FFmpeg  4.4.5
aacdec_template.c
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1 /*
2  * AAC decoder
3  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
4  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
5  * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
6  *
7  * AAC LATM decoder
8  * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
9  * Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
10  *
11  * AAC decoder fixed-point implementation
12  * Copyright (c) 2013
13  * MIPS Technologies, Inc., California.
14  *
15  * This file is part of FFmpeg.
16  *
17  * FFmpeg is free software; you can redistribute it and/or
18  * modify it under the terms of the GNU Lesser General Public
19  * License as published by the Free Software Foundation; either
20  * version 2.1 of the License, or (at your option) any later version.
21  *
22  * FFmpeg is distributed in the hope that it will be useful,
23  * but WITHOUT ANY WARRANTY; without even the implied warranty of
24  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
25  * Lesser General Public License for more details.
26  *
27  * You should have received a copy of the GNU Lesser General Public
28  * License along with FFmpeg; if not, write to the Free Software
29  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
30  */
31 
32 /**
33  * @file
34  * AAC decoder
35  * @author Oded Shimon ( ods15 ods15 dyndns org )
36  * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
37  *
38  * AAC decoder fixed-point implementation
39  * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
40  * @author Nedeljko Babic ( nedeljko.babic imgtec com )
41  */
42 
43 /*
44  * supported tools
45  *
46  * Support? Name
47  * N (code in SoC repo) gain control
48  * Y block switching
49  * Y window shapes - standard
50  * N window shapes - Low Delay
51  * Y filterbank - standard
52  * N (code in SoC repo) filterbank - Scalable Sample Rate
53  * Y Temporal Noise Shaping
54  * Y Long Term Prediction
55  * Y intensity stereo
56  * Y channel coupling
57  * Y frequency domain prediction
58  * Y Perceptual Noise Substitution
59  * Y Mid/Side stereo
60  * N Scalable Inverse AAC Quantization
61  * N Frequency Selective Switch
62  * N upsampling filter
63  * Y quantization & coding - AAC
64  * N quantization & coding - TwinVQ
65  * N quantization & coding - BSAC
66  * N AAC Error Resilience tools
67  * N Error Resilience payload syntax
68  * N Error Protection tool
69  * N CELP
70  * N Silence Compression
71  * N HVXC
72  * N HVXC 4kbits/s VR
73  * N Structured Audio tools
74  * N Structured Audio Sample Bank Format
75  * N MIDI
76  * N Harmonic and Individual Lines plus Noise
77  * N Text-To-Speech Interface
78  * Y Spectral Band Replication
79  * Y (not in this code) Layer-1
80  * Y (not in this code) Layer-2
81  * Y (not in this code) Layer-3
82  * N SinuSoidal Coding (Transient, Sinusoid, Noise)
83  * Y Parametric Stereo
84  * N Direct Stream Transfer
85  * Y (not in fixed point code) Enhanced AAC Low Delay (ER AAC ELD)
86  *
87  * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
88  * - HE AAC v2 comprises LC AAC with Spectral Band Replication and
89  Parametric Stereo.
90  */
91 
92 #include "libavutil/thread.h"
93 
95 static VLC vlc_spectral[11];
96 
97 static int output_configure(AACContext *ac,
98  uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
99  enum OCStatus oc_type, int get_new_frame);
100 
101 #define overread_err "Input buffer exhausted before END element found\n"
102 
103 static int count_channels(uint8_t (*layout)[3], int tags)
104 {
105  int i, sum = 0;
106  for (i = 0; i < tags; i++) {
107  int syn_ele = layout[i][0];
108  int pos = layout[i][2];
109  sum += (1 + (syn_ele == TYPE_CPE)) *
111  }
112  return sum;
113 }
114 
115 /**
116  * Check for the channel element in the current channel position configuration.
117  * If it exists, make sure the appropriate element is allocated and map the
118  * channel order to match the internal FFmpeg channel layout.
119  *
120  * @param che_pos current channel position configuration
121  * @param type channel element type
122  * @param id channel element id
123  * @param channels count of the number of channels in the configuration
124  *
125  * @return Returns error status. 0 - OK, !0 - error
126  */
128  enum ChannelPosition che_pos,
129  int type, int id, int *channels)
130 {
131  if (*channels >= MAX_CHANNELS)
132  return AVERROR_INVALIDDATA;
133  if (che_pos) {
134  if (!ac->che[type][id]) {
135  if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
136  return AVERROR(ENOMEM);
138  }
139  if (type != TYPE_CCE) {
140  if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
141  av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
142  return AVERROR_INVALIDDATA;
143  }
144  ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
145  if (type == TYPE_CPE ||
146  (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
147  ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
148  }
149  }
150  } else {
151  if (ac->che[type][id])
153  av_freep(&ac->che[type][id]);
154  }
155  return 0;
156 }
157 
159 {
160  AACContext *ac = avctx->priv_data;
161  int type, id, ch, ret;
162 
163  /* set channel pointers to internal buffers by default */
164  for (type = 0; type < 4; type++) {
165  for (id = 0; id < MAX_ELEM_ID; id++) {
166  ChannelElement *che = ac->che[type][id];
167  if (che) {
168  che->ch[0].ret = che->ch[0].ret_buf;
169  che->ch[1].ret = che->ch[1].ret_buf;
170  }
171  }
172  }
173 
174  /* get output buffer */
175  av_frame_unref(ac->frame);
176  if (!avctx->channels)
177  return 1;
178 
179  ac->frame->nb_samples = 2048;
180  if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
181  return ret;
182 
183  /* map output channel pointers to AVFrame data */
184  for (ch = 0; ch < avctx->channels; ch++) {
185  if (ac->output_element[ch])
186  ac->output_element[ch]->ret = (INTFLOAT *)ac->frame->extended_data[ch];
187  }
188 
189  return 0;
190 }
191 
193  uint64_t av_position;
197 };
198 
199 static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
200  uint8_t (*layout_map)[3], int offset, uint64_t left,
201  uint64_t right, int pos, uint64_t *layout)
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  if (e2c_vec[offset].av_position != UINT64_MAX)
211  *layout |= e2c_vec[offset].av_position;
212 
213  return 1;
214  } else {
215  e2c_vec[offset] = (struct elem_to_channel) {
216  .av_position = left,
217  .syn_ele = TYPE_SCE,
218  .elem_id = layout_map[offset][1],
219  .aac_position = pos
220  };
221  e2c_vec[offset + 1] = (struct elem_to_channel) {
222  .av_position = right,
223  .syn_ele = TYPE_SCE,
224  .elem_id = layout_map[offset + 1][1],
225  .aac_position = pos
226  };
227  if (left != UINT64_MAX)
228  *layout |= left;
229 
230  if (right != UINT64_MAX)
