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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, size);
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 = put_bits_left(&pb);
308  while (ret >= 32) {
309  put_bits32(&pb, get_bits_long(&gb, 32));
310  ret -= 32;
311  }
312  put_bits(&pb, ret, get_bits_long(&gb, ret));
313  flush_put_bits(&pb);
314 
315  memcpy(codecpar->extradata, buf, put_bytes_output(&pb));
316  break;
317  }
318  case AV_CODEC_ID_FLAC: {
319  uint8_t buf[13];
320 
321  init_put_bits(&pb, buf, sizeof(buf));
322  ret = init_get_bits8(&gb, codecpar->extradata, codecpar->extradata_size);
323  if (ret < 0)
324  return ret;
325 
326  put_bits32(&pb, get_bits_long(&gb, 32)); // min/max blocksize
327  put_bits63(&pb, 48, get_bits64(&gb, 48)); // min/max framesize
328  put_bits(&pb, 20, get_bits(&gb, 20)); // samplerate
329  skip_bits(&gb, 3);
330  put_bits(&pb, 3, codecpar->ch_layout.nb_channels - 1);
331  ret = put_bits_left(&pb);
332  put_bits(&pb, ret, get_bits(&gb, ret));
333  flush_put_bits(&pb);
334 
335  memcpy(codecpar->extradata, buf, sizeof(buf));
336  break;
337  }
338  }
339 
340  return 0;
341 }
342 
344  IAMFAudioElement *audio_element,
345  const IAMFCodecConfig *codec_config)
346 {
347  int nb_layers, k = 0;
348 
349  nb_layers = avio_r8(pb) >> 5; // get_bits(&gb, 3);
350  // skip_bits(&gb, 5); //reserved
351 
352  if (nb_layers > 6 || nb_layers == 0)
353  return AVERROR_INVALIDDATA;
354 
355  audio_element->layers = av_calloc(nb_layers, sizeof(*audio_element->layers));
356  if (!audio_element->layers)
357  return AVERROR(ENOMEM);
358 
359  audio_element->nb_layers = nb_layers;
360  for (int i = 0; i < nb_layers; i++) {
361  AVIAMFLayer *layer;
362  int loudspeaker_layout, output_gain_is_present_flag;
363  int substream_count, coupled_substream_count;
364  int expanded_loudspeaker_layout = -1;
365  int ret, byte = avio_r8(pb);
366 
367  layer = av_iamf_audio_element_add_layer(audio_element->element);
368  if (!layer)
369  return AVERROR(ENOMEM);
370 
371  loudspeaker_layout = byte >> 4; // get_bits(&gb, 4);
372  output_gain_is_present_flag = (byte >> 3) & 1; //get_bits1(&gb);
373  if ((byte >> 2) & 1)
374  layer->flags |= AV_IAMF_LAYER_FLAG_RECON_GAIN;
375  substream_count = avio_r8(pb);
376  coupled_substream_count = avio_r8(pb);
377 
378  if (substream_count + k > audio_element->nb_substreams)
379  return AVERROR_INVALIDDATA;
380 
381  audio_element->layers[i].substream_count = substream_count;
382  audio_element->layers[i].coupled_substream_count = coupled_substream_count;
383  if (output_gain_is_present_flag) {
384  layer->output_gain_flags = avio_r8(pb) >> 2; // get_bits(&gb, 6);
385  layer->output_gain = av_make_q(sign_extend(avio_rb16(pb), 16), 1 << 8);
386  }
387 
388  if (!i && loudspeaker_layout == 15)
389  expanded_loudspeaker_layout = avio_r8(pb);
390  if (expanded_loudspeaker_layout > 0 && expanded_loudspeaker_layout < 13)
391  av_channel_layout_copy(&layer->ch_layout, &ff_iamf_expanded_scalable_ch_layouts[expanded_loudspeaker_layout]);
392  else if (loudspeaker_layout < 10)
393  av_channel_layout_copy(&layer->ch_layout, &ff_iamf_scalable_ch_layouts[loudspeaker_layout]);
394  else
396  .nb_channels = substream_count +
397  coupled_substream_count };
398 
399  for (int j = 0; j < substream_count; j++) {
400  IAMFSubStream *substream = &audio_element->substreams[k++];
401 
402  substream->codecpar->ch_layout = coupled_substream_count-- > 0 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO :
404 
405  ret = update_extradata(substream->codecpar);
406  if (ret < 0)
407  return ret;
408  }
409 
410  }
411 
412  return 0;
413 }
414 
415 static int ambisonics_config(void *s, AVIOContext *pb,
416  IAMFAudioElement *audio_element,
417  const IAMFCodecConfig *codec_config)
