FFmpeg  4.2.1
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 
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--) {
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  INTFLOAT band_energy;
1677 #if USE_FIXED
1678  for (k = 0; k < off_len; k++) {
1680  cfo[k] = ac->random_state >> 3;
1681  }
1682 
1683  band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
1684  band_energy = fixed_sqrt(band_energy, 31);
1685  noise_scale(cfo, sf[idx], band_energy, off_len);
1686 #else
1687  float scale;
1688 
1689  for (k = 0; k < off_len; k++) {
1691  cfo[k] = ac->random_state;
1692  }
1693 
1694  band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
1695  scale = sf[idx] / sqrtf(band_energy);
1696  ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
1697 #endif /* USE_FIXED */
1698  }
1699  } else {
1700 #if !USE_FIXED
1701  const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1702 #endif /* !USE_FIXED */
1703  const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1704  VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1705  OPEN_READER(re, gb);
1706 
1707  switch (cbt_m1 >> 1) {
1708  case 0:
1709  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1710  INTFLOAT *cf = cfo;
1711  int len = off_len;
1712 
1713  do {
1714  int code;
1715  unsigned cb_idx;
1716 
1717  UPDATE_CACHE(re, gb);
1718  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1719  cb_idx = cb_vector_idx[code];
1720 #if USE_FIXED
1721  cf = DEC_SQUAD(cf, cb_idx);
1722 #else
1723  cf = VMUL4(cf, vq, cb_idx, sf + idx);
1724 #endif /* USE_FIXED */
1725  } while (len -= 4);
1726  }
1727  break;
1728 
1729  case 1:
1730  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1731  INTFLOAT *cf = cfo;
1732  int len = off_len;
1733 
1734  do {
1735  int code;
1736  unsigned nnz;
1737  unsigned cb_idx;
1738  uint32_t bits;
1739 
1740  UPDATE_CACHE(re, gb);
1741  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1742  cb_idx = cb_vector_idx[code];
1743  nnz = cb_idx >> 8 & 15;
1744  bits = nnz ? GET_CACHE(re, gb) : 0;
1745  LAST_SKIP_BITS(re, gb, nnz);
1746 #if USE_FIXED
1747  cf = DEC_UQUAD(cf, cb_idx, bits);
1748 #else
1749  cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1750 #endif /* USE_FIXED */
1751  } while (len -= 4);
1752  }
1753  break;
1754 
1755  case 2:
1756  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1757  INTFLOAT *cf = cfo;
1758  int len = off_len;
1759 
1760  do {
1761  int code;
1762  unsigned cb_idx;
1763 
1764  UPDATE_CACHE(re, gb);
1765  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1766  cb_idx = cb_vector_idx[code];
1767 #if USE_FIXED
1768  cf = DEC_SPAIR(cf, cb_idx);
1769 #else
1770  cf = VMUL2(cf, vq, cb_idx, sf + idx);
1771 #endif /* USE_FIXED */
1772  } while (len -= 2);
1773  }
1774  break;
1775 
1776  case 3:
1777  case 4:
1778  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1779  INTFLOAT *cf = cfo;
1780  int len = off_len;
1781 
1782  do {
1783  int code;
1784  unsigned nnz;
1785  unsigned cb_idx;
1786  unsigned sign;
1787 
1788  UPDATE_CACHE(re, gb);
1789  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1790  cb_idx = cb_vector_idx[code];
1791  nnz = cb_idx >> 8 & 15;
1792  sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
1793  LAST_SKIP_BITS(re, gb, nnz);
1794 #if USE_FIXED
1795  cf = DEC_UPAIR(cf, cb_idx, sign);
1796 #else
1797  cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1798 #endif /* USE_FIXED */
1799  } while (len -= 2);
1800  }
1801  break;
1802 
1803  default:
1804  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1805 #if USE_FIXED
1806  int *icf = cfo;
1807  int v;
1808 #else
1809  float *cf = cfo;
1810  uint32_t *icf = (uint32_t *) cf;
1811 #endif /* USE_FIXED */
1812  int len = off_len;
1813 
1814  do {
1815  int code;
1816  unsigned nzt, nnz;
1817  unsigned cb_idx;
1818  uint32_t bits;
1819  int j;
1820 
1821  UPDATE_CACHE(re, gb);
1822  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1823 
1824  if (!code) {
1825  *icf++ = 0;
1826  *icf++ = 0;
1827  continue;
1828  }
1829 
1830  cb_idx = cb_vector_idx[code];
1831  nnz = cb_idx >> 12;
1832  nzt = cb_idx >> 8;
1833  bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1834  LAST_SKIP_BITS(re, gb, nnz);
1835 
1836  for (j = 0; j < 2; j++) {
1837  if (nzt & 1<<j) {
1838  uint32_t b;
1839  int n;
1840  /* The total length of escape_sequence must be < 22 bits according
1841  to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1842  UPDATE_CACHE(re, gb);
1843  b = GET_CACHE(re, gb);
1844  b = 31 - av_log2(~b);
1845 
1846  if (b > 8) {
1847  av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1848  return AVERROR_INVALIDDATA;
1849  }
1850 
1851  SKIP_BITS(re, gb, b + 1);
1852  b += 4;
1853  n = (1 << b) + SHOW_UBITS(re, gb, b);
1854  LAST_SKIP_BITS(re, gb, b);
1855 #if USE_FIXED
1856  v = n;
1857  if (bits & 1U<<31)
1858  v = -v;
1859  *icf++ = v;
1860 #else
1861  *icf++ = ff_cbrt_tab[n] | (bits & 1U<<31);
1862 #endif /* USE_FIXED */
1863  bits <<= 1;
1864  } else {
1865 #if USE_FIXED
1866  v = cb_idx & 15;
1867  if (bits & 1U<<31)
1868  v = -v;
1869  *icf++ = v;
1870 #else
1871  unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1872  *icf++ = (bits & 1U<<31) | v;
1873 #endif /* USE_FIXED */
1874  bits <<= !!v;
1875  }
1876  cb_idx >>= 4;
1877  }
1878  } while (len -= 2);
1879 #if !USE_FIXED
1880  ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1881 #endif /* !USE_FIXED */
1882  }
1883  }
1884 
1885  CLOSE_READER(re, gb);
1886  }
1887  }
1888  coef += g_len << 7;
1889  }
1890 
1891  if (pulse_present) {
1892  idx = 0;
1893  for (i = 0; i < pulse->num_pulse; i++) {
1894  INTFLOAT co = coef_base[ pulse->pos[i] ];
1895  while (offsets[idx + 1] <= pulse->pos[i])
1896  idx++;
1897  if (band_type[idx] != NOISE_BT && sf[idx]) {
1898  INTFLOAT ico = -pulse->amp[i];
1899 #if USE_FIXED
1900  if (co) {
1901  ico = co + (co > 0 ? -ico : ico);
1902  }
1903  coef_base[ pulse->pos[i] ] = ico;
1904 #else
1905  if (co) {
1906  co /= sf[idx];
1907  ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1908  }
1909  coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1910 #endif /* USE_FIXED */
1911  }
1912  }
1913  }
1914 #if USE_FIXED
1915  coef = coef_base;
1916  idx = 0;
1917  for (g = 0; g < ics->num_window_groups; g++) {
1918  unsigned g_len = ics->group_len[g];
1919 
1920  for (i = 0; i < ics->max_sfb; i++, idx++) {
1921  const unsigned cbt_m1 = band_type[idx] - 1;
1922  int *cfo = coef + offsets[i];
1923  int off_len = offsets[i + 1] - offsets[i];
1924  int group;
1925 
1926  if (cbt_m1 < NOISE_BT - 1) {
1927  for (group = 0; group < (int)g_len; group++, cfo+=128) {
1928  ac->vector_pow43(cfo, off_len);
1929  ac->subband_scale(cfo, cfo, sf[idx], 34, off_len, ac->avctx);
1930  }
1931  }
1932  }
1933  coef += g_len << 7;
1934  }
1935 #endif /* USE_FIXED */
1936  return 0;
1937 }
1938 
1939 /**
1940  * Apply AAC-Main style frequency domain prediction.