231  *layout |= right;
232 
233  return 2;
234  }
235 }
236 
237 static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
238  int *current)
239 {
240  int num_pos_channels = 0;
241  int first_cpe = 0;
242  int sce_parity = 0;
243  int i;
244  for (i = *current; i < tags; i++) {
245  if (layout_map[i][2] != pos)
246  break;
247  if (layout_map[i][0] == TYPE_CPE) {
248  if (sce_parity) {
249  if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
250  sce_parity = 0;
251  } else {
252  return -1;
253  }
254  }
255  num_pos_channels += 2;
256  first_cpe = 1;
257  } else {
258  num_pos_channels++;
259  sce_parity ^= 1;
260  }
261  }
262  if (sce_parity &&
263  ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
264  return -1;
265  *current = i;
266  return num_pos_channels;
267 }
268 
269 #define PREFIX_FOR_22POINT2 (AV_CH_LAYOUT_7POINT1_WIDE_BACK|AV_CH_BACK_CENTER|AV_CH_SIDE_LEFT|AV_CH_SIDE_RIGHT|AV_CH_LOW_FREQUENCY_2)
270 static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
271 {
272  int i, n, total_non_cc_elements;
273  struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
274  int num_front_channels, num_side_channels, num_back_channels;
275  uint64_t layout = 0;
276 
277  if (FF_ARRAY_ELEMS(e2c_vec) < tags)
278  return 0;
279 
280  i = 0;
281  num_front_channels =
282  count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
283  if (num_front_channels < 0)
284  return 0;
285  num_side_channels =
286  count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
287  if (num_side_channels < 0)
288  return 0;
289  num_back_channels =
290  count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
291  if (num_back_channels < 0)
292  return 0;
293 
294  if (num_side_channels == 0 && num_back_channels >= 4) {
295  num_side_channels = 2;
296  num_back_channels -= 2;
297  }
298 
299  i = 0;
300  if (num_front_channels & 1) {
301  e2c_vec[i] = (struct elem_to_channel) {
303  .syn_ele = TYPE_SCE,
304  .elem_id = layout_map[i][1],
305  .aac_position = AAC_CHANNEL_FRONT
306  };
307  layout |= e2c_vec[i].av_position;
308  i++;
309  num_front_channels--;
310  }
311  if (num_front_channels >= 4) {
312  i += assign_pair(e2c_vec, layout_map, i,
316  num_front_channels -= 2;
317  }
318  if (num_front_channels >= 2) {
319  i += assign_pair(e2c_vec, layout_map, i,
323  num_front_channels -= 2;
324  }
325  while (num_front_channels >= 2) {
326  i += assign_pair(e2c_vec, layout_map, i,
327  UINT64_MAX,
328  UINT64_MAX,
330  num_front_channels -= 2;
331  }
332 
333  if (num_side_channels >= 2) {
334  i += assign_pair(e2c_vec, layout_map, i,
338  num_side_channels -= 2;
339  }
340  while (num_side_channels >= 2) {
341  i += assign_pair(e2c_vec, layout_map, i,
342  UINT64_MAX,
343  UINT64_MAX,
345  num_side_channels -= 2;
346  }
347 
348  while (num_back_channels >= 4) {
349  i += assign_pair(e2c_vec, layout_map, i,
350  UINT64_MAX,
351  UINT64_MAX,
353  num_back_channels -= 2;
354  }
355  if (num_back_channels >= 2) {
356  i += assign_pair(e2c_vec, layout_map, i,
360  num_back_channels -= 2;
361  }
362  if (num_back_channels) {
363  e2c_vec[i] = (struct elem_to_channel) {
365  .syn_ele = TYPE_SCE,
366  .elem_id = layout_map[i][1],
367  .aac_position = AAC_CHANNEL_BACK
368  };
369  layout |= e2c_vec[i].av_position;
370  i++;
371  num_back_channels--;
372  }
373 
374  if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
375  e2c_vec[i] = (struct elem_to_channel) {
377  .syn_ele = TYPE_LFE,
378  .elem_id = layout_map[i][1],
379  .aac_position = AAC_CHANNEL_LFE
380  };
381  layout |= e2c_vec[i].av_position;
382  i++;
383  }
384  if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
385  e2c_vec[i] = (struct elem_to_channel) {
387  .syn_ele = TYPE_LFE,
388  .elem_id = layout_map[i][1],
389  .aac_position = AAC_CHANNEL_LFE
390  };
391  layout |= e2c_vec[i].av_position;
392  i++;
393  }
394  while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
395  e2c_vec[i] = (struct elem_to_channel) {
396  .av_position = UINT64_MAX,
397  .syn_ele = TYPE_LFE,
398  .elem_id = layout_map[i][1],
399  .aac_position = AAC_CHANNEL_LFE
400  };
401  i++;
402  }
403 
404  // The previous checks would end up at 8 at this point for 22.2
405  if (layout == PREFIX_FOR_22POINT2 && tags == 16 && i == 8) {
406  const uint8_t (*reference_layout_map)[3] = aac_channel_layout_map[12];
407  for (int j = 0; j < tags; j++) {
408  if (layout_map[j][0] != reference_layout_map[j][0] ||
409  layout_map[j][2] != reference_layout_map[j][2])
410  goto end_of_layout_definition;
411  }
412 
413  e2c_vec[i] = (struct elem_to_channel) {
415  .syn_ele = layout_map[i][0],
416  .elem_id = layout_map[i][1],
417  .aac_position = layout_map[i][2]
418  }; layout |= e2c_vec[i].av_position; i++;
419  i += assign_pair(e2c_vec, layout_map, i,
423  &layout);
424  i += assign_pair(e2c_vec, layout_map, i,
428  &layout);
429  e2c_vec[i] = (struct elem_to_channel) {
431  .syn_ele = layout_map[i][0],
432  .elem_id = layout_map[i][1],
433  .aac_position = layout_map[i][2]
434  }; layout |= e2c_vec[i].av_position; i++;
435  i += assign_pair(e2c_vec, layout_map, i,
439  &layout);
440  e2c_vec[i] = (struct elem_to_channel) {
442  .syn_ele = layout_map[i][0],
443  .elem_id = layout_map[i][1],
444  .aac_position = layout_map[i][2]
445  }; layout |= e2c_vec[i].av_position; i++;
446  e2c_vec[i] = (struct elem_to_channel) {
448  .syn_ele = layout_map[i][0],
449  .elem_id = layout_map[i][1],
450  .aac_position = layout_map[i][2]
451  }; layout |= e2c_vec[i].av_position; i++;
452  i += assign_pair(e2c_vec, layout_map, i,
456  &layout);
457  }
458 
459 end_of_layout_definition:
460 
461  total_non_cc_elements = n = i;
462 
463  if (layout == AV_CH_LAYOUT_22POINT2) {
464  // For 22.2 reorder the result as needed
465  FFSWAP(struct elem_to_channel, e2c_vec[2], e2c_vec[0]); // FL & FR first (final), FC third
466  FFSWAP(struct elem_to_channel, e2c_vec[2], e2c_vec[1]); // FC second (final), FLc & FRc third
467  FFSWAP(struct elem_to_channel, e2c_vec[6], e2c_vec[2]); // LFE1 third (final), FLc & FRc seventh
468  FFSWAP(struct elem_to_channel, e2c_vec[4], e2c_vec[3]); // BL & BR fourth (final), SiL & SiR fifth
469  FFSWAP(struct elem_to_channel, e2c_vec[6], e2c_vec[4]); // FLc & FRc fifth (final), SiL & SiR seventh
470  FFSWAP(struct elem_to_channel, e2c_vec[7], e2c_vec[6]); // LFE2 seventh (final), SiL & SiR eight (final)
471  FFSWAP(struct elem_to_channel, e2c_vec[9], e2c_vec[8]); // TpFL & TpFR ninth (final), TFC tenth (final)
472  FFSWAP(struct elem_to_channel, e2c_vec[11], e2c_vec[10]); // TC eleventh (final), TpSiL & TpSiR twelth
473  FFSWAP(struct elem_to_channel, e2c_vec[12], e2c_vec[11]); // TpBL & TpBR twelth (final), TpSiL & TpSiR thirteenth (final)
474  } else {
475  // For everything else, utilize the AV channel position define as a
476  // stable sort.
