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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 static VLC vlc_scalefactors;
93 static VLC vlc_spectral[11];
94 
95 static int output_configure(AACContext *ac,
96  uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
97  enum OCStatus oc_type, int get_new_frame);
98 
99 #define overread_err "Input buffer exhausted before END element found\n"
100 
101 static int count_channels(uint8_t (*layout)[3], int tags)
102 {
103  int i, sum = 0;
104  for (i = 0; i < tags; i++) {
105  int syn_ele = layout[i][0];
106  int pos = layout[i][2];
107  sum += (1 + (syn_ele == TYPE_CPE)) *
108  (pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);
109  }
110  return sum;
111 }
112 
113 /**
114  * Check for the channel element in the current channel position configuration.
115  * If it exists, make sure the appropriate element is allocated and map the
116  * channel order to match the internal FFmpeg channel layout.
117  *
118  * @param che_pos current channel position configuration
119  * @param type channel element type
120  * @param id channel element id
121  * @param channels count of the number of channels in the configuration
122  *
123  * @return Returns error status. 0 - OK, !0 - error
124  */
125 static av_cold int che_configure(AACContext *ac,
126  enum ChannelPosition che_pos,
127  int type, int id, int *channels)
128 {
129  if (*channels >= MAX_CHANNELS)
130  return AVERROR_INVALIDDATA;
131  if (che_pos) {
132  if (!ac->che[type][id]) {
133  if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
134  return AVERROR(ENOMEM);
135  AAC_RENAME(ff_aac_sbr_ctx_init)(ac, &ac->che[type][id]->sbr);
136  }
137  if (type != TYPE_CCE) {
138  if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
139  av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
140  return AVERROR_INVALIDDATA;
141  }
142  ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
143  if (type == TYPE_CPE ||
144  (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
145  ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
146  }
147  }
148  } else {
149  if (ac->che[type][id])
150  AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][id]->sbr);
151  av_freep(&ac->che[type][id]);
152  }
153  return 0;
154 }
155 
156 static int frame_configure_elements(AVCodecContext *avctx)
157 {
158  AACContext *ac = avctx->priv_data;
159  int type, id, ch, ret;
160 
161  /* set channel pointers to internal buffers by default */
162  for (type = 0; type < 4; type++) {
163  for (id = 0; id < MAX_ELEM_ID; id++) {
164  ChannelElement *che = ac->che[type][id];
165  if (che) {
166  che->ch[0].ret = che->ch[0].ret_buf;
167  che->ch[1].ret = che->ch[1].ret_buf;
168  }
169  }
170  }
171 
172  /* get output buffer */
173  av_frame_unref(ac->frame);
174  if (!avctx->channels)
175  return 1;
176 
177  ac->frame->nb_samples = 2048;
178  if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
179  return ret;
180 
181  /* map output channel pointers to AVFrame data */
182  for (ch = 0; ch < avctx->channels; ch++) {
183  if (ac->output_element[ch])
184  ac->output_element[ch]->ret = (INTFLOAT *)ac->frame->extended_data[ch];
185  }
186 
187  return 0;
188 }
189 
190 struct elem_to_channel {
191  uint64_t av_position;
192  uint8_t syn_ele;
193  uint8_t elem_id;
194  uint8_t aac_position;
195 };
196 
197 static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
198  uint8_t (*layout_map)[3], int offset, uint64_t left,
199  uint64_t right, int pos)
200 {
201  if (layout_map[offset][0] == TYPE_CPE) {
202  e2c_vec[offset] = (struct elem_to_channel) {
203  .av_position = left | right,
204  .syn_ele = TYPE_CPE,
205  .elem_id = layout_map[offset][1],
206  .aac_position = pos
207  };
208  return 1;
209  } else {
210  e2c_vec[offset] = (struct elem_to_channel) {
211  .av_position = left,
212  .syn_ele = TYPE_SCE,
213  .elem_id = layout_map[offset][1],
214  .aac_position = pos
215  };
216  e2c_vec[offset + 1] = (struct elem_to_channel) {
217  .av_position = right,
218  .syn_ele = TYPE_SCE,
219  .elem_id = layout_map[offset + 1][1],
220  .aac_position = pos
221  };
222  return 2;
223  }
224 }
225 
226 static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
227  int *current)
228 {
229  int num_pos_channels = 0;
230  int first_cpe = 0;
231  int sce_parity = 0;
232  int i;
233  for (i = *current; i < tags; i++) {
234  if (layout_map[i][2] != pos)
235  break;
236  if (layout_map[i][0] == TYPE_CPE) {
237  if (sce_parity) {
238  if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
239  sce_parity = 0;
240  } else {
241  return -1;
242  }
243  }
244  num_pos_channels += 2;
245  first_cpe = 1;
246  } else {
247  num_pos_channels++;
248  sce_parity ^= 1;
249  }
250  }
251  if (sce_parity &&
252  ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
253  return -1;
254  *current = i;
255  return num_pos_channels;
256 }
257 
258 static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
259 {
260  int i, n, total_non_cc_elements;
261  struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
262  int num_front_channels, num_side_channels, num_back_channels;
263  uint64_t layout;
264 
265  if (FF_ARRAY_ELEMS(e2c_vec) < tags)
266  return 0;
267 
268  i = 0;
269  num_front_channels =
270  count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
271  if (num_front_channels < 0)
272  return 0;
273  num_side_channels =
274  count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
275  if (num_side_channels < 0)
276  return 0;
277  num_back_channels =
278  count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
279  if (num_back_channels < 0)
280  return 0;
281 
282  if (num_side_channels == 0 && num_back_channels >= 4) {
283  num_side_channels = 2;
284  num_back_channels -= 2;
285  }
286 
287  i = 0;
288  if (num_front_channels & 1) {
289  e2c_vec[i] = (struct elem_to_channel) {
290  .av_position = AV_CH_FRONT_CENTER,
291  .syn_ele = TYPE_SCE,
292  .elem_id = layout_map[i][1],
293  .aac_position = AAC_CHANNEL_FRONT
294  };
295  i++;
296  num_front_channels--;
297  }
298  if (num_front_channels >= 4) {
299  i += assign_pair(e2c_vec, layout_map, i,
300  AV_CH_FRONT_LEFT_OF_CENTER,
301  AV_CH_FRONT_RIGHT_OF_CENTER,
302  AAC_CHANNEL_FRONT);
303  num_front_channels -= 2;
304  }
305  if (num_front_channels >= 2) {
306  i += assign_pair(e2c_vec, layout_map, i,
307  AV_CH_FRONT_LEFT,
308  AV_CH_FRONT_RIGHT,
309  AAC_CHANNEL_FRONT);
310  num_front_channels -= 2;
311  }
312  while (num_front_channels >= 2) {
313  i += assign_pair(e2c_vec, layout_map, i,
314  UINT64_MAX,
315  UINT64_MAX,
316  AAC_CHANNEL_FRONT);
317  num_front_channels -= 2;
318  }
319 
320  if (num_side_channels >= 2) {
321  i += assign_pair(e2c_vec, layout_map, i,
322  AV_CH_SIDE_LEFT,
323  AV_CH_SIDE_RIGHT,
324  AAC_CHANNEL_FRONT);
325  num_side_channels -= 2;
326  }
327  while (num_side_channels >= 2) {
328  i += assign_pair(e2c_vec, layout_map, i,
329  UINT64_MAX,
330  UINT64_MAX,
331  AAC_CHANNEL_SIDE);
332  num_side_channels -= 2;
333  }
334 
335  while (num_back_channels >= 4) {
336  i += assign_pair(e2c_vec, layout_map, i,
337  UINT64_MAX,
338  UINT64_MAX,
339  AAC_CHANNEL_BACK);
340  num_back_channels -= 2;
341  }
342  if (num_back_channels >= 2) {
343  i += assign_pair(e2c_vec, layout_map, i,
344  AV_CH_BACK_LEFT,
345  AV_CH_BACK_RIGHT,
346  AAC_CHANNEL_BACK);
347  num_back_channels -= 2;
348  }
349  if (num_back_channels) {
350  e2c_vec[i] = (struct elem_to_channel) {
351  .av_position = AV_CH_BACK_CENTER,
352  .syn_ele = TYPE_SCE,
353  .elem_id = layout_map[i][1],
354  .aac_position = AAC_CHANNEL_BACK
355  };
356  i++;
357  num_back_channels--;
358  }
359 
360  if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
361  e2c_vec[i] = (struct elem_to_channel) {
362  .av_position = AV_CH_LOW_FREQUENCY,
363  .syn_ele = TYPE_LFE,
364  .elem_id = layout_map[i][1],
365  .aac_position = AAC_CHANNEL_LFE
366  };
367  i++;
368  }
369  while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
370  e2c_vec[i] = (struct elem_to_channel) {
371  .av_position = UINT64_MAX,
372  .syn_ele = TYPE_LFE,
373  .elem_id = layout_map[i][1],
374  .aac_position = AAC_CHANNEL_LFE
375  };
376  i++;
377  }
378 
379  // Must choose a stable sort
380  total_non_cc_elements = n = i;
381  do {
382  int next_n = 0;
383  for (i = 1; i < n; i++)
384  if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
385  FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
386  next_n = i;
387  }
388  n = next_n;
389  } while (n > 0);
390 
391  layout = 0;
392  for (i = 0; i < total_non_cc_elements; i++) {
393  layout_map[i][0] = e2c_vec[i].syn_ele;
394  layout_map[i][1] = e2c_vec[i].elem_id;
395  layout_map[i][2] = e2c_vec[i].aac_position;
396  if (e2c_vec[i].av_position != UINT64_MAX) {
397  layout |= e2c_vec[i].av_position;
398  }
399  }
400 
401  return layout;
402 }
403 
404 /**
405  * Save current output configuration if and only if it has been locked.
406  */
407 static void push_output_configuration(AACContext *ac) {
408  if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
409  ac->oc[0] = ac->oc[1];
410  }
411  ac->oc[1].status = OC_NONE;
412 }
413 
414 /**
415  * Restore the previous output configuration if and only if the current
416  * configuration is unlocked.
417  */
418 static void pop_output_configuration(AACContext *ac) {
419  if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
420  ac->oc[1] = ac->oc[0];
421  ac->avctx->channels = ac->oc[1].channels;
422  ac->avctx->channel_layout = ac->oc[1].channel_layout;
423  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
424  ac->oc[1].status, 0);
425  }
426 }
427 
428 /**
429  * Configure output channel order based on the current program
430  * configuration element.
431  *
432  * @return Returns error status. 0 - OK, !0 - error
433  */
434 static int output_configure(AACContext *ac,
435  uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
436  enum OCStatus oc_type, int get_new_frame)
437 {
438  AVCodecContext *avctx = ac->avctx;
439  int i, channels = 0, ret;
440  uint64_t layout = 0;
441  uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
442  uint8_t type_counts[TYPE_END] = { 0 };
443 
444  if (ac->oc[1].layout_map != layout_map) {
445  memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
446  ac->oc[1].layout_map_tags = tags;
447  }
448  for (i = 0; i < tags; i++) {
449  int type = layout_map[i][0];
450  int id = layout_map[i][1];
451  id_map[type][id] = type_counts[type]++;
452  }
453  // Try to sniff a reasonable channel order, otherwise output the
454  // channels in the order the PCE declared them.
455  if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
456  layout = sniff_channel_order(layout_map, tags);
457  for (i = 0; i < tags; i++) {
458  int type = layout_map[i][0];
459  int id = layout_map[i][1];
460  int iid = id_map[type][id];
461  int position = layout_map[i][2];
462  // Allocate or free elements depending on if they are in the
463  // current program configuration.
464  ret = che_configure(ac, position, type, iid, &channels);
465  if (ret < 0)
466  return ret;
467  ac->tag_che_map[type][id] = ac->che[type][iid];
468  }
469  if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
470  if (layout == AV_CH_FRONT_CENTER) {
471  layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
472  } else {
473  layout = 0;
474  }
475  }
476 
477  if (layout) avctx->channel_layout = layout;
478  ac->oc[1].channel_layout = layout;
479  avctx->channels = ac->oc[1].channels = channels;
480  ac->oc[1].status = oc_type;
481 
482  if (get_new_frame) {
483  if ((ret = frame_configure_elements(ac->avctx)) < 0)
484  return ret;
485  }
486 
487  return 0;
488 }
489 
490 static void flush(AVCodecContext *avctx)
491 {
492  AACContext *ac= avctx->priv_data;
493  int type, i, j;
494 
495  for (type = 3; type >= 0; type--) {
496  for (i = 0; i < MAX_ELEM_ID; i++) {
497  ChannelElement *che = ac->che[type][i];
498  if (che) {
499  for (j = 0; j <= 1; j++) {
500  memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
501  }
502  }
503  }
504  }
505 }
506 
507 /**
508  * Set up channel positions based on a default channel configuration
509  * as specified in table 1.17.
510  *
511  * @return Returns error status. 0 - OK, !0 - error
512  */
513 static int set_default_channel_config(AVCodecContext *avctx,
514  uint8_t (*layout_map)[3],
515  int *tags,
516  int channel_config)
517 {
518  if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
519  channel_config > 12) {
520  av_log(avctx, AV_LOG_ERROR,
521  "invalid default channel configuration (%d)\n",
522  channel_config);
523  return AVERROR_INVALIDDATA;
524  }
525  *tags = tags_per_config[channel_config];
526  memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
527  *tags * sizeof(*layout_map));
528 
529  /*
530  * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
531  * However, at least Nero AAC encoder encodes 7.1 streams using the default
532  * channel config 7, mapping the side channels of the original audio stream
533  * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
534  * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
535  * the incorrect streams as if they were correct (and as the encoder intended).
536  *
537  * As actual intended 7.1(wide) streams are very rare, default to assuming a
538  * 7.1 layout was intended.
539  */
540  if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
541  av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
542  " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
543  " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
544  layout_map[2][2] = AAC_CHANNEL_SIDE;
545  }
546 
547  return 0;
548 }
549 
550 static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
551 {
552  /* For PCE based channel configurations map the channels solely based
553  * on tags. */
554  if (!ac->oc[1].m4ac.chan_config) {
555  return ac->tag_che_map[type][elem_id];
556  }
557  // Allow single CPE stereo files to be signalled with mono configuration.
558  if (!ac->tags_mapped && type == TYPE_CPE &&
559  ac->oc[1].m4ac.chan_config == 1) {
560  uint8_t layout_map[MAX_ELEM_ID*4][3];
561  int layout_map_tags;
562  push_output_configuration(ac);
563 
564  av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
565 
566  if (set_default_channel_config(ac->avctx, layout_map,
567  &layout_map_tags, 2) < 0)
568  return NULL;
569  if (output_configure(ac, layout_map, layout_map_tags,
570  OC_TRIAL_FRAME, 1) < 0)
571  return NULL;
572 
573  ac->oc[1].m4ac.chan_config = 2;
574  ac->oc[1].m4ac.ps = 0;
575  }
576  // And vice-versa
577  if (!ac->tags_mapped && type == TYPE_SCE &&
578  ac->oc[1].m4ac.chan_config == 2) {
579  uint8_t layout_map[MAX_ELEM_ID * 4][3];
580  int layout_map_tags;
581  push_output_configuration(ac);
582 
583  av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
584 
585  if (set_default_channel_config(ac->avctx, layout_map,
586  &layout_map_tags, 1) < 0)
587  return NULL;
588  if (output_configure(ac, layout_map, layout_map_tags,
589  OC_TRIAL_FRAME, 1) < 0)
590  return NULL;
591 
592  ac->oc[1].m4ac.chan_config = 1;
593  if (ac->oc[1].m4ac.sbr)
594  ac->oc[1].m4ac.ps = -1;
595  }
596  /* For indexed channel configurations map the channels solely based
597  * on position. */
598  switch (ac->oc[1].m4ac.chan_config) {
599  case 12:
600  case 7:
601  if (ac->tags_mapped == 3 && type == TYPE_CPE) {
602  ac->tags_mapped++;
603  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
604  }
605  case 11:
606  if (ac->tags_mapped == 2 &&
607  ac->oc[1].m4ac.chan_config == 11 &&
608  type == TYPE_SCE) {
609  ac->tags_mapped++;
610  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
611  }
612  case 6:
613  /* Some streams incorrectly code 5.1 audio as
614  * SCE[0] CPE[0] CPE[1] SCE[1]
615  * instead of
616  * SCE[0] CPE[0] CPE[1] LFE[0].
