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[5];
289 
290  init_put_bits(&pb, buf, sizeof(buf));
291  ret = init_get_bits8(&gb, codecpar->extradata, codecpar->extradata_size);
292  if (ret < 0)
293  return ret;
294 
295  ret = get_bits(&gb, 5);
296  put_bits(&pb, 5, ret);
297  if (ret == AOT_ESCAPE) // violates section 3.11.2, but better check for it
298  put_bits(&pb, 6, get_bits(&gb, 6));
299  ret = get_bits(&gb, 4);
300  put_bits(&pb, 4, ret);
301  if (ret == 0x0f)
302  put_bits(&pb, 24, get_bits(&gb, 24));
303 
304  skip_bits(&gb, 4);
305  put_bits(&pb, 4, codecpar->ch_layout.nb_channels); // set channel config
306  ret = put_bits_left(&pb);
307  put_bits(&pb, ret, get_bits_long(&gb, ret));
308  flush_put_bits(&pb);
309 
310  memcpy(codecpar->extradata, buf, put_bytes_output(&pb));
311  break;
312  }
313  case AV_CODEC_ID_FLAC: {
314  uint8_t buf[13];
315 
316  init_put_bits(&pb, buf, sizeof(buf));
317  ret = init_get_bits8(&gb, codecpar->extradata, codecpar->extradata_size);
318  if (ret < 0)
319  return ret;
320 
321  put_bits32(&pb, get_bits_long(&gb, 32)); // min/max blocksize
322  put_bits64(&pb, 48, get_bits64(&gb, 48)); // min/max framesize
323  put_bits(&pb, 20, get_bits(&gb, 20)); // samplerate
324  skip_bits(&gb, 3);
325  put_bits(&pb, 3, codecpar->ch_layout.nb_channels - 1);
326  ret = put_bits_left(&pb);
327  put_bits(&pb, ret, get_bits(&gb, ret));
328  flush_put_bits(&pb);
329 
330  memcpy(codecpar->extradata, buf, sizeof(buf));
331  break;
332  }
333  }
334 
335  return 0;
336 }
337 
339  IAMFAudioElement *audio_element,
340  const IAMFCodecConfig *codec_config)
341 {
342  int nb_layers, k = 0;
343 
344  nb_layers = avio_r8(pb) >> 5; // get_bits(&gb, 3);
345  // skip_bits(&gb, 5); //reserved
346 
347  if (nb_layers > 6 || nb_layers == 0)
348  return AVERROR_INVALIDDATA;
349 
350  audio_element->layers = av_calloc(nb_layers, sizeof(*audio_element->layers));
351  if (!audio_element->layers)
352  return AVERROR(ENOMEM);
353 
354  audio_element->nb_layers = nb_layers;
355  for (int i = 0; i < nb_layers; i++) {
356  AVIAMFLayer *layer;
357  int loudspeaker_layout, output_gain_is_present_flag;
358  int substream_count, coupled_substream_count;
359  int expanded_loudspeaker_layout = -1;
360  int ret, byte = avio_r8(pb);
361 
362  layer = av_iamf_audio_element_add_layer(audio_element->element);
363  if (!layer)
364  return AVERROR(ENOMEM);
365 
366  loudspeaker_layout = byte >> 4; // get_bits(&gb, 4);
367  output_gain_is_present_flag = (byte >> 3) & 1; //get_bits1(&gb);
368  if ((byte >> 2) & 1)
369  layer->flags |= AV_IAMF_LAYER_FLAG_RECON_GAIN;
370  substream_count = avio_r8(pb);
371  coupled_substream_count = avio_r8(pb);
372 
373  if (substream_count + k > audio_element->nb_substreams)
374  return AVERROR_INVALIDDATA;
375 
376  audio_element->layers[i].substream_count = substream_count;
377  audio_element->layers[i].coupled_substream_count = coupled_substream_count;
378  if (output_gain_is_present_flag) {
379  layer->output_gain_flags = avio_r8(pb) >> 2; // get_bits(&gb, 6);
380  layer->output_gain = av_make_q(sign_extend(avio_rb16(pb), 16), 1 << 8);
381  }
382 
383  if (!i && loudspeaker_layout == 15)
384  expanded_loudspeaker_layout = avio_r8(pb);
385  if (expanded_loudspeaker_layout > 0 && expanded_loudspeaker_layout < 13)
386  av_channel_layout_copy(&layer->ch_layout, &ff_iamf_expanded_scalable_ch_layouts[expanded_loudspeaker_layout]);
387  else if (loudspeaker_layout < 10)
388  av_channel_layout_copy(&layer->ch_layout, &ff_iamf_scalable_ch_layouts[loudspeaker_layout]);
389  else
391  .nb_channels = substream_count +
392  coupled_substream_count };
