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aacdec_template.c
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1 /*
2  * AAC decoder
3  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
4  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
5  * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
6  *
7  * AAC LATM decoder
8  * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
9  * Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
10  *
11  * AAC decoder fixed-point implementation
12  * Copyright (c) 2013
13  * MIPS Technologies, Inc., California.
14  *
15  * This file is part of FFmpeg.
16  *
17  * FFmpeg is free software; you can redistribute it and/or
18  * modify it under the terms of the GNU Lesser General Public
19  * License as published by the Free Software Foundation; either
20  * version 2.1 of the License, or (at your option) any later version.
21  *
22  * FFmpeg is distributed in the hope that it will be useful,
23  * but WITHOUT ANY WARRANTY; without even the implied warranty of
24  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
25  * Lesser General Public License for more details.
26  *
27  * You should have received a copy of the GNU Lesser General Public
28  * License along with FFmpeg; if not, write to the Free Software
29  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
30  */
31 
32 /**
33  * @file
34  * AAC decoder
35  * @author Oded Shimon ( ods15 ods15 dyndns org )
36  * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
37  *
38  * AAC decoder fixed-point implementation
39  * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
40  * @author Nedeljko Babic ( nedeljko.babic imgtec com )
41  */
42 
43 /*
44  * supported tools
45  *
46  * Support? Name
47  * N (code in SoC repo) gain control
48  * Y block switching
49  * Y window shapes - standard
50  * N window shapes - Low Delay
51  * Y filterbank - standard
52  * N (code in SoC repo) filterbank - Scalable Sample Rate
53  * Y Temporal Noise Shaping
54  * Y Long Term Prediction
55  * Y intensity stereo
56  * Y channel coupling
57  * Y frequency domain prediction
58  * Y Perceptual Noise Substitution
59  * Y Mid/Side stereo
60  * N Scalable Inverse AAC Quantization
61  * N Frequency Selective Switch
62  * N upsampling filter
63  * Y quantization & coding - AAC
64  * N quantization & coding - TwinVQ
65  * N quantization & coding - BSAC
66  * N AAC Error Resilience tools
67  * N Error Resilience payload syntax
68  * N Error Protection tool
69  * N CELP
70  * N Silence Compression
71  * N HVXC
72  * N HVXC 4kbits/s VR
73  * N Structured Audio tools
74  * N Structured Audio Sample Bank Format
75  * N MIDI
76  * N Harmonic and Individual Lines plus Noise
77  * N Text-To-Speech Interface
78  * Y Spectral Band Replication
79  * Y (not in this code) Layer-1
80  * Y (not in this code) Layer-2
81  * Y (not in this code) Layer-3
82  * N SinuSoidal Coding (Transient, Sinusoid, Noise)
83  * Y Parametric Stereo
84  * N Direct Stream Transfer
85  * Y (not in fixed point code) Enhanced AAC Low Delay (ER AAC ELD)
86  *
87  * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
88  * - HE AAC v2 comprises LC AAC with Spectral Band Replication and
89  Parametric Stereo.
90  */
91 
92 #include "libavutil/thread.h"
93 
95 static VLC vlc_spectral[11];
96 
97 static int output_configure(AACContext *ac,
98  uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
99  enum OCStatus oc_type, int get_new_frame);
100 
101 #define overread_err "Input buffer exhausted before END element found\n"
102 
103 static int count_channels(uint8_t (*layout)[3], int tags)
104 {
105  int i, sum = 0;
106  for (i = 0; i < tags; i++) {
107  int syn_ele = layout[i][0];
108  int pos = layout[i][2];
109  sum += (1 + (syn_ele == TYPE_CPE)) *
110  (pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);
111  }
112  return sum;
113 }
114 
115 /**
116  * Check for the channel element in the current channel position configuration.
117  * If it exists, make sure the appropriate element is allocated and map the
118  * channel order to match the internal FFmpeg channel layout.
119  *
120  * @param che_pos current channel position configuration
121  * @param type channel element type
122  * @param id channel element id
123  * @param channels count of the number of channels in the configuration
124  *
125  * @return Returns error status. 0 - OK, !0 - error
126  */
128  enum ChannelPosition che_pos,
129  int type, int id, int *channels)
130 {
131  if (*channels >= MAX_CHANNELS)
132  return AVERROR_INVALIDDATA;
133  if (che_pos) {
134  if (!ac->che[type][id]) {
135  if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
136  return AVERROR(ENOMEM);
138  }
139  if (type != TYPE_CCE) {
140  if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
141  av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
142  return AVERROR_INVALIDDATA;
143  }
144  ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
145  if (type == TYPE_CPE ||
146  (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
147  ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
148  }
149  }
150  } else {
151  if (ac->che[type][id])
153  av_freep(&ac->che[type][id]);
154  }
155  return 0;
156 }
157 
159 {
160  AACContext *ac = avctx->priv_data;
161  int type, id, ch, ret;
162 
163  /* set channel pointers to internal buffers by default */
164  for (type = 0; type < 4; type++) {
165  for (id = 0; id < MAX_ELEM_ID; id++) {
166  ChannelElement *che = ac->che[type][id];
167  if (che) {
168  che->ch[0].ret = che->ch[0].ret_buf;
169  che->ch[1].ret = che->ch[1].ret_buf;
170  }
171  }
172  }
173 
174  /* get output buffer */
175  av_frame_unref(ac->frame);
176  if (!avctx->channels)
177  return 1;
178 
179  ac->frame->nb_samples = 2048;
180  if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
181  return ret;
182 
183  /* map output channel pointers to AVFrame data */
184  for (ch = 0; ch < avctx->channels; ch++) {
185  if (ac->output_element[ch])
186  ac->output_element[ch]->ret = (INTFLOAT *)ac->frame->extended_data[ch];
187  }
188 
189  return 0;
190 }
191 
193  uint64_t av_position;
197 };
198 
199 static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
200  uint8_t (*layout_map)[3], int offset, uint64_t left,
201  uint64_t right, int pos)
202 {
203  if (layout_map[offset][0] == TYPE_CPE) {
204  e2c_vec[offset] = (struct elem_to_channel) {
205  .av_position = left | right,
206  .syn_ele = TYPE_CPE,
207  .elem_id = layout_map[offset][1],
208  .aac_position = pos
209  };
210  return 1;
211  } else {
212  e2c_vec[offset] = (struct elem_to_channel) {
213  .av_position = left,
214  .syn_ele = TYPE_SCE,
215  .elem_id = layout_map[offset][1],
216  .aac_position = pos
217  };
218  e2c_vec[offset + 1] = (struct elem_to_channel) {
219  .av_position = right,
220  .syn_ele = TYPE_SCE,
221  .elem_id = layout_map[offset + 1][1],
222  .aac_position = pos
223  };
224  return 2;
225  }
226 }
227 
228 static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
229  int *current)
230 {
231  int num_pos_channels = 0;
232  int first_cpe = 0;
233  int sce_parity = 0;
234  int i;
235  for (i = *current; i < tags; i++) {
236  if (layout_map[i][2] != pos)
237  break;
238  if (layout_map[i][0] == TYPE_CPE) {
239  if (sce_parity) {
240  if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
241  sce_parity = 0;
242  } else {
243  return -1;
244  }
245  }
246  num_pos_channels += 2;
247  first_cpe = 1;
248  } else {
249  num_pos_channels++;
250  sce_parity ^= 1;
251  }
252  }
253  if (sce_parity &&
254  ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
255  return -1;
256  *current = i;
257  return num_pos_channels;
258 }
259 
260 static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
261 {
262  int i, n, total_non_cc_elements;
263  struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
264  int num_front_channels, num_side_channels, num_back_channels;
265  uint64_t layout;
266 
267  if (FF_ARRAY_ELEMS(e2c_vec) < tags)
268  return 0;
269 
270  i = 0;
271  num_front_channels =
272  count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
273  if (num_front_channels < 0)
274  return 0;
275  num_side_channels =
276  count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
277  if (num_side_channels < 0)
278  return 0;
279  num_back_channels =
280  count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
281  if (num_back_channels < 0)
282  return 0;
283 
284  if (num_side_channels == 0 && num_back_channels >= 4) {
285  num_side_channels = 2;
286  num_back_channels -= 2;
287  }
288 
289  i = 0;
290  if (num_front_channels & 1) {
291  e2c_vec[i] = (struct elem_to_channel) {
293  .syn_ele = TYPE_SCE,
294  .elem_id = layout_map[i][1],
295  .aac_position = AAC_CHANNEL_FRONT
296  };
297  i++;
298  num_front_channels--;
299  }
300  if (num_front_channels >= 4) {
301  i += assign_pair(e2c_vec, layout_map, i,
305  num_front_channels -= 2;
306  }
307  if (num_front_channels >= 2) {
308  i += assign_pair(e2c_vec, layout_map, i,
312  num_front_channels -= 2;
313  }
314  while (num_front_channels >= 2) {
315  i += assign_pair(e2c_vec, layout_map, i,
316  UINT64_MAX,
317  UINT64_MAX,
319  num_front_channels -= 2;
320  }
321 
322  if (num_side_channels >= 2) {
323  i += assign_pair(e2c_vec, layout_map, i,
327  num_side_channels -= 2;
328  }
329  while (num_side_channels >= 2) {
330  i += assign_pair(e2c_vec, layout_map, i,
331  UINT64_MAX,
332  UINT64_MAX,
334  num_side_channels -= 2;
335  }
336 
337  while (num_back_channels >= 4) {
338  i += assign_pair(e2c_vec, layout_map, i,
339  UINT64_MAX,
340  UINT64_MAX,
342  num_back_channels -= 2;
343  }
344  if (num_back_channels >= 2) {
345  i += assign_pair(e2c_vec, layout_map, i,
349  num_back_channels -= 2;
350  }
351  if (num_back_channels) {
352  e2c_vec[i] = (struct elem_to_channel) {
354  .syn_ele = TYPE_SCE,
355  .elem_id = layout_map[i][1],
356  .aac_position = AAC_CHANNEL_BACK
357  };
358  i++;
359  num_back_channels--;
360  }
361 
362  if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
363  e2c_vec[i] = (struct elem_to_channel) {
365  .syn_ele = TYPE_LFE,
366  .elem_id = layout_map[i][1],
367  .aac_position = AAC_CHANNEL_LFE
368  };
369  i++;
370  }
371  while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
372  e2c_vec[i] = (struct elem_to_channel) {
373  .av_position = UINT64_MAX,
374  .syn_ele = TYPE_LFE,
375  .elem_id = layout_map[i][1],
376  .aac_position = AAC_CHANNEL_LFE
377  };
378  i++;
379  }
380 
381  // Must choose a stable sort
382  total_non_cc_elements = n = i;
383  do {
384  int next_n = 0;
385  for (i = 1; i < n; i++)
386  if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
387  FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
388  next_n = i;
389  }
390  n = next_n;
391  } while (n > 0);
392 
393  layout = 0;
394  for (i = 0; i < total_non_cc_elements; i++) {
395  layout_map[i][0] = e2c_vec[i].syn_ele;
396  layout_map[i][1] = e2c_vec[i].elem_id;
397  layout_map[i][2] = e2c_vec[i].aac_position;
398  if (e2c_vec[i].av_position != UINT64_MAX) {
399  layout |= e2c_vec[i].av_position;
400  }
401  }
402 
403  return layout;
404 }
405 
406 /**
407  * Save current output configuration if and only if it has been locked.
408  */
410  int pushed = 0;
411 
412  if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
413  ac->oc[0] = ac->oc[1];
414  pushed = 1;
415  }
416  ac->oc[1].status = OC_NONE;
417  return pushed;
418 }
419 
420 /**
421  * Restore the previous output configuration if and only if the current
422  * configuration is unlocked.
423  */
425  if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
426  ac->oc[1] = ac->oc[0];
427  ac->avctx->channels = ac->oc[1].channels;
428  ac->avctx->channel_layout = ac->oc[1].channel_layout;
429  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
430  ac->oc[1].status, 0);
431  }
432 }
433 
434 /**
435  * Configure output channel order based on the current program
436  * configuration element.
437  *
438  * @return Returns error status. 0 - OK, !0 - error
439  */
441  uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
442  enum OCStatus oc_type, int get_new_frame)
443 {
444  AVCodecContext *avctx = ac->avctx;
445  int i, channels = 0, ret;
446  uint64_t layout = 0;
447  uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
448  uint8_t type_counts[TYPE_END] = { 0 };
449 
450  if (ac->oc[1].layout_map != layout_map) {
451  memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
452  ac->oc[1].layout_map_tags = tags;
453  }
454  for (i = 0; i < tags; i++) {
455  int type = layout_map[i][0];
456  int id = layout_map[i][1];
457  id_map[type][id] = type_counts[type]++;
458  if (id_map[type][id] >= MAX_ELEM_ID) {
459  avpriv_request_sample(ac->avctx, "Too large remapped id");
460  return AVERROR_PATCHWELCOME;
461  }
462  }
463  // Try to sniff a reasonable channel order, otherwise output the
464  // channels in the order the PCE declared them.
466  layout = sniff_channel_order(layout_map, tags);
467  for (i = 0; i < tags; i++) {
468  int type = layout_map[i][0];
469  int id = layout_map[i][1];
470  int iid = id_map[type][id];
471  int position = layout_map[i][2];
472  // Allocate or free elements depending on if they are in the
473  // current program configuration.
474  ret = che_configure(ac, position, type, iid, &channels);
475  if (ret < 0)
476  return ret;
477  ac->tag_che_map[type][id] = ac->che[type][iid];
478  }
479  if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
480  if (layout == AV_CH_FRONT_CENTER) {
482  } else {
483  layout = 0;
484  }
485  }
486 
487  if (layout) avctx->channel_layout = layout;
488  ac->oc[1].channel_layout = layout;
489  avctx->channels = ac->oc[1].channels = channels;
490  ac->oc[1].status = oc_type;
491 
492  if (get_new_frame) {
493  if ((ret = frame_configure_elements(ac->avctx)) < 0)
494  return ret;
495  }
496 
497  return 0;
498 }
499 
500 static void flush(AVCodecContext *avctx)
501 {
502  AACContext *ac= avctx->priv_data;
503  int type, i, j;
504 
505  for (type = 3; type >= 0; type--) {
506  for (i = 0; i < MAX_ELEM_ID; i++) {
507  ChannelElement *che = ac->che[type][i];
508  if (che) {
509  for (j = 0; j <= 1; j++) {
510  memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
511  }
512  }
513  }
514  }
515 }
516 
517 /**
518  * Set up channel positions based on a default channel configuration
519  * as specified in table 1.17.
520  *
521  * @return Returns error status. 0 - OK, !0 - error
522  */
524  uint8_t (*layout_map)[3],
525  int *tags,
526  int channel_config)
527 {
528  if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
529  channel_config > 12) {
530  av_log(avctx, AV_LOG_ERROR,
531  "invalid default channel configuration (%d)\n",
532  channel_config);
533  return AVERROR_INVALIDDATA;
534  }
535  *tags = tags_per_config[channel_config];
536  memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
537  *tags * sizeof(*layout_map));
538 
539  /*
540  * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
541  * However, at least Nero AAC encoder encodes 7.1 streams using the default
542  * channel config 7, mapping the side channels of the original audio stream
543  * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
544  * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
545  * the incorrect streams as if they were correct (and as the encoder intended).
546  *
547  * As actual intended 7.1(wide) streams are very rare, default to assuming a
548  * 7.1 layout was intended.
549  */
550  if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
551  av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
552  " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
553  " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
554  layout_map[2][2] = AAC_CHANNEL_SIDE;
555  }
556 
557  return 0;
558 }
559 
560 static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
561 {
562  /* For PCE based channel configurations map the channels solely based
563  * on tags. */
564  if (!ac->oc[1].m4ac.chan_config) {
565  return ac->tag_che_map[type][elem_id];
566  }
567  // Allow single CPE stereo files to be signalled with mono configuration.
568  if (!ac->tags_mapped && type == TYPE_CPE &&
569  ac->oc[1].m4ac.chan_config == 1) {
570  uint8_t layout_map[MAX_ELEM_ID*4][3];
571  int layout_map_tags;
573 
574  av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
575 
576  if (set_default_channel_config(ac->avctx, layout_map,
577  &layout_map_tags, 2) < 0)
578  return NULL;
579  if (output_configure(ac, layout_map, layout_map_tags,
580  OC_TRIAL_FRAME, 1) < 0)
581  return NULL;
582 
583  ac->oc[1].m4ac.chan_config = 2;
584  ac->oc[1].m4ac.ps = 0;
585  }
586  // And vice-versa
587  if (!ac->tags_mapped && type == TYPE_SCE &&
588  ac->oc[1].m4ac.chan_config == 2) {
589  uint8_t layout_map[MAX_ELEM_ID * 4][3];
590  int layout_map_tags;
592 
593  av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
594 
595  if (set_default_channel_config(ac->avctx, layout_map,
596  &layout_map_tags, 1) < 0)
597  return NULL;
598  if (output_configure(ac, layout_map, layout_map_tags,
599  OC_TRIAL_FRAME, 1) < 0)
600  return NULL;
601 
602  ac->oc[1].m4ac.chan_config = 1;
603  if (ac->oc[1].m4ac.sbr)
604  ac->oc[1].m4ac.ps = -1;
605  }
606  /* For indexed channel configurations map the channels solely based
607  * on position. */
608  switch (ac->oc[1].m4ac.chan_config) {
609  case 12:
610  case 7:
611  if (ac->tags_mapped == 3 && type == TYPE_CPE) {
612  ac->tags_mapped++;
613  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
614  }
615  case 11:
616  if (ac->tags_mapped == 2 &&
617  ac->oc[1].m4ac.chan_config == 11 &&
618  type == TYPE_SCE) {
619  ac->tags_mapped++;
620  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
621  }
622  case 6:
623  /* Some streams incorrectly code 5.1 audio as
624  * SCE[0] CPE[0] CPE[1] SCE[1]
625  * instead of
626  * SCE[0] CPE[0] CPE[1] LFE[0].
