samplefmt.c
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1 /*
2  * This file is part of Libav.
3  *
4  * Libav is free software; you can redistribute it and/or
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  *
9  * Libav is distributed in the hope that it will be useful,
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
15  * License along with Libav; if not, write to the Free Software
16  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17  */
18 
19 #include "samplefmt.h"
20 
21 #include <stdio.h>
22 #include <stdlib.h>
23 #include <string.h>
24 
25 typedef struct SampleFmtInfo {
26  const char *name;
27  int bits;
28  int planar;
30 
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 },
43 };
44 
45 const 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 
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 
62 char *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 }
74 
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
82 int 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 }
87 #endif
88 
90 {
91  if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
92  return 0;
93  return sample_fmt_info[sample_fmt].planar;
94 }
95 
96 int 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 
120 int 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 
139 int 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 }