393 
394  for (int j = 0; j < substream_count; j++) {
395  IAMFSubStream *substream = &audio_element->substreams[k++];
396 
397  substream->codecpar->ch_layout = coupled_substream_count-- > 0 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO :
399 
400  ret = update_extradata(substream->codecpar);
401  if (ret < 0)
402  return ret;
403  }
404 
405  }
406 
407  return 0;
408 }
409 
410 static int ambisonics_config(void *s, AVIOContext *pb,
411  IAMFAudioElement *audio_element,
412  const IAMFCodecConfig *codec_config)
413 {
414  AVIAMFLayer *layer;
415  unsigned ambisonics_mode;
416  int output_channel_count, substream_count, order;
417  int ret;
418 
419  ambisonics_mode = ffio_read_leb(pb);
420  if (ambisonics_mode > 1)
421  return AVERROR_INVALIDDATA;
422 
423  output_channel_count = avio_r8(pb); // C
424  substream_count = avio_r8(pb); // N
425  if (audio_element->nb_substreams != substream_count || output_channel_count == 0)
426  return AVERROR_INVALIDDATA;
427 
428  order = floor(sqrt(output_channel_count - 1));
429  /* incomplete order - some harmonics are missing */
430  if ((order + 1) * (order + 1) != output_channel_count)
431  return AVERROR_INVALIDDATA;
432 
433  audio_element->layers = av_mallocz(sizeof(*audio_element->layers));
434  if (!audio_element->layers)
435  return AVERROR(ENOMEM);
436 
437  audio_element->nb_layers = 1;
438  audio_element->layers->substream_count = substream_count;
439 
440  layer = av_iamf_audio_element_add_layer(audio_element->element);
441  if (!layer)
442  return AVERROR(ENOMEM);
443 
444  layer->ambisonics_mode = ambisonics_mode;
445  if (ambisonics_mode == 0) {
446  for (int i = 0; i < substream_count; i++) {
447  IAMFSubStream *substream = &audio_element->substreams[i];
448 
450 
451  ret = update_extradata(substream->codecpar);
452  if (ret < 0)
453  return ret;
454  }
455 
457  layer->ch_layout.nb_channels = output_channel_count;
458  layer->ch_layout.u.map = av_calloc(output_channel_count, sizeof(*layer->ch_layout.u.map));
459  if (!layer->ch_layout.u.map)
460  return AVERROR(ENOMEM);
461 
462  for (int i = 0; i < output_channel_count; i++)
463  layer->ch_layout.u.map[i].id = avio_r8(pb) + AV_CHAN_AMBISONIC_BASE;
464  } else {
465  int coupled_substream_count = avio_r8(pb); // M
466  int nb_demixing_matrix = substream_count + coupled_substream_count;
467  int demixing_matrix_size = nb_demixing_matrix * output_channel_count;
468 
469  audio_element->layers->coupled_substream_count = coupled_substream_count;
470 
471  layer->ch_layout = (AVChannelLayout){ .order = AV_CHANNEL_ORDER_AMBISONIC, .nb_channels = output_channel_count };
472  layer->demixing_matrix = av_malloc_array(demixing_matrix_size, sizeof(*layer->demixing_matrix));
473  if (!layer->demixing_matrix)
474  return AVERROR(ENOMEM);
475 
476  for (int i = 0; i < demixing_matrix_size; i++)
477  layer->demixing_matrix[i] = av_make_q(sign_extend(avio_rb16(pb), 16), 1 << 8);
478 
479  for (int i = 0; i < substream_count; i++) {
480  IAMFSubStream *substream = &audio_element->substreams[i];
481 
482  substream->codecpar->ch_layout = coupled_substream_count-- > 0 ? (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO :
484 
485 
486  ret = update_extradata(substream->codecpar);
487  if (ret < 0)
488  return ret;
489  }
490  }
491 
492  return 0;
493 }
494 
495 static int param_parse(void *s, IAMFContext *c, AVIOContext *pb,
496  unsigned int type,
497  const IAMFAudioElement *audio_element,
498  AVIAMFParamDefinition **out_param_definition)
499 {
501  AVIAMFParamDefinition *param;
502  unsigned int parameter_id, parameter_rate, mode;