1941  */
1943 {
1944  int sfb, k;
1945 
1946  if (!sce->ics.predictor_initialized) {
1948  sce->ics.predictor_initialized = 1;
1949  }
1950 
1951  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1952  for (sfb = 0;
1953  sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
1954  sfb++) {
1955  for (k = sce->ics.swb_offset[sfb];
1956  k < sce->ics.swb_offset[sfb + 1];
1957  k++) {
1958  predict(&sce->predictor_state[k], &sce->coeffs[k],
1959  sce->ics.predictor_present &&
1960  sce->ics.prediction_used[sfb]);
1961  }
1962  }
1963  if (sce->ics.predictor_reset_group)
1965  sce->ics.predictor_reset_group);
1966  } else
1968 }
1969 
1971 {
1972  // wd_num, wd_test, aloc_size
1973  static const uint8_t gain_mode[4][3] = {
1974  {1, 0, 5}, // ONLY_LONG_SEQUENCE = 0,
1975  {2, 1, 2}, // LONG_START_SEQUENCE,
1976  {8, 0, 2}, // EIGHT_SHORT_SEQUENCE,
1977  {2, 1, 5}, // LONG_STOP_SEQUENCE
1978  };
1979 
1980  const int mode = sce->ics.window_sequence[0];
1981  uint8_t bd, wd, ad;
1982 
1983  // FIXME: Store the gain control data on |sce| and do something with it.
1984  uint8_t max_band = get_bits(gb, 2);
1985  for (bd = 0; bd < max_band; bd++) {
1986  for (wd = 0; wd < gain_mode[mode][0]; wd++) {
1987  uint8_t adjust_num = get_bits(gb, 3);
1988  for (ad = 0; ad < adjust_num; ad++) {
1989  skip_bits(gb, 4 + ((wd == 0 && gain_mode[mode][1])
1990  ? 4
1991  : gain_mode[mode][2]));
1992  }
1993  }
1994  }
1995 }
1996 
1997 /**
1998  * Decode an individual_channel_stream payload; reference: table 4.44.
1999  *
2000  * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
2001  * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
2002  *
2003  * @return Returns error status. 0 - OK, !0 - error
2004  */
2006  GetBitContext *gb, int common_window, int scale_flag)
2007 {
2008  Pulse pulse;
2009  TemporalNoiseShaping *tns = &sce->tns;
2010  IndividualChannelStream *ics = &sce->ics;
2011  INTFLOAT *out = sce->coeffs;
2012  int global_gain, eld_syntax, er_syntax, pulse_present = 0;
2013  int ret;
2014 
2015  eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2016  er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
2017  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
2018  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
2019  ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2020 
2021  /* This assignment is to silence a GCC warning about the variable being used
2022  * uninitialized when in fact it always is.
2023  */
2024  pulse.num_pulse = 0;
2025 
2026  global_gain = get_bits(gb, 8);
2027 
2028  if (!common_window && !scale_flag) {
2029  ret = decode_ics_info(ac, ics, gb);
2030  if (ret < 0)
2031  goto fail;
2032  }
2033 
2034  if ((ret = decode_band_types(ac, sce->band_type,
2035  sce->band_type_run_end, gb, ics)) < 0)
2036  goto fail;
2037  if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
2038  sce->band_type, sce->band_type_run_end)) < 0)
2039  goto fail;
2040 
2041  pulse_present = 0;
2042  if (!scale_flag) {
2043  if (!eld_syntax && (pulse_present = get_bits1(gb))) {
2044  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2045  av_log(ac->avctx, AV_LOG_ERROR,
2046  "Pulse tool not allowed in eight short sequence.\n");
2047  ret = AVERROR_INVALIDDATA;
2048  goto fail;
2049  }
2050  if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
2051  av_log(ac->avctx, AV_LOG_ERROR,
2052  "Pulse data corrupt or invalid.\n");
2053  ret = AVERROR_INVALIDDATA;
2054  goto fail;
2055  }
2056  }
2057  tns->present = get_bits1(gb);
2058  if (tns->present && !er_syntax) {
2059  ret = decode_tns(ac, tns, gb, ics);
2060  if (ret < 0)
2061  goto fail;
2062  }
2063  if (!eld_syntax && get_bits1(gb)) {
2064  decode_gain_control(sce, gb);
2065  if (!ac->warned_gain_control) {
2066  avpriv_report_missing_feature(ac->avctx, "Gain control");
2067  ac->warned_gain_control = 1;
2068  }
2069  }
2070  // I see no textual basis in the spec for this occurring after SSR gain
2071  // control, but this is what both reference and real implmentations do
2072  if (tns->present && er_syntax) {
2073  ret = decode_tns(ac, tns, gb, ics);
2074  if (ret < 0)
2075  goto fail;
2076  }
2077  }
2078 
2079  ret = decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
2080  &pulse, ics, sce->band_type);
2081  if (ret < 0)
2082  goto fail;
2083 
2084  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
2085  apply_prediction(ac, sce);
2086 
2087  return 0;
2088 fail:
2089  tns->present = 0;
2090  return ret;
2091 }
2092 
2093 /**
2094  * Mid/Side stereo decoding; reference: 4.6.8.1.3.
2095  */
2097 {
2098  const IndividualChannelStream *ics = &cpe->ch[0].ics;
2099  INTFLOAT *ch0 = cpe->ch[0].coeffs;
2100  INTFLOAT *ch1 = cpe->ch[1].coeffs;
2101  int g, i, group, idx = 0;
2102  const uint16_t *offsets = ics->swb_offset;
2103  for (g = 0; g < ics->num_window_groups; g++) {
2104  for (i = 0; i < ics->max_sfb; i++, idx++) {
2105  if (cpe->ms_mask[idx] &&
2106  cpe->ch[0].band_type[idx] < NOISE_BT &&
2107  cpe->ch[1].band_type[idx] < NOISE_BT) {
2108 #if USE_FIXED
2109  for (group = 0; group < ics->group_len[g]; group++) {
2110  ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
2111  ch1 + group * 128 + offsets[i],
2112  offsets[i+1] - offsets[i]);
2113 #else
2114  for (group = 0; group < ics->group_len[g]; group++) {
2115  ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
2116  ch1 + group * 128 + offsets[i],
2117  offsets[i+1] - offsets[i]);
2118 #endif /* USE_FIXED */
2119  }
2120  }
2121  }
2122  ch0 += ics->group_len[g] * 128;
2123  ch1 += ics->group_len[g] * 128;
2124  }
2125 }
2126 
2127 /**
2128  * intensity stereo decoding; reference: 4.6.8.2.3
2129  *
2130  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
2131  * [1] mask is decoded from bitstream; [2] mask is all 1s;
2132  * [3] reserved for scalable AAC
2133  */
2135  ChannelElement *cpe, int ms_present)
2136 {
2137  const IndividualChannelStream *ics = &cpe->ch[1].ics;
2138  SingleChannelElement *sce1 = &cpe->ch[1];
2139  INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
2140  const uint16_t *offsets = ics->swb_offset;
2141  int g, group, i, idx = 0;
2142  int c;
2143  INTFLOAT scale;
2144  for (g = 0; g < ics->num_window_groups; g++) {
2145  for (i = 0; i < ics->max_sfb;) {
2146  if (sce1->band_type[idx] == INTENSITY_BT ||
2147  sce1->band_type[idx] == INTENSITY_BT2) {
2148  const int bt_run_end = sce1->band_type_run_end[idx];
2149  for (; i < bt_run_end; i++, idx++) {
2150  c = -1 + 2 * (sce1->band_type[idx] - 14);
2151  if (ms_present)
2152  c *= 1 - 2 * cpe->ms_mask[idx];
2153  scale = c * sce1->sf[idx];
2154  for (group = 0; group < ics->group_len[g]; group++)
2155 #if USE_FIXED
2156  ac->subband_scale(coef1 + group * 128 + offsets[i],
2157  coef0 + group * 128 + offsets[i],
2158  scale,
2159  23,
2160  offsets[i + 1] - offsets[i] ,ac->avctx);
2161 #else
2162  ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
2163  coef0 + group * 128 + offsets[i],
2164  scale,
2165  offsets[i + 1] - offsets[i]);
2166 #endif /* USE_FIXED */
2167  }
2168  } else {
2169  int bt_run_end = sce1->band_type_run_end[idx];
2170  idx += bt_run_end - i;
2171  i = bt_run_end;
2172  }
2173  }
2174  coef0 += ics->group_len[g] * 128;
2175  coef1 += ics->group_len[g] * 128;
2176  }
2177 }
2178 
2179 /**
2180  * Decode a channel_pair_element; reference: table 4.4.