477  do {
478  int next_n = 0;
479  for (i = 1; i < n; i++)
480  if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
481  FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
482  next_n = i;
483  }
484  n = next_n;
485  } while (n > 0);
486 
487  }
488 
489  for (i = 0; i < total_non_cc_elements; i++) {
490  layout_map[i][0] = e2c_vec[i].syn_ele;
491  layout_map[i][1] = e2c_vec[i].elem_id;
492  layout_map[i][2] = e2c_vec[i].aac_position;
493  }
494 
495  return layout;
496 }
497 
498 /**
499  * Save current output configuration if and only if it has been locked.
500  */
502  int pushed = 0;
503 
504  if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
505  ac->oc[0] = ac->oc[1];
506  pushed = 1;
507  }
508  ac->oc[1].status = OC_NONE;
509  return pushed;
510 }
511 
512 /**
513  * Restore the previous output configuration if and only if the current
514  * configuration is unlocked.
515  */
517  if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
518  ac->oc[1] = ac->oc[0];
519  ac->avctx->channels = ac->oc[1].channels;
520  ac->avctx->channel_layout = ac->oc[1].channel_layout;
521  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
522  ac->oc[1].status, 0);
523  }
524 }
525 
526 /**
527  * Configure output channel order based on the current program
528  * configuration element.
529  *
530  * @return Returns error status. 0 - OK, !0 - error
531  */
533  uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
534  enum OCStatus oc_type, int get_new_frame)
535 {
536  AVCodecContext *avctx = ac->avctx;
537  int i, channels = 0, ret;
538  uint64_t layout = 0;
539  uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
540  uint8_t type_counts[TYPE_END] = { 0 };
541 
542  if (ac->oc[1].layout_map != layout_map) {
543  memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
544  ac->oc[1].layout_map_tags = tags;
545  }
546  for (i = 0; i < tags; i++) {
547  int type = layout_map[i][0];
548  int id = layout_map[i][1];
549  id_map[type][id] = type_counts[type]++;
550  if (id_map[type][id] >= MAX_ELEM_ID) {
551  avpriv_request_sample(ac->avctx, "Too large remapped id");
552  return AVERROR_PATCHWELCOME;
553  }
554  }
555  // Try to sniff a reasonable channel order, otherwise output the
556  // channels in the order the PCE declared them.
558  layout = sniff_channel_order(layout_map, tags);
559  for (i = 0; i < tags; i++) {
560  int type = layout_map[i][0];
561  int id = layout_map[i][1];
562  int iid = id_map[type][id];
563  int position = layout_map[i][2];
564  // Allocate or free elements depending on if they are in the
565  // current program configuration.
566  ret = che_configure(ac, position, type, iid, &channels);
567  if (ret < 0)
568  return ret;
569  ac->tag_che_map[type][id] = ac->che[type][iid];
570  }
571  if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
572  if (layout == AV_CH_FRONT_CENTER) {
574  } else {
575  layout = 0;
576  }
577  }
578 
579  if (layout) avctx->channel_layout = layout;
580  ac->oc[1].channel_layout = layout;
581  avctx->channels = ac->oc[1].channels = channels;
582  ac->oc[1].status = oc_type;
583 
584  if (get_new_frame) {
585  if ((ret = frame_configure_elements(ac->avctx)) < 0)
586  return ret;
587  }
588 
589  return 0;
590 }
591 
592 static void flush(AVCodecContext *avctx)
593 {
594  AACContext *ac= avctx->priv_data;
595  int type, i, j;
596 
597  for (type = 3; type >= 0; type--) {
598  for (i = 0; i < MAX_ELEM_ID; i++) {
599  ChannelElement *che = ac->che[type][i];
600  if (che) {
601  for (j = 0; j <= 1; j++) {
602  memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
603  }
604  }
605  }
606  }
607 }
608 
609 /**
610  * Set up channel positions based on a default channel configuration
611  * as specified in table 1.17.
612  *
613  * @return Returns error status. 0 - OK, !0 - error
614  */
616  uint8_t (*layout_map)[3],
617  int *tags,
618  int channel_config)
619 {
620  if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
621  channel_config > 13) {
622  av_log(avctx, AV_LOG_ERROR,
623  "invalid default channel configuration (%d)\n",
624  channel_config);
625  return AVERROR_INVALIDDATA;
626  }
627  *tags = tags_per_config[channel_config];
628  memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
629  *tags * sizeof(*layout_map));
630 
631  /*
632  * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
633  * However, at least Nero AAC encoder encodes 7.1 streams using the default
634  * channel config 7, mapping the side channels of the original audio stream
635  * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
636  * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
637  * the incorrect streams as if they were correct (and as the encoder intended).
638  *
639  * As actual intended 7.1(wide) streams are very rare, default to assuming a
640  * 7.1 layout was intended.
641  */
642  if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
643  layout_map[2][2] = AAC_CHANNEL_SIDE;
644 
645  if (!ac || !ac->warned_71_wide++) {
646  av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
647  " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
648  " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
649  }
650  }
651 
652  return 0;
653 }
654 
655 static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
656 {
657  /* For PCE based channel configurations map the channels solely based
658  * on tags. */
659  if (!ac->oc[1].m4ac.chan_config) {
660  return ac->tag_che_map[type][elem_id];
661  }
662  // Allow single CPE stereo files to be signalled with mono configuration.
663  if (!ac->tags_mapped && type == TYPE_CPE &&
664  ac->oc[1].m4ac.chan_config == 1) {
665  uint8_t layout_map[MAX_ELEM_ID*4][3];
666  int layout_map_tags;
668 
669  av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
670 
671  if (set_default_channel_config(ac, ac->avctx, layout_map,
672  &layout_map_tags, 2) < 0)
673  return NULL;
674  if (output_configure(ac, layout_map, layout_map_tags,
675  OC_TRIAL_FRAME, 1) < 0)
676  return NULL;
677 
678  ac->oc[1].m4ac.chan_config = 2;
679  ac->oc[1].m4ac.ps = 0;
680  }
681  // And vice-versa
682  if (!ac->tags_mapped && type == TYPE_SCE &&
683  ac->oc[1].m4ac.chan_config == 2) {
684  uint8_t layout_map[MAX_ELEM_ID * 4][3];
685  int layout_map_tags;
687 
688  av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
689 
690  if (set_default_channel_config(ac, ac->avctx, layout_map,
691  &layout_map_tags, 1) < 0)
692  return NULL;
693  if (output_configure(ac, layout_map, layout_map_tags,
694  OC_TRIAL_FRAME, 1) < 0)
695  return NULL;
696 
697  ac->oc[1].m4ac.chan_config = 1;
698  if (ac->oc[1].m4ac.sbr)
699  ac->oc[1].m4ac.ps = -1;
700  }
701  /* For indexed channel configurations map the channels solely based
702  * on position. */
703  switch (ac->oc[1].m4ac.chan_config) {
704  case 13:
705  if (ac->tags_mapped > 3 && ((type == TYPE_CPE && elem_id < 8) ||
706  (type == TYPE_SCE && elem_id < 6) ||
707  (type == TYPE_LFE && elem_id < 2))) {
708  ac->tags_mapped++;
709  return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id];
710  }
711  case 12:
712  case 7:
713  if (ac->tags_mapped == 3 && type == TYPE_CPE) {
714  ac->tags_mapped++;
715  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
716  }
717  case 11:
718  if (ac->tags_mapped == 2 &&
719  ac->oc[1].m4ac.chan_config == 11 &&
720  type == TYPE_SCE) {
721  ac->tags_mapped++;
722  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
723  }
724  case 6:
725  /* Some streams incorrectly code 5.1 audio as
726  * SCE[0] CPE[0] CPE[1] SCE[1]
727  * instead of
728  * SCE[0] CPE[0] CPE[1] LFE[0].
