FFmpeg
iamf_parse.c
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1 /*
2  * Immersive Audio Model and Formats parsing
3  * Copyright (c) 2023 James Almer <jamrial@gmail.com>
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 #include "libavutil/avassert.h"
23 #include "libavutil/iamf.h"
24 #include "libavutil/intreadwrite.h"
25 #include "libavutil/log.h"
26 #include "libavutil/mem.h"
27 #include "libavcodec/get_bits.h"
28 #include "libavcodec/flac.h"
29 #include "libavcodec/leb.h"
30 #include "libavcodec/mpeg4audio.h"
31 #include "libavcodec/put_bits.h"
32 #include "avio_internal.h"
33 #include "iamf_parse.h"
34 #include "isom.h"
35 
36 static int opus_decoder_config(IAMFCodecConfig *codec_config,
37  AVIOContext *pb, int len)
38 {
39  int ret, left = len - avio_tell(pb);
40 
41  if (left < 11 || codec_config->audio_roll_distance >= 0)
42  return AVERROR_INVALIDDATA;
43 
44  codec_config->extradata = av_malloc(left + 8);
45  if (!codec_config->extradata)
46  return AVERROR(ENOMEM);
47 
48  AV_WB32A(codec_config->extradata, MKBETAG('O','p','u','s'));
49  AV_WB32A(codec_config->extradata + 4, MKBETAG('H','e','a','d'));
50  ret = ffio_read_size(pb, codec_config->extradata + 8, left);
51  if (ret < 0)
52  return ret;
53 
54  codec_config->extradata_size = left + 8;
55  codec_config->sample_rate = 48000;
56 
57  return 0;
58 }
59 
60 static int aac_decoder_config(IAMFCodecConfig *codec_config,
61  AVIOContext *pb, int len, void *logctx)
62 {
63  MPEG4AudioConfig cfg = { 0 };
64  int object_type_id, codec_id, stream_type;
65  int ret, tag, left;
66 
67  if (codec_config->audio_roll_distance >= 0)
68  return AVERROR_INVALIDDATA;
69 
70  ff_mp4_read_descr(logctx, pb, &tag);
72  return AVERROR_INVALIDDATA;
73 
74  object_type_id = avio_r8(pb);
75  if (object_type_id != 0x40)
76  return AVERROR_INVALIDDATA;
77 
78  stream_type = avio_r8(pb);
79  if (((stream_type >> 2) != 5) || ((stream_type >> 1) & 1))
80  return AVERROR_INVALIDDATA;
81 
82  avio_skip(pb, 3); // buffer size db
83  avio_skip(pb, 4); // rc_max_rate
84  avio_skip(pb, 4); // avg bitrate
85 
86  codec_id = ff_codec_get_id(ff_mp4_obj_type, object_type_id);
87  if (codec_id && codec_id != codec_config->codec_id)
88  return AVERROR_INVALIDDATA;
89 
90  left = ff_mp4_read_descr(logctx, pb, &tag);
91  if (tag != MP4DecSpecificDescrTag ||
92  !left || left > (len - avio_tell(pb)))
93  return AVERROR_INVALIDDATA;
94 
95  // We pad extradata here because avpriv_mpeg4audio_get_config2() needs it.
96  codec_config->extradata = av_malloc((size_t)left + AV_INPUT_BUFFER_PADDING_SIZE);
97  if (!codec_config->extradata)
98  return AVERROR(ENOMEM);
99 
100  ret = ffio_read_size(pb, codec_config->extradata, left);
101  if (ret < 0)
102  return ret;
103  codec_config->extradata_size = left;
104  memset(codec_config->extradata + codec_config->extradata_size, 0,
106 
107  ret = avpriv_mpeg4audio_get_config2(&cfg, codec_config->extradata,
108  codec_config->extradata_size, 1, logctx);
109  if (ret < 0)
110  return ret;
111 
112  codec_config->sample_rate = cfg.sample_rate;
113 
114  return 0;
115 }
116 
117 static int flac_decoder_config(IAMFCodecConfig *codec_config,
118  AVIOContext *pb, int len)
119 {
120  int ret, left;
121 
122  if (codec_config->audio_roll_distance)
123  return AVERROR_INVALIDDATA;
124 
125  avio_skip(pb, 4); // METADATA_BLOCK_HEADER
126 
127  left = len - avio_tell(pb);
129  return AVERROR_INVALIDDATA;
130 
131  codec_config->extradata = av_malloc(left);
132  if (!codec_config->extradata)
133  return AVERROR(ENOMEM);
134 
135  ret = ffio_read_size(pb, codec_config->extradata, left);
136  if (ret < 0)
137  return ret;
138 
139  codec_config->extradata_size = left;
140  codec_config->sample_rate = AV_RB24(codec_config->extradata + 10) >> 4;
141 
142  return 0;
143 }
144 
145 static int ipcm_decoder_config(IAMFCodecConfig *codec_config,
146  AVIOContext *pb, int len)
147 {
148  static const enum AVCodecID sample_fmt[2][3] = {
151  };
152  int sample_format = avio_r8(pb); // 0 = BE, 1 = LE
153  int sample_size = (avio_r8(pb) / 8 - 2); // 16, 24, 32
154  if (sample_format > 1 || sample_size > 2U || codec_config->audio_roll_distance)
155  return AVERROR_INVALIDDATA;
156 
157  codec_config->codec_id = sample_fmt[sample_format][sample_size];
158  codec_config->sample_rate = avio_rb32(pb);
159 
160  if (len - avio_tell(pb))
161  return AVERROR_INVALIDDATA;
162 
163  return 0;
164 }
165 
166 static int codec_config_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
167 {
168  IAMFCodecConfig **tmp, *codec_config = NULL;
169  FFIOContext b;
170  AVIOContext *pbc;
171  uint8_t *buf;
172  enum AVCodecID avcodec_id;
173  unsigned codec_config_id, nb_samples, codec_id;
174  int16_t audio_roll_distance;
175  int ret;
176 
177  buf = av_malloc(len);
178  if (!buf)
179  return AVERROR(ENOMEM);
180 
181  ret = ffio_read_size(pb, buf, len);
182  if (ret < 0)
183  goto fail;
184 
185  ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
186  pbc = &b.pub;
187 
188  codec_config_id = ffio_read_leb(pbc);
189  codec_id = avio_rb32(pbc);
190  nb_samples = ffio_read_leb(pbc);
191  audio_roll_distance = avio_rb16(pbc);
192 
193  switch(codec_id) {
194  case MKBETAG('O','p','u','s'):
195  avcodec_id = AV_CODEC_ID_OPUS;
196  break;
197  case MKBETAG('m','p','4','a'):
198  avcodec_id = AV_CODEC_ID_AAC;
199  break;
200  case MKBETAG('f','L','a','C'):
201  avcodec_id = AV_CODEC_ID_FLAC;
202  break;
203  default:
204  avcodec_id = AV_CODEC_ID_NONE;
205  break;
206  }
207 
208  for (int i = 0; i < c->nb_codec_configs; i++)
209  if (c->codec_configs[i]->codec_config_id == codec_config_id) {
211  goto fail;
212  }
213 
214  tmp = av_realloc_array(c->codec_configs, c->nb_codec_configs + 1, sizeof(*c->codec_configs));
215  if (!tmp) {
216  ret = AVERROR(ENOMEM);
217  goto fail;
218  }
219  c->codec_configs = tmp;
220 
221  codec_config = av_mallocz(sizeof(*codec_config));
222  if (!codec_config) {
223  ret = AVERROR(ENOMEM);
224  goto fail;
225  }
226 
227  codec_config->codec_config_id = codec_config_id;
228  codec_config->codec_id = avcodec_id;
229  codec_config->nb_samples = nb_samples;
230  codec_config->audio_roll_distance = audio_roll_distance;
231 
232  switch(codec_id) {
233  case MKBETAG('O','p','u','s'):
234  ret = opus_decoder_config(codec_config, pbc, len);
235  break;
236  case MKBETAG('m','p','4','a'):
237  ret = aac_decoder_config(codec_config, pbc, len, s);
238  break;
239  case MKBETAG('f','L','a','C'):
240  ret = flac_decoder_config(codec_config, pbc, len);
241  break;
242  case MKBETAG('i','p','c','m'):
243  ret = ipcm_decoder_config(codec_config, pbc, len);
244  break;
245  default:
246  break;
247  }
248  if (ret < 0)
249  goto fail;
250 
251  if ((codec_config->nb_samples > INT_MAX) || codec_config->nb_samples <= 0 ||
252  (-codec_config->audio_roll_distance > INT_MAX / codec_config->nb_samples)) {
254  goto fail;
255  }
256 
257  c->codec_configs[c->nb_codec_configs++] = codec_config;
258 
259  len -= avio_tell(pbc);
260  if (len)
261  av_log(s, AV_LOG_WARNING, "Underread in codec_config_obu. %d bytes left at the end\n", len);
262 
263  ret = 0;
264 fail:
265  av_free(buf);
266  if (ret < 0) {
267  if (codec_config)
268  av_free(codec_config->extradata);
269  av_free(codec_config);
270  }
271  return ret;
272 }
273 
274 static int update_extradata(AVCodecParameters *codecpar)
275 {
276  GetBitContext gb;
277  PutBitContext pb;
278  int ret;
279 
280  switch(codecpar->codec_id) {
281  case AV_CODEC_ID_OPUS:
282  AV_WB8(codecpar->extradata + 9, codecpar->ch_layout.nb_channels);
283  AV_WL16A(codecpar->extradata + 10, AV_RB16A(codecpar->extradata + 10)); // Byte swap pre-skip
284  AV_WL32A(codecpar->extradata + 12, AV_RB32A(codecpar->extradata + 12)); // Byte swap sample rate
285  AV_WL16A(codecpar->extradata + 16, AV_RB16A(codecpar->extradata + 16)); // Byte swap Output Gain
286  break;
287  case AV_CODEC_ID_AAC: {
288  uint8_t buf[6];
289  int size = FFMIN(codecpar->extradata_size, sizeof(buf));
290 
291  init_put_bits(&pb, buf, sizeof(buf));
292  ret = init_get_bits8(&gb, codecpar->extradata, size);
293  if (ret < 0)
294  return ret;
295 
296  ret = get_bits(&gb, 5);
297  put_bits(&pb, 5, ret);
298  if (ret == AOT_ESCAPE) // violates section 3.11.2, but better check for it
299  put_bits(&pb, 6, get_bits(&gb, 6));
300  ret = get_bits(&gb, 4);
301  put_bits(&pb, 4, ret);
302  if (ret == 0x0f)
303  put_bits(&pb, 24, get_bits(&gb, 24));
304 
305  skip_bits(&gb, 4);
306  put_bits(&pb, 4, codecpar->ch_layout.nb_channels); // set channel config
307  ret = get_bits_left(&gb);
308  if (ret < 0)
309  return AVERROR_INVALIDDATA;
310  ret = FFMIN(ret, put_bits_left(&pb));
311  while (ret >= 32) {
312  put_bits32(&pb, get_bits_long(&gb, 32));
313  ret -= 32;
314  }
315  put_bits(&pb, ret, get_bits_long(&gb, ret));
316  flush_put_bits(&pb);
317 
318  memcpy(codecpar->extradata, buf, put_bytes_output(&pb));
319  break;
320  }