617  * If we seem to have encountered such a stream, transfer
618  * the LFE[0] element to the SCE[1]'s mapping */
619  if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
620  if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
621  av_log(ac->avctx, AV_LOG_WARNING,
622  "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
623  type == TYPE_SCE ? "SCE" : "LFE", elem_id);
624  ac->warned_remapping_once++;
625  }
626  ac->tags_mapped++;
627  return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
628  }
629  case 5:
630  if (ac->tags_mapped == 2 && type == TYPE_CPE) {
631  ac->tags_mapped++;
632  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
633  }
634  case 4:
635  /* Some streams incorrectly code 4.0 audio as
636  * SCE[0] CPE[0] LFE[0]
637  * instead of
638  * SCE[0] CPE[0] SCE[1].
639  * If we seem to have encountered such a stream, transfer
640  * the SCE[1] element to the LFE[0]'s mapping */
641  if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
642  if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
643  av_log(ac->avctx, AV_LOG_WARNING,
644  "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
645  type == TYPE_SCE ? "SCE" : "LFE", elem_id);
646  ac->warned_remapping_once++;
647  }
648  ac->tags_mapped++;
649  return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
650  }
651  if (ac->tags_mapped == 2 &&
652  ac->oc[1].m4ac.chan_config == 4 &&
653  type == TYPE_SCE) {
654  ac->tags_mapped++;
655  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
656  }
657  case 3:
658  case 2:
659  if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
660  type == TYPE_CPE) {
661  ac->tags_mapped++;
662  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
663  } else if (ac->oc[1].m4ac.chan_config == 2) {
664  return NULL;
665  }
666  case 1:
667  if (!ac->tags_mapped && type == TYPE_SCE) {
668  ac->tags_mapped++;
669  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
670  }
671  default:
672  return NULL;
673  }
674 }
675 
676 /**
677  * Decode an array of 4 bit element IDs, optionally interleaved with a
678  * stereo/mono switching bit.
679  *
680  * @param type speaker type/position for these channels
681  */
682 static void decode_channel_map(uint8_t layout_map[][3],
683  enum ChannelPosition type,
684  GetBitContext *gb, int n)
685 {
686  while (n--) {
687  enum RawDataBlockType syn_ele;
688  switch (type) {
689  case AAC_CHANNEL_FRONT:
690  case AAC_CHANNEL_BACK:
691  case AAC_CHANNEL_SIDE:
692  syn_ele = get_bits1(gb);
693  break;
694  case AAC_CHANNEL_CC:
695  skip_bits1(gb);
696  syn_ele = TYPE_CCE;
697  break;
698  case AAC_CHANNEL_LFE:
699  syn_ele = TYPE_LFE;
700  break;
701  default:
702  // AAC_CHANNEL_OFF has no channel map
703  av_assert0(0);
704  }
705  layout_map[0][0] = syn_ele;
706  layout_map[0][1] = get_bits(gb, 4);
707  layout_map[0][2] = type;
708  layout_map++;
709  }
710 }
711 
712 /**
713  * Decode program configuration element; reference: table 4.2.
714  *
715  * @return Returns error status. 0 - OK, !0 - error
716  */
717 static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
718  uint8_t (*layout_map)[3],
719  GetBitContext *gb)
720 {
721  int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
722  int sampling_index;
723  int comment_len;
724  int tags;
725 
726  skip_bits(gb, 2); // object_type
727 
728  sampling_index = get_bits(gb, 4);
729  if (m4ac->sampling_index != sampling_index)
730  av_log(avctx, AV_LOG_WARNING,
731  "Sample rate index in program config element does not "
732  "match the sample rate index configured by the container.\n");
733 
734  num_front = get_bits(gb, 4);
735  num_side = get_bits(gb, 4);
736  num_back = get_bits(gb, 4);
737  num_lfe = get_bits(gb, 2);
738  num_assoc_data = get_bits(gb, 3);
739  num_cc = get_bits(gb, 4);
740 
741  if (get_bits1(gb))
742  skip_bits(gb, 4); // mono_mixdown_tag
743  if (get_bits1(gb))
744  skip_bits(gb, 4); // stereo_mixdown_tag
745 
746  if (get_bits1(gb))
747  skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
748 
749  if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
750  av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
751  return -1;
752  }
753  decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front);
754  tags = num_front;
755  decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side);
756  tags += num_side;
757  decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back);
758  tags += num_back;
759  decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe);
760  tags += num_lfe;
761 
762  skip_bits_long(gb, 4 * num_assoc_data);
763 
764  decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc);
765  tags += num_cc;
766 
767  align_get_bits(gb);
768 
769  /* comment field, first byte is length */
770  comment_len = get_bits(gb, 8) * 8;
771  if (get_bits_left(gb) < comment_len) {
772  av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
773  return AVERROR_INVALIDDATA;
774  }
775  skip_bits_long(gb, comment_len);
776  return tags;
777 }
778 
779 /**
780  * Decode GA "General Audio" specific configuration; reference: table 4.1.
781  *
782  * @param ac pointer to AACContext, may be null
783  * @param avctx pointer to AVCCodecContext, used for logging
784  *
785  * @return Returns error status. 0 - OK, !0 - error
786  */
787 static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
788  GetBitContext *gb,
789  MPEG4AudioConfig *m4ac,
790  int channel_config)
791 {
792  int extension_flag, ret, ep_config, res_flags;
793  uint8_t layout_map[MAX_ELEM_ID*4][3];
794  int tags = 0;
795 
796  if (get_bits1(gb)) { // frameLengthFlag
797  avpriv_request_sample(avctx, "960/120 MDCT window");
798  return AVERROR_PATCHWELCOME;
799  }
800  m4ac->frame_length_short = 0;
801 
802  if (get_bits1(gb)) // dependsOnCoreCoder
803  skip_bits(gb, 14); // coreCoderDelay
804  extension_flag = get_bits1(gb);
805 
806  if (m4ac->object_type == AOT_AAC_SCALABLE ||
807  m4ac->object_type == AOT_ER_AAC_SCALABLE)
808  skip_bits(gb, 3); // layerNr
809 
810  if (channel_config == 0) {
811  skip_bits(gb, 4); // element_instance_tag
812  tags = decode_pce(avctx, m4ac, layout_map, gb);
813  if (tags < 0)
814  return tags;
815  } else {
816  if ((ret = set_default_channel_config(avctx, layout_map,
817  &tags, channel_config)))
818  return ret;
819  }
820 
821  if (count_channels(layout_map, tags) > 1) {
822  m4ac->ps = 0;
823  } else if (m4ac->sbr == 1 && m4ac->ps == -1)
824  m4ac->ps = 1;
825 
826  if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
827  return ret;
828 
829  if (extension_flag) {
830  switch (m4ac->object_type) {
831  case AOT_ER_BSAC:
832  skip_bits(gb, 5); // numOfSubFrame
833  skip_bits(gb, 11); // layer_length
834  break;
835  case AOT_ER_AAC_LC:
836  case AOT_ER_AAC_LTP:
837  case AOT_ER_AAC_SCALABLE:
838  case AOT_ER_AAC_LD:
839  res_flags = get_bits(gb, 3);
840  if (res_flags) {
841  avpriv_report_missing_feature(avctx,
842  "AAC data resilience (flags %x)",
843  res_flags);
844  return AVERROR_PATCHWELCOME;
845  }
846  break;
847  }
848  skip_bits1(gb); // extensionFlag3 (TBD in version 3)
849  }
850  switch (m4ac->object_type) {
851  case AOT_ER_AAC_LC:
852  case AOT_ER_AAC_LTP:
853  case AOT_ER_AAC_SCALABLE:
854  case AOT_ER_AAC_LD:
855  ep_config = get_bits(gb, 2);
856  if (ep_config) {
857  avpriv_report_missing_feature(avctx,
858  "epConfig %d", ep_config);
859  return AVERROR_PATCHWELCOME;
860  }
861  }
862  return 0;
863 }
864 
865 static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
866  GetBitContext *gb,
867  MPEG4AudioConfig *m4ac,
868  int channel_config)
869 {
870  int ret, ep_config, res_flags;
871  uint8_t layout_map[MAX_ELEM_ID*4][3];
872  int tags = 0;
873  const int ELDEXT_TERM = 0;
874 
875  m4ac->ps = 0;
876  m4ac->sbr = 0;
877 #if USE_FIXED
878  if (get_bits1(gb)) { // frameLengthFlag
879  avpriv_request_sample(avctx, "960/120 MDCT window");
880  return AVERROR_PATCHWELCOME;
881  }
882 #else
883  m4ac->frame_length_short = get_bits1(gb);
884 #endif
885  res_flags = get_bits(gb, 3);
886  if (res_flags) {
887  avpriv_report_missing_feature(avctx,
888  "AAC data resilience (flags %x)",
889  res_flags);
890  return AVERROR_PATCHWELCOME;
891  }
892 
893  if (get_bits1(gb)) { // ldSbrPresentFlag
894  avpriv_report_missing_feature(avctx,
895  "Low Delay SBR");
896  return AVERROR_PATCHWELCOME;
897  }
898 
899  while (get_bits(gb, 4) != ELDEXT_TERM) {
900  int len = get_bits(gb, 4);
901  if (len == 15)
902  len += get_bits(gb, 8);
903  if (len == 15 + 255)
904  len += get_bits(gb, 16);
905  if (get_bits_left(gb) < len * 8 + 4) {
906  av_log(avctx, AV_LOG_ERROR, overread_err);
907  return AVERROR_INVALIDDATA;
908  }
909  skip_bits_long(gb, 8 * len);
910  }
911 
912  if ((ret = set_default_channel_config(avctx, layout_map,
913  &tags, channel_config)))
914  return ret;
915 
916  if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
917  return ret;
918 
919  ep_config = get_bits(gb, 2);
920  if (ep_config) {
921  avpriv_report_missing_feature(avctx,
922  "epConfig %d", ep_config);
923  return AVERROR_PATCHWELCOME;
924  }
925  return 0;
926 }
927 
928 /**
929  * Decode audio specific configuration; reference: table 1.13.
930  *
931  * @param ac pointer to AACContext, may be null
932  * @param avctx pointer to AVCCodecContext, used for logging
933  * @param m4ac pointer to MPEG4AudioConfig, used for parsing
934  * @param data pointer to buffer holding an audio specific config
935  * @param bit_size size of audio specific config or data in bits
936  * @param sync_extension look for an appended sync extension
937  *
938  * @return Returns error status or number of consumed bits. <0 - error
939  */
940 static int decode_audio_specific_config(AACContext *ac,
941  AVCodecContext *avctx,
942  MPEG4AudioConfig *m4ac,
943  const uint8_t *data, int64_t bit_size,
944  int sync_extension)
945 {
946  GetBitContext gb;
947  int i, ret;
948 
949  if (bit_size < 0 || bit_size > INT_MAX) {
950  av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n");
951  return AVERROR_INVALIDDATA;
952  }
953 
954  ff_dlog(avctx, "audio specific config size %d\n", (int)bit_size >> 3);
955  for (i = 0; i < bit_size >> 3; i++)
956  ff_dlog(avctx, "%02x ", data[i]);
957  ff_dlog(avctx, "\n");
958 
959  if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
960  return ret;
961 
962  if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size,
963  sync_extension)) < 0)
964  return AVERROR_INVALIDDATA;
965  if (m4ac->sampling_index > 12) {
966  av_log(avctx, AV_LOG_ERROR,
967  "invalid sampling rate index %d\n",
968  m4ac->sampling_index);
969  return AVERROR_INVALIDDATA;
970  }
971  if (m4ac->object_type == AOT_ER_AAC_LD &&
972  (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
973  av_log(avctx, AV_LOG_ERROR,
974  "invalid low delay sampling rate index %d\n",
975  m4ac->sampling_index);
976  return AVERROR_INVALIDDATA;
977  }
978 
979  skip_bits_long(&gb, i);
980 
981  switch (m4ac->object_type) {
982  case AOT_AAC_MAIN:
983  case AOT_AAC_LC:
984  case AOT_AAC_LTP:
985  case AOT_ER_AAC_LC:
986  case AOT_ER_AAC_LD:
987  if ((ret = decode_ga_specific_config(ac, avctx, &gb,
988  m4ac, m4ac->chan_config)) < 0)
989  return ret;
990  break;
991  case AOT_ER_AAC_ELD:
992  if ((ret = decode_eld_specific_config(ac, avctx, &gb,
993  m4ac, m4ac->chan_config)) < 0)
994  return ret;
995  break;
996  default:
997  avpriv_report_missing_feature(avctx,
998  "Audio object type %s%d",
999  m4ac->sbr == 1 ? "SBR+" : "",
1000  m4ac->object_type);
1001  return AVERROR(ENOSYS);
1002  }
1003 
1004  ff_dlog(avctx,
1005  "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
1006  m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
1007  m4ac->sample_rate, m4ac->sbr,
1008  m4ac->ps);
1009 
1010  return get_bits_count(&gb);
1011 }
1012 
1013 /**
1014  * linear congruential pseudorandom number generator
1015  *
1016  * @param previous_val pointer to the current state of the generator
1017  *
1018  * @return Returns a 32-bit pseudorandom integer
1019  */
1020 static av_always_inline int lcg_random(unsigned previous_val)
1021 {
1022  union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
1023  return v.s;
1024 }
1025 
1026 static void reset_all_predictors(PredictorState *ps)
1027 {
1028  int i;
1029  for (i = 0; i < MAX_PREDICTORS; i++)
1030  reset_predict_state(&ps[i]);
1031 }
1032 
1033 static int sample_rate_idx (int rate)
1034 {
1035  if (92017 <= rate) return 0;
1036  else if (75132 <= rate) return 1;
1037  else if (55426 <= rate) return 2;
1038  else if (46009 <= rate) return 3;
1039  else if (37566 <= rate) return 4;
1040  else if (27713 <= rate) return 5;
1041  else if (23004 <= rate) return 6;
1042  else if (18783 <= rate) return 7;
1043  else if (13856 <= rate) return 8;
1044  else if (11502 <= rate) return 9;
1045  else if (9391 <= rate) return 10;
1046  else return 11;
1047 }
1048 
1049 static void reset_predictor_group(PredictorState *ps, int group_num)
1050 {
1051  int i;
1052  for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
1053  reset_predict_state(&ps[i]);
1054 }
1055 
1056 #define AAC_INIT_VLC_STATIC(num, size) \
1057  INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
1058  ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \
1059  sizeof(ff_aac_spectral_bits[num][0]), \
1060  ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
1061  sizeof(ff_aac_spectral_codes[num][0]), \
1062  size);
1063 
1064 static void aacdec_init(AACContext *ac);
1065 
1066 static av_cold int aac_decode_init(AVCodecContext *avctx)
1067 {
1068  AACContext *ac = avctx->priv_data;
1069  int ret;
1070 
1071  ac->avctx = avctx;
1072  ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
1073 
1074  aacdec_init(ac);
1075 #if USE_FIXED
1076  avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
1077 #else
1078  avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1079 #endif /* USE_FIXED */
1080 
1081  if (avctx->extradata_size > 0) {
1082  if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
1083  avctx->extradata,
1084  avctx->extradata_size * 8LL,
1085  1)) < 0)
1086  return ret;
1087  } else {
1088  int sr, i;
1089  uint8_t layout_map[MAX_ELEM_ID*4][3];
1090  int layout_map_tags;
1091 
1092  sr = sample_rate_idx(avctx->sample_rate);
1093  ac->oc[1].m4ac.sampling_index = sr;
1094  ac->oc[1].m4ac.channels = avctx->channels;
1095  ac->oc[1].m4ac.sbr = -1;
1096  ac->oc[1].m4ac.ps = -1;
1097 
1098  for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
1099  if (ff_mpeg4audio_channels[i] == avctx->channels)
1100  break;
1101  if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
1102  i = 0;
1103  }
1104  ac->oc[1].m4ac.chan_config = i;
1105 
1106  if (ac->oc[1].m4ac.chan_config) {
1107  int ret = set_default_channel_config(avctx, layout_map,
1108  &layout_map_tags, ac->oc[1].m4ac.chan_config);
1109  if (!ret)
1110  output_configure(ac, layout_map, layout_map_tags,
1111  OC_GLOBAL_HDR, 0);
1112  else if (avctx->err_recognition & AV_EF_EXPLODE)
1113  return AVERROR_INVALIDDATA;
1114  }
1115  }
1116 
1117  if (avctx->channels > MAX_CHANNELS) {
1118  av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
1119  return AVERROR_INVALIDDATA;
1120  }
1121 
1122  AAC_INIT_VLC_STATIC( 0, 304);
1123  AAC_INIT_VLC_STATIC( 1, 270);
1124  AAC_INIT_VLC_STATIC( 2, 550);
1125  AAC_INIT_VLC_STATIC( 3, 300);
1126  AAC_INIT_VLC_STATIC( 4, 328);
1127  AAC_INIT_VLC_STATIC( 5, 294);
1128  AAC_INIT_VLC_STATIC( 6, 306);
1129  AAC_INIT_VLC_STATIC( 7, 268);
1130  AAC_INIT_VLC_STATIC( 8, 510);
1131  AAC_INIT_VLC_STATIC( 9, 366);
1132  AAC_INIT_VLC_STATIC(10, 462);
1133 
1134  AAC_RENAME(ff_aac_sbr_init)();
1135 
1136 #if USE_FIXED
1137  ac->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & AV_CODEC_FLAG_BITEXACT);
1138 #else
1139  ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
1140 #endif /* USE_FIXED */
1141  if (!ac->fdsp) {
1142  return AVERROR(ENOMEM);
1143  }
1144 
1145  ac->random_state = 0x1f2e3d4c;
1146 
1147  ff_aac_tableinit();
1148 
1149  INIT_VLC_STATIC(&vlc_scalefactors, 7,
1150  FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
1151  ff_aac_scalefactor_bits,
1152  sizeof(ff_aac_scalefactor_bits[0]),
1153  sizeof(ff_aac_scalefactor_bits[0]),
1154  ff_aac_scalefactor_code,
1155  sizeof(ff_aac_scalefactor_code[0]),
1156  sizeof(ff_aac_scalefactor_code[0]),
1157  352);
1158 
1159  AAC_RENAME_32(ff_mdct_init)(&ac->mdct, 11, 1, 1.0 / RANGE15(1024.0));
1160  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld, 10, 1, 1.0 / RANGE15(512.0));
1161  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small, 8, 1, 1.0 / RANGE15(128.0));
1162  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp, 11, 0, RANGE15(-2.0));
1163 #if !USE_FIXED
1164  ret = ff_imdct15_init(&ac->mdct480, 5);
1165  if (ret < 0)
1166  return ret;
1167 #endif
1168  // window initialization
1169  AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_1024), 4.0, 1024);
1170  AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_128), 6.0, 128);
1171  AAC_RENAME(ff_init_ff_sine_windows)(10);
1172  AAC_RENAME(ff_init_ff_sine_windows)( 9);
1173  AAC_RENAME(ff_init_ff_sine_windows)( 7);
1174 
1175  AAC_RENAME(cbrt_tableinit)();
1176 
1177  return 0;
1178 }
1179 
1180 /**
1181  * Skip data_stream_element; reference: table 4.10.