2181  *
2182  * @return Returns error status. 0 - OK, !0 - error
2183  */
2185 {
2186  int i, ret, common_window, ms_present = 0;
2187  int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2188 
2189  common_window = eld_syntax || get_bits1(gb);
2190  if (common_window) {
2191  if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
2192  return AVERROR_INVALIDDATA;
2193  i = cpe->ch[1].ics.use_kb_window[0];
2194  cpe->ch[1].ics = cpe->ch[0].ics;
2195  cpe->ch[1].ics.use_kb_window[1] = i;
2196  if (cpe->ch[1].ics.predictor_present &&
2197  (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
2198  if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
2199  decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
2200  ms_present = get_bits(gb, 2);
2201  if (ms_present == 3) {
2202  av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
2203  return AVERROR_INVALIDDATA;
2204  } else if (ms_present)
2205  decode_mid_side_stereo(cpe, gb, ms_present);
2206  }
2207  if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
2208  return ret;
2209  if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
2210  return ret;
2211 
2212  if (common_window) {
2213  if (ms_present)
2214  apply_mid_side_stereo(ac, cpe);
2215  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
2216  apply_prediction(ac, &cpe->ch[0]);
2217  apply_prediction(ac, &cpe->ch[1]);
2218  }
2219  }
2220 
2221  apply_intensity_stereo(ac, cpe, ms_present);
2222  return 0;
2223 }
2224 
2225 static const float cce_scale[] = {
2226  1.09050773266525765921, //2^(1/8)
2227  1.18920711500272106672, //2^(1/4)
2228  M_SQRT2,
2229  2,
2230 };
2231 
2232 /**
2233  * Decode coupling_channel_element; reference: table 4.8.
2234  *
2235  * @return Returns error status. 0 - OK, !0 - error
2236  */
2238 {
2239  int num_gain = 0;
2240  int c, g, sfb, ret;
2241  int sign;
2242  INTFLOAT scale;
2243  SingleChannelElement *sce = &che->ch[0];
2244  ChannelCoupling *coup = &che->coup;
2245 
2246  coup->coupling_point = 2 * get_bits1(gb);
2247  coup->num_coupled = get_bits(gb, 3);
2248  for (c = 0; c <= coup->num_coupled; c++) {
2249  num_gain++;
2250  coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
2251  coup->id_select[c] = get_bits(gb, 4);
2252  if (coup->type[c] == TYPE_CPE) {
2253  coup->ch_select[c] = get_bits(gb, 2);
2254  if (coup->ch_select[c] == 3)
2255  num_gain++;
2256  } else
2257  coup->ch_select[c] = 2;
2258  }
2259  coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
2260 
2261  sign = get_bits(gb, 1);
2262 #if USE_FIXED
2263  scale = get_bits(gb, 2);
2264 #else
2265  scale = cce_scale[get_bits(gb, 2)];
2266 #endif
2267 
2268  if ((ret = decode_ics(ac, sce, gb, 0, 0)))
2269  return ret;
2270 
2271  for (c = 0; c < num_gain; c++) {
2272  int idx = 0;
2273  int cge = 1;
2274  int gain = 0;
2275  INTFLOAT gain_cache = FIXR10(1.);
2276  if (c) {
2277  cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
2278  gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
2279  gain_cache = GET_GAIN(scale, gain);
2280 #if USE_FIXED
2281  if ((abs(gain_cache)-1024) >> 3 > 30)
2282  return AVERROR(ERANGE);
2283 #endif
2284  }
2285  if (coup->coupling_point == AFTER_IMDCT) {
2286  coup->gain[c][0] = gain_cache;
2287  } else {
2288  for (g = 0; g < sce->ics.num_window_groups; g++) {
2289  for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
2290  if (sce->band_type[idx] != ZERO_BT) {
2291  if (!cge) {
2292  int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
2293  if (t) {
2294  int s = 1;
2295  t = gain += t;
2296  if (sign) {
2297  s -= 2 * (t & 0x1);
2298  t >>= 1;
2299  }
2300  gain_cache = GET_GAIN(scale, t) * s;
2301 #if USE_FIXED
2302  if ((abs(gain_cache)-1024) >> 3 > 30)
2303  return AVERROR(ERANGE);
2304 #endif
2305  }
2306  }
2307  coup->gain[c][idx] = gain_cache;
2308  }
2309  }
2310  }
2311  }
2312  }
2313  return 0;
2314 }
2315 
2316 /**
2317  * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
2318  *
2319  * @return Returns number of bytes consumed.
2320  */
2322  GetBitContext *gb)
2323 {
2324  int i;
2325  int num_excl_chan = 0;
2326 
2327  do {
2328  for (i = 0; i < 7; i++)
2329  che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
2330  } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
2331 
2332  return num_excl_chan / 7;
2333 }
2334 
2335 /**
2336  * Decode dynamic range information; reference: table 4.52.
2337  *
2338  * @return Returns number of bytes consumed.
2339  */
2341  GetBitContext *gb)
2342 {
2343  int n = 1;
2344  int drc_num_bands = 1;
2345  int i;
2346 
2347  /* pce_tag_present? */
2348  if (get_bits1(gb)) {
2349  che_drc->pce_instance_tag = get_bits(gb, 4);
2350  skip_bits(gb, 4); // tag_reserved_bits
2351  n++;
2352  }
2353 
2354  /* excluded_chns_present? */
2355  if (get_bits1(gb)) {
2356  n += decode_drc_channel_exclusions(che_drc, gb);
2357  }
2358 
2359  /* drc_bands_present? */
2360  if (get_bits1(gb)) {
2361  che_drc->band_incr = get_bits(gb, 4);
2362  che_drc->interpolation_scheme = get_bits(gb, 4);
2363  n++;
2364  drc_num_bands += che_drc->band_incr;
2365  for (i = 0; i < drc_num_bands; i++) {
2366  che_drc->band_top[i] = get_bits(gb, 8);
2367  n++;
2368  }
2369  }
2370 
2371  /* prog_ref_level_present? */
2372  if (get_bits1(gb)) {
2373  che_drc->prog_ref_level = get_bits(gb, 7);
2374  skip_bits1(gb); // prog_ref_level_reserved_bits
2375  n++;
2376  }
2377 
2378  for (i = 0; i < drc_num_bands; i++) {
2379  che_drc->dyn_rng_sgn[i] = get_bits1(gb);
2380  che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
2381  n++;
2382  }
2383 
2384  return n;
2385 }
2386 
2387 static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
2388  uint8_t buf[256];
2389  int i, major, minor;
2390 
2391  if (len < 13+7*8)
2392  goto unknown;
2393 
2394  get_bits(gb, 13); len -= 13;
2395 
2396  for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
2397  buf[i] = get_bits(gb, 8);
2398 
2399  buf[i] = 0;
2400  if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
2401  av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
2402 
2403  if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
2404  ac->avctx->internal->skip_samples = 1024;
2405  }
2406 
2407 unknown:
2408  skip_bits_long(gb, len);
2409 
2410  return 0;
2411 }
2412 
2413 /**
2414  * Decode extension data (incomplete); reference: table 4.51.
