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 "config_components.h"
29 #include "libavutil/attributes.h"
30 #include "libavutil/float_dsp.h"
31 #include "libavutil/mem.h"
32 #include "libavutil/reverse.h"
33 #include "libavutil/thread.h"
34 #include "avcodec.h"
35 #include "bytestream.h"
36 #include "codec_internal.h"
37 #include "decode.h"
38 #include "encode.h"
39 #include "pcm_tablegen.h"
40 
42 {
43  avctx->frame_size = 0;
44 #if !CONFIG_HARDCODED_TABLES
45  switch (avctx->codec->id) {
46 #define INIT_ONCE(id, name) \
47  case AV_CODEC_ID_PCM_ ## id: \
48  if (CONFIG_PCM_ ## id ## _ENCODER) { \
49  static AVOnce init_static_once = AV_ONCE_INIT; \
50  ff_thread_once(&init_static_once, pcm_ ## name ## _tableinit); \
51  } \
52  break
53  INIT_ONCE(ALAW, alaw);
54  INIT_ONCE(MULAW, ulaw);
55  INIT_ONCE(VIDC, vidc);
56  default:
57  break;
58  }
59 #endif
60 
62  avctx->block_align = avctx->ch_layout.nb_channels * avctx->bits_per_coded_sample / 8;
63  avctx->bit_rate = avctx->block_align * 8LL * avctx->sample_rate;
64 
65  return 0;
66 }
67 
68 /**
69  * Write PCM samples macro
70  * @param type Datatype of native machine format
71  * @param endian bytestream_put_xxx() suffix
72  * @param src Source pointer (variable name)
73  * @param dst Destination pointer (variable name)
74  * @param n Total number of samples (variable name)
75  * @param shift Bitshift (bits)
76  * @param offset Sample value offset
77  */
78 #define ENCODE(type, endian, src, dst, n, shift, offset) \
79  samples_ ## type = (const type *) src; \
80  for (; n > 0; n--) { \
81  register type v = (*samples_ ## type++ >> shift) + offset; \
82  bytestream_put_ ## endian(&dst, v); \
83  }
84 
85 #define ENCODE_PLANAR(type, endian, dst, n, shift, offset) \
86  n /= avctx->ch_layout.nb_channels; \
87  for (c = 0; c < avctx->ch_layout.nb_channels; c++) { \
88  int i; \
89  samples_ ## type = (const type *) frame->extended_data[c]; \
90  for (i = n; i > 0; i--) { \
91  register type v = (*samples_ ## type++ >> shift) + offset; \
92  bytestream_put_ ## endian(&dst, v); \
93  } \
94  }
95 
96 static int pcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
97  const AVFrame *frame, int *got_packet_ptr)
98 {
99  int n, c, sample_size, v, ret;
100  const short *samples;
101  unsigned char *dst;
102  const uint8_t *samples_uint8_t;
103  const int16_t *samples_int16_t;
104  const int32_t *samples_int32_t;
105  const int64_t *samples_int64_t;
106  const uint16_t *samples_uint16_t;
107  const uint32_t *samples_uint32_t;
108 
109  sample_size = av_get_bits_per_sample(avctx->codec->id) / 8;
110  n = frame->nb_samples * avctx->ch_layout.nb_channels;
111  samples = (const short *)frame->data[0];
112 
113  if ((ret = ff_get_encode_buffer(avctx, avpkt, n * sample_size, 0)) < 0)
114  return ret;
115  dst = avpkt->data;
116 
117  switch (avctx->codec->id) {
119  ENCODE(uint32_t, le32, samples, dst, n, 0, 0x80000000)
120  break;
122  ENCODE(uint32_t, be32, samples, dst, n, 0, 0x80000000)
123  break;
125  ENCODE(int32_t, le24, samples, dst, n, 8, 0)
126  break;
128  ENCODE_PLANAR(int32_t, le24, dst, n, 8, 0)
129  break;
131  ENCODE(int32_t, be24, samples, dst, n, 8, 0)
132  break;
134  ENCODE(uint32_t, le24, samples, dst, n, 8, 0x800000)
135  break;
