flacenc: use uint64_t for bit counts
[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"
7689eea4 24#include "config.h"
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25
26#define SAMPLE_SIZE 16
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27#define PLANAR 0
28#include "flacdsp_template.c"
29
30#undef PLANAR
31#define PLANAR 1
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32#include "flacdsp_template.c"
33
34#undef SAMPLE_SIZE
784514a4 35#undef PLANAR
4a852834 36#define SAMPLE_SIZE 32
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37#define PLANAR 0
38#include "flacdsp_template.c"
39
40#undef PLANAR
41#define PLANAR 1
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42#include "flacdsp_template.c"
43
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44static void flac_lpc_16_c(int32_t *decoded, const int coeffs[32],
45 int pred_order, int qlevel, int len)
46{
47 int i, j;
48
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49 for (i = pred_order; i < len - 1; i += 2, decoded += 2) {
50 int c = coeffs[0];
51 int d = decoded[0];
25accf93 52 int s0 = 0, s1 = 0;
bf1cf4d5 53 for (j = 1; j < pred_order; j++) {
25accf93 54 s0 += c*d;
bf1cf4d5 55 d = decoded[j];
25accf93 56 s1 += c*d;
bf1cf4d5 57 c = coeffs[j];
25accf93 58 }
25accf93 59 s0 += c*d;
bf1cf4d5 60 d = decoded[j] += s0 >> qlevel;
25accf93 61 s1 += c*d;
bf1cf4d5 62 decoded[j + 1] += s1 >> qlevel;
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63 }
64 if (i < len) {
65 int sum = 0;
66 for (j = 0; j < pred_order; j++)
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67 sum += coeffs[j] * decoded[j];
68 decoded[j] += sum >> qlevel;
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69 }
70}
71
72static 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
bf1cf4d5 77 for (i = pred_order; i < len; i++, decoded++) {
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78 int64_t sum = 0;
79 for (j = 0; j < pred_order; j++)
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80 sum += (int64_t)coeffs[j] * decoded[j];
81 decoded[j] += sum >> qlevel;
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82 }
83
84}
85
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86av_cold void ff_flacdsp_init(FLACDSPContext *c, enum AVSampleFormat fmt,
87 int bps)
4a852834 88{
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89 if (bps > 16)
90 c->lpc = flac_lpc_32_c;
91 else
92 c->lpc = flac_lpc_16_c;
93
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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;
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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;
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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;
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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;
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121 break;
122 }
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123
124 if (ARCH_ARM)
125 ff_flacdsp_init_arm(c, fmt, bps);
4a852834 126}