2415  *
2416  * @param cnt length of TYPE_FIL syntactic element in bytes
2417  *
2418  * @return Returns number of bytes consumed
2419  */
2421  ChannelElement *che, enum RawDataBlockType elem_type)
2422 {
2423  int crc_flag = 0;
2424  int res = cnt;
2425  int type = get_bits(gb, 4);
2426 
2427  if (ac->avctx->debug & FF_DEBUG_STARTCODE)
2428  av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
2429 
2430  switch (type) { // extension type
2431  case EXT_SBR_DATA_CRC:
2432  crc_flag++;
2433  case EXT_SBR_DATA:
2434  if (!che) {
2435  av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
2436  return res;
2437  } else if (ac->oc[1].m4ac.frame_length_short) {
2438  if (!ac->warned_960_sbr)
2440  "SBR with 960 frame length");
2441  ac->warned_960_sbr = 1;
2442  skip_bits_long(gb, 8 * cnt - 4);
2443  return res;
2444  } else if (!ac->oc[1].m4ac.sbr) {
2445  av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
2446  skip_bits_long(gb, 8 * cnt - 4);
2447  return res;
2448  } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
2449  av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
2450  skip_bits_long(gb, 8 * cnt - 4);
2451  return res;
2452  } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
2453  ac->oc[1].m4ac.sbr = 1;
2454  ac->oc[1].m4ac.ps = 1;
2456  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
2457  ac->oc[1].status, 1);
2458  } else {
2459  ac->oc[1].m4ac.sbr = 1;
2461  }
2462  res = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
2463  break;
2464  case EXT_DYNAMIC_RANGE:
2465  res = decode_dynamic_range(&ac->che_drc, gb);
2466  break;
2467  case EXT_FILL:
2468  decode_fill(ac, gb, 8 * cnt - 4);
2469  break;
2470  case EXT_FILL_DATA:
2471  case EXT_DATA_ELEMENT:
2472  default:
2473  skip_bits_long(gb, 8 * cnt - 4);
2474  break;
2475  };
2476  return res;
2477 }
2478 
2479 /**
2480  * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
2481  *
2482  * @param decode 1 if tool is used normally, 0 if tool is used in LTP.
2483  * @param coef spectral coefficients
2484  */
2485 static void apply_tns(INTFLOAT coef_param[1024], TemporalNoiseShaping *tns,
2486  IndividualChannelStream *ics, int decode)
2487 {
2488  const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
2489  int w, filt, m, i;
2490  int bottom, top, order, start, end, size, inc;
2491  INTFLOAT lpc[TNS_MAX_ORDER];
2493  UINTFLOAT *coef = coef_param;
2494 
2495  if(!mmm)
2496  return;
2497 
2498  for (w = 0; w < ics->num_windows; w++) {
2499  bottom = ics->num_swb;
2500  for (filt = 0; filt < tns->n_filt[w]; filt++) {
2501  top = bottom;
2502  bottom = FFMAX(0, top - tns->length[w][filt]);
2503  order = tns->order[w][filt];
2504  if (order == 0)
2505  continue;
2506 
2507  // tns_decode_coef
2508  AAC_RENAME(compute_lpc_coefs)(tns->coef[w][filt], order, lpc, 0, 0, 0);
2509 
2510  start = ics->swb_offset[FFMIN(bottom, mmm)];
2511  end = ics->swb_offset[FFMIN( top, mmm)];
2512  if ((size = end - start) <= 0)
2513  continue;
2514  if (tns->direction[w][filt]) {
2515  inc = -1;
2516  start = end - 1;
2517  } else {
2518  inc = 1;
2519  }
2520  start += w * 128;
2521 
2522  if (decode) {
2523  // ar filter
2524  for (m = 0; m < size; m++, start += inc)
2525  for (i = 1; i <= FFMIN(m, order); i++)
2526  coef[start] -= AAC_MUL26((INTFLOAT)coef[start - i * inc], lpc[i - 1]);
2527  } else {
2528  // ma filter
2529  for (m = 0; m < size; m++, start += inc) {
2530  tmp[0] = coef[start];
2531  for (i = 1; i <= FFMIN(m, order); i++)
2532  coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
2533  for (i = order; i > 0; i--)
2534  tmp[i] = tmp[i - 1];
2535  }
2536  }
2537  }
2538  }
2539 }
2540 
2541 /**
2542  * Apply windowing and MDCT to obtain the spectral
2543  * coefficient from the predicted sample by LTP.
2544  */
2547 {
2548  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2549  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2550  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2551  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2552 
2553  if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
2554  ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
2555  } else {
2556  memset(in, 0, 448 * sizeof(*in));
2557  ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
2558  }
2559  if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
2560  ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
2561  } else {
2562  ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
2563  memset(in + 1024 + 576, 0, 448 * sizeof(*in));
2564  }
2565  ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
2566 }
2567 
2568 /**
2569  * Apply the long term prediction
2570  */
2572 {
2573  const LongTermPrediction *ltp = &sce->ics.ltp;
2574  const uint16_t *offsets = sce->ics.swb_offset;
2575  int i, sfb;
2576 
2577  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
2578  INTFLOAT *predTime = sce->ret;
2579  INTFLOAT *predFreq = ac->buf_mdct;
2580  int16_t num_samples = 2048;
2581 
2582  if (ltp->lag < 1024)
2583  num_samples = ltp->lag + 1024;
2584  for (i = 0; i < num_samples; i++)
2585  predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
2586  memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));
2587 
2588  ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
2589 
2590  if (sce->tns.present)
2591  ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
2592 
2593  for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
2594  if (ltp->used[sfb])
2595  for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
2596  sce->coeffs[i] += (UINTFLOAT)predFreq[i];
2597  }
2598 }
2599 
2600 /**
2601  * Update the LTP buffer for next frame
2602  */
2604 {
2605  IndividualChannelStream *ics = &sce->ics;
2606  INTFLOAT *saved = sce->saved;
2607  INTFLOAT *saved_ltp = sce->coeffs;
2608  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2609  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2610  int i;
2611 
2612  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2613  memcpy(saved_ltp, saved, 512 * sizeof(*saved_ltp));
2614  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2615  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2616 
2617  for (i = 0; i < 64; i++)
2618  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2619  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2620  memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(*saved_ltp));
2621  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2622  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2623 
2624  for (i = 0; i < 64; i++)
2625  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2626  } else { // LONG_STOP or ONLY_LONG
2627  ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
2628 
2629  for (i = 0; i < 512; i++)
2630  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
2631  }
2632 
2633  memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
2634  memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
2635  memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
2636 }
2637 
2638 /**
2639  * Conduct IMDCT and windowing.