729  * If we seem to have encountered such a stream, transfer
730  * the LFE[0] element to the SCE[1]'s mapping */
731  if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
732  if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
734  "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
735  type == TYPE_SCE ? "SCE" : "LFE", elem_id);
736  ac->warned_remapping_once++;
737  }
738  ac->tags_mapped++;
739  return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
740  }
741  case 5:
742  if (ac->tags_mapped == 2 && type == TYPE_CPE) {
743  ac->tags_mapped++;
744  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
745  }
746  case 4:
747  /* Some streams incorrectly code 4.0 audio as
748  * SCE[0] CPE[0] LFE[0]
749  * instead of
750  * SCE[0] CPE[0] SCE[1].
751  * If we seem to have encountered such a stream, transfer
752  * the SCE[1] element to the LFE[0]'s mapping */
753  if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
754  if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
756  "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
757  type == TYPE_SCE ? "SCE" : "LFE", elem_id);
758  ac->warned_remapping_once++;
759  }
760  ac->tags_mapped++;
761  return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
762  }
763  if (ac->tags_mapped == 2 &&
764  ac->oc[1].m4ac.chan_config == 4 &&
765  type == TYPE_SCE) {
766  ac->tags_mapped++;
767  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
768  }
769  case 3:
770  case 2:
771  if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
772  type == TYPE_CPE) {
773  ac->tags_mapped++;
774  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
775  } else if (ac->oc[1].m4ac.chan_config == 2) {
776  return NULL;
777  }
778  case 1:
779  if (!ac->tags_mapped && type == TYPE_SCE) {
780  ac->tags_mapped++;
781  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
782  }
783  default:
784  return NULL;
785  }
786 }
787 
788 /**
789  * Decode an array of 4 bit element IDs, optionally interleaved with a
790  * stereo/mono switching bit.
791  *
792  * @param type speaker type/position for these channels
793  */
794 static void decode_channel_map(uint8_t layout_map[][3],
795  enum ChannelPosition type,
796  GetBitContext *gb, int n)
797 {
798  while (n--) {
799  enum RawDataBlockType syn_ele;
800  switch (type) {
801  case AAC_CHANNEL_FRONT:
802  case AAC_CHANNEL_BACK:
803  case AAC_CHANNEL_SIDE:
804  syn_ele = get_bits1(gb);
805  break;
806  case AAC_CHANNEL_CC:
807  skip_bits1(gb);
808  syn_ele = TYPE_CCE;
809  break;
810  case AAC_CHANNEL_LFE:
811  syn_ele = TYPE_LFE;
812  break;
813  default:
814  // AAC_CHANNEL_OFF has no channel map
815  av_assert0(0);
816  }
817  layout_map[0][0] = syn_ele;
818  layout_map[0][1] = get_bits(gb, 4);
819  layout_map[0][2] = type;
820  layout_map++;
821  }
822 }
823 
824 static inline void relative_align_get_bits(GetBitContext *gb,
825  int reference_position) {
826  int n = (reference_position - get_bits_count(gb) & 7);
827  if (n)
828  skip_bits(gb, n);
829 }
830 
831 /**
832  * Decode program configuration element; reference: table 4.2.
833  *
834  * @return Returns error status. 0 - OK, !0 - error
835  */
836 static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
837  uint8_t (*layout_map)[3],
838  GetBitContext *gb, int byte_align_ref)
839 {
840  int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
841  int sampling_index;
842  int comment_len;
843  int tags;
844 
845  skip_bits(gb, 2); // object_type
846 
847  sampling_index = get_bits(gb, 4);
848  if (m4ac->sampling_index != sampling_index)
849  av_log(avctx, AV_LOG_WARNING,
850  "Sample rate index in program config element does not "
851  "match the sample rate index configured by the container.\n");
852 
853  num_front = get_bits(gb, 4);
854  num_side = get_bits(gb, 4);
855  num_back = get_bits(gb, 4);
856  num_lfe = get_bits(gb, 2);
857  num_assoc_data = get_bits(gb, 3);
858  num_cc = get_bits(gb, 4);
859 
860  if (get_bits1(gb))
861  skip_bits(gb, 4); // mono_mixdown_tag
862  if (get_bits1(gb))
863  skip_bits(gb, 4); // stereo_mixdown_tag
864 
865  if (get_bits1(gb))
866  skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
867 
868  if (get_bits_left(gb) < 5 * (num_front + num_side + num_back + num_cc) + 4 *(num_lfe + num_assoc_data + num_cc)) {
869  av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
870  return -1;
871  }
872  decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front);
873  tags = num_front;
874  decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side);
875  tags += num_side;
876  decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back);
877  tags += num_back;
878  decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe);
879  tags += num_lfe;
880 
881  skip_bits_long(gb, 4 * num_assoc_data);
882 
883  decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc);
884  tags += num_cc;
885 
886  relative_align_get_bits(gb, byte_align_ref);
887 
888  /* comment field, first byte is length */
889  comment_len = get_bits(gb, 8) * 8;
890  if (get_bits_left(gb) < comment_len) {
891  av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
892  return AVERROR_INVALIDDATA;
893  }
894  skip_bits_long(gb, comment_len);
895  return tags;
896 }
897 
898 /**
899  * Decode GA "General Audio" specific configuration; reference: table 4.1.
