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 "libavutil/attributes.h"
28 #include "libavutil/float_dsp.h"
29 #include "avcodec.h"
30 #include "bytestream.h"
31 #include "internal.h"
32 #include "mathops.h"
33 #include "pcm_tablegen.h"
34 
36 {
37  avctx->frame_size = 0;
38  switch (avctx->codec->id) {
41  break;
44  break;
47  break;
48  default:
49  break;
50  }
51 
53  avctx->block_align = avctx->channels * avctx->bits_per_coded_sample / 8;
54  avctx->bit_rate = avctx->block_align * 8LL * avctx->sample_rate;
55 
56  return 0;
57 }
58 
59 /**
60  * Write PCM samples macro
61  * @param type Datatype of native machine format
62  * @param endian bytestream_put_xxx() suffix
63  * @param src Source pointer (variable name)
64  * @param dst Destination pointer (variable name)
65  * @param n Total number of samples (variable name)
66  * @param shift Bitshift (bits)
67  * @param offset Sample value offset
68  */
69 #define ENCODE(type, endian, src, dst, n, shift, offset) \
70  samples_ ## type = (const type *) src; \
71  for (; n > 0; n--) { \
72  register type v = (*samples_ ## type++ >> shift) + offset; \
73  bytestream_put_ ## endian(&dst, v); \
74  }
75 
76 #define ENCODE_PLANAR(type, endian, dst, n, shift, offset) \
77  n /= avctx->channels; \
78  for (c = 0; c < avctx->channels; c++) { \
79  int i; \
80  samples_ ## type = (const type *) frame->extended_data[c]; \
81  for (i = n; i > 0; i--) { \
82  register type v = (*samples_ ## type++ >> shift) + offset; \
83  bytestream_put_ ## endian(&dst, v); \
84  } \
85  }
86 
87 static int pcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
88  const AVFrame *frame, int *got_packet_ptr)
89 {
90  int n, c, sample_size, v, ret;
91  const short *samples;
92  unsigned char *dst;
93  const uint8_t *samples_uint8_t;
94  const int16_t *samples_int16_t;
95  const int32_t *samples_int32_t;
96  const int64_t *samples_int64_t;
97  const uint16_t *samples_uint16_t;
98  const uint32_t *samples_uint32_t;
99 
100  sample_size = av_get_bits_per_sample(avctx->codec->id) / 8;
101  n = frame->nb_samples * avctx->channels;
102  samples = (const short *)frame->data[0];
103 
104  if ((ret = ff_alloc_packet2(avctx, avpkt, n * sample_size, n * sample_size)) < 0)
105  return ret;
106  dst = avpkt->data;
107 
108  switch (avctx->codec->id) {
110  ENCODE(uint32_t, le32, samples, dst, n, 0, 0x80000000)
111  break;
113  ENCODE(uint32_t, be32, samples, dst, n, 0, 0x80000000)
114  break;
116  ENCODE(int32_t, le24, samples, dst, n, 8, 0)
117  break;
119  ENCODE_PLANAR(int32_t, le24, dst, n, 8, 0)
120  break;
122  ENCODE(int32_t, be24, samples, dst, n, 8, 0)
123  break;
125  ENCODE(uint32_t, le24, samples, dst, n, 8, 0x800000)
126  break;
128  ENCODE(uint32_t, be24, samples, dst, n, 8, 0x800000)
129  break;
131  for (; n > 0; n--) {
132  uint32_t tmp = ff_reverse[(*samples >> 8) & 0xff] +
133  (ff_reverse[*samples & 0xff] << 8);
134  tmp <<= 4; // sync flags would go here
135  bytestream_put_be24(&dst, tmp);
136  samples++;