2640  */
2642 {
2643  IndividualChannelStream *ics = &sce->ics;
2644  INTFLOAT *in = sce->coeffs;
2645  INTFLOAT *out = sce->ret;
2646  INTFLOAT *saved = sce->saved;
2647  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2648  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2649  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2650  INTFLOAT *buf = ac->buf_mdct;
2651  INTFLOAT *temp = ac->temp;
2652  int i;
2653 
2654  // imdct
2655  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2656  for (i = 0; i < 1024; i += 128)
2657  ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
2658  } else {
2659  ac->mdct.imdct_half(&ac->mdct, buf, in);
2660 #if USE_FIXED
2661  for (i=0; i<1024; i++)
2662  buf[i] = (buf[i] + 4LL) >> 3;
2663 #endif /* USE_FIXED */
2664  }
2665 
2666  /* window overlapping
2667  * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2668  * and long to short transitions are considered to be short to short
2669  * transitions. This leaves just two cases (long to long and short to short)
2670  * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2671  */
2672  if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2674  ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
2675  } else {
2676  memcpy( out, saved, 448 * sizeof(*out));
2677 
2678  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2679  ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
2680  ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
2681  ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
2682  ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
2683  ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
2684  memcpy( out + 448 + 4*128, temp, 64 * sizeof(*out));
2685  } else {
2686  ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
2687  memcpy( out + 576, buf + 64, 448 * sizeof(*out));
2688  }
2689  }
2690 
2691  // buffer update
2692  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2693  memcpy( saved, temp + 64, 64 * sizeof(*saved));
2694  ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
2695  ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
2696  ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
2697  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2698  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2699  memcpy( saved, buf + 512, 448 * sizeof(*saved));
2700  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2701  } else { // LONG_STOP or ONLY_LONG
2702  memcpy( saved, buf + 512, 512 * sizeof(*saved));
2703  }
2704 }
2705 
2706 /**
2707  * Conduct IMDCT and windowing.
2708  */
2710 {
2711 #if !USE_FIXED
2712  IndividualChannelStream *ics = &sce->ics;
2713  INTFLOAT *in = sce->coeffs;
2714  INTFLOAT *out = sce->ret;
2715  INTFLOAT *saved = sce->saved;
2716  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_120) : AAC_RENAME(ff_sine_120);
2717  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_960) : AAC_RENAME(ff_sine_960);
2718  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_120) : AAC_RENAME(ff_sine_120);
2719  INTFLOAT *buf = ac->buf_mdct;
2720  INTFLOAT *temp = ac->temp;
2721  int i;
2722 
2723  // imdct
2724  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2725  for (i = 0; i < 8; i++)
2726  ac->mdct120->imdct_half(ac->mdct120, buf + i * 120, in + i * 128, 1);
2727  } else {
2728  ac->mdct960->imdct_half(ac->mdct960, buf, in, 1);
2729  }
2730 
2731  /* window overlapping
2732  * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2733  * and long to short transitions are considered to be short to short
2734  * transitions. This leaves just two cases (long to long and short to short)
2735  * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2736  */
2737 
2738  if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2740  ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 480);
2741  } else {
2742  memcpy( out, saved, 420 * sizeof(*out));
2743 
2744  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2745  ac->fdsp->vector_fmul_window(out + 420 + 0*120, saved + 420, buf + 0*120, swindow_prev, 60);
2746  ac->fdsp->vector_fmul_window(out + 420 + 1*120, buf + 0*120 + 60, buf + 1*120, swindow, 60);
2747  ac->fdsp->vector_fmul_window(out + 420 + 2*120, buf + 1*120 + 60, buf + 2*120, swindow, 60);
2748  ac->fdsp->vector_fmul_window(out + 420 + 3*120, buf + 2*120 + 60, buf + 3*120, swindow, 60);
2749  ac->fdsp->vector_fmul_window(temp, buf + 3*120 + 60, buf + 4*120, swindow, 60);
2750  memcpy( out + 420 + 4*120, temp, 60 * sizeof(*out));
2751  } else {
2752  ac->fdsp->vector_fmul_window(out + 420, saved + 420, buf, swindow_prev, 60);
2753  memcpy( out + 540, buf + 60, 420 * sizeof(*out));
2754  }
2755  }
2756 
2757  // buffer update
2758  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2759  memcpy( saved, temp + 60, 60 * sizeof(*saved));
2760  ac->fdsp->vector_fmul_window(saved + 60, buf + 4*120 + 60, buf + 5*120, swindow, 60);
2761  ac->fdsp->vector_fmul_window(saved + 180, buf + 5*120 + 60, buf + 6*120, swindow, 60);
2762  ac->fdsp->vector_fmul_window(saved + 300, buf + 6*120 + 60, buf + 7*120, swindow, 60);
2763  memcpy( saved + 420, buf + 7*120 + 60, 60 * sizeof(*saved));
2764  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2765  memcpy( saved, buf + 480, 420 * sizeof(*saved));
2766  memcpy( saved + 420, buf + 7*120 + 60, 60 * sizeof(*saved));
2767  } else { // LONG_STOP or ONLY_LONG
2768  memcpy( saved, buf + 480, 480 * sizeof(*saved));
2769  }
2770 #endif
2771 }
2773 {
2774  IndividualChannelStream *ics = &sce->ics;
2775  INTFLOAT *in = sce->coeffs;
2776  INTFLOAT *out = sce->ret;
2777  INTFLOAT *saved = sce->saved;
2778  INTFLOAT *buf = ac->buf_mdct;
2779 #if USE_FIXED
2780  int i;
2781 #endif /* USE_FIXED */
2782 
2783  // imdct
2784  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2785 
2786 #if USE_FIXED
2787  for (i = 0; i < 1024; i++)
2788  buf[i] = (buf[i] + 2) >> 2;
2789 #endif /* USE_FIXED */
2790 
2791  // window overlapping
2792  if (ics->use_kb_window[1]) {
2793  // AAC LD uses a low overlap sine window instead of a KBD window
2794  memcpy(out, saved, 192 * sizeof(*out));
2795  ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
2796  memcpy( out + 320, buf + 64, 192 * sizeof(*out));
2797  } else {
2798  ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);
2799  }
2800 
2801  // buffer update
2802  memcpy(saved, buf + 256, 256 * sizeof(*saved));
2803 }
2804 
2806 {
2807  INTFLOAT *in = sce->coeffs;
2808  INTFLOAT *out = sce->ret;
2809  INTFLOAT *saved = sce->saved;
2810  INTFLOAT *buf = ac->buf_mdct;
2811  int i;
2812  const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
2813  const int n2 = n >> 1;
2814  const int n4 = n >> 2;
2815  const INTFLOAT *const window = n == 480 ? AAC_RENAME(ff_aac_eld_window_480) :
2817 
2818  // Inverse transform, mapped to the conventional IMDCT by
2819  // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
2820  // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
2821  // International Conference on Audio, Language and Image Processing, ICALIP 2008.
2822  // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
2823  for (i = 0; i < n2; i+=2) {
2824  INTFLOAT temp;
2825  temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
2826  temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
2827  }
2828 #if !USE_FIXED
2829  if (n == 480)
2830  ac->mdct480->imdct_half(ac->mdct480, buf, in, 1);
2831  else
2832 #endif
2833  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2834 
2835 #if USE_FIXED
2836  for (i = 0; i < 1024; i++)
2837  buf[i] = (buf[i] + 1) >> 1;
2838 #endif /* USE_FIXED */
2839 
2840  for (i = 0; i < n; i+=2) {
2841  buf[i] = -buf[i];
2842  }
2843  // Like with the regular IMDCT at this point we still have the middle half
2844  // of a transform but with even symmetry on the left and odd symmetry on
2845  // the right
2846 
2847  // window overlapping
2848  // The spec says to use samples [0..511] but the reference decoder uses
2849  // samples [128..639].