503  unsigned int duration = 0, constant_subblock_duration = 0, nb_subblocks = 0;
504  unsigned int total_duration = 0;
505  size_t param_size;
506 
507  parameter_id = ffio_read_leb(pb);
508 
509  for (int i = 0; i < c->nb_param_definitions; i++)
510  if (c->param_definitions[i]->param->parameter_id == parameter_id) {
511  param_definition = c->param_definitions[i];
512  break;
513  }
514 
515  parameter_rate = ffio_read_leb(pb);
516  mode = avio_r8(pb) >> 7;
517 
518  if (mode == 0) {
519  duration = ffio_read_leb(pb);
520  if (!duration)
521  return AVERROR_INVALIDDATA;
522  constant_subblock_duration = ffio_read_leb(pb);
523  if (constant_subblock_duration == 0)
524  nb_subblocks = ffio_read_leb(pb);
525  else {
526  nb_subblocks = duration / constant_subblock_duration;
527  total_duration = duration;
528  }
529  }
530 
531  param = av_iamf_param_definition_alloc(type, nb_subblocks, &param_size);
532  if (!param)
533  return AVERROR(ENOMEM);
534 
535  for (int i = 0; i < nb_subblocks; i++) {
536  void *subblock = av_iamf_param_definition_get_subblock(param, i);
537  unsigned int subblock_duration = constant_subblock_duration;
538 
539  if (constant_subblock_duration == 0) {
540  subblock_duration = ffio_read_leb(pb);
541  total_duration += subblock_duration;
542  } else if (i == nb_subblocks - 1)
543  subblock_duration = duration - i * constant_subblock_duration;
544 
545  switch (type) {
547  AVIAMFMixGain *mix = subblock;
548  mix->subblock_duration = subblock_duration;
549  break;
550  }
552  AVIAMFDemixingInfo *demix = subblock;
553  demix->subblock_duration = subblock_duration;
554  // DefaultDemixingInfoParameterData
555  av_assert0(audio_element);
556  demix->dmixp_mode = avio_r8(pb) >> 5;
557  audio_element->element->default_w = avio_r8(pb) >> 4;
558  break;
559  }
561  AVIAMFReconGain *recon = subblock;
562  recon->subblock_duration = subblock_duration;
563  break;
564  }
565  default:
566  av_free(param);
567  return AVERROR_INVALIDDATA;
568  }
569  }
570 
571  if (!mode && !constant_subblock_duration && total_duration != duration) {
572  av_log(s, AV_LOG_ERROR, "Invalid subblock durations in parameter_id %u\n", parameter_id);
573  av_free(param);
574  return AVERROR_INVALIDDATA;
575  }
576 
577  param->parameter_id = parameter_id;
578  param->parameter_rate = parameter_rate;
579  param->duration = duration;
580  param->constant_subblock_duration = constant_subblock_duration;
581  param->nb_subblocks = nb_subblocks;
582 
583  if (param_definition) {
584  if (param_definition->param_size != param_size || memcmp(param_definition->param, param, param_size)) {
585  av_log(s, AV_LOG_ERROR, "Incosistent parameters for parameter_id %u\n", parameter_id);
586  av_free(param);
587  return AVERROR_INVALIDDATA;
588  }
589  } else {
590  IAMFParamDefinition **tmp = av_realloc_array(c->param_definitions, c->nb_param_definitions + 1,
591  sizeof(*c->param_definitions));
592  if (!tmp) {
593  av_free(param);
594  return AVERROR(ENOMEM);
595  }
596  c->param_definitions = tmp;
597 
599  if (!param_definition) {
600  av_free(param);
601  return AVERROR(ENOMEM);
602  }
603  param_definition->param = param;
604  param_definition->mode = !mode;
605  param_definition->param_size = param_size;
606  param_definition->audio_element = audio_element;
607 
608  c->param_definitions[c->nb_param_definitions++] = param_definition;
609  }
610 
611  av_assert0(out_param_definition);
612  *out_param_definition = param;
613 
614  return 0;
615 }
616 
617 static int audio_element_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
618 {
619  const IAMFCodecConfig *codec_config;
620  AVIAMFAudioElement *element;