137  }
138  break;
140  ENCODE(uint16_t, le16, samples, dst, n, 0, 0x8000)
141  break;
143  ENCODE(uint16_t, be16, samples, dst, n, 0, 0x8000)
144  break;
145  case AV_CODEC_ID_PCM_S8:
146  ENCODE(uint8_t, byte, samples, dst, n, 0, -128)
147  break;
149  ENCODE_PLANAR(uint8_t, byte, dst, n, 0, -128)
150  break;
151 #if HAVE_BIGENDIAN
154  ENCODE(int64_t, le64, samples, dst, n, 0, 0)
155  break;
158  ENCODE(int32_t, le32, samples, dst, n, 0, 0)
159  break;
161  ENCODE_PLANAR(int32_t, le32, dst, n, 0, 0)
162  break;
164  ENCODE(int16_t, le16, samples, dst, n, 0, 0)
165  break;
167  ENCODE_PLANAR(int16_t, le16, dst, n, 0, 0)
168  break;
174 #else
177  ENCODE(int64_t, be64, samples, dst, n, 0, 0)
178  break;
181  ENCODE(int32_t, be32, samples, dst, n, 0, 0)
182  break;
184  ENCODE(int16_t, be16, samples, dst, n, 0, 0)
185  break;
187  ENCODE_PLANAR(int16_t, be16, dst, n, 0, 0)
188  break;
194 #endif /* HAVE_BIGENDIAN */
195  case AV_CODEC_ID_PCM_U8:
196  memcpy(dst, samples, n * sample_size);
197  break;
198 #if HAVE_BIGENDIAN
200 #else
203 #endif /* HAVE_BIGENDIAN */
204  n /= avctx->channels;
205  for (c = 0; c < avctx->channels; c++) {
206  const uint8_t *src = frame->extended_data[c];
207  bytestream_put_buffer(&dst, src, n * sample_size);
208  }
209  break;
211  for (; n > 0; n--) {
212  v = *samples++;
213  *dst++ = linear_to_alaw[(v + 32768) >> 2];
214  }
215  break;
217  for (; n > 0; n--) {
218  v = *samples++;
219  *dst++ = linear_to_ulaw[(v + 32768) >> 2];
220  }
221  break;
223  for (; n > 0; n--) {
224  v = *samples++;
225  *dst++ = linear_to_vidc[(v + 32768) >> 2];
226  }
227  break;
228  default:
229  return -1;
230  }
231 
232  *got_packet_ptr = 1;
233  return 0;
234 }
235 
236 typedef struct PCMDecode {
237  short table[256];
238  void (*vector_fmul_scalar)(float *dst, const float *src, float mul,
239  int len);
240  float scale;
241 } PCMDecode;
242 
244 {
245  PCMDecode *s = avctx->priv_data;
246  AVFloatDSPContext *fdsp;
247  int i;
248 
249  if (avctx->channels <= 0) {
250  av_log(avctx, AV_LOG_ERROR, "PCM channels out of bounds\n");
251  return AVERROR(EINVAL);
252  }
253 
254  switch (avctx->codec_id) {
256  for (i = 0; i < 256; i++)
257  s->table[i] = alaw2linear(i);
258  break;
260  for (i = 0; i < 256; i++)
261  s->table[i] = ulaw2linear(i);
262  break;
264  for (i = 0; i < 256; i++)
265  s->table[i] = vidc2linear(i);
266  break;
269  if (avctx->bits_per_coded_sample < 1 || avctx->bits_per_coded_sample > 24)
270  return AVERROR_INVALIDDATA;
271 
272  s->scale = 1. / (1 << (avctx->bits_per_coded_sample - 1));
273  fdsp = avpriv_float_dsp_alloc(0);
274  if (!fdsp)
275  return AVERROR(ENOMEM);
277  av_free(fdsp);
278  break;
279  default:
280  break;
281  }
282 
283  avctx->sample_fmt = avctx->codec->sample_fmts[0];
284 
285  if (avctx->sample_fmt == AV_SAMPLE_FMT_S32)
287 
288  return 0;
289 }
290 
291 /**
292  * Read PCM samples macro
293  * @param size Data size of native machine format
294  * @param endian bytestream_get_xxx() endian suffix