1182  */
1183 static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
1184 {
1185  int byte_align = get_bits1(gb);
1186  int count = get_bits(gb, 8);
1187  if (count == 255)
1188  count += get_bits(gb, 8);
1189  if (byte_align)
1190  align_get_bits(gb);
1191 
1192  if (get_bits_left(gb) < 8 * count) {
1193  av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
1194  return AVERROR_INVALIDDATA;
1195  }
1196  skip_bits_long(gb, 8 * count);
1197  return 0;
1198 }
1199 
1200 static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
1201  GetBitContext *gb)
1202 {
1203  int sfb;
1204  if (get_bits1(gb)) {
1205  ics->predictor_reset_group = get_bits(gb, 5);
1206  if (ics->predictor_reset_group == 0 ||
1207  ics->predictor_reset_group > 30) {
1208  av_log(ac->avctx, AV_LOG_ERROR,
1209  "Invalid Predictor Reset Group.\n");
1210  return AVERROR_INVALIDDATA;
1211  }
1212  }
1213  for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
1214  ics->prediction_used[sfb] = get_bits1(gb);
1215  }
1216  return 0;
1217 }
1218 
1219 /**
1220  * Decode Long Term Prediction data; reference: table 4.xx.
1221  */
1222 static void decode_ltp(LongTermPrediction *ltp,
1223  GetBitContext *gb, uint8_t max_sfb)
1224 {
1225  int sfb;
1226 
1227  ltp->lag = get_bits(gb, 11);
1228  ltp->coef = ltp_coef[get_bits(gb, 3)];
1229  for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
1230  ltp->used[sfb] = get_bits1(gb);
1231 }
1232 
1233 /**
1234  * Decode Individual Channel Stream info; reference: table 4.6.
1235  */
1236 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
1237  GetBitContext *gb)
1238 {
1239  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
1240  const int aot = m4ac->object_type;
1241  const int sampling_index = m4ac->sampling_index;
1242  if (aot != AOT_ER_AAC_ELD) {
1243  if (get_bits1(gb)) {
1244  av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
1245  if (ac->avctx->err_recognition & AV_EF_BITSTREAM)
1246  return AVERROR_INVALIDDATA;
1247  }
1248  ics->window_sequence[1] = ics->window_sequence[0];
1249  ics->window_sequence[0] = get_bits(gb, 2);
1250  if (aot == AOT_ER_AAC_LD &&
1251  ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
1252  av_log(ac->avctx, AV_LOG_ERROR,
1253  "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
1254  "window sequence %d found.\n", ics->window_sequence[0]);
1255  ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
1256  return AVERROR_INVALIDDATA;
1257  }
1258  ics->use_kb_window[1] = ics->use_kb_window[0];
1259  ics->use_kb_window[0] = get_bits1(gb);
1260  }
1261  ics->num_window_groups = 1;
1262  ics->group_len[0] = 1;
1263  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1264  int i;
1265  ics->max_sfb = get_bits(gb, 4);
1266  for (i = 0; i < 7; i++) {
1267  if (get_bits1(gb)) {
1268  ics->group_len[ics->num_window_groups - 1]++;
1269  } else {
1270  ics->num_window_groups++;
1271  ics->group_len[ics->num_window_groups - 1] = 1;
1272  }
1273  }
1274  ics->num_windows = 8;
1275  ics->swb_offset = ff_swb_offset_128[sampling_index];
1276  ics->num_swb = ff_aac_num_swb_128[sampling_index];
1277  ics->tns_max_bands = ff_tns_max_bands_128[sampling_index];
1278  ics->predictor_present = 0;
1279  } else {
1280  ics->max_sfb = get_bits(gb, 6);
1281  ics->num_windows = 1;
1282  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
1283  if (m4ac->frame_length_short) {
1284  ics->swb_offset = ff_swb_offset_480[sampling_index];
1285  ics->num_swb = ff_aac_num_swb_480[sampling_index];
1286  ics->tns_max_bands = ff_tns_max_bands_480[sampling_index];
1287  } else {
1288  ics->swb_offset = ff_swb_offset_512[sampling_index];
1289  ics->num_swb = ff_aac_num_swb_512[sampling_index];
1290  ics->tns_max_bands = ff_tns_max_bands_512[sampling_index];
1291  }
1292  if (!ics->num_swb || !ics->swb_offset)
1293  return AVERROR_BUG;
1294  } else {
1295  ics->swb_offset = ff_swb_offset_1024[sampling_index];
1296  ics->num_swb = ff_aac_num_swb_1024[sampling_index];
1297  ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
1298  }
1299  if (aot != AOT_ER_AAC_ELD) {
1300  ics->predictor_present = get_bits1(gb);
1301  ics->predictor_reset_group = 0;
1302  }
1303  if (ics->predictor_present) {
1304  if (aot == AOT_AAC_MAIN) {
1305  if (decode_prediction(ac, ics, gb)) {
1306  goto fail;
1307  }
1308  } else if (aot == AOT_AAC_LC ||
1309  aot == AOT_ER_AAC_LC) {
1310  av_log(ac->avctx, AV_LOG_ERROR,
1311  "Prediction is not allowed in AAC-LC.\n");
1312  goto fail;
1313  } else {
1314  if (aot == AOT_ER_AAC_LD) {
1315  av_log(ac->avctx, AV_LOG_ERROR,
1316  "LTP in ER AAC LD not yet implemented.\n");
1317  return AVERROR_PATCHWELCOME;
1318  }
1319  if ((ics->ltp.present = get_bits(gb, 1)))
1320  decode_ltp(&ics->ltp, gb, ics->max_sfb);
1321  }
1322  }
1323  }
1324 
1325  if (ics->max_sfb > ics->num_swb) {
1326  av_log(ac->avctx, AV_LOG_ERROR,
1327  "Number of scalefactor bands in group (%d) "
1328  "exceeds limit (%d).\n",
1329  ics->max_sfb, ics->num_swb);
1330  goto fail;
1331  }
1332 
1333  return 0;
1334 fail:
1335  ics->max_sfb = 0;
1336  return AVERROR_INVALIDDATA;
1337 }
1338 
1339 /**
1340  * Decode band types (section_data payload); reference: table 4.46.
1341  *
1342  * @param band_type array of the used band type
1343  * @param band_type_run_end array of the last scalefactor band of a band type run
1344  *
1345  * @return Returns error status. 0 - OK, !0 - error
1346  */
1347 static int decode_band_types(AACContext *ac, enum BandType band_type[120],
1348  int band_type_run_end[120], GetBitContext *gb,
1349  IndividualChannelStream *ics)
1350 {
1351  int g, idx = 0;
1352  const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
1353  for (g = 0; g < ics->num_window_groups; g++) {
1354  int k = 0;
1355  while (k < ics->max_sfb) {
1356  uint8_t sect_end = k;
1357  int sect_len_incr;
1358  int sect_band_type = get_bits(gb, 4);
1359  if (sect_band_type == 12) {
1360  av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
1361  return AVERROR_INVALIDDATA;
1362  }
1363  do {
1364  sect_len_incr = get_bits(gb, bits);
1365  sect_end += sect_len_incr;
1366  if (get_bits_left(gb) < 0) {
1367  av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
1368  return AVERROR_INVALIDDATA;
1369  }
1370  if (sect_end > ics->max_sfb) {
1371  av_log(ac->avctx, AV_LOG_ERROR,
1372  "Number of bands (%d) exceeds limit (%d).\n",
1373  sect_end, ics->max_sfb);
1374  return AVERROR_INVALIDDATA;
1375  }
1376  } while (sect_len_incr == (1 << bits) - 1);
1377  for (; k < sect_end; k++) {
1378  band_type [idx] = sect_band_type;
1379  band_type_run_end[idx++] = sect_end;
1380  }
1381  }
1382  }
1383  return 0;
1384 }
1385 
1386 /**
1387  * Decode scalefactors; reference: table 4.47.
1388  *
1389  * @param global_gain first scalefactor value as scalefactors are differentially coded
1390  * @param band_type array of the used band type
1391  * @param band_type_run_end array of the last scalefactor band of a band type run
1392  * @param sf array of scalefactors or intensity stereo positions
1393  *
1394  * @return Returns error status. 0 - OK, !0 - error
1395  */
1396 static int decode_scalefactors(AACContext *ac, INTFLOAT sf[120], GetBitContext *gb,
1397  unsigned int global_gain,
1398  IndividualChannelStream *ics,
1399  enum BandType band_type[120],
1400  int band_type_run_end[120])
1401 {
1402  int g, i, idx = 0;
1403  int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 };
1404  int clipped_offset;
1405  int noise_flag = 1;
1406  for (g = 0; g < ics->num_window_groups; g++) {
1407  for (i = 0; i < ics->max_sfb;) {
1408  int run_end = band_type_run_end[idx];
1409  if (band_type[idx] == ZERO_BT) {
1410  for (; i < run_end; i++, idx++)
1411  sf[idx] = FIXR(0.);
1412  } else if ((band_type[idx] == INTENSITY_BT) ||
1413  (band_type[idx] == INTENSITY_BT2)) {
1414  for (; i < run_end; i++, idx++) {
1415  offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1416  clipped_offset = av_clip(offset[2], -155, 100);
1417  if (offset[2] != clipped_offset) {
1418  avpriv_request_sample(ac->avctx,
1419  "If you heard an audible artifact, there may be a bug in the decoder. "
1420  "Clipped intensity stereo position (%d -> %d)",
1421  offset[2], clipped_offset);
1422  }
1423 #if USE_FIXED
1424  sf[idx] = 100 - clipped_offset;
1425 #else
1426  sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
1427 #endif /* USE_FIXED */
1428  }
1429  } else if (band_type[idx] == NOISE_BT) {
1430  for (; i < run_end; i++, idx++) {
1431  if (noise_flag-- > 0)
1432  offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
1433  else
1434  offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1435  clipped_offset = av_clip(offset[1], -100, 155);
1436  if (offset[1] != clipped_offset) {
1437  avpriv_request_sample(ac->avctx,
1438  "If you heard an audible artifact, there may be a bug in the decoder. "
1439  "Clipped noise gain (%d -> %d)",
1440  offset[1], clipped_offset);
1441  }
1442 #if USE_FIXED
1443  sf[idx] = -(100 + clipped_offset);
1444 #else
1445  sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
1446 #endif /* USE_FIXED */
1447  }
1448  } else {
1449  for (; i < run_end; i++, idx++) {
1450  offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1451  if (offset[0] > 255U) {
1452  av_log(ac->avctx, AV_LOG_ERROR,
1453  "Scalefactor (%d) out of range.\n", offset[0]);
1454  return AVERROR_INVALIDDATA;
1455  }
1456 #if USE_FIXED
1457  sf[idx] = -offset[0];
1458 #else
1459  sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
1460 #endif /* USE_FIXED */
1461  }
1462  }
1463  }
1464  }
1465  return 0;
1466 }
1467 
1468 /**
1469  * Decode pulse data; reference: table 4.7.
1470  */
1471 static int decode_pulses(Pulse *pulse, GetBitContext *gb,
1472  const uint16_t *swb_offset, int num_swb)
1473 {
1474  int i, pulse_swb;
1475  pulse->num_pulse = get_bits(gb, 2) + 1;
1476  pulse_swb = get_bits(gb, 6);
1477  if (pulse_swb >= num_swb)
1478  return -1;
1479  pulse->pos[0] = swb_offset[pulse_swb];
1480  pulse->pos[0] += get_bits(gb, 5);
1481  if (pulse->pos[0] >= swb_offset[num_swb])
1482  return -1;
1483  pulse->amp[0] = get_bits(gb, 4);
1484  for (i = 1; i < pulse->num_pulse; i++) {
1485  pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
1486  if (pulse->pos[i] >= swb_offset[num_swb])
1487  return -1;
1488  pulse->amp[i] = get_bits(gb, 4);
1489  }
1490  return 0;
1491 }
1492 
1493 /**
1494  * Decode Temporal Noise Shaping data; reference: table 4.48.
1495  *
1496  * @return Returns error status. 0 - OK, !0 - error
1497  */
1498 static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
1499  GetBitContext *gb, const IndividualChannelStream *ics)
1500 {
1501  int w, filt, i, coef_len, coef_res, coef_compress;
1502  const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
1503  const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
1504  for (w = 0; w < ics->num_windows; w++) {
1505  if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
1506  coef_res = get_bits1(gb);
1507 
1508  for (filt = 0; filt < tns->n_filt[w]; filt++) {
1509  int tmp2_idx;
1510  tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
1511 
1512  if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
1513  av_log(ac->avctx, AV_LOG_ERROR,
1514  "TNS filter order %d is greater than maximum %d.\n",
1515  tns->order[w][filt], tns_max_order);
1516  tns->order[w][filt] = 0;
1517  return AVERROR_INVALIDDATA;
1518  }
1519  if (tns->order[w][filt]) {
1520  tns->direction[w][filt] = get_bits1(gb);
1521  coef_compress = get_bits1(gb);
1522  coef_len = coef_res + 3 - coef_compress;
1523  tmp2_idx = 2 * coef_compress + coef_res;
1524 
1525  for (i = 0; i < tns->order[w][filt]; i++)
1526  tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
1527  }
1528  }
1529  }
1530  }
1531  return 0;
1532 }
1533 
1534 /**
1535  * Decode Mid/Side data; reference: table 4.54.
1536  *
1537  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
1538  * [1] mask is decoded from bitstream; [2] mask is all 1s;
1539  * [3] reserved for scalable AAC
1540  */
1541 static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
1542  int ms_present)
1543 {
1544  int idx;
1545  int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
1546  if (ms_present == 1) {
1547  for (idx = 0; idx < max_idx; idx++)
1548  cpe->ms_mask[idx] = get_bits1(gb);
1549  } else if (ms_present == 2) {
1550  memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
1551  }
1552 }
1553 
1554 /**
1555  * Decode spectral data; reference: table 4.50.
1556  * Dequantize and scale spectral data; reference: 4.6.3.3.
