flacdec: add planar output support
[libav.git] / libavcodec / flacdsp.c
1 /*
2 * Copyright (c) 2012 Mans Rullgard <mans@mansr.com>
3 *
4 * This file is part of Libav.
5 *
6 * Libav 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 * Libav 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 Libav; 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
25 #define SAMPLE_SIZE 16
26 #define PLANAR 0
27 #include "flacdsp_template.c"
28
29 #undef PLANAR
30 #define PLANAR 1
31 #include "flacdsp_template.c"
32
33 #undef SAMPLE_SIZE
34 #undef PLANAR
35 #define SAMPLE_SIZE 32
36 #define PLANAR 0
37 #include "flacdsp_template.c"
38
39 #undef PLANAR
40 #define PLANAR 1
41 #include "flacdsp_template.c"
42
43 static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32],
44 int pred_order, int qlevel, int len)
45 {
46 int i, j;
47
48 for (i = pred_order; i < len - 1; i += 2) {
49 int c;
50 int d = decoded[i-pred_order];
51 int s0 = 0, s1 = 0;
52 for (j = pred_order-1; j > 0; j--) {
53 c = coeffs[j];
54 s0 += c*d;
55 d = decoded[i-j];
56 s1 += c*d;
57 }
58 c = coeffs[0];
59 s0 += c*d;
60 d = decoded[i] += s0 >> qlevel;
61 s1 += c*d;
62 decoded[i+1] += s1 >> qlevel;
63 }
64 if (i < len) {
65 int sum = 0;
66 for (j = 0; j < pred_order; j++)
67 sum += coeffs[j] * decoded[i-j-1];
68 decoded[i] += sum >> qlevel;
69 }
70 }
71
72 static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32],
73 int pred_order, int qlevel, int len)
74 {
75 int i, j;
76
77 for (i = pred_order; i < len; i++) {
78 int64_t sum = 0;
79 for (j = 0; j < pred_order; j++)
80 sum += (int64_t)coeffs[j] * decoded[i-j-1];
81 decoded[i] += sum >> qlevel;
82 }
83
84 }
85
86 av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt,
87 int bps)
88 {
89 if (bps > 16)
90 c->lpc = flac_lpc_32_c;
91 else
92 c->lpc = flac_lpc_16_c;
93
94 switch (fmt) {
95 case AV_SAMPLE_FMT_S32:
96 c->decorrelate[0] = flac_decorrelate_indep_c_32;
97 c->decorrelate[1] = flac_decorrelate_ls_c_32;
98 c->decorrelate[2] = flac_decorrelate_rs_c_32;
99 c->decorrelate[3] = flac_decorrelate_ms_c_32;
100 break;
101
102 case AV_SAMPLE_FMT_S32P:
103 c->decorrelate[0] = flac_decorrelate_indep_c_32p;
104 c->decorrelate[1] = flac_decorrelate_ls_c_32p;
105 c->decorrelate[2] = flac_decorrelate_rs_c_32p;
106 c->decorrelate[3] = flac_decorrelate_ms_c_32p;
107 break;
108
109 case AV_SAMPLE_FMT_S16:
110 c->decorrelate[0] = flac_decorrelate_indep_c_16;
111 c->decorrelate[1] = flac_decorrelate_ls_c_16;
112 c->decorrelate[2] = flac_decorrelate_rs_c_16;
113 c->decorrelate[3] = flac_decorrelate_ms_c_16;
114 break;
115
116 case AV_SAMPLE_FMT_S16P:
117 c->decorrelate[0] = flac_decorrelate_indep_c_16p;
118 c->decorrelate[1] = flac_decorrelate_ls_c_16p;
119 c->decorrelate[2] = flac_decorrelate_rs_c_16p;
120 c->decorrelate[3] = flac_decorrelate_ms_c_16p;
121 break;
122 }
123 }