FFmpeg  3.4.9
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);
137  AAC_RENAME(ff_aac_sbr_ctx_init)(ac, &ac->che[type][id]->sbr, type);
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])
152  AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][id]->sbr);
153  av_freep(&ac->che[type][id]);
154  }
155  return 0;
156 }
157 
159 {
160  AACContext *ac = avctx->priv_data;
161  int type, id, ch, ret;
162 
163  /* set channel pointers to internal buffers by default */
164  for (type = 0; type < 4; type++) {
165  for (id = 0; id < MAX_ELEM_ID; id++) {
166  ChannelElement *che = ac->che[type][id];
167  if (che) {
168  che->ch[0].ret = che->ch[0].ret_buf;
169  che->ch[1].ret = che->ch[1].ret_buf;
170  }
171  }
172  }
173 
174  /* get output buffer */
175  av_frame_unref(ac->frame);
176  if (!avctx->channels)
177  return 1;
178 
179  ac->frame->nb_samples = 2048;
180  if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
181  return ret;
182 
183  /* map output channel pointers to AVFrame data */
184  for (ch = 0; ch < avctx->channels; ch++) {
185  if (ac->output_element[ch])
186  ac->output_element[ch]->ret = (INTFLOAT *)ac->frame->extended_data[ch];
187  }
188 
189  return 0;
190 }
191 
193  uint64_t av_position;
197 };
198 
199 static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
200  uint8_t (*layout_map)[3], int offset, uint64_t left,
201  uint64_t right, int pos)
202 {
203  if (layout_map[offset][0] == TYPE_CPE) {
204  e2c_vec[offset] = (struct elem_to_channel) {
205  .av_position = left | right,
206  .syn_ele = TYPE_CPE,
207  .elem_id = layout_map[offset][1],
208  .aac_position = pos
209  };
210  return 1;
211  } else {
212  e2c_vec[offset] = (struct elem_to_channel) {
213  .av_position = left,
214  .syn_ele = TYPE_SCE,
215  .elem_id = layout_map[offset][1],
216  .aac_position = pos
217  };
218  e2c_vec[offset + 1] = (struct elem_to_channel) {
219  .av_position = right,
220  .syn_ele = TYPE_SCE,
221  .elem_id = layout_map[offset + 1][1],
222  .aac_position = pos
223  };
224  return 2;
225  }
226 }
227 
228 static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
229  int *current)
230 {
231  int num_pos_channels = 0;
232  int first_cpe = 0;
233  int sce_parity = 0;
234  int i;
235  for (i = *current; i < tags; i++) {
236  if (layout_map[i][2] != pos)
237  break;
238  if (layout_map[i][0] == TYPE_CPE) {
239  if (sce_parity) {
240  if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
241  sce_parity = 0;
242  } else {
243  return -1;
244  }
245  }
246  num_pos_channels += 2;
247  first_cpe = 1;
248  } else {
249  num_pos_channels++;
250  sce_parity ^= 1;
251  }
252  }
253  if (sce_parity &&
254  ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
255  return -1;
256  *current = i;
257  return num_pos_channels;
258 }
259 
260 static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
261 {
262  int i, n, total_non_cc_elements;
263  struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
264  int num_front_channels, num_side_channels, num_back_channels;
265  uint64_t layout;
266 
267  if (FF_ARRAY_ELEMS(e2c_vec) < tags)
268  return 0;
269 
270  i = 0;
271  num_front_channels =
272  count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
273  if (num_front_channels < 0)
274  return 0;
275  num_side_channels =
276  count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
277  if (num_side_channels < 0)
278  return 0;
279  num_back_channels =
280  count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
281  if (num_back_channels < 0)
282  return 0;
283 
284  if (num_side_channels == 0 && num_back_channels >= 4) {
285  num_side_channels = 2;
286  num_back_channels -= 2;
287  }
288 
289  i = 0;
290  if (num_front_channels & 1) {
291  e2c_vec[i] = (struct elem_to_channel) {
293  .syn_ele = TYPE_SCE,
294  .elem_id = layout_map[i][1],
295  .aac_position = AAC_CHANNEL_FRONT
296  };
297  i++;
298  num_front_channels--;
299  }
300  if (num_front_channels >= 4) {
301  i += assign_pair(e2c_vec, layout_map, i,
305  num_front_channels -= 2;
306  }
307  if (num_front_channels >= 2) {
308  i += assign_pair(e2c_vec, layout_map, i,
312  num_front_channels -= 2;
313  }
314  while (num_front_channels >= 2) {
315  i += assign_pair(e2c_vec, layout_map, i,
316  UINT64_MAX,
317  UINT64_MAX,
319  num_front_channels -= 2;
320  }
321 
322  if (num_side_channels >= 2) {
323  i += assign_pair(e2c_vec, layout_map, i,
327  num_side_channels -= 2;
328  }
329  while (num_side_channels >= 2) {
330  i += assign_pair(e2c_vec, layout_map, i,
331  UINT64_MAX,
332  UINT64_MAX,
334  num_side_channels -= 2;
335  }
336 
337  while (num_back_channels >= 4) {
338  i += assign_pair(e2c_vec, layout_map, i,
339  UINT64_MAX,
340  UINT64_MAX,
342  num_back_channels -= 2;
343  }
344  if (num_back_channels >= 2) {
345  i += assign_pair(e2c_vec, layout_map, i,
349  num_back_channels -= 2;
350  }
351  if (num_back_channels) {
352  e2c_vec[i] = (struct elem_to_channel) {
354  .syn_ele = TYPE_SCE,
355  .elem_id = layout_map[i][1],
356  .aac_position = AAC_CHANNEL_BACK
357  };
358  i++;
359  num_back_channels--;
360  }
361 
362  if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
363  e2c_vec[i] = (struct elem_to_channel) {
365  .syn_ele = TYPE_LFE,
366  .elem_id = layout_map[i][1],
367  .aac_position = AAC_CHANNEL_LFE
368  };
369  i++;
370  }
371  while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
372  e2c_vec[i] = (struct elem_to_channel) {
373  .av_position = UINT64_MAX,
374  .syn_ele = TYPE_LFE,
375  .elem_id = layout_map[i][1],
376  .aac_position = AAC_CHANNEL_LFE
377  };
378  i++;
379  }
380 
381  // Must choose a stable sort
382  total_non_cc_elements = n = i;
383  do {
384  int next_n = 0;
385  for (i = 1; i < n; i++)
386  if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
387  FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
388  next_n = i;
389  }
390  n = next_n;
391  } while (n > 0);
392 
393  layout = 0;
394  for (i = 0; i < total_non_cc_elements; i++) {
395  layout_map[i][0] = e2c_vec[i].syn_ele;
396  layout_map[i][1] = e2c_vec[i].elem_id;
397  layout_map[i][2] = e2c_vec[i].aac_position;
398  if (e2c_vec[i].av_position != UINT64_MAX) {
399  layout |= e2c_vec[i].av_position;
400  }
401  }
402 
403  return layout;
404 }
405 
406 /**
407  * Save current output configuration if and only if it has been locked.
408  */
410  int pushed = 0;
411 
412  if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
413  ac->oc[0] = ac->oc[1];
414  pushed = 1;
415  }
416  ac->oc[1].status = OC_NONE;
417  return pushed;
418 }
419 
420 /**
421  * Restore the previous output configuration if and only if the current
422  * configuration is unlocked.
423  */
425  if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
426  ac->oc[1] = ac->oc[0];
427  ac->avctx->channels = ac->oc[1].channels;
428  ac->avctx->channel_layout = ac->oc[1].channel_layout;
429  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
430  ac->oc[1].status, 0);
431  }
432 }
433 
434 /**
435  * Configure output channel order based on the current program
436  * configuration element.
437  *
438  * @return Returns error status. 0 - OK, !0 - error
439  */
441  uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
442  enum OCStatus oc_type, int get_new_frame)
443 {
444  AVCodecContext *avctx = ac->avctx;
445  int i, channels = 0, ret;
446  uint64_t layout = 0;
447  uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
448  uint8_t type_counts[TYPE_END] = { 0 };
449 
450  if (ac->oc[1].layout_map != layout_map) {
451  memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
452  ac->oc[1].layout_map_tags = tags;
453  }
454  for (i = 0; i < tags; i++) {
455  int type = layout_map[i][0];
456  int id = layout_map[i][1];
457  id_map[type][id] = type_counts[type]++;
458  if (id_map[type][id] >= MAX_ELEM_ID) {
459  avpriv_request_sample(ac->avctx, "Too large remapped id");
460  return AVERROR_PATCHWELCOME;
461  }
462  }
463  // Try to sniff a reasonable channel order, otherwise output the
464  // channels in the order the PCE declared them.
466  layout = sniff_channel_order(layout_map, tags);
467  for (i = 0; i < tags; i++) {
468  int type = layout_map[i][0];
469  int id = layout_map[i][1];
470  int iid = id_map[type][id];
471  int position = layout_map[i][2];
472  // Allocate or free elements depending on if they are in the
473  // current program configuration.
474  ret = che_configure(ac, position, type, iid, &channels);
475  if (ret < 0)
476  return ret;
477  ac->tag_che_map[type][id] = ac->che[type][iid];
478  }
479  if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
480  if (layout == AV_CH_FRONT_CENTER) {
482  } else {
483  layout = 0;
484  }
485  }
486 
487  if (layout) avctx->channel_layout = layout;
488  ac->oc[1].channel_layout = layout;
489  avctx->channels = ac->oc[1].channels = channels;
490  ac->oc[1].status = oc_type;
491 
492  if (get_new_frame) {
493  if ((ret = frame_configure_elements(ac->avctx)) < 0)
494  return ret;
495  }
496 
497  return 0;
498 }
499 
500 static void flush(AVCodecContext *avctx)
501 {
502  AACContext *ac= avctx->priv_data;
503  int type, i, j;
504 
505  for (type = 3; type >= 0; type--) {
506  for (i = 0; i < MAX_ELEM_ID; i++) {
507  ChannelElement *che = ac->che[type][i];
508  if (che) {
509  for (j = 0; j <= 1; j++) {
510  memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
511  }
512  }
513  }
514  }
515 }
516 
517 /**
518  * Set up channel positions based on a default channel configuration
519  * as specified in table 1.17.
520  *
521  * @return Returns error status. 0 - OK, !0 - error
522  */
524  uint8_t (*layout_map)[3],
525  int *tags,
526  int channel_config)
527 {
528  if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
529  channel_config > 12) {
530  av_log(avctx, AV_LOG_ERROR,
531  "invalid default channel configuration (%d)\n",
532  channel_config);
533  return AVERROR_INVALIDDATA;
534  }
535  *tags = tags_per_config[channel_config];
536  memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
537  *tags * sizeof(*layout_map));
538 
539  /*
540  * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
541  * However, at least Nero AAC encoder encodes 7.1 streams using the default
542  * channel config 7, mapping the side channels of the original audio stream
543  * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
544  * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
545  * the incorrect streams as if they were correct (and as the encoder intended).
546  *
547  * As actual intended 7.1(wide) streams are very rare, default to assuming a
548  * 7.1 layout was intended.
549  */
550  if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
551  av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
552  " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
553  " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
554  layout_map[2][2] = AAC_CHANNEL_SIDE;
555  }
556 
557  return 0;
558 }
559 
560 static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
561 {
562  /* For PCE based channel configurations map the channels solely based
563  * on tags. */
564  if (!ac->oc[1].m4ac.chan_config) {
565  return ac->tag_che_map[type][elem_id];
566  }
567  // Allow single CPE stereo files to be signalled with mono configuration.
568  if (!ac->tags_mapped && type == TYPE_CPE &&
569  ac->oc[1].m4ac.chan_config == 1) {
570  uint8_t layout_map[MAX_ELEM_ID*4][3];
571  int layout_map_tags;
573 
574  av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
575 
576  if (set_default_channel_config(ac->avctx, layout_map,
577  &layout_map_tags, 2) < 0)
578  return NULL;
579  if (output_configure(ac, layout_map, layout_map_tags,
580  OC_TRIAL_FRAME, 1) < 0)
581  return NULL;
582 
583  ac->oc[1].m4ac.chan_config = 2;
584  ac->oc[1].m4ac.ps = 0;
585  }
586  // And vice-versa
587  if (!ac->tags_mapped && type == TYPE_SCE &&
588  ac->oc[1].m4ac.chan_config == 2) {
589  uint8_t layout_map[MAX_ELEM_ID * 4][3];
590  int layout_map_tags;
592 
593  av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
594 
595  if (set_default_channel_config(ac->avctx, layout_map,
596  &layout_map_tags, 1) < 0)
597  return NULL;
598  if (output_configure(ac, layout_map, layout_map_tags,
599  OC_TRIAL_FRAME, 1) < 0)
600  return NULL;
601 
602  ac->oc[1].m4ac.chan_config = 1;
603  if (ac->oc[1].m4ac.sbr)
604  ac->oc[1].m4ac.ps = -1;
605  }
606  /* For indexed channel configurations map the channels solely based
607  * on position. */
608  switch (ac->oc[1].m4ac.chan_config) {
609  case 12:
610  case 7:
611  if (ac->tags_mapped == 3 && type == TYPE_CPE) {
612  ac->tags_mapped++;
613  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
614  }
615  case 11:
616  if (ac->tags_mapped == 2 &&
617  ac->oc[1].m4ac.chan_config == 11 &&
618  type == TYPE_SCE) {
619  ac->tags_mapped++;
620  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
621  }
622  case 6:
623  /* Some streams incorrectly code 5.1 audio as
624  * SCE[0] CPE[0] CPE[1] SCE[1]
625  * instead of
626  * SCE[0] CPE[0] CPE[1] LFE[0].
