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aacdec_template.c
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1 /*
2  * AAC decoder
3  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
4  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
5  * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
6  *
7  * AAC LATM decoder
8  * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
9  * Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
10  *
11  * AAC decoder fixed-point implementation
12  * Copyright (c) 2013
13  * MIPS Technologies, Inc., California.
14  *
15  * This file is part of FFmpeg.
16  *
17  * FFmpeg is free software; you can redistribute it and/or
18  * modify it under the terms of the GNU Lesser General Public
19  * License as published by the Free Software Foundation; either
20  * version 2.1 of the License, or (at your option) any later version.
21  *
22  * FFmpeg is distributed in the hope that it will be useful,
23  * but WITHOUT ANY WARRANTY; without even the implied warranty of
24  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
25  * Lesser General Public License for more details.
26  *
27  * You should have received a copy of the GNU Lesser General Public
28  * License along with FFmpeg; if not, write to the Free Software
29  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
30  */
31 
32 /**
33  * @file
34  * AAC decoder
35  * @author Oded Shimon ( ods15 ods15 dyndns org )
36  * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
37  *
38  * AAC decoder fixed-point implementation
39  * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
40  * @author Nedeljko Babic ( nedeljko.babic imgtec com )
41  */
42 
43 /*
44  * supported tools
45  *
46  * Support? Name
47  * N (code in SoC repo) gain control
48  * Y block switching
49  * Y window shapes - standard
50  * N window shapes - Low Delay
51  * Y filterbank - standard
52  * N (code in SoC repo) filterbank - Scalable Sample Rate
53  * Y Temporal Noise Shaping
54  * Y Long Term Prediction
55  * Y intensity stereo
56  * Y channel coupling
57  * Y frequency domain prediction
58  * Y Perceptual Noise Substitution
59  * Y Mid/Side stereo
60  * N Scalable Inverse AAC Quantization
61  * N Frequency Selective Switch
62  * N upsampling filter
63  * Y quantization & coding - AAC
64  * N quantization & coding - TwinVQ
65  * N quantization & coding - BSAC
66  * N AAC Error Resilience tools
67  * N Error Resilience payload syntax
68  * N Error Protection tool
69  * N CELP
70  * N Silence Compression
71  * N HVXC
72  * N HVXC 4kbits/s VR
73  * N Structured Audio tools
74  * N Structured Audio Sample Bank Format
75  * N MIDI
76  * N Harmonic and Individual Lines plus Noise
77  * N Text-To-Speech Interface
78  * Y Spectral Band Replication
79  * Y (not in this code) Layer-1
80  * Y (not in this code) Layer-2
81  * Y (not in this code) Layer-3
82  * N SinuSoidal Coding (Transient, Sinusoid, Noise)
83  * Y Parametric Stereo
84  * N Direct Stream Transfer
85  * Y (not in fixed point code) Enhanced AAC Low Delay (ER AAC ELD)
86  *
87  * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
88  * - HE AAC v2 comprises LC AAC with Spectral Band Replication and
89  Parametric Stereo.
90  */
91 
92 #include "libavutil/thread.h"
93 
94 static VLC vlc_scalefactors;
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  */
127 static av_cold int che_configure(AACContext *ac,
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 
158 static int frame_configure_elements(AVCodecContext *avctx)
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 
192 struct elem_to_channel {
193  uint64_t av_position;
194  uint8_t syn_ele;
195  uint8_t elem_id;
196  uint8_t aac_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) {
292  .av_position = AV_CH_FRONT_CENTER,
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,
302  AV_CH_FRONT_LEFT_OF_CENTER,
303  AV_CH_FRONT_RIGHT_OF_CENTER,
304  AAC_CHANNEL_FRONT);
305  num_front_channels -= 2;
306  }
307  if (num_front_channels >= 2) {
308  i += assign_pair(e2c_vec, layout_map, i,
309  AV_CH_FRONT_LEFT,
310  AV_CH_FRONT_RIGHT,
311  AAC_CHANNEL_FRONT);
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,
318  AAC_CHANNEL_FRONT);
319  num_front_channels -= 2;
320  }
321 
322  if (num_side_channels >= 2) {
323  i += assign_pair(e2c_vec, layout_map, i,
324  AV_CH_SIDE_LEFT,
325  AV_CH_SIDE_RIGHT,
326  AAC_CHANNEL_FRONT);
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,
333  AAC_CHANNEL_SIDE);
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,
341  AAC_CHANNEL_BACK);
342  num_back_channels -= 2;
343  }
344  if (num_back_channels >= 2) {
345  i += assign_pair(e2c_vec, layout_map, i,
346  AV_CH_BACK_LEFT,
347  AV_CH_BACK_RIGHT,
348  AAC_CHANNEL_BACK);
349  num_back_channels -= 2;
350  }
351  if (num_back_channels) {
352  e2c_vec[i] = (struct elem_to_channel) {
353  .av_position = AV_CH_BACK_CENTER,
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) {
364  .av_position = AV_CH_LOW_FREQUENCY,
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  */
409 static int push_output_configuration(AACContext *ac) {
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  */
424 static void pop_output_configuration(AACContext *ac) {
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  */
440 static int output_configure(AACContext *ac,
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.
465  if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
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) {
481  layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
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  */
523 static int set_default_channel_config(AVCodecContext *avctx,
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;
572  push_output_configuration(ac);
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;
591  push_output_configuration(ac);
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)) {
631  av_log(ac->avctx, AV_LOG_WARNING,
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)) {
653  av_log(ac->avctx, AV_LOG_WARNING,
654  "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
655  type == TYPE_SCE ? "SCE" : "LFE", elem_id);
656  ac->warned_remapping_once++;
657  }
658  ac->tags_mapped++;
659  return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
660  }
661  if (ac->tags_mapped == 2 &&
662  ac->oc[1].m4ac.chan_config == 4 &&
663  type == TYPE_SCE) {
664  ac->tags_mapped++;
665  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
666  }
667  case 3:
668  case 2:
669  if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
670  type == TYPE_CPE) {
671  ac->tags_mapped++;
672  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
673  } else if (ac->oc[1].m4ac.chan_config == 2) {
674  return NULL;
675  }
676  case 1:
677  if (!ac->tags_mapped && type == TYPE_SCE) {
678  ac->tags_mapped++;
679  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
680  }
681  default:
682  return NULL;
683  }
684 }
685 
686 /**
687  * Decode an array of 4 bit element IDs, optionally interleaved with a
688  * stereo/mono switching bit.
689  *
690  * @param type speaker type/position for these channels
691  */
692 static void decode_channel_map(uint8_t layout_map[][3],
693  enum ChannelPosition type,
694  GetBitContext *gb, int n)
695 {
696  while (n--) {
697  enum RawDataBlockType syn_ele;
698  switch (type) {
699  case AAC_CHANNEL_FRONT:
700  case AAC_CHANNEL_BACK:
701  case AAC_CHANNEL_SIDE:
702  syn_ele = get_bits1(gb);
703  break;
704  case AAC_CHANNEL_CC:
705  skip_bits1(gb);
706  syn_ele = TYPE_CCE;
707  break;
708  case AAC_CHANNEL_LFE:
709  syn_ele = TYPE_LFE;
710  break;
711  default:
712  // AAC_CHANNEL_OFF has no channel map
713  av_assert0(0);
714  }
715  layout_map[0][0] = syn_ele;
716  layout_map[0][1] = get_bits(gb, 4);
717  layout_map[0][2] = type;
718  layout_map++;
719  }
720 }
721 
722 static inline void relative_align_get_bits(GetBitContext *gb,
723  int reference_position) {
724  int n = (reference_position - get_bits_count(gb) & 7);
725  if (n)
726  skip_bits(gb, n);
727 }
728 
729 /**
730  * Decode program configuration element; reference: table 4.2.
731  *
732  * @return Returns error status. 0 - OK, !0 - error
733  */
734 static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
735  uint8_t (*layout_map)[3],
736  GetBitContext *gb, int byte_align_ref)
737 {
738  int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
739  int sampling_index;
740  int comment_len;
741  int tags;
742 
743  skip_bits(gb, 2); // object_type
744 
745  sampling_index = get_bits(gb, 4);
746  if (m4ac->sampling_index != sampling_index)
747  av_log(avctx, AV_LOG_WARNING,
748  "Sample rate index in program config element does not "
749  "match the sample rate index configured by the container.\n");
750 
751  num_front = get_bits(gb, 4);
752  num_side = get_bits(gb, 4);
753  num_back = get_bits(gb, 4);
754  num_lfe = get_bits(gb, 2);
755  num_assoc_data = get_bits(gb, 3);
756  num_cc = get_bits(gb, 4);
757 
758  if (get_bits1(gb))
759  skip_bits(gb, 4); // mono_mixdown_tag
760  if (get_bits1(gb))
761  skip_bits(gb, 4); // stereo_mixdown_tag
762 
763  if (get_bits1(gb))
764  skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
765 
766  if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + 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  */
804 static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
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 (get_bits1(gb)) { // frameLengthFlag
815  avpriv_request_sample(avctx, "960/120 MDCT window");
816  return AVERROR_PATCHWELCOME;
817  }
818  m4ac->frame_length_short = 0;
819 
820  if (get_bits1(gb)) // dependsOnCoreCoder
821  skip_bits(gb, 14); // coreCoderDelay
822  extension_flag = get_bits1(gb);
823 
824  if (m4ac->object_type == AOT_AAC_SCALABLE ||
825  m4ac->object_type == AOT_ER_AAC_SCALABLE)
826  skip_bits(gb, 3); // layerNr
827 
828  if (channel_config == 0) {
829  skip_bits(gb, 4); // element_instance_tag
830  tags = decode_pce(avctx, m4ac, layout_map, gb, get_bit_alignment);
831  if (tags < 0)
832  return tags;
833  } else {
834  if ((ret = set_default_channel_config(avctx, layout_map,
835  &tags, channel_config)))
836  return ret;
837  }
838 
839  if (count_channels(layout_map, tags) > 1) {
840  m4ac->ps = 0;
841  } else if (m4ac->sbr == 1 && m4ac->ps == -1)
842  m4ac->ps = 1;
843 
844  if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
845  return ret;
846 
847  if (extension_flag) {
848  switch (m4ac->object_type) {
849  case AOT_ER_BSAC:
850  skip_bits(gb, 5); // numOfSubFrame
851  skip_bits(gb, 11); // layer_length
852  break;
853  case AOT_ER_AAC_LC:
854  case AOT_ER_AAC_LTP:
855  case AOT_ER_AAC_SCALABLE:
856  case AOT_ER_AAC_LD:
857  res_flags = get_bits(gb, 3);
858  if (res_flags) {
859  avpriv_report_missing_feature(avctx,
860  "AAC data resilience (flags %x)",
861  res_flags);
862  return AVERROR_PATCHWELCOME;
863  }
864  break;
865  }
866  skip_bits1(gb); // extensionFlag3 (TBD in version 3)
867  }
868  switch (m4ac->object_type) {
869  case AOT_ER_AAC_LC:
870  case AOT_ER_AAC_LTP:
871  case AOT_ER_AAC_SCALABLE:
872  case AOT_ER_AAC_LD:
873  ep_config = get_bits(gb, 2);
874  if (ep_config) {
875  avpriv_report_missing_feature(avctx,
876  "epConfig %d", ep_config);
877  return AVERROR_PATCHWELCOME;
878  }
879  }
880  return 0;
881 }
882 
883 static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
884  GetBitContext *gb,
885  MPEG4AudioConfig *m4ac,
886  int channel_config)
887 {
888  int ret, ep_config, res_flags;
889  uint8_t layout_map[MAX_ELEM_ID*4][3];
890  int tags = 0;
891  const int ELDEXT_TERM = 0;
892 
893  m4ac->ps = 0;
894  m4ac->sbr = 0;
895 #if USE_FIXED
896  if (get_bits1(gb)) { // frameLengthFlag
897  avpriv_request_sample(avctx, "960/120 MDCT window");
898  return AVERROR_PATCHWELCOME;
899  }
900 #else
901  m4ac->frame_length_short = get_bits1(gb);
902 #endif
903  res_flags = get_bits(gb, 3);
904  if (res_flags) {
905  avpriv_report_missing_feature(avctx,
906  "AAC data resilience (flags %x)",
907  res_flags);
908  return AVERROR_PATCHWELCOME;
909  }
910 
911  if (get_bits1(gb)) { // ldSbrPresentFlag
912  avpriv_report_missing_feature(avctx,
913  "Low Delay SBR");
914  return AVERROR_PATCHWELCOME;
915  }
916 
917  while (get_bits(gb, 4) != ELDEXT_TERM) {
918  int len = get_bits(gb, 4);
919  if (len == 15)
920  len += get_bits(gb, 8);
921  if (len == 15 + 255)
922  len += get_bits(gb, 16);
923  if (get_bits_left(gb) < len * 8 + 4) {
924  av_log(avctx, AV_LOG_ERROR, overread_err);
925  return AVERROR_INVALIDDATA;
926  }
927  skip_bits_long(gb, 8 * len);
928  }
929 
930  if ((ret = set_default_channel_config(avctx, layout_map,
931  &tags, channel_config)))
932  return ret;
933 
934  if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
935  return ret;
936 
937  ep_config = get_bits(gb, 2);
938  if (ep_config) {
939  avpriv_report_missing_feature(avctx,
940  "epConfig %d", ep_config);
941  return AVERROR_PATCHWELCOME;
942  }
943  return 0;
944 }
945 
946 /**
947  * Decode audio specific configuration; reference: table 1.13.
