FFmpeg
pcm.c
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1 /*
2  * PCM codecs
3  * Copyright (c) 2001 Fabrice Bellard
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * PCM codecs
25  */
26 
27 #include "config.h"
28 #include "libavutil/attributes.h"
29 #include "libavutil/float_dsp.h"
30 #include "libavutil/thread.h"
31 #include "avcodec.h"
32 #include "bytestream.h"
33 #include "internal.h"
34 #include "mathops.h"
35 #include "pcm_tablegen.h"
36 
37 static av_cold int pcm_encode_init(AVCodecContext *avctx)
38 {
39  avctx->frame_size = 0;
40 #if !CONFIG_HARDCODED_TABLES
41  switch (avctx->codec->id) {
42 #define INIT_ONCE(id, name) \
43  case AV_CODEC_ID_PCM_ ## id: \
44  if (CONFIG_PCM_ ## id ## _ENCODER) { \
45  static AVOnce init_static_once = AV_ONCE_INIT; \
46  ff_thread_once(&init_static_once, pcm_ ## name ## _tableinit); \
47  } \
48  break
49  INIT_ONCE(ALAW, alaw);
50  INIT_ONCE(MULAW, ulaw);
51  INIT_ONCE(VIDC, vidc);
52  default:
53  break;
54  }
55 #endif
56 
57  avctx->bits_per_coded_sample = av_get_bits_per_sample(avctx->codec->id);
58  avctx->block_align = avctx->channels * avctx->bits_per_coded_sample / 8;
59  avctx->bit_rate = avctx->block_align * 8LL * avctx->sample_rate;
60 
61  return 0;
62 }
63 
64 /**
65  * Write PCM samples macro
66  * @param type Datatype of native machine format
67  * @param endian bytestream_put_xxx() suffix
68  * @param src Source pointer (variable name)
69  * @param dst Destination pointer (variable name)
70  * @param n Total number of samples (variable name)
71  * @param shift Bitshift (bits)
72  * @param offset Sample value offset
73  */
74 #define ENCODE(type, endian, src, dst, n, shift, offset) \
75  samples_ ## type = (const type *) src; \
76  for (; n > 0; n--) { \
77  register type v = (*samples_ ## type++ >> shift) + offset; \
78  bytestream_put_ ## endian(&dst, v); \
79  }
80 
81 #define ENCODE_PLANAR(type, endian, dst, n, shift, offset) \
82  n /= avctx->channels; \
83  for (c = 0; c < avctx->channels; c++) { \
84  int i; \
85  samples_ ## type = (const type *) frame->extended_data[c]; \
86  for (i = n; i > 0; i--) { \
87  register type v = (*samples_ ## type++ >> shift) + offset; \
88  bytestream_put_ ## endian(&dst, v); \
89  } \
90  }
91 
92 static int pcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
93  const AVFrame *frame, int *got_packet_ptr)
94 {
95  int n, c, sample_size, v, ret;
96  const short *samples;
97  unsigned char *dst;
98  const uint8_t *samples_uint8_t;
99  const int16_t *samples_int16_t;
100  const int32_t *samples_int32_t;
101  const int64_t *samples_int64_t;
102  const uint16_t *samples_uint16_t;
103  const uint32_t *samples_uint32_t;
104 
105  sample_size = av_get_bits_per_sample(avctx->codec->id) / 8;
106  n = frame->nb_samples * avctx->channels;
107  samples = (const short *)frame->data[0];
108 
109  if ((ret = ff_alloc_packet2(avctx, avpkt, n * sample_size, n * sample_size)) < 0)
110  return ret;
111  dst = avpkt->data;
112 
113  switch (avctx->codec->id) {
114  case AV_CODEC_ID_PCM_U32LE:
115  ENCODE(uint32_t, le32, samples, dst, n, 0, 0x80000000)
116  break;
117  case AV_CODEC_ID_PCM_U32BE:
118  ENCODE(uint32_t, be32, samples, dst, n, 0, 0x80000000)
119  break;
120  case AV_CODEC_ID_PCM_S24LE:
121  ENCODE(int32_t, le24, samples, dst, n, 8, 0)
122  break;
123  case AV_CODEC_ID_PCM_S24LE_PLANAR:
124  ENCODE_PLANAR(int32_t, le24, dst, n, 8, 0)
125  break;
126  case AV_CODEC_ID_PCM_S24BE:
127  ENCODE(int32_t, be24, samples, dst, n, 8, 0)
128  break;
129  case AV_CODEC_ID_PCM_U24LE:
130  ENCODE(uint32_t, le24, samples, dst, n, 8, 0x800000)