627  * If we seem to have encountered such a stream, transfer
628  * the LFE[0] element to the SCE[1]'s mapping */
629  if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
630  if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
632  "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
633  type == TYPE_SCE ? "SCE" : "LFE", elem_id);
634  ac->warned_remapping_once++;
635  }
636  ac->tags_mapped++;
637  return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
638  }
639  case 5:
640  if (ac->tags_mapped == 2 && type == TYPE_CPE) {
641  ac->tags_mapped++;
642  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
643  }
644  case 4:
645  /* Some streams incorrectly code 4.0 audio as
646  * SCE[0] CPE[0] LFE[0]
647  * instead of
648  * SCE[0] CPE[0] SCE[1].
649  * If we seem to have encountered such a stream, transfer
650  * the SCE[1] element to the LFE[0]'s mapping */
651  if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
652  if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
654  "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
655  type == TYPE_SCE ? "SCE" : "LFE", elem_id);
656  ac->warned_remapping_once++;
657  }
658  ac->tags_mapped++;
659  return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
660  }
661  if (ac->tags_mapped == 2 &&
662  ac->oc[1].m4ac.chan_config == 4 &&
663  type == TYPE_SCE) {
664  ac->tags_mapped++;
665  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
666  }
667  case 3:
668  case 2:
669  if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
670  type == TYPE_CPE) {
671  ac->tags_mapped++;
672  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
673  } else if (ac->oc[1].m4ac.chan_config == 2) {
674  return NULL;
675  }
676  case 1:
677  if (!ac->tags_mapped && type == TYPE_SCE) {
678  ac->tags_mapped++;
679  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
680  }
681  default:
682  return NULL;
683  }
684 }
685 
686 /**
687  * Decode an array of 4 bit element IDs, optionally interleaved with a
688  * stereo/mono switching bit.
689  *
690  * @param type speaker type/position for these channels
691  */
692 static void decode_channel_map(uint8_t layout_map[][3],
693  enum ChannelPosition type,
694  GetBitContext *gb, int n)
695 {
696  while (n--) {
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_LTP:
1001  case AOT_ER_AAC_LC:
1002  case AOT_ER_AAC_LD:
1003  if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment,
1004  m4ac, m4ac->chan_config)) < 0)
1005  return ret;
1006  break;
1007  case AOT_ER_AAC_ELD:
1008  if ((ret = decode_eld_specific_config(ac, avctx, gb,
1009  m4ac, m4ac->chan_config)) < 0)
1010  return ret;
1011  break;
1012  default:
1014  "Audio object type %s%d",
1015  m4ac->sbr == 1 ? "SBR+" : "",
1016  m4ac->object_type);
1017  return AVERROR(ENOSYS);
1018  }
1019 
1020  ff_dlog(avctx,
1021  "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
1022  m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
1023  m4ac->sample_rate, m4ac->sbr,
1024  m4ac->ps);
1025 
1026  return get_bits_count(gb);
1027 }
1028 
1030  AVCodecContext *avctx,
1031  MPEG4AudioConfig *m4ac,
1032  const uint8_t *data, int64_t bit_size,
1033  int sync_extension)
1034 {
1035  int i, ret;
1036  GetBitContext gb;
1037 
1038  if (bit_size < 0 || bit_size > INT_MAX) {
1039  av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n");
1040  return AVERROR_INVALIDDATA;
1041  }
1042 
1043  ff_dlog(avctx, "audio specific config size %d\n", (int)bit_size >> 3);
1044  for (i = 0; i < bit_size >> 3; i++)
1045  ff_dlog(avctx, "%02x ", data[i]);
1046  ff_dlog(avctx, "\n");
1047 
1048  if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
1049  return ret;
1050 
1051  return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0,
1052  sync_extension);
1053 }
1054 
1055 /**
1056  * linear congruential pseudorandom number generator
1057  *
1058  * @param previous_val pointer to the current state of the generator
1059  *
1060  * @return Returns a 32-bit pseudorandom integer
1061  */
1062 static av_always_inline int lcg_random(unsigned previous_val)
1063 {
1064  union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
1065  return v.s;
1066 }
1067 
1069 {
1070  int i;
1071  for (i = 0; i < MAX_PREDICTORS; i++)
1072  reset_predict_state(&ps[i]);
1073 }
1074 
1075 static int sample_rate_idx (int rate)
1076 {
1077  if (92017 <= rate) return 0;
1078  else if (75132 <= rate) return 1;
1079  else if (55426 <= rate) return 2;
1080  else if (46009 <= rate) return 3;
1081  else if (37566 <= rate) return 4;
1082  else if (27713 <= rate) return 5;
1083  else if (23004 <= rate) return 6;
1084  else if (18783 <= rate) return 7;
1085  else if (13856 <= rate) return 8;
1086  else if (11502 <= rate) return 9;
1087  else if (9391 <= rate) return 10;
1088  else return 11;
1089 }
1090 
1091 static void reset_predictor_group(PredictorState *ps, int group_num)
1092 {
1093  int i;
1094  for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
1095  reset_predict_state(&ps[i]);
1096 }
1097 
1098 #define AAC_INIT_VLC_STATIC(num, size) \
1099  INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
1100  ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \
1101  sizeof(ff_aac_spectral_bits[num][0]), \
1102  ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
1103  sizeof(ff_aac_spectral_codes[num][0]), \
1104  size);
1105 
1106 static void aacdec_init(AACContext *ac);
1107 
1109 {
1110  AAC_INIT_VLC_STATIC( 0, 304);
1111  AAC_INIT_VLC_STATIC( 1, 270);
1112  AAC_INIT_VLC_STATIC( 2, 550);
1113  AAC_INIT_VLC_STATIC( 3, 300);
1114  AAC_INIT_VLC_STATIC( 4, 328);
1115  AAC_INIT_VLC_STATIC( 5, 294);
1116  AAC_INIT_VLC_STATIC( 6, 306);
1117  AAC_INIT_VLC_STATIC( 7, 268);
1118  AAC_INIT_VLC_STATIC( 8, 510);
1119  AAC_INIT_VLC_STATIC( 9, 366);
1120  AAC_INIT_VLC_STATIC(10, 462);
1121 
1123 
1124  ff_aac_tableinit();
1125 
1126  INIT_VLC_STATIC(&vlc_scalefactors, 7,
1129  sizeof(ff_aac_scalefactor_bits[0]),
1130  sizeof(ff_aac_scalefactor_bits[0]),
1132  sizeof(ff_aac_scalefactor_code[0]),
1133  sizeof(ff_aac_scalefactor_code[0]),
1134  352);
1135 
1136  // window initialization
1139 #if !USE_FIXED
1142  AAC_RENAME(ff_sine_window_init)(AAC_RENAME(ff_sine_960), 960);
1143  AAC_RENAME(ff_sine_window_init)(AAC_RENAME(ff_sine_120), 120);
1144 #endif
1148 
1150 }
1151 
1153 
1155 {
1156  AACContext *ac = avctx->priv_data;
1157  int ret;
1158 
1159  if (avctx->sample_rate > 96000)
1160  return AVERROR_INVALIDDATA;
1161 
1162  ret = ff_thread_once(&aac_table_init, &aac_static_table_init);
1163  if (ret != 0)
1164  return AVERROR_UNKNOWN;
1165 
1166  ac->avctx = avctx;
1167  ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
1168 
1169  aacdec_init(ac);
1170 #if USE_FIXED
1171  avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
1172 #else
1173  avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1174 #endif /* USE_FIXED */
1175 
1176  if (avctx->extradata_size > 0) {
1177  if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
1178  avctx->extradata,
1179  avctx->extradata_size * 8LL,
1180  1)) < 0)
1181  return ret;
1182  } else {
1183  int sr, i;
1184  uint8_t layout_map[MAX_ELEM_ID*4][3];
1185  int layout_map_tags;
1186 
1187  sr = sample_rate_idx(avctx->sample_rate);
1188  ac->oc[1].m4ac.sampling_index = sr;
1189  ac->oc[1].m4ac.channels = avctx->channels;
1190  ac->oc[1].m4ac.sbr = -1;
1191  ac->oc[1].m4ac.ps = -1;
1192 
1193  for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
1194  if (ff_mpeg4audio_channels[i] == avctx->channels)
1195  break;
1197  i = 0;
1198  }
1199  ac->oc[1].m4ac.chan_config = i;
1200 
1201  if (ac->oc[1].m4ac.chan_config) {
1202  int ret = set_default_channel_config(avctx, layout_map,
1203  &layout_map_tags, ac->oc[1].m4ac.chan_config);
1204  if (!ret)
1205  output_configure(ac, layout_map, layout_map_tags,
1206  OC_GLOBAL_HDR, 0);
1207  else if (avctx->err_recognition & AV_EF_EXPLODE)
1208  return AVERROR_INVALIDDATA;
1209  }
1210  }
1211 
1212  if (avctx->channels > MAX_CHANNELS) {
1213  av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
1214  return AVERROR_INVALIDDATA;
1215  }
1216 
1217 #if USE_FIXED
1219 #else
1221 #endif /* USE_FIXED */
1222  if (!ac->fdsp) {
1223  return AVERROR(ENOMEM);
1224  }
1225 
1226  ac->random_state = 0x1f2e3d4c;
1227 
1228  AAC_RENAME_32(ff_mdct_init)(&ac->mdct, 11, 1, 1.0 / RANGE15(1024.0));
1229  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld, 10, 1, 1.0 / RANGE15(512.0));
1230  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small, 8, 1, 1.0 / RANGE15(128.0));
1231  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp, 11, 0, RANGE15(-2.0));
1232 #if !USE_FIXED
1233  ret = ff_mdct15_init(&ac->mdct120, 1, 3, 1.0f/(16*1024*120*2));
1234  if (ret < 0)
1235  return ret;
1236  ret = ff_mdct15_init(&ac->mdct480, 1, 5, 1.0f/(16*1024*960));
1237  if (ret < 0)
1238  return ret;
1239  ret = ff_mdct15_init(&ac->mdct960, 1, 6, 1.0f/(16*1024*960*2));
1240  if (ret < 0)
1241  return ret;
1242 #endif
1243 
1244  return 0;
1245 }
1246 
1247 /**
1248  * Skip data_stream_element; reference: table 4.10.
1249  */
1251 {
1252  int byte_align = get_bits1(gb);
1253  int count = get_bits(gb, 8);
1254  if (count == 255)
1255  count += get_bits(gb, 8);
1256  if (byte_align)
1257  align_get_bits(gb);
1258 
1259  if (get_bits_left(gb) < 8 * count) {
1260  av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
1261  return AVERROR_INVALIDDATA;
1262  }
1263  skip_bits_long(gb, 8 * count);
1264  return 0;
1265 }
1266 
1268  GetBitContext *gb)
1269 {
1270  int sfb;
1271  if (get_bits1(gb)) {
1272  ics->predictor_reset_group = get_bits(gb, 5);
1273  if (ics->predictor_reset_group == 0 ||
1274  ics->predictor_reset_group > 30) {
1275  av_log(ac->avctx, AV_LOG_ERROR,
1276  "Invalid Predictor Reset Group.\n");
1277  return AVERROR_INVALIDDATA;
1278  }
1279  }
1280  for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
1281  ics->prediction_used[sfb] = get_bits1(gb);
1282  }
1283  return 0;
1284 }
1285 
1286 /**
1287  * Decode Long Term Prediction data; reference: table 4.xx.
1288  */
1290  GetBitContext *gb, uint8_t max_sfb)
1291 {
1292  int sfb;
1293 
1294  ltp->lag = get_bits(gb, 11);
1295  ltp->coef = ltp_coef[get_bits(gb, 3)];
1296  for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
1297  ltp->used[sfb] = get_bits1(gb);
1298 }
1299 
1300 /**
1301  * Decode Individual Channel Stream info; reference: table 4.6.