321  case AV_CODEC_ID_FLAC: {
322  uint8_t buf[13];
323  int size = FFMIN(codecpar->extradata_size, sizeof(buf));
324 
325  init_put_bits(&pb, buf, sizeof(buf));
326  ret = init_get_bits8(&gb, codecpar->extradata, size);
327  if (ret < 0)
328  return ret;
329 
330  put_bits32(&pb, get_bits_long(&gb, 32)); // min/max blocksize
331  put_bits63(&pb, 48, get_bits64(&gb, 48)); // min/max framesize
332  put_bits(&pb, 20, get_bits(&gb, 20)); // samplerate
333  skip_bits(&gb, 3);
334  put_bits(&pb, 3, codecpar->ch_layout.nb_channels - 1);
335  ret = get_bits_left(&gb);
336  if (ret < 0)
337  return AVERROR_INVALIDDATA;
338  ret = FFMIN(ret, put_bits_left(&pb));
339  put_bits(&pb, ret, get_bits(&gb, ret));
340  flush_put_bits(&pb);
341 
342  memcpy(codecpar->extradata, buf, put_bytes_output(&pb));
343  break;
344  }
345  }
346 
347  return 0;
348 }
349 
351 {
352  if (in->u.mask & AV_CH_LAYOUT_STEREO) {
353  out->u.map[n++].id = AV_CHAN_FRONT_LEFT;
354  out->u.map[n++].id = AV_CHAN_FRONT_RIGHT;
355  in->u.mask &= ~AV_CH_LAYOUT_STEREO;
357  out->u.map[n++].id = AV_CHAN_FRONT_LEFT_OF_CENTER;
358  out->u.map[n++].id = AV_CHAN_FRONT_RIGHT_OF_CENTER;
360  } else if (in->u.mask & (AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT)) {
361  out->u.map[n++].id = AV_CHAN_SIDE_LEFT;
362  out->u.map[n++].id = AV_CHAN_SIDE_RIGHT;
364  } else if (in->u.mask & (AV_CH_BACK_LEFT|AV_CH_BACK_RIGHT)) {
365  out->u.map[n++].id = AV_CHAN_BACK_LEFT;
366  out->u.map[n++].id = AV_CHAN_BACK_RIGHT;
368  } else if (in->u.mask & (AV_CH_TOP_FRONT_LEFT|AV_CH_TOP_FRONT_RIGHT)) {
369  out->u.map[n++].id = AV_CHAN_TOP_FRONT_LEFT;
370  out->u.map[n++].id = AV_CHAN_TOP_FRONT_RIGHT;
372  } else if (in->u.mask & (AV_CH_TOP_SIDE_LEFT|AV_CH_TOP_SIDE_RIGHT)) {
373  out->u.map[n++].id = AV_CHAN_TOP_SIDE_LEFT;
374  out->u.map[n++].id = AV_CHAN_TOP_SIDE_RIGHT;
376  } else if (in->u.mask & (AV_CH_TOP_BACK_LEFT|AV_CH_TOP_BACK_RIGHT)) {
377  out->u.map[n++].id = AV_CHAN_TOP_BACK_LEFT;
378  out->u.map[n++].id = AV_CHAN_TOP_BACK_RIGHT;
380  }
381 
382  return n;
383 }
384 
386  IAMFAudioElement *audio_element,
387  const IAMFCodecConfig *codec_config)
388 {
389  int nb_layers, k = 0;
390 
391  nb_layers = avio_r8(pb) >> 5; // get_bits(&gb, 3);
392  // skip_bits(&gb, 5); //reserved
393 
394  if (nb_layers > 6 || nb_layers == 0)
395  return AVERROR_INVALIDDATA;
396 
397  audio_element->layers = av_calloc(nb_layers, sizeof(*audio_element->layers));
398  if (!audio_element->layers)
399  return AVERROR(ENOMEM);
400 
401  audio_element->nb_layers = nb_layers;
402  for (int i = 0, n = 0; i < nb_layers; i++) {
403  AVChannelLayout ch_layout = { 0 };
404  AVIAMFLayer *layer;
405  int loudspeaker_layout, output_gain_is_present_flag;
406  int substream_count, coupled_substream_count;
407  int expanded_loudspeaker_layout = -1;
408  int ret, byte = avio_r8(pb);
409 
410  layer = av_iamf_audio_element_add_layer(audio_element->element);
411  if (!layer)
412  return AVERROR(ENOMEM);
413 
414  loudspeaker_layout = byte >> 4; // get_bits(&gb, 4);
415  output_gain_is_present_flag = (byte >> 3) & 1; //get_bits1(&gb);
416  if ((byte >> 2) & 1)
417  layer->flags |= AV_IAMF_LAYER_FLAG_RECON_GAIN;
418  substream_count = avio_r8(pb);
419  coupled_substream_count = avio_r8(pb);
420 
421  if (!substream_count || coupled_substream_count > substream_count ||
422  substream_count + k > audio_element->nb_substreams)
423  return AVERROR_INVALIDDATA;
424 
425  audio_element->layers[i].substream_count = substream_count;
426  audio_element->layers[i].coupled_substream_count = coupled_substream_count;
427  if (output_gain_is_present_flag) {
428  layer->output_gain_flags = avio_r8(pb) >> 2; // get_bits(&gb, 6);
429  layer->output_gain = av_make_q(sign_extend(avio_rb16(pb), 16), 1 << 8);
430  }
431 
432  if (loudspeaker_layout == 15) {
433  if (i) {
434  av_log(s, AV_LOG_ERROR, "expanded_loudspeaker_layout set with more than one layer in Audio Element #%d\n",
435  audio_element->audio_element_id);
436  return AVERROR_INVALIDDATA;
437  }
438  expanded_loudspeaker_layout = avio_r8(pb);
439  }
440  if (expanded_loudspeaker_layout >= 0 && expanded_loudspeaker_layout < 13) {
441  av_channel_layout_copy(&ch_layout, &ff_iamf_expanded_scalable_ch_layouts[expanded_loudspeaker_layout]);
442  } else if (loudspeaker_layout < 10) {
443  av_channel_layout_copy(&ch_layout, &ff_iamf_scalable_ch_layouts[loudspeaker_layout]);
444  if (i) {
445  uint64_t mask = av_channel_layout_subset(&audio_element->element->layers[i-1]->ch_layout, UINT64_MAX);
446  // When the first layer is Mono, the second layer may not have the C channel (e.g. Stereo)
447  if (audio_element->element->layers[i-1]->ch_layout.nb_channels == 1)
448  n--;
449  else if ((ch_layout.u.mask & mask) != mask)
450  return AVERROR_INVALIDDATA;
451  ch_layout.u.mask &= ~mask;
452  }
453  } else
455  .nb_channels = substream_count +
456  coupled_substream_count };
457 
458  if (i && ch_layout.nb_channels <= audio_element->element->layers[i-1]->ch_layout.nb_channels)
459  return AVERROR_INVALIDDATA;
460 
461  for (int j = 0; j < substream_count; j++) {
462  IAMFSubStream *substream = &audio_element->substreams[k++];
463 
464  substream->codecpar->ch_layout = coupled_substream_count-- > 0 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO :
466 
467  ret = update_extradata(substream->codecpar);
468  if (ret < 0)
469  return ret;
470  }
471 
472  if (ch_layout.order == AV_CHANNEL_ORDER_NATIVE) {
474  if (ret < 0)
475  return ret;
476 
477  for (int j = 0; j < n; j++)
478  layer->ch_layout.u.map[j].id = av_channel_layout_channel_from_index(&audio_element->element->layers[i-1]->ch_layout, j);
479 
480  coupled_substream_count = audio_element->layers[i].coupled_substream_count;
481  while (coupled_substream_count--) {
482  n = parse_coupled_substream(&layer->ch_layout, &ch_layout, n);
483  }
484 
485  substream_count -= audio_element->layers[i].coupled_substream_count;
486  n = parse_coupled_substream(&layer->ch_layout, &ch_layout, n); // In case the first layer is Mono
487  while (substream_count--) {
488  if (ch_layout.u.mask & AV_CH_FRONT_CENTER) {
489  layer->ch_layout.u.map[n++].id = AV_CHAN_FRONT_CENTER;
490  ch_layout.u.mask &= ~AV_CH_FRONT_CENTER;
491  }
492  if (ch_layout.u.mask & AV_CH_LOW_FREQUENCY) {
493  layer->ch_layout.u.map[n++].id = AV_CHAN_LOW_FREQUENCY;
494  ch_layout.u.mask &= ~AV_CH_LOW_FREQUENCY;
495  }
496  }
497 
498  if (n != ch_layout.nb_channels)
499  return AVERROR_INVALIDDATA;
500 
502  if (ret < 0 && ret != AVERROR(ENOSYS))
503  return ret;
504  } else // AV_CHANNEL_ORDER_UNSPEC
505  av_channel_layout_copy(&layer->ch_layout, &ch_layout);
506  }
507 
508  if (k != audio_element->nb_substreams)
509  return AVERROR_INVALIDDATA;
510 
511  return 0;
512 }
513 
514 static int ambisonics_config(void *s, AVIOContext *pb,
515  IAMFAudioElement *audio_element,
516  const IAMFCodecConfig *codec_config)
517 {
518  AVIAMFLayer *layer;
519  unsigned ambisonics_mode;
520  int output_channel_count, substream_count, order;
521  int ret;
522 
523  ambisonics_mode = ffio_read_leb(pb);
524  if (ambisonics_mode > 1)
525  return AVERROR_INVALIDDATA;
526 
527  output_channel_count = avio_r8(pb); // C
528  substream_count = avio_r8(pb); // N
529  if (audio_element->nb_substreams != substream_count || output_channel_count == 0)
530  return AVERROR_INVALIDDATA;
531 
532  order = floor(sqrt(output_channel_count - 1));
533  /* incomplete order - some harmonics are missing */
534  if ((order + 1) * (order + 1) != output_channel_count)
535  return AVERROR_INVALIDDATA;
536 
537  audio_element->layers = av_mallocz(sizeof(*audio_element->layers));
538  if (!audio_element->layers)
539  return AVERROR(ENOMEM);
540 
541  audio_element->nb_layers = 1;
542  audio_element->layers->substream_count = substream_count;
543 
544  layer = av_iamf_audio_element_add_layer(audio_element->element);
545  if (!layer)
546  return AVERROR(ENOMEM);
547 
548  layer->ambisonics_mode = ambisonics_mode;
549  if (ambisonics_mode == 0) {
550  for (int i = 0; i < substream_count; i++) {
551  IAMFSubStream *substream = &audio_element->substreams[i];
552 
554 
555  ret = update_extradata(substream->codecpar);
556  if (ret < 0)
557  return ret;
558  }
559 
560  ret = av_channel_layout_custom_init(&layer->ch_layout, output_channel_count);
561  if (ret < 0)
562  return ret;
563 
564  for (int i = 0; i < output_channel_count; i++)
565  layer->ch_layout.u.map[i].id = avio_r8(pb) + AV_CHAN_AMBISONIC_BASE;
566 
568  if (ret < 0 && ret != AVERROR(ENOSYS))
569  return ret;
570  } else {
571  int coupled_substream_count = avio_r8(pb); // M
572  int nb_demixing_matrix = substream_count + coupled_substream_count;
573  int demixing_matrix_size = nb_demixing_matrix * output_channel_count;
574 
575  audio_element->layers->coupled_substream_count = coupled_substream_count;
576 
577  layer->ch_layout = (AVChannelLayout){ .order = AV_CHANNEL_ORDER_AMBISONIC, .nb_channels = output_channel_count };