418 {
419  AVIAMFLayer *layer;
420  unsigned ambisonics_mode;
421  int output_channel_count, substream_count, order;
422  int ret;
423 
424  ambisonics_mode = ffio_read_leb(pb);
425  if (ambisonics_mode > 1)
426  return AVERROR_INVALIDDATA;
427 
428  output_channel_count = avio_r8(pb); // C
429  substream_count = avio_r8(pb); // N
430  if (audio_element->nb_substreams != substream_count || output_channel_count == 0)
431  return AVERROR_INVALIDDATA;
432 
433  order = floor(sqrt(output_channel_count - 1));
434  /* incomplete order - some harmonics are missing */
435  if ((order + 1) * (order + 1) != output_channel_count)
436  return AVERROR_INVALIDDATA;
437 
438  audio_element->layers = av_mallocz(sizeof(*audio_element->layers));
439  if (!audio_element->layers)
440  return AVERROR(ENOMEM);
441 
442  audio_element->nb_layers = 1;
443  audio_element->layers->substream_count = substream_count;
444 
445  layer = av_iamf_audio_element_add_layer(audio_element->element);
446  if (!layer)
447  return AVERROR(ENOMEM);
448 
449  layer->ambisonics_mode = ambisonics_mode;
450  if (ambisonics_mode == 0) {
451  for (int i = 0; i < substream_count; i++) {
452  IAMFSubStream *substream = &audio_element->substreams[i];
453 
455 
456  ret = update_extradata(substream->codecpar);
457  if (ret < 0)
458  return ret;
459  }
460 
462  layer->ch_layout.nb_channels = output_channel_count;
463  layer->ch_layout.u.map = av_calloc(output_channel_count, sizeof(*layer->ch_layout.u.map));
464  if (!layer->ch_layout.u.map)
465  return AVERROR(ENOMEM);
466 
467  for (int i = 0; i < output_channel_count; i++)
468  layer->ch_layout.u.map[i].id = avio_r8(pb) + AV_CHAN_AMBISONIC_BASE;
469  } else {
470  int coupled_substream_count = avio_r8(pb); // M
471  int nb_demixing_matrix = substream_count + coupled_substream_count;
472  int demixing_matrix_size = nb_demixing_matrix * output_channel_count;
473 
474  audio_element->layers->coupled_substream_count = coupled_substream_count;
475 
476  layer->ch_layout = (AVChannelLayout){ .order = AV_CHANNEL_ORDER_AMBISONIC, .nb_channels = output_channel_count };
477  layer->demixing_matrix = av_malloc_array(demixing_matrix_size, sizeof(*layer->demixing_matrix));
478  if (!layer->demixing_matrix)
479  return AVERROR(ENOMEM);
480 
481  for (int i = 0; i < demixing_matrix_size; i++)
482  layer->demixing_matrix[i] = av_make_q(sign_extend(avio_rb16(pb), 16), 1 << 8);
483 
484  for (int i = 0; i < substream_count; i++) {
485  IAMFSubStream *substream = &audio_element->substreams[i];
486 
487  substream->codecpar->ch_layout = coupled_substream_count-- > 0 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO :
489 
490 
491  ret = update_extradata(substream->codecpar);
492  if (ret < 0)
493  return ret;
494  }
495  }
496 
497  return 0;
498 }
499 
500 static int param_parse(void *s, IAMFContext *c, AVIOContext *pb,
501  unsigned int type,
502  const IAMFAudioElement *audio_element,
503  AVIAMFParamDefinition **out_param_definition)
504 {
506  AVIAMFParamDefinition *param;
507  unsigned int parameter_id, parameter_rate, mode;
508  unsigned int duration = 0, constant_subblock_duration = 0, nb_subblocks = 0;
509  unsigned int total_duration = 0;
510  size_t param_size;
511 
512  parameter_id = ffio_read_leb(pb);
513 
514  for (int i = 0; i < c->nb_param_definitions; i++)
515  if (c->param_definitions[i]->param->parameter_id == parameter_id) {
516  param_definition = c->param_definitions[i];
517  break;
518  }
519 
520  parameter_rate = ffio_read_leb(pb);
521  mode = avio_r8(pb) >> 7;
522 
523  if (mode == 0) {
524  duration = ffio_read_leb(pb);
525  if (!duration)
526  return AVERROR_INVALIDDATA;
527  constant_subblock_duration = ffio_read_leb(pb);
528  if (constant_subblock_duration == 0)
529  nb_subblocks = ffio_read_leb(pb);