1302  */
1304  GetBitContext *gb)
1305 {
1306  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
1307  const int aot = m4ac->object_type;
1308  const int sampling_index = m4ac->sampling_index;
1309  int ret_fail = AVERROR_INVALIDDATA;
1310 
1311  if (aot != AOT_ER_AAC_ELD) {
1312  if (get_bits1(gb)) {
1313  av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
1315  return AVERROR_INVALIDDATA;
1316  }
1317  ics->window_sequence[1] = ics->window_sequence[0];
1318  ics->window_sequence[0] = get_bits(gb, 2);
1319  if (aot == AOT_ER_AAC_LD &&
1320  ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
1321  av_log(ac->avctx, AV_LOG_ERROR,
1322  "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
1323  "window sequence %d found.\n", ics->window_sequence[0]);
1325  return AVERROR_INVALIDDATA;
1326  }
1327  ics->use_kb_window[1] = ics->use_kb_window[0];
1328  ics->use_kb_window[0] = get_bits1(gb);
1329  }
1330  ics->num_window_groups = 1;
1331  ics->group_len[0] = 1;
1332  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1333  int i;
1334  ics->max_sfb = get_bits(gb, 4);
1335  for (i = 0; i < 7; i++) {
1336  if (get_bits1(gb)) {
1337  ics->group_len[ics->num_window_groups - 1]++;
1338  } else {
1339  ics->num_window_groups++;
1340  ics->group_len[ics->num_window_groups - 1] = 1;
1341  }
1342  }
1343  ics->num_windows = 8;
1344  if (m4ac->frame_length_short) {
1345  ics->swb_offset = ff_swb_offset_120[sampling_index];
1346  ics->num_swb = ff_aac_num_swb_120[sampling_index];
1347  } else {
1348  ics->swb_offset = ff_swb_offset_128[sampling_index];
1349  ics->num_swb = ff_aac_num_swb_128[sampling_index];
1350  }
1351  ics->tns_max_bands = ff_tns_max_bands_128[sampling_index];
1352  ics->predictor_present = 0;
1353  } else {
1354  ics->max_sfb = get_bits(gb, 6);
1355  ics->num_windows = 1;
1356  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
1357  if (m4ac->frame_length_short) {
1358  ics->swb_offset = ff_swb_offset_480[sampling_index];
1359  ics->num_swb = ff_aac_num_swb_480[sampling_index];
1360  ics->tns_max_bands = ff_tns_max_bands_480[sampling_index];
1361  } else {
1362  ics->swb_offset = ff_swb_offset_512[sampling_index];
1363  ics->num_swb = ff_aac_num_swb_512[sampling_index];
1364  ics->tns_max_bands = ff_tns_max_bands_512[sampling_index];
1365  }
1366  if (!ics->num_swb || !ics->swb_offset) {
1367  ret_fail = AVERROR_BUG;
1368  goto fail;
1369  }
1370  } else {
1371  if (m4ac->frame_length_short) {
1372  ics->num_swb = ff_aac_num_swb_960[sampling_index];
1373  ics->swb_offset = ff_swb_offset_960[sampling_index];
1374  } else {
1375  ics->num_swb = ff_aac_num_swb_1024[sampling_index];
1376  ics->swb_offset = ff_swb_offset_1024[sampling_index];
1377  }
1378  ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
1379  }
1380  if (aot != AOT_ER_AAC_ELD) {
1381  ics->predictor_present = get_bits1(gb);
1382  ics->predictor_reset_group = 0;
1383  }
1384  if (ics->predictor_present) {
1385  if (aot == AOT_AAC_MAIN) {
1386  if (decode_prediction(ac, ics, gb)) {
1387  goto fail;
1388  }
1389  } else if (aot == AOT_AAC_LC ||
1390  aot == AOT_ER_AAC_LC) {
1391  av_log(ac->avctx, AV_LOG_ERROR,
1392  "Prediction is not allowed in AAC-LC.\n");
1393  goto fail;
1394  } else {
1395  if (aot == AOT_ER_AAC_LD) {
1396  av_log(ac->avctx, AV_LOG_ERROR,
1397  "LTP in ER AAC LD not yet implemented.\n");
1398  ret_fail = AVERROR_PATCHWELCOME;
1399  goto fail;
1400  }
1401  if ((ics->ltp.present = get_bits(gb, 1)))
1402  decode_ltp(&ics->ltp, gb, ics->max_sfb);
1403  }
1404  }
1405  }
1406 
1407  if (ics->max_sfb > ics->num_swb) {
1408  av_log(ac->avctx, AV_LOG_ERROR,
1409  "Number of scalefactor bands in group (%d) "
1410  "exceeds limit (%d).\n",
1411  ics->max_sfb, ics->num_swb);
1412  goto fail;
1413  }
1414 
1415  return 0;
1416 fail:
1417  ics->max_sfb = 0;
1418  return ret_fail;
1419 }
1420 
1421 /**
1422  * Decode band types (section_data payload); reference: table 4.46.
1423  *
1424  * @param band_type array of the used band type
1425  * @param band_type_run_end array of the last scalefactor band of a band type run
1426  *
1427  * @return Returns error status. 0 - OK, !0 - error
1428  */
1429 static int decode_band_types(AACContext *ac, enum BandType band_type[120],
1430  int band_type_run_end[120], GetBitContext *gb,
1432 {
1433  int g, idx = 0;
1434  const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
1435  for (g = 0; g < ics->num_window_groups; g++) {
1436  int k = 0;
1437  while (k < ics->max_sfb) {
1438  uint8_t sect_end = k;
1439  int sect_len_incr;
1440  int sect_band_type = get_bits(gb, 4);
1441  if (sect_band_type == 12) {
1442  av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
1443  return AVERROR_INVALIDDATA;
1444  }
1445  do {
1446  sect_len_incr = get_bits(gb, bits);
1447  sect_end += sect_len_incr;
1448  if (get_bits_left(gb) < 0) {
1449  av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
1450  return AVERROR_INVALIDDATA;
1451  }
1452  if (sect_end > ics->max_sfb) {
1453  av_log(ac->avctx, AV_LOG_ERROR,
1454  "Number of bands (%d) exceeds limit (%d).\n",
1455  sect_end, ics->max_sfb);
1456  return AVERROR_INVALIDDATA;
1457  }
1458  } while (sect_len_incr == (1 << bits) - 1);
1459  for (; k < sect_end; k++) {
1460  band_type [idx] = sect_band_type;
1461  band_type_run_end[idx++] = sect_end;
1462  }
1463  }
1464  }
1465  return 0;
1466 }
1467 
1468 /**
1469  * Decode scalefactors; reference: table 4.47.
1470  *
1471  * @param global_gain first scalefactor value as scalefactors are differentially coded
1472  * @param band_type array of the used band type
1473  * @param band_type_run_end array of the last scalefactor band of a band type run
1474  * @param sf array of scalefactors or intensity stereo positions
1475  *
1476  * @return Returns error status. 0 - OK, !0 - error
1477  */
1479  unsigned int global_gain,
1481  enum BandType band_type[120],
1482  int band_type_run_end[120])
1483 {
1484  int g, i, idx = 0;
1485  int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 };
1486  int clipped_offset;
1487  int noise_flag = 1;
1488  for (g = 0; g < ics->num_window_groups; g++) {
1489  for (i = 0; i < ics->max_sfb;) {
1490  int run_end = band_type_run_end[idx];
1491  if (band_type[idx] == ZERO_BT) {
1492  for (; i < run_end; i++, idx++)
1493  sf[idx] = FIXR(0.);
1494  } else if ((band_type[idx] == INTENSITY_BT) ||
1495  (band_type[idx] == INTENSITY_BT2)) {
1496  for (; i < run_end; i++, idx++) {
1497  offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1498  clipped_offset = av_clip(offset[2], -155, 100);
1499  if (offset[2] != clipped_offset) {
1501  "If you heard an audible artifact, there may be a bug in the decoder. "
1502  "Clipped intensity stereo position (%d -> %d)",
1503  offset[2], clipped_offset);
1504  }
1505 #if USE_FIXED
1506  sf[idx] = 100 - clipped_offset;
1507 #else
1508  sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
1509 #endif /* USE_FIXED */
1510  }
1511  } else if (band_type[idx] == NOISE_BT) {
1512  for (; i < run_end; i++, idx++) {
1513  if (noise_flag-- > 0)
1514  offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
1515  else
1516  offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1517  clipped_offset = av_clip(offset[1], -100, 155);
1518  if (offset[1] != clipped_offset) {
1520  "If you heard an audible artifact, there may be a bug in the decoder. "
1521  "Clipped noise gain (%d -> %d)",
1522  offset[1], clipped_offset);
1523  }
1524 #if USE_FIXED
1525  sf[idx] = -(100 + clipped_offset);
1526 #else
1527  sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
1528 #endif /* USE_FIXED */
1529  }
1530  } else {
1531  for (; i < run_end; i++, idx++) {
1532  offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1533  if (offset[0] > 255U) {
1534  av_log(ac->avctx, AV_LOG_ERROR,
1535  "Scalefactor (%d) out of range.\n", offset[0]);
1536  return AVERROR_INVALIDDATA;
1537  }
1538 #if USE_FIXED
1539  sf[idx] = -offset[0];
1540 #else
1541  sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
1542 #endif /* USE_FIXED */
1543  }
1544  }
1545  }
1546  }
1547  return 0;
1548 }
1549 
1550 /**
1551  * Decode pulse data; reference: table 4.7.
1552  */
1553 static int decode_pulses(Pulse *pulse, GetBitContext *gb,
1554  const uint16_t *swb_offset, int num_swb)
1555 {
1556  int i, pulse_swb;
1557  pulse->num_pulse = get_bits(gb, 2) + 1;
1558  pulse_swb = get_bits(gb, 6);
1559  if (pulse_swb >= num_swb)
1560  return -1;
1561  pulse->pos[0] = swb_offset[pulse_swb];
1562  pulse->pos[0] += get_bits(gb, 5);
1563  if (pulse->pos[0] >= swb_offset[num_swb])
1564  return -1;
1565  pulse->amp[0] = get_bits(gb, 4);
1566  for (i = 1; i < pulse->num_pulse; i++) {
1567  pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
1568  if (pulse->pos[i] >= swb_offset[num_swb])
1569  return -1;
1570  pulse->amp[i] = get_bits(gb, 4);
1571  }
1572  return 0;
1573 }
1574 
1575 /**
1576  * Decode Temporal Noise Shaping data; reference: table 4.48.
1577  *
1578  * @return Returns error status. 0 - OK, !0 - error
1579  */
1581  GetBitContext *gb, const IndividualChannelStream *ics)
1582 {
1583  int w, filt, i, coef_len, coef_res, coef_compress;
1584  const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
1585  const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
1586  for (w = 0; w < ics->num_windows; w++) {
1587  if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
1588  coef_res = get_bits1(gb);
1589 
1590  for (filt = 0; filt < tns->n_filt[w]; filt++) {
1591  int tmp2_idx;
1592  tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
1593 
1594  if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
1595  av_log(ac->avctx, AV_LOG_ERROR,
1596  "TNS filter order %d is greater than maximum %d.\n",
1597  tns->order[w][filt], tns_max_order);
1598  tns->order[w][filt] = 0;
1599  return AVERROR_INVALIDDATA;
1600  }
1601  if (tns->order[w][filt]) {
1602  tns->direction[w][filt] = get_bits1(gb);
1603  coef_compress = get_bits1(gb);
1604  coef_len = coef_res + 3 - coef_compress;
1605  tmp2_idx = 2 * coef_compress + coef_res;
1606 
1607  for (i = 0; i < tns->order[w][filt]; i++)
1608  tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
1609  }
1610  }
1611  }
1612  }
1613  return 0;
1614 }
1615 
1616 /**
1617  * Decode Mid/Side data; reference: table 4.54.
1618  *
1619  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
1620  * [1] mask is decoded from bitstream; [2] mask is all 1s;
1621  * [3] reserved for scalable AAC
1622  */
1624  int ms_present)
1625 {
1626  int idx;
1627  int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
1628  if (ms_present == 1) {
1629  for (idx = 0; idx < max_idx; idx++)
1630  cpe->ms_mask[idx] = get_bits1(gb);
1631  } else if (ms_present == 2) {
1632  memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
1633  }
1634 }
1635 
1636 /**
1637  * Decode spectral data; reference: table 4.50.
1638  * Dequantize and scale spectral data; reference: 4.6.3.3.