948  *
949  * @param ac pointer to AACContext, may be null
950  * @param avctx pointer to AVCCodecContext, used for logging
951  * @param m4ac pointer to MPEG4AudioConfig, used for parsing
952  * @param gb buffer holding an audio specific config
953  * @param get_bit_alignment relative alignment for byte align operations
954  * @param sync_extension look for an appended sync extension
955  *
956  * @return Returns error status or number of consumed bits. <0 - error
957  */
958 static int decode_audio_specific_config_gb(AACContext *ac,
959  AVCodecContext *avctx,
960  MPEG4AudioConfig *m4ac,
961  GetBitContext *gb,
962  int get_bit_alignment,
963  int sync_extension)
964 {
965  int i, ret;
966  GetBitContext gbc = *gb;
967 
968  if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension)) < 0)
969  return AVERROR_INVALIDDATA;
970 
971  if (m4ac->sampling_index > 12) {
972  av_log(avctx, AV_LOG_ERROR,
973  "invalid sampling rate index %d\n",
974  m4ac->sampling_index);
975  return AVERROR_INVALIDDATA;
976  }
977  if (m4ac->object_type == AOT_ER_AAC_LD &&
978  (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
979  av_log(avctx, AV_LOG_ERROR,
980  "invalid low delay sampling rate index %d\n",
981  m4ac->sampling_index);
982  return AVERROR_INVALIDDATA;
983  }
984 
985  skip_bits_long(gb, i);
986 
987  switch (m4ac->object_type) {
988  case AOT_AAC_MAIN:
989  case AOT_AAC_LC:
990  case AOT_AAC_LTP:
991  case AOT_ER_AAC_LC:
992  case AOT_ER_AAC_LD:
993  if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment,
994  m4ac, m4ac->chan_config)) < 0)
995  return ret;
996  break;
997  case AOT_ER_AAC_ELD:
998  if ((ret = decode_eld_specific_config(ac, avctx, gb,
999  m4ac, m4ac->chan_config)) < 0)
1000  return ret;
1001  break;
1002  default:
1003  avpriv_report_missing_feature(avctx,
1004  "Audio object type %s%d",
1005  m4ac->sbr == 1 ? "SBR+" : "",
1006  m4ac->object_type);
1007  return AVERROR(ENOSYS);
1008  }
1009 
1010  ff_dlog(avctx,
1011  "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
1012  m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
1013  m4ac->sample_rate, m4ac->sbr,
1014  m4ac->ps);
1015 
1016  return get_bits_count(gb);
1017 }
1018 
1019 static int decode_audio_specific_config(AACContext *ac,
1020  AVCodecContext *avctx,
1021  MPEG4AudioConfig *m4ac,
1022  const uint8_t *data, int64_t bit_size,
1023  int sync_extension)
1024 {
1025  int i, ret;
1026  GetBitContext gb;
1027 
1028  if (bit_size < 0 || bit_size > INT_MAX) {
1029  av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n");
1030  return AVERROR_INVALIDDATA;
1031  }
1032 
1033  ff_dlog(avctx, "audio specific config size %d\n", (int)bit_size >> 3);
1034  for (i = 0; i < bit_size >> 3; i++)
1035  ff_dlog(avctx, "%02x ", data[i]);
1036  ff_dlog(avctx, "\n");
1037 
1038  if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
1039  return ret;
1040 
1041  return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0,
1042  sync_extension);
1043 }
1044 
1045 /**
1046  * linear congruential pseudorandom number generator
1047  *
1048  * @param previous_val pointer to the current state of the generator
1049  *
1050  * @return Returns a 32-bit pseudorandom integer
1051  */
1052 static av_always_inline int lcg_random(unsigned previous_val)
1053 {
1054  union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
1055  return v.s;
1056 }
1057 
1058 static void reset_all_predictors(PredictorState *ps)
1059 {
1060  int i;
1061  for (i = 0; i < MAX_PREDICTORS; i++)
1062  reset_predict_state(&ps[i]);
1063 }
1064 
1065 static int sample_rate_idx (int rate)
1066 {
1067  if (92017 <= rate) return 0;
1068  else if (75132 <= rate) return 1;
1069  else if (55426 <= rate) return 2;
1070  else if (46009 <= rate) return 3;
1071  else if (37566 <= rate) return 4;
1072  else if (27713 <= rate) return 5;
1073  else if (23004 <= rate) return 6;
1074  else if (18783 <= rate) return 7;
1075  else if (13856 <= rate) return 8;
1076  else if (11502 <= rate) return 9;
1077  else if (9391 <= rate) return 10;
1078  else return 11;
1079 }
1080 
1081 static void reset_predictor_group(PredictorState *ps, int group_num)
1082 {
1083  int i;
1084  for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
1085  reset_predict_state(&ps[i]);
1086 }
1087 
1088 #define AAC_INIT_VLC_STATIC(num, size) \
1089  INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
1090  ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \
1091  sizeof(ff_aac_spectral_bits[num][0]), \
1092  ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
1093  sizeof(ff_aac_spectral_codes[num][0]), \
1094  size);
1095 
1096 static void aacdec_init(AACContext *ac);
1097 
1098 static av_cold void aac_static_table_init(void)
1099 {
1100  AAC_INIT_VLC_STATIC( 0, 304);
1101  AAC_INIT_VLC_STATIC( 1, 270);
1102  AAC_INIT_VLC_STATIC( 2, 550);
1103  AAC_INIT_VLC_STATIC( 3, 300);
1104  AAC_INIT_VLC_STATIC( 4, 328);
1105  AAC_INIT_VLC_STATIC( 5, 294);
1106  AAC_INIT_VLC_STATIC( 6, 306);
1107  AAC_INIT_VLC_STATIC( 7, 268);
1108  AAC_INIT_VLC_STATIC( 8, 510);
1109  AAC_INIT_VLC_STATIC( 9, 366);
1110  AAC_INIT_VLC_STATIC(10, 462);
1111 
1112  AAC_RENAME(ff_aac_sbr_init)();
1113 
1114  ff_aac_tableinit();
1115 
1116  INIT_VLC_STATIC(&vlc_scalefactors, 7,
1117  FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
1118  ff_aac_scalefactor_bits,
1119  sizeof(ff_aac_scalefactor_bits[0]),
1120  sizeof(ff_aac_scalefactor_bits[0]),
1121  ff_aac_scalefactor_code,
1122  sizeof(ff_aac_scalefactor_code[0]),
1123  sizeof(ff_aac_scalefactor_code[0]),
1124  352);
1125 
1126  // window initialization
1127  AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_1024), 4.0, 1024);
1128  AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_128), 6.0, 128);
1129  AAC_RENAME(ff_init_ff_sine_windows)(10);
1130  AAC_RENAME(ff_init_ff_sine_windows)( 9);
1131  AAC_RENAME(ff_init_ff_sine_windows)( 7);
1132 
1133  AAC_RENAME(ff_cbrt_tableinit)();
1134 }
1135 
1136 static AVOnce aac_table_init = AV_ONCE_INIT;
1137 
1138 static av_cold int aac_decode_init(AVCodecContext *avctx)
1139 {
1140  AACContext *ac = avctx->priv_data;
1141  int ret;
1142 
1143  ret = ff_thread_once(&aac_table_init, &aac_static_table_init);
1144  if (ret != 0)
1145  return AVERROR_UNKNOWN;
1146 
1147  ac->avctx = avctx;
1148  ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
1149 
1150  aacdec_init(ac);
1151 #if USE_FIXED
1152  avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
1153 #else
1154  avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1155 #endif /* USE_FIXED */
1156 
1157  if (avctx->extradata_size > 0) {
1158  if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
1159  avctx->extradata,
1160  avctx->extradata_size * 8LL,
1161  1)) < 0)
1162  return ret;
1163  } else {
1164  int sr, i;
1165  uint8_t layout_map[MAX_ELEM_ID*4][3];
1166  int layout_map_tags;
1167 
1168  sr = sample_rate_idx(avctx->sample_rate);
1169  ac->oc[1].m4ac.sampling_index = sr;
1170  ac->oc[1].m4ac.channels = avctx->channels;
1171  ac->oc[1].m4ac.sbr = -1;
1172  ac->oc[1].m4ac.ps = -1;
1173 
1174  for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
1175  if (ff_mpeg4audio_channels[i] == avctx->channels)
1176  break;
1177  if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
1178  i = 0;
1179  }
1180  ac->oc[1].m4ac.chan_config = i;
1181 
1182  if (ac->oc[1].m4ac.chan_config) {
1183  int ret = set_default_channel_config(avctx, layout_map,
1184  &layout_map_tags, ac->oc[1].m4ac.chan_config);
1185  if (!ret)
1186  output_configure(ac, layout_map, layout_map_tags,
1187  OC_GLOBAL_HDR, 0);
1188  else if (avctx->err_recognition & AV_EF_EXPLODE)
1189  return AVERROR_INVALIDDATA;
1190  }
1191  }
1192 
1193  if (avctx->channels > MAX_CHANNELS) {
1194  av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
1195  return AVERROR_INVALIDDATA;
1196  }
1197 
1198 #if USE_FIXED
1199  ac->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & AV_CODEC_FLAG_BITEXACT);
1200 #else
1201  ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
1202 #endif /* USE_FIXED */
1203  if (!ac->fdsp) {
1204  return AVERROR(ENOMEM);
1205  }
1206 
1207  ac->random_state = 0x1f2e3d4c;
1208 
1209  AAC_RENAME_32(ff_mdct_init)(&ac->mdct, 11, 1, 1.0 / RANGE15(1024.0));
1210  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld, 10, 1, 1.0 / RANGE15(512.0));
1211  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small, 8, 1, 1.0 / RANGE15(128.0));
1212  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp, 11, 0, RANGE15(-2.0));
1213 #if !USE_FIXED
1214  ret = ff_mdct15_init(&ac->mdct480, 1, 5, -1.0f);
1215  if (ret < 0)
1216  return ret;
1217 #endif
1218 
1219  return 0;
1220 }
1221 
1222 /**
1223  * Skip data_stream_element; reference: table 4.10.
1224  */
1225 static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
1226 {
1227  int byte_align = get_bits1(gb);
1228  int count = get_bits(gb, 8);
1229  if (count == 255)
1230  count += get_bits(gb, 8);
1231  if (byte_align)
1232  align_get_bits(gb);
1233 
1234  if (get_bits_left(gb) < 8 * count) {
1235  av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
1236  return AVERROR_INVALIDDATA;
1237  }
1238  skip_bits_long(gb, 8 * count);
1239  return 0;
1240 }
1241 
1242 static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
1243  GetBitContext *gb)
1244 {
1245  int sfb;
1246  if (get_bits1(gb)) {
1247  ics->predictor_reset_group = get_bits(gb, 5);
1248  if (ics->predictor_reset_group == 0 ||
1249  ics->predictor_reset_group > 30) {
1250  av_log(ac->avctx, AV_LOG_ERROR,
1251  "Invalid Predictor Reset Group.\n");
1252  return AVERROR_INVALIDDATA;
1253  }
1254  }
1255  for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
1256  ics->prediction_used[sfb] = get_bits1(gb);
1257  }
1258  return 0;
1259 }
1260 
1261 /**
1262  * Decode Long Term Prediction data; reference: table 4.xx.
1263  */
1264 static void decode_ltp(LongTermPrediction *ltp,
1265  GetBitContext *gb, uint8_t max_sfb)
1266 {
1267  int sfb;
1268 
1269  ltp->lag = get_bits(gb, 11);
1270  ltp->coef = ltp_coef[get_bits(gb, 3)];
1271  for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
1272  ltp->used[sfb] = get_bits1(gb);
1273 }
1274 
1275 /**
1276  * Decode Individual Channel Stream info; reference: table 4.6.
1277  */
1278 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
1279  GetBitContext *gb)
1280 {
1281  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
1282  const int aot = m4ac->object_type;
1283  const int sampling_index = m4ac->sampling_index;
1284  int ret_fail = AVERROR_INVALIDDATA;
1285 
1286  if (aot != AOT_ER_AAC_ELD) {
1287  if (get_bits1(gb)) {
1288  av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
1289  if (ac->avctx->err_recognition & AV_EF_BITSTREAM)
1290  return AVERROR_INVALIDDATA;
1291  }
1292  ics->window_sequence[1] = ics->window_sequence[0];
1293  ics->window_sequence[0] = get_bits(gb, 2);
1294  if (aot == AOT_ER_AAC_LD &&
1295  ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
1296  av_log(ac->avctx, AV_LOG_ERROR,
1297  "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
1298  "window sequence %d found.\n", ics->window_sequence[0]);
1299  ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
1300  return AVERROR_INVALIDDATA;
1301  }
1302  ics->use_kb_window[1] = ics->use_kb_window[0];
1303  ics->use_kb_window[0] = get_bits1(gb);
1304  }
1305  ics->num_window_groups = 1;
1306  ics->group_len[0] = 1;
1307  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1308  int i;
1309  ics->max_sfb = get_bits(gb, 4);
1310  for (i = 0; i < 7; i++) {
1311  if (get_bits1(gb)) {
1312  ics->group_len[ics->num_window_groups - 1]++;
1313  } else {
1314  ics->num_window_groups++;
1315  ics->group_len[ics->num_window_groups - 1] = 1;
1316  }
1317  }
1318  ics->num_windows = 8;
1319  ics->swb_offset = ff_swb_offset_128[sampling_index];
1320  ics->num_swb = ff_aac_num_swb_128[sampling_index];
1321  ics->tns_max_bands = ff_tns_max_bands_128[sampling_index];
1322  ics->predictor_present = 0;
1323  } else {
1324  ics->max_sfb = get_bits(gb, 6);
1325  ics->num_windows = 1;
1326  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
1327  if (m4ac->frame_length_short) {
1328  ics->swb_offset = ff_swb_offset_480[sampling_index];
1329  ics->num_swb = ff_aac_num_swb_480[sampling_index];
1330  ics->tns_max_bands = ff_tns_max_bands_480[sampling_index];
1331  } else {
1332  ics->swb_offset = ff_swb_offset_512[sampling_index];
1333  ics->num_swb = ff_aac_num_swb_512[sampling_index];
1334  ics->tns_max_bands = ff_tns_max_bands_512[sampling_index];
1335  }
1336  if (!ics->num_swb || !ics->swb_offset) {
1337  ret_fail = AVERROR_BUG;
1338  goto fail;
1339  }
1340  } else {
1341  ics->swb_offset = ff_swb_offset_1024[sampling_index];
1342  ics->num_swb = ff_aac_num_swb_1024[sampling_index];
1343  ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
1344  }
1345  if (aot != AOT_ER_AAC_ELD) {
1346  ics->predictor_present = get_bits1(gb);
1347  ics->predictor_reset_group = 0;
1348  }
1349  if (ics->predictor_present) {
1350  if (aot == AOT_AAC_MAIN) {
1351  if (decode_prediction(ac, ics, gb)) {
1352  goto fail;
1353  }
1354  } else if (aot == AOT_AAC_LC ||
1355  aot == AOT_ER_AAC_LC) {
1356  av_log(ac->avctx, AV_LOG_ERROR,
1357  "Prediction is not allowed in AAC-LC.\n");
1358  goto fail;
1359  } else {
1360  if (aot == AOT_ER_AAC_LD) {
1361  av_log(ac->avctx, AV_LOG_ERROR,
1362  "LTP in ER AAC LD not yet implemented.\n");
1363  ret_fail = AVERROR_PATCHWELCOME;
1364  goto fail;
1365  }
1366  if ((ics->ltp.present = get_bits(gb, 1)))
1367  decode_ltp(&ics->ltp, gb, ics->max_sfb);
1368  }
1369  }
1370  }
1371 
1372  if (ics->max_sfb > ics->num_swb) {
1373  av_log(ac->avctx, AV_LOG_ERROR,
1374  "Number of scalefactor bands in group (%d) "
1375  "exceeds limit (%d).\n",
1376  ics->max_sfb, ics->num_swb);
1377  goto fail;
1378  }
1379 
1380  return 0;
1381 fail:
1382  ics->max_sfb = 0;
1383  return ret_fail;
1384 }
1385 
1386 /**
1387  * Decode band types (section_data payload); reference: table 4.46.
1388  *
1389  * @param band_type array of the used band type
1390  * @param band_type_run_end array of the last scalefactor band of a band type run
1391  *
1392  * @return Returns error status. 0 - OK, !0 - error
1393  */
1394 static int decode_band_types(AACContext *ac, enum BandType band_type[120],
1395  int band_type_run_end[120], GetBitContext *gb,
1396  IndividualChannelStream *ics)
1397 {
1398  int g, idx = 0;
1399  const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
1400  for (g = 0; g < ics->num_window_groups; g++) {
1401  int k = 0;
1402  while (k < ics->max_sfb) {
1403  uint8_t sect_end = k;
1404  int sect_len_incr;
1405  int sect_band_type = get_bits(gb, 4);
1406  if (sect_band_type == 12) {
1407  av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
1408  return AVERROR_INVALIDDATA;
1409  }
1410  do {
1411  sect_len_incr = get_bits(gb, bits);
1412  sect_end += sect_len_incr;
1413  if (get_bits_left(gb) < 0) {
1414  av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
1415  return AVERROR_INVALIDDATA;
1416  }
1417  if (sect_end > ics->max_sfb) {
1418  av_log(ac->avctx, AV_LOG_ERROR,
1419  "Number of bands (%d) exceeds limit (%d).\n",
1420  sect_end, ics->max_sfb);
1421  return AVERROR_INVALIDDATA;
1422  }
1423  } while (sect_len_incr == (1 << bits) - 1);
1424  for (; k < sect_end; k++) {
1425  band_type [idx] = sect_band_type;
1426  band_type_run_end[idx++] = sect_end;
1427  }
1428  }
1429  }
1430  return 0;
1431 }
1432 
1433 /**
1434  * Decode scalefactors; reference: table 4.47.
1435  *
1436  * @param global_gain first scalefactor value as scalefactors are differentially coded
1437  * @param band_type array of the used band type
1438  * @param band_type_run_end array of the last scalefactor band of a band type run
1439  * @param sf array of scalefactors or intensity stereo positions
1440  *
1441  * @return Returns error status. 0 - OK, !0 - error
1442  */
1443 static int decode_scalefactors(AACContext *ac, INTFLOAT sf[120], GetBitContext *gb,
1444  unsigned int global_gain,
1445  IndividualChannelStream *ics,
1446  enum BandType band_type[120],
1447  int band_type_run_end[120])
1448 {
1449  int g, i, idx = 0;
1450  int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 };
1451  int clipped_offset;
1452  int noise_flag = 1;
1453  for (g = 0; g < ics->num_window_groups; g++) {
1454  for (i = 0; i < ics->max_sfb;) {
1455  int run_end = band_type_run_end[idx];
1456  if (band_type[idx] == ZERO_BT) {
1457  for (; i < run_end; i++, idx++)
1458  sf[idx] = FIXR(0.);
1459  } else if ((band_type[idx] == INTENSITY_BT) ||
1460  (band_type[idx] == INTENSITY_BT2)) {
1461  for (; i < run_end; i++, idx++) {
1462  offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1463  clipped_offset = av_clip(offset[2], -155, 100);
1464  if (offset[2] != clipped_offset) {
1465  avpriv_request_sample(ac->avctx,
1466  "If you heard an audible artifact, there may be a bug in the decoder. "
1467  "Clipped intensity stereo position (%d -> %d)",
1468  offset[2], clipped_offset);
1469  }
1470 #if USE_FIXED
1471  sf[idx] = 100 - clipped_offset;
1472 #else
1473  sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
1474 #endif /* USE_FIXED */
1475  }
1476  } else if (band_type[idx] == NOISE_BT) {
1477  for (; i < run_end; i++, idx++) {
1478  if (noise_flag-- > 0)
1479  offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
1480  else
1481  offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1482  clipped_offset = av_clip(offset[1], -100, 155);
1483  if (offset[1] != clipped_offset) {
1484  avpriv_request_sample(ac->avctx,
1485  "If you heard an audible artifact, there may be a bug in the decoder. "
1486  "Clipped noise gain (%d -> %d)",
1487  offset[1], clipped_offset);
1488  }
1489 #if USE_FIXED
1490  sf[idx] = -(100 + clipped_offset);
1491 #else
1492  sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
1493 #endif /* USE_FIXED */
1494  }
1495  } else {
1496  for (; i < run_end; i++, idx++) {
1497  offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1498  if (offset[0] > 255U) {
1499  av_log(ac->avctx, AV_LOG_ERROR,
1500  "Scalefactor (%d) out of range.\n", offset[0]);
1501  return AVERROR_INVALIDDATA;
1502  }
1503 #if USE_FIXED
1504  sf[idx] = -offset[0];
1505 #else
1506  sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
1507 #endif /* USE_FIXED */
1508  }
1509  }
1510  }
1511  }
1512  return 0;
1513 }
1514 
1515 /**
1516  * Decode pulse data; reference: table 4.7.