131  break;
132  case AV_CODEC_ID_PCM_U24BE:
133  ENCODE(uint32_t, be24, samples, dst, n, 8, 0x800000)
134  break;
135  case AV_CODEC_ID_PCM_S24DAUD:
136  for (; n > 0; n--) {
137  uint32_t tmp = ff_reverse[(*samples >> 8) & 0xff] +
138  (ff_reverse[*samples & 0xff] << 8);
139  tmp <<= 4; // sync flags would go here
140  bytestream_put_be24(&dst, tmp);
141  samples++;
142  }
143  break;
144  case AV_CODEC_ID_PCM_U16LE:
145  ENCODE(uint16_t, le16, samples, dst, n, 0, 0x8000)
146  break;
147  case AV_CODEC_ID_PCM_U16BE:
148  ENCODE(uint16_t, be16, samples, dst, n, 0, 0x8000)
149  break;
150  case AV_CODEC_ID_PCM_S8:
151  ENCODE(uint8_t, byte, samples, dst, n, 0, -128)
152  break;
153  case AV_CODEC_ID_PCM_S8_PLANAR:
154  ENCODE_PLANAR(uint8_t, byte, dst, n, 0, -128)
155  break;
156 #if HAVE_BIGENDIAN
157  case AV_CODEC_ID_PCM_S64LE:
158  case AV_CODEC_ID_PCM_F64LE:
159  ENCODE(int64_t, le64, samples, dst, n, 0, 0)
160  break;
161  case AV_CODEC_ID_PCM_S32LE:
162  case AV_CODEC_ID_PCM_F32LE:
163  ENCODE(int32_t, le32, samples, dst, n, 0, 0)
164  break;
165  case AV_CODEC_ID_PCM_S32LE_PLANAR:
166  ENCODE_PLANAR(int32_t, le32, dst, n, 0, 0)
167  break;
168  case AV_CODEC_ID_PCM_S16LE:
169  ENCODE(int16_t, le16, samples, dst, n, 0, 0)
170  break;
171  case AV_CODEC_ID_PCM_S16LE_PLANAR:
172  ENCODE_PLANAR(int16_t, le16, dst, n, 0, 0)
173  break;
174  case AV_CODEC_ID_PCM_F64BE:
175  case AV_CODEC_ID_PCM_F32BE:
176  case AV_CODEC_ID_PCM_S64BE:
177  case AV_CODEC_ID_PCM_S32BE:
178  case AV_CODEC_ID_PCM_S16BE:
179 #else
180  case AV_CODEC_ID_PCM_S64BE:
181  case AV_CODEC_ID_PCM_F64BE:
182  ENCODE(int64_t, be64, samples, dst, n, 0, 0)
183  break;
184  case AV_CODEC_ID_PCM_F32BE:
185  case AV_CODEC_ID_PCM_S32BE:
186  ENCODE(int32_t, be32, samples, dst, n, 0, 0)
187  break;
188  case AV_CODEC_ID_PCM_S16BE:
189  ENCODE(int16_t, be16, samples, dst, n, 0, 0)
190  break;
191  case AV_CODEC_ID_PCM_S16BE_PLANAR:
192  ENCODE_PLANAR(int16_t, be16, dst, n, 0, 0)
193  break;
194  case AV_CODEC_ID_PCM_F64LE:
195  case AV_CODEC_ID_PCM_F32LE:
196  case AV_CODEC_ID_PCM_S64LE:
197  case AV_CODEC_ID_PCM_S32LE:
198  case AV_CODEC_ID_PCM_S16LE:
199 #endif /* HAVE_BIGENDIAN */
200  case AV_CODEC_ID_PCM_U8:
201  memcpy(dst, samples, n * sample_size);
202  break;
203 #if HAVE_BIGENDIAN
204  case AV_CODEC_ID_PCM_S16BE_PLANAR:
205 #else
206  case AV_CODEC_ID_PCM_S16LE_PLANAR:
207  case AV_CODEC_ID_PCM_S32LE_PLANAR:
208 #endif /* HAVE_BIGENDIAN */
209  n /= avctx->channels;
210  for (c = 0; c < avctx->channels; c++) {
211  const uint8_t *src = frame->extended_data[c];
212  bytestream_put_buffer(&dst, src, n * sample_size);
213  }
214  break;
215  case AV_CODEC_ID_PCM_ALAW:
216  for (; n > 0; n--) {
217  v = *samples++;
218  *dst++ = linear_to_alaw[(v + 32768) >> 2];
219  }
220  break;
221  case AV_CODEC_ID_PCM_MULAW:
222  for (; n > 0; n--) {
223  v = *samples++;
224  *dst++ = linear_to_ulaw[(v + 32768) >> 2];
225  }
226  break;
227  case AV_CODEC_ID_PCM_VIDC:
228  for (; n > 0; n--) {
229  v = *samples++;
230  *dst++ = linear_to_vidc[(v + 32768) >> 2];
231  }
232  break;
233  default:
234  return -1;
235  }
236 
237  *got_packet_ptr = 1;
238  return 0;
239 }
240 
241 typedef struct PCMDecode {
242  short table[256];
243  void (*vector_fmul_scalar)(float *dst, const float *src, float mul,
244  int len);
245  float scale;
246 } PCMDecode;
247 
248 static av_cold int pcm_decode_init(AVCodecContext *avctx)
249 {
250  PCMDecode *s = avctx->priv_data;
251  AVFloatDSPContext *fdsp;
252  int i;
253 
254  if (avctx->channels <= 0) {
