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aacdec_template.c
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1 /*
2  * AAC decoder
3  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
4  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
5  * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
6  *
7  * AAC LATM decoder
8  * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
9  * Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
10  *
11  * AAC decoder fixed-point implementation
12  * Copyright (c) 2013
13  * MIPS Technologies, Inc., California.
14  *
15  * This file is part of FFmpeg.
16  *
17  * FFmpeg is free software; you can redistribute it and/or
18  * modify it under the terms of the GNU Lesser General Public
19  * License as published by the Free Software Foundation; either
20  * version 2.1 of the License, or (at your option) any later version.
21  *
22  * FFmpeg is distributed in the hope that it will be useful,
23  * but WITHOUT ANY WARRANTY; without even the implied warranty of
24  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
25  * Lesser General Public License for more details.
26  *
27  * You should have received a copy of the GNU Lesser General Public
28  * License along with FFmpeg; if not, write to the Free Software
29  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
30  */
31 
32 /**
33  * @file
34  * AAC decoder
35  * @author Oded Shimon ( ods15 ods15 dyndns org )
36  * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
37  *
38  * AAC decoder fixed-point implementation
39  * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
40  * @author Nedeljko Babic ( nedeljko.babic imgtec com )
41  */
42 
43 /*
44  * supported tools
45  *
46  * Support? Name
47  * N (code in SoC repo) gain control
48  * Y block switching
49  * Y window shapes - standard
50  * N window shapes - Low Delay
51  * Y filterbank - standard
52  * N (code in SoC repo) filterbank - Scalable Sample Rate
53  * Y Temporal Noise Shaping
54  * Y Long Term Prediction
55  * Y intensity stereo
56  * Y channel coupling
57  * Y frequency domain prediction
58  * Y Perceptual Noise Substitution
59  * Y Mid/Side stereo
60  * N Scalable Inverse AAC Quantization
61  * N Frequency Selective Switch
62  * N upsampling filter
63  * Y quantization & coding - AAC
64  * N quantization & coding - TwinVQ
65  * N quantization & coding - BSAC
66  * N AAC Error Resilience tools
67  * N Error Resilience payload syntax
68  * N Error Protection tool
69  * N CELP
70  * N Silence Compression
71  * N HVXC
72  * N HVXC 4kbits/s VR
73  * N Structured Audio tools
74  * N Structured Audio Sample Bank Format
75  * N MIDI
76  * N Harmonic and Individual Lines plus Noise
77  * N Text-To-Speech Interface
78  * Y Spectral Band Replication
79  * Y (not in this code) Layer-1
80  * Y (not in this code) Layer-2
81  * Y (not in this code) Layer-3
82  * N SinuSoidal Coding (Transient, Sinusoid, Noise)
83  * Y Parametric Stereo
84  * N Direct Stream Transfer
85  * Y (not in fixed point code) Enhanced AAC Low Delay (ER AAC ELD)
86  *
87  * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
88  * - HE AAC v2 comprises LC AAC with Spectral Band Replication and
89  Parametric Stereo.
90  */
91 
92 #include "libavutil/thread.h"
93 
94 static VLC vlc_scalefactors;
95 static VLC vlc_spectral[11];
96 
97 static int output_configure(AACContext *ac,
98  uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
99  enum OCStatus oc_type, int get_new_frame);
100 
101 #define overread_err "Input buffer exhausted before END element found\n"
102 
103 static int count_channels(uint8_t (*layout)[3], int tags)
104 {
105  int i, sum = 0;
106  for (i = 0; i < tags; i++) {
107  int syn_ele = layout[i][0];
108  int pos = layout[i][2];
109  sum += (1 + (syn_ele == TYPE_CPE)) *
110  (pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);
111  }
112  return sum;
113 }
114 
115 /**
116  * Check for the channel element in the current channel position configuration.
117  * If it exists, make sure the appropriate element is allocated and map the
118  * channel order to match the internal FFmpeg channel layout.
119  *
120  * @param che_pos current channel position configuration
121  * @param type channel element type
122  * @param id channel element id
123  * @param channels count of the number of channels in the configuration
124  *
125  * @return Returns error status. 0 - OK, !0 - error
126  */
127 static av_cold int che_configure(AACContext *ac,
128  enum ChannelPosition che_pos,
129  int type, int id, int *channels)
130 {
131  if (*channels >= MAX_CHANNELS)
132  return AVERROR_INVALIDDATA;
133  if (che_pos) {
134  if (!ac->che[type][id]) {
135  if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
136  return AVERROR(ENOMEM);
137  AAC_RENAME(ff_aac_sbr_ctx_init)(ac, &ac->che[type][id]->sbr);
138  }
139  if (type != TYPE_CCE) {
140  if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
141  av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
142  return AVERROR_INVALIDDATA;
143  }
144  ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
145  if (type == TYPE_CPE ||
146  (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
147  ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
148  }
149  }
150  } else {
151  if (ac->che[type][id])
152  AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][id]->sbr);
153  av_freep(&ac->che[type][id]);
154  }
155  return 0;
156 }
157 
158 static int frame_configure_elements(AVCodecContext *avctx)
159 {
160  AACContext *ac = avctx->priv_data;
161  int type, id, ch, ret;
162 
163  /* set channel pointers to internal buffers by default */
164  for (type = 0; type < 4; type++) {
165  for (id = 0; id < MAX_ELEM_ID; id++) {
166  ChannelElement *che = ac->che[type][id];
167  if (che) {
168  che->ch[0].ret = che->ch[0].ret_buf;
169  che->ch[1].ret = che->ch[1].ret_buf;
170  }
171  }
172  }
173 
174  /* get output buffer */
175  av_frame_unref(ac->frame);
176  if (!avctx->channels)
177  return 1;
178 
179  ac->frame->nb_samples = 2048;
180  if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
181  return ret;
182 
183  /* map output channel pointers to AVFrame data */
184  for (ch = 0; ch < avctx->channels; ch++) {
185  if (ac->output_element[ch])
186  ac->output_element[ch]->ret = (INTFLOAT *)ac->frame->extended_data[ch];
187  }
188 
189  return 0;
190 }
191 
192 struct elem_to_channel {
193  uint64_t av_position;
194  uint8_t syn_ele;
195  uint8_t elem_id;
196  uint8_t aac_position;
197 };
198 
199 static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
200  uint8_t (*layout_map)[3], int offset, uint64_t left,
201  uint64_t right, int pos)
202 {
203  if (layout_map[offset][0] == TYPE_CPE) {
204  e2c_vec[offset] = (struct elem_to_channel) {
205  .av_position = left | right,
206  .syn_ele = TYPE_CPE,
207  .elem_id = layout_map[offset][1],
208  .aac_position = pos
209  };
210  return 1;
211  } else {
212  e2c_vec[offset] = (struct elem_to_channel) {
213  .av_position = left,
214  .syn_ele = TYPE_SCE,
215  .elem_id = layout_map[offset][1],
216  .aac_position = pos
217  };
218  e2c_vec[offset + 1] = (struct elem_to_channel) {
219  .av_position = right,
220  .syn_ele = TYPE_SCE,
221  .elem_id = layout_map[offset + 1][1],
222  .aac_position = pos
223  };
224  return 2;
225  }
226 }
227 
228 static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
229  int *current)
230 {
231  int num_pos_channels = 0;
232  int first_cpe = 0;
233  int sce_parity = 0;
234  int i;
235  for (i = *current; i < tags; i++) {
236  if (layout_map[i][2] != pos)
237  break;
238  if (layout_map[i][0] == TYPE_CPE) {
239  if (sce_parity) {
240  if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
241  sce_parity = 0;
242  } else {
243  return -1;
244  }
245  }
246  num_pos_channels += 2;
247  first_cpe = 1;
248  } else {
249  num_pos_channels++;
250  sce_parity ^= 1;
251  }
252  }
253  if (sce_parity &&
254  ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
255  return -1;
256  *current = i;
257  return num_pos_channels;
258 }
259 
260 static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
261 {
262  int i, n, total_non_cc_elements;
263  struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
264  int num_front_channels, num_side_channels, num_back_channels;
265  uint64_t layout;
266 
267  if (FF_ARRAY_ELEMS(e2c_vec) < tags)
268  return 0;
269 
270  i = 0;
271  num_front_channels =
272  count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
273  if (num_front_channels < 0)
274  return 0;
275  num_side_channels =
276  count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
277  if (num_side_channels < 0)
278  return 0;
279  num_back_channels =
280  count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
281  if (num_back_channels < 0)
282  return 0;
283 
284  if (num_side_channels == 0 && num_back_channels >= 4) {
285  num_side_channels = 2;
286  num_back_channels -= 2;
287  }
288 
289  i = 0;
290  if (num_front_channels & 1) {
291  e2c_vec[i] = (struct elem_to_channel) {
292  .av_position = AV_CH_FRONT_CENTER,
293  .syn_ele = TYPE_SCE,
294  .elem_id = layout_map[i][1],
295  .aac_position = AAC_CHANNEL_FRONT
296  };
297  i++;
298  num_front_channels--;
299  }
300  if (num_front_channels >= 4) {
301  i += assign_pair(e2c_vec, layout_map, i,
302  AV_CH_FRONT_LEFT_OF_CENTER,
303  AV_CH_FRONT_RIGHT_OF_CENTER,
304  AAC_CHANNEL_FRONT);
305  num_front_channels -= 2;
306  }
307  if (num_front_channels >= 2) {
308  i += assign_pair(e2c_vec, layout_map, i,
309  AV_CH_FRONT_LEFT,
310  AV_CH_FRONT_RIGHT,
311  AAC_CHANNEL_FRONT);
312  num_front_channels -= 2;
313  }
314  while (num_front_channels >= 2) {
315  i += assign_pair(e2c_vec, layout_map, i,
316  UINT64_MAX,
317  UINT64_MAX,
318  AAC_CHANNEL_FRONT);
319  num_front_channels -= 2;
320  }
321 
322  if (num_side_channels >= 2) {
323  i += assign_pair(e2c_vec, layout_map, i,
324  AV_CH_SIDE_LEFT,
325  AV_CH_SIDE_RIGHT,
326  AAC_CHANNEL_FRONT);
327  num_side_channels -= 2;
328  }
329  while (num_side_channels >= 2) {
330  i += assign_pair(e2c_vec, layout_map, i,
331  UINT64_MAX,
332  UINT64_MAX,
333  AAC_CHANNEL_SIDE);
334  num_side_channels -= 2;
335  }
336 
337  while (num_back_channels >= 4) {
338  i += assign_pair(e2c_vec, layout_map, i,
339  UINT64_MAX,
340  UINT64_MAX,
341  AAC_CHANNEL_BACK);
342  num_back_channels -= 2;
343  }
344  if (num_back_channels >= 2) {
345  i += assign_pair(e2c_vec, layout_map, i,
346  AV_CH_BACK_LEFT,
347  AV_CH_BACK_RIGHT,
348  AAC_CHANNEL_BACK);
349  num_back_channels -= 2;
350  }
351  if (num_back_channels) {
352  e2c_vec[i] = (struct elem_to_channel) {
353  .av_position = AV_CH_BACK_CENTER,
354  .syn_ele = TYPE_SCE,
355  .elem_id = layout_map[i][1],
356  .aac_position = AAC_CHANNEL_BACK
357  };
358  i++;
359  num_back_channels--;
360  }
361 
362  if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
363  e2c_vec[i] = (struct elem_to_channel) {
364  .av_position = AV_CH_LOW_FREQUENCY,
365  .syn_ele = TYPE_LFE,
366  .elem_id = layout_map[i][1],
367  .aac_position = AAC_CHANNEL_LFE
368  };
369  i++;
370  }
371  while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
372  e2c_vec[i] = (struct elem_to_channel) {
373  .av_position = UINT64_MAX,
374  .syn_ele = TYPE_LFE,
375  .elem_id = layout_map[i][1],
376  .aac_position = AAC_CHANNEL_LFE
377  };
378  i++;
379  }
380 
381  // Must choose a stable sort
382  total_non_cc_elements = n = i;
383  do {
384  int next_n = 0;
385  for (i = 1; i < n; i++)
386  if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
387  FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
388  next_n = i;
389  }
390  n = next_n;
391  } while (n > 0);
392 
393  layout = 0;
394  for (i = 0; i < total_non_cc_elements; i++) {
395  layout_map[i][0] = e2c_vec[i].syn_ele;
396  layout_map[i][1] = e2c_vec[i].elem_id;
397  layout_map[i][2] = e2c_vec[i].aac_position;
398  if (e2c_vec[i].av_position != UINT64_MAX) {
399  layout |= e2c_vec[i].av_position;
400  }
401  }
402 
403  return layout;
404 }
405 
406 /**
407  * Save current output configuration if and only if it has been locked.
408  */
409 static void push_output_configuration(AACContext *ac) {
410  if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
411  ac->oc[0] = ac->oc[1];
412  }
413  ac->oc[1].status = OC_NONE;
414 }
415 
416 /**
417  * Restore the previous output configuration if and only if the current
418  * configuration is unlocked.
419  */
420 static void pop_output_configuration(AACContext *ac) {
421  if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
422  ac->oc[1] = ac->oc[0];
423  ac->avctx->channels = ac->oc[1].channels;
424  ac->avctx->channel_layout = ac->oc[1].channel_layout;
425  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
426  ac->oc[1].status, 0);
427  }
428 }
429 
430 /**
431  * Configure output channel order based on the current program
432  * configuration element.
433  *
434  * @return Returns error status. 0 - OK, !0 - error
435  */
436 static int output_configure(AACContext *ac,
437  uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
438  enum OCStatus oc_type, int get_new_frame)
439 {
440  AVCodecContext *avctx = ac->avctx;
441  int i, channels = 0, ret;
442  uint64_t layout = 0;
443  uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
444  uint8_t type_counts[TYPE_END] = { 0 };
445 
446  if (ac->oc[1].layout_map != layout_map) {
447  memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
448  ac->oc[1].layout_map_tags = tags;
449  }
450  for (i = 0; i < tags; i++) {
451  int type = layout_map[i][0];
452  int id = layout_map[i][1];
453  id_map[type][id] = type_counts[type]++;
454  if (id_map[type][id] >= MAX_ELEM_ID) {
455  avpriv_request_sample(ac->avctx, "Remapped id too large\n");
456  return AVERROR_PATCHWELCOME;
457  }
458  }
459  // Try to sniff a reasonable channel order, otherwise output the
460  // channels in the order the PCE declared them.
461  if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
462  layout = sniff_channel_order(layout_map, tags);
463  for (i = 0; i < tags; i++) {
464  int type = layout_map[i][0];
465  int id = layout_map[i][1];
466  int iid = id_map[type][id];
467  int position = layout_map[i][2];
468  // Allocate or free elements depending on if they are in the
469  // current program configuration.
470  ret = che_configure(ac, position, type, iid, &channels);
471  if (ret < 0)
472  return ret;
473  ac->tag_che_map[type][id] = ac->che[type][iid];
474  }
475  if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
476  if (layout == AV_CH_FRONT_CENTER) {
477  layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
478  } else {
479  layout = 0;
480  }
481  }
482 
483  if (layout) avctx->channel_layout = layout;
484  ac->oc[1].channel_layout = layout;
485  avctx->channels = ac->oc[1].channels = channels;
486  ac->oc[1].status = oc_type;
487 
488  if (get_new_frame) {
489  if ((ret = frame_configure_elements(ac->avctx)) < 0)
490  return ret;
491  }
492 
493  return 0;
494 }
495 
496 static void flush(AVCodecContext *avctx)
497 {
498  AACContext *ac= avctx->priv_data;
499  int type, i, j;
500 
501  for (type = 3; type >= 0; type--) {
502  for (i = 0; i < MAX_ELEM_ID; i++) {
503  ChannelElement *che = ac->che[type][i];
504  if (che) {
505  for (j = 0; j <= 1; j++) {
506  memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
507  }
508  }
509  }
510  }
511 }
512 
513 /**
514  * Set up channel positions based on a default channel configuration
515  * as specified in table 1.17.
516  *
517  * @return Returns error status. 0 - OK, !0 - error
518  */
519 static int set_default_channel_config(AVCodecContext *avctx,
520  uint8_t (*layout_map)[3],
521  int *tags,
522  int channel_config)
523 {
524  if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
525  channel_config > 12) {
526  av_log(avctx, AV_LOG_ERROR,
527  "invalid default channel configuration (%d)\n",
528  channel_config);
529  return AVERROR_INVALIDDATA;
530  }
531  *tags = tags_per_config[channel_config];
532  memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
533  *tags * sizeof(*layout_map));
534 
535  /*
536  * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
537  * However, at least Nero AAC encoder encodes 7.1 streams using the default
538  * channel config 7, mapping the side channels of the original audio stream
539  * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
540  * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
541  * the incorrect streams as if they were correct (and as the encoder intended).
542  *
543  * As actual intended 7.1(wide) streams are very rare, default to assuming a
544  * 7.1 layout was intended.
545  */
546  if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
547  av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
548  " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
549  " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
550  layout_map[2][2] = AAC_CHANNEL_SIDE;
551  }
552 
553  return 0;
554 }
555 
556 static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
557 {
558  /* For PCE based channel configurations map the channels solely based
559  * on tags. */
560  if (!ac->oc[1].m4ac.chan_config) {
561  return ac->tag_che_map[type][elem_id];
562  }
563  // Allow single CPE stereo files to be signalled with mono configuration.