530  else {
531  nb_subblocks = duration / constant_subblock_duration;
532  total_duration = duration;
533  }
534  }
535 
536  param = av_iamf_param_definition_alloc(type, nb_subblocks, &param_size);
537  if (!param)
538  return AVERROR(ENOMEM);
539 
540  for (int i = 0; i < nb_subblocks; i++) {
541  void *subblock = av_iamf_param_definition_get_subblock(param, i);
542  unsigned int subblock_duration = constant_subblock_duration;
543 
544  if (constant_subblock_duration == 0) {
545  subblock_duration = ffio_read_leb(pb);
546  total_duration += subblock_duration;
547  } else if (i == nb_subblocks - 1)
548  subblock_duration = duration - i * constant_subblock_duration;
549 
550  switch (type) {
552  AVIAMFMixGain *mix = subblock;
553  mix->subblock_duration = subblock_duration;
554  break;
555  }
557  AVIAMFDemixingInfo *demix = subblock;
558  demix->subblock_duration = subblock_duration;
559  // DefaultDemixingInfoParameterData
560  av_assert0(audio_element);
561  demix->dmixp_mode = avio_r8(pb) >> 5;
562  audio_element->element->default_w = avio_r8(pb) >> 4;
563  break;
564  }
566  AVIAMFReconGain *recon = subblock;
567  recon->subblock_duration = subblock_duration;
568  break;
569  }
570  default:
571  av_free(param);
572  return AVERROR_INVALIDDATA;
573  }
574  }
575 
576  if (!mode && !constant_subblock_duration && total_duration != duration) {
577  av_log(s, AV_LOG_ERROR, "Invalid subblock durations in parameter_id %u\n", parameter_id);
578  av_free(param);
579  return AVERROR_INVALIDDATA;
580  }
581 
582  param->parameter_id = parameter_id;
583  param->parameter_rate = parameter_rate;
584  param->duration = duration;
585  param->constant_subblock_duration = constant_subblock_duration;
586  param->nb_subblocks = nb_subblocks;
587 
588  if (param_definition) {
589  if (param_definition->param_size != param_size || memcmp(param_definition->param, param, param_size)) {
590  av_log(s, AV_LOG_ERROR, "Incosistent parameters for parameter_id %u\n", parameter_id);
591  av_free(param);
592  return AVERROR_INVALIDDATA;
593  }
594  } else {
595  IAMFParamDefinition **tmp = av_realloc_array(c->param_definitions, c->nb_param_definitions + 1,
596  sizeof(*c->param_definitions));
597  if (!tmp) {
598  av_free(param);
599  return AVERROR(ENOMEM);
600  }
601  c->param_definitions = tmp;
602 
604  if (!param_definition) {
605  av_free(param);
606  return AVERROR(ENOMEM);
607  }
608  param_definition->param = param;
609  param_definition->mode = !mode;
610  param_definition->param_size = param_size;
611  param_definition->audio_element = audio_element;
612 
613  c->param_definitions[c->nb_param_definitions++] = param_definition;
614  }
615 
616  av_assert0(out_param_definition);
617  *out_param_definition = param;
618 
619  return 0;
620 }
621 
622 static int audio_element_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
623 {
624  const IAMFCodecConfig *codec_config;
625  AVIAMFAudioElement *element;
626  IAMFAudioElement **tmp, *audio_element = NULL;
627  FFIOContext b;
628  AVIOContext *pbc;
629  uint8_t *buf;
630  unsigned audio_element_id, nb_substreams, codec_config_id, num_parameters;
631  int audio_element_type, ret;
632 
633  buf = av_malloc(len);
634  if (!buf)
635  return AVERROR(ENOMEM);
636 
637  ret = ffio_read_size(pb, buf, len);
638  if (ret < 0)
639  goto fail;
640 
641  ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
642  pbc = &b.pub;
643 
644  audio_element_id = ffio_read_leb(pbc);
645 
646  for (int i = 0; i < c->nb_audio_elements; i++)
647  if (c->audio_elements[i]->audio_element_id == audio_element_id) {
648  av_log(s, AV_LOG_ERROR, "Duplicate audio_element_id %d\n", audio_element_id);
650  goto fail;
651  }
652 
653  audio_element_type = avio_r8(pbc) >> 5;