1517  */
1518 static int decode_pulses(Pulse *pulse, GetBitContext *gb,
1519  const uint16_t *swb_offset, int num_swb)
1520 {
1521  int i, pulse_swb;
1522  pulse->num_pulse = get_bits(gb, 2) + 1;
1523  pulse_swb = get_bits(gb, 6);
1524  if (pulse_swb >= num_swb)
1525  return -1;
1526  pulse->pos[0] = swb_offset[pulse_swb];
1527  pulse->pos[0] += get_bits(gb, 5);
1528  if (pulse->pos[0] >= swb_offset[num_swb])
1529  return -1;
1530  pulse->amp[0] = get_bits(gb, 4);
1531  for (i = 1; i < pulse->num_pulse; i++) {
1532  pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
1533  if (pulse->pos[i] >= swb_offset[num_swb])
1534  return -1;
1535  pulse->amp[i] = get_bits(gb, 4);
1536  }
1537  return 0;
1538 }
1539 
1540 /**
1541  * Decode Temporal Noise Shaping data; reference: table 4.48.
1542  *
1543  * @return Returns error status. 0 - OK, !0 - error
1544  */
1545 static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
1546  GetBitContext *gb, const IndividualChannelStream *ics)
1547 {
1548  int w, filt, i, coef_len, coef_res, coef_compress;
1549  const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
1550  const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
1551  for (w = 0; w < ics->num_windows; w++) {
1552  if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
1553  coef_res = get_bits1(gb);
1554 
1555  for (filt = 0; filt < tns->n_filt[w]; filt++) {
1556  int tmp2_idx;
1557  tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
1558 
1559  if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
1560  av_log(ac->avctx, AV_LOG_ERROR,
1561  "TNS filter order %d is greater than maximum %d.\n",
1562  tns->order[w][filt], tns_max_order);
1563  tns->order[w][filt] = 0;
1564  return AVERROR_INVALIDDATA;
1565  }
1566  if (tns->order[w][filt]) {
1567  tns->direction[w][filt] = get_bits1(gb);
1568  coef_compress = get_bits1(gb);
1569  coef_len = coef_res + 3 - coef_compress;
1570  tmp2_idx = 2 * coef_compress + coef_res;
1571 
1572  for (i = 0; i < tns->order[w][filt]; i++)
1573  tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
1574  }
1575  }
1576  }
1577  }
1578  return 0;
1579 }
1580 
1581 /**
1582  * Decode Mid/Side data; reference: table 4.54.
1583  *
1584  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
1585  * [1] mask is decoded from bitstream; [2] mask is all 1s;
1586  * [3] reserved for scalable AAC
1587  */
1588 static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
1589  int ms_present)
1590 {
1591  int idx;
1592  int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
1593  if (ms_present == 1) {
1594  for (idx = 0; idx < max_idx; idx++)
1595  cpe->ms_mask[idx] = get_bits1(gb);
1596  } else if (ms_present == 2) {
1597  memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
1598  }
1599 }
1600 
1601 /**
1602  * Decode spectral data; reference: table 4.50.
1603  * Dequantize and scale spectral data; reference: 4.6.3.3.
1604  *
1605  * @param coef array of dequantized, scaled spectral data
1606  * @param sf array of scalefactors or intensity stereo positions
1607  * @param pulse_present set if pulses are present
1608  * @param pulse pointer to pulse data struct
1609  * @param band_type array of the used band type
1610  *
1611  * @return Returns error status. 0 - OK, !0 - error
1612  */
1613 static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024],
1614  GetBitContext *gb, const INTFLOAT sf[120],
1615  int pulse_present, const Pulse *pulse,
1616  const IndividualChannelStream *ics,
1617  enum BandType band_type[120])
1618 {
1619  int i, k, g, idx = 0;
1620  const int c = 1024 / ics->num_windows;
1621  const uint16_t *offsets = ics->swb_offset;
1622  INTFLOAT *coef_base = coef;
1623 
1624  for (g = 0; g < ics->num_windows; g++)
1625  memset(coef + g * 128 + offsets[ics->max_sfb], 0,
1626  sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
1627 
1628  for (g = 0; g < ics->num_window_groups; g++) {
1629  unsigned g_len = ics->group_len[g];
1630 
1631  for (i = 0; i < ics->max_sfb; i++, idx++) {
1632  const unsigned cbt_m1 = band_type[idx] - 1;
1633  INTFLOAT *cfo = coef + offsets[i];
1634  int off_len = offsets[i + 1] - offsets[i];
1635  int group;
1636 
1637  if (cbt_m1 >= INTENSITY_BT2 - 1) {
1638  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1639  memset(cfo, 0, off_len * sizeof(*cfo));
1640  }
1641  } else if (cbt_m1 == NOISE_BT - 1) {
1642  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1643 #if !USE_FIXED
1644  float scale;
1645 #endif /* !USE_FIXED */
1646  INTFLOAT band_energy;
1647 
1648  for (k = 0; k < off_len; k++) {
1649  ac->random_state = lcg_random(ac->random_state);
1650 #if USE_FIXED
1651  cfo[k] = ac->random_state >> 3;
1652 #else
1653  cfo[k] = ac->random_state;
1654 #endif /* USE_FIXED */
1655  }
1656 
1657 #if USE_FIXED
1658  band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
1659  band_energy = fixed_sqrt(band_energy, 31);
1660  noise_scale(cfo, sf[idx], band_energy, off_len);
1661 #else
1662  band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
1663  scale = sf[idx] / sqrtf(band_energy);
1664  ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
1665 #endif /* USE_FIXED */
1666  }
1667  } else {
1668 #if !USE_FIXED
1669  const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1670 #endif /* !USE_FIXED */
1671  const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1672  VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1673  OPEN_READER(re, gb);
1674 
1675  switch (cbt_m1 >> 1) {
1676  case 0:
1677  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1678  INTFLOAT *cf = cfo;
1679  int len = off_len;
1680 
1681  do {
1682  int code;
1683  unsigned cb_idx;
1684 
1685  UPDATE_CACHE(re, gb);
1686  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1687  cb_idx = cb_vector_idx[code];
1688 #if USE_FIXED
1689  cf = DEC_SQUAD(cf, cb_idx);
1690 #else
1691  cf = VMUL4(cf, vq, cb_idx, sf + idx);
1692 #endif /* USE_FIXED */
1693  } while (len -= 4);
1694  }
1695  break;
1696 
1697  case 1:
1698  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1699  INTFLOAT *cf = cfo;
1700  int len = off_len;
1701 
1702  do {
1703  int code;
1704  unsigned nnz;
1705  unsigned cb_idx;
1706  uint32_t bits;
1707 
1708  UPDATE_CACHE(re, gb);
1709  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1710  cb_idx = cb_vector_idx[code];
1711  nnz = cb_idx >> 8 & 15;
1712  bits = nnz ? GET_CACHE(re, gb) : 0;
1713  LAST_SKIP_BITS(re, gb, nnz);
1714 #if USE_FIXED
1715  cf = DEC_UQUAD(cf, cb_idx, bits);
1716 #else
1717  cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1718 #endif /* USE_FIXED */
1719  } while (len -= 4);
1720  }
1721  break;
1722 
1723  case 2:
1724  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1725  INTFLOAT *cf = cfo;
1726  int len = off_len;
1727 
1728  do {
1729  int code;
1730  unsigned cb_idx;
1731 
1732  UPDATE_CACHE(re, gb);
1733  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1734  cb_idx = cb_vector_idx[code];
1735 #if USE_FIXED
1736  cf = DEC_SPAIR(cf, cb_idx);
1737 #else
1738  cf = VMUL2(cf, vq, cb_idx, sf + idx);
1739 #endif /* USE_FIXED */
1740  } while (len -= 2);
1741  }
1742  break;
1743 
1744  case 3:
1745  case 4:
1746  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1747  INTFLOAT *cf = cfo;
1748  int len = off_len;
1749 
1750  do {
1751  int code;
1752  unsigned nnz;
1753  unsigned cb_idx;
1754  unsigned sign;
1755 
1756  UPDATE_CACHE(re, gb);
1757  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1758  cb_idx = cb_vector_idx[code];
1759  nnz = cb_idx >> 8 & 15;
1760  sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
1761  LAST_SKIP_BITS(re, gb, nnz);
1762 #if USE_FIXED
1763  cf = DEC_UPAIR(cf, cb_idx, sign);
1764 #else
1765  cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1766 #endif /* USE_FIXED */
1767  } while (len -= 2);
1768  }
1769  break;
1770 
1771  default:
1772  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1773 #if USE_FIXED
1774  int *icf = cfo;
1775  int v;
1776 #else
1777  float *cf = cfo;
1778  uint32_t *icf = (uint32_t *) cf;
1779 #endif /* USE_FIXED */
1780  int len = off_len;
1781 
1782  do {
1783  int code;
1784  unsigned nzt, nnz;
1785  unsigned cb_idx;
1786  uint32_t bits;
1787  int j;
1788 
1789  UPDATE_CACHE(re, gb);
1790  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1791 
1792  if (!code) {
1793  *icf++ = 0;
1794  *icf++ = 0;
1795  continue;
1796  }
1797 
1798  cb_idx = cb_vector_idx[code];
1799  nnz = cb_idx >> 12;
1800  nzt = cb_idx >> 8;
1801  bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1802  LAST_SKIP_BITS(re, gb, nnz);
1803 
1804  for (j = 0; j < 2; j++) {
1805  if (nzt & 1<<j) {
1806  uint32_t b;
1807  int n;
1808  /* The total length of escape_sequence must be < 22 bits according
1809  to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1810  UPDATE_CACHE(re, gb);
1811  b = GET_CACHE(re, gb);
1812  b = 31 - av_log2(~b);
1813 
1814  if (b > 8) {
1815  av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1816  return AVERROR_INVALIDDATA;
1817  }
1818 
1819  SKIP_BITS(re, gb, b + 1);
1820  b += 4;
1821  n = (1 << b) + SHOW_UBITS(re, gb, b);
1822  LAST_SKIP_BITS(re, gb, b);
1823 #if USE_FIXED
1824  v = n;
1825  if (bits & 1U<<31)
1826  v = -v;
1827  *icf++ = v;
1828 #else
1829  *icf++ = ff_cbrt_tab[n] | (bits & 1U<<31);
1830 #endif /* USE_FIXED */
1831  bits <<= 1;
1832  } else {
1833 #if USE_FIXED
1834  v = cb_idx & 15;
1835  if (bits & 1U<<31)
1836  v = -v;
1837  *icf++ = v;
1838 #else
1839  unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1840  *icf++ = (bits & 1U<<31) | v;
1841 #endif /* USE_FIXED */
1842  bits <<= !!v;
1843  }
1844  cb_idx >>= 4;
1845  }
1846  } while (len -= 2);
1847 #if !USE_FIXED
1848  ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1849 #endif /* !USE_FIXED */
1850  }
1851  }
1852 
1853  CLOSE_READER(re, gb);
1854  }
1855  }
1856  coef += g_len << 7;
1857  }
1858 
1859  if (pulse_present) {
1860  idx = 0;
1861  for (i = 0; i < pulse->num_pulse; i++) {
1862  INTFLOAT co = coef_base[ pulse->pos[i] ];
1863  while (offsets[idx + 1] <= pulse->pos[i])
1864  idx++;
1865  if (band_type[idx] != NOISE_BT && sf[idx]) {
1866  INTFLOAT ico = -pulse->amp[i];
1867 #if USE_FIXED
1868  if (co) {
1869  ico = co + (co > 0 ? -ico : ico);
1870  }
1871  coef_base[ pulse->pos[i] ] = ico;
1872 #else
1873  if (co) {
1874  co /= sf[idx];
1875  ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1876  }
1877  coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1878 #endif /* USE_FIXED */
1879  }
1880  }
1881  }
1882 #if USE_FIXED
1883  coef = coef_base;
1884  idx = 0;
1885  for (g = 0; g < ics->num_window_groups; g++) {
1886  unsigned g_len = ics->group_len[g];
1887 
1888  for (i = 0; i < ics->max_sfb; i++, idx++) {
1889  const unsigned cbt_m1 = band_type[idx] - 1;
1890  int *cfo = coef + offsets[i];
1891  int off_len = offsets[i + 1] - offsets[i];
1892  int group;
1893 
1894  if (cbt_m1 < NOISE_BT - 1) {
1895  for (group = 0; group < (int)g_len; group++, cfo+=128) {
1896  ac->vector_pow43(cfo, off_len);
1897  ac->subband_scale(cfo, cfo, sf[idx], 34, off_len);
1898  }
1899  }
1900  }
1901  coef += g_len << 7;
1902  }
1903 #endif /* USE_FIXED */
1904  return 0;
1905 }
1906 
1907 /**
1908  * Apply AAC-Main style frequency domain prediction.
1909  */
1910 static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1911 {
1912  int sfb, k;
1913 
1914  if (!sce->ics.predictor_initialized) {
1915  reset_all_predictors(sce->predictor_state);
1916  sce->ics.predictor_initialized = 1;
1917  }
1918 
1919  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1920  for (sfb = 0;
1921  sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
1922  sfb++) {
1923  for (k = sce->ics.swb_offset[sfb];
1924  k < sce->ics.swb_offset[sfb + 1];
1925  k++) {
1926  predict(&sce->predictor_state[k], &sce->coeffs[k],
1927  sce->ics.predictor_present &&
1928  sce->ics.prediction_used[sfb]);
1929  }
1930  }
1931  if (sce->ics.predictor_reset_group)
1932  reset_predictor_group(sce->predictor_state,
1933  sce->ics.predictor_reset_group);
1934  } else
1935  reset_all_predictors(sce->predictor_state);
1936 }
1937 
1938 /**
1939  * Decode an individual_channel_stream payload; reference: table 4.44.
1940  *
1941  * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
1942  * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1943  *
1944  * @return Returns error status. 0 - OK, !0 - error
1945  */
1946 static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1947  GetBitContext *gb, int common_window, int scale_flag)
1948 {
1949  Pulse pulse;
1950  TemporalNoiseShaping *tns = &sce->tns;
1951  IndividualChannelStream *ics = &sce->ics;
1952  INTFLOAT *out = sce->coeffs;
1953  int global_gain, eld_syntax, er_syntax, pulse_present = 0;
1954  int ret;
1955 
1956  eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1957  er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
1958  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
1959  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
1960  ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1961 
1962  /* This assignment is to silence a GCC warning about the variable being used
1963  * uninitialized when in fact it always is.