578  layer->demixing_matrix = av_malloc_array(demixing_matrix_size, sizeof(*layer->demixing_matrix));
579  if (!layer->demixing_matrix)
580  return AVERROR(ENOMEM);
581 
582  for (int i = 0; i < demixing_matrix_size; i++)
583  layer->demixing_matrix[i] = av_make_q(sign_extend(avio_rb16(pb), 16), 1 << 8);
584 
585  for (int i = 0; i < substream_count; i++) {
586  IAMFSubStream *substream = &audio_element->substreams[i];
587 
588  substream->codecpar->ch_layout = coupled_substream_count-- > 0 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO :
590 
591 
592  ret = update_extradata(substream->codecpar);
593  if (ret < 0)
594  return ret;
595  }
596  }
597 
598  return 0;
599 }
600 
601 static int param_parse(void *s, IAMFContext *c, AVIOContext *pb,
602  unsigned int type,
603  const IAMFAudioElement *audio_element,
604  AVIAMFParamDefinition **out_param_definition)
605 {
607  AVIAMFParamDefinition *param;
608  unsigned int parameter_id, parameter_rate, mode;
609  unsigned int duration = 0, constant_subblock_duration = 0, nb_subblocks = 0;
610  unsigned int total_duration = 0;
611  size_t param_size;
612 
613  parameter_id = ffio_read_leb(pb);
614 
615  for (int i = 0; i < c->nb_param_definitions; i++)
616  if (c->param_definitions[i]->param->parameter_id == parameter_id) {
617  param_definition = c->param_definitions[i];
618  break;
619  }
620 
621  parameter_rate = ffio_read_leb(pb);
622  mode = avio_r8(pb) >> 7;
623 
624  if (mode == 0) {
625  duration = ffio_read_leb(pb);
626  if (!duration)
627  return AVERROR_INVALIDDATA;
628  constant_subblock_duration = ffio_read_leb(pb);
629  if (constant_subblock_duration == 0)
630  nb_subblocks = ffio_read_leb(pb);
631  else {
632  nb_subblocks = duration / constant_subblock_duration;
633  total_duration = duration;
634  }
635  }
636 
637  param = av_iamf_param_definition_alloc(type, nb_subblocks, &param_size);
638  if (!param)
639  return AVERROR(ENOMEM);
640 
641  for (int i = 0; i < nb_subblocks; i++) {
642  void *subblock = av_iamf_param_definition_get_subblock(param, i);
643  unsigned int subblock_duration = constant_subblock_duration;
644 
645  if (constant_subblock_duration == 0) {
646  subblock_duration = ffio_read_leb(pb);
647  total_duration += subblock_duration;
648  } else if (i == nb_subblocks - 1)
649  subblock_duration = duration - i * constant_subblock_duration;
650 
651  switch (type) {
653  AVIAMFMixGain *mix = subblock;
654  mix->subblock_duration = subblock_duration;
655  break;
656  }
658  AVIAMFDemixingInfo *demix = subblock;
659  demix->subblock_duration = subblock_duration;
660  // DefaultDemixingInfoParameterData
661  av_assert0(audio_element);
662  demix->dmixp_mode = avio_r8(pb) >> 5;
663  audio_element->element->default_w = avio_r8(pb) >> 4;
664  break;
665  }
667  AVIAMFReconGain *recon = subblock;
668  recon->subblock_duration = subblock_duration;
669  break;
670  }
671  default:
672  av_free(param);
673  return AVERROR_INVALIDDATA;
674  }
675  }
676 
677  if (!mode && !constant_subblock_duration && total_duration != duration) {
678  av_log(s, AV_LOG_ERROR, "Invalid subblock durations in parameter_id %u\n", parameter_id);
679  av_free(param);
680  return AVERROR_INVALIDDATA;
681  }
682 
683  param->parameter_id = parameter_id;
684  param->parameter_rate = parameter_rate;
685  param->duration = duration;
686  param->constant_subblock_duration = constant_subblock_duration;
687  param->nb_subblocks = nb_subblocks;
688 
689  if (param_definition) {
690  if (param_definition->param_size != param_size || memcmp(param_definition->param, param, param_size)) {
691  av_log(s, AV_LOG_ERROR, "Inconsistent parameters for parameter_id %u\n", parameter_id);
692  av_free(param);
693  return AVERROR_INVALIDDATA;
694  }
695  } else {
696  IAMFParamDefinition **tmp = av_realloc_array(c->param_definitions, c->nb_param_definitions + 1,
697  sizeof(*c->param_definitions));
698  if (!tmp) {
699  av_free(param);
700  return AVERROR(ENOMEM);
701  }
702  c->param_definitions = tmp;
703 
705  if (!param_definition) {
706  av_free(param);
707  return AVERROR(ENOMEM);
708  }
709  param_definition->param = param;
710  param_definition->mode = !mode;
711  param_definition->param_size = param_size;
712  param_definition->audio_element = audio_element;
713 
714  c->param_definitions[c->nb_param_definitions++] = param_definition;
715  }
716 
717  av_assert0(out_param_definition);
718  *out_param_definition = param;
719 
720  return 0;
721 }
722 
723 static int audio_element_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
724 {
725  const IAMFCodecConfig *codec_config;
726  AVIAMFAudioElement *element;
727  IAMFAudioElement **tmp, *audio_element = NULL;
728  FFIOContext b;
729  AVIOContext *pbc;
730  uint8_t *buf;
731  unsigned audio_element_id, nb_substreams, codec_config_id, num_parameters;
732  int audio_element_type, ret;
733 
734  buf = av_malloc(len);
735  if (!buf)
736  return AVERROR(ENOMEM);
737 
738  ret = ffio_read_size(pb, buf, len);
739  if (ret < 0)
740  goto fail;
741 
742  ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
743  pbc = &b.pub;
744 
745  audio_element_id = ffio_read_leb(pbc);
746 
747  for (int i = 0; i < c->nb_audio_elements; i++)
748  if (c->audio_elements[i]->audio_element_id == audio_element_id) {
749  av_log(s, AV_LOG_ERROR, "Duplicate audio_element_id %d\n", audio_element_id);
751  goto fail;
752  }
753 
754  audio_element_type = avio_r8(pbc) >> 5;
755  if (audio_element_type > AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE) {
756  av_log(s, AV_LOG_DEBUG, "Unknown audio_element_type referenced in an audio element. Ignoring\n");
757  ret = 0;
758  goto fail;
759  }
760 
761  codec_config_id = ffio_read_leb(pbc);
762 
763  codec_config = ff_iamf_get_codec_config(c, codec_config_id);
764  if (!codec_config) {
765  av_log(s, AV_LOG_ERROR, "Non existent codec config id %d referenced in an audio element\n", codec_config_id);
767  goto fail;
768  }
769 
770  if (codec_config->codec_id == AV_CODEC_ID_NONE) {
771  av_log(s, AV_LOG_DEBUG, "Unknown codec id referenced in an audio element. Ignoring\n");
772  ret = 0;
773  goto fail;
774  }
775 
776  tmp = av_realloc_array(c->audio_elements, c->nb_audio_elements + 1, sizeof(*c->audio_elements));
777  if (!tmp) {
778  ret = AVERROR(ENOMEM);
779  goto fail;
780  }
781  c->audio_elements = tmp;
782 
783  audio_element = av_mallocz(sizeof(*audio_element));
784  if (!audio_element) {
785  ret = AVERROR(ENOMEM);
786  goto fail;
787  }
788 
789  nb_substreams = ffio_read_leb(pbc);
790  audio_element->codec_config_id = codec_config_id;
791  audio_element->audio_element_id = audio_element_id;
792  audio_element->substreams = av_calloc(nb_substreams, sizeof(*audio_element->substreams));
793  if (!audio_element->substreams) {
794  ret = AVERROR(ENOMEM);
795  goto fail;
796  }
797  audio_element->nb_substreams = nb_substreams;
798 
799  element = audio_element->element = av_iamf_audio_element_alloc();
800  if (!element) {
801  ret = AVERROR(ENOMEM);
802  goto fail;
803  }
804  audio_element->celement = element;
805 
806  element->audio_element_type = audio_element_type;
807 
808  for (int i = 0; i < audio_element->nb_substreams; i++) {
809  IAMFSubStream *substream = &audio_element->substreams[i];
810 
811  substream->codecpar = avcodec_parameters_alloc();
812  if (!substream->codecpar) {
813  ret = AVERROR(ENOMEM);
814  goto fail;
815  }
816 
817  substream->audio_substream_id = ffio_read_leb(pbc);
818 
819  substream->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
820  substream->codecpar->codec_id = codec_config->codec_id;
821  substream->codecpar->frame_size = codec_config->nb_samples;
822  substream->codecpar->sample_rate = codec_config->sample_rate;
823  substream->codecpar->seek_preroll = -codec_config->audio_roll_distance * codec_config->nb_samples;
824 
825  switch(substream->codecpar->codec_id) {
826  case AV_CODEC_ID_AAC:
827  case AV_CODEC_ID_FLAC:
828  case AV_CODEC_ID_OPUS:
830  if (!substream->codecpar->extradata) {
831  ret = AVERROR(ENOMEM);
832  goto fail;
833  }
834  memcpy(substream->codecpar->extradata, codec_config->extradata, codec_config->extradata_size);
835  memset(substream->codecpar->extradata + codec_config->extradata_size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
836  substream->codecpar->extradata_size = codec_config->extradata_size;
837  break;
838  }
839  }
840 
841  num_parameters = ffio_read_leb(pbc);
842  if (num_parameters > 2 && audio_element_type == 0) {
843  av_log(s, AV_LOG_ERROR, "Audio Element parameter count %u is invalid"
844  " for Channel representations\n", num_parameters);
846  goto fail;
847  }
848  if (num_parameters && audio_element_type != 0) {
849  av_log(s, AV_LOG_ERROR, "Audio Element parameter count %u is invalid"
850  " for Scene representations\n", num_parameters);
852  goto fail;
853  }
854 
855  for (int i = 0; i < num_parameters; i++) {
856  unsigned type;
