FFmpeg: libavutil/samplefmt.c Source File

FFmpeg
samplefmt.c
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1 /*
2  * This file is part of FFmpeg.
3  *
4  * FFmpeg 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  * FFmpeg 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 FFmpeg; if not, write to the Free Software
16  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17  */
18 
19 #include "common.h"
20 #include "samplefmt.h"
21 
22 #include <stdio.h>
23 #include <stdlib.h>
24 #include <string.h>
25 
26  typedef struct SampleFmtInfo {
27   char name[8];
28   int bits;
29   int planar;
30   enum AVSampleFormat altform; ///< planar<->packed alternative form
31 } SampleFmtInfo;
32 
33 /** this table gives more information about formats */
34  static const SampleFmtInfo sample_fmt_info[AV_SAMPLE_FMT_NB] = {
35  [AV_SAMPLE_FMT_U8] = { .name = "u8", .bits = 8, .planar = 0, .altform = AV_SAMPLE_FMT_U8P },
36  [AV_SAMPLE_FMT_S16] = { .name = "s16", .bits = 16, .planar = 0, .altform = AV_SAMPLE_FMT_S16P },
37  [AV_SAMPLE_FMT_S32] = { .name = "s32", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_S32P },
38  [AV_SAMPLE_FMT_FLT] = { .name = "flt", .bits = 32, .planar = 0, .altform = AV_SAMPLE_FMT_FLTP },
39  [AV_SAMPLE_FMT_DBL] = { .name = "dbl", .bits = 64, .planar = 0, .altform = AV_SAMPLE_FMT_DBLP },
40  [AV_SAMPLE_FMT_U8P] = { .name = "u8p", .bits = 8, .planar = 1, .altform = AV_SAMPLE_FMT_U8 },
41  [AV_SAMPLE_FMT_S16P] = { .name = "s16p", .bits = 16, .planar = 1, .altform = AV_SAMPLE_FMT_S16 },
42  [AV_SAMPLE_FMT_S32P] = { .name = "s32p", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_S32 },
43  [AV_SAMPLE_FMT_FLTP] = { .name = "fltp", .bits = 32, .planar = 1, .altform = AV_SAMPLE_FMT_FLT },
44  [AV_SAMPLE_FMT_DBLP] = { .name = "dblp", .bits = 64, .planar = 1, .altform = AV_SAMPLE_FMT_DBL },
45 };
46 
47  const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt)
48 {
49  if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
50  return NULL;
51  return sample_fmt_info[sample_fmt].name;
52 }
53 
54  enum AVSampleFormat av_get_sample_fmt(const char *name)
55 {
56  int i;
57 
58  for (i = 0; i < AV_SAMPLE_FMT_NB; i++)
59  if (!strcmp(sample_fmt_info[i].name, name))
60  return i;
61  return AV_SAMPLE_FMT_NONE;
62 }
63 
64  enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar)
65 {
66  if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
67  return AV_SAMPLE_FMT_NONE;
68  if (sample_fmt_info[sample_fmt].planar == planar)
69  return sample_fmt;
70  return sample_fmt_info[sample_fmt].altform;
71 }
72 
73  enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
74 {
75  if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
76  return AV_SAMPLE_FMT_NONE;
77  if (sample_fmt_info[sample_fmt].planar)
78  return sample_fmt_info[sample_fmt].altform;
79  return sample_fmt;
80 }
81 
82  enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt)
83 {
84  if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
85  return AV_SAMPLE_FMT_NONE;
86  if (sample_fmt_info[sample_fmt].planar)
87  return sample_fmt;
88  return sample_fmt_info[sample_fmt].altform;
89 }
90 
91  char *av_get_sample_fmt_string (char *buf, int buf_size, enum AVSampleFormat sample_fmt)
92 {
93  /* print header */
94  if (sample_fmt < 0)
95  snprintf(buf, buf_size, "name " " depth");
96  else if (sample_fmt < AV_SAMPLE_FMT_NB) {
97  SampleFmtInfo info = sample_fmt_info[sample_fmt];
98  snprintf (buf, buf_size, "%-6s" " %2d ", info.name, info.bits);
99  }
100 
101  return buf;
102 }
103 
104  int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
105 {
106  return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ?
107  0 : sample_fmt_info[sample_fmt].bits >> 3;
108 }
109 
110 #if FF_API_GET_BITS_PER_SAMPLE_FMT
111  int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt)
112 {
113  return sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB ?
