_c_32;
104 c->decorrelate[3] = flac_decorrelate_ms_c_32;
105 break;
106
107 case AV_SAMPLE_FMT_S32P:
108 c->decorrelate[0] = flac_decorrelate_indep_c_32p;
109 c->decorrelate[1] = flac_decorrelate_ls_c_32p;
110 c->decorrelate[2] = flac_decorrelate_rs_c_32p;
111 c->decorrelate[3] = flac_decorrelate_ms_c_32p;
112 break;
113
114 case AV_SAMPLE_FMT_S16:
115 c->decorrelate[0] = flac_decorrelate_indep_c_16;
116 c->decorrelate[1] = flac_decorrelate_ls_c_16;
117 c->decorrelate[2] = flac_decorrelate_rs_c_16;
118 c->decorrelate[3] = flac_decorrelate_ms_c_16;
119 break;
120
121 case AV_SAMPLE_FMT_S16P:
122 c->decorrelate[0] = flac_decorrelate_indep_c_16p;
123 c->decorrelate[1] = flac_decorrelate_ls_c_16p;
124 c->decorrelate[2] = flac_decorrelate_rs_c_16p;
125 c->decorrelate[3] = flac_decorrelate_ms_c_16p;
126 break;
127 }
128
129 if (ARCH_ARM)
130 ff_flacdsp_init_arm(c, fmt, channels, bps);
131 if (ARCH_X86)
132 ff_flacdsp_init_x86(c, fmt, channels, bps);
133 }