1557  *
1558  * @param coef array of dequantized, scaled spectral data
1559  * @param sf array of scalefactors or intensity stereo positions
1560  * @param pulse_present set if pulses are present
1561  * @param pulse pointer to pulse data struct
1562  * @param band_type array of the used band type
1563  *
1564  * @return Returns error status. 0 - OK, !0 - error
1565  */
1566 static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024],
1567  GetBitContext *gb, const INTFLOAT sf[120],
1568  int pulse_present, const Pulse *pulse,
1569  const IndividualChannelStream *ics,
1570  enum BandType band_type[120])
1571 {
1572  int i, k, g, idx = 0;
1573  const int c = 1024 / ics->num_windows;
1574  const uint16_t *offsets = ics->swb_offset;
1575  INTFLOAT *coef_base = coef;
1576 
1577  for (g = 0; g < ics->num_windows; g++)
1578  memset(coef + g * 128 + offsets[ics->max_sfb], 0,
1579  sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
1580 
1581  for (g = 0; g < ics->num_window_groups; g++) {
1582  unsigned g_len = ics->group_len[g];
1583 
1584  for (i = 0; i < ics->max_sfb; i++, idx++) {
1585  const unsigned cbt_m1 = band_type[idx] - 1;
1586  INTFLOAT *cfo = coef + offsets[i];
1587  int off_len = offsets[i + 1] - offsets[i];
1588  int group;
1589 
1590  if (cbt_m1 >= INTENSITY_BT2 - 1) {
1591  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1592  memset(cfo, 0, off_len * sizeof(*cfo));
1593  }
1594  } else if (cbt_m1 == NOISE_BT - 1) {
1595  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1596 #if !USE_FIXED
1597  float scale;
1598 #endif /* !USE_FIXED */
1599  INTFLOAT band_energy;
1600 
1601  for (k = 0; k < off_len; k++) {
1602  ac->random_state = lcg_random(ac->random_state);
1603 #if USE_FIXED
1604  cfo[k] = ac->random_state >> 3;
1605 #else
1606  cfo[k] = ac->random_state;
1607 #endif /* USE_FIXED */
1608  }
1609 
1610 #if USE_FIXED
1611  band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
1612  band_energy = fixed_sqrt(band_energy, 31);
1613  noise_scale(cfo, sf[idx], band_energy, off_len);
1614 #else
1615  band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
1616  scale = sf[idx] / sqrtf(band_energy);
1617  ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
1618 #endif /* USE_FIXED */
1619  }
1620  } else {
1621 #if !USE_FIXED
1622  const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1623 #endif /* !USE_FIXED */
1624  const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1625  VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1626  OPEN_READER(re, gb);
1627 
1628  switch (cbt_m1 >> 1) {
1629  case 0:
1630  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1631  INTFLOAT *cf = cfo;
1632  int len = off_len;
1633 
1634  do {
1635  int code;
1636  unsigned cb_idx;
1637 
1638  UPDATE_CACHE(re, gb);
1639  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1640  cb_idx = cb_vector_idx[code];
1641 #if USE_FIXED
1642  cf = DEC_SQUAD(cf, cb_idx);
1643 #else
1644  cf = VMUL4(cf, vq, cb_idx, sf + idx);
1645 #endif /* USE_FIXED */
1646  } while (len -= 4);
1647  }
1648  break;
1649 
1650  case 1:
1651  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1652  INTFLOAT *cf = cfo;
1653  int len = off_len;
1654 
1655  do {
1656  int code;
1657  unsigned nnz;
1658  unsigned cb_idx;
1659  uint32_t bits;
1660 
1661  UPDATE_CACHE(re, gb);
1662  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1663  cb_idx = cb_vector_idx[code];
1664  nnz = cb_idx >> 8 & 15;
1665  bits = nnz ? GET_CACHE(re, gb) : 0;
1666  LAST_SKIP_BITS(re, gb, nnz);
1667 #if USE_FIXED
1668  cf = DEC_UQUAD(cf, cb_idx, bits);
1669 #else
1670  cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1671 #endif /* USE_FIXED */
1672  } while (len -= 4);
1673  }
1674  break;
1675 
1676  case 2:
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_SPAIR(cf, cb_idx);
1690 #else
1691  cf = VMUL2(cf, vq, cb_idx, sf + idx);
1692 #endif /* USE_FIXED */
1693  } while (len -= 2);
1694  }
1695  break;
1696 
1697  case 3:
1698  case 4:
1699  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1700  INTFLOAT *cf = cfo;
1701  int len = off_len;
1702 
1703  do {
1704  int code;
1705  unsigned nnz;
1706  unsigned cb_idx;
1707  unsigned sign;
1708 
1709  UPDATE_CACHE(re, gb);
1710  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1711  cb_idx = cb_vector_idx[code];
1712  nnz = cb_idx >> 8 & 15;
1713  sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
1714  LAST_SKIP_BITS(re, gb, nnz);
1715 #if USE_FIXED
1716  cf = DEC_UPAIR(cf, cb_idx, sign);
1717 #else
1718  cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1719 #endif /* USE_FIXED */
1720  } while (len -= 2);
1721  }
1722  break;
1723 
1724  default:
1725  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1726 #if USE_FIXED
1727  int *icf = cfo;
1728  int v;
1729 #else
1730  float *cf = cfo;
1731  uint32_t *icf = (uint32_t *) cf;
1732 #endif /* USE_FIXED */
1733  int len = off_len;
1734 
1735  do {
1736  int code;
1737  unsigned nzt, nnz;
1738  unsigned cb_idx;
1739  uint32_t bits;
1740  int j;
1741 
1742  UPDATE_CACHE(re, gb);
1743  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1744 
1745  if (!code) {
1746  *icf++ = 0;
1747  *icf++ = 0;
1748  continue;
1749  }
1750 
1751  cb_idx = cb_vector_idx[code];
1752  nnz = cb_idx >> 12;
1753  nzt = cb_idx >> 8;
1754  bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1755  LAST_SKIP_BITS(re, gb, nnz);
1756 
1757  for (j = 0; j < 2; j++) {
1758  if (nzt & 1<<j) {
1759  uint32_t b;
1760  int n;
1761  /* The total length of escape_sequence must be < 22 bits according
1762  to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1763  UPDATE_CACHE(re, gb);
1764  b = GET_CACHE(re, gb);
1765  b = 31 - av_log2(~b);
1766 
1767  if (b > 8) {
1768  av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1769  return AVERROR_INVALIDDATA;
1770  }
1771 
1772  SKIP_BITS(re, gb, b + 1);
1773  b += 4;
1774  n = (1 << b) + SHOW_UBITS(re, gb, b);
1775  LAST_SKIP_BITS(re, gb, b);
1776 #if USE_FIXED
1777  v = n;
1778  if (bits & 1U<<31)
1779  v = -v;
1780  *icf++ = v;
1781 #else
1782  *icf++ = cbrt_tab[n] | (bits & 1U<<31);
1783 #endif /* USE_FIXED */
1784  bits <<= 1;
1785  } else {
1786 #if USE_FIXED
1787  v = cb_idx & 15;
1788  if (bits & 1U<<31)
1789  v = -v;
1790  *icf++ = v;
1791 #else
1792  unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1793  *icf++ = (bits & 1U<<31) | v;
1794 #endif /* USE_FIXED */
1795  bits <<= !!v;
1796  }
1797  cb_idx >>= 4;
1798  }
1799  } while (len -= 2);
1800 #if !USE_FIXED
1801  ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1802 #endif /* !USE_FIXED */
1803  }
1804  }
1805 
1806  CLOSE_READER(re, gb);
1807  }
1808  }
1809  coef += g_len << 7;
1810  }
1811 
1812  if (pulse_present) {
1813  idx = 0;
1814  for (i = 0; i < pulse->num_pulse; i++) {
1815  INTFLOAT co = coef_base[ pulse->pos[i] ];
1816  while (offsets[idx + 1] <= pulse->pos[i])
1817  idx++;
1818  if (band_type[idx] != NOISE_BT && sf[idx]) {
1819  INTFLOAT ico = -pulse->amp[i];
1820 #if USE_FIXED
1821  if (co) {
1822  ico = co + (co > 0 ? -ico : ico);
1823  }
1824  coef_base[ pulse->pos[i] ] = ico;
1825 #else
1826  if (co) {
1827  co /= sf[idx];
1828  ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1829  }
1830  coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1831 #endif /* USE_FIXED */
1832  }
1833  }
1834  }
1835 #if USE_FIXED
1836  coef = coef_base;
1837  idx = 0;
1838  for (g = 0; g < ics->num_window_groups; g++) {
1839  unsigned g_len = ics->group_len[g];
1840 
1841  for (i = 0; i < ics->max_sfb; i++, idx++) {
1842  const unsigned cbt_m1 = band_type[idx] - 1;
1843  int *cfo = coef + offsets[i];
1844  int off_len = offsets[i + 1] - offsets[i];
1845  int group;
1846 
1847  if (cbt_m1 < NOISE_BT - 1) {
1848  for (group = 0; group < (int)g_len; group++, cfo+=128) {
1849  ac->vector_pow43(cfo, off_len);
1850  ac->subband_scale(cfo, cfo, sf[idx], 34, off_len);
1851  }
1852  }
1853  }
1854  coef += g_len << 7;
1855  }
1856 #endif /* USE_FIXED */
1857  return 0;
1858 }
1859 
1860 /**
1861  * Apply AAC-Main style frequency domain prediction.
1862  */
1863 static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1864 {
1865  int sfb, k;
1866 
1867  if (!sce->ics.predictor_initialized) {
1868  reset_all_predictors(sce->predictor_state);
1869  sce->ics.predictor_initialized = 1;
1870  }
1871 
1872  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1873  for (sfb = 0;
1874  sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
1875  sfb++) {
1876  for (k = sce->ics.swb_offset[sfb];
1877  k < sce->ics.swb_offset[sfb + 1];
1878  k++) {
1879  predict(&sce->predictor_state[k], &sce->coeffs[k],
1880  sce->ics.predictor_present &&
1881  sce->ics.prediction_used[sfb]);
1882  }
1883  }
1884  if (sce->ics.predictor_reset_group)
1885  reset_predictor_group(sce->predictor_state,
1886  sce->ics.predictor_reset_group);
1887  } else
1888  reset_all_predictors(sce->predictor_state);
1889 }
1890 
1891 /**
1892  * Decode an individual_channel_stream payload; reference: table 4.44.
1893  *
1894  * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
1895  * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1896  *
1897  * @return Returns error status. 0 - OK, !0 - error
1898  */
1899 static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1900  GetBitContext *gb, int common_window, int scale_flag)
1901 {
1902  Pulse pulse;
1903  TemporalNoiseShaping *tns = &sce->tns;
1904  IndividualChannelStream *ics = &sce->ics;
1905  INTFLOAT *out = sce->coeffs;
1906  int global_gain, eld_syntax, er_syntax, pulse_present = 0;
1907  int ret;
1908 
1909  eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1910  er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
1911  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
1912  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
1913  ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1914 
1915  /* This assignment is to silence a GCC warning about the variable being used
1916  * uninitialized when in fact it always is.
1917  */
1918  pulse.num_pulse = 0;
1919 
1920  global_gain = get_bits(gb, 8);
1921 
1922  if (!common_window && !scale_flag) {
1923  if (decode_ics_info(ac, ics, gb) < 0)
1924  return AVERROR_INVALIDDATA;
1925  }
1926 
1927  if ((ret = decode_band_types(ac, sce->band_type,
1928  sce->band_type_run_end, gb, ics)) < 0)
1929  return ret;
1930  if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
1931  sce->band_type, sce->band_type_run_end)) < 0)
1932  return ret;
1933 
1934  pulse_present = 0;
1935  if (!scale_flag) {
1936  if (!eld_syntax && (pulse_present = get_bits1(gb))) {
1937  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1938  av_log(ac->avctx, AV_LOG_ERROR,
1939  "Pulse tool not allowed in eight short sequence.\n");
1940  return AVERROR_INVALIDDATA;
1941  }
1942  if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1943  av_log(ac->avctx, AV_LOG_ERROR,
1944  "Pulse data corrupt or invalid.\n");
1945  return AVERROR_INVALIDDATA;
1946  }
1947  }
1948  tns->present = get_bits1(gb);
1949  if (tns->present && !er_syntax)
1950  if (decode_tns(ac, tns, gb, ics) < 0)
1951  return AVERROR_INVALIDDATA;
1952  if (!eld_syntax && get_bits1(gb)) {
1953  avpriv_request_sample(ac->avctx, "SSR");
1954  return AVERROR_PATCHWELCOME;
1955  }
1956  // I see no textual basis in the spec for this occurring after SSR gain
1957  // control, but this is what both reference and real implmentations do
1958  if (tns->present && er_syntax)
1959  if (decode_tns(ac, tns, gb, ics) < 0)
1960  return AVERROR_INVALIDDATA;
1961  }
1962 
1963  if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
1964  &pulse, ics, sce->band_type) < 0)
1965  return AVERROR_INVALIDDATA;
1966 
1967  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
1968  apply_prediction(ac, sce);
1969 
1970  return 0;
1971 }
1972 
1973 /**
1974  * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1975  */
1976 static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1977 {
1978  const IndividualChannelStream *ics = &cpe->ch[0].ics;
1979  INTFLOAT *ch0 = cpe->ch[0].coeffs;
1980  INTFLOAT *ch1 = cpe->ch[1].coeffs;
1981  int g, i, group, idx = 0;
1982  const uint16_t *offsets = ics->swb_offset;
1983  for (g = 0; g < ics->num_window_groups; g++) {
1984  for (i = 0; i < ics->max_sfb; i++, idx++) {
1985  if (cpe->ms_mask[idx] &&
1986  cpe->ch[0].band_type[idx] < NOISE_BT &&
1987  cpe->ch[1].band_type[idx] < NOISE_BT) {
1988 #if USE_FIXED
1989  for (group = 0; group < ics->group_len[g]; group++) {
1990  ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
1991  ch1 + group * 128 + offsets[i],
1992  offsets[i+1] - offsets[i]);
1993 #else
1994  for (group = 0; group < ics->group_len[g]; group++) {
1995  ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
1996  ch1 + group * 128 + offsets[i],
1997  offsets[i+1] - offsets[i]);
1998 #endif /* USE_FIXED */
1999  }
2000  }
2001  }
2002  ch0 += ics->group_len[g] * 128;
2003  ch1 += ics->group_len[g] * 128;
2004  }
2005 }
2006 
2007 /**
2008  * intensity stereo decoding; reference: 4.6.8.2.3
2009  *
2010  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
2011  * [1] mask is decoded from bitstream; [2] mask is all 1s;
2012  * [3] reserved for scalable AAC
2013  */
2014 static void apply_intensity_stereo(AACContext *ac,
2015  ChannelElement *cpe, int ms_present)
2016 {
2017  const IndividualChannelStream *ics = &cpe->ch[1].ics;
2018  SingleChannelElement *sce1 = &cpe->ch[1];
2019  INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
2020  const uint16_t *offsets = ics->swb_offset;
2021  int g, group, i, idx = 0;
2022  int c;
2023  INTFLOAT scale;
2024  for (g = 0; g < ics->num_window_groups; g++) {
2025  for (i = 0; i < ics->max_sfb;) {
2026  if (sce1->band_type[idx] == INTENSITY_BT ||
2027  sce1->band_type[idx] == INTENSITY_BT2) {
2028  const int bt_run_end = sce1->band_type_run_end[idx];
2029  for (; i < bt_run_end; i++, idx++) {
2030  c = -1 + 2 * (sce1->band_type[idx] - 14);
2031  if (ms_present)
2032  c *= 1 - 2 * cpe->ms_mask[idx];
2033  scale = c * sce1->sf[idx];
2034  for (group = 0; group < ics->group_len[g]; group++)
2035 #if USE_FIXED
2036  ac->subband_scale(coef1 + group * 128 + offsets[i],
2037  coef0 + group * 128 + offsets[i],
2038  scale,
2039  23,
2040  offsets[i + 1] - offsets[i]);
2041 #else
2042  ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
2043  coef0 + group * 128 + offsets[i],
2044  scale,
2045  offsets[i + 1] - offsets[i]);
2046 #endif /* USE_FIXED */
2047  }
2048  } else {
2049  int bt_run_end = sce1->band_type_run_end[idx];
2050  idx += bt_run_end - i;
2051  i = bt_run_end;
2052  }
2053  }
2054  coef0 += ics->group_len[g] * 128;
2055  coef1 += ics->group_len[g] * 128;
2056  }
2057 }
2058 
2059 /**
2060  * Decode a channel_pair_element; reference: table 4.4.