627  * If we seem to have encountered such a stream, transfer
628  * the LFE[0] element to the SCE[1]'s mapping */
629  if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
630  if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
632  "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
633  type == TYPE_SCE ? "SCE" : "LFE", elem_id);
634  ac->warned_remapping_once++;
635  }
636  ac->tags_mapped++;
637  return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
638  }
639  case 5:
640  if (ac->tags_mapped == 2 && type == TYPE_CPE) {
641  ac->tags_mapped++;
642  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
643  }
644  case 4:
645  /* Some streams incorrectly code 4.0 audio as
646  * SCE[0] CPE[0] LFE[0]
647  * instead of
648  * SCE[0] CPE[0] SCE[1].
649  * If we seem to have encountered such a stream, transfer
650  * the SCE[1] element to the LFE[0]'s mapping */
651  if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
652  if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
654  "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
655  type == TYPE_SCE ? "SCE" : "LFE", elem_id);
656  ac->warned_remapping_once++;
657  }
658  ac->tags_mapped++;
659  return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
660  }
661  if (ac->tags_mapped == 2 &&
662  ac->oc[1].m4ac.chan_config == 4 &&
663  type == TYPE_SCE) {
664  ac->tags_mapped++;
665  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
666  }
667  case 3:
668  case 2:
669  if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
670  type == TYPE_CPE) {
671  ac->tags_mapped++;
672  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
673  } else if (ac->oc[1].m4ac.chan_config == 2) {
674  return NULL;
675  }
676  case 1:
677  if (!ac->tags_mapped && type == TYPE_SCE) {
678  ac->tags_mapped++;
679  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
680  }
681  default:
682  return NULL;
683  }
684 }
685 
686 /**
687  * Decode an array of 4 bit element IDs, optionally interleaved with a
688  * stereo/mono switching bit.
689  *
690  * @param type speaker type/position for these channels
691  */
692 static void decode_channel_map(uint8_t layout_map[][3],
693  enum ChannelPosition type,
694  GetBitContext *gb, int n)
695 {
696  while (n--) {
697  enum RawDataBlockType syn_ele;
698  switch (type) {
699  case AAC_CHANNEL_FRONT:
700  case AAC_CHANNEL_BACK:
701  case AAC_CHANNEL_SIDE:
702  syn_ele = get_bits1(gb);
703  break;
704  case AAC_CHANNEL_CC:
705  skip_bits1(gb);
706  syn_ele = TYPE_CCE;
707  break;
708  case AAC_CHANNEL_LFE:
709  syn_ele = TYPE_LFE;
710  break;
711  default:
712  // AAC_CHANNEL_OFF has no channel map
713  av_assert0(0);
714  }
715  layout_map[0][0] = syn_ele;
716  layout_map[0][1] = get_bits(gb, 4);
717  layout_map[0][2] = type;
718  layout_map++;
719  }
720 }
721 
722 static inline void relative_align_get_bits(GetBitContext *gb,
723  int reference_position) {
724  int n = (reference_position - get_bits_count(gb) & 7);
725  if (n)
726  skip_bits(gb, n);
727 }
728 
729 /**
730  * Decode program configuration element; reference: table 4.2.
731  *
732  * @return Returns error status. 0 - OK, !0 - error
733  */
734 static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
735  uint8_t (*layout_map)[3],
736  GetBitContext *gb, int byte_align_ref)
737 {
738  int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
739  int sampling_index;
740  int comment_len;
741  int tags;
742 
743  skip_bits(gb, 2); // object_type
744 
745  sampling_index = get_bits(gb, 4);
746  if (m4ac->sampling_index != sampling_index)
747  av_log(avctx, AV_LOG_WARNING,
748  "Sample rate index in program config element does not "
749  "match the sample rate index configured by the container.\n");
750 
751  num_front = get_bits(gb, 4);
752  num_side = get_bits(gb, 4);
753  num_back = get_bits(gb, 4);
754  num_lfe = get_bits(gb, 2);
755  num_assoc_data = get_bits(gb, 3);
756  num_cc = get_bits(gb, 4);
757 
758  if (get_bits1(gb))
759  skip_bits(gb, 4); // mono_mixdown_tag
760  if (get_bits1(gb))
761  skip_bits(gb, 4); // stereo_mixdown_tag
762 
763  if (get_bits1(gb))
764  skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
765 
766  if (get_bits_left(gb) < 5 * (num_front + num_side + num_back + num_cc) + 4 *(num_lfe + num_assoc_data + num_cc)) {
767  av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
768  return -1;
769  }
770  decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front);
771  tags = num_front;
772  decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side);
773  tags += num_side;
774  decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back);
775  tags += num_back;
776  decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe);
777  tags += num_lfe;
778 
779  skip_bits_long(gb, 4 * num_assoc_data);
780 
781  decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc);
782  tags += num_cc;
783 
784  relative_align_get_bits(gb, byte_align_ref);
785 
786  /* comment field, first byte is length */
787  comment_len = get_bits(gb, 8) * 8;
788  if (get_bits_left(gb) < comment_len) {
789  av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
790  return AVERROR_INVALIDDATA;
791  }
792  skip_bits_long(gb, comment_len);
793  return tags;
794 }
795 
796 /**
797  * Decode GA "General Audio" specific configuration; reference: table 4.1.
798  *
799  * @param ac pointer to AACContext, may be null
800  * @param avctx pointer to AVCCodecContext, used for logging
801  *
802  * @return Returns error status. 0 - OK, !0 - error
803  */
805  GetBitContext *gb,
806  int get_bit_alignment,
807  MPEG4AudioConfig *m4ac,
808  int channel_config)
809 {
810  int extension_flag, ret, ep_config, res_flags;
811  uint8_t layout_map[MAX_ELEM_ID*4][3];
812  int tags = 0;
813 
814 #if USE_FIXED
815  if (get_bits1(gb)) { // frameLengthFlag
816  avpriv_report_missing_feature(avctx, "Fixed point 960/120 MDCT window");
817  return AVERROR_PATCHWELCOME;
818  }
819  m4ac->frame_length_short = 0;
820 #else
821  m4ac->frame_length_short = get_bits1(gb);
822  if (m4ac->frame_length_short && m4ac->sbr == 1) {
823  avpriv_report_missing_feature(avctx, "SBR with 960 frame length");
824  if (ac) ac->warned_960_sbr = 1;
825  m4ac->sbr = 0;
826  m4ac->ps = 0;
827  }
828 #endif
829 
830  if (get_bits1(gb)) // dependsOnCoreCoder
831  skip_bits(gb, 14); // coreCoderDelay
832  extension_flag = get_bits1(gb);
833 
834  if (m4ac->object_type == AOT_AAC_SCALABLE ||
836  skip_bits(gb, 3); // layerNr
837 
838  if (channel_config == 0) {
839  skip_bits(gb, 4); // element_instance_tag
840  tags = decode_pce(avctx, m4ac, layout_map, gb, get_bit_alignment);
841  if (tags < 0)
842  return tags;
843  } else {
844  if ((ret = set_default_channel_config(avctx, layout_map,
845  &tags, channel_config)))
846  return ret;
847  }
848 
849  if (count_channels(layout_map, tags) > 1) {
850  m4ac->ps = 0;
851  } else if (m4ac->sbr == 1 && m4ac->ps == -1)
852  m4ac->ps = 1;
853 
854  if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
855  return ret;
856 
857  if (extension_flag) {
858  switch (m4ac->object_type) {
859  case AOT_ER_BSAC:
860  skip_bits(gb, 5); // numOfSubFrame
861  skip_bits(gb, 11); // layer_length
862  break;
863  case AOT_ER_AAC_LC:
864  case AOT_ER_AAC_LTP:
865  case AOT_ER_AAC_SCALABLE:
866  case AOT_ER_AAC_LD:
867  res_flags = get_bits(gb, 3);
868  if (res_flags) {
870  "AAC data resilience (flags %x)",
871  res_flags);
872  return AVERROR_PATCHWELCOME;
873  }
874  break;
875  }
876  skip_bits1(gb); // extensionFlag3 (TBD in version 3)
877  }
878  switch (m4ac->object_type) {
879  case AOT_ER_AAC_LC:
880  case AOT_ER_AAC_LTP:
881  case AOT_ER_AAC_SCALABLE:
882  case AOT_ER_AAC_LD:
883  ep_config = get_bits(gb, 2);
884  if (ep_config) {
886  "epConfig %d", ep_config);
887  return AVERROR_PATCHWELCOME;
888  }
889  }
890  return 0;
891 }
892 
894  GetBitContext *gb,
895  MPEG4AudioConfig *m4ac,
896  int channel_config)
897 {
898  int ret, ep_config, res_flags;
899  uint8_t layout_map[MAX_ELEM_ID*4][3];
900  int tags = 0;
901  const int ELDEXT_TERM = 0;
902 
903  m4ac->ps = 0;
904  m4ac->sbr = 0;
905 #if USE_FIXED
906  if (get_bits1(gb)) { // frameLengthFlag
907  avpriv_request_sample(avctx, "960/120 MDCT window");
908  return AVERROR_PATCHWELCOME;
909  }
910 #else
911  m4ac->frame_length_short = get_bits1(gb);
912 #endif
913  res_flags = get_bits(gb, 3);
914  if (res_flags) {
916  "AAC data resilience (flags %x)",
917  res_flags);
918  return AVERROR_PATCHWELCOME;
919  }
920 
921  if (get_bits1(gb)) { // ldSbrPresentFlag
923  "Low Delay SBR");
924  return AVERROR_PATCHWELCOME;
925  }
926 
927  while (get_bits(gb, 4) != ELDEXT_TERM) {
928  int len = get_bits(gb, 4);
929  if (len == 15)
930  len += get_bits(gb, 8);
931  if (len == 15 + 255)
932  len += get_bits(gb, 16);
933  if (get_bits_left(gb) < len * 8 + 4) {
935  return AVERROR_INVALIDDATA;
936  }
937  skip_bits_long(gb, 8 * len);
938  }
939 
940  if ((ret = set_default_channel_config(avctx, layout_map,
941  &tags, channel_config)))
942  return ret;
943 
944  if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
945  return ret;
946 
947  ep_config = get_bits(gb, 2);
948  if (ep_config) {
950  "epConfig %d", ep_config);
951  return AVERROR_PATCHWELCOME;
952  }
953  return 0;
954 }
955 
956 /**
957  * Decode audio specific configuration; reference: table 1.13.
958  *
959  * @param ac pointer to AACContext, may be null
960  * @param avctx pointer to AVCCodecContext, used for logging
961  * @param m4ac pointer to MPEG4AudioConfig, used for parsing
962  * @param gb buffer holding an audio specific config
963  * @param get_bit_alignment relative alignment for byte align operations
964  * @param sync_extension look for an appended sync extension
965  *
966  * @return Returns error status or number of consumed bits. <0 - error
967  */
969  AVCodecContext *avctx,
970  MPEG4AudioConfig *m4ac,
971  GetBitContext *gb,
972  int get_bit_alignment,
973  int sync_extension)
974 {
975  int i, ret;
976  GetBitContext gbc = *gb;
977 
978  if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension)) < 0)
979  return AVERROR_INVALIDDATA;
980 
981  if (m4ac->sampling_index > 12) {
982  av_log(avctx, AV_LOG_ERROR,
983  "invalid sampling rate index %d\n",
984  m4ac->sampling_index);
985  return AVERROR_INVALIDDATA;
986  }
987  if (m4ac->object_type == AOT_ER_AAC_LD &&
988  (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
989  av_log(avctx, AV_LOG_ERROR,
990  "invalid low delay sampling rate index %d\n",
991  m4ac->sampling_index);
992  return AVERROR_INVALIDDATA;
993  }
994 
995  skip_bits_long(gb, i);
996 
997  switch (m4ac->object_type) {
998  case AOT_AAC_MAIN:
999  case AOT_AAC_LC:
1000  case AOT_AAC_SSR:
1001  case AOT_AAC_LTP:
1002  case AOT_ER_AAC_LC:
1003  case AOT_ER_AAC_LD:
1004  if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment,
1005  m4ac, m4ac->chan_config)) < 0)
1006  return ret;
1007  break;
1008  case AOT_ER_AAC_ELD:
1009  if ((ret = decode_eld_specific_config(ac, avctx, gb,
1010  m4ac, m4ac->chan_config)) < 0)
1011  return ret;
1012  break;
1013  default:
1015  "Audio object type %s%d",
1016  m4ac->sbr == 1 ? "SBR+" : "",
1017  m4ac->object_type);
1018  return AVERROR(ENOSYS);
1019  }
1020 
1021  ff_dlog(avctx,
1022  "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
1023  m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
1024  m4ac->sample_rate, m4ac->sbr,
1025  m4ac->ps);
1026 
1027  return get_bits_count(gb);
1028 }
1029 
1031  AVCodecContext *avctx,
1032  MPEG4AudioConfig *m4ac,
1033  const uint8_t *data, int64_t bit_size,
1034  int sync_extension)
1035 {
1036  int i, ret;
1037  GetBitContext gb;
1038 
1039  if (bit_size < 0 || bit_size > INT_MAX) {
1040  av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n");
1041  return AVERROR_INVALIDDATA;
1042  }
1043 
1044  ff_dlog(avctx, "audio specific config size %d\n", (int)bit_size >> 3);
1045  for (i = 0; i < bit_size >> 3; i++)
1046  ff_dlog(avctx, "%02x ", data[i]);
1047  ff_dlog(avctx, "\n");
1048 
1049  if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
1050  return ret;
1051 
1052  return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0,
1053  sync_extension);
1054 }
1055 
1056 /**
1057  * linear congruential pseudorandom number generator
1058  *
1059  * @param previous_val pointer to the current state of the generator
1060  *
1061  * @return Returns a 32-bit pseudorandom integer
1062  */
1063 static av_always_inline int lcg_random(unsigned previous_val)
1064 {
1065  union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
1066  return v.s;
1067 }
1068 
1070 {
1071  int i;
1072  for (i = 0; i < MAX_PREDICTORS; i++)
1073  reset_predict_state(&ps[i]);
1074 }
1075 
1076 static int sample_rate_idx (int rate)
1077 {
1078  if (92017 <= rate) return 0;
1079  else if (75132 <= rate) return 1;
1080  else if (55426 <= rate) return 2;
1081  else if (46009 <= rate) return 3;
1082  else if (37566 <= rate) return 4;
1083  else if (27713 <= rate) return 5;
1084  else if (23004 <= rate) return 6;
1085  else if (18783 <= rate) return 7;
1086  else if (13856 <= rate) return 8;
1087  else if (11502 <= rate) return 9;
1088  else if (9391 <= rate) return 10;
1089  else return 11;
1090 }
1091 
1092 static void reset_predictor_group(PredictorState *ps, int group_num)
1093 {
1094  int i;
1095  for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
1096  reset_predict_state(&ps[i]);
1097 }
1098 
1099 #define AAC_INIT_VLC_STATIC(num, size) \
1100  INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
1101  ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \
1102  sizeof(ff_aac_spectral_bits[num][0]), \
1103  ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
1104  sizeof(ff_aac_spectral_codes[num][0]), \
1105  size);
1106 
1107 static void aacdec_init(AACContext *ac);
1108 
1110 {
1111  AAC_INIT_VLC_STATIC( 0, 304);
1112  AAC_INIT_VLC_STATIC( 1, 270);
1113  AAC_INIT_VLC_STATIC( 2, 550);
1114  AAC_INIT_VLC_STATIC( 3, 300);
1115  AAC_INIT_VLC_STATIC( 4, 328);
1116  AAC_INIT_VLC_STATIC( 5, 294);
1117  AAC_INIT_VLC_STATIC( 6, 306);
1118  AAC_INIT_VLC_STATIC( 7, 268);
1119  AAC_INIT_VLC_STATIC( 8, 510);
1120  AAC_INIT_VLC_STATIC( 9, 366);
1121  AAC_INIT_VLC_STATIC(10, 462);
1122 
1124 
1125  ff_aac_tableinit();
1126 
1127  INIT_VLC_STATIC(&vlc_scalefactors, 7,
1130  sizeof(ff_aac_scalefactor_bits[0]),
1131  sizeof(ff_aac_scalefactor_bits[0]),
1133  sizeof(ff_aac_scalefactor_code[0]),
1134  sizeof(ff_aac_scalefactor_code[0]),
1135  352);
1136 
1137  // window initialization
1140 #if !USE_FIXED
1143  AAC_RENAME(ff_sine_window_init)(AAC_RENAME(ff_sine_960), 960);
1144  AAC_RENAME(ff_sine_window_init)(AAC_RENAME(ff_sine_120), 120);
1145 #endif
1149 
1151 }
1152 
1154 
1156 {
1157  AACContext *ac = avctx->priv_data;
1158  int ret;
1159 
1160  ret = ff_thread_once(&aac_table_init, &aac_static_table_init);
1161  if (ret != 0)
1162  return AVERROR_UNKNOWN;
1163 
1164  ac->avctx = avctx;
1165  ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
1166 
1167  aacdec_init(ac);
1168 #if USE_FIXED
1169  avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
1170 #else
1171  avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1172 #endif /* USE_FIXED */
1173 
1174  if (avctx->extradata_size > 0) {
1175  if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
1176  avctx->extradata,
1177  avctx->extradata_size * 8LL,
1178  1)) < 0)
1179  return ret;
1180  } else {
1181  int sr, i;
1182  uint8_t layout_map[MAX_ELEM_ID*4][3];
1183  int layout_map_tags;
1184 
1185  sr = sample_rate_idx(avctx->sample_rate);
1186  ac->oc[1].m4ac.sampling_index = sr;
1187  ac->oc[1].m4ac.channels = avctx->channels;
1188  ac->oc[1].m4ac.sbr = -1;
1189  ac->oc[1].m4ac.ps = -1;
1190 
1191  for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
1192  if (ff_mpeg4audio_channels[i] == avctx->channels)
1193  break;
1195  i = 0;
1196  }
1197  ac->oc[1].m4ac.chan_config = i;
1198 
1199  if (ac->oc[1].m4ac.chan_config) {
1200  int ret = set_default_channel_config(avctx, layout_map,
1201  &layout_map_tags, ac->oc[1].m4ac.chan_config);
1202  if (!ret)
1203  output_configure(ac, layout_map, layout_map_tags,
1204  OC_GLOBAL_HDR, 0);
1205  else if (avctx->err_recognition & AV_EF_EXPLODE)
1206  return AVERROR_INVALIDDATA;
1207  }
1208  }
1209 
1210  if (avctx->channels > MAX_CHANNELS) {
1211  av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
1212  return AVERROR_INVALIDDATA;
1213  }
1214 
1215 #if USE_FIXED
1217 #else
1219 #endif /* USE_FIXED */
1220  if (!ac->fdsp) {
1221  return AVERROR(ENOMEM);
1222  }
1223 
1224  ac->random_state = 0x1f2e3d4c;
1225 
1226  AAC_RENAME_32(ff_mdct_init)(&ac->mdct, 11, 1, 1.0 / RANGE15(1024.0));
1227  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld, 10, 1, 1.0 / RANGE15(512.0));
1228  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small, 8, 1, 1.0 / RANGE15(128.0));
1229  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp, 11, 0, RANGE15(-2.0));
1230 #if !USE_FIXED
1231  ret = ff_mdct15_init(&ac->mdct120, 1, 3, 1.0f/(16*1024*120*2));
1232  if (ret < 0)
1233  return ret;
1234  ret = ff_mdct15_init(&ac->mdct480, 1, 5, 1.0f/(16*1024*960));
1235  if (ret < 0)
1236  return ret;
1237  ret = ff_mdct15_init(&ac->mdct960, 1, 6, 1.0f/(16*1024*960*2));
1238  if (ret < 0)
1239  return ret;
1240 #endif
1241 
1242  return 0;
1243 }
1244 
1245 /**
1246  * Skip data_stream_element; reference: table 4.10.