255  av_log(avctx, AV_LOG_ERROR, "PCM channels out of bounds\n");
256  return AVERROR(EINVAL);
257  }
258 
259  switch (avctx->codec_id) {
260  case AV_CODEC_ID_PCM_ALAW:
261  for (i = 0; i < 256; i++)
262  s->table[i] = alaw2linear(i);
263  break;
264  case AV_CODEC_ID_PCM_MULAW:
265  for (i = 0; i < 256; i++)
266  s->table[i] = ulaw2linear(i);
267  break;
268  case AV_CODEC_ID_PCM_VIDC:
269  for (i = 0; i < 256; i++)
270  s->table[i] = vidc2linear(i);
271  break;
272  case AV_CODEC_ID_PCM_F16LE:
273  case AV_CODEC_ID_PCM_F24LE:
274  if (avctx->bits_per_coded_sample < 1 || avctx->bits_per_coded_sample > 24)
275  return AVERROR_INVALIDDATA;
276 
277  s->scale = 1. / (1 << (avctx->bits_per_coded_sample - 1));
278  fdsp = avpriv_float_dsp_alloc(0);
279  if (!fdsp)
280  return AVERROR(ENOMEM);
281  s->vector_fmul_scalar = fdsp->vector_fmul_scalar;
282  av_free(fdsp);
283  break;
284  default:
285  break;
286  }
287 
288  avctx->sample_fmt = avctx->codec->sample_fmts[0];
289 
290  if (avctx->sample_fmt == AV_SAMPLE_FMT_S32)
291  avctx->bits_per_raw_sample = av_get_bits_per_sample(avctx->codec_id);
292 
293  return 0;
294 }
295 
296 /**
297  * Read PCM samples macro
298  * @param size Data size of native machine format
299  * @param endian bytestream_get_xxx() endian suffix
300  * @param src Source pointer (variable name)
301  * @param dst Destination pointer (variable name)
302  * @param n Total number of samples (variable name)
303  * @param shift Bitshift (bits)
304  * @param offset Sample value offset
305  */
306 #define DECODE(size, endian, src, dst, n, shift, offset) \
307  for (; n > 0; n--) { \
308  uint ## size ## _t v = bytestream_get_ ## endian(&src); \
309  AV_WN ## size ## A(dst, (uint ## size ## _t)(v - offset) << shift); \
310  dst += size / 8; \
311  }
312 
313 #define DECODE_PLANAR(size, endian, src, dst, n, shift, offset) \
314  n /= avctx->channels; \
315  for (c = 0; c < avctx->channels; c++) { \
316  int i; \
317  dst = frame->extended_data[c]; \
318  for (i = n; i > 0; i--) { \
319  uint ## size ## _t v = bytestream_get_ ## endian(&src); \
320  AV_WN ## size ## A(dst, (uint ## size ##_t)(v - offset) << shift); \
321  dst += size / 8; \
322  } \
323  }
324 
325 static int pcm_decode_frame(AVCodecContext *avctx, void *data,
326  int *got_frame_ptr, AVPacket *avpkt)
327 {
328  const uint8_t *src = avpkt->data;
329  int buf_size = avpkt->size;
330  PCMDecode *s = avctx->priv_data;
331  AVFrame *frame = data;
332  int sample_size, c, n, ret, samples_per_block;
333  uint8_t *samples;
334  int32_t *dst_int32_t;
335 
336  sample_size = av_get_bits_per_sample(avctx->codec_id) / 8;
337 
338  /* av_get_bits_per_sample returns 0 for AV_CODEC_ID_PCM_DVD */
339  samples_per_block = 1;
340  if (avctx->codec_id == AV_CODEC_ID_PCM_LXF) {
341  /* we process 40-bit blocks per channel for LXF */
342  samples_per_block = 2;
343  sample_size = 5;
344  }
345 
346  if (sample_size == 0) {
347  av_log(avctx, AV_LOG_ERROR, "Invalid sample_size\n");
348  return AVERROR(EINVAL);
349  }
350 
351  if (avctx->channels == 0) {
352  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
353  return AVERROR(EINVAL);
354  }
355 
356  if (avctx->codec_id != avctx->codec->id) {
357  av_log(avctx, AV_LOG_ERROR, "codec ids mismatch\n");
358  return AVERROR(EINVAL);
359  }
360 
361  n = avctx->channels * sample_size;
362 
363  if (n && buf_size % n) {
364  if (buf_size < n) {
365  av_log(avctx, AV_LOG_ERROR,
366  "Invalid PCM packet, data has size %d but at least a size of %d was expected\n",
367  buf_size, n);