857 
858  type = ffio_read_leb(pbc);
862  if (element->demixing_info) {
864  goto fail;
865  }
866  ret = param_parse(s, c, pbc, type, audio_element, &element->demixing_info);
868  if (element->recon_gain_info) {
870  goto fail;
871  }
872  ret = param_parse(s, c, pbc, type, audio_element, &element->recon_gain_info);
873  } else {
874  unsigned param_definition_size = ffio_read_leb(pbc);
875  avio_skip(pbc, param_definition_size);
876  }
877  if (ret < 0)
878  goto fail;
879  }
880 
881  if (audio_element_type == AV_IAMF_AUDIO_ELEMENT_TYPE_CHANNEL) {
882  ret = scalable_channel_layout_config(s, pbc, audio_element, codec_config);
883  if (ret < 0)
884  goto fail;
885  } else if (audio_element_type == AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE) {
886  ret = ambisonics_config(s, pbc, audio_element, codec_config);
887  if (ret < 0)
888  goto fail;
889  } else {
890  av_assert0(0);
891  }
892 
893  c->audio_elements[c->nb_audio_elements++] = audio_element;
894 
895  len -= avio_tell(pbc);
896  if (len)
897  av_log(s, AV_LOG_WARNING, "Underread in audio_element_obu. %d bytes left at the end\n", len);
898 
899  ret = 0;
900 fail:
901  av_free(buf);
902  if (ret < 0)
903  ff_iamf_free_audio_element(&audio_element);
904  return ret;
905 }
906 
907 static int label_string(AVIOContext *pb, char **label)
908 {
909  uint8_t buf[128];
910 
911  avio_get_str(pb, sizeof(buf), buf, sizeof(buf));
912 
913  if (pb->error)
914  return pb->error;
915  if (pb->eof_reached)
916  return AVERROR_INVALIDDATA;
917  *label = av_strdup(buf);
918  if (!*label)
919  return AVERROR(ENOMEM);
920 
921  return 0;
922 }
923 
924 static int mix_presentation_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
925 {
927  IAMFMixPresentation **tmp, *mix_presentation = NULL;
928  FFIOContext b;
929  AVIOContext *pbc;
930  uint8_t *buf;
931  unsigned nb_submixes, mix_presentation_id;
932  int ret;
933 
934  buf = av_malloc(len);
935  if (!buf)
936  return AVERROR(ENOMEM);
937 
938  ret = ffio_read_size(pb, buf, len);
939  if (ret < 0)
940  goto fail;
941 
942  ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
943  pbc = &b.pub;
944 
945  mix_presentation_id = ffio_read_leb(pbc);
946 
947  for (int i = 0; i < c->nb_mix_presentations; i++)
948  if (c->mix_presentations[i]->mix_presentation_id == mix_presentation_id) {
949  av_log(s, AV_LOG_ERROR, "Duplicate mix_presentation_id %d\n", mix_presentation_id);
951  goto fail;
952  }
953 
954  tmp = av_realloc_array(c->mix_presentations, c->nb_mix_presentations + 1, sizeof(*c->mix_presentations));
955  if (!tmp) {
956  ret = AVERROR(ENOMEM);
957  goto fail;
958  }
959  c->mix_presentations = tmp;
960 
961  mix_presentation = av_mallocz(sizeof(*mix_presentation));
962  if (!mix_presentation) {
963  ret = AVERROR(ENOMEM);
964  goto fail;
965  }
966 
967  mix_presentation->mix_presentation_id = mix_presentation_id;
968  mix = mix_presentation->mix = av_iamf_mix_presentation_alloc();
969  if (!mix) {
970  ret = AVERROR(ENOMEM);
971  goto fail;
972  }
973  mix_presentation->cmix = mix;
974 
975  mix_presentation->count_label = ffio_read_leb(pbc);
976  mix_presentation->language_label = av_calloc(mix_presentation->count_label,
977  sizeof(*mix_presentation->language_label));
978  if (!mix_presentation->language_label) {
979  mix_presentation->count_label = 0;
980  ret = AVERROR(ENOMEM);
981  goto fail;
982  }
983 
984  for (int i = 0; i < mix_presentation->count_label; i++) {
985  ret = label_string(pbc, &mix_presentation->language_label[i]);
986  if (ret < 0)
987  goto fail;
988  }
989 
990  for (int i = 0; i < mix_presentation->count_label; i++) {
991  char *annotation = NULL;
992  ret = label_string(pbc, &annotation);
993  if (ret < 0)
994  goto fail;
995  ret = av_dict_set(&mix->annotations, mix_presentation->language_label[i], annotation,
997  if (ret < 0)
998  goto fail;
999  }
1000 
1001  nb_submixes = ffio_read_leb(pbc);
1002  for (int i = 0; i < nb_submixes; i++) {
1003  AVIAMFSubmix *sub_mix;
1004  unsigned nb_elements, nb_layouts;
1005 
1007  if (!sub_mix) {
1008  ret = AVERROR(ENOMEM);
1009  goto fail;
1010  }
1011 
1012  nb_elements = ffio_read_leb(pbc);
1013  for (int j = 0; j < nb_elements; j++) {
1014  AVIAMFSubmixElement *submix_element;
1015  IAMFAudioElement *audio_element = NULL;
1016  unsigned int rendering_config_extension_size;
1017 
1018  submix_element = av_iamf_submix_add_element(sub_mix);
1019  if (!submix_element) {
1020  ret = AVERROR(ENOMEM);
1021  goto fail;
1022  }
1023 
1024  submix_element->audio_element_id = ffio_read_leb(pbc);
1025 
1026  for (int k = 0; k < c->nb_audio_elements; k++)
1027  if (c->audio_elements[k]->audio_element_id == submix_element->audio_element_id) {
1028  audio_element = c->audio_elements[k];
1029  break;
1030  }
1031 
1032  if (!audio_element) {
1033  av_log(s, AV_LOG_ERROR, "Invalid Audio Element with id %u referenced by Mix Parameters %u\n",
1034  submix_element->audio_element_id, mix_presentation_id);
1036  goto fail;
1037  }
1038 
1039  for (int k = 0; k < mix_presentation->count_label; k++) {
1040  char *annotation = NULL;
1041  ret = label_string(pbc, &annotation);
1042  if (ret < 0)
1043  goto fail;
1044  ret = av_dict_set(&submix_element->annotations, mix_presentation->language_label[k], annotation,
1046  if (ret < 0)
1047  goto fail;
1048  }
1049 
1050  submix_element->headphones_rendering_mode = avio_r8(pbc) >> 6;
1051 
1052  rendering_config_extension_size = ffio_read_leb(pbc);
1053  avio_skip(pbc, rendering_config_extension_size);
1054 
1056  NULL,
1057  &submix_element->element_mix_config);
1058  if (ret < 0)
1059  goto fail;
1060  submix_element->default_mix_gain = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
1061  }
1062 
1064  if (ret < 0)
1065  goto fail;
1066  sub_mix->default_mix_gain = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
1067 
1068  nb_layouts = ffio_read_leb(pbc);
1069  for (int j = 0; j < nb_layouts; j++) {
1070  AVIAMFSubmixLayout *submix_layout;
1071  int info_type;
1072  int byte = avio_r8(pbc);
1073 
1074  submix_layout = av_iamf_submix_add_layout(sub_mix);
1075  if (!submix_layout) {
1076  ret = AVERROR(ENOMEM);
1077  goto fail;
1078  }
1079 
1080  submix_layout->layout_type = byte >> 6;
1083  av_log(s, AV_LOG_ERROR, "Invalid Layout type %u in a submix from Mix Presentation %u\n",
1084  submix_layout->layout_type, mix_presentation_id);
1086  goto fail;
1087  }
1088  if (submix_layout->layout_type == 2) {
1089  int sound_system;
1090  sound_system = (byte >> 2) & 0xF;
1091  if (sound_system >= FF_ARRAY_ELEMS(ff_iamf_sound_system_map)) {
1093  goto fail;
1094  }
1095  av_channel_layout_copy(&submix_layout->sound_system, &ff_iamf_sound_system_map[sound_system].layout);
1096  } else
1098 
1099  info_type = avio_r8(pbc);
1100  submix_layout->integrated_loudness = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
1101  submix_layout->digital_peak = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
1102 
1103  if (info_type & 1)
1104  submix_layout->true_peak = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
1105  if (info_type & 2) {
1106  unsigned int num_anchored_loudness = avio_r8(pbc);
1107 
1108  for (int k = 0; k < num_anchored_loudness; k++) {
1109  unsigned int anchor_element = avio_r8(pbc);
1110  AVRational anchored_loudness = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
1111  if (anchor_element == IAMF_ANCHOR_ELEMENT_DIALOGUE)
1112  submix_layout->dialogue_anchored_loudness = anchored_loudness;
1113  else if (anchor_element <= IAMF_ANCHOR_ELEMENT_ALBUM)
1114  submix_layout->album_anchored_loudness = anchored_loudness;
1115  else
1116  av_log(s, AV_LOG_DEBUG, "Unknown anchor_element. Ignoring\n");
1117  }
1118  }
1119 
1120  if (info_type & 0xFC) {
1121  unsigned int info_type_size = ffio_read_leb(pbc);
1122  avio_skip(pbc, info_type_size);
1123  }
1124  }
1125  }
1126 
1127  c->mix_presentations[c->nb_mix_presentations++] = mix_presentation;
1128 
1129  len -= avio_tell(pbc);
1130  if (len)
1131  av_log(s, AV_LOG_WARNING, "Underread in mix_presentation_obu. %d bytes left at the end\n", len);
1132 
1133  ret = 0;
1134 fail:
1135  av_free(buf);
1136  if (ret < 0)
1137  ff_iamf_free_mix_presentation(&mix_presentation);
1138  return ret;
1139 }
1140 
1141 int ff_iamf_parse_obu_header(const uint8_t *buf, int buf_size,
1142  unsigned *obu_size, int *start_pos, enum IAMF_OBU_Type *type,
1143  unsigned *skip_samples, unsigned *discard_padding)
1144 {
1145  GetBitContext gb;
1146  int ret, extension_flag, trimming, start;
1147  unsigned skip = 0, discard = 0;
1148  unsigned size;
1149 
1150  ret = init_get_bits8(&gb, buf, FFMIN(buf_size, MAX_IAMF_OBU_HEADER_SIZE));
1151  if (ret < 0)
1152  return ret;
1153 
1154  *type = get_bits(&gb, 5);
1155  /*redundant =*/ get_bits1(&gb);
1156  trimming = get_bits1(&gb);
1157  extension_flag = get_bits1(&gb);
1158 
1159  *obu_size = get_leb(&gb);
1160  if (*obu_size > INT_MAX)
1161  return AVERROR_INVALIDDATA;