1639  *
1640  * @param coef array of dequantized, scaled spectral data
1641  * @param sf array of scalefactors or intensity stereo positions
1642  * @param pulse_present set if pulses are present
1643  * @param pulse pointer to pulse data struct
1644  * @param band_type array of the used band type
1645  *
1646  * @return Returns error status. 0 - OK, !0 - error
1647  */
1649  GetBitContext *gb, const INTFLOAT sf[120],
1650  int pulse_present, const Pulse *pulse,
1651  const IndividualChannelStream *ics,
1652  enum BandType band_type[120])
1653 {
1654  int i, k, g, idx = 0;
1655  const int c = 1024 / ics->num_windows;
1656  const uint16_t *offsets = ics->swb_offset;
1657  INTFLOAT *coef_base = coef;
1658 
1659  for (g = 0; g < ics->num_windows; g++)
1660  memset(coef + g * 128 + offsets[ics->max_sfb], 0,
1661  sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
1662 
1663  for (g = 0; g < ics->num_window_groups; g++) {
1664  unsigned g_len = ics->group_len[g];
1665 
1666  for (i = 0; i < ics->max_sfb; i++, idx++) {
1667  const unsigned cbt_m1 = band_type[idx] - 1;
1668  INTFLOAT *cfo = coef + offsets[i];
1669  int off_len = offsets[i + 1] - offsets[i];
1670  int group;
1671 
1672  if (cbt_m1 >= INTENSITY_BT2 - 1) {
1673  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1674  memset(cfo, 0, off_len * sizeof(*cfo));
1675  }
1676  } else if (cbt_m1 == NOISE_BT - 1) {
1677  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1678  INTFLOAT band_energy;
1679 #if USE_FIXED
1680  for (k = 0; k < off_len; k++) {
1682  cfo[k] = ac->random_state >> 3;
1683  }
1684 
1685  band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
1686  band_energy = fixed_sqrt(band_energy, 31);
1687  noise_scale(cfo, sf[idx], band_energy, off_len);
1688 #else
1689  float scale;
1690 
1691  for (k = 0; k < off_len; k++) {
1693  cfo[k] = ac->random_state;
1694  }
1695 
1696  band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
1697  scale = sf[idx] / sqrtf(band_energy);
1698  ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
1699 #endif /* USE_FIXED */
1700  }
1701  } else {
1702 #if !USE_FIXED
1703  const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1704 #endif /* !USE_FIXED */
1705  const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1706  VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1707  OPEN_READER(re, gb);
1708 
1709  switch (cbt_m1 >> 1) {
1710  case 0:
1711  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1712  INTFLOAT *cf = cfo;
1713  int len = off_len;
1714 
1715  do {
1716  int code;
1717  unsigned cb_idx;
1718 
1719  UPDATE_CACHE(re, gb);
1720  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1721  cb_idx = cb_vector_idx[code];
1722 #if USE_FIXED
1723  cf = DEC_SQUAD(cf, cb_idx);
1724 #else
1725  cf = VMUL4(cf, vq, cb_idx, sf + idx);
1726 #endif /* USE_FIXED */
1727  } while (len -= 4);
1728  }
1729  break;
1730 
1731  case 1:
1732  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1733  INTFLOAT *cf = cfo;
1734  int len = off_len;
1735 
1736  do {
1737  int code;
1738  unsigned nnz;
1739  unsigned cb_idx;
1740  uint32_t bits;
1741 
1742  UPDATE_CACHE(re, gb);
1743  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1744  cb_idx = cb_vector_idx[code];
1745  nnz = cb_idx >> 8 & 15;
1746  bits = nnz ? GET_CACHE(re, gb) : 0;
1747  LAST_SKIP_BITS(re, gb, nnz);
1748 #if USE_FIXED
1749  cf = DEC_UQUAD(cf, cb_idx, bits);
1750 #else
1751  cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1752 #endif /* USE_FIXED */
1753  } while (len -= 4);
1754  }
1755  break;
1756 
1757  case 2:
1758  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1759  INTFLOAT *cf = cfo;
1760  int len = off_len;
1761 
1762  do {
1763  int code;
1764  unsigned cb_idx;
1765 
1766  UPDATE_CACHE(re, gb);
1767  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1768  cb_idx = cb_vector_idx[code];
1769 #if USE_FIXED
1770  cf = DEC_SPAIR(cf, cb_idx);
1771 #else
1772  cf = VMUL2(cf, vq, cb_idx, sf + idx);
1773 #endif /* USE_FIXED */
1774  } while (len -= 2);
1775  }
1776  break;
1777 
1778  case 3:
1779  case 4:
1780  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1781  INTFLOAT *cf = cfo;
1782  int len = off_len;
1783 
1784  do {
1785  int code;
1786  unsigned nnz;
1787  unsigned cb_idx;
1788  unsigned sign;
1789 
1790  UPDATE_CACHE(re, gb);
1791  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1792  cb_idx = cb_vector_idx[code];
1793  nnz = cb_idx >> 8 & 15;
1794  sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
1795  LAST_SKIP_BITS(re, gb, nnz);
1796 #if USE_FIXED
1797  cf = DEC_UPAIR(cf, cb_idx, sign);
1798 #else
1799  cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1800 #endif /* USE_FIXED */
1801  } while (len -= 2);
1802  }
1803  break;
1804 
1805  default:
1806  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1807 #if USE_FIXED
1808  int *icf = cfo;
1809  int v;
1810 #else
1811  float *cf = cfo;
1812  uint32_t *icf = (uint32_t *) cf;
1813 #endif /* USE_FIXED */
1814  int len = off_len;
1815 
1816  do {
1817  int code;
1818  unsigned nzt, nnz;
1819  unsigned cb_idx;
1820  uint32_t bits;
1821  int j;
1822 
1823  UPDATE_CACHE(re, gb);
1824  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1825 
1826  if (!code) {
1827  *icf++ = 0;
1828  *icf++ = 0;
1829  continue;
1830  }
1831 
1832  cb_idx = cb_vector_idx[code];
1833  nnz = cb_idx >> 12;
1834  nzt = cb_idx >> 8;
1835  bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1836  LAST_SKIP_BITS(re, gb, nnz);
1837 
1838  for (j = 0; j < 2; j++) {
1839  if (nzt & 1<<j) {
1840  uint32_t b;
1841  int n;
1842  /* The total length of escape_sequence must be < 22 bits according
1843  to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1844  UPDATE_CACHE(re, gb);
1845  b = GET_CACHE(re, gb);
1846  b = 31 - av_log2(~b);
1847 
1848  if (b > 8) {
1849  av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1850  return AVERROR_INVALIDDATA;
1851  }
1852 
1853  SKIP_BITS(re, gb, b + 1);
1854  b += 4;
1855  n = (1 << b) + SHOW_UBITS(re, gb, b);
1856  LAST_SKIP_BITS(re, gb, b);
1857 #if USE_FIXED
1858  v = n;
1859  if (bits & 1U<<31)
1860  v = -v;
1861  *icf++ = v;
1862 #else
1863  *icf++ = ff_cbrt_tab[n] | (bits & 1U<<31);
1864 #endif /* USE_FIXED */
1865  bits <<= 1;
1866  } else {
1867 #if USE_FIXED
1868  v = cb_idx & 15;
1869  if (bits & 1U<<31)
1870  v = -v;
1871  *icf++ = v;
1872 #else
1873  unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1874  *icf++ = (bits & 1U<<31) | v;
1875 #endif /* USE_FIXED */
1876  bits <<= !!v;
1877  }
1878  cb_idx >>= 4;
1879  }
1880  } while (len -= 2);
1881 #if !USE_FIXED
1882  ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1883 #endif /* !USE_FIXED */
1884  }
1885  }
1886 
1887  CLOSE_READER(re, gb);
1888  }
1889  }
1890  coef += g_len << 7;
1891  }
1892 
1893  if (pulse_present) {
1894  idx = 0;
1895  for (i = 0; i < pulse->num_pulse; i++) {
1896  INTFLOAT co = coef_base[ pulse->pos[i] ];
1897  while (offsets[idx + 1] <= pulse->pos[i])
1898  idx++;
1899  if (band_type[idx] != NOISE_BT && sf[idx]) {
1900  INTFLOAT ico = -pulse->amp[i];
1901 #if USE_FIXED
1902  if (co) {
1903  ico = co + (co > 0 ? -ico : ico);
1904  }
1905  coef_base[ pulse->pos[i] ] = ico;
1906 #else
1907  if (co) {
1908  co /= sf[idx];
1909  ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1910  }
1911  coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1912 #endif /* USE_FIXED */
1913  }
1914  }
1915  }
1916 #if USE_FIXED
1917  coef = coef_base;
1918  idx = 0;
1919  for (g = 0; g < ics->num_window_groups; g++) {
1920  unsigned g_len = ics->group_len[g];
1921 
1922  for (i = 0; i < ics->max_sfb; i++, idx++) {
1923  const unsigned cbt_m1 = band_type[idx] - 1;
1924  int *cfo = coef + offsets[i];
1925  int off_len = offsets[i + 1] - offsets[i];
1926  int group;
1927 
1928  if (cbt_m1 < NOISE_BT - 1) {
1929  for (group = 0; group < (int)g_len; group++, cfo+=128) {
1930  ac->vector_pow43(cfo, off_len);
1931  ac->subband_scale(cfo, cfo, sf[idx], 34, off_len);
1932  }
1933  }
1934  }
1935  coef += g_len << 7;
1936  }
1937 #endif /* USE_FIXED */
1938  return 0;
1939 }
1940 
1941 /**
1942  * Apply AAC-Main style frequency domain prediction.
1943  */
1945 {
1946  int sfb, k;
1947 
1948  if (!sce->ics.predictor_initialized) {
1950  sce->ics.predictor_initialized = 1;
1951  }
1952 
1953  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1954  for (sfb = 0;
1955  sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
1956  sfb++) {
1957  for (k = sce->ics.swb_offset[sfb];
1958  k < sce->ics.swb_offset[sfb + 1];
1959  k++) {
1960  predict(&sce->predictor_state[k], &sce->coeffs[k],
1961  sce->ics.predictor_present &&
1962  sce->ics.prediction_used[sfb]);
1963  }
1964  }
1965  if (sce->ics.predictor_reset_group)
1967  sce->ics.predictor_reset_group);
1968  } else
1970 }
1971 
1972 /**
1973  * Decode an individual_channel_stream payload; reference: table 4.44.
1974  *
1975  * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
1976  * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1977  *
1978  * @return Returns error status. 0 - OK, !0 - error
1979  */
1981  GetBitContext *gb, int common_window, int scale_flag)
1982 {
1983  Pulse pulse;
1984  TemporalNoiseShaping *tns = &sce->tns;
1985  IndividualChannelStream *ics = &sce->ics;
1986  INTFLOAT *out = sce->coeffs;
1987  int global_gain, eld_syntax, er_syntax, pulse_present = 0;
1988  int ret;
1989 
1990  eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1991  er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
1992  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
1993  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
1994  ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1995 
1996  /* This assignment is to silence a GCC warning about the variable being used
1997  * uninitialized when in fact it always is.
1998  */
1999  pulse.num_pulse = 0;
2000 
2001  global_gain = get_bits(gb, 8);
2002 
2003  if (!common_window && !scale_flag) {
2004  ret = decode_ics_info(ac, ics, gb);
2005  if (ret < 0)
2006  goto fail;
2007  }
2008 
2009  if ((ret = decode_band_types(ac, sce->band_type,
2010  sce->band_type_run_end, gb, ics)) < 0)
2011  goto fail;
2012  if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
2013  sce->band_type, sce->band_type_run_end)) < 0)
2014  goto fail;
2015 
2016  pulse_present = 0;
2017  if (!scale_flag) {
2018  if (!eld_syntax && (pulse_present = get_bits1(gb))) {
2019  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2020  av_log(ac->avctx, AV_LOG_ERROR,
2021  "Pulse tool not allowed in eight short sequence.\n");
2022  ret = AVERROR_INVALIDDATA;
2023  goto fail;
2024  }
2025  if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
2026  av_log(ac->avctx, AV_LOG_ERROR,
2027  "Pulse data corrupt or invalid.\n");
2028  ret = AVERROR_INVALIDDATA;
2029  goto fail;
2030  }
2031  }
2032  tns->present = get_bits1(gb);
2033  if (tns->present && !er_syntax) {
2034  ret = decode_tns(ac, tns, gb, ics);
2035  if (ret < 0)
2036  goto fail;
2037  }
2038  if (!eld_syntax && get_bits1(gb)) {
2039  avpriv_request_sample(ac->avctx, "SSR");
2040  ret = AVERROR_PATCHWELCOME;
2041  goto fail;
2042  }
2043  // I see no textual basis in the spec for this occurring after SSR gain
2044  // control, but this is what both reference and real implmentations do
2045  if (tns->present && er_syntax) {
2046  ret = decode_tns(ac, tns, gb, ics);
2047  if (ret < 0)
2048  goto fail;
2049  }
2050  }
2051 
2052  ret = decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
2053  &pulse, ics, sce->band_type);
2054  if (ret < 0)
2055  goto fail;
2056 
2057  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
2058  apply_prediction(ac, sce);
2059 
2060  return 0;
2061 fail:
2062  tns->present = 0;
2063  return ret;
2064 }
2065 
2066 /**
2067  * Mid/Side stereo decoding; reference: 4.6.8.1.3.