654  if (audio_element_type > AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE) {
655  av_log(s, AV_LOG_DEBUG, "Unknown audio_element_type referenced in an audio element. Ignoring\n");
656  ret = 0;
657  goto fail;
658  }
659 
660  codec_config_id = ffio_read_leb(pbc);
661 
662  codec_config = ff_iamf_get_codec_config(c, codec_config_id);
663  if (!codec_config) {
664  av_log(s, AV_LOG_ERROR, "Non existant codec config id %d referenced in an audio element\n", codec_config_id);
666  goto fail;
667  }
668 
669  if (codec_config->codec_id == AV_CODEC_ID_NONE) {
670  av_log(s, AV_LOG_DEBUG, "Unknown codec id referenced in an audio element. Ignoring\n");
671  ret = 0;
672  goto fail;
673  }
674 
675  tmp = av_realloc_array(c->audio_elements, c->nb_audio_elements + 1, sizeof(*c->audio_elements));
676  if (!tmp) {
677  ret = AVERROR(ENOMEM);
678  goto fail;
679  }
680  c->audio_elements = tmp;
681 
682  audio_element = av_mallocz(sizeof(*audio_element));
683  if (!audio_element) {
684  ret = AVERROR(ENOMEM);
685  goto fail;
686  }
687 
688  nb_substreams = ffio_read_leb(pbc);
689  audio_element->codec_config_id = codec_config_id;
690  audio_element->audio_element_id = audio_element_id;
691  audio_element->substreams = av_calloc(nb_substreams, sizeof(*audio_element->substreams));
692  if (!audio_element->substreams) {
693  ret = AVERROR(ENOMEM);
694  goto fail;
695  }
696  audio_element->nb_substreams = nb_substreams;
697 
698  element = audio_element->element = av_iamf_audio_element_alloc();
699  if (!element) {
700  ret = AVERROR(ENOMEM);
701  goto fail;
702  }
703  audio_element->celement = element;
704 
705  element->audio_element_type = audio_element_type;
706 
707  for (int i = 0; i < audio_element->nb_substreams; i++) {
708  IAMFSubStream *substream = &audio_element->substreams[i];
709 
710  substream->codecpar = avcodec_parameters_alloc();
711  if (!substream->codecpar) {
712  ret = AVERROR(ENOMEM);
713  goto fail;
714  }
715 
716  substream->audio_substream_id = ffio_read_leb(pbc);
717 
718  substream->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
719  substream->codecpar->codec_id = codec_config->codec_id;
720  substream->codecpar->frame_size = codec_config->nb_samples;
721  substream->codecpar->sample_rate = codec_config->sample_rate;
722  substream->codecpar->seek_preroll = -codec_config->audio_roll_distance * codec_config->nb_samples;
723 
724  switch(substream->codecpar->codec_id) {
725  case AV_CODEC_ID_AAC:
726  case AV_CODEC_ID_FLAC:
727  case AV_CODEC_ID_OPUS:
729  if (!substream->codecpar->extradata) {
730  ret = AVERROR(ENOMEM);
731  goto fail;
732  }
733  memcpy(substream->codecpar->extradata, codec_config->extradata, codec_config->extradata_size);
734  memset(substream->codecpar->extradata + codec_config->extradata_size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
735  substream->codecpar->extradata_size = codec_config->extradata_size;
736  break;
737  }
738  }
739 
740  num_parameters = ffio_read_leb(pbc);
741  if (num_parameters > 2 && audio_element_type == 0) {
742  av_log(s, AV_LOG_ERROR, "Audio Element parameter count %u is invalid"
743  " for Channel representations\n", num_parameters);
745  goto fail;
746  }
747  if (num_parameters && audio_element_type != 0) {
748  av_log(s, AV_LOG_ERROR, "Audio Element parameter count %u is invalid"
749  " for Scene representations\n", num_parameters);
751  goto fail;
752  }
753 
754  for (int i = 0; i < num_parameters; i++) {
755  unsigned type;
756 
757  type = ffio_read_leb(pbc);
761  if (element->demixing_info) {
763  goto fail;
764  }
765  ret = param_parse(s, c, pbc, type, audio_element, &element->demixing_info);
767  if (element->recon_gain_info) {
769  goto fail;
770  }
771  ret = param_parse(s, c, pbc, type, audio_element, &element->recon_gain_info);