1162 
1163  start = get_bits_count(&gb) / 8;
1164 
1165  if (trimming) {
1166  discard = get_leb(&gb); // num_samples_to_trim_at_end
1167  skip = get_leb(&gb); // num_samples_to_trim_at_start
1168  }
1169 
1170  if (skip_samples)
1171  *skip_samples = skip;
1172  if (discard_padding)
1173  *discard_padding = discard;
1174 
1175  if (extension_flag) {
1176  unsigned int extension_bytes;
1177  extension_bytes = get_leb(&gb);
1178  if (extension_bytes > INT_MAX / 8)
1179  return AVERROR_INVALIDDATA;
1180  skip_bits_long(&gb, extension_bytes * 8);
1181  }
1182 
1183  if (get_bits_left(&gb) < 0)
1184  return AVERROR_INVALIDDATA;
1185 
1186  size = *obu_size + start;
1187  if (size > INT_MAX)
1188  return AVERROR_INVALIDDATA;
1189 
1190  *obu_size -= get_bits_count(&gb) / 8 - start;
1191  *start_pos = size - *obu_size;
1192 
1193  return size;
1194 }
1195 
1197  int max_size, void *log_ctx)
1198 {
1200  int ret;
1201 
1202  while (1) {
1203  unsigned obu_size;
1204  enum IAMF_OBU_Type type;
1205  int start_pos, len, size;
1206 
1207  if ((ret = ffio_ensure_seekback(pb, FFMIN(MAX_IAMF_OBU_HEADER_SIZE, max_size))) < 0)
1208  return ret;
1209  size = avio_read(pb, header, FFMIN(MAX_IAMF_OBU_HEADER_SIZE, max_size));
1210  if (size < 0)
1211  return size;
1213 
1214  len = ff_iamf_parse_obu_header(header, size, &obu_size, &start_pos, &type, NULL, NULL);
1215  if (len < 0 || obu_size > max_size) {
1216  av_log(log_ctx, AV_LOG_ERROR, "Failed to read obu header\n");
1217  avio_seek(pb, -size, SEEK_CUR);
1218  return len;
1219  }
1220 
1222  avio_seek(pb, -size, SEEK_CUR);
1223  break;
1224  }
1225 
1226  avio_seek(pb, -(size - start_pos), SEEK_CUR);
1227  switch (type) {
1229  ret = codec_config_obu(log_ctx, c, pb, obu_size);
1230  break;
1232  ret = audio_element_obu(log_ctx, c, pb, obu_size);
1233  break;
1235  ret = mix_presentation_obu(log_ctx, c, pb, obu_size);
1236  break;
1237  default: {
1238  int64_t offset = avio_skip(pb, obu_size);
1239  if (offset < 0)
1240  ret = offset;
1241  break;
1242  }
1243  }
1244  if (ret < 0) {
1245  av_log(log_ctx, AV_LOG_ERROR, "Failed to read obu type %d\n", type);
1246  return ret;
1247  }
1248  max_size -= obu_size + start_pos;
1249  if (max_size < 0)
1250  return AVERROR_INVALIDDATA;
1251  if (!max_size)
1252  break;
1253  }
1254 
1255  return 0;
1256 }
update_extradata
static int update_extradata(AVCodecParameters *codecpar)
Definition: iamf_parse.c:274
AV_CODEC_ID_PCM_S16LE
@ AV_CODEC_ID_PCM_S16LE
Definition: codec_id.h:337
iamf.h
ff_iamf_free_mix_presentation
void ff_iamf_free_mix_presentation(IAMFMixPresentation **pmix_presentation)
Definition: iamf.c:158
skip_bits_long
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:280
AV_IAMF_SUBMIX_LAYOUT_TYPE_LOUDSPEAKERS
@ AV_IAMF_SUBMIX_LAYOUT_TYPE_LOUDSPEAKERS
The layout follows the loudspeaker sound system convention of ITU-2051-3.
Definition: iamf.h:498
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:216
AVCodecParameters::extradata
uint8_t * extradata
Extra binary data needed for initializing the decoder, codec-dependent.
Definition: codec_par.h:69
MP4DecConfigDescrTag
#define MP4DecConfigDescrTag
Definition: isom.h:398
ffio_init_context
void ffio_init_context(FFIOContext *s, unsigned char *buffer, int buffer_size, int write_flag, void *opaque, int(*read_packet)(void *opaque, uint8_t *buf, int buf_size), int(*write_packet)(void *opaque, const uint8_t *buf, int buf_size), int64_t(*seek)(void *opaque, int64_t offset, int whence))
Definition: aviobuf.c:50
mix
static int mix(int c0, int c1)
Definition: 4xm.c:717
get_bits_left
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:694
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
AV_CH_TOP_SIDE_LEFT
#define AV_CH_TOP_SIDE_LEFT
Definition: channel_layout.h:200
AVCodecParameters::codec_type
enum AVMediaType codec_type
General type of the encoded data.
Definition: codec_par.h:51
out
FILE * out
Definition: movenc.c:55
AV_CHANNEL_LAYOUT_STEREO
#define AV_CHANNEL_LAYOUT_STEREO
Definition: channel_layout.h:395
put_bytes_output
static int put_bytes_output(const PutBitContext *s)
Definition: put_bits.h:99
IAMF_OBU_IA_SEQUENCE_HEADER
@ IAMF_OBU_IA_SEQUENCE_HEADER
Definition: iamf.h:63
ff_mp4_obj_type
const AVCodecTag ff_mp4_obj_type[]
Definition: isom.c:34
AVCodecParameters
This struct describes the properties of an encoded stream.
Definition: codec_par.h:47
AVIAMFAudioElement::default_w
unsigned int default_w
Default weight value as defined in section 3.6 of IAMF.
Definition: iamf.h:393
IAMFAudioElement::nb_substreams
unsigned int nb_substreams
Definition: iamf.h:99
av_iamf_param_definition_alloc
AVIAMFParamDefinition * av_iamf_param_definition_alloc(enum AVIAMFParamDefinitionType type, unsigned int nb_subblocks, size_t *out_size)
Allocates memory for AVIAMFParamDefinition, plus an array of.
Definition: iamf.c:159
AVIAMFAudioElement::layers
AVIAMFLayer ** layers
Definition: iamf.h:359
get_bits_long
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
Definition: get_bits.h:424
int64_t
long long int64_t
Definition: coverity.c:34
init_put_bits
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:62
get_bits_count
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:254
AVChannelLayout::map
AVChannelCustom * map
This member must be used when the channel order is AV_CHANNEL_ORDER_CUSTOM.
Definition: channel_layout.h:370
mask
int mask
Definition: mediacodecdec_common.c:154
AV_CH_TOP_FRONT_RIGHT
#define AV_CH_TOP_FRONT_RIGHT
Definition: channel_layout.h:189
AV_RB32A
#define AV_RB32A(p)
Definition: intreadwrite.h:575
mode
Definition: swscale.c:56
put_bits
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:154
av_channel_layout_channel_from_index
enum AVChannel av_channel_layout_channel_from_index(const AVChannelLayout *channel_layout, unsigned int idx)
Get the channel with the given index in a channel layout.
Definition: channel_layout.c:673
AVCodecParameters::seek_preroll
int seek_preroll
Audio only.
Definition: codec_par.h:214
b
#define b
Definition: input.c:42
AVIOContext::error
int error
contains the error code or 0 if no error happened
Definition: avio.h:239
AVIAMFSubmixLayout::layout_type
enum AVIAMFSubmixLayoutType layout_type
Definition: iamf.h:517
AV_CH_TOP_FRONT_LEFT
#define AV_CH_TOP_FRONT_LEFT
Definition: channel_layout.h:187
put_bits32
static av_unused void put_bits32(PutBitContext *s, uint32_t value)
Write exactly 32 bits into a bitstream.
Definition: put_bits.h:301
AVIAMFParamDefinition
Parameters as defined in section 3.6.1 of IAMF.
Definition: iamf.h:193
AVIAMFSubmixElement::default_mix_gain
AVRational default_mix_gain
Default mix gain value to apply when there are no AVIAMFParamDefinition with element_mix_config's par...
Definition: iamf.h:469
AV_CODEC_ID_FLAC
@ AV_CODEC_ID_FLAC
Definition: codec_id.h:471
AVChannelLayout::order
enum AVChannelOrder order
Channel order used in this layout.
Definition: channel_layout.h:324
iamf_parse.h
AVChannelLayout::mask
uint64_t mask
This member must be used for AV_CHANNEL_ORDER_NATIVE, and may be used for AV_CHANNEL_ORDER_AMBISONIC ...
Definition: channel_layout.h:351
ambisonics_config
static int ambisonics_config(void *s, AVIOContext *pb, IAMFAudioElement *audio_element, const IAMFCodecConfig *codec_config)
Definition: iamf_parse.c:514
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:329
FFIOContext
Definition: avio_internal.h:28
flac_decoder_config
static int flac_decoder_config(IAMFCodecConfig *codec_config, AVIOContext *pb, int len)
Definition: iamf_parse.c:117
IAMFMixPresentation::cmix
const AVIAMFMixPresentation * cmix
Definition: iamf.h:108
AV_CH_TOP_BACK_LEFT
#define AV_CH_TOP_BACK_LEFT
Definition: channel_layout.h:190
MPEG4AudioConfig
Definition: mpeg4audio.h:29
AVIAMFSubmixLayout::digital_peak
AVRational digital_peak
The digital (sampled) peak value of the audio signal, as defined in ITU-1770-4.
Definition: iamf.h:535
skip_bits
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:383
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:337
mpeg4audio.h
AVIAMFSubmixLayout::integrated_loudness
AVRational integrated_loudness
The program integrated loudness information, as defined in ITU-1770-4.
Definition: iamf.h:530
av_iamf_mix_presentation_add_submix
AVIAMFSubmix * av_iamf_mix_presentation_add_submix(AVIAMFMixPresentation *mix_presentation)
Allocate a submix and add it to a given AVIAMFMixPresentation.
IAMFCodecConfig::extradata
uint8_t * extradata
Definition: iamf.h:74
MP4DecSpecificDescrTag
#define MP4DecSpecificDescrTag
Definition: isom.h:399
AOT_ESCAPE
@ AOT_ESCAPE
Y Escape Value.
Definition: mpeg4audio.h:102
AV_CODEC_ID_PCM_S16BE
@ AV_CODEC_ID_PCM_S16BE
Definition: codec_id.h:338
AV_CHAN_TOP_BACK_RIGHT
@ AV_CHAN_TOP_BACK_RIGHT
Definition: channel_layout.h:67
IAMFParamDefinition
Definition: iamf.h:121
fail
#define fail()
Definition: checkasm.h:214
GetBitContext
Definition: get_bits.h:109
AVIAMFSubmixLayout
Submix layout as defined in section 3.7.6 of IAMF.