114  0 : sample_fmt_info[sample_fmt].bits;
115 }
116 #endif
117 
118  int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
119 {
120  if (sample_fmt < 0 || sample_fmt >= AV_SAMPLE_FMT_NB)
121  return 0;
122  return sample_fmt_info[sample_fmt].planar;
123 }
124 
125  int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
126  enum AVSampleFormat sample_fmt, int align)
127 {
128  int line_size;
129  int sample_size = av_get_bytes_per_sample(sample_fmt);
130  int planar = av_sample_fmt_is_planar(sample_fmt);
131 
132  /* validate parameter ranges */
133  if (!sample_size || nb_samples <= 0 || nb_channels <= 0)
134  return AVERROR(EINVAL);
135 
136  /* auto-select alignment if not specified */
137  if (!align) {
138  if (nb_samples > INT_MAX - 31)
139  return AVERROR(EINVAL);
140  align = 1;
141  nb_samples = FFALIGN(nb_samples, 32);
142  }
143 
144  /* check for integer overflow */
145  if (nb_channels > INT_MAX / align ||
146  (int64_t)nb_channels * nb_samples > (INT_MAX - (align * nb_channels)) / sample_size)
147  return AVERROR(EINVAL);
148 
149  line_size = planar ? FFALIGN(nb_samples * sample_size, align) :
150  FFALIGN(nb_samples * sample_size * nb_channels, align);
151  if (linesize)
152  *linesize = line_size;
153 
154  return planar ? line_size * nb_channels : line_size;
155 }
156 
157  int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
158  const uint8_t *buf, int nb_channels, int nb_samples,
159  enum AVSampleFormat sample_fmt, int align)
160 {
161  int ch, planar, buf_size, line_size;
162 
163  planar = av_sample_fmt_is_planar(sample_fmt);
164  buf_size = av_samples_get_buffer_size(&line_size, nb_channels, nb_samples,
165  sample_fmt, align);
166  if (buf_size < 0)
167  return buf_size;
168 
169  audio_data[0] = (uint8_t *)buf;
170  for (ch = 1; planar && ch < nb_channels; ch++)
171  audio_data[ch] = audio_data[ch-1] + line_size;
172 
173  if (linesize)
174  *linesize = line_size;
175 
176 #if FF_API_SAMPLES_UTILS_RETURN_ZERO
177  return 0;
178 #else
179  return buf_size;
180 #endif
181 }
182 
183  int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
184  int nb_samples, enum AVSampleFormat sample_fmt, int align)
185 {
186  uint8_t *buf;
187  int size = av_samples_get_buffer_size(NULL, nb_channels, nb_samples,
188  sample_fmt, align);
189  if (size < 0)
190  return size;
191 
192  buf = av_malloc(size);
193  if (!buf)
194  return AVERROR(ENOMEM);
195 
196  size = av_samples_fill_arrays(audio_data, linesize, buf, nb_channels,
197  nb_samples, sample_fmt, align);
198  if (size < 0) {
199  av_free(buf);
200  return size;
201  }
202 
203  av_samples_set_silence(audio_data, 0, nb_samples, nb_channels, sample_fmt);
204 
205 #if FF_API_SAMPLES_UTILS_RETURN_ZERO
206  return 0;
207 #else
208  return size;
209 #endif
210 }
211 
212  int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels,
213  int nb_samples, enum AVSampleFormat sample_fmt, int align)
214 {
215  int ret, nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;
216 
217  *audio_data = av_calloc(nb_planes, sizeof(**audio_data));
218  if (!*audio_data)
219  return AVERROR(ENOMEM);
220  ret = av_samples_alloc(*audio_data, linesize, nb_channels,
221  nb_samples, sample_fmt, align);
222  if (ret < 0)
223  av_freep(audio_data);
224  return ret;
225 }
226 
227  int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset,
228  int src_offset, int nb_samples, int nb_channels,
229  enum AVSampleFormat sample_fmt)
230 {
231  int planar = av_sample_fmt_is_planar(sample_fmt);
232  int planes = planar ? nb_channels : 1;
233  int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);
234  int data_size = nb_samples * block_align;
235  int i;
236 
237  dst_offset *= block_align;
238  src_offset *= block_align;
239 
240  if((dst[0] < src[0] ? src[0] - dst[0] : dst[0] - src[0]) >= data_size) {
241  for (i = 0; i < planes; i++)
242  memcpy(dst[i] + dst_offset, src[i] + src_offset, data_size);
243  } else {
244  for (i = 0; i < planes; i++)
245  memmove(dst[i] + dst_offset, src[i] + src_offset, data_size);
246  }
247 
248  return 0;
249 }
250 
251  int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
252  int nb_channels, enum AVSampleFormat sample_fmt)
253 {
254  int planar = av_sample_fmt_is_planar(sample_fmt);
255  int planes = planar ? nb_channels : 1;
256  int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);
257  int data_size = nb_samples * block_align;
258  int fill_char = (sample_fmt == AV_SAMPLE_FMT_U8 ||
259  sample_fmt == AV_SAMPLE_FMT_U8P) ? 0x80 : 0x00;
260  int i;
261 
262  offset *= block_align;
263 
264  for (i = 0; i < planes; i++)
265  memset(audio_data[i] + offset, fill_char, data_size);
266 
267  return 0;
268 }
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