564  if (!ac->tags_mapped && type == TYPE_CPE &&
565  ac->oc[1].m4ac.chan_config == 1) {
566  uint8_t layout_map[MAX_ELEM_ID*4][3];
567  int layout_map_tags;
568  push_output_configuration(ac);
569 
570  av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
571 
572  if (set_default_channel_config(ac->avctx, layout_map,
573  &layout_map_tags, 2) < 0)
574  return NULL;
575  if (output_configure(ac, layout_map, layout_map_tags,
576  OC_TRIAL_FRAME, 1) < 0)
577  return NULL;
578 
579  ac->oc[1].m4ac.chan_config = 2;
580  ac->oc[1].m4ac.ps = 0;
581  }
582  // And vice-versa
583  if (!ac->tags_mapped && type == TYPE_SCE &&
584  ac->oc[1].m4ac.chan_config == 2) {
585  uint8_t layout_map[MAX_ELEM_ID * 4][3];
586  int layout_map_tags;
587  push_output_configuration(ac);
588 
589  av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
590 
591  if (set_default_channel_config(ac->avctx, layout_map,
592  &layout_map_tags, 1) < 0)
593  return NULL;
594  if (output_configure(ac, layout_map, layout_map_tags,
595  OC_TRIAL_FRAME, 1) < 0)
596  return NULL;
597 
598  ac->oc[1].m4ac.chan_config = 1;
599  if (ac->oc[1].m4ac.sbr)
600  ac->oc[1].m4ac.ps = -1;
601  }
602  /* For indexed channel configurations map the channels solely based
603  * on position. */
604  switch (ac->oc[1].m4ac.chan_config) {
605  case 12:
606  case 7:
607  if (ac->tags_mapped == 3 && type == TYPE_CPE) {
608  ac->tags_mapped++;
609  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
610  }
611  case 11:
612  if (ac->tags_mapped == 2 &&
613  ac->oc[1].m4ac.chan_config == 11 &&
614  type == TYPE_SCE) {
615  ac->tags_mapped++;
616  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
617  }
618  case 6:
619  /* Some streams incorrectly code 5.1 audio as
620  * SCE[0] CPE[0] CPE[1] SCE[1]
621  * instead of
622  * SCE[0] CPE[0] CPE[1] LFE[0].
623  * If we seem to have encountered such a stream, transfer
624  * the LFE[0] element to the SCE[1]'s mapping */
625  if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
626  if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
627  av_log(ac->avctx, AV_LOG_WARNING,
628  "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
629  type == TYPE_SCE ? "SCE" : "LFE", elem_id);
630  ac->warned_remapping_once++;
631  }
632  ac->tags_mapped++;
633  return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
634  }
635  case 5:
636  if (ac->tags_mapped == 2 && type == TYPE_CPE) {
637  ac->tags_mapped++;
638  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
639  }
640  case 4:
641  /* Some streams incorrectly code 4.0 audio as
642  * SCE[0] CPE[0] LFE[0]
643  * instead of
644  * SCE[0] CPE[0] SCE[1].
645  * If we seem to have encountered such a stream, transfer
646  * the SCE[1] element to the LFE[0]'s mapping */
647  if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
648  if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
649  av_log(ac->avctx, AV_LOG_WARNING,
650  "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
651  type == TYPE_SCE ? "SCE" : "LFE", elem_id);
652  ac->warned_remapping_once++;
653  }
654  ac->tags_mapped++;
655  return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
656  }
657  if (ac->tags_mapped == 2 &&
658  ac->oc[1].m4ac.chan_config == 4 &&
659  type == TYPE_SCE) {
660  ac->tags_mapped++;
661  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
662  }
663  case 3:
664  case 2:
665  if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
666  type == TYPE_CPE) {
667  ac->tags_mapped++;
668  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
669  } else if (ac->oc[1].m4ac.chan_config == 2) {
670  return NULL;
671  }
672  case 1:
673  if (!ac->tags_mapped && type == TYPE_SCE) {
674  ac->tags_mapped++;
675  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
676  }
677  default:
678  return NULL;
679  }
680 }
681 
682 /**
683  * Decode an array of 4 bit element IDs, optionally interleaved with a
684  * stereo/mono switching bit.
685  *
686  * @param type speaker type/position for these channels
687  */
688 static void decode_channel_map(uint8_t layout_map[][3],
689  enum ChannelPosition type,
690  GetBitContext *gb, int n)
691 {
692  while (n--) {
693  enum RawDataBlockType syn_ele;
694  switch (type) {
695  case AAC_CHANNEL_FRONT:
696  case AAC_CHANNEL_BACK:
697  case AAC_CHANNEL_SIDE:
698  syn_ele = get_bits1(gb);
699  break;
700  case AAC_CHANNEL_CC:
701  skip_bits1(gb);
702  syn_ele = TYPE_CCE;
703  break;
704  case AAC_CHANNEL_LFE:
705  syn_ele = TYPE_LFE;
706  break;
707  default:
708  // AAC_CHANNEL_OFF has no channel map
709  av_assert0(0);
710  }
711  layout_map[0][0] = syn_ele;
712  layout_map[0][1] = get_bits(gb, 4);
713  layout_map[0][2] = type;
714  layout_map++;
715  }
716 }
717 
718 /**
719  * Decode program configuration element; reference: table 4.2.
720  *
721  * @return Returns error status. 0 - OK, !0 - error
722  */
723 static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
724  uint8_t (*layout_map)[3],
725  GetBitContext *gb)
726 {
727  int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
728  int sampling_index;
729  int comment_len;
730  int tags;
731 
732  skip_bits(gb, 2); // object_type
733 
734  sampling_index = get_bits(gb, 4);
735  if (m4ac->sampling_index != sampling_index)
736  av_log(avctx, AV_LOG_WARNING,
737  "Sample rate index in program config element does not "
738  "match the sample rate index configured by the container.\n");
739 
740  num_front = get_bits(gb, 4);
741  num_side = get_bits(gb, 4);
742  num_back = get_bits(gb, 4);
743  num_lfe = get_bits(gb, 2);
744  num_assoc_data = get_bits(gb, 3);
745  num_cc = get_bits(gb, 4);
746 
747  if (get_bits1(gb))
748  skip_bits(gb, 4); // mono_mixdown_tag
749  if (get_bits1(gb))
750  skip_bits(gb, 4); // stereo_mixdown_tag
751 
752  if (get_bits1(gb))
753  skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
754 
755  if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
756  av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
757  return -1;
758  }
759  decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front);
760  tags = num_front;
761  decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side);
762  tags += num_side;
763  decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back);
764  tags += num_back;
765  decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe);
766  tags += num_lfe;
767 
768  skip_bits_long(gb, 4 * num_assoc_data);
769 
770  decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc);
771  tags += num_cc;
772 
773  align_get_bits(gb);
774 
775  /* comment field, first byte is length */
776  comment_len = get_bits(gb, 8) * 8;
777  if (get_bits_left(gb) < comment_len) {
778  av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
779  return AVERROR_INVALIDDATA;
780  }
781  skip_bits_long(gb, comment_len);
782  return tags;
783 }
784 
785 /**
786  * Decode GA "General Audio" specific configuration; reference: table 4.1.
787  *
788  * @param ac pointer to AACContext, may be null
789  * @param avctx pointer to AVCCodecContext, used for logging
790  *
791  * @return Returns error status. 0 - OK, !0 - error
792  */
793 static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
794  GetBitContext *gb,
795  MPEG4AudioConfig *m4ac,
796  int channel_config)
797 {
798  int extension_flag, ret, ep_config, res_flags;
799  uint8_t layout_map[MAX_ELEM_ID*4][3];
800  int tags = 0;
801 
802  if (get_bits1(gb)) { // frameLengthFlag
803  avpriv_request_sample(avctx, "960/120 MDCT window");
804  return AVERROR_PATCHWELCOME;
805  }
806  m4ac->frame_length_short = 0;
807 
808  if (get_bits1(gb)) // dependsOnCoreCoder
809  skip_bits(gb, 14); // coreCoderDelay
810  extension_flag = get_bits1(gb);
811 
812  if (m4ac->object_type == AOT_AAC_SCALABLE ||
813  m4ac->object_type == AOT_ER_AAC_SCALABLE)
814  skip_bits(gb, 3); // layerNr
815 
816  if (channel_config == 0) {
817  skip_bits(gb, 4); // element_instance_tag
818  tags = decode_pce(avctx, m4ac, layout_map, gb);
819  if (tags < 0)
820  return tags;
821  } else {
822  if ((ret = set_default_channel_config(avctx, layout_map,
823  &tags, channel_config)))
824  return ret;
825  }
826 
827  if (count_channels(layout_map, tags) > 1) {
828  m4ac->ps = 0;
829  } else if (m4ac->sbr == 1 && m4ac->ps == -1)
830  m4ac->ps = 1;
831 
832  if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
833  return ret;
834 
835  if (extension_flag) {
836  switch (m4ac->object_type) {
837  case AOT_ER_BSAC:
838  skip_bits(gb, 5); // numOfSubFrame
839  skip_bits(gb, 11); // layer_length
840  break;
841  case AOT_ER_AAC_LC:
842  case AOT_ER_AAC_LTP:
843  case AOT_ER_AAC_SCALABLE:
844  case AOT_ER_AAC_LD:
845  res_flags = get_bits(gb, 3);
846  if (res_flags) {
847  avpriv_report_missing_feature(avctx,
848  "AAC data resilience (flags %x)",
849  res_flags);
850  return AVERROR_PATCHWELCOME;
851  }
852  break;
853  }
854  skip_bits1(gb); // extensionFlag3 (TBD in version 3)
855  }
856  switch (m4ac->object_type) {
857  case AOT_ER_AAC_LC:
858  case AOT_ER_AAC_LTP:
859  case AOT_ER_AAC_SCALABLE:
860  case AOT_ER_AAC_LD:
861  ep_config = get_bits(gb, 2);
862  if (ep_config) {
863  avpriv_report_missing_feature(avctx,
864  "epConfig %d", ep_config);
865  return AVERROR_PATCHWELCOME;
866  }
867  }
868  return 0;
869 }
870 
871 static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
872  GetBitContext *gb,
873  MPEG4AudioConfig *m4ac,
874  int channel_config)
875 {
876  int ret, ep_config, res_flags;
877  uint8_t layout_map[MAX_ELEM_ID*4][3];
878  int tags = 0;
879  const int ELDEXT_TERM = 0;
880 
881  m4ac->ps = 0;
882  m4ac->sbr = 0;
883 #if USE_FIXED
884  if (get_bits1(gb)) { // frameLengthFlag
885  avpriv_request_sample(avctx, "960/120 MDCT window");
886  return AVERROR_PATCHWELCOME;
887  }
888 #else
889  m4ac->frame_length_short = get_bits1(gb);
890 #endif
891  res_flags = get_bits(gb, 3);
892  if (res_flags) {
893  avpriv_report_missing_feature(avctx,
894  "AAC data resilience (flags %x)",
895  res_flags);
896  return AVERROR_PATCHWELCOME;
897  }
898 
899  if (get_bits1(gb)) { // ldSbrPresentFlag
900  avpriv_report_missing_feature(avctx,
901  "Low Delay SBR");
902  return AVERROR_PATCHWELCOME;
903  }
904 
905  while (get_bits(gb, 4) != ELDEXT_TERM) {
906  int len = get_bits(gb, 4);
907  if (len == 15)
908  len += get_bits(gb, 8);
909  if (len == 15 + 255)
910  len += get_bits(gb, 16);
911  if (get_bits_left(gb) < len * 8 + 4) {
912  av_log(avctx, AV_LOG_ERROR, overread_err);
913  return AVERROR_INVALIDDATA;
914  }
915  skip_bits_long(gb, 8 * len);
916  }
917 
918  if ((ret = set_default_channel_config(avctx, layout_map,
919  &tags, channel_config)))
920  return ret;
921 
922  if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
923  return ret;
924 
925  ep_config = get_bits(gb, 2);
926  if (ep_config) {
927  avpriv_report_missing_feature(avctx,
928  "epConfig %d", ep_config);
929  return AVERROR_PATCHWELCOME;
930  }
931  return 0;
932 }
933 
934 /**
935  * Decode audio specific configuration; reference: table 1.13.
936  *
937  * @param ac pointer to AACContext, may be null
938  * @param avctx pointer to AVCCodecContext, used for logging
939  * @param m4ac pointer to MPEG4AudioConfig, used for parsing
940  * @param data pointer to buffer holding an audio specific config
941  * @param bit_size size of audio specific config or data in bits
942  * @param sync_extension look for an appended sync extension
943  *
944  * @return Returns error status or number of consumed bits. <0 - error
945  */
946 static int decode_audio_specific_config(AACContext *ac,
947  AVCodecContext *avctx,
948  MPEG4AudioConfig *m4ac,
949  const uint8_t *data, int64_t bit_size,
950  int sync_extension)
951 {
952  GetBitContext gb;
953  int i, ret;
954 
955  if (bit_size < 0 || bit_size > INT_MAX) {
956  av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n");
957  return AVERROR_INVALIDDATA;
958  }
959 
960  ff_dlog(avctx, "audio specific config size %d\n", (int)bit_size >> 3);
961  for (i = 0; i < bit_size >> 3; i++)
962  ff_dlog(avctx, "%02x ", data[i]);
963  ff_dlog(avctx, "\n");
964 
965  if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
966  return ret;
967 
968  if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size,
969  sync_extension)) < 0)
970  return AVERROR_INVALIDDATA;
971  if (m4ac->sampling_index > 12) {
972  av_log(avctx, AV_LOG_ERROR,
973  "invalid sampling rate index %d\n",
974  m4ac->sampling_index);
975  return AVERROR_INVALIDDATA;
976  }
977  if (m4ac->object_type == AOT_ER_AAC_LD &&
978  (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
979  av_log(avctx, AV_LOG_ERROR,
980  "invalid low delay sampling rate index %d\n",
981  m4ac->sampling_index);
982  return AVERROR_INVALIDDATA;
983  }
984 
985  skip_bits_long(&gb, i);
986 
987  switch (m4ac->object_type) {
988  case AOT_AAC_MAIN:
989  case AOT_AAC_LC:
990  case AOT_AAC_LTP:
991  case AOT_ER_AAC_LC:
992  case AOT_ER_AAC_LD:
993  if ((ret = decode_ga_specific_config(ac, avctx, &gb,
994  m4ac, m4ac->chan_config)) < 0)
995  return ret;
996  break;
997  case AOT_ER_AAC_ELD:
998  if ((ret = decode_eld_specific_config(ac, avctx, &gb,
999  m4ac, m4ac->chan_config)) < 0)
1000  return ret;
1001  break;
1002  default:
1003  avpriv_report_missing_feature(avctx,
1004  "Audio object type %s%d",
1005  m4ac->sbr == 1 ? "SBR+" : "",
1006  m4ac->object_type);
1007  return AVERROR(ENOSYS);
1008  }
1009 
1010  ff_dlog(avctx,
1011  "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
1012  m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
1013  m4ac->sample_rate, m4ac->sbr,
1014  m4ac->ps);
1015 
1016  return get_bits_count(&gb);
1017 }
1018 
1019 /**
1020  * linear congruential pseudorandom number generator
1021  *
1022  * @param previous_val pointer to the current state of the generator
1023  *
1024  * @return Returns a 32-bit pseudorandom integer
1025  */
1026 static av_always_inline int lcg_random(unsigned previous_val)
1027 {
1028  union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
1029  return v.s;
1030 }
1031 
1032 static void reset_all_predictors(PredictorState *ps)
1033 {
1034  int i;
1035  for (i = 0; i < MAX_PREDICTORS; i++)
1036  reset_predict_state(&ps[i]);
1037 }
1038 
1039 static int sample_rate_idx (int rate)
1040 {
1041  if (92017 <= rate) return 0;
1042  else if (75132 <= rate) return 1;
1043  else if (55426 <= rate) return 2;
1044  else if (46009 <= rate) return 3;
1045  else if (37566 <= rate) return 4;
1046  else if (27713 <= rate) return 5;
1047  else if (23004 <= rate) return 6;
1048  else if (18783 <= rate) return 7;
1049  else if (13856 <= rate) return 8;
1050  else if (11502 <= rate) return 9;
1051  else if (9391 <= rate) return 10;
1052  else return 11;
1053 }
1054 
1055 static void reset_predictor_group(PredictorState *ps, int group_num)
1056 {
1057  int i;
1058  for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
1059  reset_predict_state(&ps[i]);
1060 }
1061 
1062 #define AAC_INIT_VLC_STATIC(num, size) \
1063  INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
1064  ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \
1065  sizeof(ff_aac_spectral_bits[num][0]), \
1066  ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
1067  sizeof(ff_aac_spectral_codes[num][0]), \
1068  size);
1069 
1070 static void aacdec_init(AACContext *ac);
1071 
1072 static av_cold void aac_static_table_init(void)
1073 {
1074  AAC_INIT_VLC_STATIC( 0, 304);
1075  AAC_INIT_VLC_STATIC( 1, 270);
1076  AAC_INIT_VLC_STATIC( 2, 550);
1077  AAC_INIT_VLC_STATIC( 3, 300);
1078  AAC_INIT_VLC_STATIC( 4, 328);
1079  AAC_INIT_VLC_STATIC( 5, 294);
1080  AAC_INIT_VLC_STATIC( 6, 306);
1081  AAC_INIT_VLC_STATIC( 7, 268);
1082  AAC_INIT_VLC_STATIC( 8, 510);
1083  AAC_INIT_VLC_STATIC( 9, 366);
1084  AAC_INIT_VLC_STATIC(10, 462);
1085 
1086  AAC_RENAME(ff_aac_sbr_init)();
1087 
1088  ff_aac_tableinit();
1089 
1090  INIT_VLC_STATIC(&vlc_scalefactors, 7,
1091  FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
1092  ff_aac_scalefactor_bits,
1093  sizeof(ff_aac_scalefactor_bits[0]),
1094  sizeof(ff_aac_scalefactor_bits[0]),
1095  ff_aac_scalefactor_code,
1096  sizeof(ff_aac_scalefactor_code[0]),
1097  sizeof(ff_aac_scalefactor_code[0]),
1098  352);
1099 
1100  // window initialization
1101  AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_1024), 4.0, 1024);
1102  AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_128), 6.0, 128);
1103  AAC_RENAME(ff_init_ff_sine_windows)(10);
1104  AAC_RENAME(ff_init_ff_sine_windows)( 9);
1105  AAC_RENAME(ff_init_ff_sine_windows)( 7);
1106 
1107  AAC_RENAME(ff_cbrt_tableinit)();
1108 }
1109 
1110 static AVOnce aac_table_init = AV_ONCE_INIT;
1111 
1112 static av_cold int aac_decode_init(AVCodecContext *avctx)
1113 {
1114  AACContext *ac = avctx->priv_data;
1115  int ret;
1116 
1117  ret = ff_thread_once(&aac_table_init, &aac_static_table_init);
1118  if (ret != 0)
1119  return AVERROR_UNKNOWN;
1120 
1121  ac->avctx = avctx;
1122  ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
1123 
1124  aacdec_init(ac);
1125 #if USE_FIXED
1126  avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
1127 #else
1128  avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1129 #endif /* USE_FIXED */
1130 
1131  if (avctx->extradata_size > 0) {
1132  if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
1133  avctx->extradata,
1134  avctx->extradata_size * 8LL,
1135  1)) < 0)
1136  return ret;
1137  } else {
1138  int sr, i;
1139  uint8_t layout_map[MAX_ELEM_ID*4][3];
1140  int layout_map_tags;
1141 
1142  sr = sample_rate_idx(avctx->sample_rate);
1143  ac->oc[1].m4ac.sampling_index = sr;
1144  ac->oc[1].m4ac.channels = avctx->channels;
1145  ac->oc[1].m4ac.sbr = -1;
1146  ac->oc[1].m4ac.ps = -1;
1147 
1148  for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
1149  if (ff_mpeg4audio_channels[i] == avctx->channels)
1150  break;
1151  if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
1152  i = 0;
1153  }
1154  ac->oc[1].m4ac.chan_config = i;
1155 
1156  if (ac->oc[1].m4ac.chan_config) {
1157  int ret = set_default_channel_config(avctx, layout_map,
1158  &layout_map_tags, ac->oc[1].m4ac.chan_config);
1159  if (!ret)
1160  output_configure(ac, layout_map, layout_map_tags,
1161  OC_GLOBAL_HDR, 0);
1162  else if (avctx->err_recognition & AV_EF_EXPLODE)
1163  return AVERROR_INVALIDDATA;
1164  }
1165  }
1166 
1167  if (avctx->channels > MAX_CHANNELS) {
1168  av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
1169  return AVERROR_INVALIDDATA;
1170  }
1171 
1172 #if USE_FIXED
1173  ac->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & AV_CODEC_FLAG_BITEXACT);
1174 #else
1175  ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
1176 #endif /* USE_FIXED */
1177  if (!ac->fdsp) {
1178  return AVERROR(ENOMEM);
1179  }
1180 
1181  ac->random_state = 0x1f2e3d4c;
1182 
1183  AAC_RENAME_32(ff_mdct_init)(&ac->mdct, 11, 1, 1.0 / RANGE15(1024.0));
1184  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld, 10, 1, 1.0 / RANGE15(512.0));
1185  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small, 8, 1, 1.0 / RANGE15(128.0));
1186  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp, 11, 0, RANGE15(-2.0));
1187 #if !USE_FIXED
1188  ret = ff_imdct15_init(&ac->mdct480, 5);
1189  if (ret < 0)
1190  return ret;
1191 #endif
1192 
1193  return 0;
1194 }
1195 
1196 /**
1197  * Skip data_stream_element; reference: table 4.10.