Definition: iamf.h:514
AV_CH_BACK_LEFT
#define AV_CH_BACK_LEFT
Definition: channel_layout.h:179
IAMF_ANCHOR_ELEMENT_ALBUM
@ IAMF_ANCHOR_ELEMENT_ALBUM
Definition: iamf.h:142
put_bits_left
static int put_bits_left(PutBitContext *s)
Definition: put_bits.h:135
avio_tell
static av_always_inline int64_t avio_tell(AVIOContext *s)
ftell() equivalent for AVIOContext.
Definition: avio.h:494
av_iamf_audio_element_alloc
AVIAMFAudioElement * av_iamf_audio_element_alloc(void)
Allocates a AVIAMFAudioElement, and initializes its fields with default values.
Definition: iamf.c:322
type
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf type
Definition: writing_filters.txt:86
IAMF_OBU_IA_PARAMETER_BLOCK
@ IAMF_OBU_IA_PARAMETER_BLOCK
Definition: iamf.h:41
AVIAMFAudioElement::audio_element_type
enum AVIAMFAudioElementType audio_element_type
Audio element type as defined in section 3.6 of IAMF.
Definition: iamf.h:388
opus_decoder_config
static int opus_decoder_config(IAMFCodecConfig *codec_config, AVIOContext *pb, int len)
Definition: iamf_parse.c:36
AVIAMFReconGain
Recon Gain Info Parameter Data as defined in section 3.8.3 of IAMF.
Definition: iamf.h:148
AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_STEREO
Definition: channel_layout.h:218
param_parse
static int param_parse(void *s, IAMFContext *c, AVIOContext *pb, unsigned int type, const IAMFAudioElement *audio_element, AVIAMFParamDefinition **out_param_definition)
Definition: iamf_parse.c:601
codec_config_obu
static int codec_config_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
Definition: iamf_parse.c:166
AV_DICT_DONT_STRDUP_VAL
#define AV_DICT_DONT_STRDUP_VAL
Take ownership of a value that's been allocated with av_malloc() or another memory allocation functio...
Definition: dict.h:79
IAMFAudioElement::element
AVIAMFAudioElement * element
element backs celement iff the AVIAMFAudioElement is owned by this structure.
Definition: iamf.h:95
AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN
@ AV_IAMF_PARAMETER_DEFINITION_RECON_GAIN
Subblocks are of struct type AVIAMFReconGain.
Definition: iamf.h:181
AVIAMFSubmixElement::annotations
AVDictionary * annotations
A dictionary of strings describing the submix in different languages.
Definition: iamf.h:490
avassert.h
avio_rb32
unsigned int avio_rb32(AVIOContext *s)
Definition: aviobuf.c:764
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:210
FF_ARRAY_ELEMS
#define FF_ARRAY_ELEMS(a)
Definition: sinewin_tablegen.c:29
init_get_bits8
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:544
IAMFCodecConfig::sample_rate
int sample_rate
Definition: iamf.h:72
AV_CH_LOW_FREQUENCY
#define AV_CH_LOW_FREQUENCY
Definition: channel_layout.h:178
IAMF_OBU_IA_MIX_PRESENTATION
@ IAMF_OBU_IA_MIX_PRESENTATION
Definition: iamf.h:40
duration
int64_t duration
Definition: movenc.c:65
AVCodecParameters::frame_size
int frame_size
Audio only.
Definition: codec_par.h:195
av_iamf_submix_add_layout
AVIAMFSubmixLayout * av_iamf_submix_add_layout(AVIAMFSubmix *submix)
Allocate a submix layout and add it to a given AVIAMFSubmix.
avpriv_mpeg4audio_get_config2
int avpriv_mpeg4audio_get_config2(MPEG4AudioConfig *c, const uint8_t *buf, int size, int sync_extension, void *logctx)
Parse MPEG-4 systems extradata from a raw buffer to retrieve audio configuration.
Definition: mpeg4audio.c:165
intreadwrite.h
s
#define s(width, name)
Definition: cbs_vp9.c:198
IAMFSubStream::audio_substream_id
unsigned int audio_substream_id
Definition: iamf.h:83
IAMFLayer::substream_count
unsigned int substream_count
Definition: iamf.h:78
AVIAMFSubmixLayout::dialogue_anchored_loudness
AVRational dialogue_anchored_loudness
The Dialogue loudness information, as defined in ITU-1770-4.
Definition: iamf.h:543
av_realloc_array
void * av_realloc_array(void *ptr, size_t nmemb, size_t size)
Definition: mem.c:217
floor
static __device__ float floor(float a)
Definition: cuda_runtime.h:173
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:201
av_iamf_param_definition_get_subblock
static av_always_inline void * av_iamf_param_definition_get_subblock(const AVIAMFParamDefinition *par, unsigned int idx)
Get the subblock at the specified.
Definition: iamf.h:260
AV_CHANNEL_ORDER_UNSPEC
@ AV_CHANNEL_ORDER_UNSPEC
Only the channel count is specified, without any further information about the channel order.
Definition: channel_layout.h:119
AV_CHAN_SIDE_RIGHT
@ AV_CHAN_SIDE_RIGHT
Definition: channel_layout.h:60
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:41
AVIAMFSubmixElement::headphones_rendering_mode
enum AVIAMFHeadphonesMode headphones_rendering_mode
A value that indicates whether the referenced channel-based Audio Element shall be rendered to stereo...
Definition: iamf.h:478
ffio_read_leb
unsigned int ffio_read_leb(AVIOContext *s)
Read a unsigned integer coded as a variable number of up to eight little-endian bytes,...
Definition: aviobuf.c:930
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:231
AV_CH_TOP_SIDE_RIGHT
#define AV_CH_TOP_SIDE_RIGHT
Definition: channel_layout.h:201
put_bits63
static void put_bits63(PutBitContext *s, int n, uint64_t value)
Write up to 63 bits into a bitstream.
Definition: put_bits.h:344
get_bits.h
AVIAMFLayer::ch_layout
AVChannelLayout ch_layout
Definition: iamf.h:297
IAMFAudioElement::nb_layers
unsigned int nb_layers
Definition: iamf.h:104
ff_iamfdec_read_descriptors
int ff_iamfdec_read_descriptors(IAMFContext *c, AVIOContext *pb, int max_size, void *log_ctx)
Definition: iamf_parse.c:1196
codec_id
enum AVCodecID codec_id
Definition: vaapi_decode.c:410
av_mallocz
#define av_mallocz(s)
Definition: tableprint_vlc.h:31
PutBitContext
Definition: put_bits.h:50
tmp
static uint8_t tmp[40]
Definition: aes_ctr.c:52
AV_CHAN_TOP_SIDE_LEFT
@ AV_CHAN_TOP_SIDE_LEFT
Definition: channel_layout.h:77
AV_CHAN_TOP_SIDE_RIGHT
@ AV_CHAN_TOP_SIDE_RIGHT
Definition: channel_layout.h:78
IAMFAudioElement::audio_element_id
unsigned int audio_element_id
Definition: iamf.h:96
AVIAMFDemixingInfo
Demixing Info Parameter Data as defined in section 3.8.2 of IAMF.
Definition: iamf.h:128
IAMFSoundSystemMap::layout
AVChannelLayout layout
Definition: iamf.h:164
AV_CHANNEL_ORDER_AMBISONIC
@ AV_CHANNEL_ORDER_AMBISONIC
The audio is represented as the decomposition of the sound field into spherical harmonics.
Definition: channel_layout.h:155
NULL
#define NULL
Definition: coverity.c:32
get_leb
static unsigned get_leb(GetBitContext *s)
Read a unsigned integer coded as a variable number of up to eight little-endian bytes,...
Definition: leb.h:35
isom.h
IAMF_OBU_IA_AUDIO_ELEMENT
@ IAMF_OBU_IA_AUDIO_ELEMENT
Definition: iamf.h:39
AVRational
Rational number (pair of numerator and denominator).
Definition: rational.h:58
AVIAMFParamDefinition::duration
unsigned int duration
The accumulated duration of all blocks in this parameter definition, in units of 1 / parameter_rate.
Definition: iamf.h:231
ff_iamf_get_codec_config
static IAMFCodecConfig * ff_iamf_get_codec_config(const IAMFContext *c, unsigned int codec_config_id)
Definition: iamf.h:172
get_bits1
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:391
audio_element_obu
static int audio_element_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
Definition: iamf_parse.c:723
AVIAMFLayer
A layer defining a Channel Layout in the Audio Element.
Definition: iamf.h:294
IAMF_OBU_IA_CODEC_CONFIG
@ IAMF_OBU_IA_CODEC_CONFIG
Definition: iamf.h:38
IAMFSubStream
Definition: iamf.h:82
IAMFAudioElement::layers
IAMFLayer * layers
Definition: iamf.h:103
FLAC_STREAMINFO_SIZE
#define FLAC_STREAMINFO_SIZE
Definition: flac.h:32
AV_DICT_DONT_OVERWRITE
#define AV_DICT_DONT_OVERWRITE
Don't overwrite existing entries.
Definition: dict.h:81
AV_CH_FRONT_CENTER
#define AV_CH_FRONT_CENTER
Definition: channel_layout.h:177
AVCodecParameters::ch_layout
AVChannelLayout ch_layout
Audio only.
Definition: codec_par.h:180
IAMFAudioElement::celement
const AVIAMFAudioElement * celement
Definition: iamf.h:90
AV_CH_FRONT_LEFT_OF_CENTER
#define AV_CH_FRONT_LEFT_OF_CENTER
Definition: channel_layout.h:181
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
ff_mp4_read_descr
int ff_mp4_read_descr(void *logctx, AVIOContext *pb, int *tag)
Definition: isom.c:295
AVCodecParameters::sample_rate
int sample_rate
Audio only.
Definition: codec_par.h:184
AV_CHAN_FRONT_RIGHT_OF_CENTER
@ AV_CHAN_FRONT_RIGHT_OF_CENTER
Definition: channel_layout.h:57
ff_iamf_parse_obu_header
int ff_iamf_parse_obu_header(const uint8_t *buf, int buf_size, unsigned *obu_size, int *start_pos, enum IAMF_OBU_Type *type, unsigned *skip_samples, unsigned *discard_padding)
Definition: iamf_parse.c:1141
AVIAMFLayer::output_gain_flags
unsigned int output_gain_flags
Output gain channel flags as defined in section 3.6.2 of IAMF.
Definition: iamf.h:310
ff_iamf_expanded_scalable_ch_layouts
const AVChannelLayout ff_iamf_expanded_scalable_ch_layouts[13]
Definition: iamf.c:48
AVCodecID
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: codec_id.h:49
AVIAMFSubmixElement::audio_element_id
unsigned int audio_element_id
The id of the Audio Element this submix element references.