2061  *
2062  * @return Returns error status. 0 - OK, !0 - error
2063  */
2064 static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
2065 {
2066  int i, ret, common_window, ms_present = 0;
2067  int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2068 
2069  common_window = eld_syntax || get_bits1(gb);
2070  if (common_window) {
2071  if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
2072  return AVERROR_INVALIDDATA;
2073  i = cpe->ch[1].ics.use_kb_window[0];
2074  cpe->ch[1].ics = cpe->ch[0].ics;
2075  cpe->ch[1].ics.use_kb_window[1] = i;
2076  if (cpe->ch[1].ics.predictor_present &&
2077  (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
2078  if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
2079  decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
2080  ms_present = get_bits(gb, 2);
2081  if (ms_present == 3) {
2082  av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
2083  return AVERROR_INVALIDDATA;
2084  } else if (ms_present)
2085  decode_mid_side_stereo(cpe, gb, ms_present);
2086  }
2087  if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
2088  return ret;
2089  if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
2090  return ret;
2091 
2092  if (common_window) {
2093  if (ms_present)
2094  apply_mid_side_stereo(ac, cpe);
2095  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
2096  apply_prediction(ac, &cpe->ch[0]);
2097  apply_prediction(ac, &cpe->ch[1]);
2098  }
2099  }
2100 
2101  apply_intensity_stereo(ac, cpe, ms_present);
2102  return 0;
2103 }
2104 
2105 static const float cce_scale[] = {
2106  1.09050773266525765921, //2^(1/8)
2107  1.18920711500272106672, //2^(1/4)
2108  M_SQRT2,
2109  2,
2110 };
2111 
2112 /**
2113  * Decode coupling_channel_element; reference: table 4.8.
2114  *
2115  * @return Returns error status. 0 - OK, !0 - error
2116  */
2117 static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
2118 {
2119  int num_gain = 0;
2120  int c, g, sfb, ret;
2121  int sign;
2122  INTFLOAT scale;
2123  SingleChannelElement *sce = &che->ch[0];
2124  ChannelCoupling *coup = &che->coup;
2125 
2126  coup->coupling_point = 2 * get_bits1(gb);
2127  coup->num_coupled = get_bits(gb, 3);
2128  for (c = 0; c <= coup->num_coupled; c++) {
2129  num_gain++;
2130  coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
2131  coup->id_select[c] = get_bits(gb, 4);
2132  if (coup->type[c] == TYPE_CPE) {
2133  coup->ch_select[c] = get_bits(gb, 2);
2134  if (coup->ch_select[c] == 3)
2135  num_gain++;
2136  } else
2137  coup->ch_select[c] = 2;
2138  }
2139  coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
2140 
2141  sign = get_bits(gb, 1);
2142  scale = AAC_RENAME(cce_scale)[get_bits(gb, 2)];
2143 
2144  if ((ret = decode_ics(ac, sce, gb, 0, 0)))
2145  return ret;
2146 
2147  for (c = 0; c < num_gain; c++) {
2148  int idx = 0;
2149  int cge = 1;
2150  int gain = 0;
2151  INTFLOAT gain_cache = FIXR10(1.);
2152  if (c) {
2153  cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
2154  gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
2155  gain_cache = GET_GAIN(scale, gain);
2156  }
2157  if (coup->coupling_point == AFTER_IMDCT) {
2158  coup->gain[c][0] = gain_cache;
2159  } else {
2160  for (g = 0; g < sce->ics.num_window_groups; g++) {
2161  for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
2162  if (sce->band_type[idx] != ZERO_BT) {
2163  if (!cge) {
2164  int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
2165  if (t) {
2166  int s = 1;
2167  t = gain += t;
2168  if (sign) {
2169  s -= 2 * (t & 0x1);
2170  t >>= 1;
2171  }
2172  gain_cache = GET_GAIN(scale, t) * s;
2173  }
2174  }
2175  coup->gain[c][idx] = gain_cache;
2176  }
2177  }
2178  }
2179  }
2180  }
2181  return 0;
2182 }
2183 
2184 /**
2185  * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
2186  *
2187  * @return Returns number of bytes consumed.
2188  */
2189 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
2190  GetBitContext *gb)
2191 {
2192  int i;
2193  int num_excl_chan = 0;
2194 
2195  do {
2196  for (i = 0; i < 7; i++)
2197  che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
2198  } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
2199 
2200  return num_excl_chan / 7;
2201 }
2202 
2203 /**
2204  * Decode dynamic range information; reference: table 4.52.
2205  *
2206  * @return Returns number of bytes consumed.
2207  */
2208 static int decode_dynamic_range(DynamicRangeControl *che_drc,
2209  GetBitContext *gb)
2210 {
2211  int n = 1;
2212  int drc_num_bands = 1;
2213  int i;
2214 
2215  /* pce_tag_present? */
2216  if (get_bits1(gb)) {
2217  che_drc->pce_instance_tag = get_bits(gb, 4);
2218  skip_bits(gb, 4); // tag_reserved_bits
2219  n++;
2220  }
2221 
2222  /* excluded_chns_present? */
2223  if (get_bits1(gb)) {
2224  n += decode_drc_channel_exclusions(che_drc, gb);
2225  }
2226 
2227  /* drc_bands_present? */
2228  if (get_bits1(gb)) {
2229  che_drc->band_incr = get_bits(gb, 4);
2230  che_drc->interpolation_scheme = get_bits(gb, 4);
2231  n++;
2232  drc_num_bands += che_drc->band_incr;
2233  for (i = 0; i < drc_num_bands; i++) {
2234  che_drc->band_top[i] = get_bits(gb, 8);
2235  n++;
2236  }
2237  }
2238 
2239  /* prog_ref_level_present? */
2240  if (get_bits1(gb)) {
2241  che_drc->prog_ref_level = get_bits(gb, 7);
2242  skip_bits1(gb); // prog_ref_level_reserved_bits
2243  n++;
2244  }
2245 
2246  for (i = 0; i < drc_num_bands; i++) {
2247  che_drc->dyn_rng_sgn[i] = get_bits1(gb);
2248  che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
2249  n++;
2250  }
2251 
2252  return n;
2253 }
2254 
2255 static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
2256  uint8_t buf[256];
2257  int i, major, minor;
2258 
2259  if (len < 13+7*8)
2260  goto unknown;
2261 
2262  get_bits(gb, 13); len -= 13;
2263 
2264  for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
2265  buf[i] = get_bits(gb, 8);
2266 
2267  buf[i] = 0;
2268  if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
2269  av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
2270 
2271  if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
2272  ac->avctx->internal->skip_samples = 1024;
2273  }
2274 
2275 unknown:
2276  skip_bits_long(gb, len);
2277 
2278  return 0;
2279 }
2280 
2281 /**
2282  * Decode extension data (incomplete); reference: table 4.51.
2283  *
2284  * @param cnt length of TYPE_FIL syntactic element in bytes
2285  *
2286  * @return Returns number of bytes consumed
2287  */
2288 static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
2289  ChannelElement *che, enum RawDataBlockType elem_type)
2290 {
2291  int crc_flag = 0;
2292  int res = cnt;
2293  int type = get_bits(gb, 4);
2294 
2295  if (ac->avctx->debug & FF_DEBUG_STARTCODE)
2296  av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
2297 
2298  switch (type) { // extension type
2299  case EXT_SBR_DATA_CRC:
2300  crc_flag++;
2301  case EXT_SBR_DATA:
2302  if (!che) {
2303  av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
2304  return res;
2305  } else if (!ac->oc[1].m4ac.sbr) {
2306  av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
2307  skip_bits_long(gb, 8 * cnt - 4);
2308  return res;
2309  } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
2310  av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
2311  skip_bits_long(gb, 8 * cnt - 4);
2312  return res;
2313  } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
2314  ac->oc[1].m4ac.sbr = 1;
2315  ac->oc[1].m4ac.ps = 1;
2316  ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
2317  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
2318  ac->oc[1].status, 1);
2319  } else {
2320  ac->oc[1].m4ac.sbr = 1;
2321  ac->avctx->profile = FF_PROFILE_AAC_HE;
2322  }
2323  res = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
2324  break;
2325  case EXT_DYNAMIC_RANGE:
2326  res = decode_dynamic_range(&ac->che_drc, gb);
2327  break;
2328  case EXT_FILL:
2329  decode_fill(ac, gb, 8 * cnt - 4);
2330  break;
2331  case EXT_FILL_DATA:
2332  case EXT_DATA_ELEMENT:
2333  default:
2334  skip_bits_long(gb, 8 * cnt - 4);
2335  break;
2336  };
2337  return res;
2338 }
2339 
2340 /**
2341  * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
2342  *
2343  * @param decode 1 if tool is used normally, 0 if tool is used in LTP.
2344  * @param coef spectral coefficients
2345  */
2346 static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
2347  IndividualChannelStream *ics, int decode)
2348 {
2349  const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
2350  int w, filt, m, i;
2351  int bottom, top, order, start, end, size, inc;
2352  INTFLOAT lpc[TNS_MAX_ORDER];
2353  INTFLOAT tmp[TNS_MAX_ORDER+1];
2354 
2355  for (w = 0; w < ics->num_windows; w++) {
2356  bottom = ics->num_swb;
2357  for (filt = 0; filt < tns->n_filt[w]; filt++) {
2358  top = bottom;
2359  bottom = FFMAX(0, top - tns->length[w][filt]);
2360  order = tns->order[w][filt];
2361  if (order == 0)
2362  continue;
2363 
2364  // tns_decode_coef
2365  AAC_RENAME(compute_lpc_coefs)(tns->coef[w][filt], order, lpc, 0, 0, 0);
2366 
2367  start = ics->swb_offset[FFMIN(bottom, mmm)];
2368  end = ics->swb_offset[FFMIN( top, mmm)];
2369  if ((size = end - start) <= 0)
2370  continue;
2371  if (tns->direction[w][filt]) {
2372  inc = -1;
2373  start = end - 1;
2374  } else {
2375  inc = 1;
2376  }
2377  start += w * 128;
2378 
2379  if (decode) {
2380  // ar filter
2381  for (m = 0; m < size; m++, start += inc)
2382  for (i = 1; i <= FFMIN(m, order); i++)
2383  coef[start] -= AAC_MUL26(coef[start - i * inc], lpc[i - 1]);
2384  } else {
2385  // ma filter
2386  for (m = 0; m < size; m++, start += inc) {
2387  tmp[0] = coef[start];
2388  for (i = 1; i <= FFMIN(m, order); i++)
2389  coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
2390  for (i = order; i > 0; i--)
2391  tmp[i] = tmp[i - 1];
2392  }
2393  }
2394  }
2395  }
2396 }
2397 
2398 /**
2399  * Apply windowing and MDCT to obtain the spectral
2400  * coefficient from the predicted sample by LTP.
2401  */
2402 static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out,
2403  INTFLOAT *in, IndividualChannelStream *ics)
2404 {
2405  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2406  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2407  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2408  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2409 
2410  if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
2411  ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
2412  } else {
2413  memset(in, 0, 448 * sizeof(*in));
2414  ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
2415  }
2416  if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
2417  ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
2418  } else {
2419  ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
2420  memset(in + 1024 + 576, 0, 448 * sizeof(*in));
2421  }
2422  ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
2423 }
2424 
2425 /**
2426  * Apply the long term prediction
2427  */
2428 static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
2429 {
2430  const LongTermPrediction *ltp = &sce->ics.ltp;
2431  const uint16_t *offsets = sce->ics.swb_offset;
2432  int i, sfb;
2433 
2434  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
2435  INTFLOAT *predTime = sce->ret;
2436  INTFLOAT *predFreq = ac->buf_mdct;
2437  int16_t num_samples = 2048;
2438 
2439  if (ltp->lag < 1024)
2440  num_samples = ltp->lag + 1024;
2441  for (i = 0; i < num_samples; i++)
2442  predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
2443  memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));
2444 
2445  ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
2446 
2447  if (sce->tns.present)
2448  ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
2449 
2450  for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
2451  if (ltp->used[sfb])
2452  for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
2453  sce->coeffs[i] += predFreq[i];
2454  }
2455 }
2456 
2457 /**
2458  * Update the LTP buffer for next frame
2459  */
2460 static void update_ltp(AACContext *ac, SingleChannelElement *sce)
2461 {
2462  IndividualChannelStream *ics = &sce->ics;
2463  INTFLOAT *saved = sce->saved;
2464  INTFLOAT *saved_ltp = sce->coeffs;
2465  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2466  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2467  int i;
2468 
2469  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2470  memcpy(saved_ltp, saved, 512 * sizeof(*saved_ltp));
2471  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2472  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2473 
2474  for (i = 0; i < 64; i++)
2475  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2476  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2477  memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(*saved_ltp));
2478  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2479  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2480 
2481  for (i = 0; i < 64; i++)
2482  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2483  } else { // LONG_STOP or ONLY_LONG
2484  ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
2485 
2486  for (i = 0; i < 512; i++)
2487  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
2488  }
2489 
2490  memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
2491  memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
2492  memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
2493 }
2494 
2495 /**
2496  * Conduct IMDCT and windowing.
2497  */
2498 static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
2499 {
2500  IndividualChannelStream *ics = &sce->ics;
2501  INTFLOAT *in = sce->coeffs;
2502  INTFLOAT *out = sce->ret;
2503  INTFLOAT *saved = sce->saved;
2504  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2505  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2506  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2507  INTFLOAT *buf = ac->buf_mdct;
2508  INTFLOAT *temp = ac->temp;
2509  int i;
2510 
2511  // imdct
2512  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2513  for (i = 0; i < 1024; i += 128)
2514  ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
2515  } else {
2516  ac->mdct.imdct_half(&ac->mdct, buf, in);
2517 #if USE_FIXED
2518  for (i=0; i<1024; i++)
2519  buf[i] = (buf[i] + 4) >> 3;
2520 #endif /* USE_FIXED */
2521  }
2522 
2523  /* window overlapping
2524  * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2525  * and long to short transitions are considered to be short to short
2526  * transitions. This leaves just two cases (long to long and short to short)
2527  * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2528  */
2529  if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2530  (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
2531  ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
2532  } else {
2533  memcpy( out, saved, 448 * sizeof(*out));
2534 
2535  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2536  ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
2537  ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
2538  ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
2539  ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
2540  ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
2541  memcpy( out + 448 + 4*128, temp, 64 * sizeof(*out));
2542  } else {
2543  ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
2544  memcpy( out + 576, buf + 64, 448 * sizeof(*out));
2545  }
2546  }
2547 
2548  // buffer update
2549  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2550  memcpy( saved, temp + 64, 64 * sizeof(*saved));
2551  ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
2552  ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
2553  ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
2554  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2555  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2556  memcpy( saved, buf + 512, 448 * sizeof(*saved));
2557  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2558  } else { // LONG_STOP or ONLY_LONG
2559  memcpy( saved, buf + 512, 512 * sizeof(*saved));
2560  }
2561 }
2562 
2563 static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
2564 {
2565  IndividualChannelStream *ics = &sce->ics;
2566  INTFLOAT *in = sce->coeffs;
2567  INTFLOAT *out = sce->ret;
2568  INTFLOAT *saved = sce->saved;
2569  INTFLOAT *buf = ac->buf_mdct;
2570 #if USE_FIXED
2571  int i;
2572 #endif /* USE_FIXED */
2573 
2574  // imdct
2575  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2576 
2577 #if USE_FIXED
2578  for (i = 0; i < 1024; i++)
2579  buf[i] = (buf[i] + 2) >> 2;
2580 #endif /* USE_FIXED */
2581 
2582  // window overlapping
2583  if (ics->use_kb_window[1]) {
2584  // AAC LD uses a low overlap sine window instead of a KBD window
2585  memcpy(out, saved, 192 * sizeof(*out));
2586  ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
2587  memcpy( out + 320, buf + 64, 192 * sizeof(*out));
2588  } else {
2589  ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);
2590  }
2591 
2592  // buffer update
2593  memcpy(saved, buf + 256, 256 * sizeof(*saved));
2594 }
2595 
2596 static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
2597 {
2598  INTFLOAT *in = sce->coeffs;
2599  INTFLOAT *out = sce->ret;
2600  INTFLOAT *saved = sce->saved;
2601  INTFLOAT *buf = ac->buf_mdct;
2602  int i;
2603  const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
2604  const int n2 = n >> 1;
2605  const int n4 = n >> 2;
2606  const INTFLOAT *const window = n == 480 ? AAC_RENAME(ff_aac_eld_window_480) :
2607  AAC_RENAME(ff_aac_eld_window_512);
2608 
2609  // Inverse transform, mapped to the conventional IMDCT by
2610  // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
2611  // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
2612  // International Conference on Audio, Language and Image Processing, ICALIP 2008.