1198  */
1199 static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
1200 {
1201  int byte_align = get_bits1(gb);
1202  int count = get_bits(gb, 8);
1203  if (count == 255)
1204  count += get_bits(gb, 8);
1205  if (byte_align)
1206  align_get_bits(gb);
1207 
1208  if (get_bits_left(gb) < 8 * count) {
1209  av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
1210  return AVERROR_INVALIDDATA;
1211  }
1212  skip_bits_long(gb, 8 * count);
1213  return 0;
1214 }
1215 
1216 static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
1217  GetBitContext *gb)
1218 {
1219  int sfb;
1220  if (get_bits1(gb)) {
1221  ics->predictor_reset_group = get_bits(gb, 5);
1222  if (ics->predictor_reset_group == 0 ||
1223  ics->predictor_reset_group > 30) {
1224  av_log(ac->avctx, AV_LOG_ERROR,
1225  "Invalid Predictor Reset Group.\n");
1226  return AVERROR_INVALIDDATA;
1227  }
1228  }
1229  for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
1230  ics->prediction_used[sfb] = get_bits1(gb);
1231  }
1232  return 0;
1233 }
1234 
1235 /**
1236  * Decode Long Term Prediction data; reference: table 4.xx.
1237  */
1238 static void decode_ltp(LongTermPrediction *ltp,
1239  GetBitContext *gb, uint8_t max_sfb)
1240 {
1241  int sfb;
1242 
1243  ltp->lag = get_bits(gb, 11);
1244  ltp->coef = ltp_coef[get_bits(gb, 3)];
1245  for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
1246  ltp->used[sfb] = get_bits1(gb);
1247 }
1248 
1249 /**
1250  * Decode Individual Channel Stream info; reference: table 4.6.
1251  */
1252 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
1253  GetBitContext *gb)
1254 {
1255  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
1256  const int aot = m4ac->object_type;
1257  const int sampling_index = m4ac->sampling_index;
1258  if (aot != AOT_ER_AAC_ELD) {
1259  if (get_bits1(gb)) {
1260  av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
1261  if (ac->avctx->err_recognition & AV_EF_BITSTREAM)
1262  return AVERROR_INVALIDDATA;
1263  }
1264  ics->window_sequence[1] = ics->window_sequence[0];
1265  ics->window_sequence[0] = get_bits(gb, 2);
1266  if (aot == AOT_ER_AAC_LD &&
1267  ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
1268  av_log(ac->avctx, AV_LOG_ERROR,
1269  "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
1270  "window sequence %d found.\n", ics->window_sequence[0]);
1271  ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
1272  return AVERROR_INVALIDDATA;
1273  }
1274  ics->use_kb_window[1] = ics->use_kb_window[0];
1275  ics->use_kb_window[0] = get_bits1(gb);
1276  }
1277  ics->num_window_groups = 1;
1278  ics->group_len[0] = 1;
1279  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1280  int i;
1281  ics->max_sfb = get_bits(gb, 4);
1282  for (i = 0; i < 7; i++) {
1283  if (get_bits1(gb)) {
1284  ics->group_len[ics->num_window_groups - 1]++;
1285  } else {
1286  ics->num_window_groups++;
1287  ics->group_len[ics->num_window_groups - 1] = 1;
1288  }
1289  }
1290  ics->num_windows = 8;
1291  ics->swb_offset = ff_swb_offset_128[sampling_index];
1292  ics->num_swb = ff_aac_num_swb_128[sampling_index];
1293  ics->tns_max_bands = ff_tns_max_bands_128[sampling_index];
1294  ics->predictor_present = 0;
1295  } else {
1296  ics->max_sfb = get_bits(gb, 6);
1297  ics->num_windows = 1;
1298  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
1299  if (m4ac->frame_length_short) {
1300  ics->swb_offset = ff_swb_offset_480[sampling_index];
1301  ics->num_swb = ff_aac_num_swb_480[sampling_index];
1302  ics->tns_max_bands = ff_tns_max_bands_480[sampling_index];
1303  } else {
1304  ics->swb_offset = ff_swb_offset_512[sampling_index];
1305  ics->num_swb = ff_aac_num_swb_512[sampling_index];
1306  ics->tns_max_bands = ff_tns_max_bands_512[sampling_index];
1307  }
1308  if (!ics->num_swb || !ics->swb_offset)
1309  return AVERROR_BUG;
1310  } else {
1311  ics->swb_offset = ff_swb_offset_1024[sampling_index];
1312  ics->num_swb = ff_aac_num_swb_1024[sampling_index];
1313  ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
1314  }
1315  if (aot != AOT_ER_AAC_ELD) {
1316  ics->predictor_present = get_bits1(gb);
1317  ics->predictor_reset_group = 0;
1318  }
1319  if (ics->predictor_present) {
1320  if (aot == AOT_AAC_MAIN) {
1321  if (decode_prediction(ac, ics, gb)) {
1322  goto fail;
1323  }
1324  } else if (aot == AOT_AAC_LC ||
1325  aot == AOT_ER_AAC_LC) {
1326  av_log(ac->avctx, AV_LOG_ERROR,
1327  "Prediction is not allowed in AAC-LC.\n");
1328  goto fail;
1329  } else {
1330  if (aot == AOT_ER_AAC_LD) {
1331  av_log(ac->avctx, AV_LOG_ERROR,
1332  "LTP in ER AAC LD not yet implemented.\n");
1333  return AVERROR_PATCHWELCOME;
1334  }
1335  if ((ics->ltp.present = get_bits(gb, 1)))
1336  decode_ltp(&ics->ltp, gb, ics->max_sfb);
1337  }
1338  }
1339  }
1340 
1341  if (ics->max_sfb > ics->num_swb) {
1342  av_log(ac->avctx, AV_LOG_ERROR,
1343  "Number of scalefactor bands in group (%d) "
1344  "exceeds limit (%d).\n",
1345  ics->max_sfb, ics->num_swb);
1346  goto fail;
1347  }
1348 
1349  return 0;
1350 fail:
1351  ics->max_sfb = 0;
1352  return AVERROR_INVALIDDATA;
1353 }
1354 
1355 /**
1356  * Decode band types (section_data payload); reference: table 4.46.
1357  *
1358  * @param band_type array of the used band type
1359  * @param band_type_run_end array of the last scalefactor band of a band type run
1360  *
1361  * @return Returns error status. 0 - OK, !0 - error
1362  */
1363 static int decode_band_types(AACContext *ac, enum BandType band_type[120],
1364  int band_type_run_end[120], GetBitContext *gb,
1365  IndividualChannelStream *ics)
1366 {
1367  int g, idx = 0;
1368  const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
1369  for (g = 0; g < ics->num_window_groups; g++) {
1370  int k = 0;
1371  while (k < ics->max_sfb) {
1372  uint8_t sect_end = k;
1373  int sect_len_incr;
1374  int sect_band_type = get_bits(gb, 4);
1375  if (sect_band_type == 12) {
1376  av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
1377  return AVERROR_INVALIDDATA;
1378  }
1379  do {
1380  sect_len_incr = get_bits(gb, bits);
1381  sect_end += sect_len_incr;
1382  if (get_bits_left(gb) < 0) {
1383  av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
1384  return AVERROR_INVALIDDATA;
1385  }
1386  if (sect_end > ics->max_sfb) {
1387  av_log(ac->avctx, AV_LOG_ERROR,
1388  "Number of bands (%d) exceeds limit (%d).\n",
1389  sect_end, ics->max_sfb);
1390  return AVERROR_INVALIDDATA;
1391  }
1392  } while (sect_len_incr == (1 << bits) - 1);
1393  for (; k < sect_end; k++) {
1394  band_type [idx] = sect_band_type;
1395  band_type_run_end[idx++] = sect_end;
1396  }
1397  }
1398  }
1399  return 0;
1400 }
1401 
1402 /**
1403  * Decode scalefactors; reference: table 4.47.
1404  *
1405  * @param global_gain first scalefactor value as scalefactors are differentially coded
1406  * @param band_type array of the used band type
1407  * @param band_type_run_end array of the last scalefactor band of a band type run
1408  * @param sf array of scalefactors or intensity stereo positions
1409  *
1410  * @return Returns error status. 0 - OK, !0 - error
1411  */
1412 static int decode_scalefactors(AACContext *ac, INTFLOAT sf[120], GetBitContext *gb,
1413  unsigned int global_gain,
1414  IndividualChannelStream *ics,
1415  enum BandType band_type[120],
1416  int band_type_run_end[120])
1417 {
1418  int g, i, idx = 0;
1419  int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 };
1420  int clipped_offset;
1421  int noise_flag = 1;
1422  for (g = 0; g < ics->num_window_groups; g++) {
1423  for (i = 0; i < ics->max_sfb;) {
1424  int run_end = band_type_run_end[idx];
1425  if (band_type[idx] == ZERO_BT) {
1426  for (; i < run_end; i++, idx++)
1427  sf[idx] = FIXR(0.);
1428  } else if ((band_type[idx] == INTENSITY_BT) ||
1429  (band_type[idx] == INTENSITY_BT2)) {
1430  for (; i < run_end; i++, idx++) {
1431  offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1432  clipped_offset = av_clip(offset[2], -155, 100);
1433  if (offset[2] != clipped_offset) {
1434  avpriv_request_sample(ac->avctx,
1435  "If you heard an audible artifact, there may be a bug in the decoder. "
1436  "Clipped intensity stereo position (%d -> %d)",
1437  offset[2], clipped_offset);
1438  }
1439 #if USE_FIXED
1440  sf[idx] = 100 - clipped_offset;
1441 #else
1442  sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
1443 #endif /* USE_FIXED */
1444  }
1445  } else if (band_type[idx] == NOISE_BT) {
1446  for (; i < run_end; i++, idx++) {
1447  if (noise_flag-- > 0)
1448  offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
1449  else
1450  offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1451  clipped_offset = av_clip(offset[1], -100, 155);
1452  if (offset[1] != clipped_offset) {
1453  avpriv_request_sample(ac->avctx,
1454  "If you heard an audible artifact, there may be a bug in the decoder. "
1455  "Clipped noise gain (%d -> %d)",
1456  offset[1], clipped_offset);
1457  }
1458 #if USE_FIXED
1459  sf[idx] = -(100 + clipped_offset);
1460 #else
1461  sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
1462 #endif /* USE_FIXED */
1463  }
1464  } else {
1465  for (; i < run_end; i++, idx++) {
1466  offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1467  if (offset[0] > 255U) {
1468  av_log(ac->avctx, AV_LOG_ERROR,
1469  "Scalefactor (%d) out of range.\n", offset[0]);
1470  return AVERROR_INVALIDDATA;
1471  }
1472 #if USE_FIXED
1473  sf[idx] = -offset[0];
1474 #else
1475  sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
1476 #endif /* USE_FIXED */
1477  }
1478  }
1479  }
1480  }
1481  return 0;
1482 }
1483 
1484 /**
1485  * Decode pulse data; reference: table 4.7.
1486  */
1487 static int decode_pulses(Pulse *pulse, GetBitContext *gb,
1488  const uint16_t *swb_offset, int num_swb)
1489 {
1490  int i, pulse_swb;
1491  pulse->num_pulse = get_bits(gb, 2) + 1;
1492  pulse_swb = get_bits(gb, 6);
1493  if (pulse_swb >= num_swb)
1494  return -1;
1495  pulse->pos[0] = swb_offset[pulse_swb];
1496  pulse->pos[0] += get_bits(gb, 5);
1497  if (pulse->pos[0] >= swb_offset[num_swb])
1498  return -1;
1499  pulse->amp[0] = get_bits(gb, 4);
1500  for (i = 1; i < pulse->num_pulse; i++) {
1501  pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
1502  if (pulse->pos[i] >= swb_offset[num_swb])
1503  return -1;
1504  pulse->amp[i] = get_bits(gb, 4);
1505  }
1506  return 0;
1507 }
1508 
1509 /**
1510  * Decode Temporal Noise Shaping data; reference: table 4.48.
1511  *
1512  * @return Returns error status. 0 - OK, !0 - error
1513  */
1514 static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
1515  GetBitContext *gb, const IndividualChannelStream *ics)
1516 {
1517  int w, filt, i, coef_len, coef_res, coef_compress;
1518  const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
1519  const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
1520  for (w = 0; w < ics->num_windows; w++) {
1521  if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
1522  coef_res = get_bits1(gb);
1523 
1524  for (filt = 0; filt < tns->n_filt[w]; filt++) {
1525  int tmp2_idx;
1526  tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
1527 
1528  if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
1529  av_log(ac->avctx, AV_LOG_ERROR,
1530  "TNS filter order %d is greater than maximum %d.\n",
1531  tns->order[w][filt], tns_max_order);
1532  tns->order[w][filt] = 0;
1533  return AVERROR_INVALIDDATA;
1534  }
1535  if (tns->order[w][filt]) {
1536  tns->direction[w][filt] = get_bits1(gb);
1537  coef_compress = get_bits1(gb);
1538  coef_len = coef_res + 3 - coef_compress;
1539  tmp2_idx = 2 * coef_compress + coef_res;
1540 
1541  for (i = 0; i < tns->order[w][filt]; i++)
1542  tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
1543  }
1544  }
1545  }
1546  }
1547  return 0;
1548 }
1549 
1550 /**
1551  * Decode Mid/Side data; reference: table 4.54.
1552  *
1553  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
1554  * [1] mask is decoded from bitstream; [2] mask is all 1s;
1555  * [3] reserved for scalable AAC
1556  */
1557 static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
1558  int ms_present)
1559 {
1560  int idx;
1561  int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
1562  if (ms_present == 1) {
1563  for (idx = 0; idx < max_idx; idx++)
1564  cpe->ms_mask[idx] = get_bits1(gb);
1565  } else if (ms_present == 2) {
1566  memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
1567  }
1568 }
1569 
1570 /**
1571  * Decode spectral data; reference: table 4.50.
1572  * Dequantize and scale spectral data; reference: 4.6.3.3.