Definition: iamf.h:452
AV_CHAN_FRONT_RIGHT
@ AV_CHAN_FRONT_RIGHT
Definition: channel_layout.h:51
AV_CHAN_FRONT_CENTER
@ AV_CHAN_FRONT_CENTER
Definition: channel_layout.h:52
AVCodecParameters::extradata_size
int extradata_size
Size of the extradata content in bytes.
Definition: codec_par.h:73
AVIAMFSubmix
Submix layout as defined in section 3.7 of IAMF.
Definition: iamf.h:556
AVChannelLayout::u
union AVChannelLayout::@491 u
Details about which channels are present in this layout.
AV_CODEC_ID_AAC
@ AV_CODEC_ID_AAC
Definition: codec_id.h:461
IAMFAudioElement
Definition: iamf.h:89
AVIAMFReconGain::subblock_duration
unsigned int subblock_duration
Duration for the given subblock, in units of 1 / parameter_rate.
Definition: iamf.h:156
AVIOContext
Bytestream IO Context.
Definition: avio.h:160
IAMFCodecConfig::nb_samples
unsigned nb_samples
Definition: iamf.h:70
AV_CODEC_ID_PCM_S24LE
@ AV_CODEC_ID_PCM_S24LE
Definition: codec_id.h:349
AVIAMFDemixingInfo::subblock_duration
unsigned int subblock_duration
Duration for the given subblock, in units of 1 / parameter_rate.
Definition: iamf.h:136
ff_codec_get_id
enum AVCodecID ff_codec_get_id(const AVCodecTag *tags, unsigned int tag)
Definition: utils.c:143
IAMFCodecConfig
Definition: iamf.h:66
AVChannelLayout
An AVChannelLayout holds information about the channel layout of audio data.
Definition: channel_layout.h:319
ff_iamf_sound_system_map
const struct IAMFSoundSystemMap ff_iamf_sound_system_map[14]
Definition: iamf.c:120
IAMFCodecConfig::extradata_size
int extradata_size
Definition: iamf.h:73
avio_get_str
int avio_get_str(AVIOContext *pb, int maxlen, char *buf, int buflen)
Read a string from pb into buf.
Definition: aviobuf.c:869
AV_CHAN_LOW_FREQUENCY
@ AV_CHAN_LOW_FREQUENCY
Definition: channel_layout.h:53
size
int size
Definition: twinvq_data.h:10344
av_make_q
static AVRational av_make_q(int num, int den)
Create an AVRational.
Definition: rational.h:71
AV_CHAN_BACK_RIGHT
@ AV_CHAN_BACK_RIGHT
Definition: channel_layout.h:55
IAMFLayer::coupled_substream_count
unsigned int coupled_substream_count
Definition: iamf.h:79
AV_WL32A
#define AV_WL32A(p, v)
Definition: intreadwrite.h:571
MKBETAG
#define MKBETAG(a, b, c, d)
Definition: macros.h:56
AV_CHAN_SIDE_LEFT
@ AV_CHAN_SIDE_LEFT
Definition: channel_layout.h:59
AV_CODEC_ID_OPUS
@ AV_CODEC_ID_OPUS
Definition: codec_id.h:519
av_iamf_mix_presentation_alloc
AVIAMFMixPresentation * av_iamf_mix_presentation_alloc(void)
Allocates a AVIAMFMixPresentation, and initializes its fields with default values.
Definition: iamf.c:520
IAMFContext
Definition: iamf.h:128
header
static const uint8_t header[24]
Definition: sdr2.c:68
avio_r8
int avio_r8(AVIOContext *s)
Definition: aviobuf.c:606
IAMFAudioElement::substreams
IAMFSubStream * substreams
Definition: iamf.h:98
AVIAMFParamDefinition::constant_subblock_duration
unsigned int constant_subblock_duration
The duration of every subblock in the case where all subblocks, with the optional exception of the la...
Definition: iamf.h:238
av_iamf_submix_add_element
AVIAMFSubmixElement * av_iamf_submix_add_element(AVIAMFSubmix *submix)
Allocate a submix element and add it to a given AVIAMFSubmix.
av_channel_layout_retype
int av_channel_layout_retype(AVChannelLayout *channel_layout, enum AVChannelOrder order, int flags)
Change the AVChannelOrder of a channel layout.
Definition: channel_layout.c:885
AVIAMFAudioElement
Information on how to combine one or more audio streams, as defined in section 3.6 of IAMF.
Definition: iamf.h:356
ipcm_decoder_config
static int ipcm_decoder_config(IAMFCodecConfig *codec_config, AVIOContext *pb, int len)
Definition: iamf_parse.c:145
ffio_ensure_seekback
int ffio_ensure_seekback(AVIOContext *s, int64_t buf_size)
Ensures that the requested seekback buffer size will be available.
Definition: aviobuf.c:1026
AV_CHAN_TOP_FRONT_RIGHT
@ AV_CHAN_TOP_FRONT_RIGHT
Definition: channel_layout.h:64
offset
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
Definition: writing_filters.txt:86
AV_CH_TOP_BACK_RIGHT
#define AV_CH_TOP_BACK_RIGHT
Definition: channel_layout.h:192
AV_CHANNEL_ORDER_NATIVE
@ AV_CHANNEL_ORDER_NATIVE
The native channel order, i.e.
Definition: channel_layout.h:125
AVIAMFMixGain
Mix Gain Parameter Data as defined in section 3.8.1 of IAMF.
Definition: iamf.h:77
AV_CH_FRONT_RIGHT_OF_CENTER
#define AV_CH_FRONT_RIGHT_OF_CENTER
Definition: channel_layout.h:182
AV_CHAN_FRONT_LEFT_OF_CENTER
@ AV_CHAN_FRONT_LEFT_OF_CENTER
Definition: channel_layout.h:56
leb.h
av_channel_layout_custom_init
int av_channel_layout_custom_init(AVChannelLayout *channel_layout, int nb_channels)
Initialize a custom channel layout with the specified number of channels.
Definition: channel_layout.c:232
AV_RB16A
#define AV_RB16A(p)
Definition: intreadwrite.h:561
avcodec_parameters_alloc
AVCodecParameters * avcodec_parameters_alloc(void)
Allocate a new AVCodecParameters and set its fields to default values (unknown/invalid/0).
Definition: codec_par.c:57
get_bits64
static uint64_t get_bits64(GetBitContext *s, int n)
Read 0-64 bits.
Definition: get_bits.h:456
log.h
AV_CODEC_ID_NONE
@ AV_CODEC_ID_NONE
Definition: codec_id.h:50
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
AVIAMFParamDefinition::parameter_id
unsigned int parameter_id
Identifier for the parameter substream.
Definition: iamf.h:218
avio_internal.h
AVIAMFLayer::demixing_matrix
AVRational * demixing_matrix
Demixing matrix as defined in section 3.6.3 of IAMF.
Definition: iamf.h:340
av_malloc_array
#define av_malloc_array(a, b)
Definition: tableprint_vlc.h:32
AVIAMFMixPresentation
Information on how to render and mix one or more AVIAMFAudioElement to generate the final audio outpu...
Definition: iamf.h:613
aac_decoder_config
static int aac_decoder_config(IAMFCodecConfig *codec_config, AVIOContext *pb, int len, void *logctx)
Definition: iamf_parse.c:60
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
param_definition
static int param_definition(const IAMFContext *iamf, const IAMFParamDefinition *param_def, AVIOContext *dyn_bc, void *log_ctx)
Definition: iamf_writer.c:678
AV_CODEC_ID_PCM_S32BE
@ AV_CODEC_ID_PCM_S32BE
Definition: codec_id.h:346
AV_CH_SIDE_RIGHT
#define AV_CH_SIDE_RIGHT
Definition: channel_layout.h:185
len
int len
Definition: vorbis_enc_data.h:426
IAMFMixPresentation::count_label
unsigned int count_label
Definition: iamf.h:117
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:264
parse_coupled_substream
static int parse_coupled_substream(AVChannelLayout *out, AVChannelLayout *in, int n)
Definition: iamf_parse.c:350
AVIAMFParamDefinition::nb_subblocks
unsigned int nb_subblocks
Number of subblocks in the array.
Definition: iamf.h:208
AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE
@ AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE
Definition: iamf.h:346
AV_WB32A
#define AV_WB32A(p, v)
Definition: intreadwrite.h:578
AV_WB8
#define AV_WB8(p, d)
Definition: intreadwrite.h:392
tag
uint32_t tag
Definition: movenc.c:2032
ret
ret
Definition: filter_design.txt:187
avio_seek
int64_t avio_seek(AVIOContext *s, int64_t offset, int whence)
fseek() equivalent for AVIOContext.