295  * @param src Source pointer (variable name)
296  * @param dst Destination pointer (variable name)
297  * @param n Total number of samples (variable name)
298  * @param shift Bitshift (bits)
299  * @param offset Sample value offset
300  */
301 #define DECODE(size, endian, src, dst, n, shift, offset) \
302  for (; n > 0; n--) { \
303  uint ## size ## _t v = bytestream_get_ ## endian(&src); \
304  AV_WN ## size ## A(dst, (uint ## size ## _t)(v - offset) << shift); \
305  dst += size / 8; \
306  }
307 
308 #define DECODE_PLANAR(size, endian, src, dst, n, shift, offset) \
309  n /= avctx->channels; \
310  for (c = 0; c < avctx->channels; c++) { \
311  int i; \
312  dst = frame->extended_data[c]; \
313  for (i = n; i > 0; i--) { \
314  uint ## size ## _t v = bytestream_get_ ## endian(&src); \
315  AV_WN ## size ## A(dst, (uint ## size ##_t)(v - offset) << shift); \
316  dst += size / 8; \
317  } \
318  }
319 
320 static int pcm_decode_frame(AVCodecContext *avctx, void *data,
321  int *got_frame_ptr, AVPacket *avpkt)
322 {
323  const uint8_t *src = avpkt->data;
324  int buf_size = avpkt->size;
325  PCMDecode *s = avctx->priv_data;
326  AVFrame *frame = data;
327  int sample_size, c, n, ret, samples_per_block;
328  uint8_t *samples;
329  int32_t *dst_int32_t;
330 
331  sample_size = av_get_bits_per_sample(avctx->codec_id) / 8;
332 
333  /* av_get_bits_per_sample returns 0 for AV_CODEC_ID_PCM_DVD */
334  samples_per_block = 1;
335  if (avctx->codec_id == AV_CODEC_ID_PCM_LXF) {
336  /* we process 40-bit blocks per channel for LXF */
337  samples_per_block = 2;
338  sample_size = 5;
339  }
340 
341  if (sample_size == 0) {
342  av_log(avctx, AV_LOG_ERROR, "Invalid sample_size\n");
343  return AVERROR(EINVAL);
344  }
345 
346  if (avctx->channels == 0) {
347  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
348  return AVERROR(EINVAL);
349  }
350 
351  if (avctx->codec_id != avctx->codec->id) {
352  av_log(avctx, AV_LOG_ERROR, "codec ids mismatch\n");
353  return AVERROR(EINVAL);
354  }
355 
356  n = avctx->channels * sample_size;
357 
358  if (n && buf_size % n) {
359  if (buf_size < n) {
360  av_log(avctx, AV_LOG_ERROR,
361  "Invalid PCM packet, data has size %d but at least a size of %d was expected\n",
362  buf_size, n);
363  return AVERROR_INVALIDDATA;
364  } else
365  buf_size -= buf_size % n;
366  }
367 
368  n = buf_size / sample_size;
369 
370  /* get output buffer */
371  frame->nb_samples = n * samples_per_block / avctx->channels;
372  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
373  return ret;
374  samples = frame->data[0];
375 
376  switch (avctx->codec_id) {
378  DECODE(32, le32, src, samples, n, 0, 0x80000000)
379  break;
381  DECODE(32, be32, src, samples, n, 0, 0x80000000)
382  break;
384  DECODE(32, le24, src, samples, n, 8, 0)
385  break;
387  DECODE_PLANAR(32, le24, src, samples, n, 8, 0);
388  break;
390  DECODE(32, be24, src, samples, n, 8, 0)
391  break;
393  DECODE(32, le24, src, samples, n, 8, 0x800000)