137  ENCODE(uint32_t, be24, samples, dst, n, 8, 0x800000)
138  break;
140  for (; n > 0; n--) {
141  uint32_t tmp = ff_reverse[(*samples >> 8) & 0xff] +
142  (ff_reverse[*samples & 0xff] << 8);
143  tmp <<= 4; // sync flags would go here
144  bytestream_put_be24(&dst, tmp);
145  samples++;
146  }
147  break;
149  ENCODE(uint16_t, le16, samples, dst, n, 0, 0x8000)
150  break;
152  ENCODE(uint16_t, be16, samples, dst, n, 0, 0x8000)
153  break;
154  case AV_CODEC_ID_PCM_S8:
155  ENCODE(uint8_t, byte, samples, dst, n, 0, -128)
156  break;
158  ENCODE_PLANAR(uint8_t, byte, dst, n, 0, -128)
159  break;
160 #if HAVE_BIGENDIAN
163  ENCODE(int64_t, le64, samples, dst, n, 0, 0)
164  break;
167  ENCODE(int32_t, le32, samples, dst, n, 0, 0)
168  break;
170  ENCODE_PLANAR(int32_t, le32, dst, n, 0, 0)
171  break;
173  ENCODE(int16_t, le16, samples, dst, n, 0, 0)
174  break;
176  ENCODE_PLANAR(int16_t, le16, dst, n, 0, 0)
177  break;
183 #else
186  ENCODE(int64_t, be64, samples, dst, n, 0, 0)
187  break;
190  ENCODE(int32_t, be32, samples, dst, n, 0, 0)
191  break;
193  ENCODE(int16_t, be16, samples, dst, n, 0, 0)
194  break;
196  ENCODE_PLANAR(int16_t, be16, dst, n, 0, 0)
197  break;
203 #endif /* HAVE_BIGENDIAN */
204  case AV_CODEC_ID_PCM_U8:
205  memcpy(dst, samples, n * sample_size);
206  break;
207 #if HAVE_BIGENDIAN
209 #else
212 #endif /* HAVE_BIGENDIAN */
213  n /= avctx->ch_layout.nb_channels;
214  for (c = 0; c < avctx->ch_layout.nb_channels; c++) {
215  const uint8_t *src = frame->extended_data[c];
216  bytestream_put_buffer(&dst, src, n * sample_size);
217  }
218  break;
220  for (; n > 0; n--) {
221  v = *samples++;
222  *dst++ = linear_to_alaw[(v + 32768) >> 2];
223  }
224  break;
226  for (; n > 0; n--) {
227  v = *samples++;
228  *dst++ = linear_to_ulaw[(v + 32768) >> 2];
229  }
230  break;
232  for (; n > 0; n--) {
233  v = *samples++;
234  *dst++ = linear_to_vidc[(v + 32768) >> 2];
235  }
236  break;
237  default:
238  return -1;
239  }
240 
241  *got_packet_ptr = 1;
242  return 0;
243 }
244 
245 typedef struct PCMDecode {
246  short table[256];
247  void (*vector_fmul_scalar)(float *dst, const float *src, float mul,
248  int len);
249  float scale;
250 } PCMDecode;
251 
253 {
254  PCMDecode *s = avctx->priv_data;
255  AVFloatDSPContext *fdsp;
256  int i;
257 
258  switch (avctx->codec_id) {
260  for (i = 0; i < 256; i++)
261  s->table[i] = alaw2linear(i);
262  break;
264  for (i = 0; i < 256; i++)
265  s->table[i] = ulaw2linear(i);
266  break;
268  for (i = 0; i < 256; i++)
269  s->table[i] = vidc2linear(i);
270  break;
273  if (avctx->bits_per_coded_sample < 1 || avctx->bits_per_coded_sample > 24)
274  return AVERROR_INVALIDDATA;
275 
276  s->scale = 1. / (1 << (avctx->bits_per_coded_sample - 1));
277  fdsp = avpriv_float_dsp_alloc(0);
278  if (!fdsp)
279  return AVERROR(ENOMEM);
280  s->vector_fmul_scalar = fdsp->vector_fmul_scalar;
281  av_free(fdsp);
282  break;
283  default:
284  break;
285  }
286 
287  avctx->sample_fmt = avctx->codec->sample_fmts[0];
288 
289  if (avctx->sample_fmt == AV_SAMPLE_FMT_S32)
291 
292  return 0;
293 }
294 
295 /**
296  * Read PCM samples macro
297  * @param size Data size of native machine format
298  * @param endian bytestream_get_xxx() endian suffix
299  * @param src Source pointer (variable name)