1964  */
1965  pulse.num_pulse = 0;
1966 
1967  global_gain = get_bits(gb, 8);
1968 
1969  if (!common_window && !scale_flag) {
1970  ret = decode_ics_info(ac, ics, gb);
1971  if (ret < 0)
1972  goto fail;
1973  }
1974 
1975  if ((ret = decode_band_types(ac, sce->band_type,
1976  sce->band_type_run_end, gb, ics)) < 0)
1977  goto fail;
1978  if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
1979  sce->band_type, sce->band_type_run_end)) < 0)
1980  goto fail;
1981 
1982  pulse_present = 0;
1983  if (!scale_flag) {
1984  if (!eld_syntax && (pulse_present = get_bits1(gb))) {
1985  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1986  av_log(ac->avctx, AV_LOG_ERROR,
1987  "Pulse tool not allowed in eight short sequence.\n");
1988  ret = AVERROR_INVALIDDATA;
1989  goto fail;
1990  }
1991  if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1992  av_log(ac->avctx, AV_LOG_ERROR,
1993  "Pulse data corrupt or invalid.\n");
1994  ret = AVERROR_INVALIDDATA;
1995  goto fail;
1996  }
1997  }
1998  tns->present = get_bits1(gb);
1999  if (tns->present && !er_syntax) {
2000  ret = decode_tns(ac, tns, gb, ics);
2001  if (ret < 0)
2002  goto fail;
2003  }
2004  if (!eld_syntax && get_bits1(gb)) {
2005  avpriv_request_sample(ac->avctx, "SSR");
2006  ret = AVERROR_PATCHWELCOME;
2007  goto fail;
2008  }
2009  // I see no textual basis in the spec for this occurring after SSR gain
2010  // control, but this is what both reference and real implmentations do
2011  if (tns->present && er_syntax) {
2012  ret = decode_tns(ac, tns, gb, ics);
2013  if (ret < 0)
2014  goto fail;
2015  }
2016  }
2017 
2018  ret = decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
2019  &pulse, ics, sce->band_type);
2020  if (ret < 0)
2021  goto fail;
2022 
2023  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
2024  apply_prediction(ac, sce);
2025 
2026  return 0;
2027 fail:
2028  tns->present = 0;
2029  return ret;
2030 }
2031 
2032 /**
2033  * Mid/Side stereo decoding; reference: 4.6.8.1.3.
2034  */
2035 static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
2036 {
2037  const IndividualChannelStream *ics = &cpe->ch[0].ics;
2038  INTFLOAT *ch0 = cpe->ch[0].coeffs;
2039  INTFLOAT *ch1 = cpe->ch[1].coeffs;
2040  int g, i, group, idx = 0;
2041  const uint16_t *offsets = ics->swb_offset;
2042  for (g = 0; g < ics->num_window_groups; g++) {
2043  for (i = 0; i < ics->max_sfb; i++, idx++) {
2044  if (cpe->ms_mask[idx] &&
2045  cpe->ch[0].band_type[idx] < NOISE_BT &&
2046  cpe->ch[1].band_type[idx] < NOISE_BT) {
2047 #if USE_FIXED
2048  for (group = 0; group < ics->group_len[g]; group++) {
2049  ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
2050  ch1 + group * 128 + offsets[i],
2051  offsets[i+1] - offsets[i]);
2052 #else
2053  for (group = 0; group < ics->group_len[g]; group++) {
2054  ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
2055  ch1 + group * 128 + offsets[i],
2056  offsets[i+1] - offsets[i]);
2057 #endif /* USE_FIXED */
2058  }
2059  }
2060  }
2061  ch0 += ics->group_len[g] * 128;
2062  ch1 += ics->group_len[g] * 128;
2063  }
2064 }
2065 
2066 /**
2067  * intensity stereo decoding; reference: 4.6.8.2.3
2068  *
2069  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
2070  * [1] mask is decoded from bitstream; [2] mask is all 1s;
2071  * [3] reserved for scalable AAC
2072  */
2073 static void apply_intensity_stereo(AACContext *ac,
2074  ChannelElement *cpe, int ms_present)
2075 {
2076  const IndividualChannelStream *ics = &cpe->ch[1].ics;
2077  SingleChannelElement *sce1 = &cpe->ch[1];
2078  INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
2079  const uint16_t *offsets = ics->swb_offset;
2080  int g, group, i, idx = 0;
2081  int c;
2082  INTFLOAT scale;
2083  for (g = 0; g < ics->num_window_groups; g++) {
2084  for (i = 0; i < ics->max_sfb;) {
2085  if (sce1->band_type[idx] == INTENSITY_BT ||
2086  sce1->band_type[idx] == INTENSITY_BT2) {
2087  const int bt_run_end = sce1->band_type_run_end[idx];
2088  for (; i < bt_run_end; i++, idx++) {
2089  c = -1 + 2 * (sce1->band_type[idx] - 14);
2090  if (ms_present)
2091  c *= 1 - 2 * cpe->ms_mask[idx];
2092  scale = c * sce1->sf[idx];
2093  for (group = 0; group < ics->group_len[g]; group++)
2094 #if USE_FIXED
2095  ac->subband_scale(coef1 + group * 128 + offsets[i],
2096  coef0 + group * 128 + offsets[i],
2097  scale,
2098  23,
2099  offsets[i + 1] - offsets[i]);
2100 #else
2101  ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
2102  coef0 + group * 128 + offsets[i],
2103  scale,
2104  offsets[i + 1] - offsets[i]);
2105 #endif /* USE_FIXED */
2106  }
2107  } else {
2108  int bt_run_end = sce1->band_type_run_end[idx];
2109  idx += bt_run_end - i;
2110  i = bt_run_end;
2111  }
2112  }
2113  coef0 += ics->group_len[g] * 128;
2114  coef1 += ics->group_len[g] * 128;
2115  }
2116 }
2117 
2118 /**
2119  * Decode a channel_pair_element; reference: table 4.4.
2120  *
2121  * @return Returns error status. 0 - OK, !0 - error
2122  */
2123 static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
2124 {
2125  int i, ret, common_window, ms_present = 0;
2126  int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2127 
2128  common_window = eld_syntax || get_bits1(gb);
2129  if (common_window) {
2130  if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
2131  return AVERROR_INVALIDDATA;
2132  i = cpe->ch[1].ics.use_kb_window[0];
2133  cpe->ch[1].ics = cpe->ch[0].ics;
2134  cpe->ch[1].ics.use_kb_window[1] = i;
2135  if (cpe->ch[1].ics.predictor_present &&
2136  (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
2137  if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
2138  decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
2139  ms_present = get_bits(gb, 2);
2140  if (ms_present == 3) {
2141  av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
2142  return AVERROR_INVALIDDATA;
2143  } else if (ms_present)
2144  decode_mid_side_stereo(cpe, gb, ms_present);
2145  }
2146  if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
2147  return ret;
2148  if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
2149  return ret;
2150 
2151  if (common_window) {
2152  if (ms_present)
2153  apply_mid_side_stereo(ac, cpe);
2154  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
2155  apply_prediction(ac, &cpe->ch[0]);
2156  apply_prediction(ac, &cpe->ch[1]);
2157  }
2158  }
2159 
2160  apply_intensity_stereo(ac, cpe, ms_present);
2161  return 0;
2162 }
2163 
2164 static const float cce_scale[] = {
2165  1.09050773266525765921, //2^(1/8)
2166  1.18920711500272106672, //2^(1/4)
2167  M_SQRT2,
2168  2,
2169 };
2170 
2171 /**
2172  * Decode coupling_channel_element; reference: table 4.8.
2173  *
2174  * @return Returns error status. 0 - OK, !0 - error
2175  */
2176 static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
2177 {
2178  int num_gain = 0;
2179  int c, g, sfb, ret;
2180  int sign;
2181  INTFLOAT scale;
2182  SingleChannelElement *sce = &che->ch[0];
2183  ChannelCoupling *coup = &che->coup;
2184 
2185  coup->coupling_point = 2 * get_bits1(gb);
2186  coup->num_coupled = get_bits(gb, 3);
2187  for (c = 0; c <= coup->num_coupled; c++) {
2188  num_gain++;
2189  coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
2190  coup->id_select[c] = get_bits(gb, 4);
2191  if (coup->type[c] == TYPE_CPE) {
2192  coup->ch_select[c] = get_bits(gb, 2);
2193  if (coup->ch_select[c] == 3)
2194  num_gain++;
2195  } else
2196  coup->ch_select[c] = 2;
2197  }
2198  coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
2199 
2200  sign = get_bits(gb, 1);
2201 #if USE_FIXED
2202  scale = get_bits(gb, 2);
2203 #else
2204  scale = cce_scale[get_bits(gb, 2)];
2205 #endif
2206 
2207  if ((ret = decode_ics(ac, sce, gb, 0, 0)))
2208  return ret;
2209 
2210  for (c = 0; c < num_gain; c++) {
2211  int idx = 0;
2212  int cge = 1;
2213  int gain = 0;
2214  INTFLOAT gain_cache = FIXR10(1.);
2215  if (c) {
2216  cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
2217  gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
2218  gain_cache = GET_GAIN(scale, gain);
2219 #if USE_FIXED
2220  if ((abs(gain_cache)-1024) >> 3 > 30)
2221  return AVERROR(ERANGE);
2222 #endif
2223  }
2224  if (coup->coupling_point == AFTER_IMDCT) {
2225  coup->gain[c][0] = gain_cache;
2226  } else {
2227  for (g = 0; g < sce->ics.num_window_groups; g++) {
2228  for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
2229  if (sce->band_type[idx] != ZERO_BT) {
2230  if (!cge) {
2231  int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
2232  if (t) {
2233  int s = 1;
2234  t = gain += t;
2235  if (sign) {
2236  s -= 2 * (t & 0x1);
2237  t >>= 1;
2238  }
2239  gain_cache = GET_GAIN(scale, t) * s;
2240 #if USE_FIXED
2241  if ((abs(gain_cache)-1024) >> 3 > 30)
2242  return AVERROR(ERANGE);
2243 #endif
2244  }
2245  }
2246  coup->gain[c][idx] = gain_cache;
2247  }
2248  }
2249  }
2250  }
2251  }
2252  return 0;
2253 }
2254 
2255 /**
2256  * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
2257  *
2258  * @return Returns number of bytes consumed.
2259  */
2260 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
2261  GetBitContext *gb)
2262 {
2263  int i;
2264  int num_excl_chan = 0;
2265 
2266  do {
2267  for (i = 0; i < 7; i++)
2268  che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
2269  } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
2270 
2271  return num_excl_chan / 7;
2272 }
2273 
2274 /**
2275  * Decode dynamic range information; reference: table 4.52.
2276  *
2277  * @return Returns number of bytes consumed.
2278  */
2279 static int decode_dynamic_range(DynamicRangeControl *che_drc,
2280  GetBitContext *gb)
2281 {
2282  int n = 1;
2283  int drc_num_bands = 1;
2284  int i;
2285 
2286  /* pce_tag_present? */
2287  if (get_bits1(gb)) {
2288  che_drc->pce_instance_tag = get_bits(gb, 4);
2289  skip_bits(gb, 4); // tag_reserved_bits
2290  n++;
2291  }
2292 
2293  /* excluded_chns_present? */
2294  if (get_bits1(gb)) {
2295  n += decode_drc_channel_exclusions(che_drc, gb);
2296  }
2297 
2298  /* drc_bands_present? */
2299  if (get_bits1(gb)) {
2300  che_drc->band_incr = get_bits(gb, 4);
2301  che_drc->interpolation_scheme = get_bits(gb, 4);
2302  n++;
2303  drc_num_bands += che_drc->band_incr;
2304  for (i = 0; i < drc_num_bands; i++) {
2305  che_drc->band_top[i] = get_bits(gb, 8);
2306  n++;
2307  }
2308  }
2309 
2310  /* prog_ref_level_present? */
2311  if (get_bits1(gb)) {
2312  che_drc->prog_ref_level = get_bits(gb, 7);
2313  skip_bits1(gb); // prog_ref_level_reserved_bits
2314  n++;
2315  }
2316 
2317  for (i = 0; i < drc_num_bands; i++) {
2318  che_drc->dyn_rng_sgn[i] = get_bits1(gb);
2319  che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
2320  n++;
2321  }
2322 
2323  return n;
2324 }
2325 
2326 static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
2327  uint8_t buf[256];
2328  int i, major, minor;
2329 
2330  if (len < 13+7*8)
2331  goto unknown;
2332 
2333  get_bits(gb, 13); len -= 13;
2334 
2335  for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
2336  buf[i] = get_bits(gb, 8);
2337 
2338  buf[i] = 0;
2339  if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
2340  av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
2341 
2342  if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
2343  ac->avctx->internal->skip_samples = 1024;
2344  }
2345 
2346 unknown:
2347  skip_bits_long(gb, len);
2348 
2349  return 0;
2350 }
2351 
2352 /**
2353  * Decode extension data (incomplete); reference: table 4.51.
2354  *
2355  * @param cnt length of TYPE_FIL syntactic element in bytes
2356  *
2357  * @return Returns number of bytes consumed
2358  */
2359 static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
2360  ChannelElement *che, enum RawDataBlockType elem_type)
2361 {
2362  int crc_flag = 0;
2363  int res = cnt;
2364  int type = get_bits(gb, 4);
2365 
2366  if (ac->avctx->debug & FF_DEBUG_STARTCODE)
2367  av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
2368 
2369  switch (type) { // extension type
2370  case EXT_SBR_DATA_CRC:
2371  crc_flag++;
2372  case EXT_SBR_DATA:
2373  if (!che) {
2374  av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
2375  return res;
2376  } else if (!ac->oc[1].m4ac.sbr) {
2377  av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
2378  skip_bits_long(gb, 8 * cnt - 4);
2379  return res;
2380  } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
2381  av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
2382  skip_bits_long(gb, 8 * cnt - 4);
2383  return res;
2384  } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
2385  ac->oc[1].m4ac.sbr = 1;
2386  ac->oc[1].m4ac.ps = 1;
2387  ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
2388  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
2389  ac->oc[1].status, 1);
2390  } else {
2391  ac->oc[1].m4ac.sbr = 1;
2392  ac->avctx->profile = FF_PROFILE_AAC_HE;
2393  }
2394  res = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
2395  break;
2396  case EXT_DYNAMIC_RANGE:
2397  res = decode_dynamic_range(&ac->che_drc, gb);
2398  break;
2399  case EXT_FILL:
2400  decode_fill(ac, gb, 8 * cnt - 4);
2401  break;
2402  case EXT_FILL_DATA:
2403  case EXT_DATA_ELEMENT:
2404  default:
2405  skip_bits_long(gb, 8 * cnt - 4);
2406  break;
2407  };
2408  return res;
2409 }
2410 
2411 /**
2412  * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
2413  *
2414  * @param decode 1 if tool is used normally, 0 if tool is used in LTP.
2415  * @param coef spectral coefficients
2416  */
2417 static void apply_tns(INTFLOAT coef_param[1024], TemporalNoiseShaping *tns,
2418  IndividualChannelStream *ics, int decode)
2419 {
2420  const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
2421  int w, filt, m, i;
2422  int bottom, top, order, start, end, size, inc;
2423  INTFLOAT lpc[TNS_MAX_ORDER];
2424  INTFLOAT tmp[TNS_MAX_ORDER+1];
2425  UINTFLOAT *coef = coef_param;
2426 
2427  for (w = 0; w < ics->num_windows; w++) {
2428  bottom = ics->num_swb;
2429  for (filt = 0; filt < tns->n_filt[w]; filt++) {
2430  top = bottom;
2431  bottom = FFMAX(0, top - tns->length[w][filt]);
2432  order = tns->order[w][filt];
2433  if (order == 0)
2434  continue;
2435 
2436  // tns_decode_coef
2437  AAC_RENAME(compute_lpc_coefs)(tns->coef[w][filt], order, lpc, 0, 0, 0);
2438 
2439  start = ics->swb_offset[FFMIN(bottom, mmm)];
2440  end = ics->swb_offset[FFMIN( top, mmm)];
2441  if ((size = end - start) <= 0)
2442  continue;
2443  if (tns->direction[w][filt]) {
2444  inc = -1;
2445  start = end - 1;
2446  } else {
2447  inc = 1;
2448  }
2449  start += w * 128;
2450 
2451  if (decode) {
2452  // ar filter
2453  for (m = 0; m < size; m++, start += inc)
2454  for (i = 1; i <= FFMIN(m, order); i++)
2455  coef[start] -= AAC_MUL26((INTFLOAT)coef[start - i * inc], lpc[i - 1]);
2456  } else {
2457  // ma filter
2458  for (m = 0; m < size; m++, start += inc) {
2459  tmp[0] = coef[start];
2460  for (i = 1; i <= FFMIN(m, order); i++)
2461  coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
2462  for (i = order; i > 0; i--)
2463  tmp[i] = tmp[i - 1];
2464  }
2465  }
2466  }
2467  }
2468 }
2469 
2470 /**
2471  * Apply windowing and MDCT to obtain the spectral
2472  * coefficient from the predicted sample by LTP.