2850  for (i = n4; i < n2; i ++) {
2851  out[i - n4] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) +
2852  AAC_MUL31( saved[ i + n2] , window[i + n - n4]) +
2853  AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
2854  AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]);
2855  }
2856  for (i = 0; i < n2; i ++) {
2857  out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) +
2858  AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) +
2859  AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) +
2860  AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
2861  }
2862  for (i = 0; i < n4; i ++) {
2863  out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) +
2864  AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
2865  AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
2866  }
2867 
2868  // buffer update
2869  memmove(saved + n, saved, 2 * n * sizeof(*saved));
2870  memcpy( saved, buf, n * sizeof(*saved));
2871 }
2872 
2873 /**
2874  * channel coupling transformation interface
2875  *
2876  * @param apply_coupling_method pointer to (in)dependent coupling function
2877  */
2879  enum RawDataBlockType type, int elem_id,
2880  enum CouplingPoint coupling_point,
2881  void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
2882 {
2883  int i, c;
2884 
2885  for (i = 0; i < MAX_ELEM_ID; i++) {
2886  ChannelElement *cce = ac->che[TYPE_CCE][i];
2887  int index = 0;
2888 
2889  if (cce && cce->coup.coupling_point == coupling_point) {
2890  ChannelCoupling *coup = &cce->coup;
2891 
2892  for (c = 0; c <= coup->num_coupled; c++) {
2893  if (coup->type[c] == type && coup->id_select[c] == elem_id) {
2894  if (coup->ch_select[c] != 1) {
2895  apply_coupling_method(ac, &cc->ch[0], cce, index);
2896  if (coup->ch_select[c] != 0)
2897  index++;
2898  }
2899  if (coup->ch_select[c] != 2)
2900  apply_coupling_method(ac, &cc->ch[1], cce, index++);
2901  } else
2902  index += 1 + (coup->ch_select[c] == 3);
2903  }
2904  }
2905  }
2906 }
2907 
2908 /**
2909  * Convert spectral data to samples, applying all supported tools as appropriate.
2910  */
2911 static void spectral_to_sample(AACContext *ac, int samples)
2912 {
2913  int i, type;
2915  switch (ac->oc[1].m4ac.object_type) {
2916  case AOT_ER_AAC_LD:
2918  break;
2919  case AOT_ER_AAC_ELD:
2921  break;
2922  default:
2923  if (ac->oc[1].m4ac.frame_length_short)
2925  else
2927  }
2928  for (type = 3; type >= 0; type--) {
2929  for (i = 0; i < MAX_ELEM_ID; i++) {
2930  ChannelElement *che = ac->che[type][i];
2931  if (che && che->present) {
2932  if (type <= TYPE_CPE)
2934  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2935  if (che->ch[0].ics.predictor_present) {
2936  if (che->ch[0].ics.ltp.present)
2937  ac->apply_ltp(ac, &che->ch[0]);
2938  if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
2939  ac->apply_ltp(ac, &che->ch[1]);
2940  }
2941  }
2942  if (che->ch[0].tns.present)
2943  ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2944  if (che->ch[1].tns.present)
2945  ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2946  if (type <= TYPE_CPE)
2948  if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2949  imdct_and_window(ac, &che->ch[0]);
2950  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2951  ac->update_ltp(ac, &che->ch[0]);
2952  if (type == TYPE_CPE) {
2953  imdct_and_window(ac, &che->ch[1]);
2954  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2955  ac->update_ltp(ac, &che->ch[1]);
2956  }
2957  if (ac->oc[1].m4ac.sbr > 0) {
2958  AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2959  }
2960  }
2961  if (type <= TYPE_CCE)
2963 
2964 #if USE_FIXED
2965  {
2966  int j;
2967  /* preparation for resampler */
2968  for(j = 0; j<samples; j++){
2969  che->ch[0].ret[j] = (int32_t)av_clip64((int64_t)che->ch[0].ret[j]*128, INT32_MIN, INT32_MAX-0x8000)+0x8000;
2970  if(type == TYPE_CPE)
2971  che->ch[1].ret[j] = (int32_t)av_clip64((int64_t)che->ch[1].ret[j]*128, INT32_MIN, INT32_MAX-0x8000)+0x8000;
2972  }
2973  }
2974 #endif /* USE_FIXED */
2975  che->present = 0;
2976  } else if (che) {
2977  av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
2978  }
2979  }
2980  }
2981 }
2982 
2984 {
2985  int size;
2986  AACADTSHeaderInfo hdr_info;
2987  uint8_t layout_map[MAX_ELEM_ID*4][3];
2988  int layout_map_tags, ret;
2989 
2990  size = ff_adts_header_parse(gb, &hdr_info);
2991  if (size > 0) {
2992  if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
2993  // This is 2 for "VLB " audio in NSV files.
2994  // See samples/nsv/vlb_audio.
2996  "More than one AAC RDB per ADTS frame");
2997  ac->warned_num_aac_frames = 1;
2998  }
3000  if (hdr_info.chan_config) {
3001  ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
3002  if ((ret = set_default_channel_config(ac->avctx,
3003  layout_map,
3004  &layout_map_tags,
3005  hdr_info.chan_config)) < 0)
3006  return ret;
3007  if ((ret = output_configure(ac, layout_map, layout_map_tags,
3008  FFMAX(ac->oc[1].status,
3009  OC_TRIAL_FRAME), 0)) < 0)
3010  return ret;
3011  } else {
3012  ac->oc[1].m4ac.chan_config = 0;
3013  /**
3014  * dual mono frames in Japanese DTV can have chan_config 0
3015  * WITHOUT specifying PCE.
3016  * thus, set dual mono as default.
3017  */
3018  if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
3019  layout_map_tags = 2;
3020  layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
3021  layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
3022  layout_map[0][1] = 0;
3023  layout_map[1][1] = 1;
3024  if (output_configure(ac, layout_map, layout_map_tags,
3025  OC_TRIAL_FRAME, 0))
3026  return -7;
3027  }
3028  }
3029  ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
3030  ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
3031  ac->oc[1].m4ac.object_type = hdr_info.object_type;
3032  ac->oc[1].m4ac.frame_length_short = 0;
3033  if (ac->oc[0].status != OC_LOCKED ||
3034  ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
3035  ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
3036  ac->oc[1].m4ac.sbr = -1;
3037  ac->oc[1].m4ac.ps = -1;
3038  }
3039  if (!hdr_info.crc_absent)
3040  skip_bits(gb, 16);
3041  }
3042  return size;
3043 }
3044 
3045 static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
3046  int *got_frame_ptr, GetBitContext *gb)
3047 {
3048  AACContext *ac = avctx->priv_data;
3049  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
3050  ChannelElement *che;
3051  int err, i;
3052  int samples = m4ac->frame_length_short ? 960 : 1024;
3053  int chan_config = m4ac->chan_config;
3054  int aot = m4ac->object_type;
3055 
3056  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
3057  samples >>= 1;
3058 
3059  ac->frame = data;
3060 
3061  if ((err = frame_configure_elements(avctx)) < 0)
3062  return err;
3063 
3064  // The FF_PROFILE_AAC_* defines are all object_type - 1
3065  // This may lead to an undefined profile being signaled
3066  ac->avctx->profile = aot - 1;
3067 
3068  ac->tags_mapped = 0;
3069 
3070  if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
3071  avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
3072  chan_config);
3073  return AVERROR_INVALIDDATA;
3074  }
3075  for (i = 0; i < tags_per_config[chan_config]; i++) {
3076  const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
3077  const int elem_id = aac_channel_layout_map[chan_config-1][i][1];
3078  if (!(che=get_che(ac, elem_type, elem_id))) {
3079  av_log(ac->avctx, AV_LOG_ERROR,
3080  "channel element %d.%d is not allocated\n",
3081  elem_type, elem_id);
3082  return AVERROR_INVALIDDATA;
3083  }
3084  che->present = 1;
3085  if (aot != AOT_ER_AAC_ELD)
3086  skip_bits(gb, 4);
3087  switch (elem_type) {
3088  case TYPE_SCE:
3089  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3090  break;
3091  case TYPE_CPE:
3092  err = decode_cpe(ac, gb, che);
3093  break;
3094  case TYPE_LFE:
3095  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3096  break;
3097  }
3098  if (err < 0)
3099  return err;
3100  }
3101 
3102  spectral_to_sample(ac, samples);
3103 
3104  if (!ac->frame->data[0] && samples) {
3105  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
3106  return AVERROR_INVALIDDATA;
3107  }
3108 
3109  ac->frame->nb_samples = samples;
3110  ac->frame->sample_rate = avctx->sample_rate;
3111  *got_frame_ptr = 1;
3112 
3113  skip_bits_long(gb, get_bits_left(gb));
3114  return 0;
3115 }
3116 
3117 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
3118  int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
3119 {
3120  AACContext *ac = avctx->priv_data;
3121  ChannelElement *che = NULL, *che_prev = NULL;
3122  enum RawDataBlockType elem_type, che_prev_type = TYPE_END;
3123  int err, elem_id;
3124  int samples = 0, multiplier, audio_found = 0, pce_found = 0;
3125  int is_dmono, sce_count = 0;
3126  int payload_alignment;
3127  uint8_t che_presence[4][MAX_ELEM_ID] = {{0}};
3128 
3129  ac->frame = data;
3130 
3131  if (show_bits(gb, 12) == 0xfff) {
3132  if ((err = parse_adts_frame_header(ac, gb)) < 0) {
3133  av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
3134  goto fail;
3135  }
3136  if (ac->oc[1].m4ac.sampling_index > 12) {