1573  *
1574  * @param coef array of dequantized, scaled spectral data
1575  * @param sf array of scalefactors or intensity stereo positions
1576  * @param pulse_present set if pulses are present
1577  * @param pulse pointer to pulse data struct
1578  * @param band_type array of the used band type
1579  *
1580  * @return Returns error status. 0 - OK, !0 - error
1581  */
1582 static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024],
1583  GetBitContext *gb, const INTFLOAT sf[120],
1584  int pulse_present, const Pulse *pulse,
1585  const IndividualChannelStream *ics,
1586  enum BandType band_type[120])
1587 {
1588  int i, k, g, idx = 0;
1589  const int c = 1024 / ics->num_windows;
1590  const uint16_t *offsets = ics->swb_offset;
1591  INTFLOAT *coef_base = coef;
1592 
1593  for (g = 0; g < ics->num_windows; g++)
1594  memset(coef + g * 128 + offsets[ics->max_sfb], 0,
1595  sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
1596 
1597  for (g = 0; g < ics->num_window_groups; g++) {
1598  unsigned g_len = ics->group_len[g];
1599 
1600  for (i = 0; i < ics->max_sfb; i++, idx++) {
1601  const unsigned cbt_m1 = band_type[idx] - 1;
1602  INTFLOAT *cfo = coef + offsets[i];
1603  int off_len = offsets[i + 1] - offsets[i];
1604  int group;
1605 
1606  if (cbt_m1 >= INTENSITY_BT2 - 1) {
1607  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1608  memset(cfo, 0, off_len * sizeof(*cfo));
1609  }
1610  } else if (cbt_m1 == NOISE_BT - 1) {
1611  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1612 #if !USE_FIXED
1613  float scale;
1614 #endif /* !USE_FIXED */
1615  INTFLOAT band_energy;
1616 
1617  for (k = 0; k < off_len; k++) {
1618  ac->random_state = lcg_random(ac->random_state);
1619 #if USE_FIXED
1620  cfo[k] = ac->random_state >> 3;
1621 #else
1622  cfo[k] = ac->random_state;
1623 #endif /* USE_FIXED */
1624  }
1625 
1626 #if USE_FIXED
1627  band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
1628  band_energy = fixed_sqrt(band_energy, 31);
1629  noise_scale(cfo, sf[idx], band_energy, off_len);
1630 #else
1631  band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
1632  scale = sf[idx] / sqrtf(band_energy);
1633  ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
1634 #endif /* USE_FIXED */
1635  }
1636  } else {
1637 #if !USE_FIXED
1638  const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1639 #endif /* !USE_FIXED */
1640  const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1641  VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1642  OPEN_READER(re, gb);
1643 
1644  switch (cbt_m1 >> 1) {
1645  case 0:
1646  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1647  INTFLOAT *cf = cfo;
1648  int len = off_len;
1649 
1650  do {
1651  int code;
1652  unsigned cb_idx;
1653 
1654  UPDATE_CACHE(re, gb);
1655  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1656  cb_idx = cb_vector_idx[code];
1657 #if USE_FIXED
1658  cf = DEC_SQUAD(cf, cb_idx);
1659 #else
1660  cf = VMUL4(cf, vq, cb_idx, sf + idx);
1661 #endif /* USE_FIXED */
1662  } while (len -= 4);
1663  }
1664  break;
1665 
1666  case 1:
1667  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1668  INTFLOAT *cf = cfo;
1669  int len = off_len;
1670 
1671  do {
1672  int code;
1673  unsigned nnz;
1674  unsigned cb_idx;
1675  uint32_t bits;
1676 
1677  UPDATE_CACHE(re, gb);
1678  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1679  cb_idx = cb_vector_idx[code];
1680  nnz = cb_idx >> 8 & 15;
1681  bits = nnz ? GET_CACHE(re, gb) : 0;
1682  LAST_SKIP_BITS(re, gb, nnz);
1683 #if USE_FIXED
1684  cf = DEC_UQUAD(cf, cb_idx, bits);
1685 #else
1686  cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1687 #endif /* USE_FIXED */
1688  } while (len -= 4);
1689  }
1690  break;
1691 
1692  case 2:
1693  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1694  INTFLOAT *cf = cfo;
1695  int len = off_len;
1696 
1697  do {
1698  int code;
1699  unsigned cb_idx;
1700 
1701  UPDATE_CACHE(re, gb);
1702  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1703  cb_idx = cb_vector_idx[code];
1704 #if USE_FIXED
1705  cf = DEC_SPAIR(cf, cb_idx);
1706 #else
1707  cf = VMUL2(cf, vq, cb_idx, sf + idx);
1708 #endif /* USE_FIXED */
1709  } while (len -= 2);
1710  }
1711  break;
1712 
1713  case 3:
1714  case 4:
1715  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1716  INTFLOAT *cf = cfo;
1717  int len = off_len;
1718 
1719  do {
1720  int code;
1721  unsigned nnz;
1722  unsigned cb_idx;
1723  unsigned sign;
1724 
1725  UPDATE_CACHE(re, gb);
1726  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1727  cb_idx = cb_vector_idx[code];
1728  nnz = cb_idx >> 8 & 15;
1729  sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
1730  LAST_SKIP_BITS(re, gb, nnz);
1731 #if USE_FIXED
1732  cf = DEC_UPAIR(cf, cb_idx, sign);
1733 #else
1734  cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1735 #endif /* USE_FIXED */
1736  } while (len -= 2);
1737  }
1738  break;
1739 
1740  default:
1741  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1742 #if USE_FIXED
1743  int *icf = cfo;
1744  int v;
1745 #else
1746  float *cf = cfo;
1747  uint32_t *icf = (uint32_t *) cf;
1748 #endif /* USE_FIXED */
1749  int len = off_len;
1750 
1751  do {
1752  int code;
1753  unsigned nzt, nnz;
1754  unsigned cb_idx;
1755  uint32_t bits;
1756  int j;
1757 
1758  UPDATE_CACHE(re, gb);
1759  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1760 
1761  if (!code) {
1762  *icf++ = 0;
1763  *icf++ = 0;
1764  continue;
1765  }
1766 
1767  cb_idx = cb_vector_idx[code];
1768  nnz = cb_idx >> 12;
1769  nzt = cb_idx >> 8;
1770  bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1771  LAST_SKIP_BITS(re, gb, nnz);
1772 
1773  for (j = 0; j < 2; j++) {
1774  if (nzt & 1<<j) {
1775  uint32_t b;
1776  int n;
1777  /* The total length of escape_sequence must be < 22 bits according
1778  to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1779  UPDATE_CACHE(re, gb);
1780  b = GET_CACHE(re, gb);
1781  b = 31 - av_log2(~b);
1782 
1783  if (b > 8) {
1784  av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1785  return AVERROR_INVALIDDATA;
1786  }
1787 
1788  SKIP_BITS(re, gb, b + 1);
1789  b += 4;
1790  n = (1 << b) + SHOW_UBITS(re, gb, b);
1791  LAST_SKIP_BITS(re, gb, b);
1792 #if USE_FIXED
1793  v = n;
1794  if (bits & 1U<<31)
1795  v = -v;
1796  *icf++ = v;
1797 #else
1798  *icf++ = ff_cbrt_tab[n] | (bits & 1U<<31);
1799 #endif /* USE_FIXED */
1800  bits <<= 1;
1801  } else {
1802 #if USE_FIXED
1803  v = cb_idx & 15;
1804  if (bits & 1U<<31)
1805  v = -v;
1806  *icf++ = v;
1807 #else
1808  unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1809  *icf++ = (bits & 1U<<31) | v;
1810 #endif /* USE_FIXED */
1811  bits <<= !!v;
1812  }
1813  cb_idx >>= 4;
1814  }
1815  } while (len -= 2);
1816 #if !USE_FIXED
1817  ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1818 #endif /* !USE_FIXED */
1819  }
1820  }
1821 
1822  CLOSE_READER(re, gb);
1823  }
1824  }
1825  coef += g_len << 7;
1826  }
1827 
1828  if (pulse_present) {
1829  idx = 0;
1830  for (i = 0; i < pulse->num_pulse; i++) {
1831  INTFLOAT co = coef_base[ pulse->pos[i] ];
1832  while (offsets[idx + 1] <= pulse->pos[i])
1833  idx++;
1834  if (band_type[idx] != NOISE_BT && sf[idx]) {
1835  INTFLOAT ico = -pulse->amp[i];
1836 #if USE_FIXED
1837  if (co) {
1838  ico = co + (co > 0 ? -ico : ico);
1839  }
1840  coef_base[ pulse->pos[i] ] = ico;
1841 #else
1842  if (co) {
1843  co /= sf[idx];
1844  ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1845  }
1846  coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1847 #endif /* USE_FIXED */
1848  }
1849  }
1850  }
1851 #if USE_FIXED
1852  coef = coef_base;
1853  idx = 0;
1854  for (g = 0; g < ics->num_window_groups; g++) {
1855  unsigned g_len = ics->group_len[g];
1856 
1857  for (i = 0; i < ics->max_sfb; i++, idx++) {
1858  const unsigned cbt_m1 = band_type[idx] - 1;
1859  int *cfo = coef + offsets[i];
1860  int off_len = offsets[i + 1] - offsets[i];
1861  int group;
1862 
1863  if (cbt_m1 < NOISE_BT - 1) {
1864  for (group = 0; group < (int)g_len; group++, cfo+=128) {
1865  ac->vector_pow43(cfo, off_len);
1866  ac->subband_scale(cfo, cfo, sf[idx], 34, off_len);
1867  }
1868  }
1869  }
1870  coef += g_len << 7;
1871  }
1872 #endif /* USE_FIXED */
1873  return 0;
1874 }
1875 
1876 /**
1877  * Apply AAC-Main style frequency domain prediction.
1878  */
1879 static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1880 {
1881  int sfb, k;
1882 
1883  if (!sce->ics.predictor_initialized) {
1884  reset_all_predictors(sce->predictor_state);
1885  sce->ics.predictor_initialized = 1;
1886  }
1887 
1888  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1889  for (sfb = 0;
1890  sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
1891  sfb++) {
1892  for (k = sce->ics.swb_offset[sfb];
1893  k < sce->ics.swb_offset[sfb + 1];
1894  k++) {
1895  predict(&sce->predictor_state[k], &sce->coeffs[k],
1896  sce->ics.predictor_present &&
1897  sce->ics.prediction_used[sfb]);
1898  }
1899  }
1900  if (sce->ics.predictor_reset_group)
1901  reset_predictor_group(sce->predictor_state,
1902  sce->ics.predictor_reset_group);
1903  } else
1904  reset_all_predictors(sce->predictor_state);
1905 }
1906 
1907 /**
1908  * Decode an individual_channel_stream payload; reference: table 4.44.
1909  *
1910  * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
1911  * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1912  *
1913  * @return Returns error status. 0 - OK, !0 - error
1914  */
1915 static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1916  GetBitContext *gb, int common_window, int scale_flag)
1917 {
1918  Pulse pulse;
1919  TemporalNoiseShaping *tns = &sce->tns;
1920  IndividualChannelStream *ics = &sce->ics;
1921  INTFLOAT *out = sce->coeffs;
1922  int global_gain, eld_syntax, er_syntax, pulse_present = 0;
1923  int ret;
1924 
1925  eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1926  er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
1927  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
1928  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
1929  ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1930 
1931  /* This assignment is to silence a GCC warning about the variable being used
1932  * uninitialized when in fact it always is.
1933  */
1934  pulse.num_pulse = 0;
1935 
1936  global_gain = get_bits(gb, 8);
1937 
1938  if (!common_window && !scale_flag) {
1939  if (decode_ics_info(ac, ics, gb) < 0)
1940  return AVERROR_INVALIDDATA;
1941  }
1942 
1943  if ((ret = decode_band_types(ac, sce->band_type,
1944  sce->band_type_run_end, gb, ics)) < 0)
1945  return ret;
1946  if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
1947  sce->band_type, sce->band_type_run_end)) < 0)
1948  return ret;
1949 
1950  pulse_present = 0;
1951  if (!scale_flag) {
1952  if (!eld_syntax && (pulse_present = get_bits1(gb))) {
1953  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1954  av_log(ac->avctx, AV_LOG_ERROR,
1955  "Pulse tool not allowed in eight short sequence.\n");
1956  return AVERROR_INVALIDDATA;
1957  }
1958  if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1959  av_log(ac->avctx, AV_LOG_ERROR,
1960  "Pulse data corrupt or invalid.\n");
1961  return AVERROR_INVALIDDATA;
1962  }
1963  }
1964  tns->present = get_bits1(gb);
1965  if (tns->present && !er_syntax)
1966  if (decode_tns(ac, tns, gb, ics) < 0)
1967  return AVERROR_INVALIDDATA;
1968  if (!eld_syntax && get_bits1(gb)) {
1969  avpriv_request_sample(ac->avctx, "SSR");
1970  return AVERROR_PATCHWELCOME;
1971  }
1972  // I see no textual basis in the spec for this occurring after SSR gain
1973  // control, but this is what both reference and real implmentations do
1974  if (tns->present && er_syntax)
1975  if (decode_tns(ac, tns, gb, ics) < 0)
1976  return AVERROR_INVALIDDATA;
1977  }
1978 
1979  if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
1980  &pulse, ics, sce->band_type) < 0)
1981  return AVERROR_INVALIDDATA;
1982 
1983  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
1984  apply_prediction(ac, sce);
1985 
1986  return 0;
1987 }
1988 
1989 /**
1990  * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1991  */
1992 static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1993 {
1994  const IndividualChannelStream *ics = &cpe->ch[0].ics;
1995  INTFLOAT *ch0 = cpe->ch[0].coeffs;
1996  INTFLOAT *ch1 = cpe->ch[1].coeffs;
1997  int g, i, group, idx = 0;
1998  const uint16_t *offsets = ics->swb_offset;
1999  for (g = 0; g < ics->num_window_groups; g++) {
2000  for (i = 0; i < ics->max_sfb; i++, idx++) {
2001  if (cpe->ms_mask[idx] &&
2002  cpe->ch[0].band_type[idx] < NOISE_BT &&
2003  cpe->ch[1].band_type[idx] < NOISE_BT) {
2004 #if USE_FIXED
2005  for (group = 0; group < ics->group_len[g]; group++) {
2006  ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
2007  ch1 + group * 128 + offsets[i],
2008  offsets[i+1] - offsets[i]);
2009 #else
2010  for (group = 0; group < ics->group_len[g]; group++) {
2011  ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
2012  ch1 + group * 128 + offsets[i],
2013  offsets[i+1] - offsets[i]);
2014 #endif /* USE_FIXED */
2015  }
2016  }
2017  }
2018  ch0 += ics->group_len[g] * 128;
2019  ch1 += ics->group_len[g] * 128;
2020  }
2021 }
2022 
2023 /**
2024  * intensity stereo decoding; reference: 4.6.8.2.3
2025  *
2026  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
2027  * [1] mask is decoded from bitstream; [2] mask is all 1s;
2028  * [3] reserved for scalable AAC
2029  */
2030 static void apply_intensity_stereo(AACContext *ac,
2031  ChannelElement *cpe, int ms_present)
2032 {
2033  const IndividualChannelStream *ics = &cpe->ch[1].ics;
2034  SingleChannelElement *sce1 = &cpe->ch[1];
2035  INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
2036  const uint16_t *offsets = ics->swb_offset;
2037  int g, group, i, idx = 0;
2038  int c;
2039  INTFLOAT scale;
2040  for (g = 0; g < ics->num_window_groups; g++) {
2041  for (i = 0; i < ics->max_sfb;) {
2042  if (sce1->band_type[idx] == INTENSITY_BT ||
2043  sce1->band_type[idx] == INTENSITY_BT2) {
2044  const int bt_run_end = sce1->band_type_run_end[idx];
2045  for (; i < bt_run_end; i++, idx++) {
2046  c = -1 + 2 * (sce1->band_type[idx] - 14);
2047  if (ms_present)
2048  c *= 1 - 2 * cpe->ms_mask[idx];
2049  scale = c * sce1->sf[idx];
2050  for (group = 0; group < ics->group_len[g]; group++)
2051 #if USE_FIXED
2052  ac->subband_scale(coef1 + group * 128 + offsets[i],
2053  coef0 + group * 128 + offsets[i],
2054  scale,
2055  23,
2056  offsets[i + 1] - offsets[i]);
2057 #else
2058  ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
2059  coef0 + group * 128 + offsets[i],
2060  scale,
2061  offsets[i + 1] - offsets[i]);
2062 #endif /* USE_FIXED */
2063  }
2064  } else {
2065  int bt_run_end = sce1->band_type_run_end[idx];
2066  idx += bt_run_end - i;
2067  i = bt_run_end;
2068  }
2069  }
2070  coef0 += ics->group_len[g] * 128;
2071  coef1 += ics->group_len[g] * 128;
2072  }
2073 }
2074 
2075 /**
2076  * Decode a channel_pair_element; reference: table 4.4.
2077  *
2078  * @return Returns error status. 0 - OK, !0 - error
2079  */
2080 static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
2081 {
2082  int i, ret, common_window, ms_present = 0;
2083  int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2084 
2085  common_window = eld_syntax || get_bits1(gb);
2086  if (common_window) {
2087  if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
2088  return AVERROR_INVALIDDATA;
2089  i = cpe->ch[1].ics.use_kb_window[0];
2090  cpe->ch[1].ics = cpe->ch[0].ics;
2091  cpe->ch[1].ics.use_kb_window[1] = i;
2092  if (cpe->ch[1].ics.predictor_present &&
2093  (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
2094  if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
2095  decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
2096  ms_present = get_bits(gb, 2);
2097  if (ms_present == 3) {
2098  av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
2099  return AVERROR_INVALIDDATA;
2100  } else if (ms_present)
2101  decode_mid_side_stereo(cpe, gb, ms_present);
2102  }
2103  if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
2104  return ret;
2105  if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
2106  return ret;
2107 
2108  if (common_window) {
2109  if (ms_present)
2110  apply_mid_side_stereo(ac, cpe);
2111  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
2112  apply_prediction(ac, &cpe->ch[0]);
2113  apply_prediction(ac, &cpe->ch[1]);
2114  }
2115  }
2116 
2117  apply_intensity_stereo(ac, cpe, ms_present);
2118  return 0;
2119 }
2120 
2121 static const float cce_scale[] = {
2122  1.09050773266525765921, //2^(1/8)
2123  1.18920711500272106672, //2^(1/4)
2124  M_SQRT2,
2125  2,
2126 };
2127 
2128 /**
2129  * Decode coupling_channel_element; reference: table 4.8.