621  IAMFAudioElement **tmp, *audio_element = NULL;
622  FFIOContext b;
623  AVIOContext *pbc;
624  uint8_t *buf;
625  unsigned audio_element_id, nb_substreams, codec_config_id, num_parameters;
626  int audio_element_type, ret;
627 
628  buf = av_malloc(len);
629  if (!buf)
630  return AVERROR(ENOMEM);
631 
632  ret = ffio_read_size(pb, buf, len);
633  if (ret < 0)
634  goto fail;
635 
636  ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
637  pbc = &b.pub;
638 
639  audio_element_id = ffio_read_leb(pbc);
640 
641  for (int i = 0; i < c->nb_audio_elements; i++)
642  if (c->audio_elements[i]->audio_element_id == audio_element_id) {
643  av_log(s, AV_LOG_ERROR, "Duplicate audio_element_id %d\n", audio_element_id);
645  goto fail;
646  }
647 
648  audio_element_type = avio_r8(pbc) >> 5;
649  if (audio_element_type > AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE) {
650  av_log(s, AV_LOG_DEBUG, "Unknown audio_element_type referenced in an audio element. Ignoring\n");
651  ret = 0;
652  goto fail;
653  }
654 
655  codec_config_id = ffio_read_leb(pbc);
656 
657  codec_config = ff_iamf_get_codec_config(c, codec_config_id);
658  if (!codec_config) {
659  av_log(s, AV_LOG_ERROR, "Non existant codec config id %d referenced in an audio element\n", codec_config_id);
661  goto fail;
662  }
663 
664  if (codec_config->codec_id == AV_CODEC_ID_NONE) {
665  av_log(s, AV_LOG_DEBUG, "Unknown codec id referenced in an audio element. Ignoring\n");
666  ret = 0;
667  goto fail;
668  }
669 
670  tmp = av_realloc_array(c->audio_elements, c->nb_audio_elements + 1, sizeof(*c->audio_elements));
671  if (!tmp) {
672  ret = AVERROR(ENOMEM);
673  goto fail;
674  }
675  c->audio_elements = tmp;
676 
677  audio_element = av_mallocz(sizeof(*audio_element));
678  if (!audio_element) {
679  ret = AVERROR(ENOMEM);
680  goto fail;
681  }
682 
683  nb_substreams = ffio_read_leb(pbc);
684  audio_element->codec_config_id = codec_config_id;
685  audio_element->audio_element_id = audio_element_id;
686  audio_element->substreams = av_calloc(nb_substreams, sizeof(*audio_element->substreams));
687  if (!audio_element->substreams) {
688  ret = AVERROR(ENOMEM);
689  goto fail;
690  }
691  audio_element->nb_substreams = nb_substreams;
692 
693  element = audio_element->element = av_iamf_audio_element_alloc();
694  if (!element) {
695  ret = AVERROR(ENOMEM);
696  goto fail;
697  }
698  audio_element->celement = element;
699 
700  element->audio_element_type = audio_element_type;
701 
702  for (int i = 0; i < audio_element->nb_substreams; i++) {
703  IAMFSubStream *substream = &audio_element->substreams[i];
704 
705  substream->codecpar = avcodec_parameters_alloc();
706  if (!substream->codecpar) {
707  ret = AVERROR(ENOMEM);
708  goto fail;
709  }
710 
711  substream->audio_substream_id = ffio_read_leb(pbc);
712 
713  substream->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
714  substream->codecpar->codec_id = codec_config->codec_id;
715  substream->codecpar->frame_size = codec_config->nb_samples;
716  substream->codecpar->sample_rate = codec_config->sample_rate;
717  substream->codecpar->seek_preroll = -codec_config->audio_roll_distance * codec_config->nb_samples;
718 
719  switch(substream->codecpar->codec_id) {
720  case AV_CODEC_ID_AAC:
721  case AV_CODEC_ID_FLAC:
722  case AV_CODEC_ID_OPUS:
724  if (!substream->codecpar->extradata) {
725  ret = AVERROR(ENOMEM);
726  goto fail;
727  }
728  memcpy(substream->codecpar->extradata, codec_config->extradata, codec_config->extradata_size);
729  memset(substream->codecpar->extradata + codec_config->extradata_size, 0, AV_INPUT_BUFFER_PADDING_SIZE);