394  break;
396  DECODE(32, be24, src, samples, n, 8, 0x800000)
397  break;
399  for (; n > 0; n--) {
400  uint32_t v = bytestream_get_be24(&src);
401  v >>= 4; // sync flags are here
402  AV_WN16A(samples, ff_reverse[(v >> 8) & 0xff] +
403  (ff_reverse[v & 0xff] << 8));
404  samples += 2;
405  }
406  break;
408  DECODE(16, le16, src, samples, n, 0, 0x8000)
409  break;
411  DECODE(16, be16, src, samples, n, 0, 0x8000)
412  break;
413  case AV_CODEC_ID_PCM_S8:
414  for (; n > 0; n--)
415  *samples++ = *src++ + 128;
416  break;
418  n /= avctx->channels;
419  for (c = 0; c < avctx->channels; c++) {
420  int i;
421  samples = frame->extended_data[c];
422  for (i = n; i > 0; i--)
423  *samples++ = *src++ + 128;
424  }
425  break;
426 #if HAVE_BIGENDIAN
429  DECODE(64, le64, src, samples, n, 0, 0)
430  break;
435  DECODE(32, le32, src, samples, n, 0, 0)
436  break;
438  DECODE_PLANAR(32, le32, src, samples, n, 0, 0);
439  break;
441  DECODE(16, le16, src, samples, n, 0, 0)
442  break;
444  DECODE_PLANAR(16, le16, src, samples, n, 0, 0);
445  break;
451 #else
454  DECODE(64, be64, src, samples, n, 0, 0)
455  break;
458  DECODE(32, be32, src, samples, n, 0, 0)
459  break;
461  DECODE(16, be16, src, samples, n, 0, 0)
462  break;
464  DECODE_PLANAR(16, be16, src, samples, n, 0, 0);
465  break;
473 #endif /* HAVE_BIGENDIAN */
474  case AV_CODEC_ID_PCM_U8:
475  memcpy(samples, src, n * sample_size);
476  break;
477 #if HAVE_BIGENDIAN
479 #else
482 #endif /* HAVE_BIGENDIAN */
483  n /= avctx->channels;
484  for (c = 0; c < avctx->channels; c++) {
485  samples = frame->extended_data[c];
486  bytestream_get_buffer(&src, samples, n * sample_size);
487  }
488  break;
492  for (; n > 0; n--) {
493  AV_WN16A(samples, s->table[*src++]);
494  samples += 2;
495  }
496  break;
497  case AV_CODEC_ID_PCM_LXF:
498  {
499  int i;
500  n /= avctx->channels;
501  for (c = 0; c < avctx->channels; c++) {
502  dst_int32_t = (int32_t *)frame->extended_data[c];
503  for (i = 0; i < n; i++) {
504  // extract low 20 bits and expand to 32 bits
505  *dst_int32_t++ = ((uint32_t)src[2]<<28) |
506  (src[1] << 20) |
507  (src[0] << 12) |
508  ((src[2] & 0x0F) << 8) |
509  src[1];
510  // extract high 20 bits and expand to 32 bits
511  *dst_int32_t++ = ((uint32_t)src[4]<<24) |
512  (src[3] << 16) |
513  ((src[2] & 0xF0) << 8) |
514  (src[4] << 4) |
515  (src[3] >> 4);
516  src += 5;
517  }
518  }
519  break;
520  }
521  default:
522  return -1;
523  }
524 
525  if (avctx->codec_id == AV_CODEC_ID_PCM_F16LE ||
526  avctx->codec_id == AV_CODEC_ID_PCM_F24LE) {
527  s->vector_fmul_scalar((float *)frame->extended_data[0],
528  (const float *)frame->extended_data[0],
529  s->scale, FFALIGN(frame->nb_samples * avctx->channels, 4));
530  emms_c();
531  }
532 
533  *got_frame_ptr = 1;
534 
535  return buf_size;
536 }
537 
538 #define PCM_ENCODER_0(id_, sample_fmt_, name_, long_name_)
539 #define PCM_ENCODER_1(id_, sample_fmt_, name_, long_name_) \
540 AVCodec ff_ ## name_ ## _encoder = { \