300  * @param dst Destination pointer (variable name)
301  * @param n Total number of samples (variable name)
302  * @param shift Bitshift (bits)
303  * @param offset Sample value offset
304  */
305 #define DECODE(size, endian, src, dst, n, shift, offset) \
306  for (; n > 0; n--) { \
307  uint ## size ## _t v = bytestream_get_ ## endian(&src); \
308  AV_WN ## size ## A(dst, (uint ## size ## _t)(v - offset) << shift); \
309  dst += size / 8; \
310  }
311 
312 #define DECODE_PLANAR(size, endian, src, dst, n, shift, offset) \
313  n /= channels; \
314  for (c = 0; c < avctx->ch_layout.nb_channels; c++) { \
315  int i; \
316  dst = frame->extended_data[c]; \
317  for (i = n; i > 0; i--) { \
318  uint ## size ## _t v = bytestream_get_ ## endian(&src); \
319  AV_WN ## size ## A(dst, (uint ## size ##_t)(v - offset) << shift); \
320  dst += size / 8; \
321  } \
322  }
323 
325  int *got_frame_ptr, AVPacket *avpkt)
326 {
327  const uint8_t *src = avpkt->data;
328  int buf_size = avpkt->size;
329  PCMDecode *s = avctx->priv_data;
330  int channels = avctx->ch_layout.nb_channels;
331  int sample_size, c, n, ret, samples_per_block;
332  uint8_t *samples;
333  int32_t *dst_int32_t;
334 
335  sample_size = av_get_bits_per_sample(avctx->codec_id) / 8;
336 
337  /* av_get_bits_per_sample returns 0 for AV_CODEC_ID_PCM_DVD */
338  samples_per_block = 1;
339  if (avctx->codec_id == AV_CODEC_ID_PCM_LXF) {
340  /* we process 40-bit blocks per channel for LXF */
341  samples_per_block = 2;
342  sample_size = 5;
343  }
344 
345  if (sample_size == 0) {
346  av_log(avctx, AV_LOG_ERROR, "Invalid sample_size\n");
347  return AVERROR(EINVAL);
348  }
349 
350  if (channels == 0) {
351  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
352  return AVERROR(EINVAL);
353  }
354 
355  if (avctx->codec_id != avctx->codec->id) {
356  av_log(avctx, AV_LOG_ERROR, "codec ids mismatch\n");
357  return AVERROR(EINVAL);
358  }
359 
360  n = channels * sample_size;
361 
362  if (n && buf_size % n) {
363  if (buf_size < n) {
364  av_log(avctx, AV_LOG_ERROR,
365  "Invalid PCM packet, data has size %d but at least a size of %d was expected\n",
366  buf_size, n);
367  return AVERROR_INVALIDDATA;
368  } else
369  buf_size -= buf_size % n;
370  }
371 
372  n = buf_size / sample_size;
373 
374  /* get output buffer */
375  frame->nb_samples = n * samples_per_block / channels;
376  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
377  return ret;
378  samples = frame->data[0];
379 
380  switch (avctx->codec_id) {
382  DECODE(32, le32, src, samples, n, 0, 0x80000000)
383  break;
385  DECODE(32, be32, src, samples, n, 0, 0x80000000)
386  break;
388  DECODE(32, le24, src, samples, n, 8, 0)
389  break;
391  DECODE_PLANAR(32, le24, src, samples, n, 8, 0);
392  break;
394  DECODE(32, be24, src, samples, n, 8, 0)
395  break;
397  DECODE(32, le24, src, samples, n, 8, 0x800000)
398  break;
400  DECODE(32, be24, src, samples, n, 8, 0x800000)
401  break;
403  for (; n > 0; n--) {
404  uint32_t v = bytestream_get_be24(&src);
405  v >>= 4; // sync flags are here
406  AV_WN16A(samples, ff_reverse[(v >> 8) & 0xff] +
407  (ff_reverse[v & 0xff] << 8));
408  samples += 2;
409  }
410  break;
412  DECODE(16, le16, src, samples, n, 0, 0x8000)
413  break;
415  DECODE(16, be16, src, samples, n, 0, 0x8000)