1247  */
1249 {
1250  int byte_align = get_bits1(gb);
1251  int count = get_bits(gb, 8);
1252  if (count == 255)
1253  count += get_bits(gb, 8);
1254  if (byte_align)
1255  align_get_bits(gb);
1256 
1257  if (get_bits_left(gb) < 8 * count) {
1258  av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
1259  return AVERROR_INVALIDDATA;
1260  }
1261  skip_bits_long(gb, 8 * count);
1262  return 0;
1263 }
1264 
1266  GetBitContext *gb)
1267 {
1268  int sfb;
1269  if (get_bits1(gb)) {
1270  ics->predictor_reset_group = get_bits(gb, 5);
1271  if (ics->predictor_reset_group == 0 ||
1272  ics->predictor_reset_group > 30) {
1273  av_log(ac->avctx, AV_LOG_ERROR,
1274  "Invalid Predictor Reset Group.\n");
1275  return AVERROR_INVALIDDATA;
1276  }
1277  }
1278  for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
1279  ics->prediction_used[sfb] = get_bits1(gb);
1280  }
1281  return 0;
1282 }
1283 
1284 /**
1285  * Decode Long Term Prediction data; reference: table 4.xx.
1286  */
1288  GetBitContext *gb, uint8_t max_sfb)
1289 {
1290  int sfb;
1291 
1292  ltp->lag = get_bits(gb, 11);
1293  ltp->coef = ltp_coef[get_bits(gb, 3)];
1294  for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
1295  ltp->used[sfb] = get_bits1(gb);
1296 }
1297 
1298 /**
1299  * Decode Individual Channel Stream info; reference: table 4.6.
1300  */
1302  GetBitContext *gb)
1303 {
1304  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
1305  const int aot = m4ac->object_type;
1306  const int sampling_index = m4ac->sampling_index;
1307  int ret_fail = AVERROR_INVALIDDATA;
1308 
1309  if (aot != AOT_ER_AAC_ELD) {
1310  if (get_bits1(gb)) {
1311  av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
1313  return AVERROR_INVALIDDATA;
1314  }
1315  ics->window_sequence[1] = ics->window_sequence[0];
1316  ics->window_sequence[0] = get_bits(gb, 2);
1317  if (aot == AOT_ER_AAC_LD &&
1318  ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
1319  av_log(ac->avctx, AV_LOG_ERROR,
1320  "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
1321  "window sequence %d found.\n", ics->window_sequence[0]);
1323  return AVERROR_INVALIDDATA;
1324  }
1325  ics->use_kb_window[1] = ics->use_kb_window[0];
1326  ics->use_kb_window[0] = get_bits1(gb);
1327  }
1328  ics->num_window_groups = 1;
1329  ics->group_len[0] = 1;
1330  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1331  int i;
1332  ics->max_sfb = get_bits(gb, 4);
1333  for (i = 0; i < 7; i++) {
1334  if (get_bits1(gb)) {
1335  ics->group_len[ics->num_window_groups - 1]++;
1336  } else {
1337  ics->num_window_groups++;
1338  ics->group_len[ics->num_window_groups - 1] = 1;
1339  }
1340  }
1341  ics->num_windows = 8;
1342  if (m4ac->frame_length_short) {
1343  ics->swb_offset = ff_swb_offset_120[sampling_index];
1344  ics->num_swb = ff_aac_num_swb_120[sampling_index];
1345  } else {
1346  ics->swb_offset = ff_swb_offset_128[sampling_index];
1347  ics->num_swb = ff_aac_num_swb_128[sampling_index];
1348  }
1349  ics->tns_max_bands = ff_tns_max_bands_128[sampling_index];
1350  ics->predictor_present = 0;
1351  } else {
1352  ics->max_sfb = get_bits(gb, 6);
1353  ics->num_windows = 1;
1354  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
1355  if (m4ac->frame_length_short) {
1356  ics->swb_offset = ff_swb_offset_480[sampling_index];
1357  ics->num_swb = ff_aac_num_swb_480[sampling_index];
1358  ics->tns_max_bands = ff_tns_max_bands_480[sampling_index];
1359  } else {
1360  ics->swb_offset = ff_swb_offset_512[sampling_index];
1361  ics->num_swb = ff_aac_num_swb_512[sampling_index];
1362  ics->tns_max_bands = ff_tns_max_bands_512[sampling_index];
1363  }
1364  if (!ics->num_swb || !ics->swb_offset) {
1365  ret_fail = AVERROR_BUG;
1366  goto fail;
1367  }
1368  } else {
1369  if (m4ac->frame_length_short) {
1370  ics->num_swb = ff_aac_num_swb_960[sampling_index];
1371  ics->swb_offset = ff_swb_offset_960[sampling_index];
1372  } else {
1373  ics->num_swb = ff_aac_num_swb_1024[sampling_index];
1374  ics->swb_offset = ff_swb_offset_1024[sampling_index];
1375  }
1376  ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
1377  }
1378  if (aot != AOT_ER_AAC_ELD) {
1379  ics->predictor_present = get_bits1(gb);
1380  ics->predictor_reset_group = 0;
1381  }
1382  if (ics->predictor_present) {
1383  if (aot == AOT_AAC_MAIN) {
1384  if (decode_prediction(ac, ics, gb)) {
1385  goto fail;
1386  }
1387  } else if (aot == AOT_AAC_LC ||
1388  aot == AOT_ER_AAC_LC) {
1389  av_log(ac->avctx, AV_LOG_ERROR,
1390  "Prediction is not allowed in AAC-LC.\n");
1391  goto fail;
1392  } else {
1393  if (aot == AOT_ER_AAC_LD) {
1394  av_log(ac->avctx, AV_LOG_ERROR,
1395  "LTP in ER AAC LD not yet implemented.\n");
1396  ret_fail = AVERROR_PATCHWELCOME;
1397  goto fail;
1398  }
1399  if ((ics->ltp.present = get_bits(gb, 1)))
1400  decode_ltp(&ics->ltp, gb, ics->max_sfb);
1401  }
1402  }
1403  }
1404 
1405  if (ics->max_sfb > ics->num_swb) {
1406  av_log(ac->avctx, AV_LOG_ERROR,
1407  "Number of scalefactor bands in group (%d) "
1408  "exceeds limit (%d).\n",
1409  ics->max_sfb, ics->num_swb);
1410  goto fail;
1411  }
1412 
1413  return 0;
1414 fail:
1415  ics->max_sfb = 0;
1416  return ret_fail;
1417 }
1418 
1419 /**
1420  * Decode band types (section_data payload); reference: table 4.46.
1421  *
1422  * @param band_type array of the used band type
1423  * @param band_type_run_end array of the last scalefactor band of a band type run
1424  *
1425  * @return Returns error status. 0 - OK, !0 - error
1426  */
1427 static int decode_band_types(AACContext *ac, enum BandType band_type[120],
1428  int band_type_run_end[120], GetBitContext *gb,
1430 {
1431  int g, idx = 0;
1432  const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
1433  for (g = 0; g < ics->num_window_groups; g++) {
1434  int k = 0;
1435  while (k < ics->max_sfb) {
1436  uint8_t sect_end = k;
1437  int sect_len_incr;
1438  int sect_band_type = get_bits(gb, 4);
1439  if (sect_band_type == 12) {
1440  av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
1441  return AVERROR_INVALIDDATA;
1442  }
1443  do {
1444  sect_len_incr = get_bits(gb, bits);
1445  sect_end += sect_len_incr;
1446  if (get_bits_left(gb) < 0) {
1447  av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
1448  return AVERROR_INVALIDDATA;
1449  }
1450  if (sect_end > ics->max_sfb) {
1451  av_log(ac->avctx, AV_LOG_ERROR,
1452  "Number of bands (%d) exceeds limit (%d).\n",
1453  sect_end, ics->max_sfb);
1454  return AVERROR_INVALIDDATA;
1455  }
1456  } while (sect_len_incr == (1 << bits) - 1);
1457  for (; k < sect_end; k++) {
1458  band_type [idx] = sect_band_type;
1459  band_type_run_end[idx++] = sect_end;
1460  }
1461  }
1462  }
1463  return 0;
1464 }
1465 
1466 /**
1467  * Decode scalefactors; reference: table 4.47.
1468  *
1469  * @param global_gain first scalefactor value as scalefactors are differentially coded
1470  * @param band_type array of the used band type
1471  * @param band_type_run_end array of the last scalefactor band of a band type run
1472  * @param sf array of scalefactors or intensity stereo positions
1473  *
1474  * @return Returns error status. 0 - OK, !0 - error
1475  */
1477  unsigned int global_gain,
1479  enum BandType band_type[120],
1480  int band_type_run_end[120])
1481 {
1482  int g, i, idx = 0;
1483  int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 };
1484  int clipped_offset;
1485  int noise_flag = 1;
1486  for (g = 0; g < ics->num_window_groups; g++) {
1487  for (i = 0; i < ics->max_sfb;) {
1488  int run_end = band_type_run_end[idx];
1489  if (band_type[idx] == ZERO_BT) {
1490  for (; i < run_end; i++, idx++)
1491  sf[idx] = FIXR(0.);
1492  } else if ((band_type[idx] == INTENSITY_BT) ||
1493  (band_type[idx] == INTENSITY_BT2)) {
1494  for (; i < run_end; i++, idx++) {
1495  offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1496  clipped_offset = av_clip(offset[2], -155, 100);
1497  if (offset[2] != clipped_offset) {
1499  "If you heard an audible artifact, there may be a bug in the decoder. "
1500  "Clipped intensity stereo position (%d -> %d)",
1501  offset[2], clipped_offset);
1502  }
1503 #if USE_FIXED
1504  sf[idx] = 100 - clipped_offset;
1505 #else
1506  sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
1507 #endif /* USE_FIXED */
1508  }
1509  } else if (band_type[idx] == NOISE_BT) {
1510  for (; i < run_end; i++, idx++) {
1511  if (noise_flag-- > 0)
1512  offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
1513  else
1514  offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1515  clipped_offset = av_clip(offset[1], -100, 155);
1516  if (offset[1] != clipped_offset) {
1518  "If you heard an audible artifact, there may be a bug in the decoder. "
1519  "Clipped noise gain (%d -> %d)",
1520  offset[1], clipped_offset);
1521  }
1522 #if USE_FIXED
1523  sf[idx] = -(100 + clipped_offset);
1524 #else
1525  sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
1526 #endif /* USE_FIXED */
1527  }
1528  } else {
1529  for (; i < run_end; i++, idx++) {
1530  offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1531  if (offset[0] > 255U) {
1532  av_log(ac->avctx, AV_LOG_ERROR,
1533  "Scalefactor (%d) out of range.\n", offset[0]);
1534  return AVERROR_INVALIDDATA;
1535  }
1536 #if USE_FIXED
1537  sf[idx] = -offset[0];
1538 #else
1539  sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
1540 #endif /* USE_FIXED */
1541  }
1542  }
1543  }
1544  }
1545  return 0;
1546 }
1547 
1548 /**
1549  * Decode pulse data; reference: table 4.7.
1550  */
1551 static int decode_pulses(Pulse *pulse, GetBitContext *gb,
1552  const uint16_t *swb_offset, int num_swb)
1553 {
1554  int i, pulse_swb;
1555  pulse->num_pulse = get_bits(gb, 2) + 1;
1556  pulse_swb = get_bits(gb, 6);
1557  if (pulse_swb >= num_swb)
1558  return -1;
1559  pulse->pos[0] = swb_offset[pulse_swb];
1560  pulse->pos[0] += get_bits(gb, 5);
1561  if (pulse->pos[0] >= swb_offset[num_swb])
1562  return -1;
1563  pulse->amp[0] = get_bits(gb, 4);
1564  for (i = 1; i < pulse->num_pulse; i++) {
1565  pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
1566  if (pulse->pos[i] >= swb_offset[num_swb])
1567  return -1;
1568  pulse->amp[i] = get_bits(gb, 4);
1569  }
1570  return 0;
1571 }
1572 
1573 /**
1574  * Decode Temporal Noise Shaping data; reference: table 4.48.
1575  *
1576  * @return Returns error status. 0 - OK, !0 - error
1577  */
1579  GetBitContext *gb, const IndividualChannelStream *ics)
1580 {
1581  int w, filt, i, coef_len, coef_res, coef_compress;
1582  const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
1583  const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
1584  for (w = 0; w < ics->num_windows; w++) {
1585  if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
1586  coef_res = get_bits1(gb);
1587 
1588  for (filt = 0; filt < tns->n_filt[w]; filt++) {
1589  int tmp2_idx;
1590  tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
1591 
1592  if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
1593  av_log(ac->avctx, AV_LOG_ERROR,
1594  "TNS filter order %d is greater than maximum %d.\n",
1595  tns->order[w][filt], tns_max_order);
1596  tns->order[w][filt] = 0;
1597  return AVERROR_INVALIDDATA;
1598  }
1599  if (tns->order[w][filt]) {
1600  tns->direction[w][filt] = get_bits1(gb);
1601  coef_compress = get_bits1(gb);
1602  coef_len = coef_res + 3 - coef_compress;
1603  tmp2_idx = 2 * coef_compress + coef_res;
1604 
1605  for (i = 0; i < tns->order[w][filt]; i++)
1606  tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
1607  }
1608  }
1609  }
1610  }
1611  return 0;
1612 }
1613 
1614 /**
1615  * Decode Mid/Side data; reference: table 4.54.
1616  *
1617  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
1618  * [1] mask is decoded from bitstream; [2] mask is all 1s;
1619  * [3] reserved for scalable AAC
1620  */
1622  int ms_present)
1623 {
1624  int idx;
1625  int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
1626  if (ms_present == 1) {
1627  for (idx = 0; idx < max_idx; idx++)
1628  cpe->ms_mask[idx] = get_bits1(gb);
1629  } else if (ms_present == 2) {
1630  memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
1631  }
1632 }
1633 
1634 /**
1635  * Decode spectral data; reference: table 4.50.
1636  * Dequantize and scale spectral data; reference: 4.6.3.3.