730  substream->codecpar->extradata_size = codec_config->extradata_size;
731  break;
732  }
733  }
734 
735  num_parameters = ffio_read_leb(pbc);
736  if (num_parameters && audio_element_type != 0) {
737  av_log(s, AV_LOG_ERROR, "Audio Element parameter count %u is invalid"
738  " for Scene representations\n", num_parameters);
740  goto fail;
741  }
742 
743  for (int i = 0; i < num_parameters; i++) {
744  unsigned type;
745 
746  type = ffio_read_leb(pbc);
750  ret = param_parse(s, c, pbc, type, audio_element, &element->demixing_info);
752  ret = param_parse(s, c, pbc, type, audio_element, &element->recon_gain_info);
753  else {
754  unsigned param_definition_size = ffio_read_leb(pbc);
755  avio_skip(pbc, param_definition_size);
756  }
757  if (ret < 0)
758  goto fail;
759  }
760 
761  if (audio_element_type == AV_IAMF_AUDIO_ELEMENT_TYPE_CHANNEL) {
762  ret = scalable_channel_layout_config(s, pbc, audio_element, codec_config);
763  if (ret < 0)
764  goto fail;
765  } else if (audio_element_type == AV_IAMF_AUDIO_ELEMENT_TYPE_SCENE) {
766  ret = ambisonics_config(s, pbc, audio_element, codec_config);
767  if (ret < 0)
768  goto fail;
769  } else {
770  av_assert0(0);
771  }
772 
773  c->audio_elements[c->nb_audio_elements++] = audio_element;
774 
775  len -= avio_tell(pbc);
776  if (len)
777  av_log(s, AV_LOG_WARNING, "Underread in audio_element_obu. %d bytes left at the end\n", len);
778 
779  ret = 0;
780 fail:
781  av_free(buf);
782  if (ret < 0)
783  ff_iamf_free_audio_element(&audio_element);
784  return ret;
785 }
786 
787 static int label_string(AVIOContext *pb, char **label)
788 {
789  uint8_t buf[128];
790 
791  avio_get_str(pb, sizeof(buf), buf, sizeof(buf));
792 
793  if (pb->error)
794  return pb->error;
795  if (pb->eof_reached)
796  return AVERROR_INVALIDDATA;
797  *label = av_strdup(buf);
798  if (!*label)
799  return AVERROR(ENOMEM);
800 
801  return 0;
802 }
803 
804 static int mix_presentation_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
805 {
807  IAMFMixPresentation **tmp, *mix_presentation = NULL;
808  FFIOContext b;
809  AVIOContext *pbc;
810  uint8_t *buf;
811  unsigned nb_submixes, mix_presentation_id;
812  int ret;
813 
814  buf = av_malloc(len);
815  if (!buf)
816  return AVERROR(ENOMEM);
817 
818  ret = ffio_read_size(pb, buf, len);
819  if (ret < 0)
820  goto fail;
821 
822  ffio_init_context(&b, buf, len, 0, NULL, NULL, NULL, NULL);
823  pbc = &b.pub;
824 
825  mix_presentation_id = ffio_read_leb(pbc);
826 
827  for (int i = 0; i < c->nb_mix_presentations; i++)
828  if (c->mix_presentations[i]->mix_presentation_id == mix_presentation_id) {
829  av_log(s, AV_LOG_ERROR, "Duplicate mix_presentation_id %d\n", mix_presentation_id);
831  goto fail;
832  }
833 
834  tmp = av_realloc_array(c->mix_presentations, c->nb_mix_presentations + 1, sizeof(*c->mix_presentations));
835  if (!tmp) {
836  ret = AVERROR(ENOMEM);
837  goto fail;
838  }
839  c->mix_presentations = tmp;
840 
841  mix_presentation = av_mallocz(sizeof(*mix_presentation));
842  if (!mix_presentation) {
843  ret = AVERROR(ENOMEM);
844  goto fail;
845  }
846 
847  mix_presentation->mix_presentation_id = mix_presentation_id;
848  mix = mix_presentation->mix = av_iamf_mix_presentation_alloc();
849  if (!mix) {
850  ret = AVERROR(ENOMEM);
851  goto fail;
852  }
853  mix_presentation->cmix = mix;
854 
855  mix_presentation->count_label = ffio_read_leb(pbc);
856  mix_presentation->language_label = av_calloc(mix_presentation->count_label,
857  sizeof(*mix_presentation->language_label));
858  if (!mix_presentation->language_label) {
859  mix_presentation->count_label = 0;
860  ret = AVERROR(ENOMEM);
861  goto fail;