772  } else {
773  unsigned param_definition_size = ffio_read_leb(pbc);
774  avio_skip(pbc, param_definition_size);
775  }
776  if (ret < 0)
777  goto fail;
778  }
779 
780  if (audio_element_type == AV_IAMF_AUDIO_ELEMENT_TYPE_CHANNEL) {
781  ret = scalable_channel_layout_config(s, pbc, audio_element, codec_config);
782  if (ret < 0)
783  goto fail;
784  } else if (audio_element_type == AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE) {
785  ret = ambisonics_config(s, pbc, audio_element, codec_config);
786  if (ret < 0)
787  goto fail;
788  } else {
789  av_assert0(0);
790  }
791 
792  c->audio_elements[c->nb_audio_elements++] = audio_element;
793 
794  len -= avio_tell(pbc);
795  if (len)
796  av_log(s, AV_LOG_WARNING, "Underread in audio_element_obu. %d bytes left at the end\n", len);
797 
798  ret = 0;
799 fail:
800  av_free(buf);
801  if (ret < 0)
802  ff_iamf_free_audio_element(&audio_element);
803  return ret;
804 }
805 
806 static int label_string(AVIOContext *pb, char **label)
807 {
808  uint8_t buf[128];
809 
810  avio_get_str(pb, sizeof(buf), buf, sizeof(buf));
811 
812  if (pb->error)
813  return pb->error;
814  if (pb->eof_reached)
815  return AVERROR_INVALIDDATA;
816  *label = av_strdup(buf);
817  if (!*label)
818  return AVERROR(ENOMEM);
819 
820  return 0;
821 }
822 
823 static int mix_presentation_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
824 {
826  IAMFMixPresentation **tmp, *mix_presentation = NULL;
827  FFIOContext b;
828  AVIOContext *pbc;
829  uint8_t *buf;
830  unsigned nb_submixes, mix_presentation_id;
831  int ret;
832 
833  buf = av_malloc(len);
834  if (!buf)
835  return AVERROR(ENOMEM);
836 
837  ret = ffio_read_size(pb, buf, len);
838  if (ret < 0)
839  goto fail;
840 
841  ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
842  pbc = &b.pub;
843 
844  mix_presentation_id = ffio_read_leb(pbc);
845 
846  for (int i = 0; i < c->nb_mix_presentations; i++)
847  if (c->mix_presentations[i]->mix_presentation_id == mix_presentation_id) {
848  av_log(s, AV_LOG_ERROR, "Duplicate mix_presentation_id %d\n", mix_presentation_id);
850  goto fail;
851  }
852 
853  tmp = av_realloc_array(c->mix_presentations, c->nb_mix_presentations + 1, sizeof(*c->mix_presentations));
854  if (!tmp) {
855  ret = AVERROR(ENOMEM);
856  goto fail;
857  }
858  c->mix_presentations = tmp;
859 
860  mix_presentation = av_mallocz(sizeof(*mix_presentation));
861  if (!mix_presentation) {
862  ret = AVERROR(ENOMEM);
863  goto fail;
864  }
865 
866  mix_presentation->mix_presentation_id = mix_presentation_id;
867  mix = mix_presentation->mix = av_iamf_mix_presentation_alloc();
868  if (!mix) {
869  ret = AVERROR(ENOMEM);
870  goto fail;
871  }
872  mix_presentation->cmix = mix;
873 
874  mix_presentation->count_label = ffio_read_leb(pbc);
875  mix_presentation->language_label = av_calloc(mix_presentation->count_label,
876  sizeof(*mix_presentation->language_label));
877  if (!mix_presentation->language_label) {
878  mix_presentation->count_label = 0;
879  ret = AVERROR(ENOMEM);
880  goto fail;
881  }
882 
883  for (int i = 0; i < mix_presentation->count_label; i++) {
884  ret = label_string(pbc, &mix_presentation->language_label[i]);
885  if (ret < 0)
886  goto fail;
887  }
888 
889  for (int i = 0; i < mix_presentation->count_label; i++) {
890  char *annotation = NULL;
891  ret = label_string(pbc, &annotation);
892  if (ret < 0)
893  goto fail;
894  ret = av_dict_set(&mix->annotations, mix_presentation->language_label[i], annotation,
896  if (ret < 0)
897  goto fail;
898  }
899 
900  nb_submixes = ffio_read_leb(pbc);
901  for (int i = 0; i < nb_submixes; i++) {
902  AVIAMFSubmix *sub_mix;
903  unsigned nb_elements, nb_layouts;
904 
906  if (!sub_mix) {
907  ret = AVERROR(ENOMEM);
908  goto fail;
909  }