1637  *
1638  * @param coef array of dequantized, scaled spectral data
1639  * @param sf array of scalefactors or intensity stereo positions
1640  * @param pulse_present set if pulses are present
1641  * @param pulse pointer to pulse data struct
1642  * @param band_type array of the used band type
1643  *
1644  * @return Returns error status. 0 - OK, !0 - error
1645  */
1647  GetBitContext *gb, const INTFLOAT sf[120],
1648  int pulse_present, const Pulse *pulse,
1649  const IndividualChannelStream *ics,
1650  enum BandType band_type[120])
1651 {
1652  int i, k, g, idx = 0;
1653  const int c = 1024 / ics->num_windows;
1654  const uint16_t *offsets = ics->swb_offset;
1655  INTFLOAT *coef_base = coef;
1656 
1657  for (g = 0; g < ics->num_windows; g++)
1658  memset(coef + g * 128 + offsets[ics->max_sfb], 0,
1659  sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
1660 
1661  for (g = 0; g < ics->num_window_groups; g++) {
1662  unsigned g_len = ics->group_len[g];
1663 
1664  for (i = 0; i < ics->max_sfb; i++, idx++) {
1665  const unsigned cbt_m1 = band_type[idx] - 1;
1666  INTFLOAT *cfo = coef + offsets[i];
1667  int off_len = offsets[i + 1] - offsets[i];
1668  int group;
1669 
1670  if (cbt_m1 >= INTENSITY_BT2 - 1) {
1671  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1672  memset(cfo, 0, off_len * sizeof(*cfo));
1673  }
1674  } else if (cbt_m1 == NOISE_BT - 1) {
1675  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1676 #if !USE_FIXED
1677  float scale;
1678 #endif /* !USE_FIXED */
1679  INTFLOAT band_energy;
1680 
1681  for (k = 0; k < off_len; k++) {
1683 #if USE_FIXED
1684  cfo[k] = ac->random_state >> 3;
1685 #else
1686  cfo[k] = ac->random_state;
1687 #endif /* USE_FIXED */
1688  }
1689 
1690 #if USE_FIXED
1691  band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
1692  band_energy = fixed_sqrt(band_energy, 31);
1693  noise_scale(cfo, sf[idx], band_energy, off_len);
1694 #else
1695  band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
1696  scale = sf[idx] / sqrtf(band_energy);
1697  ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
1698 #endif /* USE_FIXED */
1699  }
1700  } else {
1701 #if !USE_FIXED
1702  const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1703 #endif /* !USE_FIXED */
1704  const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1705  VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1706  OPEN_READER(re, gb);
1707 
1708  switch (cbt_m1 >> 1) {
1709  case 0:
1710  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1711  INTFLOAT *cf = cfo;
1712  int len = off_len;
1713 
1714  do {
1715  int code;
1716  unsigned cb_idx;
1717 
1718  UPDATE_CACHE(re, gb);
1719  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1720  cb_idx = cb_vector_idx[code];
1721 #if USE_FIXED
1722  cf = DEC_SQUAD(cf, cb_idx);
1723 #else
1724  cf = VMUL4(cf, vq, cb_idx, sf + idx);
1725 #endif /* USE_FIXED */
1726  } while (len -= 4);
1727  }
1728  break;
1729 
1730  case 1:
1731  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1732  INTFLOAT *cf = cfo;
1733  int len = off_len;
1734 
1735  do {
1736  int code;
1737  unsigned nnz;
1738  unsigned cb_idx;
1739  uint32_t bits;
1740 
1741  UPDATE_CACHE(re, gb);
1742  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1743  cb_idx = cb_vector_idx[code];
1744  nnz = cb_idx >> 8 & 15;
1745  bits = nnz ? GET_CACHE(re, gb) : 0;
1746  LAST_SKIP_BITS(re, gb, nnz);
1747 #if USE_FIXED
1748  cf = DEC_UQUAD(cf, cb_idx, bits);
1749 #else
1750  cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1751 #endif /* USE_FIXED */
1752  } while (len -= 4);
1753  }
1754  break;
1755 
1756  case 2:
1757  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1758  INTFLOAT *cf = cfo;
1759  int len = off_len;
1760 
1761  do {
1762  int code;
1763  unsigned cb_idx;
1764 
1765  UPDATE_CACHE(re, gb);
1766  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1767  cb_idx = cb_vector_idx[code];
1768 #if USE_FIXED
1769  cf = DEC_SPAIR(cf, cb_idx);
1770 #else
1771  cf = VMUL2(cf, vq, cb_idx, sf + idx);
1772 #endif /* USE_FIXED */
1773  } while (len -= 2);
1774  }
1775  break;
1776 
1777  case 3:
1778  case 4:
1779  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1780  INTFLOAT *cf = cfo;
1781  int len = off_len;
1782 
1783  do {
1784  int code;
1785  unsigned nnz;
1786  unsigned cb_idx;
1787  unsigned sign;
1788 
1789  UPDATE_CACHE(re, gb);
1790  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1791  cb_idx = cb_vector_idx[code];
1792  nnz = cb_idx >> 8 & 15;
1793  sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
1794  LAST_SKIP_BITS(re, gb, nnz);
1795 #if USE_FIXED
1796  cf = DEC_UPAIR(cf, cb_idx, sign);
1797 #else
1798  cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1799 #endif /* USE_FIXED */
1800  } while (len -= 2);
1801  }
1802  break;
1803 
1804  default:
1805  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1806 #if USE_FIXED
1807  int *icf = cfo;
1808  int v;
1809 #else
1810  float *cf = cfo;
1811  uint32_t *icf = (uint32_t *) cf;
1812 #endif /* USE_FIXED */
1813  int len = off_len;
1814 
1815  do {
1816  int code;
1817  unsigned nzt, nnz;
1818  unsigned cb_idx;
1819  uint32_t bits;
1820  int j;
1821 
1822  UPDATE_CACHE(re, gb);
1823  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1824 
1825  if (!code) {
1826  *icf++ = 0;
1827  *icf++ = 0;
1828  continue;
1829  }
1830 
1831  cb_idx = cb_vector_idx[code];
1832  nnz = cb_idx >> 12;
1833  nzt = cb_idx >> 8;
1834  bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1835  LAST_SKIP_BITS(re, gb, nnz);
1836 
1837  for (j = 0; j < 2; j++) {
1838  if (nzt & 1<<j) {
1839  uint32_t b;
1840  int n;
1841  /* The total length of escape_sequence must be < 22 bits according
1842  to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1843  UPDATE_CACHE(re, gb);
1844  b = GET_CACHE(re, gb);
1845  b = 31 - av_log2(~b);
1846 
1847  if (b > 8) {
1848  av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1849  return AVERROR_INVALIDDATA;
1850  }
1851 
1852  SKIP_BITS(re, gb, b + 1);
1853  b += 4;
1854  n = (1 << b) + SHOW_UBITS(re, gb, b);
1855  LAST_SKIP_BITS(re, gb, b);
1856 #if USE_FIXED
1857  v = n;
1858  if (bits & 1U<<31)
1859  v = -v;
1860  *icf++ = v;
1861 #else
1862  *icf++ = ff_cbrt_tab[n] | (bits & 1U<<31);
1863 #endif /* USE_FIXED */
1864  bits <<= 1;
1865  } else {
1866 #if USE_FIXED
1867  v = cb_idx & 15;
1868  if (bits & 1U<<31)
1869  v = -v;
1870  *icf++ = v;
1871 #else
1872  unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1873  *icf++ = (bits & 1U<<31) | v;
1874 #endif /* USE_FIXED */
1875  bits <<= !!v;
1876  }
1877  cb_idx >>= 4;
1878  }
1879  } while (len -= 2);
1880 #if !USE_FIXED
1881  ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1882 #endif /* !USE_FIXED */
1883  }
1884  }
1885 
1886  CLOSE_READER(re, gb);
1887  }
1888  }
1889  coef += g_len << 7;
1890  }
1891 
1892  if (pulse_present) {
1893  idx = 0;
1894  for (i = 0; i < pulse->num_pulse; i++) {
1895  INTFLOAT co = coef_base[ pulse->pos[i] ];
1896  while (offsets[idx + 1] <= pulse->pos[i])
1897  idx++;
1898  if (band_type[idx] != NOISE_BT && sf[idx]) {
1899  INTFLOAT ico = -pulse->amp[i];
1900 #if USE_FIXED
1901  if (co) {
1902  ico = co + (co > 0 ? -ico : ico);
1903  }
1904  coef_base[ pulse->pos[i] ] = ico;
1905 #else
1906  if (co) {
1907  co /= sf[idx];
1908  ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1909  }
1910  coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1911 #endif /* USE_FIXED */
1912  }
1913  }
1914  }
1915 #if USE_FIXED
1916  coef = coef_base;
1917  idx = 0;
1918  for (g = 0; g < ics->num_window_groups; g++) {
1919  unsigned g_len = ics->group_len[g];
1920 
1921  for (i = 0; i < ics->max_sfb; i++, idx++) {
1922  const unsigned cbt_m1 = band_type[idx] - 1;
1923  int *cfo = coef + offsets[i];
1924  int off_len = offsets[i + 1] - offsets[i];
1925  int group;
1926 
1927  if (cbt_m1 < NOISE_BT - 1) {
1928  for (group = 0; group < (int)g_len; group++, cfo+=128) {
1929  ac->vector_pow43(cfo, off_len);
1930  ac->subband_scale(cfo, cfo, sf[idx], 34, off_len);
1931  }
1932  }
1933  }
1934  coef += g_len << 7;
1935  }
1936 #endif /* USE_FIXED */
1937  return 0;
1938 }
1939 
1940 /**
1941  * Apply AAC-Main style frequency domain prediction.
1942  */
1944 {
1945  int sfb, k;
1946 
1947  if (!sce->ics.predictor_initialized) {
1949  sce->ics.predictor_initialized = 1;
1950  }
1951 
1952  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1953  for (sfb = 0;
1954  sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
1955  sfb++) {
1956  for (k = sce->ics.swb_offset[sfb];
1957  k < sce->ics.swb_offset[sfb + 1];
1958  k++) {
1959  predict(&sce->predictor_state[k], &sce->coeffs[k],
1960  sce->ics.predictor_present &&
1961  sce->ics.prediction_used[sfb]);
1962  }
1963  }
1964  if (sce->ics.predictor_reset_group)
1966  sce->ics.predictor_reset_group);
1967  } else
1969 }
1970 
1972 {
1973  // wd_num, wd_test, aloc_size
1974  static const uint8_t gain_mode[4][3] = {
1975  {1, 0, 5}, // ONLY_LONG_SEQUENCE = 0,
1976  {2, 1, 2}, // LONG_START_SEQUENCE,
1977  {8, 0, 2}, // EIGHT_SHORT_SEQUENCE,
1978  {2, 1, 5}, // LONG_STOP_SEQUENCE
1979  };
1980 
1981  const int mode = sce->ics.window_sequence[0];
1982  uint8_t bd, wd, ad;
1983 
1984  // FIXME: Store the gain control data on |sce| and do something with it.
1985  uint8_t max_band = get_bits(gb, 2);
1986  for (bd = 0; bd < max_band; bd++) {
1987  for (wd = 0; wd < gain_mode[mode][0]; wd++) {
1988  uint8_t adjust_num = get_bits(gb, 3);
1989  for (ad = 0; ad < adjust_num; ad++) {
1990  skip_bits(gb, 4 + ((wd == 0 && gain_mode[mode][1])
1991  ? 4
1992  : gain_mode[mode][2]));
1993  }
1994  }
1995  }
1996 }
1997 
1998 /**
1999  * Decode an individual_channel_stream payload; reference: table 4.44.
2000  *
2001  * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
2002  * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
2003  *
2004  * @return Returns error status. 0 - OK, !0 - error
2005  */
2007  GetBitContext *gb, int common_window, int scale_flag)
2008 {
2009  Pulse pulse;
2010  TemporalNoiseShaping *tns = &sce->tns;
2011  IndividualChannelStream *ics = &sce->ics;
2012  INTFLOAT *out = sce->coeffs;
2013  int global_gain, eld_syntax, er_syntax, pulse_present = 0;
2014  int ret;
2015 
2016  eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2017  er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
2018  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
2019  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
2020  ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2021 
2022  /* This assignment is to silence a GCC warning about the variable being used
2023  * uninitialized when in fact it always is.
2024  */
2025  pulse.num_pulse = 0;
2026 
2027  global_gain = get_bits(gb, 8);
2028 
2029  if (!common_window && !scale_flag) {
2030  ret = decode_ics_info(ac, ics, gb);
2031  if (ret < 0)
2032  goto fail;
2033  }
2034 
2035  if ((ret = decode_band_types(ac, sce->band_type,
2036  sce->band_type_run_end, gb, ics)) < 0)
2037  goto fail;
2038  if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
2039  sce->band_type, sce->band_type_run_end)) < 0)
2040  goto fail;
2041 
2042  pulse_present = 0;
2043  if (!scale_flag) {
2044  if (!eld_syntax && (pulse_present = get_bits1(gb))) {
2045  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2046  av_log(ac->avctx, AV_LOG_ERROR,
2047  "Pulse tool not allowed in eight short sequence.\n");
2048  ret = AVERROR_INVALIDDATA;
2049  goto fail;
2050  }
2051  if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
2052  av_log(ac->avctx, AV_LOG_ERROR,
2053  "Pulse data corrupt or invalid.\n");
2054  ret = AVERROR_INVALIDDATA;
2055  goto fail;
2056  }
2057  }
2058  tns->present = get_bits1(gb);
2059  if (tns->present && !er_syntax) {
2060  ret = decode_tns(ac, tns, gb, ics);
2061  if (ret < 0)
2062  goto fail;
2063  }
2064  if (!eld_syntax && get_bits1(gb)) {
2065  decode_gain_control(sce, gb);
2066  if (!ac->warned_gain_control) {
2067  avpriv_report_missing_feature(ac->avctx, "Gain control");
2068  ac->warned_gain_control = 1;
2069  }
2070  }
2071  // I see no textual basis in the spec for this occurring after SSR gain
2072  // control, but this is what both reference and real implmentations do
2073  if (tns->present && er_syntax) {
2074  ret = decode_tns(ac, tns, gb, ics);
2075  if (ret < 0)
2076  goto fail;
2077  }
2078  }
2079 
2080  ret = decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
2081  &pulse, ics, sce->band_type);
2082  if (ret < 0)
2083  goto fail;
2084 
2085  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
2086  apply_prediction(ac, sce);
2087 
2088  return 0;
2089 fail:
2090  tns->present = 0;
2091  return ret;
2092 }
2093 
2094 /**
2095  * Mid/Side stereo decoding; reference: 4.6.8.1.3.
2096  */
2098 {
2099  const IndividualChannelStream *ics = &cpe->ch[0].ics;
2100  INTFLOAT *ch0 = cpe->ch[0].coeffs;
2101  INTFLOAT *ch1 = cpe->ch[1].coeffs;
2102  int g, i, group, idx = 0;
2103  const uint16_t *offsets = ics->swb_offset;
2104  for (g = 0; g < ics->num_window_groups; g++) {
2105  for (i = 0; i < ics->max_sfb; i++, idx++) {
2106  if (cpe->ms_mask[idx] &&
2107  cpe->ch[0].band_type[idx] < NOISE_BT &&
2108  cpe->ch[1].band_type[idx] < NOISE_BT) {
2109 #if USE_FIXED
2110  for (group = 0; group < ics->group_len[g]; group++) {
2111  ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
2112  ch1 + group * 128 + offsets[i],
2113  offsets[i+1] - offsets[i]);
2114 #else
2115  for (group = 0; group < ics->group_len[g]; group++) {
2116  ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
2117  ch1 + group * 128 + offsets[i],
2118  offsets[i+1] - offsets[i]);
2119 #endif /* USE_FIXED */
2120  }
2121  }
2122  }
2123  ch0 += ics->group_len[g] * 128;
2124  ch1 += ics->group_len[g] * 128;
2125  }
2126 }
2127 
2128 /**
2129  * intensity stereo decoding; reference: 4.6.8.2.3
2130  *
2131  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
2132  * [1] mask is decoded from bitstream; [2] mask is all 1s;
2133  * [3] reserved for scalable AAC
2134  */
2136  ChannelElement *cpe, int ms_present)
2137 {
2138  const IndividualChannelStream *ics = &cpe->ch[1].ics;
2139  SingleChannelElement *sce1 = &cpe->ch[1];
2140  INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
2141  const uint16_t *offsets = ics->swb_offset;
2142  int g, group, i, idx = 0;
2143  int c;
2144  INTFLOAT scale;
2145  for (g = 0; g < ics->num_window_groups; g++) {
2146  for (i = 0; i < ics->max_sfb;) {
2147  if (sce1->band_type[idx] == INTENSITY_BT ||
2148  sce1->band_type[idx] == INTENSITY_BT2) {
2149  const int bt_run_end = sce1->band_type_run_end[idx];
2150  for (; i < bt_run_end; i++, idx++) {
2151  c = -1 + 2 * (sce1->band_type[idx] - 14);
2152  if (ms_present)
2153  c *= 1 - 2 * cpe->ms_mask[idx];
2154  scale = c * sce1->sf[idx];
2155  for (group = 0; group < ics->group_len[g]; group++)
2156 #if USE_FIXED
2157  ac->subband_scale(coef1 + group * 128 + offsets[i],
2158  coef0 + group * 128 + offsets[i],
2159  scale,
2160  23,
2161  offsets[i + 1] - offsets[i]);
2162 #else
2163  ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
2164  coef0 + group * 128 + offsets[i],
2165  scale,
2166  offsets[i + 1] - offsets[i]);
2167 #endif /* USE_FIXED */
2168  }
2169  } else {
2170  int bt_run_end = sce1->band_type_run_end[idx];
2171  idx += bt_run_end - i;
2172  i = bt_run_end;
2173  }
2174  }
2175  coef0 += ics->group_len[g] * 128;
2176  coef1 += ics->group_len[g] * 128;
2177  }
2178 }
2179 
2180 /**
2181  * Decode a channel_pair_element; reference: table 4.4.