541  .name = #name_, \
542  .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
543  .type = AVMEDIA_TYPE_AUDIO, \
544  .id = AV_CODEC_ID_ ## id_, \
545  .init = pcm_encode_init, \
546  .encode2 = pcm_encode_frame, \
547  .capabilities = AV_CODEC_CAP_VARIABLE_FRAME_SIZE, \
548  .sample_fmts = (const enum AVSampleFormat[]){ sample_fmt_, \
549  AV_SAMPLE_FMT_NONE }, \
550 }
551 
552 #define PCM_ENCODER_2(cf, id, sample_fmt, name, long_name) \
553  PCM_ENCODER_ ## cf(id, sample_fmt, name, long_name)
554 #define PCM_ENCODER_3(cf, id, sample_fmt, name, long_name) \
555  PCM_ENCODER_2(cf, id, sample_fmt, name, long_name)
556 #define PCM_ENCODER(id, sample_fmt, name, long_name) \
557  PCM_ENCODER_3(CONFIG_ ## id ## _ENCODER, id, sample_fmt, name, long_name)
558 
559 #define PCM_DECODER_0(id, sample_fmt, name, long_name)
560 #define PCM_DECODER_1(id_, sample_fmt_, name_, long_name_) \
561 AVCodec ff_ ## name_ ## _decoder = { \
562  .name = #name_, \
563  .long_name = NULL_IF_CONFIG_SMALL(long_name_), \
564  .type = AVMEDIA_TYPE_AUDIO, \
565  .id = AV_CODEC_ID_ ## id_, \
566  .priv_data_size = sizeof(PCMDecode), \
567  .init = pcm_decode_init, \
568  .decode = pcm_decode_frame, \
569  .capabilities = AV_CODEC_CAP_DR1, \
570  .sample_fmts = (const enum AVSampleFormat[]){ sample_fmt_, \
571  AV_SAMPLE_FMT_NONE }, \
572 }
573 
574 #define PCM_DECODER_2(cf, id, sample_fmt, name, long_name) \
575  PCM_DECODER_ ## cf(id, sample_fmt, name, long_name)
576 #define PCM_DECODER_3(cf, id, sample_fmt, name, long_name) \
577  PCM_DECODER_2(cf, id, sample_fmt, name, long_name)
578 #define PCM_DECODER(id, sample_fmt, name, long_name) \
579  PCM_DECODER_3(CONFIG_ ## id ## _DECODER, id, sample_fmt, name, long_name)
580 
581 #define PCM_CODEC(id, sample_fmt_, name, long_name_) \
582  PCM_ENCODER(id, sample_fmt_, name, long_name_); \
583  PCM_DECODER(id, sample_fmt_, name, long_name_)
584 
585 /* Note: Do not forget to add new entries to the Makefile as well. */
586 PCM_CODEC (PCM_ALAW, AV_SAMPLE_FMT_S16, pcm_alaw, "PCM A-law / G.711 A-law");
587 PCM_DECODER(PCM_F16LE, AV_SAMPLE_FMT_FLT, pcm_f16le, "PCM 16.8 floating point little-endian");
588 PCM_DECODER(PCM_F24LE, AV_SAMPLE_FMT_FLT, pcm_f24le, "PCM 24.0 floating point little-endian");
589 PCM_CODEC (PCM_F32BE, AV_SAMPLE_FMT_FLT, pcm_f32be, "PCM 32-bit floating point big-endian");
590 PCM_CODEC (PCM_F32LE, AV_SAMPLE_FMT_FLT, pcm_f32le, "PCM 32-bit floating point little-endian");
591 PCM_CODEC (PCM_F64BE, AV_SAMPLE_FMT_DBL, pcm_f64be, "PCM 64-bit floating point big-endian");
592 PCM_CODEC (PCM_F64LE, AV_SAMPLE_FMT_DBL, pcm_f64le, "PCM 64-bit floating point little-endian");
593 PCM_DECODER(PCM_LXF, AV_SAMPLE_FMT_S32P,pcm_lxf, "PCM signed 20-bit little-endian planar");
594 PCM_CODEC (PCM_MULAW, AV_SAMPLE_FMT_S16, pcm_mulaw, "PCM mu-law / G.711 mu-law");
595 PCM_CODEC (PCM_S8, AV_SAMPLE_FMT_U8, pcm_s8, "PCM signed 8-bit");
596 PCM_CODEC (PCM_S8_PLANAR, AV_SAMPLE_FMT_U8P, pcm_s8_planar, "PCM signed 8-bit planar");