2473  */
2474 static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out,
2475  INTFLOAT *in, IndividualChannelStream *ics)
2476 {
2477  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2478  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2479  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2480  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2481 
2482  if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
2483  ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
2484  } else {
2485  memset(in, 0, 448 * sizeof(*in));
2486  ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
2487  }
2488  if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
2489  ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
2490  } else {
2491  ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
2492  memset(in + 1024 + 576, 0, 448 * sizeof(*in));
2493  }
2494  ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
2495 }
2496 
2497 /**
2498  * Apply the long term prediction
2499  */
2500 static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
2501 {
2502  const LongTermPrediction *ltp = &sce->ics.ltp;
2503  const uint16_t *offsets = sce->ics.swb_offset;
2504  int i, sfb;
2505 
2506  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
2507  INTFLOAT *predTime = sce->ret;
2508  INTFLOAT *predFreq = ac->buf_mdct;
2509  int16_t num_samples = 2048;
2510 
2511  if (ltp->lag < 1024)
2512  num_samples = ltp->lag + 1024;
2513  for (i = 0; i < num_samples; i++)
2514  predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
2515  memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));
2516 
2517  ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
2518 
2519  if (sce->tns.present)
2520  ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
2521 
2522  for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
2523  if (ltp->used[sfb])
2524  for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
2525  sce->coeffs[i] += predFreq[i];
2526  }
2527 }
2528 
2529 /**
2530  * Update the LTP buffer for next frame
2531  */
2532 static void update_ltp(AACContext *ac, SingleChannelElement *sce)
2533 {
2534  IndividualChannelStream *ics = &sce->ics;
2535  INTFLOAT *saved = sce->saved;
2536  INTFLOAT *saved_ltp = sce->coeffs;
2537  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2538  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2539  int i;
2540 
2541  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2542  memcpy(saved_ltp, saved, 512 * sizeof(*saved_ltp));
2543  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2544  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2545 
2546  for (i = 0; i < 64; i++)
2547  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2548  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2549  memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(*saved_ltp));
2550  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2551  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2552 
2553  for (i = 0; i < 64; i++)
2554  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2555  } else { // LONG_STOP or ONLY_LONG
2556  ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
2557 
2558  for (i = 0; i < 512; i++)
2559  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
2560  }
2561 
2562  memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
2563  memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
2564  memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
2565 }
2566 
2567 /**
2568  * Conduct IMDCT and windowing.
2569  */
2570 static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
2571 {
2572  IndividualChannelStream *ics = &sce->ics;
2573  INTFLOAT *in = sce->coeffs;
2574  INTFLOAT *out = sce->ret;
2575  INTFLOAT *saved = sce->saved;
2576  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2577  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2578  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2579  INTFLOAT *buf = ac->buf_mdct;
2580  INTFLOAT *temp = ac->temp;
2581  int i;
2582 
2583  // imdct
2584  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2585  for (i = 0; i < 1024; i += 128)
2586  ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
2587  } else {
2588  ac->mdct.imdct_half(&ac->mdct, buf, in);
2589 #if USE_FIXED
2590  for (i=0; i<1024; i++)
2591  buf[i] = (buf[i] + 4) >> 3;
2592 #endif /* USE_FIXED */
2593  }
2594 
2595  /* window overlapping
2596  * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2597  * and long to short transitions are considered to be short to short
2598  * transitions. This leaves just two cases (long to long and short to short)
2599  * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2600  */
2601  if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2602  (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
2603  ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
2604  } else {
2605  memcpy( out, saved, 448 * sizeof(*out));
2606 
2607  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2608  ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
2609  ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
2610  ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
2611  ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
2612  ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
2613  memcpy( out + 448 + 4*128, temp, 64 * sizeof(*out));
2614  } else {
2615  ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
2616  memcpy( out + 576, buf + 64, 448 * sizeof(*out));
2617  }
2618  }
2619 
2620  // buffer update
2621  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2622  memcpy( saved, temp + 64, 64 * sizeof(*saved));
2623  ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
2624  ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
2625  ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
2626  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2627  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2628  memcpy( saved, buf + 512, 448 * sizeof(*saved));
2629  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2630  } else { // LONG_STOP or ONLY_LONG
2631  memcpy( saved, buf + 512, 512 * sizeof(*saved));
2632  }
2633 }
2634 
2635 static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
2636 {
2637  IndividualChannelStream *ics = &sce->ics;
2638  INTFLOAT *in = sce->coeffs;
2639  INTFLOAT *out = sce->ret;
2640  INTFLOAT *saved = sce->saved;
2641  INTFLOAT *buf = ac->buf_mdct;
2642 #if USE_FIXED
2643  int i;
2644 #endif /* USE_FIXED */
2645 
2646  // imdct
2647  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2648 
2649 #if USE_FIXED
2650  for (i = 0; i < 1024; i++)
2651  buf[i] = (buf[i] + 2) >> 2;
2652 #endif /* USE_FIXED */
2653 
2654  // window overlapping
2655  if (ics->use_kb_window[1]) {
2656  // AAC LD uses a low overlap sine window instead of a KBD window
2657  memcpy(out, saved, 192 * sizeof(*out));
2658  ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
2659  memcpy( out + 320, buf + 64, 192 * sizeof(*out));
2660  } else {
2661  ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);
2662  }
2663 
2664  // buffer update
2665  memcpy(saved, buf + 256, 256 * sizeof(*saved));
2666 }
2667 
2668 static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
2669 {
2670  INTFLOAT *in = sce->coeffs;
2671  INTFLOAT *out = sce->ret;
2672  INTFLOAT *saved = sce->saved;
2673  INTFLOAT *buf = ac->buf_mdct;
2674  int i;
2675  const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
2676  const int n2 = n >> 1;
2677  const int n4 = n >> 2;
2678  const INTFLOAT *const window = n == 480 ? AAC_RENAME(ff_aac_eld_window_480) :
2679  AAC_RENAME(ff_aac_eld_window_512);
2680 
2681  // Inverse transform, mapped to the conventional IMDCT by
2682  // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
2683  // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
2684  // International Conference on Audio, Language and Image Processing, ICALIP 2008.
2685  // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
2686  for (i = 0; i < n2; i+=2) {
2687  INTFLOAT temp;
2688  temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
2689  temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
2690  }
2691 #if !USE_FIXED
2692  if (n == 480)
2693  ac->mdct480->imdct_half(ac->mdct480, buf, in, 1, -1.f/(16*1024*960));
2694  else
2695 #endif
2696  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2697 
2698 #if USE_FIXED
2699  for (i = 0; i < 1024; i++)
2700  buf[i] = (buf[i] + 1) >> 1;
2701 #endif /* USE_FIXED */
2702 
2703  for (i = 0; i < n; i+=2) {
2704  buf[i] = -buf[i];
2705  }
2706  // Like with the regular IMDCT at this point we still have the middle half
2707  // of a transform but with even symmetry on the left and odd symmetry on
2708  // the right
2709 
2710  // window overlapping
2711  // The spec says to use samples [0..511] but the reference decoder uses
2712  // samples [128..639].
2713  for (i = n4; i < n2; i ++) {
2714  out[i - n4] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) +
2715  AAC_MUL31( saved[ i + n2] , window[i + n - n4]) +
2716  AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
2717  AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]);
2718  }
2719  for (i = 0; i < n2; i ++) {
2720  out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) +
2721  AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) +
2722  AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) +
2723  AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
2724  }
2725  for (i = 0; i < n4; i ++) {
2726  out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) +
2727  AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
2728  AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
2729  }
2730 
2731  // buffer update
2732  memmove(saved + n, saved, 2 * n * sizeof(*saved));
2733  memcpy( saved, buf, n * sizeof(*saved));
2734 }
2735 
2736 /**
2737  * channel coupling transformation interface
2738  *
2739  * @param apply_coupling_method pointer to (in)dependent coupling function
2740  */
2741 static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
2742  enum RawDataBlockType type, int elem_id,
2743  enum CouplingPoint coupling_point,
2744  void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
2745 {
2746  int i, c;
2747 
2748  for (i = 0; i < MAX_ELEM_ID; i++) {
2749  ChannelElement *cce = ac->che[TYPE_CCE][i];
2750  int index = 0;
2751 
2752  if (cce && cce->coup.coupling_point == coupling_point) {
2753  ChannelCoupling *coup = &cce->coup;
2754 
2755  for (c = 0; c <= coup->num_coupled; c++) {
2756  if (coup->type[c] == type && coup->id_select[c] == elem_id) {
2757  if (coup->ch_select[c] != 1) {
2758  apply_coupling_method(ac, &cc->ch[0], cce, index);
2759  if (coup->ch_select[c] != 0)
2760  index++;
2761  }
2762  if (coup->ch_select[c] != 2)
2763  apply_coupling_method(ac, &cc->ch[1], cce, index++);
2764  } else
2765  index += 1 + (coup->ch_select[c] == 3);
2766  }
2767  }
2768  }
2769 }
2770 
2771 /**
2772  * Convert spectral data to samples, applying all supported tools as appropriate.
2773  */
2774 static void spectral_to_sample(AACContext *ac, int samples)
2775 {
2776  int i, type;
2777  void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
2778  switch (ac->oc[1].m4ac.object_type) {
2779  case AOT_ER_AAC_LD:
2780  imdct_and_window = imdct_and_windowing_ld;
2781  break;
2782  case AOT_ER_AAC_ELD:
2783  imdct_and_window = imdct_and_windowing_eld;
2784  break;
2785  default:
2786  imdct_and_window = ac->imdct_and_windowing;
2787  }
2788  for (type = 3; type >= 0; type--) {
2789  for (i = 0; i < MAX_ELEM_ID; i++) {
2790  ChannelElement *che = ac->che[type][i];
2791  if (che && che->present) {
2792  if (type <= TYPE_CPE)
2793  apply_channel_coupling(ac, che, type, i, BEFORE_TNS, AAC_RENAME(apply_dependent_coupling));
2794  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2795  if (che->ch[0].ics.predictor_present) {
2796  if (che->ch[0].ics.ltp.present)
2797  ac->apply_ltp(ac, &che->ch[0]);
2798  if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
2799  ac->apply_ltp(ac, &che->ch[1]);
2800  }
2801  }
2802  if (che->ch[0].tns.present)
2803  ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2804  if (che->ch[1].tns.present)
2805  ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2806  if (type <= TYPE_CPE)
2807  apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, AAC_RENAME(apply_dependent_coupling));
2808  if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2809  imdct_and_window(ac, &che->ch[0]);
2810  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2811  ac->update_ltp(ac, &che->ch[0]);
2812  if (type == TYPE_CPE) {
2813  imdct_and_window(ac, &che->ch[1]);
2814  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2815  ac->update_ltp(ac, &che->ch[1]);
2816  }
2817  if (ac->oc[1].m4ac.sbr > 0) {
2818  AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2819  }
2820  }
2821  if (type <= TYPE_CCE)
2822  apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, AAC_RENAME(apply_independent_coupling));
2823 
2824 #if USE_FIXED
2825  {
2826  int j;
2827  /* preparation for resampler */
2828  for(j = 0; j<samples; j++){
2829  che->ch[0].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[0].ret[j]<<7)+0x8000;
2830  if(type == TYPE_CPE)
2831  che->ch[1].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[1].ret[j]<<7)+0x8000;
2832  }
2833  }
2834 #endif /* USE_FIXED */
2835  che->present = 0;
2836  } else if (che) {
2837  av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
2838  }
2839  }
2840  }
2841 }
2842 
2843 static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
2844 {
2845  int size;
2846  AACADTSHeaderInfo hdr_info;
2847  uint8_t layout_map[MAX_ELEM_ID*4][3];
2848  int layout_map_tags, ret;
2849 
2850  size = avpriv_aac_parse_header(gb, &hdr_info);
2851  if (size > 0) {
2852  if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
2853  // This is 2 for "VLB " audio in NSV files.
2854  // See samples/nsv/vlb_audio.
2855  avpriv_report_missing_feature(ac->avctx,
2856  "More than one AAC RDB per ADTS frame");
2857  ac->warned_num_aac_frames = 1;
2858  }
2859  push_output_configuration(ac);
2860  if (hdr_info.chan_config) {
2861  ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
2862  if ((ret = set_default_channel_config(ac->avctx,
2863  layout_map,
2864  &layout_map_tags,
2865  hdr_info.chan_config)) < 0)
2866  return ret;
2867  if ((ret = output_configure(ac, layout_map, layout_map_tags,
2868  FFMAX(ac->oc[1].status,
2869  OC_TRIAL_FRAME), 0)) < 0)
2870  return ret;
2871  } else {
2872  ac->oc[1].m4ac.chan_config = 0;
2873  /**
2874  * dual mono frames in Japanese DTV can have chan_config 0
2875  * WITHOUT specifying PCE.
2876  * thus, set dual mono as default.
