libavutil: add utility functions to simplify allocation of audio buffers.
[libav.git] / libavutil / samplefmt.c
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d63e456a 1/*
2912e87a 2 * This file is part of Libav.
d63e456a 3 *
2912e87a 4 * Libav is free software; you can redistribute it and/or
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5 * modify it under the terms of the GNU Lesser General Public
6 * License as published by the Free Software Foundation; either
7 * version 2.1 of the License, or (at your option) any later version.
8 *
2912e87a 9 * Libav is distributed in the hope that it will be useful,
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10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Lesser General Public License for more details.
13 *
14 * You should have received a copy of the GNU Lesser General Public
2912e87a 15 * License along with Libav; if not, write to the Free Software
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16 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17 */
18
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19#include "samplefmt.h"
20
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21#include <stdio.h>
22#include <stdlib.h>
23#include <string.h>
24
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25typedef struct SampleFmtInfo {
26 const char *name;
27 int bits;
8889cc4f 28 int planar;
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29} SampleFmtInfo;
30
31/** this table gives more information about formats */
32static const SampleFmtInfo sample_fmt_info[AV_SAMPLE_FMT_NB] = {
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33 [AV_SAMPLE_FMT_U8] = { .name = "u8", .bits = 8, .planar = 0 },
34 [AV_SAMPLE_FMT_S16] = { .name = "s16", .bits = 16, .planar = 0 },
35 [AV_SAMPLE_FMT_S32] = { .name = "s32", .bits = 32, .planar = 0 },
36 [AV_SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32, .planar = 0 },
37 [AV_SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64, .planar = 0 },
38 [AV_SAMPLE_FMT_U8P] = { .name = "u8p", .bits = 8, .planar = 1 },
39 [AV_SAMPLE_FMT_S16P] = { .name = "s16p", .bits = 16, .planar = 1 },
40 [AV_SAMPLE_FMT_S32P] = { .name = "s32p", .bits = 32, .planar = 1 },
41 [AV_SAMPLE_FMT_FLTP] = { .name = "fltp", .bits = 32, .planar = 1 },
42 [AV_SAMPLE_FMT_DBLP] = { .name = "dblp", .bits = 64, .planar = 1 },
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43};
44
45const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
46{
47 if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
48 return NULL;
49 return sample_fmt_info[sample_fmt].name;
50}
51
52enum AVSampleFormat av_get_sample_fmt(const char *name)
53{
54 int i;
55
56 for (i = 0; i < AV_SAMPLE_FMT_NB; i++)
57 if (!strcmp(sample_fmt_info[i].name, name))
58 return i;
59 return AV_SAMPLE_FMT_NONE;
60}
61
62char *av_get_sample_fmt_string (char *buf, int buf_size, enum AVSampleFormat sample_fmt)
63{
64 /* print header */
65 if (sample_fmt < 0)
66 snprintf(buf, buf_size, "name " " depth");
67 else if (sample_fmt < AV_SAMPLE_FMT_NB) {
68 SampleFmtInfo info = sample_fmt_info[sample_fmt];
69 snprintf (buf, buf_size, "%-6s" " %2d ", info.name, info.bits);
70 }
71
72 return buf;
73}
6f84cd12 74
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75int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
76{
77 return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ?
78 0 : sample_fmt_info[sample_fmt].bits >> 3;
79}
80
81#if FF_API_GET_BITS_PER_SAMPLE_FMT
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82int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt)
83{
84 return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ?
85 0 : sample_fmt_info[sample_fmt].bits;
86}
a6703faa 87#endif
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88
89int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
90{
91 if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
92 return 0;
93 return sample_fmt_info[sample_fmt].planar;
94}
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95
96int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
97 enum AVSampleFormat sample_fmt, int align)
98{
99 int line_size;
100 int sample_size = av_get_bytes_per_sample(sample_fmt);
101 int planar = av_sample_fmt_is_planar(sample_fmt);
102
103 /* validate parameter ranges */
104 if (!sample_size || nb_samples <= 0 || nb_channels <= 0)
105 return AVERROR(EINVAL);
106
107 /* check for integer overflow */
108 if (nb_channels > INT_MAX / align ||
109 (int64_t)nb_channels * nb_samples > (INT_MAX - (align * nb_channels)) / sample_size)
110 return AVERROR(EINVAL);
111
112 line_size = planar ? FFALIGN(nb_samples * sample_size, align) :
113 FFALIGN(nb_samples * sample_size * nb_channels, align);
114 if (linesize)
115 *linesize = line_size;
116
117 return planar ? line_size * nb_channels : line_size;
118}
119
120int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
121 uint8_t *buf, int nb_channels, int nb_samples,
122 enum AVSampleFormat sample_fmt, int align)
123{
124 int ch, planar, buf_size;
125
126 planar = av_sample_fmt_is_planar(sample_fmt);
127 buf_size = av_samples_get_buffer_size(linesize, nb_channels, nb_samples,
128 sample_fmt, align);
129 if (buf_size < 0)
130 return buf_size;
131
132 audio_data[0] = buf;
133 for (ch = 1; planar && ch < nb_channels; ch++)
134 audio_data[ch] = audio_data[ch-1] + *linesize;
135
136 return 0;
137}
138
139int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
140 int nb_samples, enum AVSampleFormat sample_fmt, int align)
141{
142 uint8_t *buf;
143 int size = av_samples_get_buffer_size(NULL, nb_channels, nb_samples,
144 sample_fmt, align);
145 if (size < 0)
146 return size;
147
148 buf = av_mallocz(size);
149 if (!buf)
150 return AVERROR(ENOMEM);
151
152 size = av_samples_fill_arrays(audio_data, linesize, buf, nb_channels,
153 nb_samples, sample_fmt, align);
154 if (size < 0) {
155 av_free(buf);
156 return size;
157 }
158 return 0;
159}