compat/vsnprintf: return number of bytes required on truncation.
[libav.git] / libavcodec / flacdsp.c
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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
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26#define PLANAR 0
27#include "flacdsp_template.c"
28
29#undef PLANAR
30#define PLANAR 1
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31#include "flacdsp_template.c"
32
33#undef SAMPLE_SIZE
784514a4 34#undef PLANAR
4a852834 35#define SAMPLE_SIZE 32
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36#define PLANAR 0
37#include "flacdsp_template.c"
38
39#undef PLANAR
40#define PLANAR 1
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41#include "flacdsp_template.c"
42
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43static 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
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48 for (i = pred_order; i < len - 1; i += 2, decoded += 2) {
49 int c = coeffs[0];
50 int d = decoded[0];
25accf93 51 int s0 = 0, s1 = 0;
bf1cf4d5 52 for (j = 1; j < pred_order; j++) {
25accf93 53 s0 += c*d;
bf1cf4d5 54 d = decoded[j];
25accf93 55 s1 += c*d;
bf1cf4d5 56 c = coeffs[j];
25accf93 57 }
25accf93 58 s0 += c*d;
bf1cf4d5 59 d = decoded[j] += s0 >> qlevel;
25accf93 60 s1 += c*d;
bf1cf4d5 61 decoded[j + 1] += s1 >> qlevel;
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62 }
63 if (i < len) {
64 int sum = 0;
65 for (j = 0; j < pred_order; j++)
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66 sum += coeffs[j] * decoded[j];
67 decoded[j] += sum >> qlevel;
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68 }
69}
70
71static void flac_lpc_32_c(int32_t *decoded, const int coeffs[32],
72 int pred_order, int qlevel, int len)
73{
74 int i, j;
75
bf1cf4d5 76 for (i = pred_order; i < len; i++, decoded++) {
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77 int64_t sum = 0;
78 for (j = 0; j < pred_order; j++)
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79 sum += (int64_t)coeffs[j] * decoded[j];
80 decoded[j] += sum >> qlevel;
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81 }
82
83}
84
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85av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt,
86 int bps)
4a852834 87{
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88 if (bps > 16)
89 c->lpc = flac_lpc_32_c;
90 else
91 c->lpc = flac_lpc_16_c;
92
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93 switch (fmt) {
94 case AV_SAMPLE_FMT_S32:
95 c->decorrelate[0] = flac_decorrelate_indep_c_32;
96 c->decorrelate[1] = flac_decorrelate_ls_c_32;
97 c->decorrelate[2] = flac_decorrelate_rs_c_32;
98 c->decorrelate[3] = flac_decorrelate_ms_c_32;
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99 break;
100
101 case AV_SAMPLE_FMT_S32P:
102 c->decorrelate[0] = flac_decorrelate_indep_c_32p;
103 c->decorrelate[1] = flac_decorrelate_ls_c_32p;
104 c->decorrelate[2] = flac_decorrelate_rs_c_32p;
105 c->decorrelate[3] = flac_decorrelate_ms_c_32p;
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106 break;
107
108 case AV_SAMPLE_FMT_S16:
109 c->decorrelate[0] = flac_decorrelate_indep_c_16;
110 c->decorrelate[1] = flac_decorrelate_ls_c_16;
111 c->decorrelate[2] = flac_decorrelate_rs_c_16;
112 c->decorrelate[3] = flac_decorrelate_ms_c_16;
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113 break;
114
115 case AV_SAMPLE_FMT_S16P:
116 c->decorrelate[0] = flac_decorrelate_indep_c_16p;
117 c->decorrelate[1] = flac_decorrelate_ls_c_16p;
118 c->decorrelate[2] = flac_decorrelate_rs_c_16p;
119 c->decorrelate[3] = flac_decorrelate_ms_c_16p;
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120 break;
121 }
122}