2877  */
2878  if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
2879  layout_map_tags = 2;
2880  layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
2881  layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
2882  layout_map[0][1] = 0;
2883  layout_map[1][1] = 1;
2884  if (output_configure(ac, layout_map, layout_map_tags,
2885  OC_TRIAL_FRAME, 0))
2886  return -7;
2887  }
2888  }
2889  ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
2890  ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
2891  ac->oc[1].m4ac.object_type = hdr_info.object_type;
2892  ac->oc[1].m4ac.frame_length_short = 0;
2893  if (ac->oc[0].status != OC_LOCKED ||
2894  ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
2895  ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
2896  ac->oc[1].m4ac.sbr = -1;
2897  ac->oc[1].m4ac.ps = -1;
2898  }
2899  if (!hdr_info.crc_absent)
2900  skip_bits(gb, 16);
2901  }
2902  return size;
2903 }
2904 
2905 static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
2906  int *got_frame_ptr, GetBitContext *gb)
2907 {
2908  AACContext *ac = avctx->priv_data;
2909  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
2910  ChannelElement *che;
2911  int err, i;
2912  int samples = m4ac->frame_length_short ? 960 : 1024;
2913  int chan_config = m4ac->chan_config;
2914  int aot = m4ac->object_type;
2915 
2916  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
2917  samples >>= 1;
2918 
2919  ac->frame = data;
2920 
2921  if ((err = frame_configure_elements(avctx)) < 0)
2922  return err;
2923 
2924  // The FF_PROFILE_AAC_* defines are all object_type - 1
2925  // This may lead to an undefined profile being signaled
2926  ac->avctx->profile = aot - 1;
2927 
2928  ac->tags_mapped = 0;
2929 
2930  if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
2931  avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
2932  chan_config);
2933  return AVERROR_INVALIDDATA;
2934  }
2935  for (i = 0; i < tags_per_config[chan_config]; i++) {
2936  const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
2937  const int elem_id = aac_channel_layout_map[chan_config-1][i][1];
2938  if (!(che=get_che(ac, elem_type, elem_id))) {
2939  av_log(ac->avctx, AV_LOG_ERROR,
2940  "channel element %d.%d is not allocated\n",
2941  elem_type, elem_id);
2942  return AVERROR_INVALIDDATA;
2943  }
2944  che->present = 1;
2945  if (aot != AOT_ER_AAC_ELD)
2946  skip_bits(gb, 4);
2947  switch (elem_type) {
2948  case TYPE_SCE:
2949  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2950  break;
2951  case TYPE_CPE:
2952  err = decode_cpe(ac, gb, che);
2953  break;
2954  case TYPE_LFE:
2955  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2956  break;
2957  }
2958  if (err < 0)
2959  return err;
2960  }
2961 
2962  spectral_to_sample(ac, samples);
2963 
2964  if (!ac->frame->data[0] && samples) {
2965  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
2966  return AVERROR_INVALIDDATA;
2967  }
2968 
2969  ac->frame->nb_samples = samples;
2970  ac->frame->sample_rate = avctx->sample_rate;
2971  *got_frame_ptr = 1;
2972 
2973  skip_bits_long(gb, get_bits_left(gb));
2974  return 0;
2975 }
2976 
2977 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
2978  int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
2979 {
2980  AACContext *ac = avctx->priv_data;
2981  ChannelElement *che = NULL, *che_prev = NULL;
2982  enum RawDataBlockType elem_type, che_prev_type = TYPE_END;
2983  int err, elem_id;
2984  int samples = 0, multiplier, audio_found = 0, pce_found = 0;
2985  int is_dmono, sce_count = 0;
2986  int payload_alignment;
2987 
2988  ac->frame = data;
2989 
2990  if (show_bits(gb, 12) == 0xfff) {
2991  if ((err = parse_adts_frame_header(ac, gb)) < 0) {
2992  av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
2993  goto fail;
2994  }
2995  if (ac->oc[1].m4ac.sampling_index > 12) {
2996  av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
2997  err = AVERROR_INVALIDDATA;
2998  goto fail;
2999  }
3000  }
3001 
3002  if ((err = frame_configure_elements(avctx)) < 0)
3003  goto fail;
3004 
3005  // The FF_PROFILE_AAC_* defines are all object_type - 1
3006  // This may lead to an undefined profile being signaled
3007  ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
3008 
3009  payload_alignment = get_bits_count(gb);
3010  ac->tags_mapped = 0;
3011  // parse
3012  while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
3013  elem_id = get_bits(gb, 4);
3014 
3015  if (avctx->debug & FF_DEBUG_STARTCODE)
3016  av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
3017 
3018  if (!avctx->channels && elem_type != TYPE_PCE) {
3019  err = AVERROR_INVALIDDATA;
3020  goto fail;
3021  }
3022 
3023  if (elem_type < TYPE_DSE) {
3024  if (!(che=get_che(ac, elem_type, elem_id))) {
3025  av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
3026  elem_type, elem_id);
3027  err = AVERROR_INVALIDDATA;
3028  goto fail;
3029  }
3030  samples = 1024;
3031  che->present = 1;
3032  }
3033 
3034  switch (elem_type) {
3035 
3036  case TYPE_SCE:
3037  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3038  audio_found = 1;
3039  sce_count++;
3040  break;
3041 
3042  case TYPE_CPE:
3043  err = decode_cpe(ac, gb, che);
3044  audio_found = 1;
3045  break;
3046 
3047  case TYPE_CCE:
3048  err = decode_cce(ac, gb, che);
3049  break;
3050 
3051  case TYPE_LFE:
3052  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3053  audio_found = 1;
3054  break;
3055 
3056  case TYPE_DSE:
3057  err = skip_data_stream_element(ac, gb);
3058  break;
3059 
3060  case TYPE_PCE: {
3061  uint8_t layout_map[MAX_ELEM_ID*4][3];
3062  int tags;
3063 
3064  int pushed = push_output_configuration(ac);
3065  if (pce_found && !pushed) {
3066  err = AVERROR_INVALIDDATA;
3067  goto fail;
3068  }
3069 
3070  tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb,
3071  payload_alignment);
3072  if (tags < 0) {
3073  err = tags;
3074  break;
3075  }
3076  if (pce_found) {
3077  av_log(avctx, AV_LOG_ERROR,
3078  "Not evaluating a further program_config_element as this construct is dubious at best.\n");
3079  pop_output_configuration(ac);
3080  } else {
3081  err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
3082  if (!err)
3083  ac->oc[1].m4ac.chan_config = 0;
3084  pce_found = 1;
3085  }
3086  break;
3087  }
3088 
3089  case TYPE_FIL:
3090  if (elem_id == 15)
3091  elem_id += get_bits(gb, 8) - 1;
3092  if (get_bits_left(gb) < 8 * elem_id) {
3093  av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
3094  err = AVERROR_INVALIDDATA;
3095  goto fail;
3096  }
3097  while (elem_id > 0)
3098  elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type);
3099  err = 0; /* FIXME */
3100  break;
3101 
3102  default:
3103  err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
3104  break;
3105  }
3106 
3107  if (elem_type < TYPE_DSE) {
3108  che_prev = che;
3109  che_prev_type = elem_type;
3110  }
3111 
3112  if (err)
3113  goto fail;
3114 
3115  if (get_bits_left(gb) < 3) {
3116  av_log(avctx, AV_LOG_ERROR, overread_err);
3117  err = AVERROR_INVALIDDATA;
3118  goto fail;
3119  }
3120  }
3121 
3122  if (!avctx->channels) {
3123  *got_frame_ptr = 0;
3124  return 0;
3125  }
3126 
3127  multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
3128  samples <<= multiplier;
3129 
3130  spectral_to_sample(ac, samples);
3131 
3132  if (ac->oc[1].status && audio_found) {
3133  avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
3134  avctx->frame_size = samples;
3135  ac->oc[1].status = OC_LOCKED;
3136  }
3137 
3138  if (multiplier)
3139  avctx->internal->skip_samples_multiplier = 2;
3140 
3141  if (!ac->frame->data[0] && samples) {
3142  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
3143  err = AVERROR_INVALIDDATA;
3144  goto fail;
3145  }
3146 
3147  if (samples) {
3148  ac->frame->nb_samples = samples;
3149  ac->frame->sample_rate = avctx->sample_rate;
3150  } else
3151  av_frame_unref(ac->frame);
3152  *got_frame_ptr = !!samples;
3153 
3154  /* for dual-mono audio (SCE + SCE) */
3155  is_dmono = ac->dmono_mode && sce_count == 2 &&
3156  ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
3157  if (is_dmono) {
3158  if (ac->dmono_mode == 1)
3159  ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
3160  else if (ac->dmono_mode == 2)
3161  ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
3162  }
3163 
3164  return 0;
3165 fail:
3166  pop_output_configuration(ac);
3167  return err;
3168 }
3169 
3170 static int aac_decode_frame(AVCodecContext *avctx, void *data,
3171  int *got_frame_ptr, AVPacket *avpkt)
3172 {
3173  AACContext *ac = avctx->priv_data;
3174  const uint8_t *buf = avpkt->data;
3175  int buf_size = avpkt->size;
3176  GetBitContext gb;
3177  int buf_consumed;
3178  int buf_offset;
3179  int err;
3180  int new_extradata_size;
3181  const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
3182  AV_PKT_DATA_NEW_EXTRADATA,
3183  &new_extradata_size);
3184  int jp_dualmono_size;
3185  const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
3186  AV_PKT_DATA_JP_DUALMONO,
3187  &jp_dualmono_size);
3188 
3189  if (new_extradata && 0) {
3190  av_free(avctx->extradata);
3191  avctx->extradata = av_mallocz(new_extradata_size +
3192  AV_INPUT_BUFFER_PADDING_SIZE);
3193  if (!avctx->extradata)
3194  return AVERROR(ENOMEM);
3195  avctx->extradata_size = new_extradata_size;
3196  memcpy(avctx->extradata, new_extradata, new_extradata_size);
3197  push_output_configuration(ac);
3198  if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
3199  avctx->extradata,
3200  avctx->extradata_size*8LL, 1) < 0) {
3201  pop_output_configuration(ac);
3202  return AVERROR_INVALIDDATA;
3203  }
3204  }
3205 
3206  ac->dmono_mode = 0;
3207  if (jp_dualmono && jp_dualmono_size > 0)
3208  ac->dmono_mode = 1 + *jp_dualmono;
3209  if (ac->force_dmono_mode >= 0)
3210  ac->dmono_mode = ac->force_dmono_mode;
3211 
3212  if (INT_MAX / 8 <= buf_size)
3213  return AVERROR_INVALIDDATA;
3214 
3215  if ((err = init_get_bits8(&gb, buf, buf_size)) < 0)
3216  return err;
3217 
3218  switch (ac->oc[1].m4ac.object_type) {
3219  case AOT_ER_AAC_LC:
3220  case AOT_ER_AAC_LTP:
3221  case AOT_ER_AAC_LD:
3222  case AOT_ER_AAC_ELD:
3223  err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
3224  break;
3225  default:
3226  err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
3227  }
3228  if (err < 0)
3229  return err;
3230 
3231  buf_consumed = (get_bits_count(&gb) + 7) >> 3;
3232  for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
3233  if (buf[buf_offset])
3234  break;
3235 
3236  return buf_size > buf_offset ? buf_consumed : buf_size;
3237 }
3238 
3239 static av_cold int aac_decode_close(AVCodecContext *avctx)
3240 {
3241  AACContext *ac = avctx->priv_data;
3242  int i, type;
3243 
3244  for (i = 0; i < MAX_ELEM_ID; i++) {
3245  for (type = 0; type < 4; type++) {
3246  if (ac->che[type][i])
3247  AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][i]->sbr);
3248  av_freep(&ac->che[type][i]);
3249  }
3250  }
3251 
3252  ff_mdct_end(&ac->mdct);
3253  ff_mdct_end(&ac->mdct_small);
3254  ff_mdct_end(&ac->mdct_ld);
3255  ff_mdct_end(&ac->mdct_ltp);
3256 #if !USE_FIXED
3257  ff_mdct15_uninit(&ac->mdct480);
3258 #endif
3259  av_freep(&ac->fdsp);
3260  return 0;
3261 }
3262 
3263 static void aacdec_init(AACContext *c)
3264 {
3265  c->imdct_and_windowing = imdct_and_windowing;
3266  c->apply_ltp = apply_ltp;
3267  c->apply_tns = apply_tns;
3268  c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
3269  c->update_ltp = update_ltp;
3270 #if USE_FIXED
3271  c->vector_pow43 = vector_pow43;
3272  c->subband_scale = subband_scale;
3273 #endif
3274 
3275 #if !USE_FIXED
3276  if(ARCH_MIPS)
3277  ff_aacdec_init_mips(c);
3278 #endif /* !USE_FIXED */
3279 }
3280 /**
3281  * AVOptions for Japanese DTV specific extensions (ADTS only)
3282  */
3283 #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
3284 static const AVOption options[] = {
3285  {"dual_mono_mode", "Select the channel to decode for dual mono",
3286  offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
3287  AACDEC_FLAGS, "dual_mono_mode"},
3288 
3289  {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3290  {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3291  {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3292  {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3293 
3294  {NULL},
3295 };
3296 
3297 static const AVClass aac_decoder_class = {
3298  .class_name = "AAC decoder",
3299  .item_name = av_default_item_name,
3300  .option = options,
3301  .version = LIBAVUTIL_VERSION_INT,
3302 };
IndividualChannelStream::predictor_initialized
int predictor_initialized
Definition: aac.h:187
UINTFLOAT
float UINTFLOAT
Definition: aac_defines.h:87
VMUL4S
static float * VMUL4S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
Definition: aacdec.c:124
AACContext::fdsp
AVFloatDSPContext * fdsp
Definition: aac.h:331
apply_prediction
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
Apply AAC-Main style frequency domain prediction.
Definition: aacdec_template.c:1910
AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
imdct_and_windowing
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
Conduct IMDCT and windowing.
Definition: aacdec_template.c:2570
apply_ltp
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
Apply the long term prediction.
Definition: aacdec_template.c:2500
NULL
#define NULL
Definition: coverity.c:32
s
const char * s
Definition: avisynth_c.h:768
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
TYPE_DSE
Definition: aac.h:60
elem_to_channel::elem_id
uint8_t elem_id
Definition: aacdec_template.c:195
decode_pulses
static int decode_pulses(Pulse *pulse, GetBitContext *gb, const uint16_t *swb_offset, int num_swb)
Decode pulse data; reference: table 4.7.
Definition: aacdec_template.c:1518
IndividualChannelStream::use_kb_window
uint8_t use_kb_window[2]
If set, use Kaiser-Bessel window, otherwise use a sine window.
Definition: aac.h:177
AACContext::buf_mdct
INTFLOAT buf_mdct[1024]
Definition: aac.h:316
AACContext::subband_scale
void(* subband_scale)(int *dst, int *src, int scale, int offset, int len)
Definition: aac.h:366
overread_err
#define overread_err
Definition: aacdec_template.c:101
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:187
AACADTSHeaderInfo::object_type
uint8_t object_type
Definition: aacadtsdec.h:36
AVOption
AVOption.
Definition: opt.h:246
data
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
flush
static void flush(AVCodecContext *avctx)
Definition: aacdec_template.c:500
tags_per_config
static const int8_t tags_per_config[16]
Definition: aacdectab.h:38
AACContext::avctx
AVCodecContext * avctx
Definition: aac.h:295
Pulse
Definition: aac.h:224
FFTContext::mdct_calc
void(* mdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:109
DEC_UPAIR
static int * DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
Definition: aacdec_fixed.c:125
aac_table_init
static AVOnce aac_table_init
Definition: aacdec_template.c:1136
re
float re
Definition: fft.c:82
AAC_MUL26
#define AAC_MUL26(x, y)
Definition: aac_defines.h:100
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:261
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
ff_kbd_window_init
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
Generate a Kaiser-Bessel Derived Window.
Definition: kbdwin.c:26
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
SCALE_DIFF_ZERO
#define SCALE_DIFF_ZERO
codebook index corresponding to zero scalefactor indices difference
Definition: aac.h:152
temp
else temp
Definition: vf_mcdeint.c:259
TYPE_END
Definition: aac.h:63
cce_scale
static const float cce_scale[]
Definition: aacdec_template.c:2164
skip_bits_long
static void skip_bits_long(GetBitContext *s, int n)
Definition: get_bits.h:204
g
const char * g
Definition: vf_curves.c:112
AV_OPT_TYPE_INT
Definition: opt.h:223
VMUL2S
static float * VMUL2S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
Definition: aacdec.c:107
AACDEC_FLAGS
#define AACDEC_FLAGS
AVOptions for Japanese DTV specific extensions (ADTS only)
Definition: aacdec_template.c:3283
imdct_and_windowing_eld
static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
Definition: aacdec_template.c:2668
elem_to_channel
Definition: aacdec_template.c:192
INIT_VLC_STATIC
#define INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
Definition: vlc.h:75
aacdec_init
static void aacdec_init(AACContext *ac)
Definition: aacdec_template.c:3263
FIXR10
#define FIXR10(x)
Definition: aac_defines.h:93
DEC_SQUAD
static int * DEC_SQUAD(int *dst, unsigned idx)
Definition: aacdec_fixed.c:115
decode_audio_specific_config_gb
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: aacdec_template.c:958
TYPE_SCE
Definition: aac.h:56
TYPE_CPE
Definition: aac.h:57
AACContext::che
ChannelElement * che[4][MAX_ELEM_ID]
Definition: aac.h:305
AVPacket::size
int size
Definition: avcodec.h:1658
b
const char * b
Definition: vf_curves.c:113
AVFloatDSPContext::scalarproduct_float
float(* scalarproduct_float)(const float *v1, const float *v2, int len)
Calculate the scalar product of two vectors of floats.
Definition: float_dsp.h:159
av_log2
int av_log2(unsigned v)
Definition: intmath.c:26
SingleChannelElement::ret
INTFLOAT * ret
PCM output.
Definition: aac.h:269
ChannelElement::present
int present
Definition: aac.h:276
update_ltp
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
Update the LTP buffer for next frame.
Definition: aacdec_template.c:2532
AACContext::update_ltp
void(* update_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:364
AACContext::imdct_and_windowing
void(* imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:358
vector_pow43
static void vector_pow43(int *coefs, int len)
Definition: aacdec_fixed.c:151
OutputConfiguration::channel_layout
uint64_t channel_layout
Definition: aac.h:128
SingleChannelElement::sf
INTFLOAT sf[120]
scalefactors
Definition: aac.h:255
AVFloatDSPContext::vector_fmul_reverse
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:138
AV_EF_BITSTREAM
#define AV_EF_BITSTREAM
detect bitstream specification deviations
Definition: avcodec.h:3029
decode_ga_specific_config
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.
Definition: aacdec_template.c:804
ChannelElement::ms_mask
uint8_t ms_mask[128]
Set if mid/side stereo is used for each scalefactor window band.
Definition: aac.h:281
apply_independent_coupling
static void apply_independent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply independent channel coupling (applied after IMDCT).
Definition: aacdec.c:246
av_mallocz
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:222
subband_scale
static void subband_scale(int *dst, int *src, int scale, int offset, int len)
Definition: aacdec_fixed.c:165
MAX_LTP_LONG_SFB
#define MAX_LTP_LONG_SFB
Definition: aac.h:51
GET_GAIN
#define GET_GAIN(x, y)
Definition: aac_defines.h:98
DynamicRangeControl
Dynamic Range Control - decoded from the bitstream but not processed further.
Definition: aac.h:211
che_configure
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.
Definition: aacdec_template.c:127
vlc_scalefactors
static VLC vlc_scalefactors
Definition: aacdec_template.c:94
NOISE_PRE
#define NOISE_PRE
preamble for NOISE_BT, put in bitstream with the first noise band
Definition: aac.h:156
FF_PROFILE_AAC_HE_V2
#define FF_PROFILE_AAC_HE_V2
Definition: avcodec.h:3244
predict
static av_always_inline void predict(PredictorState *ps, float *coef, int output_enable)
Definition: aacdec.c:174
ChannelCoupling::type
enum RawDataBlockType type[8]
Type of channel element to be coupled - SCE or CPE.
Definition: aac.h:237
AVCodecContext::profile
int profile
profile
Definition: avcodec.h:3235
output_configure
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.
Definition: aacdec_template.c:440
ChannelPosition
ChannelPosition
Definition: aac.h:94
decode_channel_map
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...
Definition: aacdec_template.c:692
aac_decode_er_frame
static int aac_decode_er_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb)
Definition: aacdec_template.c:2905
NOISE_BT
Spectral data are scaled white noise not coded in the bitstream.
Definition: aac.h:87
TYPE_CCE
Definition: aac.h:58
decode_cce
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
Decode coupling_channel_element; reference: table 4.8.
Definition: aacdec_template.c:2176
USE_FIXED
#define USE_FIXED
Definition: aac_defines.h:25
lcg_random
static av_always_inline int lcg_random(unsigned previous_val)
linear congruential pseudorandom number generator
Definition: aacdec_template.c:1052
DynamicRangeControl::band_incr
int band_incr
Number of DRC bands greater than 1 having DRC info.