3137  av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
3138  err = AVERROR_INVALIDDATA;
3139  goto fail;
3140  }
3141  }
3142 
3143  if ((err = frame_configure_elements(avctx)) < 0)
3144  goto fail;
3145 
3146  // The FF_PROFILE_AAC_* defines are all object_type - 1
3147  // This may lead to an undefined profile being signaled
3148  ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
3149 
3150  payload_alignment = get_bits_count(gb);
3151  ac->tags_mapped = 0;
3152  // parse
3153  while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
3154  elem_id = get_bits(gb, 4);
3155 
3156  if (avctx->debug & FF_DEBUG_STARTCODE)
3157  av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
3158 
3159  if (!avctx->channels && elem_type != TYPE_PCE) {
3160  err = AVERROR_INVALIDDATA;
3161  goto fail;
3162  }
3163 
3164  if (elem_type < TYPE_DSE) {
3165  if (che_presence[elem_type][elem_id]) {
3166  int error = che_presence[elem_type][elem_id] > 1;
3167  av_log(ac->avctx, error ? AV_LOG_ERROR : AV_LOG_DEBUG, "channel element %d.%d duplicate\n",
3168  elem_type, elem_id);
3169  if (error) {
3170  err = AVERROR_INVALIDDATA;
3171  goto fail;
3172  }
3173  }
3174  che_presence[elem_type][elem_id]++;
3175 
3176  if (!(che=get_che(ac, elem_type, elem_id))) {
3177  av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
3178  elem_type, elem_id);
3179  err = AVERROR_INVALIDDATA;
3180  goto fail;
3181  }
3182  samples = ac->oc[1].m4ac.frame_length_short ? 960 : 1024;
3183  che->present = 1;
3184  }
3185 
3186  switch (elem_type) {
3187 
3188  case TYPE_SCE:
3189  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3190  audio_found = 1;
3191  sce_count++;
3192  break;
3193 
3194  case TYPE_CPE:
3195  err = decode_cpe(ac, gb, che);
3196  audio_found = 1;
3197  break;
3198 
3199  case TYPE_CCE:
3200  err = decode_cce(ac, gb, che);
3201  break;
3202 
3203  case TYPE_LFE:
3204  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3205  audio_found = 1;
3206  break;
3207 
3208  case TYPE_DSE:
3209  err = skip_data_stream_element(ac, gb);
3210  break;
3211 
3212  case TYPE_PCE: {
3213  uint8_t layout_map[MAX_ELEM_ID*4][3];
3214  int tags;
3215 
3216  int pushed = push_output_configuration(ac);
3217  if (pce_found && !pushed) {
3218  err = AVERROR_INVALIDDATA;
3219  goto fail;
3220  }
3221 
3222  tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb,
3223  payload_alignment);
3224  if (tags < 0) {
3225  err = tags;
3226  break;
3227  }
3228  if (pce_found) {
3229  av_log(avctx, AV_LOG_ERROR,
3230  "Not evaluating a further program_config_element as this construct is dubious at best.\n");
3232  } else {
3233  err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
3234  if (!err)
3235  ac->oc[1].m4ac.chan_config = 0;
3236  pce_found = 1;
3237  }
3238  break;
3239  }
3240 
3241  case TYPE_FIL:
3242  if (elem_id == 15)
3243  elem_id += get_bits(gb, 8) - 1;
3244  if (get_bits_left(gb) < 8 * elem_id) {
3245  av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
3246  err = AVERROR_INVALIDDATA;
3247  goto fail;
3248  }
3249  while (elem_id > 0)
3250  elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type);
3251  err = 0; /* FIXME */
3252  break;
3253 
3254  default:
3255  err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
3256  break;
3257  }
3258 
3259  if (elem_type < TYPE_DSE) {
3260  che_prev = che;
3261  che_prev_type = elem_type;
3262  }
3263 
3264  if (err)
3265  goto fail;
3266 
3267  if (get_bits_left(gb) < 3) {
3268  av_log(avctx, AV_LOG_ERROR, overread_err);
3269  err = AVERROR_INVALIDDATA;
3270  goto fail;
3271  }
3272  }
3273 
3274  if (!avctx->channels) {
3275  *got_frame_ptr = 0;
3276  return 0;
3277  }
3278 
3279  multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
3280  samples <<= multiplier;
3281 
3282  spectral_to_sample(ac, samples);
3283 
3284  if (ac->oc[1].status && audio_found) {
3285  avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
3286  avctx->frame_size = samples;
3287  ac->oc[1].status = OC_LOCKED;
3288  }
3289 
3290  if (multiplier)
3291  avctx->internal->skip_samples_multiplier = 2;
3292 
3293  if (!ac->frame->data[0] && samples) {
3294  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
3295  err = AVERROR_INVALIDDATA;
3296  goto fail;
3297  }
3298 
3299  if (samples) {
3300  ac->frame->nb_samples = samples;
3301  ac->frame->sample_rate = avctx->sample_rate;
3302  } else
3303  av_frame_unref(ac->frame);
3304  *got_frame_ptr = !!samples;
3305 
3306  /* for dual-mono audio (SCE + SCE) */
3307  is_dmono = ac->dmono_mode && sce_count == 2 &&
3309  if (is_dmono) {
3310  if (ac->dmono_mode == 1)
3311  ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
3312  else if (ac->dmono_mode == 2)
3313  ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
3314  }
3315 
3316  return 0;
3317 fail:
3319  return err;
3320 }
3321 
3322 static int aac_decode_frame(AVCodecContext *avctx, void *data,
3323  int *got_frame_ptr, AVPacket *avpkt)
3324 {
3325  AACContext *ac = avctx->priv_data;
3326  const uint8_t *buf = avpkt->data;
3327  int buf_size = avpkt->size;
3328  GetBitContext gb;
3329  int buf_consumed;
3330  int buf_offset;
3331  int err;
3332  int new_extradata_size;
3333  const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
3335  &new_extradata_size);
3336  int jp_dualmono_size;
3337  const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
3339  &jp_dualmono_size);
3340 
3341  if (new_extradata) {
3342  /* discard previous configuration */
3343  ac->oc[1].status = OC_NONE;
3344  err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
3345  new_extradata,
3346  new_extradata_size * 8LL, 1);
3347  if (err < 0) {
3348  return err;
3349  }
3350  }
3351 
3352  ac->dmono_mode = 0;
3353  if (jp_dualmono && jp_dualmono_size > 0)
3354  ac->dmono_mode = 1 + *jp_dualmono;
3355  if (ac->force_dmono_mode >= 0)
3356  ac->dmono_mode = ac->force_dmono_mode;
3357 
3358  if (INT_MAX / 8 <= buf_size)
3359  return AVERROR_INVALIDDATA;
3360 
3361  if ((err = init_get_bits8(&gb, buf, buf_size)) < 0)
3362  return err;
3363 
3364  switch (ac->oc[1].m4ac.object_type) {
3365  case AOT_ER_AAC_LC:
3366  case AOT_ER_AAC_LTP:
3367  case AOT_ER_AAC_LD:
3368  case AOT_ER_AAC_ELD:
3369  err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
3370  break;
3371  default:
3372  err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
3373  }
3374  if (err < 0)
3375  return err;
3376 
3377  buf_consumed = (get_bits_count(&gb) + 7) >> 3;
3378  for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
3379  if (buf[buf_offset])
3380  break;
3381 
3382  return buf_size > buf_offset ? buf_consumed : buf_size;
3383 }
3384 
3386 {
3387  AACContext *ac = avctx->priv_data;
3388  int i, type;
3389 
3390  for (i = 0; i < MAX_ELEM_ID; i++) {
3391  for (type = 0; type < 4; type++) {
3392  if (ac->che[type][i])
3394  av_freep(&ac->che[type][i]);
3395  }
3396  }
3397 
3398  ff_mdct_end(&ac->mdct);
3399  ff_mdct_end(&ac->mdct_small);
3400  ff_mdct_end(&ac->mdct_ld);
3401  ff_mdct_end(&ac->mdct_ltp);
3402 #if !USE_FIXED
3403  ff_mdct15_uninit(&ac->mdct120);
3404  ff_mdct15_uninit(&ac->mdct480);
3405  ff_mdct15_uninit(&ac->mdct960);
3406 #endif
3407  av_freep(&ac->fdsp);
3408  return 0;
3409 }
3410 
3411 static void aacdec_init(AACContext *c)
3412 {
3414  c->apply_ltp = apply_ltp;
3415  c->apply_tns = apply_tns;
3417  c->update_ltp = update_ltp;
3418 #if USE_FIXED
3421 #endif
3422 
3423 #if !USE_FIXED
3424  if(ARCH_MIPS)
3426 #endif /* !USE_FIXED */
3427 }
3428 /**
3429  * AVOptions for Japanese DTV specific extensions (ADTS only)
3430  */
3431 #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
3432 static const AVOption options[] = {
3433  {"dual_mono_mode", "Select the channel to decode for dual mono",
3434  offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
3435  AACDEC_FLAGS, "dual_mono_mode"},
3436 
3437  {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3438  {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3439  {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3440  {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3441 
3442  {NULL},
3443 };
3444 
3445 static const AVClass aac_decoder_class = {
3446  .class_name = "AAC decoder",
3447  .item_name = av_default_item_name,
3448  .option = options,
3449  .version = LIBAVUTIL_VERSION_INT,
3450 };
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
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
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
#define overread_err
int size
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
uint8_t object_type
Definition: adts_header.h:33
AVOption.
Definition: opt.h:246
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
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:379
#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:291
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
#define avpriv_request_sample(...)
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:1478