2068  */
2070 {
2071  const IndividualChannelStream *ics = &cpe->ch[0].ics;
2072  INTFLOAT *ch0 = cpe->ch[0].coeffs;
2073  INTFLOAT *ch1 = cpe->ch[1].coeffs;
2074  int g, i, group, idx = 0;
2075  const uint16_t *offsets = ics->swb_offset;
2076  for (g = 0; g < ics->num_window_groups; g++) {
2077  for (i = 0; i < ics->max_sfb; i++, idx++) {
2078  if (cpe->ms_mask[idx] &&
2079  cpe->ch[0].band_type[idx] < NOISE_BT &&
2080  cpe->ch[1].band_type[idx] < NOISE_BT) {
2081 #if USE_FIXED
2082  for (group = 0; group < ics->group_len[g]; group++) {
2083  ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
2084  ch1 + group * 128 + offsets[i],
2085  offsets[i+1] - offsets[i]);
2086 #else
2087  for (group = 0; group < ics->group_len[g]; group++) {
2088  ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
2089  ch1 + group * 128 + offsets[i],
2090  offsets[i+1] - offsets[i]);
2091 #endif /* USE_FIXED */
2092  }
2093  }
2094  }
2095  ch0 += ics->group_len[g] * 128;
2096  ch1 += ics->group_len[g] * 128;
2097  }
2098 }
2099 
2100 /**
2101  * intensity stereo decoding; reference: 4.6.8.2.3
2102  *
2103  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
2104  * [1] mask is decoded from bitstream; [2] mask is all 1s;
2105  * [3] reserved for scalable AAC
2106  */
2108  ChannelElement *cpe, int ms_present)
2109 {
2110  const IndividualChannelStream *ics = &cpe->ch[1].ics;
2111  SingleChannelElement *sce1 = &cpe->ch[1];
2112  INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
2113  const uint16_t *offsets = ics->swb_offset;
2114  int g, group, i, idx = 0;
2115  int c;
2116  INTFLOAT scale;
2117  for (g = 0; g < ics->num_window_groups; g++) {
2118  for (i = 0; i < ics->max_sfb;) {
2119  if (sce1->band_type[idx] == INTENSITY_BT ||
2120  sce1->band_type[idx] == INTENSITY_BT2) {
2121  const int bt_run_end = sce1->band_type_run_end[idx];
2122  for (; i < bt_run_end; i++, idx++) {
2123  c = -1 + 2 * (sce1->band_type[idx] - 14);
2124  if (ms_present)
2125  c *= 1 - 2 * cpe->ms_mask[idx];
2126  scale = c * sce1->sf[idx];
2127  for (group = 0; group < ics->group_len[g]; group++)
2128 #if USE_FIXED
2129  ac->subband_scale(coef1 + group * 128 + offsets[i],
2130  coef0 + group * 128 + offsets[i],
2131  scale,
2132  23,
2133  offsets[i + 1] - offsets[i]);
2134 #else
2135  ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
2136  coef0 + group * 128 + offsets[i],
2137  scale,
2138  offsets[i + 1] - offsets[i]);
2139 #endif /* USE_FIXED */
2140  }
2141  } else {
2142  int bt_run_end = sce1->band_type_run_end[idx];
2143  idx += bt_run_end - i;
2144  i = bt_run_end;
2145  }
2146  }
2147  coef0 += ics->group_len[g] * 128;
2148  coef1 += ics->group_len[g] * 128;
2149  }
2150 }
2151 
2152 /**
2153  * Decode a channel_pair_element; reference: table 4.4.
2154  *
2155  * @return Returns error status. 0 - OK, !0 - error
2156  */
2158 {
2159  int i, ret, common_window, ms_present = 0;
2160  int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2161 
2162  common_window = eld_syntax || get_bits1(gb);
2163  if (common_window) {
2164  if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
2165  return AVERROR_INVALIDDATA;
2166  i = cpe->ch[1].ics.use_kb_window[0];
2167  cpe->ch[1].ics = cpe->ch[0].ics;
2168  cpe->ch[1].ics.use_kb_window[1] = i;
2169  if (cpe->ch[1].ics.predictor_present &&
2170  (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
2171  if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
2172  decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
2173  ms_present = get_bits(gb, 2);
2174  if (ms_present == 3) {
2175  av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
2176  return AVERROR_INVALIDDATA;
2177  } else if (ms_present)
2178  decode_mid_side_stereo(cpe, gb, ms_present);
2179  }
2180  if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
2181  return ret;
2182  if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
2183  return ret;
2184 
2185  if (common_window) {
2186  if (ms_present)
2187  apply_mid_side_stereo(ac, cpe);
2188  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
2189  apply_prediction(ac, &cpe->ch[0]);
2190  apply_prediction(ac, &cpe->ch[1]);
2191  }
2192  }
2193 
2194  apply_intensity_stereo(ac, cpe, ms_present);
2195  return 0;
2196 }
2197 
2198 static const float cce_scale[] = {
2199  1.09050773266525765921, //2^(1/8)
2200  1.18920711500272106672, //2^(1/4)
2201  M_SQRT2,
2202  2,
2203 };
2204 
2205 /**
2206  * Decode coupling_channel_element; reference: table 4.8.
2207  *
2208  * @return Returns error status. 0 - OK, !0 - error
2209  */
2211 {
2212  int num_gain = 0;
2213  int c, g, sfb, ret;
2214  int sign;
2215  INTFLOAT scale;
2216  SingleChannelElement *sce = &che->ch[0];
2217  ChannelCoupling *coup = &che->coup;
2218 
2219  coup->coupling_point = 2 * get_bits1(gb);
2220  coup->num_coupled = get_bits(gb, 3);
2221  for (c = 0; c <= coup->num_coupled; c++) {
2222  num_gain++;
2223  coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
2224  coup->id_select[c] = get_bits(gb, 4);
2225  if (coup->type[c] == TYPE_CPE) {
2226  coup->ch_select[c] = get_bits(gb, 2);
2227  if (coup->ch_select[c] == 3)
2228  num_gain++;
2229  } else
2230  coup->ch_select[c] = 2;
2231  }
2232  coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
2233 
2234  sign = get_bits(gb, 1);
2235 #if USE_FIXED
2236  scale = get_bits(gb, 2);
2237 #else
2238  scale = cce_scale[get_bits(gb, 2)];
2239 #endif
2240 
2241  if ((ret = decode_ics(ac, sce, gb, 0, 0)))
2242  return ret;
2243 
2244  for (c = 0; c < num_gain; c++) {
2245  int idx = 0;
2246  int cge = 1;
2247  int gain = 0;
2248  INTFLOAT gain_cache = FIXR10(1.);
2249  if (c) {
2250  cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
2251  gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
2252  gain_cache = GET_GAIN(scale, gain);
2253 #if USE_FIXED
2254  if ((abs(gain_cache)-1024) >> 3 > 30)
2255  return AVERROR(ERANGE);
2256 #endif
2257  }
2258  if (coup->coupling_point == AFTER_IMDCT) {
2259  coup->gain[c][0] = gain_cache;
2260  } else {
2261  for (g = 0; g < sce->ics.num_window_groups; g++) {
2262  for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
2263  if (sce->band_type[idx] != ZERO_BT) {
2264  if (!cge) {
2265  int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
2266  if (t) {
2267  int s = 1;
2268  t = gain += t;
2269  if (sign) {
2270  s -= 2 * (t & 0x1);
2271  t >>= 1;
2272  }
2273  gain_cache = GET_GAIN(scale, t) * s;
2274 #if USE_FIXED
2275  if ((abs(gain_cache)-1024) >> 3 > 30)
2276  return AVERROR(ERANGE);
2277 #endif
2278  }
2279  }
2280  coup->gain[c][idx] = gain_cache;
2281  }
2282  }
2283  }
2284  }
2285  }
2286  return 0;
2287 }
2288 
2289 /**
2290  * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
2291  *
2292  * @return Returns number of bytes consumed.
2293  */
2295  GetBitContext *gb)
2296 {
2297  int i;
2298  int num_excl_chan = 0;
2299 
2300  do {
2301  for (i = 0; i < 7; i++)
2302  che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
2303  } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
2304 
2305  return num_excl_chan / 7;
2306 }
2307 
2308 /**
2309  * Decode dynamic range information; reference: table 4.52.
2310  *
2311  * @return Returns number of bytes consumed.
2312  */
2314  GetBitContext *gb)
2315 {
2316  int n = 1;
2317  int drc_num_bands = 1;
2318  int i;
2319 
2320  /* pce_tag_present? */
2321  if (get_bits1(gb)) {
2322  che_drc->pce_instance_tag = get_bits(gb, 4);
2323  skip_bits(gb, 4); // tag_reserved_bits
2324  n++;
2325  }
2326 
2327  /* excluded_chns_present? */
2328  if (get_bits1(gb)) {
2329  n += decode_drc_channel_exclusions(che_drc, gb);
2330  }
2331 
2332  /* drc_bands_present? */
2333  if (get_bits1(gb)) {
2334  che_drc->band_incr = get_bits(gb, 4);
2335  che_drc->interpolation_scheme = get_bits(gb, 4);
2336  n++;
2337  drc_num_bands += che_drc->band_incr;
2338  for (i = 0; i < drc_num_bands; i++) {
2339  che_drc->band_top[i] = get_bits(gb, 8);
2340  n++;
2341  }
2342  }
2343 
2344  /* prog_ref_level_present? */
2345  if (get_bits1(gb)) {
2346  che_drc->prog_ref_level = get_bits(gb, 7);
2347  skip_bits1(gb); // prog_ref_level_reserved_bits
2348  n++;
2349  }
2350 
2351  for (i = 0; i < drc_num_bands; i++) {
2352  che_drc->dyn_rng_sgn[i] = get_bits1(gb);
2353  che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
2354  n++;
2355  }
2356 
2357  return n;
2358 }
2359 
2360 static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
2361  uint8_t buf[256];
2362  int i, major, minor;
2363 
2364  if (len < 13+7*8)
2365  goto unknown;
2366 
2367  get_bits(gb, 13); len -= 13;
2368 
2369  for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
2370  buf[i] = get_bits(gb, 8);
2371 
2372  buf[i] = 0;
2373  if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
2374  av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
2375 
2376  if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
2377  ac->avctx->internal->skip_samples = 1024;
2378  }
2379 
2380 unknown:
2381  skip_bits_long(gb, len);
2382 
2383  return 0;
2384 }
2385 
2386 /**
2387  * Decode extension data (incomplete); reference: table 4.51.
2388  *
2389  * @param cnt length of TYPE_FIL syntactic element in bytes
2390  *
2391  * @return Returns number of bytes consumed
2392  */
2394  ChannelElement *che, enum RawDataBlockType elem_type)
2395 {
2396  int crc_flag = 0;
2397  int res = cnt;
2398  int type = get_bits(gb, 4);
2399 
2400  if (ac->avctx->debug & FF_DEBUG_STARTCODE)
2401  av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
2402 
2403  switch (type) { // extension type
2404  case EXT_SBR_DATA_CRC:
2405  crc_flag++;
2406  case EXT_SBR_DATA:
2407  if (!che) {
2408  av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
2409  return res;
2410  } else if (ac->oc[1].m4ac.frame_length_short) {
2411  if (!ac->warned_960_sbr)
2413  "SBR with 960 frame length");
2414  ac->warned_960_sbr = 1;
2415  skip_bits_long(gb, 8 * cnt - 4);
2416  return res;
2417  } else if (!ac->oc[1].m4ac.sbr) {
2418  av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
2419  skip_bits_long(gb, 8 * cnt - 4);
2420  return res;
2421  } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
2422  av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
2423  skip_bits_long(gb, 8 * cnt - 4);
2424  return res;
2425  } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
2426  ac->oc[1].m4ac.sbr = 1;
2427  ac->oc[1].m4ac.ps = 1;
2429  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
2430  ac->oc[1].status, 1);
2431  } else {
2432  ac->oc[1].m4ac.sbr = 1;
2434  }
2435  res = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
2436  break;
2437  case EXT_DYNAMIC_RANGE:
2438  res = decode_dynamic_range(&ac->che_drc, gb);
2439  break;
2440  case EXT_FILL:
2441  decode_fill(ac, gb, 8 * cnt - 4);
2442  break;
2443  case EXT_FILL_DATA:
2444  case EXT_DATA_ELEMENT:
2445  default:
2446  skip_bits_long(gb, 8 * cnt - 4);
2447  break;
2448  };
2449  return res;
2450 }
2451 
2452 /**
2453  * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
2454  *
2455  * @param decode 1 if tool is used normally, 0 if tool is used in LTP.