Definition: aac.h:216
ff_aac_num_swb_128
const uint8_t ff_aac_num_swb_128[]
Definition: aactab.c:55
AAC_RENAME_32
#define AAC_RENAME_32(x)
Definition: aac_defines.h:85
ff_cbrt_tableinit
void ff_cbrt_tableinit(void)
Definition: cbrt_tablegen.h:40
AACContext::dmono_mode
int dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
Definition: aac.h:349
IndividualChannelStream::swb_offset
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
AOT_ER_AAC_LTP
N Error Resilient Long Term Prediction.
Definition: mpeg4audio.h:86
INTFLOAT
float INTFLOAT
Definition: aac_defines.h:86
AVClass::class_name
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
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
AV_OPT_TYPE_CONST
Definition: opt.h:232
avpriv_request_sample
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
decode_ics_info
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb)
Decode Individual Channel Stream info; reference: table 4.6.
Definition: aacdec_template.c:1278
AVFloatDSPContext::vector_fmul_window
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:103
EXT_FILL
Definition: aac.h:67
BandType
BandType
Definition: aac.h:82
bits
uint8_t bits
Definition: crc.c:296
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2502
uint8_t
uint8_t
Definition: audio_convert.c:194
FIXR
#define FIXR(x)
Definition: aac_defines.h:92
av_cold
#define av_cold
Definition: attributes.h:82
OutputConfiguration::layout_map
uint8_t layout_map[MAX_ELEM_ID *4][3]
Definition: aac.h:125
OC_TRIAL_PCE
Output configuration under trial specified by an inband PCE.
Definition: aac.h:117
ff_swb_offset_480
const uint16_t *const ff_swb_offset_480[]
Definition: aactab.c:1248
FF_DEBUG_PICT_INFO
#define FF_DEBUG_PICT_INFO
Definition: avcodec.h:2974
ChannelElement::ch
SingleChannelElement ch[2]
Definition: aac.h:284
ff_swb_offset_512
const uint16_t *const ff_swb_offset_512[]
Definition: aactab.c:1240
TYPE_LFE
Definition: aac.h:59
ff_tns_max_bands_480
const uint8_t ff_tns_max_bands_480[]
Definition: aactab.c:1282
end
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
SingleChannelElement::tns
TemporalNoiseShaping tns
Definition: aac.h:250
AOT_ER_AAC_LD
N Error Resilient Low Delay.
Definition: mpeg4audio.h:90
decode_extension_payload
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.
Definition: aacdec_template.c:2359
ff_aac_scalefactor_bits
const uint8_t ff_aac_scalefactor_bits[121]
Definition: aactab.c:82
CouplingPoint
CouplingPoint
The point during decoding at which channel coupling is applied.
Definition: aac.h:106
AVCodecContext::extradata
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1847
ChannelCoupling::num_coupled
int num_coupled
number of target elements
Definition: aac.h:236
AV_CH_LOW_FREQUENCY
#define AV_CH_LOW_FREQUENCY
Definition: channel_layout.h:52
ff_mdct15_init
av_cold int ff_mdct15_init(MDCT15Context **ps, int inverse, int N, double scale)
Init an (i)MDCT of the length 2 * 15 * (2^N)
Definition: mdct15.c:101
DynamicRangeControl::exclude_mask
int exclude_mask[MAX_CHANNELS]
Channels to be excluded from DRC processing.
Definition: aac.h:215
TemporalNoiseShaping::n_filt
int n_filt[8]
Definition: aac.h:200
AACContext::mdct_ltp
FFTContext mdct_ltp
Definition: aac.h:326
AACContext::vector_pow43
void(* vector_pow43)(int *coefs, int len)
Definition: aac.h:365
AACContext::output_element
SingleChannelElement * output_element[MAX_CHANNELS]
Points to each SingleChannelElement.
Definition: aac.h:340
aac_decode_init
static av_cold int aac_decode_init(AVCodecContext *avctx)
Definition: aacdec_template.c:1138
AVPacket::data
uint8_t * data
Definition: avcodec.h:1657
get_bits_count
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:199
AAC_MUL31
#define AAC_MUL31(x, y)
Definition: aac_defines.h:102
count_channels
static int count_channels(uint8_t(*layout)[3], int tags)
Definition: aacdec_template.c:103
ff_dlog
#define ff_dlog(a,...)
Definition: tableprint_vlc.h:29
INTENSITY_BT
Scalefactor data are intensity stereo positions (in phase).
Definition: aac.h:89
AV_LOG_VERBOSE
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
sample_rate_idx
static int sample_rate_idx(int rate)
Definition: aacdec_template.c:1065
decode_tns
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, GetBitContext *gb, const IndividualChannelStream *ics)
Decode Temporal Noise Shaping data; reference: table 4.48.
Definition: aacdec_template.c:1545
AV_CH_BACK_LEFT
#define AV_CH_BACK_LEFT
Definition: channel_layout.h:53
ChannelCoupling::id_select
int id_select[8]
element id
Definition: aac.h:238
size
ptrdiff_t size
Definition: opengl_enc.c:101
ff_aac_codebook_vector_vals
const float *const ff_aac_codebook_vector_vals[]
Definition: aactab.c:1064
avpriv_alloc_fixed_dsp
AVFixedDSPContext * avpriv_alloc_fixed_dsp(int bit_exact)
Allocate and initialize a fixed DSP context.
Definition: fixed_dsp.c:148
fixed_sqrt
static av_always_inline int fixed_sqrt(int x, int bits)
Calculate the square root.
Definition: fixed_dsp.h:176
AOT_ER_AAC_LC
N Error Resilient Low Complexity.
Definition: mpeg4audio.h:85
AACContext::tag_che_map
ChannelElement * tag_che_map[4][MAX_ELEM_ID]
Definition: aac.h:306
AVOnce
#define AVOnce
Definition: thread.h:154
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
OC_GLOBAL_HDR
Output configuration set in a global header but not yet locked.
Definition: aac.h:119
spectral_to_sample
static void spectral_to_sample(AACContext *ac, int samples)
Convert spectral data to samples, applying all supported tools as appropriate.
Definition: aacdec_template.c:2774
AACContext::random_state
int random_state
Definition: aac.h:333
AACContext::mdct480
MDCT15Context * mdct480
Definition: aac.h:330
U
#define U(x)
Definition: vp56_arith.h:37
parse_adts_frame_header
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
Definition: aacdec_template.c:2843
get_bits_left
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:587
OutputConfiguration::m4ac
MPEG4AudioConfig m4ac
Definition: aac.h:124
DynamicRangeControl::dyn_rng_sgn
int dyn_rng_sgn[17]
DRC sign information; 0 - positive, 1 - negative.
Definition: aac.h:213
ff_sbr_apply
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.
Definition: aacsbr_template.c:1467
ff_cbrt_tab
uint32_t ff_cbrt_tab[1<< 13]
pop_output_configuration
static void pop_output_configuration(AACContext *ac)
Restore the previous output configuration if and only if the current configuration is unlocked...
Definition: aacdec_template.c:424
decode_fill
static int decode_fill(AACContext *ac, GetBitContext *gb, int len)
Definition: aacdec_template.c:2326
UPDATE_CACHE
#define UPDATE_CACHE(name, gb)
Definition: get_bits.h:160
SingleChannelElement::predictor_state
PredictorState predictor_state[MAX_PREDICTORS]
Definition: aac.h:268
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
relative_align_get_bits
static void relative_align_get_bits(GetBitContext *gb, int reference_position)
Definition: aacdec_template.c:722
ChannelElement::sbr
SpectralBandReplication sbr
Definition: aac.h:287
AACContext::mdct_small
FFTContext mdct_small
Definition: aac.h:324
AVFloatDSPContext::vector_fmul
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
av_default_item_name
av_default_item_name
Definition: libopenh264enc.c:78
ChannelCoupling::coupling_point
enum CouplingPoint coupling_point
The point during decoding at which coupling is applied.
Definition: aac.h:235
AVERROR
#define AVERROR(e)
Definition: error.h:43
elem_to_channel::av_position
uint64_t av_position
Definition: aacdec_template.c:193
av_packet_get_side_data
uint8_t * av_packet_get_side_data(const AVPacket *pkt, enum AVPacketSideDataType type, int *size)
Get side information from packet.
Definition: avpacket.c:350
ff_aac_num_swb_1024
const uint8_t ff_aac_num_swb_1024[]
Definition: aactab.c:43
ff_mpeg4audio_get_config_gb
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
AVFloatDSPContext::butterflies_float
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:148
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
ff_aac_kbd_long_1024
float ff_aac_kbd_long_1024[1024]
Definition: aactab.c:38
AACContext::temp
INTFLOAT temp[128]
Definition: aac.h:352
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1827
decode_eld_specific_config
static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx, GetBitContext *gb, MPEG4AudioConfig *m4ac, int channel_config)
Definition: aacdec_template.c:883
AACADTSHeaderInfo::sampling_index
uint8_t sampling_index
Definition: aacadtsdec.h:37
Pulse::amp
int amp[4]
Definition: aac.h:228
assign_pair
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)
Definition: aacdec_template.c:199
IndividualChannelStream::max_sfb
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:175
ff_mdct_init
#define ff_mdct_init
Definition: fft.h:169
ff_aac_eld_window_512
const float ff_aac_eld_window_512[1920]
Definition: aactab.c:1291
TYPE_FIL
Definition: aac.h:62
offset
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
count
GLsizei count
Definition: opengl_enc.c:109
CLOSE_READER
#define CLOSE_READER(name, gb)
Definition: get_bits.h:131
IndividualChannelStream::num_swb
int num_swb
number of scalefactor window bands
Definition: aac.h:183
count_paired_channels
static int count_paired_channels(uint8_t(*layout_map)[3], int tags, int pos, int *current)
Definition: aacdec_template.c:228
FFMAX
#define FFMAX(a, b)
Definition: common.h:94
fail
#define fail()
Definition: checkasm.h:89
DynamicRangeControl::prog_ref_level
int prog_ref_level
A reference level for the long-term program audio level for all channels combined.
Definition: aac.h:219
OC_LOCKED
Output configuration locked in place.
Definition: aac.h:120
PredictorState
Predictor State.
Definition: aac.h:135
AACADTSHeaderInfo::chan_config
uint8_t chan_config
Definition: aacadtsdec.h:38
VLC
Definition: vlc.h:26
ff_aac_pow2sf_tab
float ff_aac_pow2sf_tab[428]
Definition: aactab.c:35
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2545
SKIP_BITS
#define SKIP_BITS(name, gb, num)
Definition: get_bits.h:175
AAC_RENAME
#define AAC_RENAME(x)
Definition: aac_defines.h:84
AACContext::warned_remapping_once
int warned_remapping_once
Definition: aac.h:308
SingleChannelElement::ret_buf
INTFLOAT ret_buf[2048]
PCM output buffer.
Definition: aac.h:264
AOT_ER_AAC_SCALABLE
N Error Resilient Scalable.
Definition: mpeg4audio.h:87
window
static SDL_Window * window
Definition: ffplay.c:362
reset_predictor_group
static void reset_predictor_group(PredictorState *ps, int group_num)
Definition: aacdec_template.c:1081
MDCT15Context::imdct_half
void(* imdct_half)(struct MDCT15Context *s, float *dst, const float *src, ptrdiff_t src_stride, float scale)
Calculate the middle half of the iMDCT.
Definition: mdct15.h:53
get_che
static ChannelElement * get_che(AACContext *ac, int type, int elem_id)
Definition: aacdec_template.c:560
IndividualChannelStream::window_sequence
enum WindowSequence window_sequence[2]
Definition: aac.h:176
SingleChannelElement::ltp_state
INTFLOAT ltp_state[3072]
time signal for LTP
Definition: aac.h:265
AV_CODEC_FLAG_BITEXACT
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:921
ff_aac_num_swb_512
const uint8_t ff_aac_num_swb_512[]
Definition: aactab.c:47
AVCodecContext::err_recognition
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
Definition: avcodec.h:3020
aac_decode_frame
static int aac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: aacdec_template.c:3170
IndividualChannelStream::predictor_reset_group
int predictor_reset_group
Definition: aac.h:188
frame_configure_elements
static int frame_configure_elements(AVCodecContext *avctx)
Definition: aacdec_template.c:158
FFMIN
#define FFMIN(a, b)
Definition: common.h:96
DynamicRangeControl::dyn_rng_ctl
int dyn_rng_ctl[17]
DRC magnitude information.
Definition: aac.h:214
AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
ltp_coef
static const INTFLOAT ltp_coef[8]
Definition: aactab.h:94
decode_mid_side_stereo
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, int ms_present)
Decode Mid/Side data; reference: table 4.54.
Definition: aacdec_template.c:1588
void
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
apply_intensity_stereo
static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
intensity stereo decoding; reference: 4.6.8.2.3
Definition: aacdec_template.c:2073
AACADTSHeaderInfo::num_aac_frames
uint8_t num_aac_frames
Definition: aacadtsdec.h:39
Pulse::pos
int pos[4]
Definition: aac.h:227
AOT_AAC_MAIN
Y Main.
Definition: mpeg4audio.h:71
int32_t
int32_t
Definition: audio_convert.c:194
show_bits
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
Definition: get_bits.h:296
AACContext::mdct_ld
FFTContext mdct_ld
Definition: aac.h:325
ff_aacdec_init_mips
void ff_aacdec_init_mips(AACContext *c)
Definition: aacdec_mips.c:433
ff_decode_sbr_extension
int AAC_RENAME() ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr, GetBitContext *gb, int crc, int cnt, int id_aac)
Decode one SBR element.
Definition: aacsbr_template.c:1103
LAST_SKIP_BITS
#define LAST_SKIP_BITS(name, gb, num)
Definition: get_bits.h:181
get_vlc2
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:554
TemporalNoiseShaping::length
int length[8][4]
Definition: aac.h:201
AVFloatDSPContext::vector_fmul_scalar
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:69
aac_static_table_init
static av_cold void aac_static_table_init(void)
Definition: aacdec_template.c:1098
ff_aac_sbr_ctx_close
void AAC_RENAME() ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
Close one SBR context.
Definition: aacsbr_template.c:103
apply_channel_coupling
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
Definition: aacdec_template.c:2741
AV_CH_FRONT_LEFT_OF_CENTER
#define AV_CH_FRONT_LEFT_OF_CENTER
Definition: channel_layout.h:55
AV_EF_EXPLODE
#define AV_EF_EXPLODE
abort decoding on minor error detection
Definition: avcodec.h:3031
n
int n
Definition: avisynth_c.h:684
ff_tns_max_bands_1024
const uint8_t ff_tns_max_bands_1024[]
Definition: aactab.c:1274
GET_VLC
#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:477
compute_lpc_coefs
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
AV_CH_FRONT_CENTER
#define AV_CH_FRONT_CENTER
Definition: channel_layout.h:51
decode_ltp
static void decode_ltp(LongTermPrediction *ltp, GetBitContext *gb, uint8_t max_sfb)
Decode Long Term Prediction data; reference: table 4.xx.
Definition: aacdec_template.c:1264
aac_decode_frame_int
static int aac_decode_frame_int(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
Definition: aacdec_template.c:2977
apply_dependent_coupling
static void apply_dependent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply dependent channel coupling (applied before IMDCT).
Definition: aacdec.c:210
decode_spectrum_and_dequant
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.
Definition: aacdec_template.c:1613
ff_aac_sbr_init
void AAC_RENAME() ff_aac_sbr_init(void)
Initialize SBR.
Definition: aacsbr_template.c:37
DynamicRangeControl::pce_instance_tag
int pce_instance_tag
Indicates with which program the DRC info is associated.
Definition: aac.h:212
windowing_and_mdct_ltp
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...
Definition: aacdec_template.c:2474
AOT_AAC_SCALABLE
N Scalable.
Definition: mpeg4audio.h:76
tns_tmp2_map
static const INTFLOAT *const tns_tmp2_map[4]
Definition: aactab.h:126
elem_to_channel::aac_position
uint8_t aac_position
Definition: aacdec_template.c:196
SHOW_UBITS
#define SHOW_UBITS(name, gb, num)
Definition: get_bits.h:193
push_output_configuration
static int push_output_configuration(AACContext *ac)
Save current output configuration if and only if it has been locked.
Definition: aacdec_template.c:409
FF_ARRAY_ELEMS
#define FF_ARRAY_ELEMS(a)
Definition: sinewin_tablegen_template.c:36
AV_CH_FRONT_RIGHT_OF_CENTER
#define AV_CH_FRONT_RIGHT_OF_CENTER
Definition: channel_layout.h:56
DynamicRangeControl::interpolation_scheme
int interpolation_scheme
Indicates the interpolation scheme used in the SBR QMF domain.