416  break;
417  case AV_CODEC_ID_PCM_S8:
418  for (; n > 0; n--)
419  *samples++ = *src++ + 128;
420  break;
421  case AV_CODEC_ID_PCM_SGA:
422  for (; n > 0; n--) {
423  int sign = *src >> 7;
424  int magn = *src & 0x7f;
425  *samples++ = sign ? 128 - magn : 128 + magn;
426  src++;
427  }
428  break;
430  n /= avctx->ch_layout.nb_channels;
431  for (c = 0; c < avctx->ch_layout.nb_channels; c++) {
432  int i;
433  samples = frame->extended_data[c];
434  for (i = n; i > 0; i--)
435  *samples++ = *src++ + 128;
436  }
437  break;
438 #if HAVE_BIGENDIAN
441  DECODE(64, le64, src, samples, n, 0, 0)
442  break;
447  DECODE(32, le32, src, samples, n, 0, 0)
448  break;
450  DECODE_PLANAR(32, le32, src, samples, n, 0, 0);
451  break;
453  DECODE(16, le16, src, samples, n, 0, 0)
454  break;
456  DECODE_PLANAR(16, le16, src, samples, n, 0, 0);
457  break;
463 #else
466  DECODE(64, be64, src, samples, n, 0, 0)
467  break;
470  DECODE(32, be32, src, samples, n, 0, 0)
471  break;
473  DECODE(16, be16, src, samples, n, 0, 0)
474  break;
476  DECODE_PLANAR(16, be16, src, samples, n, 0, 0);
477  break;
485 #endif /* HAVE_BIGENDIAN */
486  case AV_CODEC_ID_PCM_U8:
487  memcpy(samples, src, n * sample_size);
488  break;
489 #if HAVE_BIGENDIAN
491 #else
494 #endif /* HAVE_BIGENDIAN */
495  n /= avctx->ch_layout.nb_channels;
496  for (c = 0; c < avctx->ch_layout.nb_channels; c++) {
497  samples = frame->extended_data[c];
498  bytestream_get_buffer(&src, samples, n * sample_size);
499  }
500  break;
504  for (; n > 0; n--) {
505  AV_WN16A(samples, s->table[*src++]);
506  samples += 2;
507  }
508  break;
509  case AV_CODEC_ID_PCM_LXF:
510  {
511  int i;
512  n /= channels;
513  for (c = 0; c < channels; c++) {
514  dst_int32_t = (int32_t *)frame->extended_data[c];
515  for (i = 0; i < n; i++) {
516  // extract low 20 bits and expand to 32 bits
517  *dst_int32_t++ = ((uint32_t)src[2]<<28) |
518  (src[1] << 20) |
519  (src[0] << 12) |
520  ((src[2] & 0x0F) << 8) |
521  src[1];
522  // extract high 20 bits and expand to 32 bits
523  *dst_int32_t++ = ((uint32_t)src[4]<<24) |
524  (src[3] << 16) |
525  ((src[2] & 0xF0) << 8) |
526  (src[4] << 4) |
527  (src[3] >> 4);
528  src += 5;
529  }
530  }
531  break;
532  }
533  default:
534  return -1;
535  }
536 
537  if (avctx->codec_id == AV_CODEC_ID_PCM_F16LE ||
538  avctx->codec_id == AV_CODEC_ID_PCM_F24LE) {
539  s->vector_fmul_scalar((float *)frame->extended_data[0],
540  (const float *)frame->extended_data[0],
541  s->scale, FFALIGN(frame->nb_samples * avctx->ch_layout.nb_channels, 4));
542  }
543 
544  *got_frame_ptr = 1;
545 
546  return buf_size;
547 }
548 
549 #define PCM_ENCODER_0(id_, sample_fmt_, name_, long_name_)
550 #define PCM_ENCODER_1(id_, sample_fmt_, name_, long_name_) \
551 const FFCodec ff_ ## name_ ## _encoder = { \
552  .p.name = #name_, \
553  CODEC_LONG_NAME(long_name_), \
554  .p.type = AVMEDIA_TYPE_AUDIO, \
555  .p.id = AV_CODEC_ID_ ## id_, \
556  .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_VARIABLE_FRAME_SIZE | \
557  AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE, \
558  .init = pcm_encode_init, \
559  FF_CODEC_ENCODE_CB(pcm_encode_frame), \
560  .p.sample_fmts = (const enum AVSampleFormat[]){ sample_fmt_, \
561  AV_SAMPLE_FMT_NONE }, \
562 }
563 