368  return AVERROR_INVALIDDATA;
369  } else
370  buf_size -= buf_size % n;
371  }
372 
373  n = buf_size / sample_size;
374 
375  /* get output buffer */
376  frame->nb_samples = n * samples_per_block / avctx->channels;
377  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
378  return ret;
379  samples = frame->data[0];
380 
381  switch (avctx->codec_id) {
382  case AV_CODEC_ID_PCM_U32LE:
383  DECODE(32, le32, src, samples, n, 0, 0x80000000)
384  break;
385  case AV_CODEC_ID_PCM_U32BE:
386  DECODE(32, be32, src, samples, n, 0, 0x80000000)
387  break;
388  case AV_CODEC_ID_PCM_S24LE:
389  DECODE(32, le24, src, samples, n, 8, 0)
390  break;
391  case AV_CODEC_ID_PCM_S24LE_PLANAR:
392  DECODE_PLANAR(32, le24, src, samples, n, 8, 0);
393  break;
394  case AV_CODEC_ID_PCM_S24BE:
395  DECODE(32, be24, src, samples, n, 8, 0)
396  break;
397  case AV_CODEC_ID_PCM_U24LE:
398  DECODE(32, le24, src, samples, n, 8, 0x800000)
399  break;
400  case AV_CODEC_ID_PCM_U24BE:
401  DECODE(32, be24, src, samples, n, 8, 0x800000)
402  break;
403  case AV_CODEC_ID_PCM_S24DAUD:
404  for (; n > 0; n--) {
405  uint32_t v = bytestream_get_be24(&src);
406  v >>= 4; // sync flags are here
407  AV_WN16A(samples, ff_reverse[(v >> 8) & 0xff] +
408  (ff_reverse[v & 0xff] << 8));
409  samples += 2;
410  }
411  break;
412  case AV_CODEC_ID_PCM_U16LE:
413  DECODE(16, le16, src, samples, n, 0, 0x8000)
414  break;
415  case AV_CODEC_ID_PCM_U16BE:
416  DECODE(16, be16, src, samples, n, 0, 0x8000)
417  break;
418  case AV_CODEC_ID_PCM_S8:
419  for (; n > 0; n--)
420  *samples++ = *src++ + 128;
421  break;
422  case AV_CODEC_ID_PCM_S8_PLANAR:
423  n /= avctx->channels;
424  for (c = 0; c < avctx->channels; c++) {
425  int i;
426  samples = frame->extended_data[c];
427  for (i = n; i > 0; i--)
428  *samples++ = *src++ + 128;
429  }
430  break;
431 #if HAVE_BIGENDIAN
432  case AV_CODEC_ID_PCM_S64LE:
433  case AV_CODEC_ID_PCM_F64LE:
434  DECODE(64, le64, src, samples, n, 0, 0)
435  break;
436  case AV_CODEC_ID_PCM_S32LE:
437  case AV_CODEC_ID_PCM_F32LE:
438  case AV_CODEC_ID_PCM_F24LE:
439  case AV_CODEC_ID_PCM_F16LE:
440  DECODE(32, le32, src, samples, n, 0, 0)
441  break;
442  case AV_CODEC_ID_PCM_S32LE_PLANAR:
443  DECODE_PLANAR(32, le32, src, samples, n, 0, 0);
444  break;
445  case AV_CODEC_ID_PCM_S16LE:
446  DECODE(16, le16, src, samples, n, 0, 0)
447  break;
448  case AV_CODEC_ID_PCM_S16LE_PLANAR:
449  DECODE_PLANAR(16, le16, src, samples, n, 0, 0);
450  break;
451  case AV_CODEC_ID_PCM_F64BE:
452  case AV_CODEC_ID_PCM_F32BE:
453  case AV_CODEC_ID_PCM_S64BE:
454  case AV_CODEC_ID_PCM_S32BE:
455  case AV_CODEC_ID_PCM_S16BE:
456 #else
457  case AV_CODEC_ID_PCM_S64BE:
458  case AV_CODEC_ID_PCM_F64BE:
459  DECODE(64, be64, src, samples, n, 0, 0)
460  break;
461  case AV_CODEC_ID_PCM_F32BE:
462  case AV_CODEC_ID_PCM_S32BE:
463  DECODE(32, be32, src, samples, n, 0, 0)
464  break;
465  case AV_CODEC_ID_PCM_S16BE:
466  DECODE(16, be16, src, samples, n, 0, 0)
467  break;
468  case AV_CODEC_ID_PCM_S16BE_PLANAR:
469  DECODE_PLANAR(16, be16, src, samples, n, 0, 0);
470  break;
471  case AV_CODEC_ID_PCM_F64LE:
472  case AV_CODEC_ID_PCM_F32LE:
473  case AV_CODEC_ID_PCM_F24LE:
474  case AV_CODEC_ID_PCM_F16LE:
475  case AV_CODEC_ID_PCM_S64LE:
476  case AV_CODEC_ID_PCM_S32LE:
477  case AV_CODEC_ID_PCM_S16LE:
478 #endif /* HAVE_BIGENDIAN */
479  case AV_CODEC_ID_PCM_U8:
480  memcpy(samples, src, n * sample_size);
481  break;
482 #if HAVE_BIGENDIAN
483  case AV_CODEC_ID_PCM_S16BE_PLANAR:
484 #else
485  case AV_CODEC_ID_PCM_S16LE_PLANAR:
486  case AV_CODEC_ID_PCM_S32LE_PLANAR:
487 #endif /* HAVE_BIGENDIAN */
488  n /= avctx->channels;
489  for (c = 0; c < avctx->channels; c++) {
490  samples = frame->extended_data[c];
491  bytestream_get_buffer(&src, samples, n * sample_size);
492  }
493  break;
494  case AV_CODEC_ID_PCM_ALAW:
495  case AV_CODEC_ID_PCM_MULAW:
496  case AV_CODEC_ID_PCM_VIDC:
497  for (; n > 0; n--) {
498  AV_WN16A(samples, s->table[*src++]);
499  samples += 2;
500  }
501  break;
502  case AV_CODEC_ID_PCM_LXF:
503  {
504  int i;
505  n /= avctx->channels;
506  for (c = 0; c < avctx->channels; c++) {
507  dst_int32_t = (int32_t *)frame->extended_data[c];
508  for (i = 0; i < n; i++) {
509  // extract low 20 bits and expand to 32 bits
510  *dst_int32_t++ = ((uint32_t)src[2]<<28) |
511  (src[1] << 20) |
512  (src[0] << 12) |
513  ((src[2] & 0x0F) << 8) |
514  src[1];
515  // extract high 20 bits and expand to 32 bits
516  *dst_int32_t++ = ((uint32_t)src[4]<<24) |
517  (src[3] << 16) |
518  ((src[2] & 0xF0) << 8) |
519  (src[4] << 4) |
520  (src[3] >> 4);
521  src += 5;
522  }
523  }
524  break;
525  }
526  default:
527  return -1;
528  }
529 
530  if (avctx->codec_id == AV_CODEC_ID_PCM_F16LE ||
531  avctx->codec_id == AV_CODEC_ID_PCM_F24LE) {
532  s->vector_fmul_scalar((float *)frame->extended_data[0],
533  (const float *)frame->extended_data[0],
534  s->scale, FFALIGN(frame->nb_samples * avctx->channels, 4));
535  emms_c();
536  }
537 
538  *got_frame_ptr = 1;
539 
540  return buf_size;
541 }
542 
543 #define PCM_ENCODER_0(id_, sample_fmt_, name_, long_name_)
544 #define PCM_ENCODER_1(id_, sample_fmt_, name_, long_name_) \
545 AVCodec ff_ ## name_ ## _encoder = { \
546  .name = #name_, \
547  .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
548  .type = AVMEDIA_TYPE_AUDIO, \
549  .id = AV_CODEC_ID_ ## id_, \
550  .init = pcm_encode_init, \
551  .encode2 = pcm_encode_frame, \
552  .capabilities = AV_CODEC_CAP_VARIABLE_FRAME_SIZE, \
553  .sample_fmts = (const enum AVSampleFormat[]){ sample_fmt_, \
554  AV_SAMPLE_FMT_NONE }, \
555  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, \
556 }
557 
558 #define PCM_ENCODER_2(cf, id, sample_fmt, name, long_name) \
559  PCM_ENCODER_ ## cf(id, sample_fmt, name, long_name)
560 #define PCM_ENCODER_3(cf, id, sample_fmt, name, long_name) \
561  PCM_ENCODER_2(cf, id, sample_fmt, name, long_name)
562 #define PCM_ENCODER(id, sample_fmt, name, long_name) \
563  PCM_ENCODER_3(CONFIG_ ## id ## _ENCODER, id, sample_fmt, name, long_name)
564 
565 #define PCM_DECODER_0(id, sample_fmt, name, long_name)
566 #define PCM_DECODER_1(id_, sample_fmt_, name_, long_name_) \
567 AVCodec ff_ ## name_ ## _decoder = { \
568  .name = #name_, \
569  .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
570  .type = AVMEDIA_TYPE_AUDIO, \
571  .id = AV_CODEC_ID_ ## id_, \
572  .priv_data_size = sizeof(PCMDecode), \
573  .init = pcm_decode_init, \
574  .decode = pcm_decode_frame, \
575  .capabilities = AV_CODEC_CAP_DR1, \
576  .sample_fmts = (const enum AVSampleFormat[]){ sample_fmt_, \
577  AV_SAMPLE_FMT_NONE }, \
578  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, \
579 }
580 
581 #define PCM_DECODER_2(cf, id, sample_fmt, name, long_name) \
582  PCM_DECODER_ ## cf(id, sample_fmt, name, long_name)
583 #define PCM_DECODER_3(cf, id, sample_fmt, name, long_name) \
584  PCM_DECODER_2(cf, id, sample_fmt, name, long_name)
585 #define PCM_DECODER(id, sample_fmt, name, long_name) \
586  PCM_DECODER_3(CONFIG_ ## id ## _DECODER, id, sample_fmt, name, long_name)
587 
588 #define PCM_CODEC(id, sample_fmt_, name, long_name_) \