2613  // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
2614  for (i = 0; i < n2; i+=2) {
2615  INTFLOAT temp;
2616  temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
2617  temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
2618  }
2619 #if !USE_FIXED
2620  if (n == 480)
2621  ac->mdct480->imdct_half(ac->mdct480, buf, in, 1, -1.f/(16*1024*960));
2622  else
2623 #endif
2624  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2625 
2626 #if USE_FIXED
2627  for (i = 0; i < 1024; i++)
2628  buf[i] = (buf[i] + 1) >> 1;
2629 #endif /* USE_FIXED */
2630 
2631  for (i = 0; i < n; i+=2) {
2632  buf[i] = -buf[i];
2633  }
2634  // Like with the regular IMDCT at this point we still have the middle half
2635  // of a transform but with even symmetry on the left and odd symmetry on
2636  // the right
2637 
2638  // window overlapping
2639  // The spec says to use samples [0..511] but the reference decoder uses
2640  // samples [128..639].
2641  for (i = n4; i < n2; i ++) {
2642  out[i - n4] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) +
2643  AAC_MUL31( saved[ i + n2] , window[i + n - n4]) +
2644  AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
2645  AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]);
2646  }
2647  for (i = 0; i < n2; i ++) {
2648  out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) +
2649  AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) +
2650  AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) +
2651  AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
2652  }
2653  for (i = 0; i < n4; i ++) {
2654  out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) +
2655  AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
2656  AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
2657  }
2658 
2659  // buffer update
2660  memmove(saved + n, saved, 2 * n * sizeof(*saved));
2661  memcpy( saved, buf, n * sizeof(*saved));
2662 }
2663 
2664 /**
2665  * channel coupling transformation interface
2666  *
2667  * @param apply_coupling_method pointer to (in)dependent coupling function
2668  */
2669 static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
2670  enum RawDataBlockType type, int elem_id,
2671  enum CouplingPoint coupling_point,
2672  void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
2673 {
2674  int i, c;
2675 
2676  for (i = 0; i < MAX_ELEM_ID; i++) {
2677  ChannelElement *cce = ac->che[TYPE_CCE][i];
2678  int index = 0;
2679 
2680  if (cce && cce->coup.coupling_point == coupling_point) {
2681  ChannelCoupling *coup = &cce->coup;
2682 
2683  for (c = 0; c <= coup->num_coupled; c++) {
2684  if (coup->type[c] == type && coup->id_select[c] == elem_id) {
2685  if (coup->ch_select[c] != 1) {
2686  apply_coupling_method(ac, &cc->ch[0], cce, index);
2687  if (coup->ch_select[c] != 0)
2688  index++;
2689  }
2690  if (coup->ch_select[c] != 2)
2691  apply_coupling_method(ac, &cc->ch[1], cce, index++);
2692  } else
2693  index += 1 + (coup->ch_select[c] == 3);
2694  }
2695  }
2696  }
2697 }
2698 
2699 /**
2700  * Convert spectral data to samples, applying all supported tools as appropriate.
2701  */
2702 static void spectral_to_sample(AACContext *ac, int samples)
2703 {
2704  int i, type;
2705  void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
2706  switch (ac->oc[1].m4ac.object_type) {
2707  case AOT_ER_AAC_LD:
2708  imdct_and_window = imdct_and_windowing_ld;
2709  break;
2710  case AOT_ER_AAC_ELD:
2711  imdct_and_window = imdct_and_windowing_eld;
2712  break;
2713  default:
2714  imdct_and_window = ac->imdct_and_windowing;
2715  }
2716  for (type = 3; type >= 0; type--) {
2717  for (i = 0; i < MAX_ELEM_ID; i++) {
2718  ChannelElement *che = ac->che[type][i];
2719  if (che && che->present) {
2720  if (type <= TYPE_CPE)
2721  apply_channel_coupling(ac, che, type, i, BEFORE_TNS, AAC_RENAME(apply_dependent_coupling));
2722  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2723  if (che->ch[0].ics.predictor_present) {
2724  if (che->ch[0].ics.ltp.present)
2725  ac->apply_ltp(ac, &che->ch[0]);
2726  if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
2727  ac->apply_ltp(ac, &che->ch[1]);
2728  }
2729  }
2730  if (che->ch[0].tns.present)
2731  ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2732  if (che->ch[1].tns.present)
2733  ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2734  if (type <= TYPE_CPE)
2735  apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, AAC_RENAME(apply_dependent_coupling));
2736  if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2737  imdct_and_window(ac, &che->ch[0]);
2738  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2739  ac->update_ltp(ac, &che->ch[0]);
2740  if (type == TYPE_CPE) {
2741  imdct_and_window(ac, &che->ch[1]);
2742  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2743  ac->update_ltp(ac, &che->ch[1]);
2744  }
2745  if (ac->oc[1].m4ac.sbr > 0) {
2746  AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2747  }
2748  }
2749  if (type <= TYPE_CCE)
2750  apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, AAC_RENAME(apply_independent_coupling));
2751 
2752 #if USE_FIXED
2753  {
2754  int j;
2755  /* preparation for resampler */
2756  for(j = 0; j<samples; j++){
2757  che->ch[0].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[0].ret[j]<<7)+0x8000;
2758  if(type == TYPE_CPE)
2759  che->ch[1].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[1].ret[j]<<7)+0x8000;
2760  }
2761  }
2762 #endif /* USE_FIXED */
2763  che->present = 0;
2764  } else if (che) {
2765  av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
2766  }
2767  }
2768  }
2769 }
2770 
2771 static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
2772 {
2773  int size;
2774  AACADTSHeaderInfo hdr_info;
2775  uint8_t layout_map[MAX_ELEM_ID*4][3];
2776  int layout_map_tags, ret;
2777 
2778  size = avpriv_aac_parse_header(gb, &hdr_info);
2779  if (size > 0) {
2780  if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
2781  // This is 2 for "VLB " audio in NSV files.
2782  // See samples/nsv/vlb_audio.
2783  avpriv_report_missing_feature(ac->avctx,
2784  "More than one AAC RDB per ADTS frame");
2785  ac->warned_num_aac_frames = 1;
2786  }
2787  push_output_configuration(ac);
2788  if (hdr_info.chan_config) {
2789  ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
2790  if ((ret = set_default_channel_config(ac->avctx,
2791  layout_map,
2792  &layout_map_tags,
2793  hdr_info.chan_config)) < 0)
2794  return ret;
2795  if ((ret = output_configure(ac, layout_map, layout_map_tags,
2796  FFMAX(ac->oc[1].status,
2797  OC_TRIAL_FRAME), 0)) < 0)
2798  return ret;
2799  } else {
2800  ac->oc[1].m4ac.chan_config = 0;
2801  /**
2802  * dual mono frames in Japanese DTV can have chan_config 0
2803  * WITHOUT specifying PCE.
2804  * thus, set dual mono as default.
2805  */
2806  if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
2807  layout_map_tags = 2;
2808  layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
2809  layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
2810  layout_map[0][1] = 0;
2811  layout_map[1][1] = 1;
2812  if (output_configure(ac, layout_map, layout_map_tags,
2813  OC_TRIAL_FRAME, 0))
2814  return -7;
2815  }
2816  }
2817  ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
2818  ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
2819  ac->oc[1].m4ac.object_type = hdr_info.object_type;
2820  ac->oc[1].m4ac.frame_length_short = 0;
2821  if (ac->oc[0].status != OC_LOCKED ||
2822  ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
2823  ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
2824  ac->oc[1].m4ac.sbr = -1;
2825  ac->oc[1].m4ac.ps = -1;
2826  }
2827  if (!hdr_info.crc_absent)
2828  skip_bits(gb, 16);
2829  }
2830  return size;
2831 }
2832 
2833 static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
2834  int *got_frame_ptr, GetBitContext *gb)
2835 {
2836  AACContext *ac = avctx->priv_data;
2837  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
2838  ChannelElement *che;
2839  int err, i;
2840  int samples = m4ac->frame_length_short ? 960 : 1024;
2841  int chan_config = m4ac->chan_config;
2842  int aot = m4ac->object_type;
2843 
2844  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
2845  samples >>= 1;
2846 
2847  ac->frame = data;
2848 
2849  if ((err = frame_configure_elements(avctx)) < 0)
2850  return err;
2851 
2852  // The FF_PROFILE_AAC_* defines are all object_type - 1
2853  // This may lead to an undefined profile being signaled
2854  ac->avctx->profile = aot - 1;
2855 
2856  ac->tags_mapped = 0;
2857 
2858  if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
2859  avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
2860  chan_config);
2861  return AVERROR_INVALIDDATA;
2862  }
2863  for (i = 0; i < tags_per_config[chan_config]; i++) {
2864  const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
2865  const int elem_id = aac_channel_layout_map[chan_config-1][i][1];
2866  if (!(che=get_che(ac, elem_type, elem_id))) {
2867  av_log(ac->avctx, AV_LOG_ERROR,
2868  "channel element %d.%d is not allocated\n",
2869  elem_type, elem_id);
2870  return AVERROR_INVALIDDATA;
2871  }
2872  che->present = 1;
2873  if (aot != AOT_ER_AAC_ELD)
2874  skip_bits(gb, 4);
2875  switch (elem_type) {
2876  case TYPE_SCE:
2877  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2878  break;
2879  case TYPE_CPE:
2880  err = decode_cpe(ac, gb, che);
2881  break;
2882  case TYPE_LFE:
2883  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2884  break;
2885  }
2886  if (err < 0)
2887  return err;
2888  }
2889 
2890  spectral_to_sample(ac, samples);
2891 
2892  ac->frame->nb_samples = samples;
2893  ac->frame->sample_rate = avctx->sample_rate;
2894  *got_frame_ptr = 1;
2895 
2896  skip_bits_long(gb, get_bits_left(gb));
2897  return 0;
2898 }
2899 
2900 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
2901  int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
2902 {
2903  AACContext *ac = avctx->priv_data;
2904  ChannelElement *che = NULL, *che_prev = NULL;
2905  enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
2906  int err, elem_id;
2907  int samples = 0, multiplier, audio_found = 0, pce_found = 0;
2908  int is_dmono, sce_count = 0;
2909 
2910  ac->frame = data;
2911 
2912  if (show_bits(gb, 12) == 0xfff) {
2913  if ((err = parse_adts_frame_header(ac, gb)) < 0) {
2914  av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
2915  goto fail;
2916  }
2917  if (ac->oc[1].m4ac.sampling_index > 12) {
2918  av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
2919  err = AVERROR_INVALIDDATA;
2920  goto fail;
2921  }
2922  }
2923 
2924  if ((err = frame_configure_elements(avctx)) < 0)
2925  goto fail;
2926 
2927  // The FF_PROFILE_AAC_* defines are all object_type - 1
2928  // This may lead to an undefined profile being signaled
2929  ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
2930 
2931  ac->tags_mapped = 0;
2932  // parse
2933  while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
2934  elem_id = get_bits(gb, 4);
2935 
2936  if (avctx->debug & FF_DEBUG_STARTCODE)
2937  av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
2938 
2939  if (!avctx->channels && elem_type != TYPE_PCE) {
2940  err = AVERROR_INVALIDDATA;
2941  goto fail;
2942  }
2943 
2944  if (elem_type < TYPE_DSE) {
2945  if (!(che=get_che(ac, elem_type, elem_id))) {
2946  av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
2947  elem_type, elem_id);
2948  err = AVERROR_INVALIDDATA;
2949  goto fail;
2950  }
2951  samples = 1024;
2952  che->present = 1;
2953  }
2954 
2955  switch (elem_type) {
2956 
2957  case TYPE_SCE:
2958  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2959  audio_found = 1;
2960  sce_count++;
2961  break;
2962 
2963  case TYPE_CPE:
2964  err = decode_cpe(ac, gb, che);
2965  audio_found = 1;
2966  break;
2967 
2968  case TYPE_CCE:
2969  err = decode_cce(ac, gb, che);
2970  break;
2971 
2972  case TYPE_LFE:
2973  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2974  audio_found = 1;
2975  break;
2976 
2977  case TYPE_DSE:
2978  err = skip_data_stream_element(ac, gb);
2979  break;
2980 
2981  case TYPE_PCE: {
2982  uint8_t layout_map[MAX_ELEM_ID*4][3];
2983  int tags;
2984  push_output_configuration(ac);
2985  tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);
2986  if (tags < 0) {
2987  err = tags;
2988  break;
2989  }
2990  if (pce_found) {
2991  av_log(avctx, AV_LOG_ERROR,
2992  "Not evaluating a further program_config_element as this construct is dubious at best.\n");
2993  } else {
2994  err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
2995  if (!err)
2996  ac->oc[1].m4ac.chan_config = 0;
2997  pce_found = 1;
2998  }
2999  break;
3000  }
3001 
3002  case TYPE_FIL:
3003  if (elem_id == 15)
3004  elem_id += get_bits(gb, 8) - 1;
3005  if (get_bits_left(gb) < 8 * elem_id) {
3006  av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
3007  err = AVERROR_INVALIDDATA;
3008  goto fail;
3009  }
3010  while (elem_id > 0)
3011  elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
3012  err = 0; /* FIXME */
3013  break;
3014 
3015  default:
3016  err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
3017  break;
3018  }
3019 
3020  che_prev = che;
3021  elem_type_prev = elem_type;
3022 
3023  if (err)
3024  goto fail;
3025 
3026  if (get_bits_left(gb) < 3) {
3027  av_log(avctx, AV_LOG_ERROR, overread_err);
3028  err = AVERROR_INVALIDDATA;
3029  goto fail;
3030  }
3031  }
3032 
3033  if (!avctx->channels) {
3034  *got_frame_ptr = 0;
3035  return 0;
3036  }
3037 
3038  multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
3039  samples <<= multiplier;
3040 
3041  spectral_to_sample(ac, samples);
3042 
3043  if (ac->oc[1].status && audio_found) {
3044  avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
3045  avctx->frame_size = samples;
3046  ac->oc[1].status = OC_LOCKED;
3047  }
3048 
3049  if (multiplier) {
3050  int side_size;
3051  const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
3052  if (side && side_size>=4)
3053  AV_WL32(side, 2*AV_RL32(side));
3054  }
3055 
3056  if (!ac->frame->data[0] && samples) {
3057  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
3058  err = AVERROR_INVALIDDATA;
3059  goto fail;
3060  }
3061 
3062  if (samples) {
3063  ac->frame->nb_samples = samples;
3064  ac->frame->sample_rate = avctx->sample_rate;
3065  } else
3066  av_frame_unref(ac->frame);
3067  *got_frame_ptr = !!samples;
3068 
3069  /* for dual-mono audio (SCE + SCE) */
3070  is_dmono = ac->dmono_mode && sce_count == 2 &&
3071  ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
3072  if (is_dmono) {
3073  if (ac->dmono_mode == 1)
3074  ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
3075  else if (ac->dmono_mode == 2)
3076  ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
3077  }
3078 
3079  return 0;
3080 fail:
3081  pop_output_configuration(ac);
3082  return err;
3083 }
3084 
3085 static int aac_decode_frame(AVCodecContext *avctx, void *data,
3086  int *got_frame_ptr, AVPacket *avpkt)
3087 {
3088  AACContext *ac = avctx->priv_data;
3089  const uint8_t *buf = avpkt->data;
3090  int buf_size = avpkt->size;
3091  GetBitContext gb;
3092  int buf_consumed;
3093  int buf_offset;
3094  int err;
3095  int new_extradata_size;
3096  const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
3097  AV_PKT_DATA_NEW_EXTRADATA,
3098  &new_extradata_size);
3099  int jp_dualmono_size;
3100  const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
3101  AV_PKT_DATA_JP_DUALMONO,
3102  &jp_dualmono_size);
3103 
3104  if (new_extradata && 0) {
3105  av_free(avctx->extradata);
3106  avctx->extradata = av_mallocz(new_extradata_size +
3107  AV_INPUT_BUFFER_PADDING_SIZE);
3108  if (!avctx->extradata)
3109  return AVERROR(ENOMEM);
3110  avctx->extradata_size = new_extradata_size;
3111  memcpy(avctx->extradata, new_extradata, new_extradata_size);
3112  push_output_configuration(ac);
3113  if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
3114  avctx->extradata,
3115  avctx->extradata_size*8LL, 1) < 0) {
3116  pop_output_configuration(ac);
3117  return AVERROR_INVALIDDATA;
3118  }
3119  }
3120 
3121  ac->dmono_mode = 0;
3122  if (jp_dualmono && jp_dualmono_size > 0)
3123  ac->dmono_mode = 1 + *jp_dualmono;
3124  if (ac->force_dmono_mode >= 0)
3125  ac->dmono_mode = ac->force_dmono_mode;
3126 
3127  if (INT_MAX / 8 <= buf_size)
3128  return AVERROR_INVALIDDATA;
3129 
3130  if ((err = init_get_bits8(&gb, buf, buf_size)) < 0)
3131  return err;
3132 
3133  switch (ac->oc[1].m4ac.object_type) {
3134  case AOT_ER_AAC_LC:
3135  case AOT_ER_AAC_LTP:
3136  case AOT_ER_AAC_LD:
3137  case AOT_ER_AAC_ELD:
3138  err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
3139  break;
3140  default:
3141  err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
3142  }
3143  if (err < 0)
3144  return err;
3145 
3146  buf_consumed = (get_bits_count(&gb) + 7) >> 3;
3147  for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
3148  if (buf[buf_offset])
3149  break;
3150 
3151  return buf_size > buf_offset ? buf_consumed : buf_size;
3152 }
3153 
3154 static av_cold int aac_decode_close(AVCodecContext *avctx)
3155 {
3156  AACContext *ac = avctx->priv_data;
3157  int i, type;
3158 
3159  for (i = 0; i < MAX_ELEM_ID; i++) {
3160  for (type = 0; type < 4; type++) {
3161  if (ac->che[type][i])
3162  AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][i]->sbr);
3163  av_freep(&ac->che[type][i]);
3164  }
3165  }
3166 
3167  ff_mdct_end(&ac->mdct);
3168  ff_mdct_end(&ac->mdct_small);
3169  ff_mdct_end(&ac->mdct_ld);
3170  ff_mdct_end(&ac->mdct_ltp);
3171 #if !USE_FIXED
3172  ff_imdct15_uninit(&ac->mdct480);
3173 #endif
3174  av_freep(&ac->fdsp);
3175  return 0;
3176 }
3177 
3178 static void aacdec_init(AACContext *c)
3179 {
3180  c->imdct_and_windowing = imdct_and_windowing;
3181  c->apply_ltp = apply_ltp;
3182  c->apply_tns = apply_tns;
3183  c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
3184  c->update_ltp = update_ltp;
3185 #if USE_FIXED
3186  c->vector_pow43 = vector_pow43;
3187  c->subband_scale = subband_scale;
3188 #endif
3189 
3190 #if !USE_FIXED
3191  if(ARCH_MIPS)
3192  ff_aacdec_init_mips(c);
3193 #endif /* !USE_FIXED */
3194 }
3195 /**
3196  * AVOptions for Japanese DTV specific extensions (ADTS only)
3197  */
3198 #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
3199 static const AVOption options[] = {
3200  {"dual_mono_mode", "Select the channel to decode for dual mono",
3201  offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
3202  AACDEC_FLAGS, "dual_mono_mode"},
3203 
3204  {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3205  {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3206  {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3207  {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3208 
3209  {NULL},
3210 };
3211 
3212 static const AVClass aac_decoder_class = {
3213  .class_name = "AAC decoder",
3214  .item_name = av_default_item_name,
3215  .option = options,
3216  .version = LIBAVUTIL_VERSION_INT,
3217 };
3218 
3219 static const AVProfile profiles[] = {
3220  { FF_PROFILE_AAC_MAIN, "Main" },
3221  { FF_PROFILE_AAC_LOW, "LC" },
3222  { FF_PROFILE_AAC_SSR, "SSR" },
3223  { FF_PROFILE_AAC_LTP, "LTP" },
3224  { FF_PROFILE_AAC_HE, "HE-AAC" },
3225  { FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
3226  { FF_PROFILE_AAC_LD, "LD" },
3227  { FF_PROFILE_AAC_ELD, "ELD" },
3228  { FF_PROFILE_UNKNOWN },
3229 };
IndividualChannelStream::predictor_initialized
int predictor_initialized
Definition: aac.h:184
VMUL4S
static float * VMUL4S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
Definition: aacdec.c:123
AACContext::fdsp
AVFloatDSPContext * fdsp
Definition: aac.h:326
apply_prediction
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
Apply AAC-Main style frequency domain prediction.