862  }
863 
864  for (int i = 0; i < mix_presentation->count_label; i++) {
865  ret = label_string(pbc, &mix_presentation->language_label[i]);
866  if (ret < 0)
867  goto fail;
868  }
869 
870  for (int i = 0; i < mix_presentation->count_label; i++) {
871  char *annotation = NULL;
872  ret = label_string(pbc, &annotation);
873  if (ret < 0)
874  goto fail;
875  ret = av_dict_set(&mix->annotations, mix_presentation->language_label[i], annotation,
877  if (ret < 0)
878  goto fail;
879  }
880 
881  nb_submixes = ffio_read_leb(pbc);
882  for (int i = 0; i < nb_submixes; i++) {
883  AVIAMFSubmix *sub_mix;
884  unsigned nb_elements, nb_layouts;
885 
887  if (!sub_mix) {
888  ret = AVERROR(ENOMEM);
889  goto fail;
890  }
891 
892  nb_elements = ffio_read_leb(pbc);
893  for (int j = 0; j < nb_elements; j++) {
894  AVIAMFSubmixElement *submix_element;
895  IAMFAudioElement *audio_element = NULL;
896  unsigned int rendering_config_extension_size;
897 
898  submix_element = av_iamf_submix_add_element(sub_mix);
899  if (!submix_element) {
900  ret = AVERROR(ENOMEM);
901  goto fail;
902  }
903 
904  submix_element->audio_element_id = ffio_read_leb(pbc);
905 
906  for (int k = 0; k < c->nb_audio_elements; k++)
907  if (c->audio_elements[k]->audio_element_id == submix_element->audio_element_id) {
908  audio_element = c->audio_elements[k];
909  break;
910  }
911 
912  if (!audio_element) {
913  av_log(s, AV_LOG_ERROR, "Invalid Audio Element with id %u referenced by Mix Parameters %u\n",
914  submix_element->audio_element_id, mix_presentation_id);
916  goto fail;
917  }
918 
919  for (int k = 0; k < mix_presentation->count_label; k++) {
920  char *annotation = NULL;
921  ret = label_string(pbc, &annotation);
922  if (ret < 0)
923  goto fail;
924  ret = av_dict_set(&submix_element->annotations, mix_presentation->language_label[k], annotation,
926  if (ret < 0)
927  goto fail;
928  }
929 
930  submix_element->headphones_rendering_mode = avio_r8(pbc) >> 6;
931 
932  rendering_config_extension_size = ffio_read_leb(pbc);
933  avio_skip(pbc, rendering_config_extension_size);
934 
936  NULL,
937  &submix_element->element_mix_config);
938  if (ret < 0)
939  goto fail;
940  submix_element->default_mix_gain = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
941  }
942 
944  if (ret < 0)
945  goto fail;
946  sub_mix->default_mix_gain = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
947 
948  nb_layouts = ffio_read_leb(pbc);
949  for (int j = 0; j < nb_layouts; j++) {
950  AVIAMFSubmixLayout *submix_layout;
951  int info_type;
952  int byte = avio_r8(pbc);
953 
954  submix_layout = av_iamf_submix_add_layout(sub_mix);
955  if (!submix_layout) {
956  ret = AVERROR(ENOMEM);
957  goto fail;
958  }
959 
960  submix_layout->layout_type = byte >> 6;
963  av_log(s, AV_LOG_ERROR, "Invalid Layout type %u in a submix from Mix Presentation %u\n",
964  submix_layout->layout_type, mix_presentation_id);
966  goto fail;
967  }
968  if (submix_layout->layout_type == 2) {
969  int sound_system;
970  sound_system = (byte >> 2) & 0xF;
971  if (sound_system >= FF_ARRAY_ELEMS(ff_iamf_sound_system_map)) {
973  goto fail;
974  }
975  av_channel_layout_copy(&submix_layout->sound_system, &ff_iamf_sound_system_map[sound_system].layout);
976  } else
978 
979  info_type = avio_r8(pbc);
980  submix_layout->integrated_loudness = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
981  submix_layout->digital_peak = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
982 
983  if (info_type & 1)
984  submix_layout->true_peak = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
985  if (info_type & 2) {