2456  * @param coef spectral coefficients
2457  */
2458 static void apply_tns(INTFLOAT coef_param[1024], TemporalNoiseShaping *tns,
2459  IndividualChannelStream *ics, int decode)
2460 {
2461  const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
2462  int w, filt, m, i;
2463  int bottom, top, order, start, end, size, inc;
2464  INTFLOAT lpc[TNS_MAX_ORDER];
2466  UINTFLOAT *coef = coef_param;
2467 
2468  if(!mmm)
2469  return;
2470 
2471  for (w = 0; w < ics->num_windows; w++) {
2472  bottom = ics->num_swb;
2473  for (filt = 0; filt < tns->n_filt[w]; filt++) {
2474  top = bottom;
2475  bottom = FFMAX(0, top - tns->length[w][filt]);
2476  order = tns->order[w][filt];
2477  if (order == 0)
2478  continue;
2479 
2480  // tns_decode_coef
2481  AAC_RENAME(compute_lpc_coefs)(tns->coef[w][filt], order, lpc, 0, 0, 0);
2482 
2483  start = ics->swb_offset[FFMIN(bottom, mmm)];
2484  end = ics->swb_offset[FFMIN( top, mmm)];
2485  if ((size = end - start) <= 0)
2486  continue;
2487  if (tns->direction[w][filt]) {
2488  inc = -1;
2489  start = end - 1;
2490  } else {
2491  inc = 1;
2492  }
2493  start += w * 128;
2494 
2495  if (decode) {
2496  // ar filter
2497  for (m = 0; m < size; m++, start += inc)
2498  for (i = 1; i <= FFMIN(m, order); i++)
2499  coef[start] -= AAC_MUL26((INTFLOAT)coef[start - i * inc], lpc[i - 1]);
2500  } else {
2501  // ma filter
2502  for (m = 0; m < size; m++, start += inc) {
2503  tmp[0] = coef[start];
2504  for (i = 1; i <= FFMIN(m, order); i++)
2505  coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
2506  for (i = order; i > 0; i--)
2507  tmp[i] = tmp[i - 1];
2508  }
2509  }
2510  }
2511  }
2512 }
2513 
2514 /**
2515  * Apply windowing and MDCT to obtain the spectral
2516  * coefficient from the predicted sample by LTP.
2517  */
2520 {
2521  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2522  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2523  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2524  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2525 
2526  if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
2527  ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
2528  } else {
2529  memset(in, 0, 448 * sizeof(*in));
2530  ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
2531  }
2532  if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
2533  ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
2534  } else {
2535  ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
2536  memset(in + 1024 + 576, 0, 448 * sizeof(*in));
2537  }
2538  ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
2539 }
2540 
2541 /**
2542  * Apply the long term prediction
2543  */
2545 {
2546  const LongTermPrediction *ltp = &sce->ics.ltp;
2547  const uint16_t *offsets = sce->ics.swb_offset;
2548  int i, sfb;
2549 
2550  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
2551  INTFLOAT *predTime = sce->ret;
2552  INTFLOAT *predFreq = ac->buf_mdct;
2553  int16_t num_samples = 2048;
2554 
2555  if (ltp->lag < 1024)
2556  num_samples = ltp->lag + 1024;
2557  for (i = 0; i < num_samples; i++)
2558  predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
2559  memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));
2560 
2561  ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
2562 
2563  if (sce->tns.present)
2564  ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
2565 
2566  for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
2567  if (ltp->used[sfb])
2568  for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
2569  sce->coeffs[i] += (UINTFLOAT)predFreq[i];
2570  }
2571 }
2572 
2573 /**
2574  * Update the LTP buffer for next frame
2575  */
2577 {
2578  IndividualChannelStream *ics = &sce->ics;
2579  INTFLOAT *saved = sce->saved;
2580  INTFLOAT *saved_ltp = sce->coeffs;
2581  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2582  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2583  int i;
2584 
2585  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2586  memcpy(saved_ltp, saved, 512 * sizeof(*saved_ltp));
2587  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2588  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2589 
2590  for (i = 0; i < 64; i++)
2591  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2592  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2593  memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(*saved_ltp));
2594  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2595  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2596 
2597  for (i = 0; i < 64; i++)
2598  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2599  } else { // LONG_STOP or ONLY_LONG
2600  ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
2601 
2602  for (i = 0; i < 512; i++)
2603  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
2604  }
2605 
2606  memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
2607  memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
2608  memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
2609 }
2610 
2611 /**
2612  * Conduct IMDCT and windowing.
2613  */
2615 {
2616  IndividualChannelStream *ics = &sce->ics;
2617  INTFLOAT *in = sce->coeffs;
2618  INTFLOAT *out = sce->ret;
2619  INTFLOAT *saved = sce->saved;
2620  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2621  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2622  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2623  INTFLOAT *buf = ac->buf_mdct;
2624  INTFLOAT *temp = ac->temp;
2625  int i;
2626 
2627  // imdct
2628  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2629  for (i = 0; i < 1024; i += 128)
2630  ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
2631  } else {
2632  ac->mdct.imdct_half(&ac->mdct, buf, in);
2633 #if USE_FIXED
2634  for (i=0; i<1024; i++)
2635  buf[i] = (buf[i] + 4LL) >> 3;
2636 #endif /* USE_FIXED */
2637  }
2638 
2639  /* window overlapping
2640  * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2641  * and long to short transitions are considered to be short to short
2642  * transitions. This leaves just two cases (long to long and short to short)
2643  * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2644  */
2645  if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2647  ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
2648  } else {
2649  memcpy( out, saved, 448 * sizeof(*out));
2650 
2651  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2652  ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
2653  ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
2654  ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
2655  ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
2656  ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
2657  memcpy( out + 448 + 4*128, temp, 64 * sizeof(*out));
2658  } else {
2659  ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
2660  memcpy( out + 576, buf + 64, 448 * sizeof(*out));
2661  }
2662  }
2663 
2664  // buffer update
2665  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2666  memcpy( saved, temp + 64, 64 * sizeof(*saved));
2667  ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
2668  ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
2669  ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
2670  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2671  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2672  memcpy( saved, buf + 512, 448 * sizeof(*saved));
2673  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2674  } else { // LONG_STOP or ONLY_LONG
2675  memcpy( saved, buf + 512, 512 * sizeof(*saved));
2676  }
2677 }
2678 
2679 /**
2680  * Conduct IMDCT and windowing.
2681  */
2683 {
2684 #if !USE_FIXED
2685  IndividualChannelStream *ics = &sce->ics;
2686  INTFLOAT *in = sce->coeffs;
2687  INTFLOAT *out = sce->ret;
2688  INTFLOAT *saved = sce->saved;
2689  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_120) : AAC_RENAME(ff_sine_120);
2690  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_960) : AAC_RENAME(ff_sine_960);
2691  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_120) : AAC_RENAME(ff_sine_120);
2692  INTFLOAT *buf = ac->buf_mdct;
2693  INTFLOAT *temp = ac->temp;
2694  int i;
2695 
2696  // imdct
2697  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2698  for (i = 0; i < 8; i++)
2699  ac->mdct120->imdct_half(ac->mdct120, buf + i * 120, in + i * 128, 1);
2700  } else {
2701  ac->mdct960->imdct_half(ac->mdct960, buf, in, 1);
2702  }
2703 
2704  /* window overlapping
2705  * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2706  * and long to short transitions are considered to be short to short
2707  * transitions. This leaves just two cases (long to long and short to short)
2708  * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2709  */
2710 
2711  if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2713  ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 480);
2714  } else {
2715  memcpy( out, saved, 420 * sizeof(*out));
2716 
2717  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2718  ac->fdsp->vector_fmul_window(out + 420 + 0*120, saved + 420, buf + 0*120, swindow_prev, 60);
2719  ac->fdsp->vector_fmul_window(out + 420 + 1*120, buf + 0*120 + 60, buf + 1*120, swindow, 60);
2720  ac->fdsp->vector_fmul_window(out + 420 + 2*120, buf + 1*120 + 60, buf + 2*120, swindow, 60);
2721  ac->fdsp->vector_fmul_window(out + 420 + 3*120, buf + 2*120 + 60, buf + 3*120, swindow, 60);
2722  ac->fdsp->vector_fmul_window(temp, buf + 3*120 + 60, buf + 4*120, swindow, 60);
2723  memcpy( out + 420 + 4*120, temp, 60 * sizeof(*out));
2724  } else {
2725  ac->fdsp->vector_fmul_window(out + 420, saved + 420, buf, swindow_prev, 60);
2726  memcpy( out + 540, buf + 60, 420 * sizeof(*out));
2727  }
2728  }
2729 
2730  // buffer update
2731  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2732  memcpy( saved, temp + 60, 60 * sizeof(*saved));
2733  ac->fdsp->vector_fmul_window(saved + 60, buf + 4*120 + 60, buf + 5*120, swindow, 60);
2734  ac->fdsp->vector_fmul_window(saved + 180, buf + 5*120 + 60, buf + 6*120, swindow, 60);
2735  ac->fdsp->vector_fmul_window(saved + 300, buf + 6*120 + 60, buf + 7*120, swindow, 60);
2736  memcpy( saved + 420, buf + 7*120 + 60, 60 * sizeof(*saved));
2737  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2738  memcpy( saved, buf + 480, 420 * sizeof(*saved));
2739  memcpy( saved + 420, buf + 7*120 + 60, 60 * sizeof(*saved));
2740  } else { // LONG_STOP or ONLY_LONG
2741  memcpy( saved, buf + 480, 480 * sizeof(*saved));
2742  }
2743 #endif
2744 }
2746 {
2747  IndividualChannelStream *ics = &sce->ics;
2748  INTFLOAT *in = sce->coeffs;
2749  INTFLOAT *out = sce->ret;
2750  INTFLOAT *saved = sce->saved;
2751  INTFLOAT *buf = ac->buf_mdct;
2752 #if USE_FIXED
2753  int i;
2754 #endif /* USE_FIXED */
2755 
2756  // imdct
2757  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2758 
2759 #if USE_FIXED
2760  for (i = 0; i < 1024; i++)
2761  buf[i] = (buf[i] + 2) >> 2;
2762 #endif /* USE_FIXED */
2763 
2764  // window overlapping
2765  if (ics->use_kb_window[1]) {
2766  // AAC LD uses a low overlap sine window instead of a KBD window
2767  memcpy(out, saved, 192 * sizeof(*out));
2768  ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
2769  memcpy( out + 320, buf + 64, 192 * sizeof(*out));
2770  } else {
2771  ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);
2772  }
2773 
2774  // buffer update
2775  memcpy(saved, buf + 256, 256 * sizeof(*saved));
2776 }
2777 
2779 {
2780  UINTFLOAT *in = sce->coeffs;
2781  INTFLOAT *out = sce->ret;
2782  INTFLOAT *saved = sce->saved;
2783  INTFLOAT *buf = ac->buf_mdct;
2784  int i;
2785  const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
2786  const int n2 = n >> 1;
2787  const int n4 = n >> 2;
2788  const INTFLOAT *const window = n == 480 ? AAC_RENAME(ff_aac_eld_window_480) :
2790 
2791  // Inverse transform, mapped to the conventional IMDCT by
2792  // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
2793  // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
2794  // International Conference on Audio, Language and Image Processing, ICALIP 2008.
2795  // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
2796  for (i = 0; i < n2; i+=2) {
2797  INTFLOAT temp;
2798  temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
2799  temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
2800  }
2801 #if !USE_FIXED
2802  if (n == 480)
2803  ac->mdct480->imdct_half(ac->mdct480, buf, in, 1);
2804  else
2805 #endif
2806  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2807 
2808 #if USE_FIXED
2809  for (i = 0; i < 1024; i++)
2810  buf[i] = (buf[i] + 1) >> 1;
2811 #endif /* USE_FIXED */
2812 
2813  for (i = 0; i < n; i+=2) {
2814  buf[i] = -buf[i];
2815  }
2816  // Like with the regular IMDCT at this point we still have the middle half
2817  // of a transform but with even symmetry on the left and odd symmetry on
2818  // the right
2819 
2820  // window overlapping
2821  // The spec says to use samples [0..511] but the reference decoder uses
2822  // samples [128..639].