2182  *
2183  * @return Returns error status. 0 - OK, !0 - error
2184  */
2186 {
2187  int i, ret, common_window, ms_present = 0;
2188  int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2189 
2190  common_window = eld_syntax || get_bits1(gb);
2191  if (common_window) {
2192  if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
2193  return AVERROR_INVALIDDATA;
2194  i = cpe->ch[1].ics.use_kb_window[0];
2195  cpe->ch[1].ics = cpe->ch[0].ics;
2196  cpe->ch[1].ics.use_kb_window[1] = i;
2197  if (cpe->ch[1].ics.predictor_present &&
2198  (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
2199  if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
2200  decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
2201  ms_present = get_bits(gb, 2);
2202  if (ms_present == 3) {
2203  av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
2204  return AVERROR_INVALIDDATA;
2205  } else if (ms_present)
2206  decode_mid_side_stereo(cpe, gb, ms_present);
2207  }
2208  if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
2209  return ret;
2210  if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
2211  return ret;
2212 
2213  if (common_window) {
2214  if (ms_present)
2215  apply_mid_side_stereo(ac, cpe);
2216  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
2217  apply_prediction(ac, &cpe->ch[0]);
2218  apply_prediction(ac, &cpe->ch[1]);
2219  }
2220  }
2221 
2222  apply_intensity_stereo(ac, cpe, ms_present);
2223  return 0;
2224 }
2225 
2226 static const float cce_scale[] = {
2227  1.09050773266525765921, //2^(1/8)
2228  1.18920711500272106672, //2^(1/4)
2229  M_SQRT2,
2230  2,
2231 };
2232 
2233 /**
2234  * Decode coupling_channel_element; reference: table 4.8.
2235  *
2236  * @return Returns error status. 0 - OK, !0 - error
2237  */
2239 {
2240  int num_gain = 0;
2241  int c, g, sfb, ret;
2242  int sign;
2243  INTFLOAT scale;
2244  SingleChannelElement *sce = &che->ch[0];
2245  ChannelCoupling *coup = &che->coup;
2246 
2247  coup->coupling_point = 2 * get_bits1(gb);
2248  coup->num_coupled = get_bits(gb, 3);
2249  for (c = 0; c <= coup->num_coupled; c++) {
2250  num_gain++;
2251  coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
2252  coup->id_select[c] = get_bits(gb, 4);
2253  if (coup->type[c] == TYPE_CPE) {
2254  coup->ch_select[c] = get_bits(gb, 2);
2255  if (coup->ch_select[c] == 3)
2256  num_gain++;
2257  } else
2258  coup->ch_select[c] = 2;
2259  }
2260  coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
2261 
2262  sign = get_bits(gb, 1);
2263 #if USE_FIXED
2264  scale = get_bits(gb, 2);
2265 #else
2266  scale = cce_scale[get_bits(gb, 2)];
2267 #endif
2268 
2269  if ((ret = decode_ics(ac, sce, gb, 0, 0)))
2270  return ret;
2271 
2272  for (c = 0; c < num_gain; c++) {
2273  int idx = 0;
2274  int cge = 1;
2275  int gain = 0;
2276  INTFLOAT gain_cache = FIXR10(1.);
2277  if (c) {
2278  cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
2279  gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
2280  gain_cache = GET_GAIN(scale, gain);
2281 #if USE_FIXED
2282  if ((abs(gain_cache)-1024) >> 3 > 30)
2283  return AVERROR(ERANGE);
2284 #endif
2285  }
2286  if (coup->coupling_point == AFTER_IMDCT) {
2287  coup->gain[c][0] = gain_cache;
2288  } else {
2289  for (g = 0; g < sce->ics.num_window_groups; g++) {
2290  for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
2291  if (sce->band_type[idx] != ZERO_BT) {
2292  if (!cge) {
2293  int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
2294  if (t) {
2295  int s = 1;
2296  t = gain += t;
2297  if (sign) {
2298  s -= 2 * (t & 0x1);
2299  t >>= 1;
2300  }
2301  gain_cache = GET_GAIN(scale, t) * s;
2302 #if USE_FIXED
2303  if ((abs(gain_cache)-1024) >> 3 > 30)
2304  return AVERROR(ERANGE);
2305 #endif
2306  }
2307  }
2308  coup->gain[c][idx] = gain_cache;
2309  }
2310  }
2311  }
2312  }
2313  }
2314  return 0;
2315 }
2316 
2317 /**
2318  * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
2319  *
2320  * @return Returns number of bytes consumed.
2321  */
2323  GetBitContext *gb)
2324 {
2325  int i;
2326  int num_excl_chan = 0;
2327 
2328  do {
2329  for (i = 0; i < 7; i++)
2330  che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
2331  } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
2332 
2333  return num_excl_chan / 7;
2334 }
2335 
2336 /**
2337  * Decode dynamic range information; reference: table 4.52.
2338  *
2339  * @return Returns number of bytes consumed.
2340  */
2342  GetBitContext *gb)
2343 {
2344  int n = 1;
2345  int drc_num_bands = 1;
2346  int i;
2347 
2348  /* pce_tag_present? */
2349  if (get_bits1(gb)) {
2350  che_drc->pce_instance_tag = get_bits(gb, 4);
2351  skip_bits(gb, 4); // tag_reserved_bits
2352  n++;
2353  }
2354 
2355  /* excluded_chns_present? */
2356  if (get_bits1(gb)) {
2357  n += decode_drc_channel_exclusions(che_drc, gb);
2358  }
2359 
2360  /* drc_bands_present? */
2361  if (get_bits1(gb)) {
2362  che_drc->band_incr = get_bits(gb, 4);
2363  che_drc->interpolation_scheme = get_bits(gb, 4);
2364  n++;
2365  drc_num_bands += che_drc->band_incr;
2366  for (i = 0; i < drc_num_bands; i++) {
2367  che_drc->band_top[i] = get_bits(gb, 8);
2368  n++;
2369  }
2370  }
2371 
2372  /* prog_ref_level_present? */
2373  if (get_bits1(gb)) {
2374  che_drc->prog_ref_level = get_bits(gb, 7);
2375  skip_bits1(gb); // prog_ref_level_reserved_bits
2376  n++;
2377  }
2378 
2379  for (i = 0; i < drc_num_bands; i++) {
2380  che_drc->dyn_rng_sgn[i] = get_bits1(gb);
2381  che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
2382  n++;
2383  }
2384 
2385  return n;
2386 }
2387 
2388 static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
2389  uint8_t buf[256];
2390  int i, major, minor;
2391 
2392  if (len < 13+7*8)
2393  goto unknown;
2394 
2395  get_bits(gb, 13); len -= 13;
2396 
2397  for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
2398  buf[i] = get_bits(gb, 8);
2399 
2400  buf[i] = 0;
2401  if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
2402  av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
2403 
2404  if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
2405  ac->avctx->internal->skip_samples = 1024;
2406  }
2407 
2408 unknown:
2409  skip_bits_long(gb, len);
2410 
2411  return 0;
2412 }
2413 
2414 /**
2415  * Decode extension data (incomplete); reference: table 4.51.
2416  *
2417  * @param cnt length of TYPE_FIL syntactic element in bytes
2418  *
2419  * @return Returns number of bytes consumed
2420  */
2422  ChannelElement *che, enum RawDataBlockType elem_type)
2423 {
2424  int crc_flag = 0;
2425  int res = cnt;
2426  int type = get_bits(gb, 4);
2427 
2428  if (ac->avctx->debug & FF_DEBUG_STARTCODE)
2429  av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
2430 
2431  switch (type) { // extension type
2432  case EXT_SBR_DATA_CRC:
2433  crc_flag++;
2434  case EXT_SBR_DATA:
2435  if (!che) {
2436  av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
2437  return res;
2438  } else if (ac->oc[1].m4ac.frame_length_short) {
2439  if (!ac->warned_960_sbr)
2441  "SBR with 960 frame length");
2442  ac->warned_960_sbr = 1;
2443  skip_bits_long(gb, 8 * cnt - 4);
2444  return res;
2445  } else if (!ac->oc[1].m4ac.sbr) {
2446  av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
2447  skip_bits_long(gb, 8 * cnt - 4);
2448  return res;
2449  } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
2450  av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
2451  skip_bits_long(gb, 8 * cnt - 4);
2452  return res;
2453  } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
2454  ac->oc[1].m4ac.sbr = 1;
2455  ac->oc[1].m4ac.ps = 1;
2457  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
2458  ac->oc[1].status, 1);
2459  } else {
2460  ac->oc[1].m4ac.sbr = 1;
2462  }
2463  res = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
2464  break;
2465  case EXT_DYNAMIC_RANGE:
2466  res = decode_dynamic_range(&ac->che_drc, gb);
2467  break;
2468  case EXT_FILL:
2469  decode_fill(ac, gb, 8 * cnt - 4);
2470  break;
2471  case EXT_FILL_DATA:
2472  case EXT_DATA_ELEMENT:
2473  default:
2474  skip_bits_long(gb, 8 * cnt - 4);
2475  break;
2476  };
2477  return res;
2478 }
2479 
2480 /**
2481  * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
2482  *
2483  * @param decode 1 if tool is used normally, 0 if tool is used in LTP.
2484  * @param coef spectral coefficients
2485  */
2486 static void apply_tns(INTFLOAT coef_param[1024], TemporalNoiseShaping *tns,
2487  IndividualChannelStream *ics, int decode)
2488 {
2489  const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
2490  int w, filt, m, i;
2491  int bottom, top, order, start, end, size, inc;
2492  INTFLOAT lpc[TNS_MAX_ORDER];
2494  UINTFLOAT *coef = coef_param;
2495 
2496  for (w = 0; w < ics->num_windows; w++) {
2497  bottom = ics->num_swb;
2498  for (filt = 0; filt < tns->n_filt[w]; filt++) {
2499  top = bottom;
2500  bottom = FFMAX(0, top - tns->length[w][filt]);
2501  order = tns->order[w][filt];
2502  if (order == 0)
2503  continue;
2504 
2505  // tns_decode_coef
2506  AAC_RENAME(compute_lpc_coefs)(tns->coef[w][filt], order, lpc, 0, 0, 0);
2507 
2508  start = ics->swb_offset[FFMIN(bottom, mmm)];
2509  end = ics->swb_offset[FFMIN( top, mmm)];
2510  if ((size = end - start) <= 0)
2511  continue;
2512  if (tns->direction[w][filt]) {
2513  inc = -1;
2514  start = end - 1;
2515  } else {
2516  inc = 1;
2517  }
2518  start += w * 128;
2519 
2520  if (decode) {
2521  // ar filter
2522  for (m = 0; m < size; m++, start += inc)
2523  for (i = 1; i <= FFMIN(m, order); i++)
2524  coef[start] -= AAC_MUL26((INTFLOAT)coef[start - i * inc], lpc[i - 1]);
2525  } else {
2526  // ma filter
2527  for (m = 0; m < size; m++, start += inc) {
2528  tmp[0] = coef[start];
2529  for (i = 1; i <= FFMIN(m, order); i++)
2530  coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
2531  for (i = order; i > 0; i--)
2532  tmp[i] = tmp[i - 1];
2533  }
2534  }
2535  }
2536  }
2537 }
2538 
2539 /**
2540  * Apply windowing and MDCT to obtain the spectral
2541  * coefficient from the predicted sample by LTP.
2542  */
2545 {
2546  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2547  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2548  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2549  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2550 
2551  if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
2552  ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
2553  } else {
2554  memset(in, 0, 448 * sizeof(*in));
2555  ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
2556  }
2557  if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
2558  ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
2559  } else {
2560  ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
2561  memset(in + 1024 + 576, 0, 448 * sizeof(*in));
2562  }
2563  ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
2564 }
2565 
2566 /**
2567  * Apply the long term prediction
2568  */
2570 {
2571  const LongTermPrediction *ltp = &sce->ics.ltp;
2572  const uint16_t *offsets = sce->ics.swb_offset;
2573  int i, sfb;
2574 
2575  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
2576  INTFLOAT *predTime = sce->ret;
2577  INTFLOAT *predFreq = ac->buf_mdct;
2578  int16_t num_samples = 2048;
2579 
2580  if (ltp->lag < 1024)
2581  num_samples = ltp->lag + 1024;
2582  for (i = 0; i < num_samples; i++)
2583  predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
2584  memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));
2585 
2586  ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
2587 
2588  if (sce->tns.present)
2589  ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
2590 
2591  for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
2592  if (ltp->used[sfb])
2593  for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
2594  sce->coeffs[i] += (UINTFLOAT)predFreq[i];
2595  }
2596 }
2597 
2598 /**
2599  * Update the LTP buffer for next frame
2600  */
2602 {
2603  IndividualChannelStream *ics = &sce->ics;
2604  INTFLOAT *saved = sce->saved;
2605  INTFLOAT *saved_ltp = sce->coeffs;
2606  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2607  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2608  int i;
2609 
2610  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2611  memcpy(saved_ltp, saved, 512 * sizeof(*saved_ltp));
2612  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2613  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2614 
2615  for (i = 0; i < 64; i++)
2616  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2617  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2618  memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(*saved_ltp));
2619  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2620  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2621 
2622  for (i = 0; i < 64; i++)
2623  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2624  } else { // LONG_STOP or ONLY_LONG
2625  ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
2626 
2627  for (i = 0; i < 512; i++)
2628  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
2629  }
2630 
2631  memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
2632  memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
2633  memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
2634 }
2635 
2636 /**
2637  * Conduct IMDCT and windowing.
2638  */
2640 {
2641  IndividualChannelStream *ics = &sce->ics;
2642  INTFLOAT *in = sce->coeffs;
2643  INTFLOAT *out = sce->ret;
2644  INTFLOAT *saved = sce->saved;
2645  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2646  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2647  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2648  INTFLOAT *buf = ac->buf_mdct;
2649  INTFLOAT *temp = ac->temp;
2650  int i;
2651 
2652  // imdct
2653  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2654  for (i = 0; i < 1024; i += 128)
2655  ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
2656  } else {
2657  ac->mdct.imdct_half(&ac->mdct, buf, in);
2658 #if USE_FIXED
2659  for (i=0; i<1024; i++)
2660  buf[i] = (buf[i] + 4) >> 3;
2661 #endif /* USE_FIXED */
2662  }
2663 
2664  /* window overlapping
2665  * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2666  * and long to short transitions are considered to be short to short
2667  * transitions. This leaves just two cases (long to long and short to short)
2668  * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2669  */
2670  if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2672  ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
2673  } else {
2674  memcpy( out, saved, 448 * sizeof(*out));
2675 
2676  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2677  ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
2678  ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
2679  ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
2680  ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
2681  ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
2682  memcpy( out + 448 + 4*128, temp, 64 * sizeof(*out));
2683  } else {
2684  ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
2685  memcpy( out + 576, buf + 64, 448 * sizeof(*out));
2686  }
2687  }
2688 
2689  // buffer update
2690  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2691  memcpy( saved, temp + 64, 64 * sizeof(*saved));
2692  ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
2693  ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
2694  ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
2695  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2696  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2697  memcpy( saved, buf + 512, 448 * sizeof(*saved));
2698  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2699  } else { // LONG_STOP or ONLY_LONG
2700  memcpy( saved, buf + 512, 512 * sizeof(*saved));
2701  }
2702 }
2703 
2704 /**
2705  * Conduct IMDCT and windowing.
2706  */
2708 {
2709 #if !USE_FIXED
2710  IndividualChannelStream *ics = &sce->ics;
2711  INTFLOAT *in = sce->coeffs;
2712  INTFLOAT *out = sce->ret;
2713  INTFLOAT *saved = sce->saved;
2714  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_120) : AAC_RENAME(ff_sine_120);
2715  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_960) : AAC_RENAME(ff_sine_960);
2716  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_120) : AAC_RENAME(ff_sine_120);
2717  INTFLOAT *buf = ac->buf_mdct;
2718  INTFLOAT *temp = ac->temp;
2719  int i;
2720 
2721  // imdct
2722  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2723  for (i = 0; i < 8; i++)
2724  ac->mdct120->imdct_half(ac->mdct120, buf + i * 120, in + i * 128, 1);
2725  } else {
2726  ac->mdct960->imdct_half(ac->mdct960, buf, in, 1);
2727  }
2728 
2729  /* window overlapping
2730  * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2731  * and long to short transitions are considered to be short to short
2732  * transitions. This leaves just two cases (long to long and short to short)
2733  * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2734  */
2735 
2736  if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2738  ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 480);
2739  } else {
2740  memcpy( out, saved, 420 * sizeof(*out));
2741 
2742  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2743  ac->fdsp->vector_fmul_window(out + 420 + 0*120, saved + 420, buf + 0*120, swindow_prev, 60);
2744  ac->fdsp->vector_fmul_window(out + 420 + 1*120, buf + 0*120 + 60, buf + 1*120, swindow, 60);
2745  ac->fdsp->vector_fmul_window(out + 420 + 2*120, buf + 1*120 + 60, buf + 2*120, swindow, 60);
2746  ac->fdsp->vector_fmul_window(out + 420 + 3*120, buf + 2*120 + 60, buf + 3*120, swindow, 60);
2747  ac->fdsp->vector_fmul_window(temp, buf + 3*120 + 60, buf + 4*120, swindow, 60);
2748  memcpy( out + 420 + 4*120, temp, 60 * sizeof(*out));
2749  } else {
2750  ac->fdsp->vector_fmul_window(out + 420, saved + 420, buf, swindow_prev, 60);
2751  memcpy( out + 540, buf + 60, 420 * sizeof(*out));
2752  }
2753  }
2754 
2755  // buffer update
2756  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2757  memcpy( saved, temp + 60, 60 * sizeof(*saved));
2758  ac->fdsp->vector_fmul_window(saved + 60, buf + 4*120 + 60, buf + 5*120, swindow, 60);
2759  ac->fdsp->vector_fmul_window(saved + 180, buf + 5*120 + 60, buf + 6*120, swindow, 60);
2760  ac->fdsp->vector_fmul_window(saved + 300, buf + 6*120 + 60, buf + 7*120, swindow, 60);
2761  memcpy( saved + 420, buf + 7*120 + 60, 60 * sizeof(*saved));
2762  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2763  memcpy( saved, buf + 480, 420 * sizeof(*saved));
2764  memcpy( saved + 420, buf + 7*120 + 60, 60 * sizeof(*saved));
2765  } else { // LONG_STOP or ONLY_LONG
2766  memcpy( saved, buf + 480, 480 * sizeof(*saved));
2767  }
2768 #endif
2769 }
2771 {
2772  IndividualChannelStream *ics = &sce->ics;
2773  INTFLOAT *in = sce->coeffs;
2774  INTFLOAT *out = sce->ret;
2775  INTFLOAT *saved = sce->saved;
2776  INTFLOAT *buf = ac->buf_mdct;
2777 #if USE_FIXED
2778  int i;
2779 #endif /* USE_FIXED */
2780 
2781  // imdct
2782  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2783 
2784 #if USE_FIXED
2785  for (i = 0; i < 1024; i++)
2786  buf[i] = (buf[i] + 2) >> 2;
2787 #endif /* USE_FIXED */
2788 
2789  // window overlapping
2790  if (ics->use_kb_window[1]) {
2791  // AAC LD uses a low overlap sine window instead of a KBD window
2792  memcpy(out, saved, 192 * sizeof(*out));
2793  ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
2794  memcpy( out + 320, buf + 64, 192 * sizeof(*out));
2795  } else {
2796  ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);
2797  }
2798 
2799  // buffer update
2800  memcpy(saved, buf + 256, 256 * sizeof(*saved));
2801 }
2802 
2804 {
2805  INTFLOAT *in = sce->coeffs;
2806  INTFLOAT *out = sce->ret;
2807  INTFLOAT *saved = sce->saved;
2808  INTFLOAT *buf = ac->buf_mdct;
2809  int i;
2810  const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
2811  const int n2 = n >> 1;
2812  const int n4 = n >> 2;
2813  const INTFLOAT *const window = n == 480 ? AAC_RENAME(ff_aac_eld_window_480) :
2815 
2816  // Inverse transform, mapped to the conventional IMDCT by
2817  // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
2818  // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
2819  // International Conference on Audio, Language and Image Processing, ICALIP 2008.