986  unsigned int num_anchored_loudness = avio_r8(pbc);
987 
988  for (int k = 0; k < num_anchored_loudness; k++) {
989  unsigned int anchor_element = avio_r8(pbc);
990  AVRational anchored_loudness = av_make_q(sign_extend(avio_rb16(pbc), 16), 1 << 8);
991  if (anchor_element == IAMF_ANCHOR_ELEMENT_DIALOGUE)
992  submix_layout->dialogue_anchored_loudness = anchored_loudness;
993  else if (anchor_element <= IAMF_ANCHOR_ELEMENT_ALBUM)
994  submix_layout->album_anchored_loudness = anchored_loudness;
995  else
996  av_log(s, AV_LOG_DEBUG, "Unknown anchor_element. Ignoring\n");
997  }
998  }
999 
1000  if (info_type & 0xFC) {
1001  unsigned int info_type_size = ffio_read_leb(pbc);
1002  avio_skip(pbc, info_type_size);
1003  }
1004  }
1005  }
1006 
1007  c->mix_presentations[c->nb_mix_presentations++] = mix_presentation;
1008 
1009  len -= avio_tell(pbc);
1010  if (len)
1011  av_log(s, AV_LOG_WARNING, "Underread in mix_presentation_obu. %d bytes left at the end\n", len);
1012 
1013  ret = 0;
1014 fail:
1015  av_free(buf);
1016  if (ret < 0)
1017  ff_iamf_free_mix_presentation(&mix_presentation);
1018  return ret;
1019 }
1020 
1021 int ff_iamf_parse_obu_header(const uint8_t *buf, int buf_size,
1022  unsigned *obu_size, int *start_pos, enum IAMF_OBU_Type *type,
1023  unsigned *skip_samples, unsigned *discard_padding)
1024 {
1025  GetBitContext gb;
1026  int ret, extension_flag, trimming, start;
1027  unsigned skip = 0, discard = 0;
1028  unsigned size;
1029 
1030  ret = init_get_bits8(&gb, buf, FFMIN(buf_size, MAX_IAMF_OBU_HEADER_SIZE));
1031  if (ret < 0)
1032  return ret;
1033 
1034  *type = get_bits(&gb, 5);
1035  /*redundant =*/ get_bits1(&gb);
1036  trimming = get_bits1(&gb);
1037  extension_flag = get_bits1(&gb);
1038 
1039  *obu_size = get_leb(&gb);
1040  if (*obu_size > INT_MAX)
1041  return AVERROR_INVALIDDATA;
1042 
1043  start = get_bits_count(&gb) / 8;
1044 
1045  if (trimming) {
1046  discard = get_leb(&gb); // num_samples_to_trim_at_end
1047  skip = get_leb(&gb); // num_samples_to_trim_at_start
1048  }
1049 
1050  if (skip_samples)
1051  *skip_samples = skip;
1052  if (discard_padding)
1053  *discard_padding = discard;
1054 
1055  if (extension_flag) {
1056  unsigned int extension_bytes;
1057  extension_bytes = get_leb(&gb);
1058  if (extension_bytes > INT_MAX / 8)
1059  return AVERROR_INVALIDDATA;
1060  skip_bits_long(&gb, extension_bytes * 8);
1061  }
1062 
1063  if (get_bits_left(&gb) < 0)
1064  return AVERROR_INVALIDDATA;
1065 
1066  size = *obu_size + start;
1067  if (size > INT_MAX)
1068  return AVERROR_INVALIDDATA;
1069 
1070  *obu_size -= get_bits_count(&gb) / 8 - start;
1071  *start_pos = size - *obu_size;
1072 
1073  return size;
1074 }
1075 
1077  int max_size, void *log_ctx)
1078 {
1080  int ret;
1081 
1082  while (1) {
1083  unsigned obu_size;
1084  enum IAMF_OBU_Type type;
1085  int start_pos, len, size;
1086 
1087  if ((ret = ffio_ensure_seekback(pb, FFMIN(MAX_IAMF_OBU_HEADER_SIZE, max_size))) < 0)
1088  return ret;
1089  size = avio_read(pb, header, FFMIN(MAX_IAMF_OBU_HEADER_SIZE, max_size));
1090  if (size < 0)
1091  return size;
1093 
1094  len = ff_iamf_parse_obu_header(header, size, &obu_size, &start_pos, &type, NULL, NULL);
1095  if (len < 0 || obu_size > max_size) {
1096  av_log(log_ctx, AV_LOG_ERROR, "Failed to read obu header\n");
1097  avio_seek(pb, -size, SEEK_CUR);
1098  return len;
1099  }
1100 
1102  avio_seek(pb, -size, SEEK_CUR);
1103  break;
1104  }
1105 
1106  avio_seek(pb, -(size - start_pos), SEEK_CUR);
1107  switch (type) {
1109  ret = codec_config_obu(log_ctx, c, pb, obu_size);
1110  break;
1112  ret = audio_element_obu(log_ctx, c, pb, obu_size);