564 #define PCM_ENCODER_2(cf, id, sample_fmt, name, long_name) \
565  PCM_ENCODER_ ## cf(id, sample_fmt, name, long_name)
566 #define PCM_ENCODER_3(cf, id, sample_fmt, name, long_name) \
567  PCM_ENCODER_2(cf, id, sample_fmt, name, long_name)
568 #define PCM_ENCODER(id, sample_fmt, name, long_name) \
569  PCM_ENCODER_3(CONFIG_ ## id ## _ENCODER, id, sample_fmt, name, long_name)
570 
571 #define PCM_DECODER_0(id, sample_fmt, name, long_name)
572 #define PCM_DECODER_1(id_, sample_fmt_, name_, long_name_) \
573 const FFCodec ff_ ## name_ ## _decoder = { \
574  .p.name = #name_, \
575  CODEC_LONG_NAME(long_name_), \
576  .p.type = AVMEDIA_TYPE_AUDIO, \
577  .p.id = AV_CODEC_ID_ ## id_, \
578  .priv_data_size = sizeof(PCMDecode), \
579  .init = pcm_decode_init, \
580  FF_CODEC_DECODE_CB(pcm_decode_frame), \
581  .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_PARAM_CHANGE, \
582  .p.sample_fmts = (const enum AVSampleFormat[]){ sample_fmt_, \
583  AV_SAMPLE_FMT_NONE }, \
584 }
585 
586 #define PCM_DECODER_2(cf, id, sample_fmt, name, long_name) \
587  PCM_DECODER_ ## cf(id, sample_fmt, name, long_name)
588 #define PCM_DECODER_3(cf, id, sample_fmt, name, long_name) \
589  PCM_DECODER_2(cf, id, sample_fmt, name, long_name)
590 #define PCM_DECODER(id, sample_fmt, name, long_name) \
591  PCM_DECODER_3(CONFIG_ ## id ## _DECODER, id, sample_fmt, name, long_name)
592 
593 #define PCM_CODEC(id, sample_fmt_, name, long_name_) \
594  PCM_ENCODER(id, sample_fmt_, name, long_name_); \
595  PCM_DECODER(id, sample_fmt_, name, long_name_)
596 
597 /* Note: Do not forget to add new entries to the Makefile as well. */
598 PCM_CODEC (PCM_ALAW, AV_SAMPLE_FMT_S16, pcm_alaw, "PCM A-law / G.711 A-law");
599 PCM_DECODER(PCM_F16LE, AV_SAMPLE_FMT_FLT, pcm_f16le, "PCM 16.8 floating point little-endian");
600 PCM_DECODER(PCM_F24LE, AV_SAMPLE_FMT_FLT, pcm_f24le, "PCM 24.0 floating point little-endian");
601 PCM_CODEC (PCM_F32BE, AV_SAMPLE_FMT_FLT, pcm_f32be, "PCM 32-bit floating point big-endian");
602 PCM_CODEC (PCM_F32LE, AV_SAMPLE_FMT_FLT, pcm_f32le, "PCM 32-bit floating point little-endian");
603 PCM_CODEC (PCM_F64BE, AV_SAMPLE_FMT_DBL, pcm_f64be, "PCM 64-bit floating point big-endian");
604 PCM_CODEC (PCM_F64LE, AV_SAMPLE_FMT_DBL, pcm_f64le, "PCM 64-bit floating point little-endian");
605 PCM_DECODER(PCM_LXF, AV_SAMPLE_FMT_S32P,pcm_lxf, "PCM signed 20-bit little-endian planar");
606 PCM_CODEC (PCM_MULAW, AV_SAMPLE_FMT_S16, pcm_mulaw, "PCM mu-law / G.711 mu-law");
607 PCM_CODEC (PCM_S8, AV_SAMPLE_FMT_U8, pcm_s8, "PCM signed 8-bit");
608 PCM_CODEC (PCM_S8_PLANAR, AV_SAMPLE_FMT_U8P, pcm_s8_planar, "PCM signed 8-bit planar");
609 PCM_CODEC (PCM_S16BE, AV_SAMPLE_FMT_S16, pcm_s16be, "PCM signed 16-bit big-endian");
610 PCM_CODEC (PCM_S16BE_PLANAR, AV_SAMPLE_FMT_S16P,pcm_s16be_planar, "PCM signed 16-bit big-endian planar");
611 PCM_CODEC (PCM_S16LE, AV_SAMPLE_FMT_S16, pcm_s16le, "PCM signed 16-bit little-endian");
612 PCM_CODEC (PCM_S16LE_PLANAR, AV_SAMPLE_FMT_S16P,pcm_s16le_planar, "PCM signed 16-bit little-endian planar");
613 PCM_CODEC (PCM_S24BE, AV_SAMPLE_FMT_S32, pcm_s24be, "PCM signed 24-bit big-endian");
614 PCM_CODEC (PCM_S24DAUD, AV_SAMPLE_FMT_S16, pcm_s24daud, "PCM D-Cinema audio signed 24-bit");
615 PCM_CODEC (PCM_S24LE, AV_SAMPLE_FMT_S32, pcm_s24le, "PCM signed 24-bit little-endian");