2820  // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
2821  for (i = 0; i < n2; i+=2) {
2822  INTFLOAT temp;
2823  temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
2824  temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
2825  }
2826 #if !USE_FIXED
2827  if (n == 480)
2828  ac->mdct480->imdct_half(ac->mdct480, buf, in, 1);
2829  else
2830 #endif
2831  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2832 
2833 #if USE_FIXED
2834  for (i = 0; i < 1024; i++)
2835  buf[i] = (buf[i] + 1) >> 1;
2836 #endif /* USE_FIXED */
2837 
2838  for (i = 0; i < n; i+=2) {
2839  buf[i] = -buf[i];
2840  }
2841  // Like with the regular IMDCT at this point we still have the middle half
2842  // of a transform but with even symmetry on the left and odd symmetry on
2843  // the right
2844 
2845  // window overlapping
2846  // The spec says to use samples [0..511] but the reference decoder uses
2847  // samples [128..639].
2848  for (i = n4; i < n2; i ++) {
2849  out[i - n4] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) +
2850  AAC_MUL31( saved[ i + n2] , window[i + n - n4]) +
2851  AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
2852  AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]);
2853  }
2854  for (i = 0; i < n2; i ++) {
2855  out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) +
2856  AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) +
2857  AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) +
2858  AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
2859  }
2860  for (i = 0; i < n4; i ++) {
2861  out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) +
2862  AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
2863  AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
2864  }
2865 
2866  // buffer update
2867  memmove(saved + n, saved, 2 * n * sizeof(*saved));
2868  memcpy( saved, buf, n * sizeof(*saved));
2869 }
2870 
2871 /**
2872  * channel coupling transformation interface
2873  *
2874  * @param apply_coupling_method pointer to (in)dependent coupling function
2875  */
2877  enum RawDataBlockType type, int elem_id,
2878  enum CouplingPoint coupling_point,
2879  void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
2880 {
2881  int i, c;
2882 
2883  for (i = 0; i < MAX_ELEM_ID; i++) {
2884  ChannelElement *cce = ac->che[TYPE_CCE][i];
2885  int index = 0;
2886 
2887  if (cce && cce->coup.coupling_point == coupling_point) {
2888  ChannelCoupling *coup = &cce->coup;
2889 
2890  for (c = 0; c <= coup->num_coupled; c++) {
2891  if (coup->type[c] == type && coup->id_select[c] == elem_id) {
2892  if (coup->ch_select[c] != 1) {
2893  apply_coupling_method(ac, &cc->ch[0], cce, index);
2894  if (coup->ch_select[c] != 0)
2895  index++;
2896  }
2897  if (coup->ch_select[c] != 2)
2898  apply_coupling_method(ac, &cc->ch[1], cce, index++);
2899  } else
2900  index += 1 + (coup->ch_select[c] == 3);
2901  }
2902  }
2903  }
2904 }
2905 
2906 /**
2907  * Convert spectral data to samples, applying all supported tools as appropriate.
2908  */
2909 static void spectral_to_sample(AACContext *ac, int samples)
2910 {
2911  int i, type;
2913  switch (ac->oc[1].m4ac.object_type) {
2914  case AOT_ER_AAC_LD:
2916  break;
2917  case AOT_ER_AAC_ELD:
2919  break;
2920  default:
2921  if (ac->oc[1].m4ac.frame_length_short)
2923  else
2925  }
2926  for (type = 3; type >= 0; type--) {
2927  for (i = 0; i < MAX_ELEM_ID; i++) {
2928  ChannelElement *che = ac->che[type][i];
2929  if (che && che->present) {
2930  if (type <= TYPE_CPE)
2932  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2933  if (che->ch[0].ics.predictor_present) {
2934  if (che->ch[0].ics.ltp.present)
2935  ac->apply_ltp(ac, &che->ch[0]);
2936  if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
2937  ac->apply_ltp(ac, &che->ch[1]);
2938  }
2939  }
2940  if (che->ch[0].tns.present)
2941  ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2942  if (che->ch[1].tns.present)
2943  ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2944  if (type <= TYPE_CPE)
2946  if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2947  imdct_and_window(ac, &che->ch[0]);
2948  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2949  ac->update_ltp(ac, &che->ch[0]);
2950  if (type == TYPE_CPE) {
2951  imdct_and_window(ac, &che->ch[1]);
2952  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2953  ac->update_ltp(ac, &che->ch[1]);
2954  }
2955  if (ac->oc[1].m4ac.sbr > 0) {
2956  AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2957  }
2958  }
2959  if (type <= TYPE_CCE)
2961 
2962 #if USE_FIXED
2963  {
2964  int j;
2965  /* preparation for resampler */
2966  for(j = 0; j<samples; j++){
2967  che->ch[0].ret[j] = (int32_t)av_clip64((int64_t)che->ch[0].ret[j]*128, INT32_MIN, INT32_MAX-0x8000)+0x8000;
2968  if(type == TYPE_CPE)
2969  che->ch[1].ret[j] = (int32_t)av_clip64((int64_t)che->ch[1].ret[j]*128, INT32_MIN, INT32_MAX-0x8000)+0x8000;
2970  }
2971  }
2972 #endif /* USE_FIXED */
2973  che->present = 0;
2974  } else if (che) {
2975  av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
2976  }
2977  }
2978  }
2979 }
2980 
2982 {
2983  int size;
2984  AACADTSHeaderInfo hdr_info;
2985  uint8_t layout_map[MAX_ELEM_ID*4][3];
2986  int layout_map_tags, ret;
2987 
2988  size = ff_adts_header_parse(gb, &hdr_info);
2989  if (size > 0) {
2990  if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
2991  // This is 2 for "VLB " audio in NSV files.
2992  // See samples/nsv/vlb_audio.
2994  "More than one AAC RDB per ADTS frame");
2995  ac->warned_num_aac_frames = 1;
2996  }
2998  if (hdr_info.chan_config) {
2999  ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
3000  if ((ret = set_default_channel_config(ac->avctx,
3001  layout_map,
3002  &layout_map_tags,
3003  hdr_info.chan_config)) < 0)
3004  return ret;
3005  if ((ret = output_configure(ac, layout_map, layout_map_tags,
3006  FFMAX(ac->oc[1].status,
3007  OC_TRIAL_FRAME), 0)) < 0)
3008  return ret;
3009  } else {
3010  ac->oc[1].m4ac.chan_config = 0;
3011  /**
3012  * dual mono frames in Japanese DTV can have chan_config 0
3013  * WITHOUT specifying PCE.
3014  * thus, set dual mono as default.
3015  */
3016  if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
3017  layout_map_tags = 2;
3018  layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
3019  layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
3020  layout_map[0][1] = 0;
3021  layout_map[1][1] = 1;
3022  if (output_configure(ac, layout_map, layout_map_tags,
3023  OC_TRIAL_FRAME, 0))
3024  return -7;
3025  }
3026  }
3027  ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
3028  ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
3029  ac->oc[1].m4ac.object_type = hdr_info.object_type;
3030  ac->oc[1].m4ac.frame_length_short = 0;
3031  if (ac->oc[0].status != OC_LOCKED ||
3032  ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
3033  ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
3034  ac->oc[1].m4ac.sbr = -1;
3035  ac->oc[1].m4ac.ps = -1;
3036  }
3037  if (!hdr_info.crc_absent)
3038  skip_bits(gb, 16);
3039  }
3040  return size;
3041 }
3042 
3043 static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
3044  int *got_frame_ptr, GetBitContext *gb)
3045 {
3046  AACContext *ac = avctx->priv_data;
3047  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
3048  ChannelElement *che;
3049  int err, i;
3050  int samples = m4ac->frame_length_short ? 960 : 1024;
3051  int chan_config = m4ac->chan_config;
3052  int aot = m4ac->object_type;
3053 
3054  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
3055  samples >>= 1;
3056 
3057  ac->frame = data;
3058 
3059  if ((err = frame_configure_elements(avctx)) < 0)
3060  return err;
3061 
3062  // The FF_PROFILE_AAC_* defines are all object_type - 1
3063  // This may lead to an undefined profile being signaled
3064  ac->avctx->profile = aot - 1;
3065 
3066  ac->tags_mapped = 0;
3067 
3068  if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
3069  avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
3070  chan_config);
3071  return AVERROR_INVALIDDATA;
3072  }
3073  for (i = 0; i < tags_per_config[chan_config]; i++) {
3074  const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
3075  const int elem_id = aac_channel_layout_map[chan_config-1][i][1];
3076  if (!(che=get_che(ac, elem_type, elem_id))) {
3077  av_log(ac->avctx, AV_LOG_ERROR,
3078  "channel element %d.%d is not allocated\n",
3079  elem_type, elem_id);
3080  return AVERROR_INVALIDDATA;
3081  }
3082  che->present = 1;
3083  if (aot != AOT_ER_AAC_ELD)
3084  skip_bits(gb, 4);
3085  switch (elem_type) {
3086  case TYPE_SCE:
3087  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3088  break;
3089  case TYPE_CPE:
3090  err = decode_cpe(ac, gb, che);
3091  break;
3092  case TYPE_LFE:
3093  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3094  break;
3095  }
3096  if (err < 0)
3097  return err;
3098  }
3099 
3100  spectral_to_sample(ac, samples);
3101 
3102  if (!ac->frame->data[0] && samples) {
3103  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
3104  return AVERROR_INVALIDDATA;
3105  }
3106 
3107  ac->frame->nb_samples = samples;
3108  ac->frame->sample_rate = avctx->sample_rate;
3109  *got_frame_ptr = 1;
3110 
3111  skip_bits_long(gb, get_bits_left(gb));
3112  return 0;
3113 }
3114 
3115 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
3116  int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
3117 {
3118  AACContext *ac = avctx->priv_data;
3119  ChannelElement *che = NULL, *che_prev = NULL;
3120  enum RawDataBlockType elem_type, che_prev_type = TYPE_END;
3121  int err, elem_id;
3122  int samples = 0, multiplier, audio_found = 0, pce_found = 0;
3123  int is_dmono, sce_count = 0;
3124  int payload_alignment;
3125  uint8_t che_presence[4][MAX_ELEM_ID] = {{0}};
3126 
3127  ac->frame = data;
3128 
3129  if (show_bits(gb, 12) == 0xfff) {
3130  if ((err = parse_adts_frame_header(ac, gb)) < 0) {
3131  av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
3132  goto fail;
3133  }
3134  if (ac->oc[1].m4ac.sampling_index > 12) {
3135  av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
3136  err = AVERROR_INVALIDDATA;
3137  goto fail;
3138  }
3139  }
3140 
3141  if ((err = frame_configure_elements(avctx)) < 0)
3142  goto fail;
3143 
3144  // The FF_PROFILE_AAC_* defines are all object_type - 1
3145  // This may lead to an undefined profile being signaled
3146  ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
3147 
3148  payload_alignment = get_bits_count(gb);
3149  ac->tags_mapped = 0;
3150  // parse
3151  while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
3152  elem_id = get_bits(gb, 4);
3153 
3154  if (avctx->debug & FF_DEBUG_STARTCODE)
3155  av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
3156 
3157  if (!avctx->channels && elem_type != TYPE_PCE) {
3158  err = AVERROR_INVALIDDATA;
3159  goto fail;
3160  }
3161 
3162  if (elem_type < TYPE_DSE) {
3163  if (che_presence[elem_type][elem_id]) {
3164  int error = che_presence[elem_type][elem_id] > 1;
3165  av_log(ac->avctx, error ? AV_LOG_ERROR : AV_LOG_DEBUG, "channel element %d.%d duplicate\n",
3166  elem_type, elem_id);
3167  if (error) {
3168  err = AVERROR_INVALIDDATA;
3169  goto fail;
3170  }
3171  }
3172  che_presence[elem_type][elem_id]++;
3173 
3174  if (!(che=get_che(ac, elem_type, elem_id))) {
3175  av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
3176  elem_type, elem_id);
3177  err = AVERROR_INVALIDDATA;
3178  goto fail;
3179  }
3180  samples = ac->oc[1].m4ac.frame_length_short ? 960 : 1024;
3181  che->present = 1;
3182  }
3183 
3184  switch (elem_type) {
3185 
3186  case TYPE_SCE:
3187  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3188  audio_found = 1;
3189  sce_count++;
3190  break;
3191 
3192  case TYPE_CPE:
3193  err = decode_cpe(ac, gb, che);
3194  audio_found = 1;
3195  break;
3196 
3197  case TYPE_CCE:
3198  err = decode_cce(ac, gb, che);
3199  break;
3200 
3201  case TYPE_LFE:
3202  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3203  audio_found = 1;
3204  break;
3205 
3206  case TYPE_DSE:
3207  err = skip_data_stream_element(ac, gb);
3208  break;
3209 
3210  case TYPE_PCE: {
3211  uint8_t layout_map[MAX_ELEM_ID*4][3];
3212  int tags;
3213 
3214  int pushed = push_output_configuration(ac);
3215  if (pce_found && !pushed) {
3216  err = AVERROR_INVALIDDATA;
3217  goto fail;
3218  }
3219 
3220  tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb,
3221  payload_alignment);
3222  if (tags < 0) {
3223  err = tags;
3224  break;
3225  }
3226  if (pce_found) {
3227  av_log(avctx, AV_LOG_ERROR,
3228  "Not evaluating a further program_config_element as this construct is dubious at best.\n");
3230  } else {
3231  err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
3232  if (!err)
3233  ac->oc[1].m4ac.chan_config = 0;
3234  pce_found = 1;
3235  }
3236  break;
3237  }
3238 
3239  case TYPE_FIL:
3240  if (elem_id == 15)
3241  elem_id += get_bits(gb, 8) - 1;
3242  if (get_bits_left(gb) < 8 * elem_id) {
3243  av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
3244  err = AVERROR_INVALIDDATA;
3245  goto fail;
3246  }
3247  while (elem_id > 0)
3248  elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type);
3249  err = 0; /* FIXME */
3250  break;
3251 
3252  default:
3253  err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
3254  break;
3255  }
3256 
3257  if (elem_type < TYPE_DSE) {
3258  che_prev = che;
3259  che_prev_type = elem_type;
3260  }
3261 
3262  if (err)
3263  goto fail;
3264 
3265  if (get_bits_left(gb) < 3) {
3266  av_log(avctx, AV_LOG_ERROR, overread_err);
3267  err = AVERROR_INVALIDDATA;
3268  goto fail;
3269  }
3270  }
3271 
3272  if (!avctx->channels) {
3273  *got_frame_ptr = 0;
3274  return 0;
3275  }
3276 
3277  multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
3278  samples <<= multiplier;
3279 
3280  spectral_to_sample(ac, samples);
3281 
3282  if (ac->oc[1].status && audio_found) {
3283  avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
3284  avctx->frame_size = samples;
3285  ac->oc[1].status = OC_LOCKED;
3286  }
3287 
3288  if (multiplier)
3289  avctx->internal->skip_samples_multiplier = 2;
3290 
3291  if (!ac->frame->data[0] && samples) {
3292  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
3293  err = AVERROR_INVALIDDATA;
3294  goto fail;
3295  }
3296 
3297  if (samples) {
3298  ac->frame->nb_samples = samples;
3299  ac->frame->sample_rate = avctx->sample_rate;
3300  } else
3301  av_frame_unref(ac->frame);
3302  *got_frame_ptr = !!samples;
3303 
3304  /* for dual-mono audio (SCE + SCE) */
3305  is_dmono = ac->dmono_mode && sce_count == 2 &&
3307  if (is_dmono) {
3308  if (ac->dmono_mode == 1)
3309  ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
3310  else if (ac->dmono_mode == 2)
3311  ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
3312  }
3313 
3314  return 0;
3315 fail:
3317  return err;
3318 }
3319 
3320 static int aac_decode_frame(AVCodecContext *avctx, void *data,
3321  int *got_frame_ptr, AVPacket *avpkt)
3322 {
3323  AACContext *ac = avctx->priv_data;
3324  const uint8_t *buf = avpkt->data;
3325  int buf_size = avpkt->size;
3326  GetBitContext gb;
3327  int buf_consumed;
3328  int buf_offset;
3329  int err;
3330  int new_extradata_size;
3331  const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
3333  &new_extradata_size);
3334  int jp_dualmono_size;
3335  const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
3337  &jp_dualmono_size);
3338 
3339  if (new_extradata) {
3340  /* discard previous configuration */
3341  ac->oc[1].status = OC_NONE;
3342  err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
3343  new_extradata,
3344  new_extradata_size * 8LL, 1);
3345  if (err < 0) {
3346  return err;
3347  }
3348  }
3349 
3350  ac->dmono_mode = 0;
3351  if (jp_dualmono && jp_dualmono_size > 0)
3352  ac->dmono_mode = 1 + *jp_dualmono;
3353  if (ac->force_dmono_mode >= 0)
3354  ac->dmono_mode = ac->force_dmono_mode;
3355 
3356  if (INT_MAX / 8 <= buf_size)
3357  return AVERROR_INVALIDDATA;
3358 
3359  if ((err = init_get_bits8(&gb, buf, buf_size)) < 0)
3360  return err;
3361 
3362  switch (ac->oc[1].m4ac.object_type) {
3363  case AOT_ER_AAC_LC:
3364  case AOT_ER_AAC_LTP:
3365  case AOT_ER_AAC_LD:
3366  case AOT_ER_AAC_ELD:
3367  err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
3368  break;
3369  default:
3370  err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
3371  }
3372  if (err < 0)
3373  return err;
3374 
3375  buf_consumed = (get_bits_count(&gb) + 7) >> 3;
3376  for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
3377  if (buf[buf_offset])
3378  break;
3379 
3380  return buf_size > buf_offset ? buf_consumed : buf_size;
3381 }
3382 
3384 {
3385  AACContext *ac = avctx->priv_data;
3386  int i, type;
3387 
3388  for (i = 0; i < MAX_ELEM_ID; i++) {
3389  for (type = 0; type < 4; type++) {
3390  if (ac->che[type][i])
3392  av_freep(&ac->che[type][i]);
3393  }
3394  }
3395 
3396  ff_mdct_end(&ac->mdct);
3397  ff_mdct_end(&ac->mdct_small);
3398  ff_mdct_end(&ac->mdct_ld);
3399  ff_mdct_end(&ac->mdct_ltp);
3400 #if !USE_FIXED
3401  ff_mdct15_uninit(&ac->mdct120);
3402  ff_mdct15_uninit(&ac->mdct480);
3403  ff_mdct15_uninit(&ac->mdct960);
3404 #endif
3405  av_freep(&ac->fdsp);
3406  return 0;
3407 }
3408 
3409 static void aacdec_init(AACContext *c)
3410 {
3412  c->apply_ltp = apply_ltp;
3413  c->apply_tns = apply_tns;
3415  c->update_ltp = update_ltp;
3416 #if USE_FIXED
3419 #endif
3420 
3421 #if !USE_FIXED
3422  if(ARCH_MIPS)
3424 #endif /* !USE_FIXED */
3425 }
3426 /**
3427  * AVOptions for Japanese DTV specific extensions (ADTS only)
3428  */
3429 #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
3430 static const AVOption options[] = {
3431  {"dual_mono_mode", "Select the channel to decode for dual mono",
3432  offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
3433  AACDEC_FLAGS, "dual_mono_mode"},
3434 
3435  {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3436  {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3437  {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3438  {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3439 
3440  {NULL},
3441 };
3442 
3443 static const AVClass aac_decoder_class = {
3444  .class_name = "AAC decoder",
3445  .item_name = av_default_item_name,
3446  .option = options,
3447  .version = LIBAVUTIL_VERSION_INT,
3448 };
int predictor_initialized
Definition: aac.h:187
float UINTFLOAT
Definition: aac_defines.h:87
static float * VMUL4S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
Definition: aacdec.c:124
AVFloatDSPContext * fdsp
Definition: aac.h:333
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
Apply AAC-Main style frequency domain prediction.