910 
911  nb_elements = ffio_read_leb(pbc);
912  for (int j = 0; j < nb_elements; j++) {
913  AVIAMFSubmixElement *submix_element;
914  IAMFAudioElement *audio_element = NULL;
915  unsigned int rendering_config_extension_size;
916 
917  submix_element = av_iamf_submix_add_element(sub_mix);
918  if (!submix_element) {
919  ret = AVERROR(ENOMEM);
920  goto fail;
921  }
922 
923  submix_element->audio_element_id = ffio_read_leb(pbc);
924 
925  for (int k = 0; k < c->nb_audio_elements; k++)
926  if (c->audio_elements[k]->audio_element_id == submix_element->audio_element_id) {
927  audio_element = c->audio_elements[k];
928  break;
929  }
930 
931  if (!audio_element) {
932  av_log(s, AV_LOG_ERROR, "Invalid Audio Element with id %u referenced by Mix Parameters %u\n",
933  submix_element->audio_element_id, mix_presentation_id);
935  goto fail;
936  }
937 
938  for (int k = 0; k < mix_presentation->count_label; k++) {
939  char *annotation = NULL;
940  ret = label_string(pbc, &annotation);
941  if (ret < 0)
942  goto fail;
943  ret = av_dict_set(&submix_element->annotations, mix_presentation->language_label[k], annotation,
945  if (ret < 0)
946  goto fail;
947  }
948 
949  submix_element->headphones_rendering_mode = avio_r8(pbc) >> 6;
950 
951  rendering_config_extension_size = ffio_read_leb(pbc);
952  avio_skip(pbc, rendering_config_extension_size);
953 
955  NULL,
956  &submix_element->element_mix_config);
957  if (ret < 0)
958  goto fail;
959  submix_element->default_mix_gain = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
960  }
961 
963  if (ret < 0)
964  goto fail;
965  sub_mix->default_mix_gain = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
966 
967  nb_layouts = ffio_read_leb(pbc);
968  for (int j = 0; j < nb_layouts; j++) {
969  AVIAMFSubmixLayout *submix_layout;
970  int info_type;
971  int byte = avio_r8(pbc);
972 
973  submix_layout = av_iamf_submix_add_layout(sub_mix);
974  if (!submix_layout) {
975  ret = AVERROR(ENOMEM);
976  goto fail;
977  }
978 
979  submix_layout->layout_type = byte >> 6;
982  av_log(s, AV_LOG_ERROR, "Invalid Layout type %u in a submix from Mix Presentation %u\n",
983  submix_layout->layout_type, mix_presentation_id);
985  goto fail;
986  }
987  if (submix_layout->layout_type == 2) {
988  int sound_system;
989  sound_system = (byte >> 2) & 0xF;
990  if (sound_system >= FF_ARRAY_ELEMS(ff_iamf_sound_system_map)) {
992  goto fail;
993  }
994  av_channel_layout_copy(&submix_layout->sound_system, &ff_iamf_sound_system_map[sound_system].layout);
995  } else
997 
998  info_type = avio_r8(pbc);
999  submix_layout->integrated_loudness = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
1000  submix_layout->digital_peak = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
1001 
1002  if (info_type & 1)
1003  submix_layout->true_peak = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
1004  if (info_type & 2) {
1005  unsigned int num_anchored_loudness = avio_r8(pbc);
1006 
1007  for (int k = 0; k < num_anchored_loudness; k++) {
1008  unsigned int anchor_element = avio_r8(pbc);
1009  AVRational anchored_loudness = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
1010  if (anchor_element == IAMF_ANCHOR_ELEMENT_DIALOGUE)
1011  submix_layout->dialogue_anchored_loudness = anchored_loudness;
1012  else if (anchor_element <= IAMF_ANCHOR_ELEMENT_ALBUM)
1013  submix_layout->album_anchored_loudness = anchored_loudness;
1014  else
1015  av_log(s, AV_LOG_DEBUG, "Unknown anchor_element. Ignoring\n");
1016  }
1017  }
1018 
1019  if (info_type & 0xFC) {
1020  unsigned int info_type_size = ffio_read_leb(pbc);
1021  avio_skip(pbc, info_type_size);
1022  }
1023  }
1024  }
1025 
1026  c->mix_presentations[c->nb_mix_presentations++] = mix_presentation;
1027 
1028  len -= avio_tell(pbc);