2130  *
2131  * @return Returns error status. 0 - OK, !0 - error
2132  */
2133 static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
2134 {
2135  int num_gain = 0;
2136  int c, g, sfb, ret;
2137  int sign;
2138  INTFLOAT scale;
2139  SingleChannelElement *sce = &che->ch[0];
2140  ChannelCoupling *coup = &che->coup;
2141 
2142  coup->coupling_point = 2 * get_bits1(gb);
2143  coup->num_coupled = get_bits(gb, 3);
2144  for (c = 0; c <= coup->num_coupled; c++) {
2145  num_gain++;
2146  coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
2147  coup->id_select[c] = get_bits(gb, 4);
2148  if (coup->type[c] == TYPE_CPE) {
2149  coup->ch_select[c] = get_bits(gb, 2);
2150  if (coup->ch_select[c] == 3)
2151  num_gain++;
2152  } else
2153  coup->ch_select[c] = 2;
2154  }
2155  coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
2156 
2157  sign = get_bits(gb, 1);
2158  scale = AAC_RENAME(cce_scale)[get_bits(gb, 2)];
2159 
2160  if ((ret = decode_ics(ac, sce, gb, 0, 0)))
2161  return ret;
2162 
2163  for (c = 0; c < num_gain; c++) {
2164  int idx = 0;
2165  int cge = 1;
2166  int gain = 0;
2167  INTFLOAT gain_cache = FIXR10(1.);
2168  if (c) {
2169  cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
2170  gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
2171  gain_cache = GET_GAIN(scale, gain);
2172  }
2173  if (coup->coupling_point == AFTER_IMDCT) {
2174  coup->gain[c][0] = gain_cache;
2175  } else {
2176  for (g = 0; g < sce->ics.num_window_groups; g++) {
2177  for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
2178  if (sce->band_type[idx] != ZERO_BT) {
2179  if (!cge) {
2180  int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
2181  if (t) {
2182  int s = 1;
2183  t = gain += t;
2184  if (sign) {
2185  s -= 2 * (t & 0x1);
2186  t >>= 1;
2187  }
2188  gain_cache = GET_GAIN(scale, t) * s;
2189  }
2190  }
2191  coup->gain[c][idx] = gain_cache;
2192  }
2193  }
2194  }
2195  }
2196  }
2197  return 0;
2198 }
2199 
2200 /**
2201  * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
2202  *
2203  * @return Returns number of bytes consumed.
2204  */
2205 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
2206  GetBitContext *gb)
2207 {
2208  int i;
2209  int num_excl_chan = 0;
2210 
2211  do {
2212  for (i = 0; i < 7; i++)
2213  che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
2214  } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
2215 
2216  return num_excl_chan / 7;
2217 }
2218 
2219 /**
2220  * Decode dynamic range information; reference: table 4.52.
2221  *
2222  * @return Returns number of bytes consumed.
2223  */
2224 static int decode_dynamic_range(DynamicRangeControl *che_drc,
2225  GetBitContext *gb)
2226 {
2227  int n = 1;
2228  int drc_num_bands = 1;
2229  int i;
2230 
2231  /* pce_tag_present? */
2232  if (get_bits1(gb)) {
2233  che_drc->pce_instance_tag = get_bits(gb, 4);
2234  skip_bits(gb, 4); // tag_reserved_bits
2235  n++;
2236  }
2237 
2238  /* excluded_chns_present? */
2239  if (get_bits1(gb)) {
2240  n += decode_drc_channel_exclusions(che_drc, gb);
2241  }
2242 
2243  /* drc_bands_present? */
2244  if (get_bits1(gb)) {
2245  che_drc->band_incr = get_bits(gb, 4);
2246  che_drc->interpolation_scheme = get_bits(gb, 4);
2247  n++;
2248  drc_num_bands += che_drc->band_incr;
2249  for (i = 0; i < drc_num_bands; i++) {
2250  che_drc->band_top[i] = get_bits(gb, 8);
2251  n++;
2252  }
2253  }
2254 
2255  /* prog_ref_level_present? */
2256  if (get_bits1(gb)) {
2257  che_drc->prog_ref_level = get_bits(gb, 7);
2258  skip_bits1(gb); // prog_ref_level_reserved_bits
2259  n++;
2260  }
2261 
2262  for (i = 0; i < drc_num_bands; i++) {
2263  che_drc->dyn_rng_sgn[i] = get_bits1(gb);
2264  che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
2265  n++;
2266  }
2267 
2268  return n;
2269 }
2270 
2271 static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
2272  uint8_t buf[256];
2273  int i, major, minor;
2274 
2275  if (len < 13+7*8)
2276  goto unknown;
2277 
2278  get_bits(gb, 13); len -= 13;
2279 
2280  for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
2281  buf[i] = get_bits(gb, 8);
2282 
2283  buf[i] = 0;
2284  if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
2285  av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
2286 
2287  if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
2288  ac->avctx->internal->skip_samples = 1024;
2289  }
2290 
2291 unknown:
2292  skip_bits_long(gb, len);
2293 
2294  return 0;
2295 }
2296 
2297 /**
2298  * Decode extension data (incomplete); reference: table 4.51.
2299  *
2300  * @param cnt length of TYPE_FIL syntactic element in bytes
2301  *
2302  * @return Returns number of bytes consumed
2303  */
2304 static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
2305  ChannelElement *che, enum RawDataBlockType elem_type)
2306 {
2307  int crc_flag = 0;
2308  int res = cnt;
2309  int type = get_bits(gb, 4);
2310 
2311  if (ac->avctx->debug & FF_DEBUG_STARTCODE)
2312  av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
2313 
2314  switch (type) { // extension type
2315  case EXT_SBR_DATA_CRC:
2316  crc_flag++;
2317  case EXT_SBR_DATA:
2318  if (!che) {
2319  av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
2320  return res;
2321  } else if (!ac->oc[1].m4ac.sbr) {
2322  av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
2323  skip_bits_long(gb, 8 * cnt - 4);
2324  return res;
2325  } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
2326  av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
2327  skip_bits_long(gb, 8 * cnt - 4);
2328  return res;
2329  } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
2330  ac->oc[1].m4ac.sbr = 1;
2331  ac->oc[1].m4ac.ps = 1;
2332  ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
2333  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
2334  ac->oc[1].status, 1);
2335  } else {
2336  ac->oc[1].m4ac.sbr = 1;
2337  ac->avctx->profile = FF_PROFILE_AAC_HE;
2338  }
2339  res = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
2340  break;
2341  case EXT_DYNAMIC_RANGE:
2342  res = decode_dynamic_range(&ac->che_drc, gb);
2343  break;
2344  case EXT_FILL:
2345  decode_fill(ac, gb, 8 * cnt - 4);
2346  break;
2347  case EXT_FILL_DATA:
2348  case EXT_DATA_ELEMENT:
2349  default:
2350  skip_bits_long(gb, 8 * cnt - 4);
2351  break;
2352  };
2353  return res;
2354 }
2355 
2356 /**
2357  * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
2358  *
2359  * @param decode 1 if tool is used normally, 0 if tool is used in LTP.
2360  * @param coef spectral coefficients
2361  */
2362 static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
2363  IndividualChannelStream *ics, int decode)
2364 {
2365  const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
2366  int w, filt, m, i;
2367  int bottom, top, order, start, end, size, inc;
2368  INTFLOAT lpc[TNS_MAX_ORDER];
2369  INTFLOAT tmp[TNS_MAX_ORDER+1];
2370 
2371  for (w = 0; w < ics->num_windows; w++) {
2372  bottom = ics->num_swb;
2373  for (filt = 0; filt < tns->n_filt[w]; filt++) {
2374  top = bottom;
2375  bottom = FFMAX(0, top - tns->length[w][filt]);
2376  order = tns->order[w][filt];
2377  if (order == 0)
2378  continue;
2379 
2380  // tns_decode_coef
2381  AAC_RENAME(compute_lpc_coefs)(tns->coef[w][filt], order, lpc, 0, 0, 0);
2382 
2383  start = ics->swb_offset[FFMIN(bottom, mmm)];
2384  end = ics->swb_offset[FFMIN( top, mmm)];
2385  if ((size = end - start) <= 0)
2386  continue;
2387  if (tns->direction[w][filt]) {
2388  inc = -1;
2389  start = end - 1;
2390  } else {
2391  inc = 1;
2392  }
2393  start += w * 128;
2394 
2395  if (decode) {
2396  // ar filter
2397  for (m = 0; m < size; m++, start += inc)
2398  for (i = 1; i <= FFMIN(m, order); i++)
2399  coef[start] -= AAC_MUL26(coef[start - i * inc], lpc[i - 1]);
2400  } else {
2401  // ma filter
2402  for (m = 0; m < size; m++, start += inc) {
2403  tmp[0] = coef[start];
2404  for (i = 1; i <= FFMIN(m, order); i++)
2405  coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
2406  for (i = order; i > 0; i--)
2407  tmp[i] = tmp[i - 1];
2408  }
2409  }
2410  }
2411  }
2412 }
2413 
2414 /**
2415  * Apply windowing and MDCT to obtain the spectral
2416  * coefficient from the predicted sample by LTP.
2417  */
2418 static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out,
2419  INTFLOAT *in, IndividualChannelStream *ics)
2420 {
2421  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2422  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2423  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2424  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2425 
2426  if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
2427  ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
2428  } else {
2429  memset(in, 0, 448 * sizeof(*in));
2430  ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
2431  }
2432  if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
2433  ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
2434  } else {
2435  ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
2436  memset(in + 1024 + 576, 0, 448 * sizeof(*in));
2437  }
2438  ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
2439 }
2440 
2441 /**
2442  * Apply the long term prediction
2443  */
2444 static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
2445 {
2446  const LongTermPrediction *ltp = &sce->ics.ltp;
2447  const uint16_t *offsets = sce->ics.swb_offset;
2448  int i, sfb;
2449 
2450  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
2451  INTFLOAT *predTime = sce->ret;
2452  INTFLOAT *predFreq = ac->buf_mdct;
2453  int16_t num_samples = 2048;
2454 
2455  if (ltp->lag < 1024)
2456  num_samples = ltp->lag + 1024;
2457  for (i = 0; i < num_samples; i++)
2458  predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
2459  memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));
2460 
2461  ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
2462 
2463  if (sce->tns.present)
2464  ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
2465 
2466  for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
2467  if (ltp->used[sfb])
2468  for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
2469  sce->coeffs[i] += predFreq[i];
2470  }
2471 }
2472 
2473 /**
2474  * Update the LTP buffer for next frame
2475  */
2476 static void update_ltp(AACContext *ac, SingleChannelElement *sce)
2477 {
2478  IndividualChannelStream *ics = &sce->ics;
2479  INTFLOAT *saved = sce->saved;
2480  INTFLOAT *saved_ltp = sce->coeffs;
2481  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2482  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2483  int i;
2484 
2485  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2486  memcpy(saved_ltp, saved, 512 * sizeof(*saved_ltp));
2487  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2488  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2489 
2490  for (i = 0; i < 64; i++)
2491  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2492  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2493  memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(*saved_ltp));
2494  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2495  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2496 
2497  for (i = 0; i < 64; i++)
2498  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2499  } else { // LONG_STOP or ONLY_LONG
2500  ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
2501 
2502  for (i = 0; i < 512; i++)
2503  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
2504  }
2505 
2506  memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
2507  memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
2508  memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
2509 }
2510 
2511 /**
2512  * Conduct IMDCT and windowing.
2513  */
2514 static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
2515 {
2516  IndividualChannelStream *ics = &sce->ics;
2517  INTFLOAT *in = sce->coeffs;
2518  INTFLOAT *out = sce->ret;
2519  INTFLOAT *saved = sce->saved;
2520  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2521  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2522  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2523  INTFLOAT *buf = ac->buf_mdct;
2524  INTFLOAT *temp = ac->temp;
2525  int i;
2526 
2527  // imdct
2528  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2529  for (i = 0; i < 1024; i += 128)
2530  ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
2531  } else {
2532  ac->mdct.imdct_half(&ac->mdct, buf, in);
2533 #if USE_FIXED
2534  for (i=0; i<1024; i++)
2535  buf[i] = (buf[i] + 4) >> 3;
2536 #endif /* USE_FIXED */
2537  }
2538 
2539  /* window overlapping
2540  * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2541  * and long to short transitions are considered to be short to short
2542  * transitions. This leaves just two cases (long to long and short to short)
2543  * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2544  */
2545  if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2546  (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
2547  ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
2548  } else {
2549  memcpy( out, saved, 448 * sizeof(*out));
2550 
2551  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2552  ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
2553  ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
2554  ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
2555  ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
2556  ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
2557  memcpy( out + 448 + 4*128, temp, 64 * sizeof(*out));
2558  } else {
2559  ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
2560  memcpy( out + 576, buf + 64, 448 * sizeof(*out));
2561  }
2562  }
2563 
2564  // buffer update
2565  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2566  memcpy( saved, temp + 64, 64 * sizeof(*saved));
2567  ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
2568  ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
2569  ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
2570  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2571  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2572  memcpy( saved, buf + 512, 448 * sizeof(*saved));
2573  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2574  } else { // LONG_STOP or ONLY_LONG
2575  memcpy( saved, buf + 512, 512 * sizeof(*saved));
2576  }
2577 }
2578 
2579 static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
2580 {
2581  IndividualChannelStream *ics = &sce->ics;
2582  INTFLOAT *in = sce->coeffs;
2583  INTFLOAT *out = sce->ret;
2584  INTFLOAT *saved = sce->saved;
2585  INTFLOAT *buf = ac->buf_mdct;
2586 #if USE_FIXED
2587  int i;
2588 #endif /* USE_FIXED */
2589 
2590  // imdct
2591  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2592 
2593 #if USE_FIXED
2594  for (i = 0; i < 1024; i++)
2595  buf[i] = (buf[i] + 2) >> 2;
2596 #endif /* USE_FIXED */
2597 
2598  // window overlapping
2599  if (ics->use_kb_window[1]) {
2600  // AAC LD uses a low overlap sine window instead of a KBD window
2601  memcpy(out, saved, 192 * sizeof(*out));
2602  ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
2603  memcpy( out + 320, buf + 64, 192 * sizeof(*out));
2604  } else {
2605  ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);
2606  }
2607 
2608  // buffer update
2609  memcpy(saved, buf + 256, 256 * sizeof(*saved));
2610 }
2611 
2612 static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
2613 {
2614  INTFLOAT *in = sce->coeffs;
2615  INTFLOAT *out = sce->ret;
2616  INTFLOAT *saved = sce->saved;
2617  INTFLOAT *buf = ac->buf_mdct;
2618  int i;
2619  const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
2620  const int n2 = n >> 1;
2621  const int n4 = n >> 2;
2622  const INTFLOAT *const window = n == 480 ? AAC_RENAME(ff_aac_eld_window_480) :
2623  AAC_RENAME(ff_aac_eld_window_512);
2624 
2625  // Inverse transform, mapped to the conventional IMDCT by
2626  // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
2627  // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
2628  // International Conference on Audio, Language and Image Processing, ICALIP 2008.
2629  // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
2630  for (i = 0; i < n2; i+=2) {
2631  INTFLOAT temp;
2632  temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
2633  temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
2634  }
2635 #if !USE_FIXED
2636  if (n == 480)
2637  ac->mdct480->imdct_half(ac->mdct480, buf, in, 1, -1.f/(16*1024*960));
2638  else
2639 #endif
2640  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2641 
2642 #if USE_FIXED
2643  for (i = 0; i < 1024; i++)
2644  buf[i] = (buf[i] + 1) >> 1;
2645 #endif /* USE_FIXED */
2646 
2647  for (i = 0; i < n; i+=2) {
2648  buf[i] = -buf[i];
2649  }
2650  // Like with the regular IMDCT at this point we still have the middle half
2651  // of a transform but with even symmetry on the left and odd symmetry on
2652  // the right
2653 
2654  // window overlapping
2655  // The spec says to use samples [0..511] but the reference decoder uses
2656  // samples [128..639].
2657  for (i = n4; i < n2; i ++) {
2658  out[i - n4] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) +
2659  AAC_MUL31( saved[ i + n2] , window[i + n - n4]) +
2660  AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
2661  AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]);
2662  }
2663  for (i = 0; i < n2; i ++) {
2664  out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) +
2665  AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) +
2666  AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) +
2667  AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
2668  }
2669  for (i = 0; i < n4; i ++) {
2670  out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) +
2671  AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
2672  AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
2673  }
2674 
2675  // buffer update
2676  memmove(saved + n, saved, 2 * n * sizeof(*saved));
2677  memcpy( saved, buf, n * sizeof(*saved));
2678 }
2679 
2680 /**
2681  * channel coupling transformation interface
2682  *
2683  * @param apply_coupling_method pointer to (in)dependent coupling function
2684  */
2685 static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
2686  enum RawDataBlockType type, int elem_id,
2687  enum CouplingPoint coupling_point,
2688  void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
2689 {
2690  int i, c;
2691 
2692  for (i = 0; i < MAX_ELEM_ID; i++) {
2693  ChannelElement *cce = ac->che[TYPE_CCE][i];
2694  int index = 0;
2695 
2696  if (cce && cce->coup.coupling_point == coupling_point) {
2697  ChannelCoupling *coup = &cce->coup;
2698 
2699  for (c = 0; c <= coup->num_coupled; c++) {
2700  if (coup->type[c] == type && coup->id_select[c] == elem_id) {
2701  if (coup->ch_select[c] != 1) {
2702  apply_coupling_method(ac, &cc->ch[0], cce, index);
2703  if (coup->ch_select[c] != 0)
2704  index++;
2705  }
2706  if (coup->ch_select[c] != 2)
2707  apply_coupling_method(ac, &cc->ch[1], cce, index++);
2708  } else
2709  index += 1 + (coup->ch_select[c] == 3);
2710  }
2711  }
2712  }
2713 }
2714 
2715 /**
2716  * Convert spectral data to samples, applying all supported tools as appropriate.