float, planar
Definition: samplefmt.h:69
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
Conduct IMDCT and windowing.
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
Apply the long term prediction.
#define NULL
Definition: coverity.c:32
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
Definition: aac.h:60
static int decode_pulses(Pulse *pulse, GetBitContext *gb, const uint16_t *swb_offset, int num_swb)
Decode pulse data; reference: table 4.7.
uint8_t use_kb_window[2]
If set, use Kaiser-Bessel window, otherwise use a sine window.
Definition: aac.h:177
float ff_aac_kbd_short_120[120]
Definition: aactab.c:41
INTFLOAT buf_mdct[1024]
Definition: aac.h:316
void(* subband_scale)(int *dst, int *src, int scale, int offset, int len)
Definition: aac.h:371
#define overread_err
This structure describes decoded (raw) audio or video data.
Definition: frame.h:226
uint8_t object_type
Definition: adts_header.h:33
AVOption.
Definition: opt.h:246
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
static void flush(AVCodecContext *avctx)
static const int8_t tags_per_config[16]
Definition: aacdectab.h:38
AVCodecContext * avctx
Definition: aac.h:295
Definition: aac.h:224
void(* mdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:109
static int * DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
Definition: aacdec_fixed.c:125
static AVOnce aac_table_init
float re
Definition: fft.c:82
#define AAC_MUL26(x, y)
Definition: aac_defines.h:100
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:381
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
Generate a Kaiser-Bessel Derived Window.
Definition: kbdwin.c:26
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
#define SCALE_DIFF_ZERO
codebook index corresponding to zero scalefactor indices difference
Definition: aac.h:152
else temp
Definition: vf_mcdeint.c:256
Definition: aac.h:63
static const float cce_scale[]
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:293
const char * g
Definition: vf_curves.c:115
static float * VMUL2S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
Definition: aacdec.c:107
#define AACDEC_FLAGS
AVOptions for Japanese DTV specific extensions (ADTS only)
static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
#define INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
Definition: vlc.h:75
static void aacdec_init(AACContext *ac)
#define FIXR10(x)
Definition: aac_defines.h:93
static int * DEC_SQUAD(int *dst, unsigned idx)
Definition: aacdec_fixed.c:115
static int decode_audio_specific_config_gb(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, GetBitContext *gb, int get_bit_alignment, int sync_extension)
Decode audio specific configuration; reference: table 1.13.
Definition: aac.h:56
Definition: aac.h:57
channels
Definition: aptx.c:30
ChannelElement * che[4][MAX_ELEM_ID]
Definition: aac.h:305
int size
Definition: avcodec.h:1453
const char * b
Definition: vf_curves.c:116
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
float(* scalarproduct_float)(const float *v1, const float *v2, int len)
Calculate the scalar product of two vectors of floats.
Definition: float_dsp.h:175
int av_log2(unsigned v)
Definition: intmath.c:26
INTFLOAT * ret
PCM output.
Definition: aac.h:269
int present
Definition: aac.h:276
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
Update the LTP buffer for next frame.
void(* update_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:369
void(* imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:363
static void vector_pow43(int *coefs, int len)
Definition: aacdec_fixed.c:151
uint64_t channel_layout
Definition: aac.h:128
INTFLOAT sf[120]
scalefactors
Definition: aac.h:255
void(* vector_fmul_reverse)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats, and store the result in a vector of floats...
Definition: float_dsp.h:154
#define AV_EF_BITSTREAM
detect bitstream specification deviations
Definition: avcodec.h:2674
static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx, GetBitContext *gb, int get_bit_alignment, MPEG4AudioConfig *m4ac, int channel_config)
Decode GA "General Audio" specific configuration; reference: table 4.1.
uint8_t ms_mask[128]
Set if mid/side stereo is used for each scalefactor window band.
Definition: aac.h:281
static void apply_independent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply independent channel coupling (applied after IMDCT).
Definition: aacdec.c:246
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:236
static void subband_scale(int *dst, int *src, int scale, int offset, int len)
Definition: aacdec_fixed.c:165
#define MAX_LTP_LONG_SFB
Definition: aac.h:51
#define GET_GAIN(x, y)
Definition: aac_defines.h:98
Dynamic Range Control - decoded from the bitstream but not processed further.
Definition: aac.h:211
static av_cold int che_configure(AACContext *ac, enum ChannelPosition che_pos, int type, int id, int *channels)
Check for the channel element in the current channel position configuration.
static VLC vlc_scalefactors
#define NOISE_PRE
preamble for NOISE_BT, put in bitstream with the first noise band
Definition: aac.h:156
#define FF_PROFILE_AAC_HE_V2
Definition: avcodec.h:2878
static av_always_inline void predict(PredictorState *ps, float *coef, int output_enable)
Definition: aacdec.c:174
enum RawDataBlockType type[8]
Type of channel element to be coupled - SCE or CPE.
Definition: aac.h:237
int profile
profile
Definition: avcodec.h:2869
static int output_configure(AACContext *ac, uint8_t layout_map[MAX_ELEM_ID *4][3], int tags, enum OCStatus oc_type, int get_new_frame)
Configure output channel order based on the current program configuration element.
ChannelPosition
Definition: aac.h:94
static void decode_channel_map(uint8_t layout_map[][3], enum ChannelPosition type, GetBitContext *gb, int n)
Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit...
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:42
static int aac_decode_er_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb)
Spectral data are scaled white noise not coded in the bitstream.
Definition: aac.h:87
Definition: aac.h:58
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
Decode coupling_channel_element; reference: table 4.8.
#define USE_FIXED
Definition: aac_defines.h:25
static av_always_inline int lcg_random(unsigned previous_val)
linear congruential pseudorandom number generator
int band_incr
Number of DRC bands greater than 1 having DRC info.
Definition: aac.h:216
const uint8_t ff_aac_num_swb_128[]
Definition: aactab.c:61
#define AAC_RENAME_32(x)
Definition: aac_defines.h:85
void ff_cbrt_tableinit(void)
Definition: cbrt_tablegen.h:40
int dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
Definition: aac.h:351
const uint16_t * swb_offset
table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular wind...
Definition: aac.h:181
N Error Resilient Long Term Prediction.
Definition: mpeg4audio.h:90
float INTFLOAT
Definition: aac_defines.h:86
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb)
Decode Individual Channel Stream info; reference: table 4.6.
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
Overlap/add with window function.
Definition: float_dsp.h:119
Definition: aac.h:67
BandType
Definition: aac.h:82
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2204
uint8_t
#define FIXR(x)
Definition: aac_defines.h:92
#define av_cold
Definition: attributes.h:82
float ff_aac_kbd_long_960[960]
Definition: aactab.c:40
uint8_t layout_map[MAX_ELEM_ID *4][3]
Definition: aac.h:125
Output configuration under trial specified by an inband PCE.
Definition: aac.h:117
const uint16_t *const ff_swb_offset_480[]
Definition: aactab.c:1360
#define FF_DEBUG_PICT_INFO
Definition: avcodec.h:2622
int warned_960_sbr
Definition: aac.h:358
SingleChannelElement ch[2]
Definition: aac.h:284
const uint16_t *const ff_swb_offset_512[]
Definition: aactab.c:1352
Definition: aac.h:59
const uint8_t ff_tns_max_bands_480[]
Definition: aactab.c:1402
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
TemporalNoiseShaping tns
Definition: aac.h:250
N Error Resilient Low Delay.
Definition: mpeg4audio.h:94
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt, ChannelElement *che, enum RawDataBlockType elem_type)
Decode extension data (incomplete); reference: table 4.51.
const uint8_t ff_aac_scalefactor_bits[121]
Definition: aactab.c:92
CouplingPoint
The point during decoding at which channel coupling is applied.
Definition: aac.h:106
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1641
int num_coupled
number of target elements
Definition: aac.h:236
#define u(width, name, range_min, range_max)
Definition: cbs_h2645.c:253
#define AV_CH_LOW_FREQUENCY
av_cold int ff_mdct15_init(MDCT15Context **ps, int inverse, int N, double scale)
Definition: mdct15.c:247
int exclude_mask[MAX_CHANNELS]
Channels to be excluded from DRC processing.
Definition: aac.h:215
int n_filt[8]
Definition: aac.h:200
FFTContext mdct_ltp
Definition: aac.h:326
void(* vector_pow43)(int *coefs, int len)
Definition: aac.h:370
SingleChannelElement * output_element[MAX_CHANNELS]
Points to each SingleChannelElement.
Definition: aac.h:342
static av_cold int aac_decode_init(AVCodecContext *avctx)
uint8_t * data
Definition: avcodec.h:1452
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
#define AAC_MUL31(x, y)
Definition: aac_defines.h:102
static int count_channels(uint8_t(*layout)[3], int tags)
#define ff_dlog(a,...)
Scalefactor data are intensity stereo positions (in phase).
Definition: aac.h:89
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
static int sample_rate_idx(int rate)
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, GetBitContext *gb, const IndividualChannelStream *ics)
Decode Temporal Noise Shaping data; reference: table 4.48.
#define AV_CH_BACK_LEFT
int id_select[8]
element id
Definition: aac.h:238
ptrdiff_t size
Definition: opengl_enc.c:101
const float *const ff_aac_codebook_vector_vals[]
Definition: aactab.c:1074
static av_always_inline int fixed_sqrt(int x, int bits)
Calculate the square root.
Definition: fixed_dsp.h:176
N Error Resilient Low Complexity.
Definition: mpeg4audio.h:89
ChannelElement * tag_che_map[4][MAX_ELEM_ID]
Definition: aac.h:306
#define AVOnce
Definition: thread.h:159
#define av_log(a,...)
Output configuration set in a global header but not yet locked.
Definition: aac.h:119
static void spectral_to_sample(AACContext *ac, int samples)
Convert spectral data to samples, applying all supported tools as appropriate.
int random_state
Definition: aac.h:335
MDCT15Context * mdct480
Definition: aac.h:331
#define U(x)
Definition: vp56_arith.h:37
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:814
MPEG4AudioConfig m4ac
Definition: aac.h:124
int dyn_rng_sgn[17]
DRC sign information; 0 - positive, 1 - negative.
Definition: aac.h:213
void AAC_RENAME() ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac, INTFLOAT *L, INTFLOAT *R)
Apply one SBR element to one AAC element.
uint32_t ff_cbrt_tab[1<< 13]
static void pop_output_configuration(AACContext *ac)
Restore the previous output configuration if and only if the current configuration is unlocked...
static int decode_fill(AACContext *ac, GetBitContext *gb, int len)
#define UPDATE_CACHE(name, gb)
Definition: get_bits.h:178
PredictorState predictor_state[MAX_PREDICTORS]
Definition: aac.h:268
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
void(* imdct_half)(struct MDCT15Context *s, float *dst, const float *src, ptrdiff_t stride)
Definition: mdct15.h:52
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
const uint8_t ff_aac_num_swb_960[]
Definition: aactab.c:49
static void relative_align_get_bits(GetBitContext *gb, int reference_position)
SpectralBandReplication sbr
Definition: aac.h:287
FFTContext mdct_small
Definition: aac.h:324
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats and store the result in a vector of floats...
Definition: float_dsp.h:38
enum CouplingPoint coupling_point
The point during decoding at which coupling is applied.
Definition: aac.h:235
#define AVERROR(e)
Definition: error.h:43
const uint16_t *const ff_swb_offset_120[]
Definition: aactab.c:1378
uint8_t * av_packet_get_side_data(const AVPacket *pkt, enum AVPacketSideDataType type, int *size)
Get side information from packet.
Definition: avpacket.c:350
const uint8_t ff_aac_num_swb_1024[]
Definition: aactab.c:45
int ff_mpeg4audio_get_config_gb(MPEG4AudioConfig *c, GetBitContext *gb, int sync_extension)
Parse MPEG-4 systems extradata from a potentially unaligned GetBitContext to retrieve audio configura...
Definition: mpeg4audio.c:86
static void imdct_and_windowing_960(AACContext *ac, SingleChannelElement *sce)
Conduct IMDCT and windowing.
void(* butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
Definition: float_dsp.h:164
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
float ff_aac_kbd_long_1024[1024]
Definition: aactab.c:38
INTFLOAT temp[128]
Definition: aac.h:354
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1620
static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx, GetBitContext *gb, MPEG4AudioConfig *m4ac, int channel_config)
uint8_t sampling_index
Definition: adts_header.h:34
int amp[4]
Definition: aac.h:228
static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID], uint8_t(*layout_map)[3], int offset, uint64_t left, uint64_t right, int pos)
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:175
#define ff_mdct_init
Definition: fft.h:169
const float ff_aac_eld_window_512[1920]
Definition: aactab.c:1411
Definition: aac.h:62
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
GLsizei count
Definition: opengl_enc.c:109
#define CLOSE_READER(name, gb)
Definition: get_bits.h:149
int num_swb
number of scalefactor window bands
Definition: aac.h:183
static int count_paired_channels(uint8_t(*layout_map)[3], int tags, int pos, int *current)
#define FFMAX(a, b)
Definition: common.h:94
#define fail()
Definition: checkasm.h:117
int prog_ref_level
A reference level for the long-term program audio level for all channels combined.