1029  if (len)
1030  av_log(s, AV_LOG_WARNING, "Underread in mix_presentation_obu. %d bytes left at the end\n", len);
1031 
1032  ret = 0;
1033 fail:
1034  av_free(buf);
1035  if (ret < 0)
1036  ff_iamf_free_mix_presentation(&mix_presentation);
1037  return ret;
1038 }
1039 
1040 int ff_iamf_parse_obu_header(const uint8_t *buf, int buf_size,
1041  unsigned *obu_size, int *start_pos, enum IAMF_OBU_Type *type,
1042  unsigned *skip_samples, unsigned *discard_padding)
1043 {
1044  GetBitContext gb;
1045  int ret, extension_flag, trimming, start;
1046  unsigned skip = 0, discard = 0;
1047  unsigned size;
1048 
1049  ret = init_get_bits8(&gb, buf, FFMIN(buf_size, MAX_IAMF_OBU_HEADER_SIZE));
1050  if (ret < 0)
1051  return ret;
1052 
1053  *type = get_bits(&gb, 5);
1054  /*redundant =*/ get_bits1(&gb);
1055  trimming = get_bits1(&gb);
1056  extension_flag = get_bits1(&gb);
1057 
1058  *obu_size = get_leb(&gb);
1059  if (*obu_size > INT_MAX)
1060  return AVERROR_INVALIDDATA;
1061 
1062  start = get_bits_count(&gb) / 8;
1063 
1064  if (trimming) {
1065  discard = get_leb(&gb); // num_samples_to_trim_at_end
1066  skip = get_leb(&gb); // num_samples_to_trim_at_start
1067  }
1068 
1069  if (skip_samples)
1070  *skip_samples = skip;
1071  if (discard_padding)
1072  *discard_padding = discard;
1073 
1074  if (extension_flag) {
1075  unsigned int extension_bytes;
1076  extension_bytes = get_leb(&gb);
1077  if (extension_bytes > INT_MAX / 8)
1078  return AVERROR_INVALIDDATA;
1079  skip_bits_long(&gb, extension_bytes * 8);
1080  }
1081 
1082  if (get_bits_left(&gb) < 0)
1083  return AVERROR_INVALIDDATA;
1084 
1085  size = *obu_size + start;
1086  if (size > INT_MAX)
1087  return AVERROR_INVALIDDATA;
1088 
1089  *obu_size -= get_bits_count(&gb) / 8 - start;
1090  *start_pos = size - *obu_size;
1091 
1092  return size;
1093 }
1094 
1096  int max_size, void *log_ctx)
1097 {
1099  int ret;
1100 
1101  while (1) {
1102  unsigned obu_size;
1103  enum IAMF_OBU_Type type;
1104  int start_pos, len, size;
1105 
1106  if ((ret = ffio_ensure_seekback(pb, FFMIN(MAX_IAMF_OBU_HEADER_SIZE, max_size))) < 0)
1107  return ret;
1108  size = avio_read(pb, header, FFMIN(MAX_IAMF_OBU_HEADER_SIZE, max_size));
1109  if (size < 0)
1110  return size;
1112 
1113  len = ff_iamf_parse_obu_header(header, size, &obu_size, &start_pos, &type, NULL, NULL);
1114  if (len < 0 || obu_size > max_size) {
1115  av_log(log_ctx, AV_LOG_ERROR, "Failed to read obu header\n");
1116  avio_seek(pb, -size, SEEK_CUR);
1117  return len;
1118  }
1119 
1121  avio_seek(pb, -size, SEEK_CUR);
1122  break;
1123  }
1124 
1125  avio_seek(pb, -(size - start_pos), SEEK_CUR);
1126  switch (type) {
1128  ret = codec_config_obu(log_ctx, c, pb, obu_size);
1129  break;
1131  ret = audio_element_obu(log_ctx, c, pb, obu_size);
1132  break;
1134  ret = mix_presentation_obu(log_ctx, c, pb, obu_size);
1135  break;
1136  default: {
1137  int64_t offset = avio_skip(pb, obu_size);
1138  if (offset < 0)
1139  ret = offset;
1140  break;
1141  }
1142  }
1143  if (ret < 0) {
1144  av_log(log_ctx, AV_LOG_ERROR, "Failed to read obu type %d\n", type);
1145  return ret;
1146  }
1147  max_size -= obu_size + start_pos;
1148  if (max_size < 0)
1149  return AVERROR_INVALIDDATA;
1150  if (!max_size)
1151  break;
1152  }
1153 
1154  return 0;
1155 }
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:336
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:261
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:392
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:716
get_bits_left
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:678
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
put_bits32
static void av_unused put_bits32(PutBitContext *s, uint32_t value)
Write exactly 32 bits into a bitstream.