616 PCM_CODEC (PCM_S24LE_PLANAR, AV_SAMPLE_FMT_S32P,pcm_s24le_planar, "PCM signed 24-bit little-endian planar");
617 PCM_CODEC (PCM_S32BE, AV_SAMPLE_FMT_S32, pcm_s32be, "PCM signed 32-bit big-endian");
618 PCM_CODEC (PCM_S32LE, AV_SAMPLE_FMT_S32, pcm_s32le, "PCM signed 32-bit little-endian");
619 PCM_CODEC (PCM_S32LE_PLANAR, AV_SAMPLE_FMT_S32P,pcm_s32le_planar, "PCM signed 32-bit little-endian planar");
620 PCM_CODEC (PCM_U8, AV_SAMPLE_FMT_U8, pcm_u8, "PCM unsigned 8-bit");
621 PCM_CODEC (PCM_U16BE, AV_SAMPLE_FMT_S16, pcm_u16be, "PCM unsigned 16-bit big-endian");
622 PCM_CODEC (PCM_U16LE, AV_SAMPLE_FMT_S16, pcm_u16le, "PCM unsigned 16-bit little-endian");
623 PCM_CODEC (PCM_U24BE, AV_SAMPLE_FMT_S32, pcm_u24be, "PCM unsigned 24-bit big-endian");
624 PCM_CODEC (PCM_U24LE, AV_SAMPLE_FMT_S32, pcm_u24le, "PCM unsigned 24-bit little-endian");
625 PCM_CODEC (PCM_U32BE, AV_SAMPLE_FMT_S32, pcm_u32be, "PCM unsigned 32-bit big-endian");
626 PCM_CODEC (PCM_U32LE, AV_SAMPLE_FMT_S32, pcm_u32le, "PCM unsigned 32-bit little-endian");
627 PCM_CODEC (PCM_S64BE, AV_SAMPLE_FMT_S64, pcm_s64be, "PCM signed 64-bit big-endian");
628 PCM_CODEC (PCM_S64LE, AV_SAMPLE_FMT_S64, pcm_s64le, "PCM signed 64-bit little-endian");
629 PCM_CODEC (PCM_VIDC, AV_SAMPLE_FMT_S16, pcm_vidc, "PCM Archimedes VIDC");
630 PCM_DECODER(PCM_SGA, AV_SAMPLE_FMT_U8, pcm_sga, "PCM SGA");
AV_CODEC_ID_PCM_S16LE
@ AV_CODEC_ID_PCM_S16LE
Definition: codec_id.h:328
PCMDecode::scale
float scale
Definition: pcm.c:249
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1077
PCM_CODEC
#define PCM_CODEC(id, sample_fmt_, name, long_name_)
Definition: pcm.c:593
linear_to_alaw
static uint8_t linear_to_alaw[16384]
Definition: pcm_tablegen.h:99
AV_CODEC_ID_PCM_F32BE
@ AV_CODEC_ID_PCM_F32BE
Definition: codec_id.h:348
le32
uint64_t_TMPL AV_WL64 unsigned int_TMPL le32
Definition: bytestream.h:92
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
ENCODE
#define ENCODE(type, endian, src, dst, n, shift, offset)
Write PCM samples macro.
Definition: pcm.c:78
pcm_tablegen.h
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:1050
thread.h
vidc2linear
static av_cold int vidc2linear(unsigned char u_val)
Definition: pcm_tablegen.h:78
AV_CODEC_ID_PCM_S32LE_PLANAR
@ AV_CODEC_ID_PCM_S32LE_PLANAR
Definition: codec_id.h:357
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:374
tmp
static uint8_t tmp[11]
Definition: aes_ctr.c:28
AVPacket::data
uint8_t * data
Definition: packet.h:520
AV_CODEC_ID_PCM_S16BE_PLANAR
@ AV_CODEC_ID_PCM_S16BE_PLANAR
Definition: codec_id.h:358
encode.h
ff_reverse
const uint8_t ff_reverse[256]
Definition: reverse.c:23
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:65
AV_CODEC_ID_PCM_U24LE
@ AV_CODEC_ID_PCM_U24LE
Definition: codec_id.h:342
AV_CODEC_ID_PCM_SGA
@ AV_CODEC_ID_PCM_SGA
Definition: codec_id.h:364
reverse.h
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:321
AV_CODEC_ID_PCM_S16LE_PLANAR
@ AV_CODEC_ID_PCM_S16LE_PLANAR
Definition: codec_id.h:346
AV_CODEC_ID_PCM_S64LE
@ AV_CODEC_ID_PCM_S64LE
Definition: codec_id.h:359
AVCodecContext::codec
const struct AVCodec * codec
Definition: avcodec.h:454
AV_CODEC_ID_PCM_S16BE
@ AV_CODEC_ID_PCM_S16BE
Definition: codec_id.h:329
AVCodecContext::ch_layout
AVChannelLayout ch_layout
Audio channel layout.