589  PCM_ENCODER(id, sample_fmt_, name, long_name_); \
590  PCM_DECODER(id, sample_fmt_, name, long_name_)
591 
592 /* Note: Do not forget to add new entries to the Makefile as well. */
593 PCM_CODEC (PCM_ALAW, AV_SAMPLE_FMT_S16, pcm_alaw, "PCM A-law / G.711 A-law");
594 PCM_DECODER(PCM_F16LE, AV_SAMPLE_FMT_FLT, pcm_f16le, "PCM 16.8 floating point little-endian");
595 PCM_DECODER(PCM_F24LE, AV_SAMPLE_FMT_FLT, pcm_f24le, "PCM 24.0 floating point little-endian");
596 PCM_CODEC (PCM_F32BE, AV_SAMPLE_FMT_FLT, pcm_f32be, "PCM 32-bit floating point big-endian");
597 PCM_CODEC (PCM_F32LE, AV_SAMPLE_FMT_FLT, pcm_f32le, "PCM 32-bit floating point little-endian");
598 PCM_CODEC (PCM_F64BE, AV_SAMPLE_FMT_DBL, pcm_f64be, "PCM 64-bit floating point big-endian");
599 PCM_CODEC (PCM_F64LE, AV_SAMPLE_FMT_DBL, pcm_f64le, "PCM 64-bit floating point little-endian");
600 PCM_DECODER(PCM_LXF, AV_SAMPLE_FMT_S32P,pcm_lxf, "PCM signed 20-bit little-endian planar");
601 PCM_CODEC (PCM_MULAW, AV_SAMPLE_FMT_S16, pcm_mulaw, "PCM mu-law / G.711 mu-law");
602 PCM_CODEC (PCM_S8, AV_SAMPLE_FMT_U8, pcm_s8, "PCM signed 8-bit");
603 PCM_CODEC (PCM_S8_PLANAR, AV_SAMPLE_FMT_U8P, pcm_s8_planar, "PCM signed 8-bit planar");
604 PCM_CODEC (PCM_S16BE, AV_SAMPLE_FMT_S16, pcm_s16be, "PCM signed 16-bit big-endian");
605 PCM_CODEC (PCM_S16BE_PLANAR, AV_SAMPLE_FMT_S16P,pcm_s16be_planar, "PCM signed 16-bit big-endian planar");
606 PCM_CODEC (PCM_S16LE, AV_SAMPLE_FMT_S16, pcm_s16le, "PCM signed 16-bit little-endian");
607 PCM_CODEC (PCM_S16LE_PLANAR, AV_SAMPLE_FMT_S16P,pcm_s16le_planar, "PCM signed 16-bit little-endian planar");
608 PCM_CODEC (PCM_S24BE, AV_SAMPLE_FMT_S32, pcm_s24be, "PCM signed 24-bit big-endian");
609 PCM_CODEC (PCM_S24DAUD, AV_SAMPLE_FMT_S16, pcm_s24daud, "PCM D-Cinema audio signed 24-bit");
610 PCM_CODEC (PCM_S24LE, AV_SAMPLE_FMT_S32, pcm_s24le, "PCM signed 24-bit little-endian");
611 PCM_CODEC (PCM_S24LE_PLANAR, AV_SAMPLE_FMT_S32P,pcm_s24le_planar, "PCM signed 24-bit little-endian planar");
612 PCM_CODEC (PCM_S32BE, AV_SAMPLE_FMT_S32, pcm_s32be, "PCM signed 32-bit big-endian");
613 PCM_CODEC (PCM_S32LE, AV_SAMPLE_FMT_S32, pcm_s32le, "PCM signed 32-bit little-endian");
614 PCM_CODEC (PCM_S32LE_PLANAR, AV_SAMPLE_FMT_S32P,pcm_s32le_planar, "PCM signed 32-bit little-endian planar");
615 PCM_CODEC (PCM_U8, AV_SAMPLE_FMT_U8, pcm_u8, "PCM unsigned 8-bit");
616 PCM_CODEC (PCM_U16BE, AV_SAMPLE_FMT_S16, pcm_u16be, "PCM unsigned 16-bit big-endian");
617 PCM_CODEC (PCM_U16LE, AV_SAMPLE_FMT_S16, pcm_u16le, "PCM unsigned 16-bit little-endian");
618 PCM_CODEC (PCM_U24BE, AV_SAMPLE_FMT_S32, pcm_u24be, "PCM unsigned 24-bit big-endian");
619 PCM_CODEC (PCM_U24LE, AV_SAMPLE_FMT_S32, pcm_u24le, "PCM unsigned 24-bit little-endian");
620 PCM_CODEC (PCM_U32BE, AV_SAMPLE_FMT_S32, pcm_u32be, "PCM unsigned 32-bit big-endian");
621 PCM_CODEC (PCM_U32LE, AV_SAMPLE_FMT_S32, pcm_u32le, "PCM unsigned 32-bit little-endian");
622 PCM_CODEC (PCM_S64BE, AV_SAMPLE_FMT_S64, pcm_s64be, "PCM signed 64-bit big-endian");
623 PCM_CODEC (PCM_S64LE, AV_SAMPLE_FMT_S64, pcm_s64le, "PCM signed 64-bit little-endian");
624 PCM_CODEC (PCM_VIDC, AV_SAMPLE_FMT_S16, pcm_vidc, "PCM Archimedes VIDC");
AV_CODEC_ID_PCM_S24BE
Definition: codec_id.h:324
AVCodecContext::codec
const struct AVCodec * codec
Definition: avcodec.h:540
vidc2linear
static av_cold int vidc2linear(unsigned char u_val)
Definition: pcm_tablegen.h:78
AV_CODEC_ID_PCM_S24DAUD
Definition: codec_id.h:327
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
AV_CODEC_ID_PCM_S8
Definition: codec_id.h:315
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:314