2717  */
2718 static void spectral_to_sample(AACContext *ac, int samples)
2719 {
2720  int i, type;
2721  void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
2722  switch (ac->oc[1].m4ac.object_type) {
2723  case AOT_ER_AAC_LD:
2724  imdct_and_window = imdct_and_windowing_ld;
2725  break;
2726  case AOT_ER_AAC_ELD:
2727  imdct_and_window = imdct_and_windowing_eld;
2728  break;
2729  default:
2730  imdct_and_window = ac->imdct_and_windowing;
2731  }
2732  for (type = 3; type >= 0; type--) {
2733  for (i = 0; i < MAX_ELEM_ID; i++) {
2734  ChannelElement *che = ac->che[type][i];
2735  if (che && che->present) {
2736  if (type <= TYPE_CPE)
2737  apply_channel_coupling(ac, che, type, i, BEFORE_TNS, AAC_RENAME(apply_dependent_coupling));
2738  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2739  if (che->ch[0].ics.predictor_present) {
2740  if (che->ch[0].ics.ltp.present)
2741  ac->apply_ltp(ac, &che->ch[0]);
2742  if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
2743  ac->apply_ltp(ac, &che->ch[1]);
2744  }
2745  }
2746  if (che->ch[0].tns.present)
2747  ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2748  if (che->ch[1].tns.present)
2749  ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2750  if (type <= TYPE_CPE)
2751  apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, AAC_RENAME(apply_dependent_coupling));
2752  if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2753  imdct_and_window(ac, &che->ch[0]);
2754  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2755  ac->update_ltp(ac, &che->ch[0]);
2756  if (type == TYPE_CPE) {
2757  imdct_and_window(ac, &che->ch[1]);
2758  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2759  ac->update_ltp(ac, &che->ch[1]);
2760  }
2761  if (ac->oc[1].m4ac.sbr > 0) {
2762  AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2763  }
2764  }
2765  if (type <= TYPE_CCE)
2766  apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, AAC_RENAME(apply_independent_coupling));
2767 
2768 #if USE_FIXED
2769  {
2770  int j;
2771  /* preparation for resampler */
2772  for(j = 0; j<samples; j++){
2773  che->ch[0].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[0].ret[j]<<7)+0x8000;
2774  if(type == TYPE_CPE)
2775  che->ch[1].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[1].ret[j]<<7)+0x8000;
2776  }
2777  }
2778 #endif /* USE_FIXED */
2779  che->present = 0;
2780  } else if (che) {
2781  av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
2782  }
2783  }
2784  }
2785 }
2786 
2787 static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
2788 {
2789  int size;
2790  AACADTSHeaderInfo hdr_info;
2791  uint8_t layout_map[MAX_ELEM_ID*4][3];
2792  int layout_map_tags, ret;
2793 
2794  size = avpriv_aac_parse_header(gb, &hdr_info);
2795  if (size > 0) {
2796  if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
2797  // This is 2 for "VLB " audio in NSV files.
2798  // See samples/nsv/vlb_audio.
2799  avpriv_report_missing_feature(ac->avctx,
2800  "More than one AAC RDB per ADTS frame");
2801  ac->warned_num_aac_frames = 1;
2802  }
2803  push_output_configuration(ac);
2804  if (hdr_info.chan_config) {
2805  ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
2806  if ((ret = set_default_channel_config(ac->avctx,
2807  layout_map,
2808  &layout_map_tags,
2809  hdr_info.chan_config)) < 0)
2810  return ret;
2811  if ((ret = output_configure(ac, layout_map, layout_map_tags,
2812  FFMAX(ac->oc[1].status,
2813  OC_TRIAL_FRAME), 0)) < 0)
2814  return ret;
2815  } else {
2816  ac->oc[1].m4ac.chan_config = 0;
2817  /**
2818  * dual mono frames in Japanese DTV can have chan_config 0
2819  * WITHOUT specifying PCE.
2820  * thus, set dual mono as default.
2821  */
2822  if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
2823  layout_map_tags = 2;
2824  layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
2825  layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
2826  layout_map[0][1] = 0;
2827  layout_map[1][1] = 1;
2828  if (output_configure(ac, layout_map, layout_map_tags,
2829  OC_TRIAL_FRAME, 0))
2830  return -7;
2831  }
2832  }
2833  ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
2834  ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
2835  ac->oc[1].m4ac.object_type = hdr_info.object_type;
2836  ac->oc[1].m4ac.frame_length_short = 0;
2837  if (ac->oc[0].status != OC_LOCKED ||
2838  ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
2839  ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
2840  ac->oc[1].m4ac.sbr = -1;
2841  ac->oc[1].m4ac.ps = -1;
2842  }
2843  if (!hdr_info.crc_absent)
2844  skip_bits(gb, 16);
2845  }
2846  return size;
2847 }
2848 
2849 static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
2850  int *got_frame_ptr, GetBitContext *gb)
2851 {
2852  AACContext *ac = avctx->priv_data;
2853  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
2854  ChannelElement *che;
2855  int err, i;
2856  int samples = m4ac->frame_length_short ? 960 : 1024;
2857  int chan_config = m4ac->chan_config;
2858  int aot = m4ac->object_type;
2859 
2860  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
2861  samples >>= 1;
2862 
2863  ac->frame = data;
2864 
2865  if ((err = frame_configure_elements(avctx)) < 0)
2866  return err;
2867 
2868  // The FF_PROFILE_AAC_* defines are all object_type - 1
2869  // This may lead to an undefined profile being signaled
2870  ac->avctx->profile = aot - 1;
2871 
2872  ac->tags_mapped = 0;
2873 
2874  if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
2875  avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
2876  chan_config);
2877  return AVERROR_INVALIDDATA;
2878  }
2879  for (i = 0; i < tags_per_config[chan_config]; i++) {
2880  const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
2881  const int elem_id = aac_channel_layout_map[chan_config-1][i][1];
2882  if (!(che=get_che(ac, elem_type, elem_id))) {
2883  av_log(ac->avctx, AV_LOG_ERROR,
2884  "channel element %d.%d is not allocated\n",
2885  elem_type, elem_id);
2886  return AVERROR_INVALIDDATA;
2887  }
2888  che->present = 1;
2889  if (aot != AOT_ER_AAC_ELD)
2890  skip_bits(gb, 4);
2891  switch (elem_type) {
2892  case TYPE_SCE:
2893  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2894  break;
2895  case TYPE_CPE:
2896  err = decode_cpe(ac, gb, che);
2897  break;
2898  case TYPE_LFE:
2899  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2900  break;
2901  }
2902  if (err < 0)
2903  return err;
2904  }
2905 
2906  spectral_to_sample(ac, samples);
2907 
2908  if (!ac->frame->data[0] && samples) {
2909  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
2910  return AVERROR_INVALIDDATA;
2911  }
2912 
2913  ac->frame->nb_samples = samples;
2914  ac->frame->sample_rate = avctx->sample_rate;
2915  *got_frame_ptr = 1;
2916 
2917  skip_bits_long(gb, get_bits_left(gb));
2918  return 0;
2919 }
2920 
2921 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
2922  int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
2923 {
2924  AACContext *ac = avctx->priv_data;
2925  ChannelElement *che = NULL, *che_prev = NULL;
2926  enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
2927  int err, elem_id;
2928  int samples = 0, multiplier, audio_found = 0, pce_found = 0;
2929  int is_dmono, sce_count = 0;
2930 
2931  ac->frame = data;
2932 
2933  if (show_bits(gb, 12) == 0xfff) {
2934  if ((err = parse_adts_frame_header(ac, gb)) < 0) {
2935  av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
2936  goto fail;
2937  }
2938  if (ac->oc[1].m4ac.sampling_index > 12) {
2939  av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
2940  err = AVERROR_INVALIDDATA;
2941  goto fail;
2942  }
2943  }
2944 
2945  if ((err = frame_configure_elements(avctx)) < 0)
2946  goto fail;
2947 
2948  // The FF_PROFILE_AAC_* defines are all object_type - 1
2949  // This may lead to an undefined profile being signaled
2950  ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
2951 
2952  ac->tags_mapped = 0;
2953  // parse
2954  while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
2955  elem_id = get_bits(gb, 4);
2956 
2957  if (avctx->debug & FF_DEBUG_STARTCODE)
2958  av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
2959 
2960  if (!avctx->channels && elem_type != TYPE_PCE) {
2961  err = AVERROR_INVALIDDATA;
2962  goto fail;
2963  }
2964 
2965  if (elem_type < TYPE_DSE) {
2966  if (!(che=get_che(ac, elem_type, elem_id))) {
2967  av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
2968  elem_type, elem_id);
2969  err = AVERROR_INVALIDDATA;
2970  goto fail;
2971  }
2972  samples = 1024;
2973  che->present = 1;
2974  }
2975 
2976  switch (elem_type) {
2977 
2978  case TYPE_SCE:
2979  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2980  audio_found = 1;
2981  sce_count++;
2982  break;
2983 
2984  case TYPE_CPE:
2985  err = decode_cpe(ac, gb, che);
2986  audio_found = 1;
2987  break;
2988 
2989  case TYPE_CCE:
2990  err = decode_cce(ac, gb, che);
2991  break;
2992 
2993  case TYPE_LFE:
2994  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2995  audio_found = 1;
2996  break;
2997 
2998  case TYPE_DSE:
2999  err = skip_data_stream_element(ac, gb);
3000  break;
3001 
3002  case TYPE_PCE: {
3003  uint8_t layout_map[MAX_ELEM_ID*4][3];
3004  int tags;
3005  push_output_configuration(ac);
3006  tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);
3007  if (tags < 0) {
3008  err = tags;
3009  break;
3010  }
3011  if (pce_found) {
3012  av_log(avctx, AV_LOG_ERROR,
3013  "Not evaluating a further program_config_element as this construct is dubious at best.\n");
3014  } else {
3015  err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
3016  if (!err)
3017  ac->oc[1].m4ac.chan_config = 0;
3018  pce_found = 1;
3019  }
3020  break;
3021  }
3022 
3023  case TYPE_FIL:
3024  if (elem_id == 15)
3025  elem_id += get_bits(gb, 8) - 1;
3026  if (get_bits_left(gb) < 8 * elem_id) {
3027  av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
3028  err = AVERROR_INVALIDDATA;
3029  goto fail;
3030  }
3031  while (elem_id > 0)
3032  elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
3033  err = 0; /* FIXME */
3034  break;
3035 
3036  default:
3037  err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
3038  break;
3039  }
3040 
3041  che_prev = che;
3042  elem_type_prev = elem_type;
3043 
3044  if (err)
3045  goto fail;
3046 
3047  if (get_bits_left(gb) < 3) {
3048  av_log(avctx, AV_LOG_ERROR, overread_err);
3049  err = AVERROR_INVALIDDATA;
3050  goto fail;
3051  }
3052  }
3053 
3054  if (!avctx->channels) {
3055  *got_frame_ptr = 0;
3056  return 0;
3057  }
3058 
3059  multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
3060  samples <<= multiplier;
3061 
3062  spectral_to_sample(ac, samples);
3063 
3064  if (ac->oc[1].status && audio_found) {
3065  avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
3066  avctx->frame_size = samples;
3067  ac->oc[1].status = OC_LOCKED;
3068  }
3069 
3070  if (multiplier) {
3071  int side_size;
3072  const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
3073  if (side && side_size>=4)
3074  AV_WL32(side, 2*AV_RL32(side));
3075  }
3076 
3077  if (!ac->frame->data[0] && samples) {
3078  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
3079  err = AVERROR_INVALIDDATA;
3080  goto fail;
3081  }
3082 
3083  if (samples) {
3084  ac->frame->nb_samples = samples;
3085  ac->frame->sample_rate = avctx->sample_rate;
3086  } else
3087  av_frame_unref(ac->frame);
3088  *got_frame_ptr = !!samples;
3089 
3090  /* for dual-mono audio (SCE + SCE) */
3091  is_dmono = ac->dmono_mode && sce_count == 2 &&
3092  ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
3093  if (is_dmono) {
3094  if (ac->dmono_mode == 1)
3095  ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
3096  else if (ac->dmono_mode == 2)
3097  ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
3098  }
3099 
3100  return 0;
3101 fail:
3102  pop_output_configuration(ac);
3103  return err;
3104 }
3105 
3106 static int aac_decode_frame(AVCodecContext *avctx, void *data,
3107  int *got_frame_ptr, AVPacket *avpkt)
3108 {
3109  AACContext *ac = avctx->priv_data;
3110  const uint8_t *buf = avpkt->data;
3111  int buf_size = avpkt->size;
3112  GetBitContext gb;
3113  int buf_consumed;
3114  int buf_offset;
3115  int err;
3116  int new_extradata_size;
3117  const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
3118  AV_PKT_DATA_NEW_EXTRADATA,
3119  &new_extradata_size);
3120  int jp_dualmono_size;
3121  const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
3122  AV_PKT_DATA_JP_DUALMONO,
3123  &jp_dualmono_size);
3124 
3125  if (new_extradata && 0) {
3126  av_free(avctx->extradata);
3127  avctx->extradata = av_mallocz(new_extradata_size +
3128  AV_INPUT_BUFFER_PADDING_SIZE);
3129  if (!avctx->extradata)
3130  return AVERROR(ENOMEM);
3131  avctx->extradata_size = new_extradata_size;
3132  memcpy(avctx->extradata, new_extradata, new_extradata_size);
3133  push_output_configuration(ac);
3134  if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
3135  avctx->extradata,
3136  avctx->extradata_size*8LL, 1) < 0) {
3137  pop_output_configuration(ac);
3138  return AVERROR_INVALIDDATA;
3139  }
3140  }
3141 
3142  ac->dmono_mode = 0;
3143  if (jp_dualmono && jp_dualmono_size > 0)
3144  ac->dmono_mode = 1 + *jp_dualmono;
3145  if (ac->force_dmono_mode >= 0)
3146  ac->dmono_mode = ac->force_dmono_mode;
3147 
3148  if (INT_MAX / 8 <= buf_size)
3149  return AVERROR_INVALIDDATA;
3150 
3151  if ((err = init_get_bits8(&gb, buf, buf_size)) < 0)
3152  return err;
3153 
3154  switch (ac->oc[1].m4ac.object_type) {
3155  case AOT_ER_AAC_LC:
3156  case AOT_ER_AAC_LTP:
3157  case AOT_ER_AAC_LD:
3158  case AOT_ER_AAC_ELD:
3159  err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
3160  break;
3161  default:
3162  err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
3163  }
3164  if (err < 0)
3165  return err;
3166 
3167  buf_consumed = (get_bits_count(&gb) + 7) >> 3;
3168  for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
3169  if (buf[buf_offset])
3170  break;
3171 
3172  return buf_size > buf_offset ? buf_consumed : buf_size;
3173 }
3174 
3175 static av_cold int aac_decode_close(AVCodecContext *avctx)
3176 {
3177  AACContext *ac = avctx->priv_data;
3178  int i, type;
3179 
3180  for (i = 0; i < MAX_ELEM_ID; i++) {
3181  for (type = 0; type < 4; type++) {
3182  if (ac->che[type][i])
3183  AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][i]->sbr);
3184  av_freep(&ac->che[type][i]);
3185  }
3186  }
3187 
3188  ff_mdct_end(&ac->mdct);
3189  ff_mdct_end(&ac->mdct_small);
3190  ff_mdct_end(&ac->mdct_ld);
3191  ff_mdct_end(&ac->mdct_ltp);
3192 #if !USE_FIXED
3193  ff_imdct15_uninit(&ac->mdct480);
3194 #endif
3195  av_freep(&ac->fdsp);
3196  return 0;
3197 }
3198 
3199 static void aacdec_init(AACContext *c)
3200 {
3201  c->imdct_and_windowing = imdct_and_windowing;
3202  c->apply_ltp = apply_ltp;
3203  c->apply_tns = apply_tns;
3204  c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
3205  c->update_ltp = update_ltp;
3206 #if USE_FIXED
3207  c->vector_pow43 = vector_pow43;
3208  c->subband_scale = subband_scale;
3209 #endif
3210 
3211 #if !USE_FIXED
3212  if(ARCH_MIPS)
3213  ff_aacdec_init_mips(c);
3214 #endif /* !USE_FIXED */
3215 }
3216 /**
3217  * AVOptions for Japanese DTV specific extensions (ADTS only)
3218  */
3219 #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
3220 static const AVOption options[] = {
3221  {"dual_mono_mode", "Select the channel to decode for dual mono",
3222  offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
3223  AACDEC_FLAGS, "dual_mono_mode"},
3224 
3225  {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3226  {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3227  {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3228  {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3229 
3230  {NULL},
3231 };
3232 
3233 static const AVClass aac_decoder_class = {
3234  .class_name = "AAC decoder",
3235  .item_name = av_default_item_name,
3236  .option = options,
3237  .version = LIBAVUTIL_VERSION_INT,
3238 };
IndividualChannelStream::predictor_initialized
int predictor_initialized
Definition: aac.h:187
VMUL4S
static float * VMUL4S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
Definition: aacdec.c:124
AACContext::fdsp
AVFloatDSPContext * fdsp
Definition: aac.h:331
apply_prediction
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
Apply AAC-Main style frequency domain prediction.
Definition: aacdec_template.c:1879
AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
imdct_and_windowing
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
Conduct IMDCT and windowing.
Definition: aacdec_template.c:2514
apply_ltp
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
Apply the long term prediction.