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi - 0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63))) #define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_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), };static void cpy1(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, len);} static void cpy2(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 2 *len);} static void cpy4(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 4 *len);} static void cpy8(uint8_t **dst, const uint8_t **src, int len){ memcpy(*dst, *src, 8 *len);} AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags) { 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) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;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)){ case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;} } if(HAVE_X86ASM &&HAVE_MMX) 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);return ctx;} void swri_audio_convert_free(AudioConvert **ctx) { av_freep(ctx);} int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) { int ch;int off=0;const int os=(out->planar ? 1 :out->ch_count) *out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask) { int planes=in->planar ? in->ch_count :1;unsigned m=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) { int planes=out->planar ? out->ch_count :1;unsigned m=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){ int planes=out->planar ? out->ch_count :1;for(ch=0;ch< planes;ch++){ ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
const char * b
Definition: vf_curves.c:116
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:191
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.
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
#define AV_EF_BITSTREAM
detect bitstream specification deviations
Definition: avcodec.h:2703
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
#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:2907
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:2898
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
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb)
Decode Individual Channel Stream info; reference: table 4.6.
Definition: aac.h:67
BandType
Definition: aac.h:82
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2233
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:2651
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
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
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:1666
int num_coupled
number of target elements
Definition: aac.h:236
#define u(width, name, range_min, range_max)
Definition: cbs_h2645.c:252
#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
const char data[16]
Definition: mxf.c:91
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:1477
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
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:849
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
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 i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
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
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.
#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:1645
static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx, GetBitContext *gb, MPEG4AudioConfig *m4ac, int channel_config)
void(* mdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:109
uint8_t sampling_index
Definition: adts_header.h:34
int amp[4]
Definition: aac.h:228
void(* apply_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:364
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)
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
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:175
uint8_t bits
Definition: vp3data.h:202
#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
#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:120
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:2276
#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:367
static void reset_predictor_group(PredictorState *ps, int group_num)
void(* apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns, IndividualChannelStream *ics, int decode)
Definition: aac.h:365
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:908
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:2694
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.
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:446
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:797
int length[8][4]
Definition: aac.h:201
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:2705
int n
Definition: avisynth_c.h:760
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:706
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
#define av_log2
Definition: intmath.h:83
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:2245
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:1199
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
void(* imdct_half)(struct MDCT15Context *s, float *dst, const float *src, ptrdiff_t stride)
Definition: mdct15.h:52
typedef void(RENAME(mix_any_func_type))
#define AAC_INIT_VLC_STATIC(num, size)
Temporal Noise Shaping.
Definition: aac.h:198
int sample_rate
samples per second
Definition: avcodec.h:2225
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:677
#define AV_CH_LAYOUT_NATIVE
Channel mask value used for AVCodecContext.request_channel_layout to indicate that the user requests ...
#define abs(x)
Definition: cuda_runtime.h:35
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:2650
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:1565
#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:1964
IndividualChannelStream ics
Definition: aac.h:249
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
void * buf
Definition: avisynth_c.h:766
#define MAX_PREDICTORS
Definition: aac.h:146
static av_always_inline float cbrtf(float x)
Definition: libm.h:61
int extradata_size
Definition: avcodec.h:1667
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:498
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;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);return NULL;} return ac;} 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;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->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);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:538
#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:467
static av_cold int aac_decode_close(AVCodecContext *avctx)
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
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:659
#define GET_CACHE(name, gb)
Definition: get_bits.h:215
cl_device_type type
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 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 ARCH_MIPS
Definition: config.h:26
#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:2630
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
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:108
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:309
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:1310
int
const uint8_t ff_aac_pred_sfb_max[]
Definition: aactab.c:69
int direction[8][4]
Definition: aac.h:202
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(* vector_pow43)(int *coefs, int len)
Definition: aac.h:370
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
void(* update_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:369
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:1592
static void subband_scale(int *dst, int *src, int scale, int offset, int len, void *log_context)
Definition: aacdec_fixed.c:165
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:2664
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:2226
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:1600
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
void(* subband_scale)(int *dst, int *src, int scale, int offset, int len, void *log_context)
Definition: aac.h:371
uint8_t crc_absent
Definition: adts_header.h:32
static const uint8_t * align_get_bits(GetBitContext *s)
Definition: get_bits.h:693
uint64_t layout
#define FF_PROFILE_AAC_HE
Definition: avcodec.h:2906
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)
void(* windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out, INTFLOAT *in, IndividualChannelStream *ics)
Definition: aac.h:367
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 INT64 count
Definition: avisynth_c.h:766
void INT64 start
Definition: avisynth_c.h:766
#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.
void(* imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:363
#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:342
uint64_t request_channel_layout
Request decoder to use this channel layout if it can (0 for default)
Definition: avcodec.h:2283
int layout_map_tags
Definition: aac.h:126
enum AVCodecID id
This structure stores compressed data.
Definition: avcodec.h:1454
mode
Use these values in ebur128_init (or&#39;ed).
Definition: ebur128.h:83
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:2628
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
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
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
static uint8_t tmp[11]
Definition: aes_ctr.c:26