900  *
901  * @param ac pointer to AACContext, may be null
902  * @param avctx pointer to AVCCodecContext, used for logging
903  *
904  * @return Returns error status. 0 - OK, !0 - error
905  */
907  GetBitContext *gb,
908  int get_bit_alignment,
909  MPEG4AudioConfig *m4ac,
910  int channel_config)
911 {
912  int extension_flag, ret, ep_config, res_flags;
913  uint8_t layout_map[MAX_ELEM_ID*4][3];
914  int tags = 0;
915 
916 #if USE_FIXED
917  if (get_bits1(gb)) { // frameLengthFlag
918  avpriv_report_missing_feature(avctx, "Fixed point 960/120 MDCT window");
919  return AVERROR_PATCHWELCOME;
920  }
921  m4ac->frame_length_short = 0;
922 #else
923  m4ac->frame_length_short = get_bits1(gb);
924  if (m4ac->frame_length_short && m4ac->sbr == 1) {
925  avpriv_report_missing_feature(avctx, "SBR with 960 frame length");
926  if (ac) ac->warned_960_sbr = 1;
927  m4ac->sbr = 0;
928  m4ac->ps = 0;
929  }
930 #endif
931 
932  if (get_bits1(gb)) // dependsOnCoreCoder
933  skip_bits(gb, 14); // coreCoderDelay
934  extension_flag = get_bits1(gb);
935 
936  if (m4ac->object_type == AOT_AAC_SCALABLE ||
938  skip_bits(gb, 3); // layerNr
939 
940  if (channel_config == 0) {
941  skip_bits(gb, 4); // element_instance_tag
942  tags = decode_pce(avctx, m4ac, layout_map, gb, get_bit_alignment);
943  if (tags < 0)
944  return tags;
945  } else {
946  if ((ret = set_default_channel_config(ac, avctx, layout_map,
947  &tags, channel_config)))
948  return ret;
949  }
950 
951  if (count_channels(layout_map, tags) > 1) {
952  m4ac->ps = 0;
953  } else if (m4ac->sbr == 1 && m4ac->ps == -1)
954  m4ac->ps = 1;
955 
956  if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
957  return ret;
958 
959  if (extension_flag) {
960  switch (m4ac->object_type) {
961  case AOT_ER_BSAC:
962  skip_bits(gb, 5); // numOfSubFrame
963  skip_bits(gb, 11); // layer_length
964  break;
965  case AOT_ER_AAC_LC:
966  case AOT_ER_AAC_LTP:
967  case AOT_ER_AAC_SCALABLE:
968  case AOT_ER_AAC_LD:
969  res_flags = get_bits(gb, 3);
970  if (res_flags) {
972  "AAC data resilience (flags %x)",
973  res_flags);
974  return AVERROR_PATCHWELCOME;
975  }
976  break;
977  }
978  skip_bits1(gb); // extensionFlag3 (TBD in version 3)
979  }
980  switch (m4ac->object_type) {
981  case AOT_ER_AAC_LC:
982  case AOT_ER_AAC_LTP:
983  case AOT_ER_AAC_SCALABLE:
984  case AOT_ER_AAC_LD:
985  ep_config = get_bits(gb, 2);
986  if (ep_config) {
988  "epConfig %d", ep_config);
989  return AVERROR_PATCHWELCOME;
990  }
991  }
992  return 0;
993 }
994 
996  GetBitContext *gb,
997  MPEG4AudioConfig *m4ac,
998  int channel_config)
999 {
1000  int ret, ep_config, res_flags;
1001  uint8_t layout_map[MAX_ELEM_ID*4][3];
1002  int tags = 0;
1003  const int ELDEXT_TERM = 0;
1004 
1005  m4ac->ps = 0;
1006  m4ac->sbr = 0;
1007 #if USE_FIXED
1008  if (get_bits1(gb)) { // frameLengthFlag
1009  avpriv_request_sample(avctx, "960/120 MDCT window");
1010  return AVERROR_PATCHWELCOME;
1011  }
1012 #else
1013  m4ac->frame_length_short = get_bits1(gb);
1014 #endif
1015  res_flags = get_bits(gb, 3);
1016  if (res_flags) {
1018  "AAC data resilience (flags %x)",
1019  res_flags);
1020  return AVERROR_PATCHWELCOME;
1021  }
1022 
1023  if (get_bits1(gb)) { // ldSbrPresentFlag
1025  "Low Delay SBR");
1026  return AVERROR_PATCHWELCOME;
1027  }
1028 
1029  while (get_bits(gb, 4) != ELDEXT_TERM) {
1030  int len = get_bits(gb, 4);
1031  if (len == 15)
1032  len += get_bits(gb, 8);
1033  if (len == 15 + 255)
1034  len += get_bits(gb, 16);
1035  if (get_bits_left(gb) < len * 8 + 4) {
1036  av_log(avctx, AV_LOG_ERROR, overread_err);
1037  return AVERROR_INVALIDDATA;
1038  }
1039  skip_bits_long(gb, 8 * len);
1040  }
1041 
1042  if ((ret = set_default_channel_config(ac, avctx, layout_map,
1043  &tags, channel_config)))
1044  return ret;
1045 
1046  if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
1047  return ret;
1048 
1049  ep_config = get_bits(gb, 2);
1050  if (ep_config) {
1052  "epConfig %d", ep_config);
1053  return AVERROR_PATCHWELCOME;
1054  }
1055  return 0;
1056 }
1057 
1058 /**
1059  * Decode audio specific configuration; reference: table 1.13.
1060  *
1061  * @param ac pointer to AACContext, may be null
1062  * @param avctx pointer to AVCCodecContext, used for logging
1063  * @param m4ac pointer to MPEG4AudioConfig, used for parsing
1064  * @param gb buffer holding an audio specific config
1065  * @param get_bit_alignment relative alignment for byte align operations
1066  * @param sync_extension look for an appended sync extension
1067  *
1068  * @return Returns error status or number of consumed bits. <0 - error
1069  */
1071  AVCodecContext *avctx,
1072  MPEG4AudioConfig *m4ac,
1073  GetBitContext *gb,
1074  int get_bit_alignment,
1075  int sync_extension)
1076 {
1077  int i, ret;
1078  GetBitContext gbc = *gb;
1079  MPEG4AudioConfig m4ac_bak = *m4ac;
1080 
1081  if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension, avctx)) < 0) {
1082  *m4ac = m4ac_bak;
1083  return AVERROR_INVALIDDATA;
1084  }
1085 
1086  if (m4ac->sampling_index > 12) {
1087  av_log(avctx, AV_LOG_ERROR,
1088  "invalid sampling rate index %d\n",
1089  m4ac->sampling_index);
1090  *m4ac = m4ac_bak;
1091  return AVERROR_INVALIDDATA;
1092  }
1093  if (m4ac->object_type == AOT_ER_AAC_LD &&
1094  (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
1095  av_log(avctx, AV_LOG_ERROR,
1096  "invalid low delay sampling rate index %d\n",
1097  m4ac->sampling_index);
1098  *m4ac = m4ac_bak;
1099  return AVERROR_INVALIDDATA;
1100  }
1101 
1102  skip_bits_long(gb, i);
1103 
1104  switch (m4ac->object_type) {
1105  case AOT_AAC_MAIN:
1106  case AOT_AAC_LC:
1107  case AOT_AAC_SSR:
1108  case AOT_AAC_LTP:
1109  case AOT_ER_AAC_LC:
1110  case AOT_ER_AAC_LD:
1111  if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment,
1112  m4ac, m4ac->chan_config)) < 0)
1113  return ret;
1114  break;
1115  case AOT_ER_AAC_ELD:
1116  if ((ret = decode_eld_specific_config(ac, avctx, gb,