Definition: aacdec_template.c:2444
NULL
#define NULL
Definition: coverity.c:32
s
const char * s
Definition: avisynth_c.h:768
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:723
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
TYPE_DSE
Definition: aac.h:60
elem_to_channel::elem_id
uint8_t elem_id
Definition: aacdec_template.c:195
decode_pulses
static int decode_pulses(Pulse *pulse, GetBitContext *gb, const uint16_t *swb_offset, int num_swb)
Decode pulse data; reference: table 4.7.
Definition: aacdec_template.c:1487
IndividualChannelStream::use_kb_window
uint8_t use_kb_window[2]
If set, use Kaiser-Bessel window, otherwise use a sine window.
Definition: aac.h:177
AACContext::buf_mdct
INTFLOAT buf_mdct[1024]
Definition: aac.h:316
AACContext::subband_scale
void(* subband_scale)(int *dst, int *src, int scale, int offset, int len)
Definition: aac.h:366
overread_err
#define overread_err
Definition: aacdec_template.c:101
AACContext::mdct480
IMDCT15Context * mdct480
Definition: aac.h:330
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
AACADTSHeaderInfo::object_type
uint8_t object_type
Definition: aacadtsdec.h:36
AVOption
AVOption.
Definition: opt.h:245
data
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
flush
static void flush(AVCodecContext *avctx)
Definition: aacdec_template.c:496
tags_per_config
static const int8_t tags_per_config[16]
Definition: aacdectab.h:38
AACContext::avctx
AVCodecContext * avctx
Definition: aac.h:295
Pulse
Definition: aac.h:224
id
enum AVCodecID id
Definition: mxfenc.c:104
FFTContext::mdct_calc
void(* mdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:109
DEC_UPAIR
static int * DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
Definition: aacdec_fixed.c:125
aac_table_init
static AVOnce aac_table_init
Definition: aacdec_template.c:1110
re
float re
Definition: fft.c:82
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:247
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
ff_kbd_window_init
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
Generate a Kaiser-Bessel Derived Window.
Definition: kbdwin.c:26
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
SCALE_DIFF_ZERO
#define SCALE_DIFF_ZERO
codebook index corresponding to zero scalefactor indices difference
Definition: aac.h:152
temp
else temp
Definition: vf_mcdeint.c:259
TYPE_END
Definition: aac.h:63
cce_scale
static const float cce_scale[]
Definition: aacdec_template.c:2121
skip_bits_long
static void skip_bits_long(GetBitContext *s, int n)
Definition: get_bits.h:204
g
const char * g
Definition: vf_curves.c:112
AV_OPT_TYPE_INT
Definition: opt.h:223
VMUL2S
static float * VMUL2S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
Definition: aacdec.c:107
AACDEC_FLAGS
#define AACDEC_FLAGS
AVOptions for Japanese DTV specific extensions (ADTS only)
Definition: aacdec_template.c:3219
imdct_and_windowing_eld
static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
Definition: aacdec_template.c:2612
elem_to_channel
Definition: aacdec_template.c:192
INIT_VLC_STATIC
#define INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
Definition: vlc.h:57
aacdec_init
static void aacdec_init(AACContext *ac)
Definition: aacdec_template.c:3199
FIXR10
#define FIXR10(x)
Definition: aac_defines.h:91
DEC_SQUAD
static int * DEC_SQUAD(int *dst, unsigned idx)
Definition: aacdec_fixed.c:115
TYPE_SCE
Definition: aac.h:56
TYPE_CPE
Definition: aac.h:57
AACContext::che
ChannelElement * che[4][MAX_ELEM_ID]
Definition: aac.h:305
AVPacket::size
int size
Definition: avcodec.h:1602
b
const char * b
Definition: vf_curves.c:113
AVFloatDSPContext::scalarproduct_float
float(* scalarproduct_float)(const float *v1, const float *v2, int len)
Calculate the scalar product of two vectors of floats.
Definition: float_dsp.h:159
av_log2
int av_log2(unsigned v)
Definition: intmath.c:26
SingleChannelElement::ret
INTFLOAT * ret
PCM output.
Definition: aac.h:269
ChannelElement::present
int present
Definition: aac.h:276
update_ltp
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
Update the LTP buffer for next frame.
Definition: aacdec_template.c:2476
AACContext::update_ltp
void(* update_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:364
AACContext::imdct_and_windowing
void(* imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:358
vector_pow43
static void vector_pow43(int *coefs, int len)
Definition: aacdec_fixed.c:151
OutputConfiguration::channel_layout
uint64_t channel_layout
Definition: aac.h:128
SingleChannelElement::sf
INTFLOAT sf[120]
scalefactors
Definition: aac.h:255
AVFloatDSPContext::vector_fmul_reverse
void(* vector_fmul_reverse)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats, and store the result in a vector of floats...
Definition: float_dsp.h:138
AV_EF_BITSTREAM
#define AV_EF_BITSTREAM
detect bitstream specification deviations
Definition: avcodec.h:2973
ChannelElement::ms_mask
uint8_t ms_mask[128]
Set if mid/side stereo is used for each scalefactor window band.
Definition: aac.h:281
apply_independent_coupling
static void apply_independent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply independent channel coupling (applied after IMDCT).
Definition: aacdec.c:246
av_mallocz
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:252
subband_scale
static void subband_scale(int *dst, int *src, int scale, int offset, int len)
Definition: aacdec_fixed.c:165
MAX_LTP_LONG_SFB
#define MAX_LTP_LONG_SFB
Definition: aac.h:51
GET_GAIN
#define GET_GAIN(x, y)
Definition: aac_defines.h:96
DynamicRangeControl
Dynamic Range Control - decoded from the bitstream but not processed further.
Definition: aac.h:211
che_configure
static av_cold int che_configure(AACContext *ac, enum ChannelPosition che_pos, int type, int id, int *channels)
Check for the channel element in the current channel position configuration.
Definition: aacdec_template.c:127
vlc_scalefactors
static VLC vlc_scalefactors
Definition: aacdec_template.c:94
NOISE_PRE
#define NOISE_PRE
preamble for NOISE_BT, put in bitstream with the first noise band
Definition: aac.h:156
FF_PROFILE_AAC_HE_V2
#define FF_PROFILE_AAC_HE_V2
Definition: avcodec.h:3190
predict
static av_always_inline void predict(PredictorState *ps, float *coef, int output_enable)
Definition: aacdec.c:174
ChannelCoupling::type
enum RawDataBlockType type[8]
Type of channel element to be coupled - SCE or CPE.
Definition: aac.h:237
AVCodecContext::profile
int profile
profile
Definition: avcodec.h:3181
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:436
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:688
aac_decode_er_frame
static int aac_decode_er_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb)
Definition: aacdec_template.c:2849
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:2133
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:2362
lcg_random
static av_always_inline int lcg_random(unsigned previous_val)
linear congruential pseudorandom number generator
Definition: aacdec_template.c:1026
DynamicRangeControl::band_incr
int band_incr
Number of DRC bands greater than 1 having DRC info.
Definition: aac.h:216
ff_aac_num_swb_128
const uint8_t ff_aac_num_swb_128[]
Definition: aactab.c:55
AAC_RENAME_32
#define AAC_RENAME_32(x)
Definition: aac_defines.h:84
ff_cbrt_tableinit
void ff_cbrt_tableinit(void)
Definition: cbrt_tablegen.h:40
AACContext::dmono_mode
int dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
Definition: aac.h:349
IndividualChannelStream::swb_offset
const uint16_t * swb_offset
table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular wind...
Definition: aac.h:181
AOT_ER_AAC_LTP
N Error Resilient Long Term Prediction.
Definition: mpeg4audio.h:76
INTFLOAT
float INTFLOAT
Definition: aac_defines.h:85
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:1252
AVFloatDSPContext::vector_fmul_window
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
Overlap/add with window function.
Definition: float_dsp.h:103
EXT_FILL
Definition: aac.h:67
BandType
BandType
Definition: aac.h:82
bits
uint8_t bits
Definition: crc.c:296
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2446
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:82
OutputConfiguration::layout_map
uint8_t layout_map[MAX_ELEM_ID *4][3]
Definition: aac.h:125
OC_TRIAL_PCE
Output configuration under trial specified by an inband PCE.
Definition: aac.h:117
ff_swb_offset_480
const uint16_t *const ff_swb_offset_480[]
Definition: aactab.c:1248
FF_DEBUG_PICT_INFO
#define FF_DEBUG_PICT_INFO
Definition: avcodec.h:2917
ChannelElement::ch
SingleChannelElement ch[2]
Definition: aac.h:284
ff_swb_offset_512
const uint16_t *const ff_swb_offset_512[]
Definition: aactab.c:1240
TYPE_LFE
Definition: aac.h:59
ff_tns_max_bands_480
const uint8_t ff_tns_max_bands_480[]
Definition: aactab.c:1282
end
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
SingleChannelElement::tns
TemporalNoiseShaping tns
Definition: aac.h:250
AOT_ER_AAC_LD
N Error Resilient Low Delay.
Definition: mpeg4audio.h: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:2304
ff_aac_scalefactor_bits
const uint8_t ff_aac_scalefactor_bits[121]
Definition: aactab.c:82
CouplingPoint
CouplingPoint
The point during decoding at which channel coupling is applied.
Definition: aac.h:106
AVCodecContext::extradata
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1791
ChannelCoupling::num_coupled
int num_coupled
number of target elements
Definition: aac.h:236
AV_CH_LOW_FREQUENCY
#define AV_CH_LOW_FREQUENCY
Definition: channel_layout.h:52
DynamicRangeControl::exclude_mask
int exclude_mask[MAX_CHANNELS]
Channels to be excluded from DRC processing.
Definition: aac.h:215
TemporalNoiseShaping::n_filt
int n_filt[8]
Definition: aac.h:200
AACContext::mdct_ltp
FFTContext mdct_ltp
Definition: aac.h:326
AACContext::vector_pow43
void(* vector_pow43)(int *coefs, int len)
Definition: aac.h:365
AACContext::output_element
SingleChannelElement * output_element[MAX_CHANNELS]
Points to each SingleChannelElement.
Definition: aac.h:340
aac_decode_init
static av_cold int aac_decode_init(AVCodecContext *avctx)
Definition: aacdec_template.c:1112
AVPacket::data
uint8_t * data
Definition: avcodec.h:1601
get_bits_count
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:199
AAC_MUL31
#define AAC_MUL31(x, y)
Definition: aac_defines.h:100
count_channels
static int count_channels(uint8_t(*layout)[3], int tags)
Definition: aacdec_template.c:103
ff_dlog
#define ff_dlog(a,...)
Definition: tableprint_vlc.h:29
INTENSITY_BT
Scalefactor data are intensity stereo positions (in phase).
Definition: aac.h:89
AV_LOG_VERBOSE
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
sample_rate_idx
static int sample_rate_idx(int rate)
Definition: aacdec_template.c:1039
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:1514
AV_CH_BACK_LEFT
#define AV_CH_BACK_LEFT
Definition: channel_layout.h:53
ChannelCoupling::id_select
int id_select[8]
element id
Definition: aac.h:238
size
ptrdiff_t size
Definition: opengl_enc.c:101
ff_aac_codebook_vector_vals
const float *const ff_aac_codebook_vector_vals[]
Definition: aactab.c:1064
avpriv_alloc_fixed_dsp
AVFixedDSPContext * avpriv_alloc_fixed_dsp(int bit_exact)
Allocate and initialize a fixed DSP context.
Definition: fixed_dsp.c:148
fixed_sqrt
static av_always_inline int fixed_sqrt(int x, int bits)
Calculate the square root.
Definition: fixed_dsp.h:176
AOT_ER_AAC_LC
N Error Resilient Low Complexity.
Definition: mpeg4audio.h:75
AACContext::tag_che_map
ChannelElement * tag_che_map[4][MAX_ELEM_ID]
Definition: aac.h:306
AVOnce
#define AVOnce
Definition: thread.h:158
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
OC_GLOBAL_HDR
Output configuration set in a global header but not yet locked.
Definition: aac.h:119
spectral_to_sample
static void spectral_to_sample(AACContext *ac, int samples)
Convert spectral data to samples, applying all supported tools as appropriate.
Definition: aacdec_template.c:2718
AACContext::random_state
int random_state
Definition: aac.h:333
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:2787
get_bits_left
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:568
OutputConfiguration::m4ac
MPEG4AudioConfig m4ac
Definition: aac.h:124
DynamicRangeControl::dyn_rng_sgn
int dyn_rng_sgn[17]
DRC sign information; 0 - positive, 1 - negative.
Definition: aac.h:213
ff_sbr_apply
void AAC_RENAME() ff_sbr_apply(AACContext *ac, SpectralBandReplication *sbr, int id_aac, INTFLOAT *L, INTFLOAT *R)
Apply one SBR element to one AAC element.
Definition: aacsbr_template.c:1467
ff_cbrt_tab
uint32_t ff_cbrt_tab[1<< 13]
pop_output_configuration
static void pop_output_configuration(AACContext *ac)
Restore the previous output configuration if and only if the current configuration is unlocked...
Definition: aacdec_template.c:420
decode_fill
static int decode_fill(AACContext *ac, GetBitContext *gb, int len)
Definition: aacdec_template.c:2271
UPDATE_CACHE
#define UPDATE_CACHE(name, gb)
Definition: get_bits.h:160
SingleChannelElement::predictor_state
PredictorState predictor_state[MAX_PREDICTORS]
Definition: aac.h:268
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
ChannelElement::sbr
SpectralBandReplication sbr
Definition: aac.h:287
AACContext::mdct_small
FFTContext mdct_small
Definition: aac.h:324
AVFloatDSPContext::vector_fmul
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats and store the result in a vector of floats...
Definition: float_dsp.h:38
av_default_item_name
av_default_item_name
Definition: libopenh264enc.c:78
ChannelCoupling::coupling_point
enum CouplingPoint coupling_point
The point during decoding at which coupling is applied.
Definition: aac.h:235
AVERROR
#define AVERROR(e)
Definition: error.h:43
elem_to_channel::av_position
uint64_t av_position
Definition: aacdec_template.c:193
ff_aac_num_swb_1024
const uint8_t ff_aac_num_swb_1024[]
Definition: aactab.c:43
AVFloatDSPContext::butterflies_float
void(* butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
Definition: float_dsp.h:148
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
ff_aac_kbd_long_1024
float ff_aac_kbd_long_1024[1024]
Definition: aactab.c:38
AACContext::temp
INTFLOAT temp[128]
Definition: aac.h:352
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1771
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:871
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:228
assign_pair
static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID], uint8_t(*layout_map)[3], int offset, uint64_t left, uint64_t right, int pos)
Definition: aacdec_template.c:199
IndividualChannelStream::max_sfb
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:175
ff_mdct_init
#define ff_mdct_init
Definition: fft.h:169
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:409
ff_aac_eld_window_512
const float ff_aac_eld_window_512[1920]
Definition: aactab.c:1291
TYPE_FIL
Definition: aac.h:62
offset
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
count
GLsizei count
Definition: opengl_enc.c:109
CLOSE_READER
#define CLOSE_READER(name, gb)
Definition: get_bits.h:131
IndividualChannelStream::num_swb
int num_swb
number of scalefactor window bands
Definition: aac.h:183
count_paired_channels
static int count_paired_channels(uint8_t(*layout_map)[3], int tags, int pos, int *current)
Definition: aacdec_template.c:228
FFMAX
#define FFMAX(a, b)
Definition: common.h:94
fail
#define fail()
Definition: checkasm.h:83
DynamicRangeControl::prog_ref_level
int prog_ref_level
A reference level for the long-term program audio level for all channels combined.
Definition: aac.h:219
OC_LOCKED
Output configuration locked in place.
Definition: aac.h:120
PredictorState
Predictor State.
Definition: aac.h:135
AACADTSHeaderInfo::chan_config
uint8_t chan_config
Definition: aacadtsdec.h:38
VLC
Definition: vlc.h:26
ff_aac_pow2sf_tab
float ff_aac_pow2sf_tab[428]
Definition: aactab.c:35
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2489
SKIP_BITS
#define SKIP_BITS(name, gb, num)
Definition: get_bits.h:175
AAC_RENAME
#define AAC_RENAME(x)
Definition: aac_defines.h:83
AACContext::warned_remapping_once
int warned_remapping_once
Definition: aac.h:308
SingleChannelElement::ret_buf
INTFLOAT ret_buf[2048]
PCM output buffer.
Definition: aac.h:264
AOT_ER_AAC_SCALABLE
N Error Resilient Scalable.
Definition: mpeg4audio.h:77
window
static SDL_Window * window
Definition: ffplay.c:361
reset_predictor_group
static void reset_predictor_group(PredictorState *ps, int group_num)
Definition: aacdec_template.c:1055
get_che
static ChannelElement * get_che(AACContext *ac, int type, int elem_id)
Definition: aacdec_template.c:556
IndividualChannelStream::window_sequence
enum WindowSequence window_sequence[2]
Definition: aac.h:176
SingleChannelElement::ltp_state
INTFLOAT ltp_state[3072]
time signal for LTP
Definition: aac.h:265
AV_CODEC_FLAG_BITEXACT
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:886
ff_aac_num_swb_512
const uint8_t ff_aac_num_swb_512[]
Definition: aactab.c:47
AVCodecContext::err_recognition
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
Definition: avcodec.h:2964
aac_decode_frame
static int aac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: aacdec_template.c:3106
IndividualChannelStream::predictor_reset_group
int predictor_reset_group
Definition: aac.h:188
frame_configure_elements
static int frame_configure_elements(AVCodecContext *avctx)
Definition: aacdec_template.c:158
FFMIN
#define FFMIN(a, b)
Definition: common.h:96
DynamicRangeControl::dyn_rng_ctl
int dyn_rng_ctl[17]
DRC magnitude information.
