FFmpeg: libavutil/samplefmt.c Source File

FFmpeg
samplefmt.c
Go to the documentation of this file.
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  align = 1;
139  nb_samples = FFALIGN(nb_samples, 32);
140  }
141 
142  /* check for integer overflow */
143  if (nb_channels > INT_MAX / align ||
144  (int64_t)nb_channels * nb_samples > (INT_MAX - (align * nb_channels)) / sample_size)
145  return AVERROR(EINVAL);
146 
147  line_size = planar ? FFALIGN(nb_samples * sample_size, align) :
148  FFALIGN(nb_samples * sample_size * nb_channels, align);
149  if (linesize)
150  *linesize = line_size;
151 
152  return planar ? line_size * nb_channels : line_size;
153 }
154 
155  int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
156  const uint8_t *buf, int nb_channels, int nb_samples,
157  enum AVSampleFormat sample_fmt, int align)
158 {
159  int ch, planar, buf_size, line_size;
160 
161  planar = av_sample_fmt_is_planar(sample_fmt);
162  buf_size = av_samples_get_buffer_size(&line_size, nb_channels, nb_samples,
163  sample_fmt, align);
164  if (buf_size < 0)
165  return buf_size;
166 
167  audio_data[0] = (uint8_t *)buf;
168  for (ch = 1; planar && ch < nb_channels; ch++)
169  audio_data[ch] = audio_data[ch-1] + line_size;
170 
171  if (linesize)
172  *linesize = line_size;
173 
174 #if FF_API_SAMPLES_UTILS_RETURN_ZERO
175  return 0;
176 #else
177  return buf_size;
178 #endif
179 }
180 
181  int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
182  int nb_samples, enum AVSampleFormat sample_fmt, int align)
183 {
184  uint8_t *buf;
185  int size = av_samples_get_buffer_size(NULL, nb_channels, nb_samples,
186  sample_fmt, align);
187  if (size < 0)
188  return size;
189 
190  buf = av_malloc(size);
191  if (!buf)
192  return AVERROR(ENOMEM);
193 
194  size = av_samples_fill_arrays(audio_data, linesize, buf, nb_channels,
195  nb_samples, sample_fmt, align);
196  if (size < 0) {
197  av_free(buf);
198  return size;
199  }
200 
201  av_samples_set_silence(audio_data, 0, nb_samples, nb_channels, sample_fmt);
202 
203 #if FF_API_SAMPLES_UTILS_RETURN_ZERO
204  return 0;
205 #else
206  return size;
207 #endif
208 }
209 
210  int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset,
211  int src_offset, int nb_samples, int nb_channels,
212  enum AVSampleFormat sample_fmt)
213 {
214  int planar = av_sample_fmt_is_planar(sample_fmt);
215  int planes = planar ? nb_channels : 1;
216  int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);
217  int data_size = nb_samples * block_align;
218  int i;
219 
220  dst_offset *= block_align;
221  src_offset *= block_align;
222 
223  if((dst[0] < src[0] ? src[0] - dst[0] : dst[0] - src[0]) >= data_size) {
224  for (i = 0; i < planes; i++)
225  memcpy(dst[i] + dst_offset, src[i] + src_offset, data_size);
226  } else {
227  for (i = 0; i < planes; i++)
228  memmove(dst[i] + dst_offset, src[i] + src_offset, data_size);
229  }
230 
231  return 0;
232 }
233 
234  int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
235  int nb_channels, enum AVSampleFormat sample_fmt)
236 {
237  int planar = av_sample_fmt_is_planar(sample_fmt);
238  int planes = planar ? nb_channels : 1;
239  int block_align = av_get_bytes_per_sample(sample_fmt) * (planar ? 1 : nb_channels);
240  int data_size = nb_samples * block_align;
241  int fill_char = (sample_fmt == AV_SAMPLE_FMT_U8 ||
242  sample_fmt == AV_SAMPLE_FMT_U8P) ? 0x80 : 0x00;
243  int i;
244 
245  offset *= block_align;
246 
247  for (i = 0; i < planes; i++)
248  memset(audio_data[i] + offset, fill_char, data_size);
249 
250  return 0;
251 }
Generated on Sat May 25 2013 04:01:21 for FFmpeg by   doxygen 1.8.2

AltStyle によって変換されたページ (->オリジナル) /