1117  m4ac, m4ac->chan_config)) < 0)
1118  return ret;
1119  break;
1120  default:
1122  "Audio object type %s%d",
1123  m4ac->sbr == 1 ? "SBR+" : "",
1124  m4ac->object_type);
1125  return AVERROR(ENOSYS);
1126  }
1127 
1128  ff_dlog(avctx,
1129  "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
1130  m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
1131  m4ac->sample_rate, m4ac->sbr,
1132  m4ac->ps);
1133 
1134  return get_bits_count(gb);
1135 }
1136 
1138  AVCodecContext *avctx,
1139  MPEG4AudioConfig *m4ac,
1140  const uint8_t *data, int64_t bit_size,
1141  int sync_extension)
1142 {
1143  int i, ret;
1144  GetBitContext gb;
1145 
1146  if (bit_size < 0 || bit_size > INT_MAX) {
1147  av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n");
1148  return AVERROR_INVALIDDATA;
1149  }
1150 
1151  ff_dlog(avctx, "audio specific config size %d\n", (int)bit_size >> 3);
1152  for (i = 0; i < bit_size >> 3; i++)
1153  ff_dlog(avctx, "%02x ", data[i]);
1154  ff_dlog(avctx, "\n");
1155 
1156  if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
1157  return ret;
1158 
1159  return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0,
1160  sync_extension);
1161 }
1162 
1163 /**
1164  * linear congruential pseudorandom number generator
1165  *
1166  * @param previous_val pointer to the current state of the generator
1167  *
1168  * @return Returns a 32-bit pseudorandom integer
1169  */
1170 static av_always_inline int lcg_random(unsigned previous_val)
1171 {
1172  union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
1173  return v.s;
1174 }
1175 
1177 {
1178  int i;
1179  for (i = 0; i < MAX_PREDICTORS; i++)
1180  reset_predict_state(&ps[i]);
1181 }
1182 
1183 static int sample_rate_idx (int rate)
1184 {
1185  if (92017 <= rate) return 0;
1186  else if (75132 <= rate) return 1;
1187  else if (55426 <= rate) return 2;
1188  else if (46009 <= rate) return 3;
1189  else if (37566 <= rate) return 4;
1190  else if (27713 <= rate) return 5;
1191  else if (23004 <= rate) return 6;
1192  else if (18783 <= rate) return 7;
1193  else if (13856 <= rate) return 8;
1194  else if (11502 <= rate) return 9;
1195  else if (9391 <= rate) return 10;
1196  else return 11;
1197 }
1198 
1199 static void reset_predictor_group(PredictorState *ps, int group_num)
1200 {
1201  int i;
1202  for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
1203  reset_predict_state(&ps[i]);
1204 }
1205 
1206 static void aacdec_init(AACContext *ac);
1207 
1209 {
1210  static VLC_TYPE vlc_buf[304 + 270 + 550 + 300 + 328 +
1211  294 + 306 + 268 + 510 + 366 + 462][2];
1212  for (unsigned i = 0, offset = 0; i < 11; i++) {
1217  sizeof(ff_aac_spectral_bits[i][0]),
1219  sizeof(ff_aac_spectral_codes[i][0]),
1221  sizeof(ff_aac_codebook_vector_idx[i][0]),
1224  }
1225 
1227 
1228  ff_aac_tableinit();
1229 
1233  sizeof(ff_aac_scalefactor_bits[0]),
1234  sizeof(ff_aac_scalefactor_bits[0]),
1236  sizeof(ff_aac_scalefactor_code[0]),
1237  sizeof(ff_aac_scalefactor_code[0]),
1238  352);
1239 
1240  // window initialization
1241 #if !USE_FIXED
1248 #else
1249  AAC_RENAME(ff_kbd_window_init)(AAC_RENAME2(aac_kbd_long_1024), 4.0, 1024);
1250  AAC_RENAME(ff_kbd_window_init)(AAC_RENAME2(aac_kbd_short_128), 6.0, 128);
1252 #endif
1253 
1255 }
1256 
1258 
1260 {
1261  AACContext *ac = avctx->priv_data;
1262  int ret;
1263 
1264  if (avctx->sample_rate > 96000)
1265  return AVERROR_INVALIDDATA;
1266 
1268  if (ret != 0)
1269  return AVERROR_UNKNOWN;
1270 
1271  ac->avctx = avctx;
1272  ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
1273 
1274  aacdec_init(ac);
1275 #if USE_FIXED
1276  avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
1277 #else
1278  avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1279 #endif /* USE_FIXED */
1280 
1281  if (avctx->extradata_size > 0) {
1282  if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
1283  avctx->extradata,
1284  avctx->extradata_size * 8LL,
1285  1)) < 0)
1286  return ret;
1287  } else {
1288  int sr, i;
1289  uint8_t layout_map[MAX_ELEM_ID*4][3];
1290  int layout_map_tags;
1291 
1292  sr = sample_rate_idx(avctx->sample_rate);
1293  ac->oc[1].m4ac.sampling_index = sr;
1294  ac->oc[1].m4ac.channels = avctx->channels;
1295  ac->oc[1].m4ac.sbr = -1;
1296  ac->oc[1].m4ac.ps = -1;
1297 
1298  for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
1299  if (ff_mpeg4audio_channels[i] == avctx->channels)
1300  break;
1302  i = 0;
1303  }
1304  ac->oc[1].m4ac.chan_config = i;
1305 
1306  if (ac->oc[1].m4ac.chan_config) {
1307  int ret = set_default_channel_config(ac, avctx, layout_map,
1308  &layout_map_tags, ac->oc[1].m4ac.chan_config);
1309  if (!ret)
1310  output_configure(ac, layout_map, layout_map_tags,
1311  OC_GLOBAL_HDR, 0);
1312  else if (avctx->err_recognition & AV_EF_EXPLODE)
1313  return AVERROR_INVALIDDATA;
1314  }
1315  }
1316 
1317  if (avctx->channels > MAX_CHANNELS) {
1318  av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
1319  return AVERROR_INVALIDDATA;
1320  }
1321 
1322 #if USE_FIXED
1324 #else
1326 #endif /* USE_FIXED */
1327  if (!ac->fdsp) {
1328  return AVERROR(ENOMEM);
1329  }
1330 
1331  ac->random_state = 0x1f2e3d4c;
1332 
1333  AAC_RENAME_32(ff_mdct_init)(&ac->mdct, 11, 1, 1.0 / RANGE15(1024.0));
1334  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld, 10, 1, 1.0 / RANGE15(512.0));
1335  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small, 8, 1, 1.0 / RANGE15(128.0));
1336  AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp, 11, 0, RANGE15(-2.0));
1337 #if !USE_FIXED
1338  ret = ff_mdct15_init(&ac->mdct120, 1, 3, 1.0f/(16*1024*120*2));
1339  if (ret < 0)
1340  return ret;
1341  ret = ff_mdct15_init(&ac->mdct480, 1, 5, 1.0f/(16*1024*960));
1342  if (ret < 0)
1343  return ret;
1344  ret = ff_mdct15_init(&ac->mdct960, 1, 6, 1.0f/(16*1024*960*2));
1345  if (ret < 0)
1346  return ret;
1347 #endif
1348 
1349  return 0;
1350 }
1351 
1352 /**
1353  * Skip data_stream_element; reference: table 4.10.