Definition: avcodec.h:1065
AVCodec::sample_fmts
enum AVSampleFormat * sample_fmts
array of supported sample formats, or NULL if unknown, array is terminated by -1
Definition: codec.h:211
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:97
le24
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL le24
Definition: bytestream.h:93
PCM_DECODER
#define PCM_DECODER(id, sample_fmt, name, long_name)
Definition: pcm.c:590
av_get_bits_per_sample
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
Definition: utils.c:547
AV_CODEC_ID_PCM_S8
@ AV_CODEC_ID_PCM_S8
Definition: codec_id.h:332
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
av_cold
#define av_cold
Definition: attributes.h:90
s
#define s(width, name)
Definition: cbs_vp9.c:198
AV_CODEC_ID_PCM_LXF
@ AV_CODEC_ID_PCM_LXF
Definition: codec_id.h:353
linear_to_ulaw
static uint8_t linear_to_ulaw[16384]
Definition: pcm_tablegen.h:100
AVCodecContext::bits_per_raw_sample
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:1575
AV_CODEC_ID_PCM_F24LE
@ AV_CODEC_ID_PCM_F24LE
Definition: codec_id.h:362
channels
channels
Definition: aptx.h:31
decode.h
AV_WN16A
#define AV_WN16A(p, v)
Definition: intreadwrite.h:530
AV_CODEC_ID_PCM_MULAW
@ AV_CODEC_ID_PCM_MULAW
Definition: codec_id.h:334
AV_CODEC_ID_PCM_U16BE
@ AV_CODEC_ID_PCM_U16BE
Definition: codec_id.h:331
AVCodecContext::codec_id
enum AVCodecID codec_id
Definition: avcodec.h:455
if
if(ret)
Definition: filter_design.txt:179
AV_CODEC_ID_PCM_ALAW
@ AV_CODEC_ID_PCM_ALAW
Definition: codec_id.h:335
AV_CODEC_ID_PCM_U24BE
@ AV_CODEC_ID_PCM_U24BE
Definition: codec_id.h:343
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
AV_CODEC_ID_PCM_U32BE
@ AV_CODEC_ID_PCM_U32BE
Definition: codec_id.h:339
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:495
AV_CODEC_ID_PCM_S64BE
@ AV_CODEC_ID_PCM_S64BE
Definition: codec_id.h:360
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:95
DECODE_PLANAR
#define DECODE_PLANAR(size, endian, src, dst, n, shift, offset)
Definition: pcm.c:312
PCMDecode::vector_fmul_scalar
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Definition: pcm.c:247
pcm_decode_frame
static int pcm_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
Definition: pcm.c:324
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:96
pcm_decode_init
static av_cold int pcm_decode_init(AVCodecContext *avctx)
Definition: pcm.c:252
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_S24LE_PLANAR
@ AV_CODEC_ID_PCM_S24LE_PLANAR
Definition: codec_id.h:356
float_dsp.h
AV_CODEC_ID_PCM_VIDC
@ AV_CODEC_ID_PCM_VIDC
Definition: codec_id.h:363
pcm_encode_frame
static int pcm_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: pcm.c:96
AV_CODEC_ID_PCM_S24LE
@ AV_CODEC_ID_PCM_S24LE
Definition: codec_id.h:340
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1575
AVPacket::size
int size
Definition: packet.h:521
AV_SAMPLE_FMT_U8P
@ AV_SAMPLE_FMT_U8P
unsigned 8 bits, planar
Definition: samplefmt.h:63
codec_internal.h
alaw2linear
static av_cold int alaw2linear(unsigned char a_val)
Definition: pcm_tablegen.h:46
for
for(k=2;k<=8;++k)
Definition: h264pred_template.c:425
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1057
AVFloatDSPContext
Definition: float_dsp.h:24
DECODE
#define DECODE(size, endian, src, dst, n, shift, offset)
Read PCM samples macro.