Definition: aacdec_template.c:1863
AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:70
imdct_and_windowing
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
Conduct IMDCT and windowing.
Definition: aacdec_template.c:2498
apply_ltp
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
Apply the long term prediction.
Definition: aacdec_template.c:2428
NULL
#define NULL
Definition: coverity.c:32
v
float v
Definition: avf_showspectrum.c:96
s
const char * s
Definition: avisynth_c.h:631
decode_pce
static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac, uint8_t(*layout_map)[3], GetBitContext *gb)
Decode program configuration element; reference: table 4.2.
Definition: aacdec_template.c:717
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:193
FF_PROFILE_AAC_SSR
#define FF_PROFILE_AAC_SSR
Definition: avcodec.h:3121
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:1471
IndividualChannelStream::use_kb_window
uint8_t use_kb_window[2]
If set, use Kaiser-Bessel window, otherwise use a sine window.
Definition: aac.h:174
AACContext::subband_scale
void(* subband_scale)(int *dst, int *src, int scale, int offset, int len)
Definition: aac.h:361
overread_err
#define overread_err
Definition: aacdec_template.c:99
AACContext::mdct480
IMDCT15Context * mdct480
Definition: aac.h:325
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:171
FF_PROFILE_AAC_ELD
#define FF_PROFILE_AAC_ELD
Definition: avcodec.h:3126
AACADTSHeaderInfo::object_type
uint8_t object_type
Definition: aacadtsdec.h:36
AVOption
AVOption.
Definition: opt.h:255
data
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
flush
static void flush(AVCodecContext *avctx)
Definition: aacdec_template.c:490
tags_per_config
static const int8_t tags_per_config[16]
Definition: aacdectab.h:46
AACContext::avctx
AVCodecContext * avctx
Definition: aac.h:290
Pulse
Definition: aac.h:221
id
enum AVCodecID id
Definition: mxfenc.c:101
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:124
AAC_MUL26
#define AAC_MUL26(x, y)
Definition: aac_defines.h:98
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:260
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:62
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:257
TYPE_END
Definition: aac.h:63
cce_scale
static const float cce_scale[]
Definition: aacdec_template.c:2105
skip_bits_long
static void skip_bits_long(GetBitContext *s, int n)
Definition: get_bits.h:217
g
const char * g
Definition: vf_curves.c:108
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:106
AACDEC_FLAGS
#define AACDEC_FLAGS
AVOptions for Japanese DTV specific extensions (ADTS only)
Definition: aacdec_template.c:3198
imdct_and_windowing_eld
static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
Definition: aacdec_template.c:2596
elem_to_channel
Definition: aacdec_template.c:190
aacdec_init
static void aacdec_init(AACContext *ac)
Definition: aacdec_template.c:3178
FIXR10
#define FIXR10(x)
Definition: aac_defines.h:91
DEC_SQUAD
static int * DEC_SQUAD(int *dst, unsigned idx)
Definition: aacdec_fixed.c:114
TYPE_SCE
Definition: aac.h:56
TYPE_CPE
Definition: aac.h:57
AACContext::che
ChannelElement * che[4][MAX_ELEM_ID]
Definition: aac.h:300
AVPacket::size
int size
Definition: avcodec.h:1424
b
const char * b
Definition: vf_curves.c:109
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
SingleChannelElement::ret
INTFLOAT * ret
PCM output.
Definition: aac.h:264
ChannelElement::present
int present
Definition: aac.h:271
update_ltp
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
Update the LTP buffer for next frame.
Definition: aacdec_template.c:2460
AACContext::update_ltp
void(* update_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:359
AACContext::imdct_and_windowing
void(* imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:353
vector_pow43
static void vector_pow43(int *coefs, int len)
Definition: aacdec_fixed.c:150
OutputConfiguration::channel_layout
uint64_t channel_layout
Definition: aac.h:128
SingleChannelElement::sf
INTFLOAT sf[120]
scalefactors
Definition: aac.h:252
VLC_TYPE
#define VLC_TYPE
Definition: get_bits.h:61
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
ltp_coef
static const INTFLOAT ltp_coef[8]
Definition: aacdectab.h:41
ChannelElement::ms_mask
uint8_t ms_mask[128]
Set if mid/side stereo is used for each scalefactor window band.
Definition: aac.h:276
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:245
subband_scale
static void subband_scale(int *dst, int *src, int scale, int offset, int len)
Definition: aacdec_fixed.c:164
MAX_LTP_LONG_SFB
#define MAX_LTP_LONG_SFB
Definition: aac.h:51
GET_GAIN
#define GET_GAIN(x, y)
Definition: aac_defines.h:96
DynamicRangeControl
Dynamic Range Control - decoded from the bitstream but not processed further.
Definition: aac.h:208
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:125
vlc_scalefactors
static VLC vlc_scalefactors
Definition: aacdec_template.c:92
NOISE_PRE
#define NOISE_PRE
preamble for NOISE_BT, put in bitstream with the first noise band
Definition: aac.h:154
predict
static av_always_inline void predict(PredictorState *ps, float *coef, int output_enable)
Definition: aacdec.c:173
ChannelCoupling::type
enum RawDataBlockType type[8]
Type of channel element to be coupled - SCE or CPE.
Definition: aac.h:234
AVCodecContext::profile
int profile
profile
Definition: avcodec.h:3115
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:434
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:682
aac_decode_er_frame
static int aac_decode_er_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb)
Definition: aacdec_template.c:2833
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:2117
USE_FIXED
#define USE_FIXED
Definition: aac_defines.h:25
apply_tns
static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns, IndividualChannelStream *ics, int decode)
Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4...
Definition: aacdec_template.c:2346
lcg_random
static av_always_inline int lcg_random(unsigned previous_val)
linear congruential pseudorandom number generator
Definition: aacdec_template.c:1020
DynamicRangeControl::band_incr
int band_incr
Number of DRC bands greater than 1 having DRC info.
Definition: aac.h:213
ff_aac_num_swb_128
const uint8_t ff_aac_num_swb_128[]
Definition: aactab.c:53
AAC_RENAME_32
#define AAC_RENAME_32(x)
Definition: aac_defines.h:84
AACContext::dmono_mode
int dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
Definition: aac.h:344
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:178
AOT_ER_AAC_LTP
N Error Resilient Long Term Prediction.
Definition: mpeg4audio.h:76
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:231
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:1236
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:295
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2270
uint8_t
uint8_t
Definition: audio_convert.c:194
FIXR
#define FIXR(x)
Definition: aac_defines.h:90
av_cold
#define av_cold
Definition: attributes.h:74
OutputConfiguration::layout_map
uint8_t layout_map[MAX_ELEM_ID *4][3]
Definition: aac.h:125
SingleChannelElement::saved
float saved[1536]
overlap
Definition: aac.h:259
OC_TRIAL_PCE
Output configuration under trial specified by an inband PCE.
Definition: aac.h:117
INIT_VLC_STATIC
#define INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
Definition: get_bits.h:476
ff_swb_offset_480
const uint16_t *const ff_swb_offset_480[]
Definition: aactab.c:1246
ChannelElement::ch
SingleChannelElement ch[2]
Definition: aac.h:279
INTFLOAT
#define INTFLOAT
Definition: aac_defines.h:85
ff_swb_offset_512
const uint16_t *const ff_swb_offset_512[]
Definition: aactab.c:1238
TYPE_LFE
Definition: aac.h:59
ff_tns_max_bands_480
const uint8_t ff_tns_max_bands_480[]
Definition: aactab.c:1280
end
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
SingleChannelElement::tns
TemporalNoiseShaping tns
Definition: aac.h:247
AOT_ER_AAC_LD
N Error Resilient Low Delay.
Definition: mpeg4audio.h:80
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:2288
FF_PROFILE_UNKNOWN
#define FF_PROFILE_UNKNOWN
Definition: avcodec.h:3116
ff_aac_scalefactor_bits
const uint8_t ff_aac_scalefactor_bits[121]
Definition: aactab.c:80
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:1617
ChannelCoupling::num_coupled
int num_coupled
number of target elements
Definition: aac.h:233
AV_CH_LOW_FREQUENCY
#define AV_CH_LOW_FREQUENCY
Definition: channel_layout.h:52
DynamicRangeControl::exclude_mask
int exclude_mask[MAX_CHANNELS]
Channels to be excluded from DRC processing.
Definition: aac.h:212
TemporalNoiseShaping::n_filt
int n_filt[8]
Definition: aac.h:197
AACContext::mdct_ltp
FFTContext mdct_ltp
Definition: aac.h:321
AACContext::vector_pow43
void(* vector_pow43)(int *coefs, int len)
Definition: aac.h:360
AACContext::output_element
SingleChannelElement * output_element[MAX_CHANNELS]
Points to each SingleChannelElement.
Definition: aac.h:335
aac_decode_init
static av_cold int aac_decode_init(AVCodecContext *avctx)
Definition: aacdec_template.c:1066
AVPacket::data
uint8_t * data
Definition: avcodec.h:1423
get_bits_count
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:212
AAC_MUL31
#define AAC_MUL31(x, y)
Definition: aac_defines.h:100
count_channels
static int count_channels(uint8_t(*layout)[3], int tags)
Definition: aacdec_template.c:101
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:1033
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:1498
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:235
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:1062
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:174
AOT_ER_AAC_LC
N Error Resilient Low Complexity.
Definition: mpeg4audio.h:75
AACContext::tag_che_map
ChannelElement * tag_che_map[4][MAX_ELEM_ID]
Definition: aac.h:301
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
m
unsigned m
Definition: audioconvert.c:187
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:2702
AACContext::random_state
int random_state
Definition: aac.h:328
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:2771
get_bits_left
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:588
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:210
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:1438
SingleChannelElement::coeffs
float coeffs[1024]
coefficients for IMDCT, maybe processed
Definition: aac.h:258
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:418
decode_fill
static int decode_fill(AACContext *ac, GetBitContext *gb, int len)
Definition: aacdec_template.c:2255
UPDATE_CACHE
#define UPDATE_CACHE(name, gb)
Definition: get_bits.h:173
SingleChannelElement::predictor_state
PredictorState predictor_state[MAX_PREDICTORS]
Definition: aac.h:263
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
FF_PROFILE_AAC_HE
#define FF_PROFILE_AAC_HE
Definition: avcodec.h:3123
FF_PROFILE_AAC_HE_V2
#define FF_PROFILE_AAC_HE_V2
Definition: avcodec.h:3124
ChannelElement::sbr
SpectralBandReplication sbr
Definition: aac.h:282
FF_PROFILE_AAC_MAIN
#define FF_PROFILE_AAC_MAIN
Definition: avcodec.h:3119
AACContext::mdct_small
FFTContext mdct_small
Definition: aac.h:319
AV_EF_EXPLODE
#define AV_EF_EXPLODE
abort decoding on minor error detection
Definition: avcodec.h:2901
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:59
ChannelCoupling::coupling_point
enum CouplingPoint coupling_point
The point during decoding at which coupling is applied.
Definition: aac.h:232
AVERROR
#define AVERROR(e)
Definition: error.h:43
elem_to_channel::av_position
uint64_t av_position
Definition: aacdec_template.c:191
ff_aac_num_swb_1024
const uint8_t ff_aac_num_swb_1024[]
Definition: aactab.c:41
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:36
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1597
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:865
ff_imdct15_uninit
av_cold void ff_imdct15_uninit(IMDCT15Context **ps)
Free an iMDCT.
Definition: imdct15.c:69
AACADTSHeaderInfo::sampling_index
uint8_t sampling_index
Definition: aacadtsdec.h:37
Pulse::amp
int amp[4]
Definition: aac.h:225
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:197
AACContext::temp
float temp[128]
Definition: aac.h:347
IndividualChannelStream::max_sfb
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:172
ff_mdct_init
#define ff_mdct_init
Definition: fft.h:167
push_output_configuration
static void push_output_configuration(AACContext *ac)
Save current output configuration if and only if it has been locked.
Definition: aacdec_template.c:407
ff_aac_eld_window_512
const float ff_aac_eld_window_512[1920]
Definition: aactab.c:1289
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:144
IndividualChannelStream::num_swb
int num_swb
number of scalefactor window bands
Definition: aac.h:180
count_paired_channels
static int count_paired_channels(uint8_t(*layout_map)[3], int tags, int pos, int *current)
Definition: aacdec_template.c:226
FFMAX
#define FFMAX(a, b)
Definition: common.h:79
fail
#define fail()
Definition: checkasm.h:57
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:216
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: get_bits.h:63
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2323
SKIP_BITS
#define SKIP_BITS(name, gb, num)
Definition: get_bits.h:188
POW_SF2_ZERO
#define POW_SF2_ZERO
ff_aac_pow2sf_tab index corresponding to pow(2, 0);
Definition: aac_tablegen_decl.h:26
AAC_RENAME
#define AAC_RENAME(x)
Definition: aac_defines.h:83
AACContext::warned_remapping_once
int warned_remapping_once
Definition: aac.h:303
FF_DEBUG_STARTCODE
#define FF_DEBUG_STARTCODE
Definition: avcodec.h:2856
AOT_ER_AAC_SCALABLE
N Error Resilient Scalable.