2823  for (i = n4; i < n2; i ++) {
2824  out[i - n4] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) +
2825  AAC_MUL31( saved[ i + n2] , window[i + n - n4]) +
2826  AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
2827  AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]);
2828  }
2829  for (i = 0; i < n2; i ++) {
2830  out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) +
2831  AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) +
2832  AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) +
2833  AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
2834  }
2835  for (i = 0; i < n4; i ++) {
2836  out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) +
2837  AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
2838  AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
2839  }
2840 
2841  // buffer update
2842  memmove(saved + n, saved, 2 * n * sizeof(*saved));
2843  memcpy( saved, buf, n * sizeof(*saved));
2844 }
2845 
2846 /**
2847  * channel coupling transformation interface
2848  *
2849  * @param apply_coupling_method pointer to (in)dependent coupling function
2850  */
2852  enum RawDataBlockType type, int elem_id,
2853  enum CouplingPoint coupling_point,
2854  void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
2855 {
2856  int i, c;
2857 
2858  for (i = 0; i < MAX_ELEM_ID; i++) {
2859  ChannelElement *cce = ac->che[TYPE_CCE][i];
2860  int index = 0;
2861 
2862  if (cce && cce->coup.coupling_point == coupling_point) {
2863  ChannelCoupling *coup = &cce->coup;
2864 
2865  for (c = 0; c <= coup->num_coupled; c++) {
2866  if (coup->type[c] == type && coup->id_select[c] == elem_id) {
2867  if (coup->ch_select[c] != 1) {
2868  apply_coupling_method(ac, &cc->ch[0], cce, index);
2869  if (coup->ch_select[c] != 0)
2870  index++;
2871  }
2872  if (coup->ch_select[c] != 2)
2873  apply_coupling_method(ac, &cc->ch[1], cce, index++);
2874  } else
2875  index += 1 + (coup->ch_select[c] == 3);
2876  }
2877  }
2878  }
2879 }
2880 
2881 /**
2882  * Convert spectral data to samples, applying all supported tools as appropriate.
2883  */
2884 static void spectral_to_sample(AACContext *ac, int samples)
2885 {
2886  int i, type;
2888  switch (ac->oc[1].m4ac.object_type) {
2889  case AOT_ER_AAC_LD:
2891  break;
2892  case AOT_ER_AAC_ELD:
2894  break;
2895  default:
2896  if (ac->oc[1].m4ac.frame_length_short)
2898  else
2900  }
2901  for (type = 3; type >= 0; type--) {
2902  for (i = 0; i < MAX_ELEM_ID; i++) {
2903  ChannelElement *che = ac->che[type][i];
2904  if (che && che->present) {
2905  if (type <= TYPE_CPE)
2907  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2908  if (che->ch[0].ics.predictor_present) {
2909  if (che->ch[0].ics.ltp.present)
2910  ac->apply_ltp(ac, &che->ch[0]);
2911  if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
2912  ac->apply_ltp(ac, &che->ch[1]);
2913  }
2914  }
2915  if (che->ch[0].tns.present)
2916  ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2917  if (che->ch[1].tns.present)
2918  ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2919  if (type <= TYPE_CPE)
2921  if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2922  imdct_and_window(ac, &che->ch[0]);
2923  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2924  ac->update_ltp(ac, &che->ch[0]);
2925  if (type == TYPE_CPE) {
2926  imdct_and_window(ac, &che->ch[1]);
2927  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2928  ac->update_ltp(ac, &che->ch[1]);
2929  }
2930  if (ac->oc[1].m4ac.sbr > 0) {
2931  AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2932  }
2933  }
2934  if (type <= TYPE_CCE)
2936 
2937 #if USE_FIXED
2938  {
2939  int j;
2940  /* preparation for resampler */
2941  for(j = 0; j<samples; j++){
2942  che->ch[0].ret[j] = (int32_t)av_clip64((int64_t)che->ch[0].ret[j]*128, INT32_MIN, INT32_MAX-0x8000)+0x8000;
2943  if(type == TYPE_CPE)
2944  che->ch[1].ret[j] = (int32_t)av_clip64((int64_t)che->ch[1].ret[j]*128, INT32_MIN, INT32_MAX-0x8000)+0x8000;
2945  }
2946  }
2947 #endif /* USE_FIXED */
2948  che->present = 0;
2949  } else if (che) {
2950  av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
2951  }
2952  }
2953  }
2954 }
2955 
2957 {
2958  int size;
2959  AACADTSHeaderInfo hdr_info;
2960  uint8_t layout_map[MAX_ELEM_ID*4][3];
2961  int layout_map_tags, ret;
2962 
2963  size = avpriv_aac_parse_header(gb, &hdr_info);
2964  if (size > 0) {
2965  if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
2966  // This is 2 for "VLB " audio in NSV files.
2967  // See samples/nsv/vlb_audio.
2969  "More than one AAC RDB per ADTS frame");
2970  ac->warned_num_aac_frames = 1;
2971  }
2973  if (hdr_info.chan_config) {
2974  ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
2975  if ((ret = set_default_channel_config(ac->avctx,
2976  layout_map,
2977  &layout_map_tags,
2978  hdr_info.chan_config)) < 0)
2979  return ret;
2980  if ((ret = output_configure(ac, layout_map, layout_map_tags,
2981  FFMAX(ac->oc[1].status,
2982  OC_TRIAL_FRAME), 0)) < 0)
2983  return ret;
2984  } else {
2985  ac->oc[1].m4ac.chan_config = 0;
2986  /**
2987  * dual mono frames in Japanese DTV can have chan_config 0
2988  * WITHOUT specifying PCE.
2989  * thus, set dual mono as default.
2990  */
2991  if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
2992  layout_map_tags = 2;
2993  layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
2994  layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
2995  layout_map[0][1] = 0;
2996  layout_map[1][1] = 1;
2997  if (output_configure(ac, layout_map, layout_map_tags,
2998  OC_TRIAL_FRAME, 0))
2999  return -7;
3000  }
3001  }
3002  ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
3003  ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
3004  ac->oc[1].m4ac.object_type = hdr_info.object_type;
3005  ac->oc[1].m4ac.frame_length_short = 0;
3006  if (ac->oc[0].status != OC_LOCKED ||
3007  ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
3008  ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
3009  ac->oc[1].m4ac.sbr = -1;
3010  ac->oc[1].m4ac.ps = -1;
3011  }
3012  if (!hdr_info.crc_absent)
3013  skip_bits(gb, 16);
3014  }
3015  return size;
3016 }
3017 
3018 static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
3019  int *got_frame_ptr, GetBitContext *gb)
3020 {
3021  AACContext *ac = avctx->priv_data;
3022  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
3023  ChannelElement *che;
3024  int err, i;
3025  int samples = m4ac->frame_length_short ? 960 : 1024;
3026  int chan_config = m4ac->chan_config;
3027  int aot = m4ac->object_type;
3028 
3029  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
3030  samples >>= 1;
3031 
3032  ac->frame = data;
3033 
3034  if ((err = frame_configure_elements(avctx)) < 0)
3035  return err;
3036 
3037  // The FF_PROFILE_AAC_* defines are all object_type - 1
3038  // This may lead to an undefined profile being signaled
3039  ac->avctx->profile = aot - 1;
3040 
3041  ac->tags_mapped = 0;
3042 
3043  if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
3044  avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
3045  chan_config);
3046  return AVERROR_INVALIDDATA;
3047  }
3048  for (i = 0; i < tags_per_config[chan_config]; i++) {
3049  const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
3050  const int elem_id = aac_channel_layout_map[chan_config-1][i][1];
3051  if (!(che=get_che(ac, elem_type, elem_id))) {
3052  av_log(ac->avctx, AV_LOG_ERROR,
3053  "channel element %d.%d is not allocated\n",
3054  elem_type, elem_id);
3055  return AVERROR_INVALIDDATA;
3056  }
3057  che->present = 1;
3058  if (aot != AOT_ER_AAC_ELD)
3059  skip_bits(gb, 4);
3060  switch (elem_type) {
3061  case TYPE_SCE:
3062  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3063  break;
3064  case TYPE_CPE:
3065  err = decode_cpe(ac, gb, che);
3066  break;
3067  case TYPE_LFE:
3068  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3069  break;
3070  }
3071  if (err < 0)
3072  return err;
3073  }
3074 
3075  spectral_to_sample(ac, samples);
3076 
3077  if (!ac->frame->data[0] && samples) {
3078  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
3079  return AVERROR_INVALIDDATA;
3080  }
3081 
3082  ac->frame->nb_samples = samples;
3083  ac->frame->sample_rate = avctx->sample_rate;
3084  *got_frame_ptr = 1;
3085 
3086  skip_bits_long(gb, get_bits_left(gb));
3087  return 0;
3088 }
3089 
3090 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
3091  int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
3092 {
3093  AACContext *ac = avctx->priv_data;
3094  ChannelElement *che = NULL, *che_prev = NULL;
3095  enum RawDataBlockType elem_type, che_prev_type = TYPE_END;
3096  int err, elem_id;
3097  int samples = 0, multiplier, audio_found = 0, pce_found = 0;
3098  int is_dmono, sce_count = 0;
3099  int payload_alignment;
3100 
3101  ac->frame = data;
3102 
3103  if (show_bits(gb, 12) == 0xfff) {
3104  if ((err = parse_adts_frame_header(ac, gb)) < 0) {
3105  av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
3106  goto fail;
3107  }
3108  if (ac->oc[1].m4ac.sampling_index > 12) {
3109  av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
3110  err = AVERROR_INVALIDDATA;
3111  goto fail;
3112  }
3113  }
3114 
3115  if ((err = frame_configure_elements(avctx)) < 0)
3116  goto fail;
3117 
3118  // The FF_PROFILE_AAC_* defines are all object_type - 1
3119  // This may lead to an undefined profile being signaled
3120  ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
3121 
3122  payload_alignment = get_bits_count(gb);
3123  ac->tags_mapped = 0;
3124  // parse
3125  while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
3126  elem_id = get_bits(gb, 4);
3127 
3128  if (avctx->debug & FF_DEBUG_STARTCODE)
3129  av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
3130 
3131  if (!avctx->channels && elem_type != TYPE_PCE) {
3132  err = AVERROR_INVALIDDATA;
3133  goto fail;
3134  }
3135 
3136  if (elem_type < TYPE_DSE) {
3137  if (!(che=get_che(ac, elem_type, elem_id))) {
3138  av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
3139  elem_type, elem_id);
3140  err = AVERROR_INVALIDDATA;
3141  goto fail;
3142  }
3143  samples = ac->oc[1].m4ac.frame_length_short ? 960 : 1024;
3144  che->present = 1;
3145  }
3146 
3147  switch (elem_type) {
3148 
3149  case TYPE_SCE:
3150  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3151  audio_found = 1;
3152  sce_count++;
3153  break;
3154 
3155  case TYPE_CPE:
3156  err = decode_cpe(ac, gb, che);
3157  audio_found = 1;
3158  break;
3159 
3160  case TYPE_CCE:
3161  err = decode_cce(ac, gb, che);
3162  break;
3163 
3164  case TYPE_LFE:
3165  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3166  audio_found = 1;
3167  break;
3168 
3169  case TYPE_DSE:
3170  err = skip_data_stream_element(ac, gb);
3171  break;
3172 
3173  case TYPE_PCE: {
3174  uint8_t layout_map[MAX_ELEM_ID*4][3];
3175  int tags;
3176 
3177  int pushed = push_output_configuration(ac);
3178  if (pce_found && !pushed) {
3179  err = AVERROR_INVALIDDATA;
3180  goto fail;
3181  }
3182 
3183  tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb,
3184  payload_alignment);
3185  if (tags < 0) {
3186  err = tags;
3187  break;
3188  }
3189  if (pce_found) {
3190  av_log(avctx, AV_LOG_ERROR,
3191  "Not evaluating a further program_config_element as this construct is dubious at best.\n");
3193  } else {
3194  err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
3195  if (!err)
3196  ac->oc[1].m4ac.chan_config = 0;
3197  pce_found = 1;
3198  }
3199  break;
3200  }
3201 
3202  case TYPE_FIL:
3203  if (elem_id == 15)
3204  elem_id += get_bits(gb, 8) - 1;
3205  if (get_bits_left(gb) < 8 * elem_id) {
3206  av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
3207  err = AVERROR_INVALIDDATA;
3208  goto fail;
3209  }
3210  while (elem_id > 0)
3211  elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type);
3212  err = 0; /* FIXME */
3213  break;
3214 
3215  default:
3216  err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
3217  break;
3218  }
3219 
3220  if (elem_type < TYPE_DSE) {
3221  che_prev = che;
3222  che_prev_type = elem_type;
3223  }
3224 
3225  if (err)
3226  goto fail;
3227 
3228  if (get_bits_left(gb) < 3) {
3229  av_log(avctx, AV_LOG_ERROR, overread_err);
3230  err = AVERROR_INVALIDDATA;
3231  goto fail;
3232  }
3233  }
3234 
3235  if (!avctx->channels) {
3236  *got_frame_ptr = 0;
3237  return 0;
3238  }
3239 
3240  multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
3241  samples <<= multiplier;
3242 
3243  spectral_to_sample(ac, samples);
3244 
3245  if (ac->oc[1].status && audio_found) {
3246  avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
3247  avctx->frame_size = samples;
3248  ac->oc[1].status = OC_LOCKED;
3249  }
3250 
3251  if (multiplier)
3252  avctx->internal->skip_samples_multiplier = 2;
3253 
3254  if (!ac->frame->data[0] && samples) {
3255  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
3256  err = AVERROR_INVALIDDATA;
3257  goto fail;
3258  }
3259 
3260  if (samples) {
3261  ac->frame->nb_samples = samples;
3262  ac->frame->sample_rate = avctx->sample_rate;
3263  } else
3264  av_frame_unref(ac->frame);
3265  *got_frame_ptr = !!samples;
3266 
3267  /* for dual-mono audio (SCE + SCE) */
3268  is_dmono = ac->dmono_mode && sce_count == 2 &&
3270  if (is_dmono) {
3271  if (ac->dmono_mode == 1)
3272  ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
3273  else if (ac->dmono_mode == 2)
3274  ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
3275  }
3276 
3277  return 0;
3278 fail:
3280  return err;
3281 }
3282 
3283 static int aac_decode_frame(AVCodecContext *avctx, void *data,