Definition: aac.h:214
AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
ltp_coef
static const INTFLOAT ltp_coef[8]
Definition: aactab.h:94
decode_mid_side_stereo
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, int ms_present)
Decode Mid/Side data; reference: table 4.54.
Definition: aacdec_template.c:1557
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:2030
AACADTSHeaderInfo::num_aac_frames
uint8_t num_aac_frames
Definition: aacadtsdec.h:39
Pulse::pos
int pos[4]
Definition: aac.h:227
AOT_AAC_MAIN
Y Main.
Definition: mpeg4audio.h: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:282
AACContext::mdct_ld
FFTContext mdct_ld
Definition: aac.h:325
ff_aacdec_init_mips
void ff_aacdec_init_mips(AACContext *c)
Definition: aacdec_mips.c:433
ff_decode_sbr_extension
int AAC_RENAME() ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr, GetBitContext *gb, int crc, int cnt, int id_aac)
Decode one SBR element.
Definition: aacsbr_template.c:1104
LAST_SKIP_BITS
#define LAST_SKIP_BITS(name, gb, num)
Definition: get_bits.h:181
get_vlc2
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
Definition: get_bits.h:535
TemporalNoiseShaping::length
int length[8][4]
Definition: aac.h:201
AVFloatDSPContext::vector_fmul_scalar
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
Definition: float_dsp.h:69
aac_static_table_init
static av_cold void aac_static_table_init(void)
Definition: aacdec_template.c:1072
ff_aac_sbr_ctx_close
void AAC_RENAME() ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
Close one SBR context.
Definition: aacsbr_template.c:102
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:2685
AV_CH_FRONT_LEFT_OF_CENTER
#define AV_CH_FRONT_LEFT_OF_CENTER
Definition: channel_layout.h:55
AV_EF_EXPLODE
#define AV_EF_EXPLODE
abort decoding on minor error detection
Definition: avcodec.h:2975
n
int n
Definition: avisynth_c.h:684
ff_tns_max_bands_1024
const uint8_t ff_tns_max_bands_1024[]
Definition: aactab.c:1274
GET_VLC
#define GET_VLC(code, name, gb, table, bits, max_depth)
If the vlc code is invalid and max_depth=1, then no bits will be removed.
Definition: get_bits.h:458
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:1238
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:2921
apply_dependent_coupling
static void apply_dependent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply dependent channel coupling (applied before IMDCT).
Definition: aacdec.c:210
decode_spectrum_and_dequant
static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024], GetBitContext *gb, const INTFLOAT sf[120], int pulse_present, const Pulse *pulse, const IndividualChannelStream *ics, enum BandType band_type[120])
Decode spectral data; reference: table 4.50.
Definition: aacdec_template.c:1582
ff_aac_sbr_init
void AAC_RENAME() ff_aac_sbr_init(void)
Initialize SBR.
Definition: aacsbr_template.c:37
DynamicRangeControl::pce_instance_tag
int pce_instance_tag
Indicates with which program the DRC info is associated.
Definition: aac.h:212
windowing_and_mdct_ltp
static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out, INTFLOAT *in, IndividualChannelStream *ics)
Apply windowing and MDCT to obtain the spectral coefficient from the predicted sample by LTP...
Definition: aacdec_template.c:2418
AOT_AAC_SCALABLE
N Scalable.
Definition: mpeg4audio.h:66
tns_tmp2_map
static const INTFLOAT *const tns_tmp2_map[4]
Definition: aactab.h:126
elem_to_channel::aac_position
uint8_t aac_position
Definition: aacdec_template.c:196
SHOW_UBITS
#define SHOW_UBITS(name, gb, num)
Definition: get_bits.h:193
FF_ARRAY_ELEMS
#define FF_ARRAY_ELEMS(a)
Definition: sinewin_tablegen_template.c:36
AV_CH_FRONT_RIGHT_OF_CENTER
#define AV_CH_FRONT_RIGHT_OF_CENTER
Definition: channel_layout.h:56
DynamicRangeControl::interpolation_scheme
int interpolation_scheme
Indicates the interpolation scheme used in the SBR QMF domain.
Definition: aac.h:217
ChannelCoupling
coupling parameters
Definition: aac.h:234
AACContext::tags_mapped
int tags_mapped
Definition: aac.h:307
reset_all_predictors
static void reset_all_predictors(PredictorState *ps)
Definition: aacdec_template.c:1032
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:1199
AVERROR_PATCHWELCOME
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
ChannelCoupling::ch_select
int ch_select[8]
[0] shared list of gains; [1] list of gains for right channel; [2] list of gains for left channel; [3...
Definition: aac.h:239
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2458
AACContext::force_dmono_mode
int force_dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
Definition: aac.h:348
AV_PKT_DATA_NEW_EXTRADATA
The AV_PKT_DATA_NEW_EXTRADATA is used to notify the codec or the format that the extradata buffer was...
Definition: avcodec.h:1372
TemporalNoiseShaping::order
int order[8][4]
Definition: aac.h:203
AV_LOG_INFO
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
AV_ONCE_INIT
#define AV_ONCE_INIT
Definition: thread.h:159
AACContext::warned_num_aac_frames
int warned_num_aac_frames
Definition: aac.h:355
AAC_INIT_VLC_STATIC
#define AAC_INIT_VLC_STATIC(num, size)
Definition: aacdec_template.c:1062
TemporalNoiseShaping
Temporal Noise Shaping.
Definition: aac.h:198
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:2438
ff_aac_kbd_short_128
float ff_aac_kbd_short_128[128]
Definition: aactab.c:39
options
static const AVOption options[]
Definition: aacdec_template.c:3220
init_get_bits8
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:437
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:1412
AVCodecContext::debug
int debug
debug
Definition: avcodec.h:2916
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:2080
LongTermPrediction
Long Term Prediction.
Definition: aac.h:163
AVCodecContext
main external API structure.
Definition: avcodec.h:1676
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:157
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:947
OPEN_READER
#define OPEN_READER(name, gb)
Definition: get_bits.h:120
SingleChannelElement::ics
IndividualChannelStream ics
Definition: aac.h:249
in
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method!=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt)>2){ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc){av_free(ac);returnNULL;}returnac;}in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar){ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar?ac->channels:1;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
Definition: audio_convert.c:194
avpriv_aac_parse_header
int avpriv_aac_parse_header(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
Parse AAC frame header.
Definition: aacadtsdec.c:29
buf
void * buf
Definition: avisynth_c.h:690
MAX_PREDICTORS
#define MAX_PREDICTORS
Definition: aac.h:146
cbrtf
static av_always_inline float cbrtf(float x)
Definition: libm.h:61
FFTContext::imdct_half
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:108
type
GLint GLenum type
Definition: opengl_enc.c:105
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:1792
IndividualChannelStream::group_len
uint8_t group_len[8]
Definition: aac.h:179
AVERROR_BUG
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
get_bits1
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:299
skip_bits1
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:324
AACContext::apply_tns
void(* apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns, IndividualChannelStream *ics, int decode)
Definition: aac.h:360
MAX_ELEM_ID
#define MAX_ELEM_ID
Definition: aac.h:48
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
AVFrame::sample_rate
int sample_rate
Sample rate of the audio data.
Definition: frame.h:348
aac_decode_close
static av_cold int aac_decode_close(AVCodecContext *avctx)
Definition: aacdec_template.c:3175
skip_bits
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:292
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:946
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:260
decode_drc_channel_exclusions
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext *gb)
Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4...
Definition: aacdec_template.c:2205
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:191
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:406
AV_PKT_DATA_SKIP_SAMPLES
Recommmends skipping the specified number of samples.
Definition: avcodec.h:1473
GET_CACHE
#define GET_CACHE(name, gb)
Definition: get_bits.h:197
elem_to_channel::syn_ele
uint8_t syn_ele
Definition: aacdec_template.c:194
VMUL2
static float * VMUL2(float *dst, const float *v, unsigned idx, const float *scale)
Definition: aacdec.c:83
OCStatus
OCStatus
Output configuration status.
Definition: aac.h:115
AVCodecInternal::skip_samples
int skip_samples
Number of audio samples to skip at the start of the next decoded frame.
Definition: internal.h:157
MAX_CHANNELS
#define MAX_CHANNELS
Definition: aac.h:47
AOT_ER_BSAC
N Error Resilient Bit-Sliced Arithmetic Coding.
Definition: mpeg4audio.h: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:84
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:119
FF_COMPLIANCE_STRICT
#define FF_COMPLIANCE_STRICT
Strictly conform to all the things in the spec no matter what consequences.
Definition: avcodec.h:2896
AACContext
main AAC context
Definition: aac.h:293
u
#define u(width,...)
Definition: vaapi_encode_h26x.h:36
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:63
IndividualChannelStream::ltp
LongTermPrediction ltp
Definition: aac.h:180
ChannelElement::coup
ChannelCoupling coup
Definition: aac.h:286
OC_TRIAL_FRAME
Output configuration under trial specified by a frame header.
Definition: aac.h:118
MPEG4AudioConfig::frame_length_short
int frame_length_short
Definition: mpeg4audio.h:41
decode_prediction
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb)
Definition: aacdec_template.c:1216
ff_tns_max_bands_128
const uint8_t ff_tns_max_bands_128[]
Definition: aactab.c:1286
NOISE_OFFSET
#define NOISE_OFFSET
subtracted from global gain, used as offset for the preamble
Definition: aac.h:158
imdct_and_window
static void imdct_and_window(TwinVQContext *tctx, enum TwinVQFrameType ftype, int wtype, float *in, float *prev, int ch)
Definition: twinvq.c:327
av_frame_unref
void av_frame_unref(AVFrame *frame)
Unreference all the buffers referenced by frame and reset the frame fields.
Definition: frame.c:493
avpriv_report_missing_feature
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
filt
static const int8_t filt[NUMTAPS]
Definition: af_earwax.c:39
SingleChannelElement::band_type_run_end
int band_type_run_end[120]
band type run end points
Definition: aac.h:254
decode_band_types
static int decode_band_types(AACContext *ac, enum BandType band_type[120], int band_type_run_end[120], GetBitContext *gb, IndividualChannelStream *ics)
Decode band types (section_data payload); reference: table 4.46.
Definition: aacdec_template.c:1363
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:198
DynamicRangeControl::band_top
int band_top[17]
Indicates the top of the i-th DRC band in units of 4 spectral lines.
Definition: aac.h:218
AV_CH_SIDE_RIGHT
#define AV_CH_SIDE_RIGHT
Definition: channel_layout.h:59
SingleChannelElement::coeffs
INTFLOAT coeffs[1024]
coefficients for IMDCT, maybe processed
Definition: aac.h:262
vlc_spectral
static VLC vlc_spectral[11]
Definition: aacdec_template.c:95
OutputConfiguration::status
enum OCStatus status
Definition: aac.h:129
ChannelCoupling::gain
INTFLOAT gain[16][120]
Definition: aac.h:242
INTENSITY_BT2
Scalefactor data are intensity stereo positions (out of phase).
Definition: aac.h:88
AOT_ER_AAC_ELD
N Error Resilient Enhanced Low Delay.
Definition: mpeg4audio.h:96
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:519
M_SQRT2
#define M_SQRT2
Definition: mathematics.h:61
RANGE15
#define RANGE15(x)
Definition: aac_defines.h:95
TemporalNoiseShaping::coef
INTFLOAT coef[8][4][TNS_MAX_ORDER]
Definition: aac.h:205
LongTermPrediction::lag
int16_t lag
Definition: aac.h:165
AACContext::che_drc
DynamicRangeControl che_drc
Definition: aac.h:299
reset_predict_state
static av_always_inline void reset_predict_state(PredictorState *ps)
Definition: aacdec.c:72
AACContext::frame
AVFrame * frame
Definition: aac.h:296
AACContext::oc
OutputConfiguration oc[2]
Definition: aac.h:354
AV_PKT_DATA_JP_DUALMONO
An AV_PKT_DATA_JP_DUALMONO side data packet indicates that the packet may contain "dual mono" audio s...
Definition: avcodec.h:1483
ff_aac_pred_sfb_max
const uint8_t ff_aac_pred_sfb_max[]
Definition: aactab.c:59
TemporalNoiseShaping::direction
int direction[8][4]
Definition: aac.h:202
AACContext::apply_ltp
void(* apply_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:359
IndividualChannelStream::prediction_used
uint8_t prediction_used[41]
Definition: aac.h:190
ff_aac_eld_window_480
const float ff_aac_eld_window_480[1800]
Definition: aactab.c:2258
SingleChannelElement::saved
INTFLOAT saved[1536]
overlap
Definition: aac.h:263
SingleChannelElement
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:248
ff_mdct_end
#define ff_mdct_end
Definition: fft.h:170
ff_aac_num_swb_480
const uint8_t ff_aac_num_swb_480[]
Definition: aactab.c:51
c
static double c[64]
Definition: vsrc_mptestsrc.c:87
ff_swb_offset_1024
const uint16_t *const ff_swb_offset_1024[]
Definition: aactab.c:1232
AAC_SIGNE
unsigned AAC_SIGNE
Definition: aac_defines.h:89
AACContext::windowing_and_mdct_ltp
void(* windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out, INTFLOAT *in, IndividualChannelStream *ics)
Definition: aac.h:362
TYPE_PCE
Definition: aac.h:61
IndividualChannelStream
Individual Channel Stream.
Definition: aac.h:174
LongTermPrediction::coef
INTFLOAT coef
Definition: aac.h:167
ch
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_YASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
AVERROR_UNKNOWN
#define AVERROR_UNKNOWN
Unknown error, typically from an external library.
Definition: error.h:71
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:1073
ff_aac_tableinit
static void ff_aac_tableinit(void)
Definition: aactab.h:45
AV_INPUT_BUFFER_PADDING_SIZE
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:734
ChannelElement
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:275
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:1718
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:1915
av_free
#define av_free(p)
Definition: tableprint_vlc.h:34
FF_DEBUG_STARTCODE
#define FF_DEBUG_STARTCODE
Definition: avcodec.h:2930
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:793
ff_tns_max_bands_512
const uint8_t ff_tns_max_bands_512[]
Definition: aactab.c:1278
len
int len
Definition: vorbis_enc_data.h:452
ZERO_BT
Scalefactors and spectral data are all zero.
Definition: aac.h:83
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:2439
Pulse::num_pulse
int num_pulse
Definition: aac.h:225
DEC_SPAIR
static int * DEC_SPAIR(int *dst, unsigned idx)
Definition: aacdec_fixed.c:107
AVCodecContext::internal
struct AVCodecInternal * internal
Private context used for internal data.
Definition: avcodec.h:1726
tmp
static uint8_t tmp[8]
Definition: des.c:38
ff_mpeg4audio_channels
const uint8_t ff_mpeg4audio_channels[8]
Definition: mpeg4audio.c:62
ff_thread_once
static int ff_thread_once(char *control, void(*routine)(void))
Definition: thread.h:161
VLC::table
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
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:445
layout
uint64_t layout
Definition: channel_layout.c:77
FF_PROFILE_AAC_HE
#define FF_PROFILE_AAC_HE
Definition: avcodec.h:3189
SingleChannelElement::band_type
enum BandType band_type[128]
band types
Definition: aac.h:252
decode_dynamic_range
static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext *gb)
Decode dynamic range information; reference: table 4.52.
Definition: aacdec_template.c:2224
AV_CH_FRONT_RIGHT
#define AV_CH_FRONT_RIGHT
Definition: channel_layout.h:50
POW_SF2_ZERO
#define POW_SF2_ZERO
ff_aac_pow2sf_tab index corresponding to pow(2, 0);
Definition: aac.h:154
imdct_and_windowing_ld
static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
Definition: aacdec_template.c:2579
out
FILE * out
Definition: movenc.c:54
AACContext::mdct
FFTContext mdct
Definition: aac.h:323
MPEG4AudioConfig::sbr
int sbr
-1 implicit, 1 presence
Definition: mpeg4audio.h:34
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
start
void INT64 start
Definition: avisynth_c.h:690
decode
static int decode(AVCodecContext *avctx, AVFrame *frame, int *got_frame, AVPacket *pkt)
Definition: ffmpeg.c:2035
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:338
av_always_inline
#define av_always_inline
Definition: attributes.h:39
apply_mid_side_stereo
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
Mid/Side stereo decoding; reference: 4.6.8.1.3.
Definition: aacdec_template.c:1992
VLC_TYPE
#define VLC_TYPE
Definition: vlc.h:24
AV_CH_SIDE_LEFT
#define AV_CH_SIDE_LEFT
Definition: channel_layout.h:58
FFSWAP
#define FFSWAP(type, a, b)
Definition: common.h:99
MPEG4AudioConfig::ps
int ps
-1 implicit, 1 presence
Definition: mpeg4audio.h:40
LongTermPrediction::used
int8_t used[MAX_LTP_LONG_SFB]
Definition: aac.h:168
ff_swb_offset_128
const uint16_t *const ff_swb_offset_128[]
Definition: aactab.c:1256
LongTermPrediction::present
int8_t present
Definition: aac.h:164
AACADTSHeaderInfo::sample_rate
uint32_t sample_rate
Definition: aacadtsdec.h:32
aac_decoder_class
static const AVClass aac_decoder_class
Definition: aacdec_template.c:3233
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:231
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:2496
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:1578
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
AVCodecContext::strict_std_compliance
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:2894
ff_init_ff_sine_windows
void AAC_RENAME() ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
Definition: sinewin_tablegen.h:72
DEC_UQUAD
static int * DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
Definition: aacdec_fixed.c:133
AV_CH_BACK_RIGHT
#define AV_CH_BACK_RIGHT
Definition: channel_layout.h:54
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:94
OC_NONE
Output unconfigured.
Definition: aac.h:116
aac_channel_layout_map
static const uint8_t aac_channel_layout_map[16][5][3]
Definition: aacdectab.h:40
RawDataBlockType
RawDataBlockType
Definition: aac.h:55
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