Definition: aac.h:219
Output configuration locked in place.
Definition: aac.h:120
Predictor State.
Definition: aac.h:135
uint8_t chan_config
Definition: adts_header.h:35
Definition: vlc.h:26
float ff_aac_pow2sf_tab[428]
Definition: aactab.c:35
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2247
#define SKIP_BITS(name, gb, num)
Definition: get_bits.h:193
#define AAC_RENAME(x)
Definition: aac_defines.h:84
int warned_remapping_once
Definition: aac.h:308
INTFLOAT ret_buf[2048]
PCM output buffer.
Definition: aac.h:264
N Error Resilient Scalable.
Definition: mpeg4audio.h:91
static SDL_Window * window
Definition: ffplay.c:365
static void reset_predictor_group(PredictorState *ps, int group_num)
static ChannelElement * get_che(AACContext *ac, int type, int elem_id)
enum WindowSequence window_sequence[2]
Definition: aac.h:176
INTFLOAT ltp_state[3072]
time signal for LTP
Definition: aac.h:265
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:895
const uint8_t ff_aac_num_swb_512[]
Definition: aactab.c:53
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
Definition: avcodec.h:2665
static int aac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
int predictor_reset_group
Definition: aac.h:188
static int frame_configure_elements(AVCodecContext *avctx)
#define FFMIN(a, b)
Definition: common.h:96
int dyn_rng_ctl[17]
DRC magnitude information.
Definition: aac.h:214
signed 32 bits, planar
Definition: samplefmt.h:68
static const INTFLOAT ltp_coef[8]
Definition: aactab.h:94
uint8_t w
Definition: llviddspenc.c:38
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, int ms_present)
Decode Mid/Side data; reference: table 4.54.
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
intensity stereo decoding; reference: 4.6.8.2.3
uint8_t num_aac_frames
Definition: adts_header.h:36
int pos[4]
Definition: aac.h:227
MDCT15Context * mdct120
Definition: aac.h:330
Y Main.
Definition: mpeg4audio.h:75
int32_t
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
Definition: get_bits.h:443
FFTContext mdct_ld
Definition: aac.h:325
#define s(width, name)
Definition: cbs_vp9.c:257
void ff_aacdec_init_mips(AACContext *c)
Definition: aacdec_mips.c:433
int AAC_RENAME() ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr, GetBitContext *gb, int crc, int cnt, int id_aac)
Decode one SBR element.
#define LAST_SKIP_BITS(name, gb, num)
Definition: get_bits.h:199
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
Definition: get_bits.h:762
int length[8][4]
Definition: aac.h:201
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
Definition: float_dsp.h:85
static av_cold void aac_static_table_init(void)
void AAC_RENAME() ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
Close one SBR context.
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc, enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point, void(*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
channel coupling transformation interface
#define AV_CH_FRONT_LEFT_OF_CENTER
#define AV_EF_EXPLODE
abort decoding on minor error detection
Definition: avcodec.h:2676
int n
Definition: avisynth_c.h:684
const uint8_t ff_tns_max_bands_1024[]
Definition: aactab.c:1394
#define GET_VLC(code, name, gb, table, bits, max_depth)
If the vlc code is invalid and max_depth=1, then no bits will be removed.
Definition: get_bits.h:671
static int AAC_RENAME() compute_lpc_coefs(const LPC_TYPE *autoc, int max_order, LPC_TYPE *lpc, int lpc_stride, int fail, int normalize)
Levinson-Durbin recursion.
Definition: lpc.h:166
#define AV_CH_FRONT_CENTER
static void decode_ltp(LongTermPrediction *ltp, GetBitContext *gb, uint8_t max_sfb)
Decode Long Term Prediction data; reference: table 4.xx.
static int aac_decode_frame_int(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
static void apply_dependent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply dependent channel coupling (applied before IMDCT).
Definition: aacdec.c:210
static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024], GetBitContext *gb, const INTFLOAT sf[120], int pulse_present, const Pulse *pulse, const IndividualChannelStream *ics, enum BandType band_type[120])
Decode spectral data; reference: table 4.50.
void AAC_RENAME() ff_aac_sbr_init(void)
Initialize SBR.
int pce_instance_tag
Indicates with which program the DRC info is associated.
Definition: aac.h:212
N (code in SoC repo) Scalable Sample Rate.
Definition: mpeg4audio.h:77
static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out, INTFLOAT *in, IndividualChannelStream *ics)
Apply windowing and MDCT to obtain the spectral coefficient from the predicted sample by LTP...
N Scalable.
Definition: mpeg4audio.h:80
static void error(const char *err)
static const INTFLOAT *const tns_tmp2_map[4]
Definition: aactab.h:126
#define SHOW_UBITS(name, gb, num)
Definition: get_bits.h:211
static int push_output_configuration(AACContext *ac)
Save current output configuration if and only if it has been locked.
#define FF_ARRAY_ELEMS(a)
#define AV_CH_FRONT_RIGHT_OF_CENTER
int interpolation_scheme
Indicates the interpolation scheme used in the SBR QMF domain.
Definition: aac.h:217
coupling parameters
Definition: aac.h:234
int tags_mapped
Definition: aac.h:307
static void reset_all_predictors(PredictorState *ps)
MDCT15Context * mdct960
Definition: aac.h:332
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
Skip data_stream_element; reference: table 4.10.
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
int ff_adts_header_parse(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
Parse the ADTS frame header to the end of the variable header, which is the first 54 bits...
Definition: adts_header.c:30
int ch_select[8]
[0] shared list of gains; [1] list of gains for right channel; [2] list of gains for left channel; [3...
Definition: aac.h:239
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2216
int force_dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
Definition: aac.h:350
The AV_PKT_DATA_NEW_EXTRADATA is used to notify the codec or the format that the extradata buffer was...
Definition: avcodec.h:1174
int order[8][4]
Definition: aac.h:203
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
#define AV_ONCE_INIT
Definition: thread.h:160
int warned_num_aac_frames
Definition: aac.h:357
#define AAC_INIT_VLC_STATIC(num, size)
Temporal Noise Shaping.
Definition: aac.h:198
int sample_rate
samples per second
Definition: avcodec.h:2196
float ff_aac_kbd_short_128[128]
Definition: aactab.c:39
void AAC_RENAME() ff_sine_window_init(INTFLOAT *window, int n)
Generate a sine window.
static const AVOption options[]
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:650
#define AV_CH_LAYOUT_NATIVE
Channel mask value used for AVCodecContext.request_channel_layout to indicate that the user requests ...
static int decode_scalefactors(AACContext *ac, INTFLOAT sf[120], GetBitContext *gb, unsigned int global_gain, IndividualChannelStream *ics, enum BandType band_type[120], int band_type_run_end[120])
Decode scalefactors; reference: table 4.47.
int debug
debug
Definition: avcodec.h:2621
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
Decode a channel_pair_element; reference: table 4.4.
Long Term Prediction.
Definition: aac.h:163
static void apply_tns(INTFLOAT coef_param[1024], TemporalNoiseShaping *tns, IndividualChannelStream *ics, int decode)
Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4...
main external API structure.
Definition: avcodec.h:1540
#define AV_CH_FRONT_LEFT
int skip_samples_multiplier
Definition: internal.h:217
#define NOISE_PRE_BITS
length of preamble
Definition: aac.h:157
#define OPEN_READER(name, gb)
Definition: get_bits.h:138
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1919
IndividualChannelStream ics
Definition: aac.h:249
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
void * buf
Definition: avisynth_c.h:690
#define MAX_PREDICTORS
Definition: aac.h:146
static av_always_inline float cbrtf(float x)
Definition: libm.h:61
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:108
GLint GLenum type
Definition: opengl_enc.c:105
int extradata_size
Definition: avcodec.h:1642
void AAC_RENAME() ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr, int id_aac)
Initialize one SBR context.
uint8_t group_len[8]
Definition: aac.h:179
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:487
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:523
void(* apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns, IndividualChannelStream *ics, int decode)
Definition: aac.h:365
#define MAX_ELEM_ID
Definition: aac.h:48
Describe the class of an AVClass context structure.
Definition: log.h:67
int sample_rate
Sample rate of the audio data.
Definition: frame.h:398
static av_cold int aac_decode_close(AVCodecContext *avctx)
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:460
static int decode_audio_specific_config(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, const uint8_t *data, int64_t bit_size, int sync_extension)
#define AAC_MUL30(x, y)
Definition: aac_defines.h:101
static uint64_t sniff_channel_order(uint8_t(*layout_map)[3], int tags)
const uint16_t *const ff_swb_offset_960[]
Definition: aactab.c:1344
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext *gb)
Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4...
int index
Definition: gxfenc.c:89
static void noise_scale(int *coefs, int scale, int band_energy, int len)
Definition: aacdec_fixed.c:196
static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac, uint8_t(*layout_map)[3], GetBitContext *gb, int byte_align_ref)
Decode program configuration element; reference: table 4.2.
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:615
#define GET_CACHE(name, gb)
Definition: get_bits.h:215
static float * VMUL2(float *dst, const float *v, unsigned idx, const float *scale)
Definition: aacdec.c:83
OCStatus
Output configuration status.
Definition: aac.h:115
int skip_samples
Number of audio samples to skip at the start of the next decoded frame.
Definition: internal.h:185
#define MAX_CHANNELS
Definition: aac.h:47
N Error Resilient Bit-Sliced Arithmetic Coding.
Definition: mpeg4audio.h:93
#define TNS_MAX_ORDER
Definition: aac.h:50
#define FF_COMPLIANCE_STRICT
Strictly conform to all the things in the spec no matter what consequences.
Definition: avcodec.h:2601
main AAC context
Definition: aac.h:293
const uint32_t ff_aac_scalefactor_code[121]
Definition: aactab.c:73
LongTermPrediction ltp
Definition: aac.h:180
ChannelCoupling coup
Definition: aac.h:286
Output configuration under trial specified by a frame header.
Definition: aac.h:118
int frame_length_short
Definition: mpeg4audio.h:45
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb)
const uint8_t ff_tns_max_bands_128[]
Definition: aactab.c:1406
#define NOISE_OFFSET
subtracted from global gain, used as offset for the preamble
Definition: aac.h:158
static void imdct_and_window(TwinVQContext *tctx, enum TwinVQFrameType ftype, int wtype, float *in, float *prev, int ch)
Definition: twinvq.c:327
void av_frame_unref(AVFrame *frame)
Unreference all the buffers referenced by frame and reset the frame fields.
Definition: frame.c:553
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
static const int8_t filt[NUMTAPS]
Definition: af_earwax.c:39
int band_type_run_end[120]
band type run end points
Definition: aac.h:254
static int decode_band_types(AACContext *ac, enum BandType band_type[120], int band_type_run_end[120], GetBitContext *gb, IndividualChannelStream *ics)
Decode band types (section_data payload); reference: table 4.46.
#define AV_CH_BACK_CENTER
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:240
int band_top[17]
Indicates the top of the i-th DRC band in units of 4 spectral lines.
Definition: aac.h:218
#define AV_CH_SIDE_RIGHT
INTFLOAT coeffs[1024]
coefficients for IMDCT, maybe processed
Definition: aac.h:262
AVFixedDSPContext * avpriv_alloc_fixed_dsp(int bit_exact)
Allocate and initialize a fixed DSP context.
Definition: fixed_dsp.c:148
static VLC vlc_spectral[11]
enum OCStatus status
Definition: aac.h:129
INTFLOAT gain[16][120]
Definition: aac.h:242
Scalefactor data are intensity stereo positions (out of phase).
Definition: aac.h:88
N Error Resilient Enhanced Low Delay.
Definition: mpeg4audio.h:110
static int set_default_channel_config(AVCodecContext *avctx, uint8_t(*layout_map)[3], int *tags, int channel_config)
Set up channel positions based on a default channel configuration as specified in table 1...
#define M_SQRT2
Definition: mathematics.h:61
#define RANGE15(x)
Definition: aac_defines.h:97
INTFLOAT coef[8][4][TNS_MAX_ORDER]
Definition: aac.h:205
int16_t lag
Definition: aac.h:165
const uint8_t ff_aac_num_swb_120[]
Definition: aactab.c:65
DynamicRangeControl che_drc
Definition: aac.h:299
static av_always_inline void reset_predict_state(PredictorState *ps)
Definition: aacdec.c:72
AVFrame * frame
Definition: aac.h:296
OutputConfiguration oc[2]
Definition: aac.h:356
An AV_PKT_DATA_JP_DUALMONO side data packet indicates that the packet may contain "dual mono" audio s...
Definition: avcodec.h:1285
int
const uint8_t ff_aac_pred_sfb_max[]
Definition: aactab.c:69
int direction[8][4]
Definition: aac.h:202
void(* apply_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:364
uint8_t prediction_used[41]
Definition: aac.h:190
const float ff_aac_eld_window_480[1800]
Definition: aactab.c:2378
INTFLOAT saved[1536]
overlap
Definition: aac.h:263
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:248
#define ff_mdct_end
Definition: fft.h:170
const uint8_t ff_aac_num_swb_480[]
Definition: aactab.c:57
static double c[64]
const uint16_t *const ff_swb_offset_1024[]
Definition: aactab.c:1336
unsigned AAC_SIGNE
Definition: aac_defines.h:91
void(* windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out, INTFLOAT *in, IndividualChannelStream *ics)
Definition: aac.h:367
Definition: aac.h:61
Individual Channel Stream.
Definition: aac.h:174
INTFLOAT coef
Definition: aac.h:167
#define AVERROR_UNKNOWN
Unknown error, typically from an external library.
Definition: error.h:71
const uint16_t *const ff_aac_codebook_vector_idx[]
Definition: aactab.c:1083
static void ff_aac_tableinit(void)
Definition: aactab.h:45
av_cold void ff_mdct15_uninit(MDCT15Context **ps)
Definition: mdct15.c:43
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:275
static void decode_gain_control(SingleChannelElement *sce, GetBitContext *gb)
void * priv_data
Definition: avcodec.h:1567
int warned_gain_control
Definition: aac.h:360
static int decode_ics(AACContext *ac, SingleChannelElement *sce, GetBitContext *gb, int common_window, int scale_flag)
Decode an individual_channel_stream payload; reference: table 4.44.
#define FF_DEBUG_STARTCODE
Definition: avcodec.h:2635
const uint8_t ff_tns_max_bands_512[]
Definition: aactab.c:1398
int len
Scalefactors and spectral data are all zero.
Definition: aac.h:83
int channels
number of audio channels
Definition: avcodec.h:2197
int num_pulse
Definition: aac.h:225
static int * DEC_SPAIR(int *dst, unsigned idx)
Definition: aacdec_fixed.c:107
struct AVCodecInternal * internal
Private context used for internal data.
Definition: avcodec.h:1575
const uint8_t ff_mpeg4audio_channels[8]
Definition: mpeg4audio.c:67
static int ff_thread_once(char *control, void(*routine)(void))
Definition: thread.h:162
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
Y Long Term Prediction.
Definition: mpeg4audio.h:78
uint8_t crc_absent
Definition: adts_header.h:32
static const uint8_t * align_get_bits(GetBitContext *s)
Definition: get_bits.h:658
uint64_t layout
#define FF_PROFILE_AAC_HE
Definition: avcodec.h:2877
enum BandType band_type[128]
band types
Definition: aac.h:252
static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext *gb)
Decode dynamic range information; reference: table 4.52.
#define AV_CH_FRONT_RIGHT
#define POW_SF2_ZERO
ff_aac_pow2sf_tab index corresponding to pow(2, 0);
Definition: aac.h:154
static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
FILE * out
Definition: movenc.c:54
FFTContext mdct
Definition: aac.h:323
int sbr
-1 implicit, 1 presence
Definition: mpeg4audio.h:38
#define av_freep(p)
void INT64 start
Definition: avisynth_c.h:690
#define av_always_inline
Definition: attributes.h:39
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
Mid/Side stereo decoding; reference: 4.6.8.1.3.
#define VLC_TYPE
Definition: vlc.h:24
#define AV_CH_SIDE_LEFT
#define FFSWAP(type, a, b)
Definition: common.h:99
int ps
-1 implicit, 1 presence
Definition: mpeg4audio.h:44
int8_t used[MAX_LTP_LONG_SFB]
Definition: aac.h:168
const uint16_t *const ff_swb_offset_128[]
Definition: aactab.c:1368
int8_t present
Definition: aac.h:164
uint32_t sample_rate
Definition: adts_header.h:29
static const AVClass aac_decoder_class
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:273
uint64_t request_channel_layout
Request decoder to use this channel layout if it can (0 for default)
Definition: avcodec.h:2254
int layout_map_tags
Definition: aac.h:126
enum AVCodecID id
This structure stores compressed data.
Definition: avcodec.h:1429
mode
Use these values in ebur128_init (or'ed).
Definition: ebur128.h:83
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:292
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:2599
void AAC_RENAME() ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
static int * DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
Definition: aacdec_fixed.c:133
#define AV_CH_BACK_RIGHT
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
Y Low Complexity.
Definition: mpeg4audio.h:76
static float * VMUL4(float *dst, const float *v, unsigned idx, const float *scale)
Definition: aacdec.c:94
Output unconfigured.
Definition: aac.h:116
static const uint8_t aac_channel_layout_map[16][5][3]
Definition: aacdectab.h:40
RawDataBlockType
Definition: aac.h:55
static uint8_t tmp[11]
Definition: aes_ctr.c:26