597 PCM_CODEC (PCM_S16BE, AV_SAMPLE_FMT_S16, pcm_s16be, "PCM signed 16-bit big-endian");
598 PCM_CODEC (PCM_S16BE_PLANAR, AV_SAMPLE_FMT_S16P,pcm_s16be_planar, "PCM signed 16-bit big-endian planar");
599 PCM_CODEC (PCM_S16LE, AV_SAMPLE_FMT_S16, pcm_s16le, "PCM signed 16-bit little-endian");
600 PCM_CODEC (PCM_S16LE_PLANAR, AV_SAMPLE_FMT_S16P,pcm_s16le_planar, "PCM signed 16-bit little-endian planar");
601 PCM_CODEC (PCM_S24BE, AV_SAMPLE_FMT_S32, pcm_s24be, "PCM signed 24-bit big-endian");
602 PCM_CODEC (PCM_S24DAUD, AV_SAMPLE_FMT_S16, pcm_s24daud, "PCM D-Cinema audio signed 24-bit");
603 PCM_CODEC (PCM_S24LE, AV_SAMPLE_FMT_S32, pcm_s24le, "PCM signed 24-bit little-endian");
604 PCM_CODEC (PCM_S24LE_PLANAR, AV_SAMPLE_FMT_S32P,pcm_s24le_planar, "PCM signed 24-bit little-endian planar");
605 PCM_CODEC (PCM_S32BE, AV_SAMPLE_FMT_S32, pcm_s32be, "PCM signed 32-bit big-endian");
606 PCM_CODEC (PCM_S32LE, AV_SAMPLE_FMT_S32, pcm_s32le, "PCM signed 32-bit little-endian");
607 PCM_CODEC (PCM_S32LE_PLANAR, AV_SAMPLE_FMT_S32P,pcm_s32le_planar, "PCM signed 32-bit little-endian planar");
608 PCM_CODEC (PCM_U8, AV_SAMPLE_FMT_U8, pcm_u8, "PCM unsigned 8-bit");
609 PCM_CODEC (PCM_U16BE, AV_SAMPLE_FMT_S16, pcm_u16be, "PCM unsigned 16-bit big-endian");
610 PCM_CODEC (PCM_U16LE, AV_SAMPLE_FMT_S16, pcm_u16le, "PCM unsigned 16-bit little-endian");
611 PCM_CODEC (PCM_U24BE, AV_SAMPLE_FMT_S32, pcm_u24be, "PCM unsigned 24-bit big-endian");
612 PCM_CODEC (PCM_U24LE, AV_SAMPLE_FMT_S32, pcm_u24le, "PCM unsigned 24-bit little-endian");
613 PCM_CODEC (PCM_U32BE, AV_SAMPLE_FMT_S32, pcm_u32be, "PCM unsigned 32-bit big-endian");
614 PCM_CODEC (PCM_U32LE, AV_SAMPLE_FMT_S32, pcm_u32le, "PCM unsigned 32-bit little-endian");
615 PCM_CODEC (PCM_S64BE, AV_SAMPLE_FMT_S64, pcm_s64be, "PCM signed 64-bit big-endian");
616 PCM_CODEC (PCM_S64LE, AV_SAMPLE_FMT_S64, pcm_s64le, "PCM signed 64-bit little-endian");
617 PCM_CODEC (PCM_VIDC, AV_SAMPLE_FMT_S16, pcm_vidc, "PCM Archimedes VIDC");
const struct AVCodec * codec
Definition: avcodec.h:535
static av_cold int vidc2linear(unsigned char u_val)
Definition: pcm_tablegen.h:78
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:308
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
static void pcm_alaw_tableinit(void)
Definition: pcm_tablegen.h:127
int64_t bit_rate
the average bitrate
Definition: avcodec.h:576
static void pcm_ulaw_tableinit(void)
Definition: pcm_tablegen.h:132
const uint8_t ff_reverse[256]
Definition: reverse.c:23
#define DECODE_PLANAR(size, endian, src, dst, n, shift, offset)
Definition: pcm.c:308
static av_cold int ulaw2linear(unsigned char u_val)
Definition: pcm_tablegen.h:61
static int pcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: pcm.c:87
int size
Definition: packet.h:364
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Definition: pcm.c:238
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:1757
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:1223
Macro definitions for various function/variable attributes.