Definition: aac.h:217
ChannelCoupling
coupling parameters
Definition: aac.h:234
AACContext::tags_mapped
int tags_mapped
Definition: aac.h:307
reset_all_predictors
static void reset_all_predictors(PredictorState *ps)
Definition: aacdec_template.c:1058
skip_data_stream_element
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
Skip data_stream_element; reference: table 4.10.
Definition: aacdec_template.c:1225
AVERROR_PATCHWELCOME
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
ChannelCoupling::ch_select
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
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2514
AACContext::force_dmono_mode
int force_dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
Definition: aac.h:348
AV_PKT_DATA_NEW_EXTRADATA
The AV_PKT_DATA_NEW_EXTRADATA is used to notify the codec or the format that the extradata buffer was...
Definition: avcodec.h:1412
TemporalNoiseShaping::order
int order[8][4]
Definition: aac.h:203
AV_LOG_INFO
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
AV_ONCE_INIT
#define AV_ONCE_INIT
Definition: thread.h:155
AACContext::warned_num_aac_frames
int warned_num_aac_frames
Definition: aac.h:355
AAC_INIT_VLC_STATIC
#define AAC_INIT_VLC_STATIC(num, size)
Definition: aacdec_template.c:1088
TemporalNoiseShaping
Temporal Noise Shaping.
Definition: aac.h:198
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:2494
ff_aac_kbd_short_128
float ff_aac_kbd_short_128[128]
Definition: aactab.c:39
options
static const AVOption options[]
Definition: aacdec_template.c:3284
init_get_bits8
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:456
AV_CH_LAYOUT_NATIVE
#define AV_CH_LAYOUT_NATIVE
Channel mask value used for AVCodecContext.request_channel_layout to indicate that the user requests ...
Definition: channel_layout.h:78
decode_scalefactors
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.
Definition: aacdec_template.c:1443
AVCodecContext::debug
int debug
debug
Definition: avcodec.h:2973
decode_cpe
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
Decode a channel_pair_element; reference: table 4.4.
Definition: aacdec_template.c:2123
LongTermPrediction
Long Term Prediction.
Definition: aac.h:163
apply_tns
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...
Definition: aacdec_template.c:2417
AVCodecContext
main external API structure.
Definition: avcodec.h:1732
AV_CH_FRONT_LEFT
#define AV_CH_FRONT_LEFT
Definition: channel_layout.h:49
AVCodecInternal::skip_samples_multiplier
int skip_samples_multiplier
Definition: internal.h:178
NOISE_PRE_BITS
#define NOISE_PRE_BITS
length of preamble
Definition: aac.h:157
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:953
OPEN_READER
#define OPEN_READER(name, gb)
Definition: get_bits.h:120
SingleChannelElement::ics
IndividualChannelStream ics
Definition: aac.h:249
in
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
Definition: audio_convert.c:194
avpriv_aac_parse_header
int avpriv_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
Parse AAC frame header.
Definition: aacadtsdec.c:29
buf
void * buf
Definition: avisynth_c.h:690
MAX_PREDICTORS
#define MAX_PREDICTORS
Definition: aac.h:146
cbrtf
static av_always_inline float cbrtf(float x)
Definition: libm.h:61
FFTContext::imdct_half
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:108
type
GLint GLenum type
Definition: opengl_enc.c:105
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:1848
ff_aac_sbr_ctx_init
void AAC_RENAME() ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr, int id_aac)
Initialize one SBR context.
Definition: aacsbr_template.c:84
IndividualChannelStream::group_len
uint8_t group_len[8]
Definition: aac.h:179
AVERROR_BUG
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
get_bits1
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:313
skip_bits1
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:338
AACContext::apply_tns
void(* apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns, IndividualChannelStream *ics, int decode)
Definition: aac.h:360
MAX_ELEM_ID
#define MAX_ELEM_ID
Definition: aac.h:48
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
AVFrame::sample_rate
int sample_rate
Sample rate of the audio data.
Definition: frame.h:351
aac_decode_close
static av_cold int aac_decode_close(AVCodecContext *avctx)
Definition: aacdec_template.c:3239
skip_bits
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:306
decode_audio_specific_config
static int decode_audio_specific_config(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, const uint8_t *data, int64_t bit_size, int sync_extension)
Definition: aacdec_template.c:1019
AAC_MUL30
#define AAC_MUL30(x, y)
Definition: aac_defines.h:101
sniff_channel_order
static uint64_t sniff_channel_order(uint8_t(*layout_map)[3], int tags)
Definition: aacdec_template.c:260
decode_drc_channel_exclusions
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...
Definition: aacdec_template.c:2260
index
int index
Definition: gxfenc.c:89
noise_scale
static void noise_scale(int *coefs, int scale, int band_energy, int len)
Definition: aacdec_fixed.c:196
decode_pce
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.
Definition: aacdec_template.c:734
init_get_bits
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:425
GET_CACHE
#define GET_CACHE(name, gb)
Definition: get_bits.h:197
elem_to_channel::syn_ele
uint8_t syn_ele
Definition: aacdec_template.c:194
VMUL2
static float * VMUL2(float *dst, const float *v, unsigned idx, const float *scale)
Definition: aacdec.c:83
OCStatus
OCStatus
Output configuration status.
Definition: aac.h:115
AVCodecInternal::skip_samples
int skip_samples
Number of audio samples to skip at the start of the next decoded frame.
Definition: internal.h:157
MAX_CHANNELS
#define MAX_CHANNELS
Definition: aac.h:47
AOT_ER_BSAC
N Error Resilient Bit-Sliced Arithmetic Coding.
Definition: mpeg4audio.h:89
TNS_MAX_ORDER
#define TNS_MAX_ORDER
Definition: aac.h:50
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:119
FF_COMPLIANCE_STRICT
#define FF_COMPLIANCE_STRICT
Strictly conform to all the things in the spec no matter what consequences.
Definition: avcodec.h:2953
AACContext
main AAC context
Definition: aac.h:293
u
#define u(width,...)
Definition: vaapi_encode_h26x.h:36
ff_aac_scalefactor_code
const uint32_t ff_aac_scalefactor_code[121]
Definition: aactab.c:63
IndividualChannelStream::ltp
LongTermPrediction ltp
Definition: aac.h:180
ChannelElement::coup
ChannelCoupling coup
Definition: aac.h:286
OC_TRIAL_FRAME
Output configuration under trial specified by a frame header.
Definition: aac.h:118
MPEG4AudioConfig::frame_length_short
int frame_length_short
Definition: mpeg4audio.h:41
decode_prediction
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb)
Definition: aacdec_template.c:1242
ff_tns_max_bands_128
const uint8_t ff_tns_max_bands_128[]
Definition: aactab.c:1286
NOISE_OFFSET
#define NOISE_OFFSET
subtracted from global gain, used as offset for the preamble
Definition: aac.h:158
imdct_and_window
static void imdct_and_window(TwinVQContext *tctx, enum TwinVQFrameType ftype, int wtype, float *in, float *prev, int ch)
Definition: twinvq.c:327
av_frame_unref
void av_frame_unref(AVFrame *frame)
Unreference all the buffers referenced by frame and reset the frame fields.
Definition: frame.c:498
avpriv_report_missing_feature
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
filt
static const int8_t filt[NUMTAPS]
Definition: af_earwax.c:39
SingleChannelElement::band_type_run_end
int band_type_run_end[120]
band type run end points
Definition: aac.h:254
decode_band_types
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.
Definition: aacdec_template.c:1394
AV_CH_BACK_CENTER
#define AV_CH_BACK_CENTER
Definition: channel_layout.h:57
AVFrame::data
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:201
DynamicRangeControl::band_top
int band_top[17]
Indicates the top of the i-th DRC band in units of 4 spectral lines.
Definition: aac.h:218
AV_CH_SIDE_RIGHT
#define AV_CH_SIDE_RIGHT
Definition: channel_layout.h:59
SingleChannelElement::coeffs
INTFLOAT coeffs[1024]
coefficients for IMDCT, maybe processed
Definition: aac.h:262
vlc_spectral
static VLC vlc_spectral[11]
Definition: aacdec_template.c:95
OutputConfiguration::status
enum OCStatus status
Definition: aac.h:129
ChannelCoupling::gain
INTFLOAT gain[16][120]
Definition: aac.h:242
INTENSITY_BT2
Scalefactor data are intensity stereo positions (out of phase).
Definition: aac.h:88
AOT_ER_AAC_ELD
N Error Resilient Enhanced Low Delay.
Definition: mpeg4audio.h:106
set_default_channel_config
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...
Definition: aacdec_template.c:523
M_SQRT2
#define M_SQRT2
Definition: mathematics.h:61
RANGE15
#define RANGE15(x)
Definition: aac_defines.h:97
TemporalNoiseShaping::coef
INTFLOAT coef[8][4][TNS_MAX_ORDER]
Definition: aac.h:205
LongTermPrediction::lag
int16_t lag
Definition: aac.h:165
AACContext::che_drc
DynamicRangeControl che_drc
Definition: aac.h:299
reset_predict_state
static av_always_inline void reset_predict_state(PredictorState *ps)
Definition: aacdec.c:72
AACContext::frame
AVFrame * frame
Definition: aac.h:296
AACContext::oc
OutputConfiguration oc[2]
Definition: aac.h:354
AV_PKT_DATA_JP_DUALMONO
An AV_PKT_DATA_JP_DUALMONO side data packet indicates that the packet may contain "dual mono" audio s...
Definition: avcodec.h:1523
int
int
Definition: ffmpeg_filter.c:190
ff_aac_pred_sfb_max
const uint8_t ff_aac_pred_sfb_max[]
Definition: aactab.c:59
TemporalNoiseShaping::direction
int direction[8][4]
Definition: aac.h:202
AACContext::apply_ltp
void(* apply_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:359
IndividualChannelStream::prediction_used
uint8_t prediction_used[41]
Definition: aac.h:190
ff_aac_eld_window_480
const float ff_aac_eld_window_480[1800]
Definition: aactab.c:2258
SingleChannelElement::saved
INTFLOAT saved[1536]
overlap
Definition: aac.h:263
SingleChannelElement
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:248
ff_mdct_end
#define ff_mdct_end
Definition: fft.h:170
ff_aac_num_swb_480
const uint8_t ff_aac_num_swb_480[]
Definition: aactab.c:51
c
static double c[64]
Definition: vsrc_mptestsrc.c:87
ff_swb_offset_1024
const uint16_t *const ff_swb_offset_1024[]
Definition: aactab.c:1232
AAC_SIGNE
unsigned AAC_SIGNE
Definition: aac_defines.h:91
AACContext::windowing_and_mdct_ltp
void(* windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out, INTFLOAT *in, IndividualChannelStream *ics)
Definition: aac.h:362
TYPE_PCE
Definition: aac.h:61
IndividualChannelStream
Individual Channel Stream.
Definition: aac.h:174
LongTermPrediction::coef
INTFLOAT coef
Definition: aac.h:167
ch
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_YASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
AVERROR_UNKNOWN
#define AVERROR_UNKNOWN
Unknown error, typically from an external library.
Definition: error.h:71
ff_aac_codebook_vector_idx
const uint16_t *const ff_aac_codebook_vector_idx[]
Definition: aactab.c:1073
ff_aac_tableinit
static void ff_aac_tableinit(void)
Definition: aactab.h:45
AV_INPUT_BUFFER_PADDING_SIZE
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:769
ff_mdct15_uninit
av_cold void ff_mdct15_uninit(MDCT15Context **ps)
Frees a context.
Definition: mdct15.c:48
ChannelElement
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:275
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:1774
decode_ics
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.
Definition: aacdec_template.c:1946
av_free
#define av_free(p)
Definition: tableprint_vlc.h:34
FF_DEBUG_STARTCODE
#define FF_DEBUG_STARTCODE
Definition: avcodec.h:2987
ff_tns_max_bands_512
const uint8_t ff_tns_max_bands_512[]
Definition: aactab.c:1278
len
int len
Definition: vorbis_enc_data.h:452
ZERO_BT
Scalefactors and spectral data are all zero.
Definition: aac.h:83
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:2495
Pulse::num_pulse
int num_pulse
Definition: aac.h:225
DEC_SPAIR
static int * DEC_SPAIR(int *dst, unsigned idx)
Definition: aacdec_fixed.c:107
AVCodecContext::internal
struct AVCodecInternal * internal
Private context used for internal data.
Definition: avcodec.h:1782
ff_mpeg4audio_channels
const uint8_t ff_mpeg4audio_channels[8]
Definition: mpeg4audio.c:67
ff_thread_once
static int ff_thread_once(char *control, void(*routine)(void))
Definition: thread.h:157
VLC::table
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
AOT_AAC_LTP
Y Long Term Prediction.
Definition: mpeg4audio.h:74
AACADTSHeaderInfo::crc_absent
uint8_t crc_absent
Definition: aacadtsdec.h:35
align_get_bits
static const uint8_t * align_get_bits(GetBitContext *s)
Definition: get_bits.h:464
layout
uint64_t layout
Definition: channel_layout.c:77
FF_PROFILE_AAC_HE
#define FF_PROFILE_AAC_HE
Definition: avcodec.h:3243
SingleChannelElement::band_type
enum BandType band_type[128]
band types
Definition: aac.h:252
decode_dynamic_range
static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext *gb)
Decode dynamic range information; reference: table 4.52.
Definition: aacdec_template.c:2279
AV_CH_FRONT_RIGHT
#define AV_CH_FRONT_RIGHT
Definition: channel_layout.h:50
POW_SF2_ZERO
#define POW_SF2_ZERO
ff_aac_pow2sf_tab index corresponding to pow(2, 0);
Definition: aac.h:154
imdct_and_windowing_ld
static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
Definition: aacdec_template.c:2635
out
FILE * out
Definition: movenc.c:54
AACContext::mdct
FFTContext mdct
Definition: aac.h:323
MPEG4AudioConfig::sbr
int sbr
-1 implicit, 1 presence
Definition: mpeg4audio.h:34
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
start
void INT64 start
Definition: avisynth_c.h:690
decode
static int decode(AVCodecContext *avctx, AVFrame *frame, int *got_frame, AVPacket *pkt)
Definition: ffmpeg.c:2257
av_always_inline
#define av_always_inline
Definition: attributes.h:39
apply_mid_side_stereo
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
Mid/Side stereo decoding; reference: 4.6.8.1.3.
Definition: aacdec_template.c:2035
VLC_TYPE
#define VLC_TYPE
Definition: vlc.h:24
AV_CH_SIDE_LEFT
#define AV_CH_SIDE_LEFT
Definition: channel_layout.h:58
FFSWAP
#define FFSWAP(type, a, b)
Definition: common.h:99
MPEG4AudioConfig::ps
int ps
-1 implicit, 1 presence
Definition: mpeg4audio.h:40
LongTermPrediction::used
int8_t used[MAX_LTP_LONG_SFB]
Definition: aac.h:168
ff_swb_offset_128
const uint16_t *const ff_swb_offset_128[]
Definition: aactab.c:1256
LongTermPrediction::present
int8_t present
Definition: aac.h:164
AACADTSHeaderInfo::sample_rate
uint32_t sample_rate
Definition: aacadtsdec.h:32
aac_decoder_class
static const AVClass aac_decoder_class
Definition: aacdec_template.c:3297
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:234
AVCodecContext::request_channel_layout
uint64_t request_channel_layout
Request decoder to use this channel layout if it can (0 for default)
Definition: avcodec.h:2552
OutputConfiguration::layout_map_tags
int layout_map_tags
Definition: aac.h:126
id
enum AVCodecID id
Definition: extract_extradata_bsf.c:244
AVPacket
This structure stores compressed data.
Definition: avcodec.h:1634
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:244
AVCodecContext::strict_std_compliance
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:2951
ff_init_ff_sine_windows
void AAC_RENAME() ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
Definition: sinewin_tablegen.h:72
DEC_UQUAD
static int * DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
Definition: aacdec_fixed.c:133
AV_CH_BACK_RIGHT
#define AV_CH_BACK_RIGHT
Definition: channel_layout.h:54
AOT_AAC_LC
Y Low Complexity.
Definition: mpeg4audio.h:72
VMUL4
static float * VMUL4(float *dst, const float *v, unsigned idx, const float *scale)
Definition: aacdec.c:94
OC_NONE
Output unconfigured.
Definition: aac.h:116
aac_channel_layout_map
static const uint8_t aac_channel_layout_map[16][5][3]
Definition: aacdectab.h:40
RawDataBlockType
RawDataBlockType
Definition: aac.h:55
tmp
static uint8_t tmp[11]
Definition: aes_ctr.c:26
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