Definition: pcm.c:305
ENCODE_PLANAR
#define ENCODE_PLANAR(type, endian, dst, n, shift, offset)
Definition: pcm.c:85
attributes.h
AV_SAMPLE_FMT_S16P
@ AV_SAMPLE_FMT_S16P
signed 16 bits, planar
Definition: samplefmt.h:64
AVCodec::id
enum AVCodecID id
Definition: codec.h:201
AVCodecContext::bits_per_coded_sample
int bits_per_coded_sample
bits per sample/pixel from the demuxer (needed for huffyuv).
Definition: avcodec.h:1568
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
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
PCMDecode::table
short table[256]
Definition: pcm.c:246
AV_SAMPLE_FMT_U8
@ AV_SAMPLE_FMT_U8
unsigned 8 bits
Definition: samplefmt.h:57
AV_CODEC_ID_PCM_F64BE
@ AV_CODEC_ID_PCM_F64BE
Definition: codec_id.h:350
AV_CODEC_ID_PCM_S32BE
@ AV_CODEC_ID_PCM_S32BE
Definition: codec_id.h:337
AV_SAMPLE_FMT_S16
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:58
le64
uint64_t_TMPL le64
Definition: bytestream.h:91
AV_CODEC_ID_PCM_F16LE
@ AV_CODEC_ID_PCM_F16LE
Definition: codec_id.h:361
len
int len
Definition: vorbis_enc_data.h:426
avcodec.h
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
ret
ret
Definition: filter_design.txt:187
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:1083
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:264
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:98
PCMDecode
Definition: pcm.c:245
le16
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL le16
Definition: bytestream.h:94
linear_to_vidc
static uint8_t linear_to_vidc[16384]
Definition: pcm_tablegen.h:101
AVCodecContext
main external API structure.
Definition: avcodec.h:445
ff_get_encode_buffer
int ff_get_encode_buffer(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int flags)
Get a buffer for a packet.
Definition: encode.c:106
INIT_ONCE
#define INIT_ONCE(id, name)
AV_CODEC_ID_PCM_U32LE
@ AV_CODEC_ID_PCM_U32LE
Definition: codec_id.h:338
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
AV_CODEC_ID_PCM_S32LE
@ AV_CODEC_ID_PCM_S32LE
Definition: codec_id.h:336
mem.h
AV_CODEC_ID_PCM_U8
@ AV_CODEC_ID_PCM_U8
Definition: codec_id.h:333
AV_CODEC_ID_PCM_S24DAUD
@ AV_CODEC_ID_PCM_S24DAUD
Definition: codec_id.h:344
pcm_encode_init
static av_cold int pcm_encode_init(AVCodecContext *avctx)
Definition: pcm.c:41
AV_CODEC_ID_PCM_F64LE
@ AV_CODEC_ID_PCM_F64LE
Definition: codec_id.h:351
av_free
#define av_free(p)
Definition: tableprint_vlc.h:33
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:78
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:472
AVPacket
This structure stores compressed data.
Definition: packet.h:497
src
INIT_CLIP pixel * src
Definition: h264pred_template.c:418
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:146
AV_CODEC_ID_PCM_S8_PLANAR
@ AV_CODEC_ID_PCM_S8_PLANAR
Definition: codec_id.h:355
int32_t
int32_t
Definition: audioconvert.c:56
bytestream.h
AV_CODEC_ID_PCM_U16LE
@ AV_CODEC_ID_PCM_U16LE
Definition: codec_id.h:330
AV_CODEC_ID_PCM_F32LE
@ AV_CODEC_ID_PCM_F32LE
Definition: codec_id.h:349
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:61
AV_SAMPLE_FMT_DBL
@ AV_SAMPLE_FMT_DBL
double
Definition: samplefmt.h:61
AV_SAMPLE_FMT_S32
@ AV_SAMPLE_FMT_S32
signed 32 bits
Definition: samplefmt.h:59
AV_CODEC_ID_PCM_S24BE
@ AV_CODEC_ID_PCM_S24BE
Definition: codec_id.h:341
AV_SAMPLE_FMT_FLT
@ AV_SAMPLE_FMT_FLT
float
Definition: samplefmt.h:60
ulaw2linear
static av_cold int ulaw2linear(unsigned char u_val)
Definition: pcm_tablegen.h:61
AV_SAMPLE_FMT_S64
@ AV_SAMPLE_FMT_S64
signed 64 bits
Definition: samplefmt.h:68