uint64_t_TMPL AV_WL64 unsigned int_TMPL le32
Definition: bytestream.h:91
Definition: pcm.c:236
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
short table[256]
Definition: pcm.c:237
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1194
uint8_t
#define av_cold
Definition: attributes.h:88
AV_SAMPLE_FMT_U8
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
uint8_t * data
Definition: packet.h:363
static uint8_t linear_to_ulaw[16384]
Definition: pcm_tablegen.h:100
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
Definition: avcodec.h:1750
signed 32 bits
Definition: samplefmt.h:62
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
#define FFALIGN(x, a)
Definition: macros.h:48
#define av_log(a,...)
#define src
Definition: vp8dsp.c:254
static int pcm_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: pcm.c:320
enum AVCodecID id
Definition: codec.h:204
#define PCM_CODEC(id, sample_fmt_, name, long_name_)
Definition: pcm.c:581
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
Definition: utils.c:1572
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
#define DECODE(size, endian, src, dst, n, shift, offset)
Read PCM samples macro.
Definition: pcm.c:301
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
#define ENCODE_PLANAR(type, endian, dst, n, shift, offset)
Definition: pcm.c:76
static float mul(float src0, float src1)
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
static void pcm_vidc_tableinit(void)
Definition: pcm_tablegen.h:137
#define ENCODE(type, endian, src, dst, n, shift, offset)
Write PCM samples macro.
Definition: pcm.c:69
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL le24
Definition: bytestream.h:91
static av_cold int pcm_encode_init(AVCodecContext *avctx)
Definition: pcm.c:35
static av_cold int alaw2linear(unsigned char a_val)
Definition: pcm_tablegen.h:46
signed 32 bits, planar
Definition: samplefmt.h:68
uint64_t_TMPL le64
Definition: bytestream.h:91
signed 64 bits
Definition: samplefmt.h:71
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
int32_t
float scale
Definition: pcm.c:240
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
#define s(width, name)
Definition: cbs_vp9.c:257
unsigned 8 bits, planar
Definition: samplefmt.h:66
#define AV_WN16A(p, v)
Definition: intreadwrite.h:534
static av_always_inline unsigned int bytestream_get_buffer(const uint8_t **b, uint8_t *dst, unsigned int size)
Definition: bytestream.h:363
if(ret)
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1206
Libavcodec external API header.
enum AVCodecID codec_id
Definition: avcodec.h:536
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
int sample_rate
samples per second
Definition: avcodec.h:1186
main external API structure.
Definition: avcodec.h:526
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1879
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
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL le16
Definition: bytestream.h:91
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
static av_cold int pcm_decode_init(AVCodecContext *avctx)
Definition: pcm.c:243
#define PCM_DECODER(id, sample_fmt, name, long_name)
Definition: pcm.c:578
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:322
common internal api header.
signed 16 bits
Definition: samplefmt.h:61
static uint8_t linear_to_vidc[16384]
Definition: pcm_tablegen.h:101
static uint8_t linear_to_alaw[16384]
Definition: pcm_tablegen.h:99
static av_always_inline void bytestream_put_buffer(uint8_t **b, const uint8_t *src, unsigned int size)
Definition: bytestream.h:372
void * priv_data
Definition: avcodec.h:553
#define av_free(p)
int len
int channels
number of audio channels
Definition: avcodec.h:1187
Filter the word “frame” indicates either a video frame or a group of audio samples
signed 16 bits, planar
Definition: samplefmt.h:67
enum AVSampleFormat * sample_fmts
array of supported sample formats, or NULL if unknown, array is terminated by -1
Definition: codec.h:213
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
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:355
This structure stores compressed data.
Definition: packet.h:340
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:374
for(j=16;j >0;--j)
int i
Definition: input.c:407
static uint8_t tmp[11]
Definition: aes_ctr.c:26