data
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
AV_CODEC_ID_PCM_U16LE
Definition: codec_id.h:313
AV_CODEC_ID_PCM_S32LE_PLANAR
Definition: codec_id.h:340
ret
ret
Definition: filter_design.txt:187
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:581
ff_reverse
const uint8_t ff_reverse[256]
Definition: reverse.c:23
DECODE_PLANAR
#define DECODE_PLANAR(size, endian, src, dst, n, shift, offset)
Definition: pcm.c:313
ulaw2linear
static av_cold int ulaw2linear(unsigned char u_val)
Definition: pcm_tablegen.h:61
pcm_encode_frame
static int pcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: pcm.c:92
AVPacket::size
int size
Definition: packet.h:364
float_dsp.h
PCMDecode::vector_fmul_scalar
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Definition: pcm.c:243
AVCodecContext::bits_per_raw_sample
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:1762
AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:63
AVCodecContext::block_align
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:1228
AV_CODEC_ID_PCM_S24LE_PLANAR
Definition: codec_id.h:339
attributes.h
Macro definitions for various function/variable attributes.
le32
uint64_t_TMPL AV_WL64 unsigned int_TMPL le32
Definition: bytestream.h:91
PCMDecode
Definition: pcm.c:241
ff_alloc_packet2
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: encode.c:33
PCMDecode::table
short table[256]
Definition: pcm.c:242
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1199
uint8_t
uint8_t
Definition: audio_convert.c:194
av_cold
#define av_cold
Definition: attributes.h:88
AV_SAMPLE_FMT_U8
AV_SAMPLE_FMT_U8
Definition: audio_convert.c:194
AV_CODEC_ID_PCM_S8_PLANAR
Definition: codec_id.h:338
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
AV_CODEC_ID_PCM_VIDC
Definition: codec_id.h:347
AV_CODEC_ID_PCM_S64LE
Definition: codec_id.h:343
AV_CODEC_ID_PCM_U24LE
Definition: codec_id.h:325
AVPacket::data
uint8_t * data
Definition: packet.h:363
linear_to_ulaw
static uint8_t linear_to_ulaw[16384]
Definition: pcm_tablegen.h:100
AVCodecContext::bits_per_coded_sample
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
Definition: avcodec.h:1755
AV_SAMPLE_FMT_S32
signed 32 bits
Definition: samplefmt.h:62
be24
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL be24
Definition: bytestream.h:91
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:48
AV_CODEC_ID_PCM_S16LE_PLANAR
Definition: codec_id.h:329
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
src
#define src
Definition: vp8dsp.c:255
pcm_decode_frame
static int pcm_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: pcm.c:325
AVCodec::id
enum AVCodecID id
Definition: codec.h:204
PCM_CODEC
#define PCM_CODEC(id, sample_fmt_, name, long_name_)
Definition: pcm.c:588
AV_CODEC_ID_PCM_ALAW
Definition: codec_id.h:318
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
av_get_bits_per_sample
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
Definition: utils.c:1595
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
DECODE
#define DECODE(size, endian, src, dst, n, shift, offset)
Read PCM samples macro.
Definition: pcm.c:306
be64
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL be64
Definition: bytestream.h:91
ENCODE_PLANAR
#define ENCODE_PLANAR(type, endian, dst, n, shift, offset)
Definition: pcm.c:81
mul
static float mul(float src0, float src1)
Definition: dnn_backend_native_layer_mathbinary.c:40
be16
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL be16
Definition: bytestream.h:91
ENCODE
#define ENCODE(type, endian, src, dst, n, shift, offset)
Write PCM samples macro.