Definition: put_bits.h:291
AVCodecParameters::codec_type
enum AVMediaType codec_type
General type of the encoded data.
Definition: codec_par.h:51
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:89
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
get_bits_long
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
Definition: get_bits.h:404
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:249
AVChannelLayout::map
AVChannelCustom * map
This member must be used when the channel order is AV_CHANNEL_ORDER_CUSTOM.
Definition: channel_layout.h:370
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:223
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
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:461
AVChannelLayout::order
enum AVChannelOrder order
Channel order used in this layout.
Definition: channel_layout.h:324
iamf_parse.h
ambisonics_config
static int ambisonics_config(void *s, AVIOContext *pb, IAMFAudioElement *audio_element, const IAMFCodecConfig *codec_config)
Definition: iamf_parse.c:415
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
av_malloc
#define av_malloc(s)
Definition: tableprint_vlc.h:31
IAMFMixPresentation::cmix
const AVIAMFMixPresentation * cmix
Definition: iamf.h:108
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:364
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:318
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:393
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:337
IAMFParamDefinition
Definition: iamf.h:121
fail
#define fail()
Definition: checkasm.h:196
GetBitContext
Definition: get_bits.h:108
AVIAMFSubmixLayout
Submix layout as defined in section 3.7.6 of IAMF.
Definition: iamf.h:514
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:125
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
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:500
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:761
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:528
IAMFCodecConfig::sample_rate
int sample_rate
Definition: iamf.h:72
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_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:40
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:927
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:231
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:334
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:1095
codec_id
enum AVCodecID codec_id
Definition: vaapi_decode.c:410
PutBitContext
Definition: put_bits.h:50
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
tmp
static uint8_t tmp[20]
Definition: aes_ctr.c:47
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:371
audio_element_obu
static int audio_element_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
Definition: iamf_parse.c:622
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
AVCodecParameters::ch_layout
AVChannelLayout ch_layout
Audio only.
Definition: codec_par.h:180
IAMFAudioElement::celement
const AVIAMFAudioElement * celement
Definition: iamf.h:90
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
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:1040
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
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
AV_CODEC_ID_AAC
@ AV_CODEC_ID_AAC
Definition: codec_id.h:451
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:348
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:136
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:866
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
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_CODEC_ID_OPUS
@ AV_CODEC_ID_OPUS
Definition: codec_id.h:509
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:603
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.
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:1023
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
AVIAMFMixGain
Mix Gain Parameter Data as defined in section 3.8.1 of IAMF.
Definition: iamf.h:77
leb.h
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:56
get_bits64
static uint64_t get_bits64(GetBitContext *s, int n)
Read 0-64 bits.
Definition: get_bits.h:436
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 paremeter 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:622
AV_CODEC_ID_PCM_S32BE
@ AV_CODEC_ID_PCM_S32BE
Definition: codec_id.h:345
av_mallocz
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:256
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
AVIAMFParamDefinition::nb_subblocks
unsigned int nb_subblocks
Number of subblocks in the array.
Definition: iamf.h:208
AVChannelLayout::u
union AVChannelLayout::@459 u
Details about which channels are present in this layout.
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:1911
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:231
ff_iamf_free_audio_element
void ff_iamf_free_audio_element(IAMFAudioElement **paudio_element)
Definition: iamf.c:143
avio_rb16
unsigned int avio_rb16(AVIOContext *s)
Definition: aviobuf.c:746
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:806
IAMFMixPresentation
Definition: iamf.h:107
AV_CHANNEL_ORDER_CUSTOM
@ AV_CHANNEL_ORDER_CUSTOM
The channel order does not correspond to any other predefined order and is stored as an explicit map.
Definition: channel_layout.h:132
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
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:612
sign_extend
static av_const int sign_extend(int val, unsigned bits)
Definition: mathops.h:131
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:318
mix_presentation_obu
static int mix_presentation_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
Definition: iamf_parse.c:823
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:344
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:143
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
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 paremeter 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
ffio_read_size
int ffio_read_size(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:662
AVIAMFAudioElement::recon_gain_info
AVIAMFParamDefinition * recon_gain_info
Recon gain information used to reconstruct a scalable channel audio representation.
Definition: iamf.h:383
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:343
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:349
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:375
AV_WL16A
#define AV_WL16A(p, v)
Definition: intreadwrite.h:557
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