Definition: aviobuf.c:236
ff_iamf_free_audio_element
void ff_iamf_free_audio_element(IAMFAudioElement **paudio_element)
Definition: iamf.c:143
av_malloc
void * av_malloc(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:98
avio_rb16
unsigned int avio_rb16(AVIOContext *s)
Definition: aviobuf.c:749
AV_INPUT_BUFFER_PADDING_SIZE
#define AV_INPUT_BUFFER_PADDING_SIZE
Definition: defs.h:40
left
Tag MUST be and< 10hcoeff half pel interpolation filter coefficients, hcoeff[0] are the 2 middle coefficients[1] are the next outer ones and so on, resulting in a filter like:...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ... the sign of the coefficients is not explicitly stored but alternates after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,... hcoeff[0] is not explicitly stored but found by subtracting the sum of all stored coefficients with signs from 32 hcoeff[0]=32 - hcoeff[1] - hcoeff[2] - ... a good choice for hcoeff and htaps is htaps=6 hcoeff={40,-10, 2} an alternative which requires more computations at both encoder and decoder side and may or may not be better is htaps=8 hcoeff={42,-14, 6,-2}ref_frames minimum of the number of available reference frames and max_ref_frames for example the first frame after a key frame always has ref_frames=1spatial_decomposition_type wavelet type 0 is a 9/7 symmetric compact integer wavelet 1 is a 5/3 symmetric compact integer wavelet others are reserved stored as delta from last, last is reset to 0 if always_reset||keyframeqlog quality(logarithmic quantizer scale) stored as delta from last, last is reset to 0 if always_reset||keyframemv_scale stored as delta from last, last is reset to 0 if always_reset||keyframe FIXME check that everything works fine if this changes between framesqbias dequantization bias stored as delta from last, last is reset to 0 if always_reset||keyframeblock_max_depth maximum depth of the block tree stored as delta from last, last is reset to 0 if always_reset||keyframequant_table quantization tableHighlevel bitstream structure:==============================--------------------------------------------|Header|--------------------------------------------|------------------------------------|||Block0||||split?||||yes no||||......... intra?||||:Block01 :yes no||||:Block02 :....... ..........||||:Block03 ::y DC ::ref index:||||:Block04 ::cb DC ::motion x :||||......... :cr DC ::motion y :||||....... ..........|||------------------------------------||------------------------------------|||Block1|||...|--------------------------------------------|------------ ------------ ------------|||Y subbands||Cb subbands||Cr subbands||||--- ---||--- ---||--- ---|||||LL0||HL0||||LL0||HL0||||LL0||HL0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||LH0||HH0||||LH0||HH0||||LH0||HH0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HL1||LH1||||HL1||LH1||||HL1||LH1|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HH1||HL2||||HH1||HL2||||HH1||HL2|||||...||...||...|||------------ ------------ ------------|--------------------------------------------Decoding process:=================------------|||Subbands|------------||||------------|Intra DC||||LL0 subband prediction ------------|\ Dequantization ------------------- \||Reference frames|\ IDWT|------- -------|Motion \|||Frame 0||Frame 1||Compensation . OBMC v -------|------- -------|--------------. \------> Frame n output Frame Frame<----------------------------------/|...|------------------- Range Coder:============Binary Range Coder:------------------- The implemented range coder is an adapted version based upon "Range encoding: an algorithm for removing redundancy from a digitised message." by G. N. N. Martin. The symbols encoded by the Snow range coder are bits(0|1). The associated probabilities are not fix but change depending on the symbol mix seen so far. bit seen|new state ---------+----------------------------------------------- 0|256 - state_transition_table[256 - old_state];1|state_transition_table[old_state];state_transition_table={ 0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, 165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, 180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, 195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, 210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, 226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, 241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0};FIXME Range Coding of integers:------------------------- FIXME Neighboring Blocks:===================left and top are set to the respective blocks unless they are outside of the image in which case they are set to the Null block top-left is set to the top left block unless it is outside of the image in which case it is set to the left block if this block has no larger parent block or it is at the left side of its parent block and the top right block is not outside of the image then the top right block is used for top-right else the top-left block is used Null block y, cb, cr are 128 level, ref, mx and my are 0 Motion Vector Prediction:=========================1. the motion vectors of all the neighboring blocks are scaled to compensate for the difference of reference frames scaled_mv=(mv *(256 *(current_reference+1)/(mv.reference+1))+128)> the median of the scaled left
Definition: snow.txt:386
U
#define U(x)
Definition: vpx_arith.h:37
IAMF_OBU_Type
IAMF_OBU_Type
Definition: iamf.h:37
label_string
static int label_string(AVIOContext *pb, char **label)
Definition: iamf_parse.c:907
IAMFMixPresentation
Definition: iamf.h:107
IAMFMixPresentation::language_label
char ** language_label
Definition: iamf.h:118
AVIAMFSubmix::default_mix_gain
AVRational default_mix_gain
Default mix gain value to apply when there are no AVIAMFParamDefinition with output_mix_config's para...
Definition: iamf.h:603
AVIAMFSubmixLayout::album_anchored_loudness
AVRational album_anchored_loudness
The Album loudness information, as defined in ITU-1770-4.
Definition: iamf.h:547
av_channel_layout_subset
uint64_t av_channel_layout_subset(const AVChannelLayout *channel_layout, uint64_t mask)
Find out what channels from a given set are present in a channel layout, without regard for their pos...
Definition: channel_layout.c:865
AV_CHAN_TOP_BACK_LEFT
@ AV_CHAN_TOP_BACK_LEFT
Definition: channel_layout.h:65
AVIAMFSubmixLayout::true_peak
AVRational true_peak
The true peak of the audio signal, as defined in ITU-1770-4.
Definition: iamf.h:539
mode
mode
Definition: ebur128.h:83
AVIAMFSubmixLayout::sound_system
AVChannelLayout sound_system
Channel layout matching one of Sound Systems A to J of ITU-2051-3, plus 7.1.2ch, 3....
Definition: iamf.h:525
IAMFCodecConfig::codec_id
enum AVCodecID codec_id
Definition: iamf.h:68
IAMFCodecConfig::codec_config_id
unsigned codec_config_id
Definition: iamf.h:67
avio_read
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:615
sign_extend
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:132
AV_CHAN_BACK_LEFT
@ AV_CHAN_BACK_LEFT
Definition: channel_layout.h:54
AVIOContext::eof_reached
int eof_reached
true if was unable to read due to error or eof
Definition: avio.h:238
IAMF_ANCHOR_ELEMENT_DIALOGUE
@ IAMF_ANCHOR_ELEMENT_DIALOGUE
Definition: iamf.h:141
AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN
@ AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN
Subblocks are of struct type AVIAMFMixGain.
Definition: iamf.h:173
MAX_IAMF_OBU_HEADER_SIZE
#define MAX_IAMF_OBU_HEADER_SIZE
Definition: iamf.h:34
avio_skip
int64_t avio_skip(AVIOContext *s, int64_t offset)
Skip given number of bytes forward.
Definition: aviobuf.c:321
mix_presentation_obu
static int mix_presentation_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
Definition: iamf_parse.c:924
av_iamf_audio_element_add_layer
AVIAMFLayer * av_iamf_audio_element_add_layer(AVIAMFAudioElement *audio_element)
Allocate a layer and add it to a given AVIAMFAudioElement.
av_channel_layout_copy
int av_channel_layout_copy(AVChannelLayout *dst, const AVChannelLayout *src)
Make a copy of a channel layout.
Definition: channel_layout.c:449
av_strdup
char * av_strdup(const char *s)
Duplicate a string.
Definition: mem.c:272
AV_CODEC_ID_PCM_S32LE
@ AV_CODEC_ID_PCM_S32LE
Definition: codec_id.h:345
AVIAMFAudioElement::demixing_info
AVIAMFParamDefinition * demixing_info
Demixing information used to reconstruct a scalable channel audio representation.
Definition: iamf.h:376
AVIAMFDemixingInfo::dmixp_mode
unsigned int dmixp_mode
Pre-defined combination of demixing parameters.
Definition: iamf.h:140
mem.h
AVIAMFSubmix::output_mix_config
AVIAMFParamDefinition * output_mix_config
Information required for post-processing the mixed audio signal to generate the audio signal for play...
Definition: iamf.h:595
AVIAMFLayer::ambisonics_mode
enum AVIAMFAmbisonicsMode ambisonics_mode
Ambisonics mode as defined in section 3.6.3 of IAMF.
Definition: iamf.h:328
sample_format
enum AVSampleFormat sample_format
Definition: mediacodecdec_common.c:101
flush_put_bits
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:153
AV_CHANNEL_LAYOUT_BINAURAL
#define AV_CHANNEL_LAYOUT_BINAURAL
Definition: channel_layout.h:431
AV_CHANNEL_LAYOUT_MONO
#define AV_CHANNEL_LAYOUT_MONO
Definition: channel_layout.h:394
AV_IAMF_SUBMIX_LAYOUT_TYPE_BINAURAL
@ AV_IAMF_SUBMIX_LAYOUT_TYPE_BINAURAL
The layout is binaural.
Definition: iamf.h:505
AVIAMFLayer::flags
unsigned int flags
A bitmask which may contain a combination of AV_IAMF_LAYER_FLAG_* flags.
Definition: iamf.h:302
av_free
#define av_free(p)
Definition: tableprint_vlc.h:34
AVIAMFLayer::output_gain
AVRational output_gain
Output gain as defined in section 3.6.2 of IAMF.
Definition: iamf.h:316
AVCodecParameters::codec_id
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: codec_par.h:55
av_dict_set
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
Definition: dict.c:86
AV_CHAN_TOP_FRONT_LEFT
@ AV_CHAN_TOP_FRONT_LEFT
Definition: channel_layout.h:62
AVIAMFSubmixElement::element_mix_config
AVIAMFParamDefinition * element_mix_config
Information required required for applying any processing to the referenced and rendered Audio Elemen...
Definition: iamf.h:461
AV_CHAN_AMBISONIC_BASE
@ AV_CHAN_AMBISONIC_BASE
Range of channels between AV_CHAN_AMBISONIC_BASE and AV_CHAN_AMBISONIC_END represent Ambisonic compon...
Definition: channel_layout.h:108
AVIAMFParamDefinition::parameter_rate
unsigned int parameter_rate
Sample rate for the parameter substream.
Definition: iamf.h:222
ff_iamf_scalable_ch_layouts
const AVChannelLayout ff_iamf_scalable_ch_layouts[10]
Definition: iamf.c:27
AVIAMFSubmixElement
Submix element as defined in section 3.7 of IAMF.
Definition: iamf.h:446
AV_IAMF_AUDIO_ELEMENT_TYPE_CHANNEL
@ AV_IAMF_AUDIO_ELEMENT_TYPE_CHANNEL
Definition: iamf.h:345
IAMFAudioElement::codec_config_id
unsigned int codec_config_id
Definition: iamf.h:101
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_CH_BACK_RIGHT
#define AV_CH_BACK_RIGHT
Definition: channel_layout.h:180
ffio_read_size
int ffio_read_size(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:665
AVIAMFAudioElement::recon_gain_info
AVIAMFParamDefinition * recon_gain_info
Recon gain information used to reconstruct a scalable channel audio representation.
Definition: iamf.h:383
AV_CHAN_FRONT_LEFT
@ AV_CHAN_FRONT_LEFT
Definition: channel_layout.h:50
IAMFMixPresentation::mix
AVIAMFMixPresentation * mix
mix backs cmix iff the AVIAMFMixPresentation is owned by this structure.
Definition: iamf.h:113
scalable_channel_layout_config
static int scalable_channel_layout_config(void *s, AVIOContext *pb, IAMFAudioElement *audio_element, const IAMFCodecConfig *codec_config)
Definition: iamf_parse.c:385
flac.h
AV_RB24
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_RB24
Definition: bytestream.h:97
IAMFSubStream::codecpar
AVCodecParameters * codecpar
Definition: iamf.h:86
put_bits.h
IAMFCodecConfig::audio_roll_distance
int audio_roll_distance
Definition: iamf.h:71
IAMFMixPresentation::mix_presentation_id
unsigned int mix_presentation_id
Definition: iamf.h:114
AV_CODEC_ID_PCM_S24BE
@ AV_CODEC_ID_PCM_S24BE
Definition: codec_id.h:350
AVChannelCustom::id
enum AVChannel id
Definition: channel_layout.h:284
skip
static void BS_FUNC() skip(BSCTX *bc, unsigned int n)
Skip n bits in the buffer.
Definition: bitstream_template.h:383
AV_WL16A
#define AV_WL16A(p, v)
Definition: intreadwrite.h:557
AV_CH_SIDE_LEFT
#define AV_CH_SIDE_LEFT
Definition: channel_layout.h:184
AV_IAMF_PARAMETER_DEFINITION_DEMIXING
@ AV_IAMF_PARAMETER_DEFINITION_DEMIXING
Subblocks are of struct type AVIAMFDemixingInfo.
Definition: iamf.h:177
MPEG4AudioConfig::sample_rate
int sample_rate
Definition: mpeg4audio.h:32