1354  */
1356 {
1357  int byte_align = get_bits1(gb);
1358  int count = get_bits(gb, 8);
1359  if (count == 255)
1360  count += get_bits(gb, 8);
1361  if (byte_align)
1362  align_get_bits(gb);
1363 
1364  if (get_bits_left(gb) < 8 * count) {
1365  av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
1366  return AVERROR_INVALIDDATA;
1367  }
1368  skip_bits_long(gb, 8 * count);
1369  return 0;
1370 }
1371 
1373  GetBitContext *gb)
1374 {
1375  int sfb;
1376  if (get_bits1(gb)) {
1377  ics->predictor_reset_group = get_bits(gb, 5);
1378  if (ics->predictor_reset_group == 0 ||
1379  ics->predictor_reset_group > 30) {
1380  av_log(ac->avctx, AV_LOG_ERROR,
1381  "Invalid Predictor Reset Group.\n");
1382  return AVERROR_INVALIDDATA;
1383  }
1384  }
1385  for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
1386  ics->prediction_used[sfb] = get_bits1(gb);
1387  }
1388  return 0;
1389 }
1390 
1391 /**
1392  * Decode Long Term Prediction data; reference: table 4.xx.
1393  */
1395  GetBitContext *gb, uint8_t max_sfb)
1396 {
1397  int sfb;
1398 
1399  ltp->lag = get_bits(gb, 11);
1400  ltp->coef = ltp_coef[get_bits(gb, 3)];
1401  for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
1402  ltp->used[sfb] = get_bits1(gb);
1403 }
1404 
1405 /**
1406  * Decode Individual Channel Stream info; reference: table 4.6.
1407  */
1409  GetBitContext *gb)
1410 {
1411  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
1412  const int aot = m4ac->object_type;
1413  const int sampling_index = m4ac->sampling_index;
1414  int ret_fail = AVERROR_INVALIDDATA;
1415 
1416  if (aot != AOT_ER_AAC_ELD) {
1417  if (get_bits1(gb)) {
1418  av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
1420  return AVERROR_INVALIDDATA;
1421  }
1422  ics->window_sequence[1] = ics->window_sequence[0];
1423  ics->window_sequence[0] = get_bits(gb, 2);
1424  if (aot == AOT_ER_AAC_LD &&
1425  ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
1426  av_log(ac->avctx, AV_LOG_ERROR,
1427  "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
1428  "window sequence %d found.\n", ics->window_sequence[0]);
1430  return AVERROR_INVALIDDATA;
1431  }
1432  ics->use_kb_window[1] = ics->use_kb_window[0];
1433  ics->use_kb_window[0] = get_bits1(gb);
1434  }
1435  ics->num_window_groups = 1;
1436  ics->group_len[0] = 1;
1437  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1438  int i;
1439  ics->max_sfb = get_bits(gb, 4);
1440  for (i = 0; i < 7; i++) {
1441  if (get_bits1(gb)) {
1442  ics->group_len[ics->num_window_groups - 1]++;
1443  } else {
1444  ics->num_window_groups++;
1445  ics->group_len[ics->num_window_groups - 1] = 1;
1446  }
1447  }
1448  ics->num_windows = 8;
1449  if (m4ac->frame_length_short) {
1450  ics->swb_offset = ff_swb_offset_120[sampling_index];
1451  ics->num_swb = ff_aac_num_swb_120[sampling_index];
1452  } else {
1453  ics->swb_offset = ff_swb_offset_128[sampling_index];
1454  ics->num_swb = ff_aac_num_swb_128[sampling_index];
1455  }
1456  ics->tns_max_bands = ff_tns_max_bands_128[sampling_index];
1457  ics->predictor_present = 0;
1458  } else {
1459  ics->max_sfb = get_bits(gb, 6);
1460  ics->num_windows = 1;
1461  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
1462  if (m4ac->frame_length_short) {
1463  ics->swb_offset = ff_swb_offset_480[sampling_index];
1464  ics->num_swb = ff_aac_num_swb_480[sampling_index];
1465  ics->tns_max_bands = ff_tns_max_bands_480[sampling_index];
1466  } else {
1467  ics->swb_offset = ff_swb_offset_512[sampling_index];
1468  ics->num_swb = ff_aac_num_swb_512[sampling_index];
1469  ics->tns_max_bands = ff_tns_max_bands_512[sampling_index];
1470  }
1471  if (!ics->num_swb || !ics->swb_offset) {
1472  ret_fail = AVERROR_BUG;
1473  goto fail;
1474  }
1475  } else {
1476  if (m4ac->frame_length_short) {
1477  ics->num_swb = ff_aac_num_swb_960[sampling_index];
1478  ics->swb_offset = ff_swb_offset_960[sampling_index];
1479  } else {
1480  ics->num_swb = ff_aac_num_swb_1024[sampling_index];
1481  ics->swb_offset = ff_swb_offset_1024[sampling_index];
1482  }
1483  ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
1484  }
1485  if (aot != AOT_ER_AAC_ELD) {
1486  ics->predictor_present = get_bits1(gb);
1487  ics->predictor_reset_group = 0;
1488  }
1489  if (ics->predictor_present) {
1490  if (aot == AOT_AAC_MAIN) {
1491  if (decode_prediction(ac, ics, gb)) {
1492  goto fail;
1493  }
1494  } else if (aot == AOT_AAC_LC ||
1495  aot == AOT_ER_AAC_LC) {
1496  av_log(ac->avctx, AV_LOG_ERROR,
1497  "Prediction is not allowed in AAC-LC.\n");
1498  goto fail;
1499  } else {
1500  if (aot == AOT_ER_AAC_LD) {
1501  av_log(ac->avctx, AV_LOG_ERROR,
1502  "LTP in ER AAC LD not yet implemented.\n");
1503  ret_fail = AVERROR_PATCHWELCOME;
1504  goto fail;
1505  }
1506  if ((ics->ltp.present = get_bits(gb, 1)))
1507  decode_ltp(&ics->ltp, gb, ics->max_sfb);
1508  }
1509  }
1510  }
1511 
1512  if (ics->max_sfb > ics->num_swb) {
1513  av_log(ac->avctx, AV_LOG_ERROR,
1514  "Number of scalefactor bands in group (%d) "
1515  "exceeds limit (%d).\n",
1516  ics->max_sfb, ics->num_swb);
1517  goto fail;
1518  }
1519 
1520  return 0;
1521 fail:
1522  ics->max_sfb = 0;
1523  return ret_fail;
1524 }
1525 
1526 /**
1527  * Decode band types (section_data payload); reference: table 4.46.
1528  *
1529  * @param band_type array of the used band type
1530  * @param band_type_run_end array of the last scalefactor band of a band type run
1531  *
1532  * @return Returns error status. 0 - OK, !0 - error
1533  */
1534 static int decode_band_types(AACContext *ac, enum BandType band_type[120],
1535  int band_type_run_end[120], GetBitContext *gb,
1537 {
1538  int g, idx = 0;
1539  const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
1540  for (g = 0; g < ics->num_window_groups; g++) {
1541  int k = 0;
1542  while (k < ics->max_sfb) {
1543  uint8_t sect_end = k;
1544  int sect_len_incr;
1545  int sect_band_type = get_bits(gb, 4);
1546  if (sect_band_type == 12) {
1547  av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
1548  return AVERROR_INVALIDDATA;
1549  }
1550  do {
1551  sect_len_incr = get_bits(gb, bits);
1552  sect_end += sect_len_incr;
1553  if (get_bits_left(gb) < 0) {
1554  av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
1555  return AVERROR_INVALIDDATA;
1556  }
1557  if (sect_end > ics->max_sfb) {
1558  av_log(ac->avctx, AV_LOG_ERROR,
1559  "Number of bands (%d) exceeds limit (%d).\n",
1560  sect_end, ics->max_sfb);
1561  return AVERROR_INVALIDDATA;
1562  }
1563  } while (sect_len_incr == (1 << bits) - 1);
1564  for (; k < sect_end; k++) {
1565  band_type [idx] = sect_band_type;
1566  band_type_run_end[idx++] = sect_end;
1567  }
1568  }
1569  }
1570  return 0;
1571 }
1572 
1573 /**
1574  * Decode scalefactors; reference: table 4.47.