1113  break;
1115  ret = mix_presentation_obu(log_ctx, c, pb, obu_size);
1116  break;
1117  default: {
1118  int64_t offset = avio_skip(pb, obu_size);
1119  if (offset < 0)
1120  ret = offset;
1121  break;
1122  }
1123  }
1124  if (ret < 0) {
1125  av_log(log_ctx, AV_LOG_ERROR, "Failed to read obu type %d\n", type);
1126  return ret;
1127  }
1128  max_size -= obu_size + start_pos;
1129  if (max_size < 0)
1130  return AVERROR_INVALIDDATA;
1131  if (!max_size)
1132  break;
1133  }
1134 
1135  return 0;
1136 }
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:335
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:278
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:215
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:391
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:695
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:421
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:266
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:52
put_bits
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:223
tmp
static uint8_t tmp[11]
Definition: aes_ctr.c:28
put_bits64
static void put_bits64(PutBitContext *s, int n, uint64_t value)
Write up to 64 bits into a bitstream.
Definition: put_bits.h:334
AVCodecParameters::seek_preroll
int seek_preroll
Audio only.
Definition: codec_par.h:214
b
#define b
Definition: input.c:41
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:460
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:410
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:329
AVChannelLayout::u
union AVChannelLayout::@432 u
Details about which channels are present in this layout.
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:30
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:381
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:335
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:392
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:336
IAMFParamDefinition
Definition: iamf.h:121
fail
#define fail()
Definition: checkasm.h:193
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:495
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:209
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:545
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:202
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:230
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:1076
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
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:388
audio_element_obu
static int audio_element_obu(void *s, IAMFContext *c, AVIOContext *pb, int len)
Definition: iamf_parse.c:617
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:1021
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:450
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:347
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:133
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:508
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:453
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:31
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:344
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
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:1879
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:787
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:804
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:343
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:33
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:88
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:338
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:348
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