Definition: pcm.c:74
le24
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL le24
Definition: bytestream.h:91
pcm_encode_init
static av_cold int pcm_encode_init(AVCodecContext *avctx)
Definition: pcm.c:37
AV_CODEC_ID_PCM_U24BE
Definition: codec_id.h:326
alaw2linear
static av_cold int alaw2linear(unsigned char a_val)
Definition: pcm_tablegen.h:46
AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
le64
uint64_t_TMPL le64
Definition: bytestream.h:91
AV_SAMPLE_FMT_S64
signed 64 bits
Definition: samplefmt.h:71
AV_CODEC_ID_PCM_S64BE
Definition: codec_id.h:344
void
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
int32_t
int32_t
Definition: audio_convert.c:194
AV_CODEC_ID_PCM_U16BE
Definition: codec_id.h:314
PCMDecode::scale
float scale
Definition: pcm.c:245
AV_CODEC_ID_PCM_F64BE
Definition: codec_id.h:333
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:255
s
#define s(width, name)
Definition: cbs_vp9.c:257
AV_SAMPLE_FMT_DBL
double
Definition: samplefmt.h:64
AV_SAMPLE_FMT_U8P
unsigned 8 bits, planar
Definition: samplefmt.h:66
AV_WN16A
#define AV_WN16A(p, v)
Definition: intreadwrite.h:534
bytestream_get_buffer
static av_always_inline unsigned int bytestream_get_buffer(const uint8_t **b, uint8_t *dst, unsigned int size)
Definition: bytestream.h:363
pcm_tablegen.h
AV_CODEC_ID_PCM_S16BE
Definition: codec_id.h:312
if
if(ret)
Definition: filter_design.txt:179
AV_CODEC_ID_PCM_U8
Definition: codec_id.h:316
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1211
avcodec.h
Libavcodec external API header.
AVCodecContext::codec_id
enum AVCodecID codec_id
Definition: avcodec.h:541
byte
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_WB32 unsigned int_TMPL AV_WB24 unsigned int_TMPL AV_WB16 unsigned int_TMPL byte
Definition: bytestream.h:91
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1191
AVCodecContext
main external API structure.
Definition: avcodec.h:531
AV_CODEC_ID_PCM_MULAW
Definition: codec_id.h:317
AV_CODEC_ID_PCM_U32LE
Definition: codec_id.h:321
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1876
AV_CODEC_ID_PCM_F32BE
Definition: codec_id.h:331
be32
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL be32
Definition: bytestream.h:91
AV_CODEC_ID_PCM_LXF
Definition: codec_id.h:336
le16
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL le16
Definition: bytestream.h:91
AVFloatDSPContext::vector_fmul_scalar
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
Definition: float_dsp.h:85
pcm_decode_init
static av_cold int pcm_decode_init(AVCodecContext *avctx)
Definition: pcm.c:248
AV_CODEC_ID_PCM_S24LE
Definition: codec_id.h:323
PCM_DECODER
#define PCM_DECODER(id, sample_fmt, name, long_name)
Definition: pcm.c:585
AV_CODEC_ID_PCM_F64LE
Definition: codec_id.h:334
AV_CODEC_ID_PCM_F24LE
Definition: codec_id.h:346
AV_CODEC_ID_PCM_F16LE
Definition: codec_id.h:345
AVFrame::data
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:328
AV_CODEC_ID_PCM_F32LE
Definition: codec_id.h:332
internal.h
common internal api header.
AV_CODEC_ID_PCM_U32BE
Definition: codec_id.h:322
AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
INIT_ONCE
#define INIT_ONCE(id, name)
AV_CODEC_ID_PCM_S16LE
Definition: codec_id.h:311
linear_to_vidc
static uint8_t linear_to_vidc[16384]
Definition: pcm_tablegen.h:101
linear_to_alaw
static uint8_t linear_to_alaw[16384]
Definition: pcm_tablegen.h:99
bytestream_put_buffer
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
Definition: bytestream.h:372
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:558
av_free
#define av_free(p)
Definition: tableprint_vlc.h:34
len
int len
Definition: vorbis_enc_data.h:452
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:1192
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:2
AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:67
AV_CODEC_ID_PCM_S32LE
Definition: codec_id.h:319
AVCodec::sample_fmts
enum AVSampleFormat * sample_fmts
array of supported sample formats, or NULL if unknown, array is terminated by -1
Definition: codec.h:213
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:361
AVPacket
This structure stores compressed data.
Definition: packet.h:340
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:380
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
i
int i
Definition: input.c:407
AV_CODEC_ID_PCM_S16BE_PLANAR
Definition: codec_id.h:341
AV_CODEC_ID_PCM_S32BE
Definition: codec_id.h:320
tmp
static uint8_t tmp[11]
Definition: aes_ctr.c:27
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