Definition: mpeg4audio.h:77
reset_predictor_group
static void reset_predictor_group(PredictorState *ps, int group_num)
Definition: aacdec_template.c:1049
get_che
static ChannelElement * get_che(AACContext *ac, int type, int elem_id)
Definition: aacdec_template.c:550
IndividualChannelStream::window_sequence
enum WindowSequence window_sequence[2]
Definition: aac.h:173
AV_CODEC_FLAG_BITEXACT
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:788
ff_aac_num_swb_512
const uint8_t ff_aac_num_swb_512[]
Definition: aactab.c:45
AVCodecContext::err_recognition
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
Definition: avcodec.h:2890
aac_decode_frame
static int aac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: aacdec_template.c:3085
IndividualChannelStream::predictor_reset_group
int predictor_reset_group
Definition: aac.h:185
frame_configure_elements
static int frame_configure_elements(AVCodecContext *avctx)
Definition: aacdec_template.c:156
FFMIN
#define FFMIN(a, b)
Definition: common.h:81
DynamicRangeControl::dyn_rng_ctl
int dyn_rng_ctl[17]
DRC magnitude information.
Definition: aac.h:211
AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:69
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:1541
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:2014
AACADTSHeaderInfo::num_aac_frames
uint8_t num_aac_frames
Definition: aacadtsdec.h:39
Pulse::pos
int pos[4]
Definition: aac.h:224
AOT_AAC_MAIN
Y Main.
Definition: mpeg4audio.h:61
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:287
AACContext::mdct_ld
FFTContext mdct_ld
Definition: aac.h:320
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:1093
LAST_SKIP_BITS
#define LAST_SKIP_BITS(name, gb, num)
Definition: get_bits.h:194
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:555
TemporalNoiseShaping::length
int length[8][4]
Definition: aac.h:198
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
ff_aac_sbr_ctx_close
void AAC_RENAME() ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
Close one SBR context.
Definition: aacsbr_template.c:99
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:2669
AV_CH_FRONT_LEFT_OF_CENTER
#define AV_CH_FRONT_LEFT_OF_CENTER
Definition: channel_layout.h:55
u
float u
Definition: avf_showspectrum.c:96
n
int n
Definition: avisynth_c.h:547
ff_tns_max_bands_1024
const uint8_t ff_tns_max_bands_1024[]
Definition: aactab.c:1272
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:489
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:163
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:1222
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:2900
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:209
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:1566
ff_aac_sbr_init
void AAC_RENAME() ff_aac_sbr_init(void)
Initialize SBR.
Definition: aacsbr_template.c:35
DynamicRangeControl::pce_instance_tag
int pce_instance_tag
Indicates with which program the DRC info is associated.
Definition: aac.h:209
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:2402
AOT_AAC_SCALABLE
N Scalable.
Definition: mpeg4audio.h:66
tns_tmp2_map
static const INTFLOAT *const tns_tmp2_map[4]
Definition: aactab.h:71
elem_to_channel::aac_position
uint8_t aac_position
Definition: aacdec_template.c:194
SHOW_UBITS
#define SHOW_UBITS(name, gb, num)
Definition: get_bits.h:206
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
av_log2
#define av_log2
Definition: intmath.h:100
DynamicRangeControl::interpolation_scheme
int interpolation_scheme
Indicates the interpolation scheme used in the SBR QMF domain.
Definition: aac.h:214
ChannelCoupling
coupling parameters
Definition: aac.h:231
AACContext::tags_mapped
int tags_mapped
Definition: aac.h:302
reset_all_predictors
static void reset_all_predictors(PredictorState *ps)
Definition: aacdec_template.c:1026
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:1183
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:236
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2282
AACContext::force_dmono_mode
int force_dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
Definition: aac.h:343
AV_PKT_DATA_NEW_EXTRADATA
Definition: avcodec.h:1226
TemporalNoiseShaping::order
int order[8][4]
Definition: aac.h:200
AV_LOG_INFO
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
AACContext::warned_num_aac_frames
int warned_num_aac_frames
Definition: aac.h:350
profiles
static const AVProfile profiles[]
Definition: aacdec_template.c:3219
AAC_INIT_VLC_STATIC
#define AAC_INIT_VLC_STATIC(num, size)
Definition: aacdec_template.c:1056
TemporalNoiseShaping
Temporal Noise Shaping.
Definition: aac.h:195
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:2262
ff_aac_kbd_short_128
float ff_aac_kbd_short_128[128]
Definition: aactab.c:37
cbrt_tab
static uint32_t cbrt_tab[1<< 13]
Definition: cbrt_tablegen.h:46
options
static const AVOption options[]
Definition: aacdec_template.c:3199
init_get_bits8
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:441
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:1396
FF_PROFILE_AAC_LTP
#define FF_PROFILE_AAC_LTP
Definition: avcodec.h:3122
AVCodecContext::debug
int debug
debug
Definition: avcodec.h:2842
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:2064
LongTermPrediction
Long Term Prediction.
Definition: aac.h:161
AVCodecContext
main external API structure.
Definition: avcodec.h:1502
AV_CH_FRONT_LEFT
#define AV_CH_FRONT_LEFT
Definition: channel_layout.h:49
NOISE_PRE_BITS
#define NOISE_PRE_BITS
length of preamble
Definition: aac.h:155
FF_PROFILE_AAC_LOW
#define FF_PROFILE_AAC_LOW
Definition: avcodec.h:3120
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:1040
OPEN_READER
#define OPEN_READER(name, gb)
Definition: get_bits.h:133
SingleChannelElement::ics
IndividualChannelStream ics
Definition: aac.h:246
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:553
MAX_PREDICTORS
#define MAX_PREDICTORS
Definition: aac.h:146
cbrtf
static av_always_inline float cbrtf(float x)
Definition: libm.h:59
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:1618
IndividualChannelStream::group_len
uint8_t group_len[8]
Definition: aac.h:176
AVERROR_BUG
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
cbrt_tableinit
static av_cold void AAC_RENAME() cbrt_tableinit(void)
Definition: cbrt_tablegen.h:48
get_bits1
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:304
skip_bits1
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:329
AACContext::apply_tns
void(* apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns, IndividualChannelStream *ics, int decode)
Definition: aac.h:355
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:422
aac_decode_close
static av_cold int aac_decode_close(AVCodecContext *avctx)
Definition: aacdec_template.c:3154
skip_bits
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:297
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)
Decode audio specific configuration; reference: table 1.13.
Definition: aacdec_template.c:940
AAC_MUL30
#define AAC_MUL30(x, y)
Definition: aac_defines.h:99
sniff_channel_order
static uint64_t sniff_channel_order(uint8_t(*layout_map)[3], int tags)
Definition: aacdec_template.c:258
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:2189
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:190
IMDCT15Context::imdct_half
void(* imdct_half)(struct IMDCT15Context *s, float *dst, const float *src, ptrdiff_t src_stride, float scale)
Calculate the middle half of the iMDCT.
Definition: imdct15.h:40
init_get_bits
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:410
AV_PKT_DATA_SKIP_SAMPLES
Recommmends skipping the specified number of samples.
Definition: avcodec.h:1314
GET_CACHE
#define GET_CACHE(name, gb)
Definition: get_bits.h:210
elem_to_channel::syn_ele
uint8_t syn_ele
Definition: aacdec_template.c:192
VMUL2
static float * VMUL2(float *dst, const float *v, unsigned idx, const float *scale)
Definition: aacdec.c:82
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:154
MAX_CHANNELS
#define MAX_CHANNELS
Definition: aac.h:47
AOT_ER_BSAC
N Error Resilient Bit-Sliced Arithmetic Coding.
Definition: mpeg4audio.h:79
TNS_MAX_ORDER
#define TNS_MAX_ORDER
Definition: aac.h:50
ff_aac_sbr_ctx_init
void AAC_RENAME() ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr)
Initialize one SBR context.
Definition: aacsbr_template.c:81
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:143
AACContext
main AAC context
Definition: aac.h:288
ff_imdct15_init
av_cold int ff_imdct15_init(IMDCT15Context **ps, int N)
Init an iMDCT of the length 2 * 15 * (2^N)
Definition: imdct15.c:90
ff_aac_scalefactor_code
const uint32_t ff_aac_scalefactor_code[121]
Definition: aactab.c:61
IndividualChannelStream::ltp
LongTermPrediction ltp
Definition: aac.h:177
ChannelElement::coup
ChannelCoupling coup
Definition: aac.h:281
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:1200
ff_tns_max_bands_128
const uint8_t ff_tns_max_bands_128[]
Definition: aactab.c:1284
NOISE_OFFSET
#define NOISE_OFFSET
subtracted from global gain, used as offset for the preamble
Definition: aac.h:156
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
SingleChannelElement::ltp_state
float ltp_state[3072]
time signal for LTP
Definition: aac.h:261
av_frame_unref
void av_frame_unref(AVFrame *frame)
Unreference all the buffers referenced by frame and reset the frame fields.
Definition: frame.c:464
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:251
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:1347
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:182
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:215
AV_CH_SIDE_RIGHT
#define AV_CH_SIDE_RIGHT
Definition: channel_layout.h:59
FF_DEBUG_PICT_INFO
#define FF_DEBUG_PICT_INFO
Definition: avcodec.h:2843
vlc_spectral
static VLC vlc_spectral[11]
Definition: aacdec_template.c:93
OutputConfiguration::status
enum OCStatus status
Definition: aac.h:129
ChannelCoupling::gain
INTFLOAT gain[16][120]
Definition: aac.h:239
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:96
ff_aac_tableinit
av_cold void ff_aac_tableinit(void)
Definition: aac_tablegen.h:35
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:513
M_SQRT2
#define M_SQRT2
Definition: mathematics.h:55
RANGE15
#define RANGE15(x)
Definition: aac_defines.h:95
decode
static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)
Definition: ccaption_dec.c:523
TemporalNoiseShaping::coef
INTFLOAT coef[8][4][TNS_MAX_ORDER]
Definition: aac.h:202
LongTermPrediction::lag
int16_t lag
Definition: aac.h:163
AACContext::che_drc
DynamicRangeControl che_drc
Definition: aac.h:294
reset_predict_state
static av_always_inline void reset_predict_state(PredictorState *ps)
Definition: aacdec.c:71
AACContext::frame
AVFrame * frame
Definition: aac.h:291
AACContext::oc
OutputConfiguration oc[2]
Definition: aac.h:349
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:1324
ff_aac_pred_sfb_max
const uint8_t ff_aac_pred_sfb_max[]
Definition: aactab.c:57
TemporalNoiseShaping::direction
int direction[8][4]
Definition: aac.h:199
AACContext::apply_ltp
void(* apply_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:354
IndividualChannelStream::prediction_used
uint8_t prediction_used[41]
Definition: aac.h:187
ff_aac_eld_window_480
const float ff_aac_eld_window_480[1800]
Definition: aactab.c:2256
SingleChannelElement
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:245
ff_mdct_end
#define ff_mdct_end
Definition: fft.h:168
ff_aac_num_swb_480
const uint8_t ff_aac_num_swb_480[]
Definition: aactab.c:49
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:1230
AACContext::windowing_and_mdct_ltp
void(* windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out, INTFLOAT *in, IndividualChannelStream *ics)
Definition: aac.h:357
TYPE_PCE
Definition: aac.h:61
AVProfile
AVProfile.
Definition: avcodec.h:3460
AV_EF_BITSTREAM
#define AV_EF_BITSTREAM
detect bitstream specification deviations
Definition: avcodec.h:2899
IndividualChannelStream
Individual Channel Stream.
Definition: aac.h:171
ff_aac_pow2sf_tab
float ff_aac_pow2sf_tab[428]
Definition: aac_tablegen.h:32
LongTermPrediction::coef
INTFLOAT coef
Definition: aac.h:164
avpriv_mpeg4audio_get_config
int avpriv_mpeg4audio_get_config(MPEG4AudioConfig *c, const uint8_t *buf, int bit_size, int sync_extension)
Parse MPEG-4 systems extradata to retrieve audio configuration.
Definition: mpeg4audio.c:81
ff_aac_codebook_vector_idx
const uint16_t *const ff_aac_codebook_vector_idx[]
Definition: aactab.c:1071
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:636
ChannelElement
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:270
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:1544
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:1899
re
float re
Definition: fft-test.c:73
av_free
#define av_free(p)
Definition: tableprint_vlc.h:34
decode_ga_specific_config
static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx, GetBitContext *gb, MPEG4AudioConfig *m4ac, int channel_config)
Decode GA "General Audio" specific configuration; reference: table 4.1.
Definition: aacdec_template.c:787
ff_tns_max_bands_512
const uint8_t ff_tns_max_bands_512[]
Definition: aactab.c:1276
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:2263
Pulse::num_pulse
int num_pulse
Definition: aac.h:222
DEC_SPAIR
static int * DEC_SPAIR(int *dst, unsigned idx)
Definition: aacdec_fixed.c:106
AVCodecContext::internal
struct AVCodecInternal * internal
Private context used for internal data.
Definition: avcodec.h:1552
ff_mpeg4audio_channels
const uint8_t ff_mpeg4audio_channels[8]
Definition: mpeg4audio.c:62
VLC::table
VLC_TYPE(* table)[2]
code, bits
Definition: get_bits.h:65
FF_COMPLIANCE_STRICT
#define FF_COMPLIANCE_STRICT
Strictly conform to all the things in the spec no matter what consequences.
Definition: avcodec.h:2822
AOT_AAC_LTP
Y Long Term Prediction.
Definition: mpeg4audio.h:64
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:449
layout
uint64_t layout
Definition: channel_layout.c:77
SingleChannelElement::band_type
enum BandType band_type[128]
band types
Definition: aac.h:249
out
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-> out
Definition: audio_convert.c:194
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:2208
AV_CH_FRONT_RIGHT
#define AV_CH_FRONT_RIGHT
Definition: channel_layout.h:50
SingleChannelElement::ret_buf
float ret_buf[2048]
PCM output buffer.
Definition: aac.h:260
imdct_and_windowing_ld
static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
Definition: aacdec_template.c:2563
AAC_SIGNE
#define AAC_SIGNE
Definition: aac_defines.h:89
AACContext::mdct
FFTContext mdct
Definition: aac.h:318
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:553
av_packet_get_side_data
uint8_t * av_packet_get_side_data(AVPacket *pkt, enum AVPacketSideDataType type, int *size)
Get side information from packet.
Definition: avpacket.c:324
av_always_inline
#define av_always_inline
Definition: attributes.h:37
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:1976
AV_CH_SIDE_LEFT
#define AV_CH_SIDE_LEFT
Definition: channel_layout.h:58
FFSWAP
#define FFSWAP(type, a, b)
Definition: common.h:84
MPEG4AudioConfig::ps
int ps
-1 implicit, 1 presence
Definition: mpeg4audio.h:40
LongTermPrediction::used
int8_t used[MAX_LTP_LONG_SFB]
Definition: aac.h:165
ff_swb_offset_128
const uint16_t *const ff_swb_offset_128[]
Definition: aactab.c:1254
LongTermPrediction::present
int8_t present
Definition: aac.h:162
AACADTSHeaderInfo::sample_rate
uint32_t sample_rate
Definition: aacadtsdec.h:32
aac_decoder_class
static const AVClass aac_decoder_class
Definition: aacdec_template.c:3212
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:215
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:2330
OutputConfiguration::layout_map_tags
int layout_map_tags
Definition: aac.h:126
AV_RL32
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_RL32
Definition: bytestream.h:87
AVPacket
This structure stores compressed data.
Definition: avcodec.h:1400
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:225
av_mallocz
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:252
AVCodecContext::strict_std_compliance
int strict_std_compliance
strictly follow the standard (MPEG4, ...).
Definition: avcodec.h:2820
FF_PROFILE_AAC_LD
#define FF_PROFILE_AAC_LD
Definition: avcodec.h:3125
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:132
AV_CH_BACK_RIGHT
#define AV_CH_BACK_RIGHT
Definition: channel_layout.h:54
AV_WL32
#define AV_WL32(p, v)
Definition: intreadwrite.h:426
AOT_AAC_LC
Y Low Complexity.
Definition: mpeg4audio.h:62
VMUL4
static float * VMUL4(float *dst, const float *v, unsigned idx, const float *scale)
Definition: aacdec.c:93
AACContext::buf_mdct
float buf_mdct[1024]
Definition: aac.h:311
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:48
RawDataBlockType
RawDataBlockType
Definition: aac.h:55
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