3284  int *got_frame_ptr, AVPacket *avpkt)
3285 {
3286  AACContext *ac = avctx->priv_data;
3287  const uint8_t *buf = avpkt->data;
3288  int buf_size = avpkt->size;
3289  GetBitContext gb;
3290  int buf_consumed;
3291  int buf_offset;
3292  int err;
3293  int new_extradata_size;
3294  const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
3296  &new_extradata_size);
3297  int jp_dualmono_size;
3298  const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
3300  &jp_dualmono_size);
3301 
3302  if (new_extradata && 0) {
3303  av_free(avctx->extradata);
3304  avctx->extradata = av_mallocz(new_extradata_size +
3306  if (!avctx->extradata)
3307  return AVERROR(ENOMEM);
3308  avctx->extradata_size = new_extradata_size;
3309  memcpy(avctx->extradata, new_extradata, new_extradata_size);
3311  if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
3312  avctx->extradata,
3313  avctx->extradata_size*8LL, 1) < 0) {
3315  return AVERROR_INVALIDDATA;
3316  }
3317  }
3318 
3319  ac->dmono_mode = 0;
3320  if (jp_dualmono && jp_dualmono_size > 0)
3321  ac->dmono_mode = 1 + *jp_dualmono;
3322  if (ac->force_dmono_mode >= 0)
3323  ac->dmono_mode = ac->force_dmono_mode;
3324 
3325  if (INT_MAX / 8 <= buf_size)
3326  return AVERROR_INVALIDDATA;
3327 
3328  if ((err = init_get_bits8(&gb, buf, buf_size)) < 0)
3329  return err;
3330 
3331  switch (ac->oc[1].m4ac.object_type) {
3332  case AOT_ER_AAC_LC:
3333  case AOT_ER_AAC_LTP:
3334  case AOT_ER_AAC_LD:
3335  case AOT_ER_AAC_ELD:
3336  err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
3337  break;
3338  default:
3339  err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
3340  }
3341  if (err < 0)
3342  return err;
3343 
3344  buf_consumed = (get_bits_count(&gb) + 7) >> 3;
3345  for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
3346  if (buf[buf_offset])
3347  break;
3348 
3349  return buf_size > buf_offset ? buf_consumed : buf_size;
3350 }
3351 
3353 {
3354  AACContext *ac = avctx->priv_data;
3355  int i, type;
3356 
3357  for (i = 0; i < MAX_ELEM_ID; i++) {
3358  for (type = 0; type < 4; type++) {
3359  if (ac->che[type][i])
3360  AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][i]->sbr);
3361  av_freep(&ac->che[type][i]);
3362  }
3363  }
3364 
3365  ff_mdct_end(&ac->mdct);
3366  ff_mdct_end(&ac->mdct_small);
3367  ff_mdct_end(&ac->mdct_ld);
3368  ff_mdct_end(&ac->mdct_ltp);
3369 #if !USE_FIXED
3370  ff_mdct15_uninit(&ac->mdct120);
3371  ff_mdct15_uninit(&ac->mdct480);
3372  ff_mdct15_uninit(&ac->mdct960);
3373 #endif
3374  av_freep(&ac->fdsp);
3375  return 0;
3376 }
3377 
3378 static void aacdec_init(AACContext *c)
3379 {
3381  c->apply_ltp = apply_ltp;
3382  c->apply_tns = apply_tns;
3384  c->update_ltp = update_ltp;
3385 #if USE_FIXED
3388 #endif
3389 
3390 #if !USE_FIXED
3391  if(ARCH_MIPS)
3393 #endif /* !USE_FIXED */
3394 }
3395 /**
3396  * AVOptions for Japanese DTV specific extensions (ADTS only)
3397  */
3398 #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
3399 static const AVOption options[] = {
3400  {"dual_mono_mode", "Select the channel to decode for dual mono",
3401  offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
3402  AACDEC_FLAGS, "dual_mono_mode"},
3403 
3404  {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3405  {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3406  {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3407  {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3408 
3409  {NULL},
3410 };
3411 
3412 static const AVClass aac_decoder_class = {
3413  .class_name = "AAC decoder",
3414  .item_name = av_default_item_name,
3415  .option = options,
3416  .version = LIBAVUTIL_VERSION_INT,
3417 };
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
const char * s
Definition: avisynth_c.h:768
#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:201
uint8_t object_type
Definition: aacadtsdec.h:36
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:262
#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:86
#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)
Definition: get_bits.h:205
const char * g
Definition: vf_curves.c:112
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
ChannelElement * che[4][MAX_ELEM_ID]
Definition: aac.h:305
int size
Definition: avcodec.h:1680
const char * b
Definition: vf_curves.c:113
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:3059
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
static void subband_scale(int *dst, int *src, int scale, int offset, int len)
Definition: aacdec_fixed.c:165
#define MAX_LTP_LONG_SFB
Definition: aac.h:51
#define GET_GAIN(x, y)
Definition: aac_defines.h:98
Dynamic Range Control - decoded from the bitstream but not processed further.
Definition: aac.h:211
static av_cold int che_configure(AACContext *ac, enum ChannelPosition che_pos, int type, int id, int *channels)
Check for the channel element in the current channel position configuration.
static VLC vlc_scalefactors
#define NOISE_PRE
preamble for NOISE_BT, put in bitstream with the first noise band
Definition: aac.h:156
#define FF_PROFILE_AAC_HE_V2
Definition: avcodec.h:3275
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:3266
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:86
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
uint8_t bits
Definition: crc.c:296
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2531
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:3004
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:90
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:1876
int num_coupled
number of target elements
Definition: aac.h:236
#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:90
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:1679
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:200
#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:85
ChannelElement * tag_che_map[4][MAX_ELEM_ID]
Definition: aac.h:306
#define AVOnce
Definition: thread.h:157
#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:589
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:161
PredictorState predictor_state[MAX_PREDICTORS]
Definition: aac.h:268
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:127
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
av_default_item_name
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:1856
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: aacadtsdec.h:37
int amp[4]
Definition: aac.h:228
void(* apply_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:362
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:222
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:175
#define ff_mdct_init
Definition: fft.h:169
const float ff_aac_eld_window_512[1920]
Definition: aactab.c:1411
Definition: aac.h:62
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
#define CLOSE_READER(name, gb)
Definition: get_bits.h:132
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:109
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: aacadtsdec.h:38
Definition: vlc.h:26
float ff_aac_pow2sf_tab[428]
Definition: aactab.c:35
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2574
#define SKIP_BITS(name, gb, num)
Definition: get_bits.h:176
#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:87
static SDL_Window * window
Definition: ffplay.c:362
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:363
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:929
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:3050
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
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: aacadtsdec.h:39
int pos[4]
Definition: aac.h:227
MDCT15Context * mdct120
Definition: aac.h:330
Y Main.
Definition: mpeg4audio.h:71
int32_t
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
Definition: get_bits.h:297
FFTContext mdct_ld
Definition: aac.h:325
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:182
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:556
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:3061
int n
Definition: avisynth_c.h:684
const uint8_t ff_tns_max_bands_1024[]
Definition: aactab.c:1394
#define GET_VLC(code, name, gb, table, bits, max_depth)
If the vlc code is invalid and max_depth=1, then no bits will be removed.
Definition: get_bits.h:478
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
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:76
static const INTFLOAT *const tns_tmp2_map[4]
Definition: aactab.h:126
#define SHOW_UBITS(name, gb, num)
Definition: get_bits.h:194
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 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:2543
int force_dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
Definition: aac.h:350
void(* subband_scale)(int *dst, int *src, int scale, int offset, int len)
Definition: aac.h:369
The AV_PKT_DATA_NEW_EXTRADATA is used to notify the codec or the format that the extradata buffer was...
Definition: avcodec.h:1420
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:158
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:2523
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:457
#define AV_CH_LAYOUT_NATIVE
Channel mask value used for AVCodecContext.request_channel_layout to indicate that the user requests ...
static int decode_scalefactors(AACContext *ac, INTFLOAT sf[120], GetBitContext *gb, unsigned int global_gain, IndividualChannelStream *ics, enum BandType band_type[120], int band_type_run_end[120])
Decode scalefactors; reference: table 4.47.
int debug
debug
Definition: avcodec.h:3003
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:1761
#define AV_CH_FRONT_LEFT
int skip_samples_multiplier
Definition: internal.h:219
#define NOISE_PRE_BITS
length of preamble
Definition: aac.h:157
#define OPEN_READER(name, gb)
Definition: get_bits.h:121
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1669
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
int avpriv_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
Parse AAC frame header.
Definition: aacadtsdec.c:29
void * buf
Definition: avisynth_c.h:690
#define MAX_PREDICTORS
Definition: aac.h:146
static av_always_inline float cbrtf(float x)
Definition: libm.h:61
int extradata_size
Definition: avcodec.h:1877
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:314
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:339
#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:374
static av_cold int aac_decode_close(AVCodecContext *avctx)
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:307
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:426
#define GET_CACHE(name, gb)
Definition: get_bits.h:198
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:187
#define MAX_CHANNELS
Definition: aac.h:47
N Error Resilient Bit-Sliced Arithmetic Coding.
Definition: mpeg4audio.h:89
#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:2983
main AAC context
Definition: aac.h:293
#define u(width,...)
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 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:41
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:505
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:215
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:149
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:106
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:1531
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:368
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:367
static void ff_aac_tableinit(void)
Definition: aactab.h:45
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:777
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
void * priv_data
Definition: avcodec.h:1803
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 av_free(p)
#define FF_DEBUG_STARTCODE
Definition: avcodec.h:3017
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:2524
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:1811
const uint8_t ff_mpeg4audio_channels[8]
Definition: mpeg4audio.c:67
static int ff_thread_once(char *control, void(*routine)(void))
Definition: thread.h:160
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
Y Long Term Prediction.
Definition: mpeg4audio.h:74
uint8_t crc_absent
Definition: aacadtsdec.h:35
static const uint8_t * align_get_bits(GetBitContext *s)
Definition: get_bits.h:465
uint64_t layout
#define FF_PROFILE_AAC_HE
Definition: avcodec.h:3274
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:365
FILE * out
Definition: movenc.c:54
FFTContext mdct
Definition: aac.h:323
int sbr
-1 implicit, 1 presence
Definition: mpeg4audio.h:34
#define av_freep(p)
void INT64 INT64 count
Definition: avisynth_c.h:690
void INT64 start
Definition: avisynth_c.h:690
#define av_always_inline
Definition: attributes.h:39
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
Mid/Side stereo decoding; reference: 4.6.8.1.3.
void(* imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:361
#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:40
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: aacadtsdec.h:32
static const AVClass aac_decoder_class
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:248
uint64_t request_channel_layout
Request decoder to use this channel layout if it can (0 for default)
Definition: avcodec.h:2581
int layout_map_tags
Definition: aac.h:126
enum AVCodecID id
This structure stores compressed data.
Definition: avcodec.h:1656
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:267
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:2981
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:72
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