1575  *
1576  * @param global_gain first scalefactor value as scalefactors are differentially coded
1577  * @param band_type array of the used band type
1578  * @param band_type_run_end array of the last scalefactor band of a band type run
1579  * @param sf array of scalefactors or intensity stereo positions
1580  *
1581  * @return Returns error status. 0 - OK, !0 - error
1582  */
1584  unsigned int global_gain,
1586  enum BandType band_type[120],
1587  int band_type_run_end[120])
1588 {
1589  int g, i, idx = 0;
1590  int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 };
1591  int clipped_offset;
1592  int noise_flag = 1;
1593  for (g = 0; g < ics->num_window_groups; g++) {
1594  for (i = 0; i < ics->max_sfb;) {
1595  int run_end = band_type_run_end[idx];
1596  if (band_type[idx] == ZERO_BT) {
1597  for (; i < run_end; i++, idx++)
1598  sf[idx] = FIXR(0.);
1599  } else if ((band_type[idx] == INTENSITY_BT) ||
1600  (band_type[idx] == INTENSITY_BT2)) {
1601  for (; i < run_end; i++, idx++) {
1603  clipped_offset = av_clip(offset[2], -155, 100);
1604  if (offset[2] != clipped_offset) {
1606  "If you heard an audible artifact, there may be a bug in the decoder. "
1607  "Clipped intensity stereo position (%d -> %d)",
1608  offset[2], clipped_offset);
1609  }
1610 #if USE_FIXED
1611  sf[idx] = 100 - clipped_offset;
1612 #else
1613  sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
1614 #endif /* USE_FIXED */
1615  }
1616  } else if (band_type[idx] == NOISE_BT) {
1617  for (; i < run_end; i++, idx++) {
1618  if (noise_flag-- > 0)
1619  offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
1620  else
1622  clipped_offset = av_clip(offset[1], -100, 155);
1623  if (offset[1] != clipped_offset) {
1625  "If you heard an audible artifact, there may be a bug in the decoder. "
1626  "Clipped noise gain (%d -> %d)",
1627  offset[1], clipped_offset);
1628  }
1629 #if USE_FIXED
1630  sf[idx] = -(100 + clipped_offset);
1631 #else
1632  sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
1633 #endif /* USE_FIXED */
1634  }
1635  } else {
1636  for (; i < run_end; i++, idx++) {
1638  if (offset[0] > 255U) {
1639  av_log(ac->avctx, AV_LOG_ERROR,
1640  "Scalefactor (%d) out of range.\n", offset[0]);
1641  return AVERROR_INVALIDDATA;
1642  }
1643 #if USE_FIXED
1644  sf[idx] = -offset[0];
1645 #else
1646  sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
1647 #endif /* USE_FIXED */
1648  }
1649  }
1650  }
1651  }
1652  return 0;
1653 }
1654 
1655 /**
1656  * Decode pulse data; reference: table 4.7.
1657  */
1658 static int decode_pulses(Pulse *pulse, GetBitContext *gb,
1659  const uint16_t *swb_offset, int num_swb)
1660 {
1661  int i, pulse_swb;
1662  pulse->num_pulse = get_bits(gb, 2) + 1;
1663  pulse_swb = get_bits(gb, 6);
1664  if (pulse_swb >= num_swb)
1665  return -1;
1666  pulse->pos[0] = swb_offset[pulse_swb];
1667  pulse->pos[0] += get_bits(gb, 5);
1668  if (pulse->pos[0] >= swb_offset[num_swb])
1669  return -1;
1670  pulse->amp[0] = get_bits(gb, 4);
1671  for (i = 1; i < pulse->num_pulse; i++) {
1672  pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
1673  if (pulse->pos[i] >= swb_offset[num_swb])
1674  return -1;
1675  pulse->amp[i] = get_bits(gb, 4);
1676  }
1677  return 0;
1678 }
1679 
1680 /**
1681  * Decode Temporal Noise Shaping data; reference: table 4.48.
1682  *
1683  * @return Returns error status. 0 - OK, !0 - error
1684  */
1686  GetBitContext *gb, const IndividualChannelStream *ics)
1687 {
1688  int w, filt, i, coef_len, coef_res, coef_compress;
1689  const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
1690  const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
1691  for (w = 0; w < ics->num_windows; w++) {
1692  if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
1693  coef_res = get_bits1(gb);
1694 
1695  for (filt = 0; filt < tns->n_filt[w]; filt++) {
1696  int tmp2_idx;
1697  tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
1698 
1699  if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
1700  av_log(ac->avctx, AV_LOG_ERROR,
1701  "TNS filter order %d is greater than maximum %d.\n",
1702  tns->order[w][filt], tns_max_order);
1703  tns->order[w][filt] = 0;
1704  return AVERROR_INVALIDDATA;
1705  }
1706  if (tns->order[w][filt]) {
1707  tns->direction[w][filt] = get_bits1(gb);
1708  coef_compress = get_bits1(gb);
1709  coef_len = coef_res + 3 - coef_compress;
1710  tmp2_idx = 2 * coef_compress + coef_res;
1711 
1712  for (i = 0; i < tns->order[w][filt]; i++)
1713  tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
1714  }
1715  }
1716  }
1717  }
1718  return 0;
1719 }
1720 
1721 /**
1722  * Decode Mid/Side data; reference: table 4.54.
1723  *
1724  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
1725  * [1] mask is decoded from bitstream; [2] mask is all 1s;
1726  * [3] reserved for scalable AAC
1727  */
1729  int ms_present)
1730 {
1731  int idx;
1732  int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
1733  if (ms_present == 1) {
1734  for (idx = 0; idx < max_idx; idx++)
1735  cpe->ms_mask[idx] = get_bits1(gb);
1736  } else if (ms_present == 2) {
1737  memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
1738  }
1739 }
1740 
1741 /**
1742  * Decode spectral data; reference: table 4.50.
1743  * Dequantize and scale spectral data; reference: 4.6.3.3.
1744  *
1745  * @param coef array of dequantized, scaled spectral data
1746  * @param sf array of scalefactors or intensity stereo positions
1747  * @param pulse_present set if pulses are present
1748  * @param pulse pointer to pulse data struct
1749  * @param band_type array of the used band type
1750  *
1751  * @return Returns error status. 0 - OK, !0 - error
1752  */
1754  GetBitContext *gb, const INTFLOAT sf[120],
1755  int pulse_present, const Pulse *pulse,
1756  const IndividualChannelStream *ics,
1757  enum BandType band_type[120])
1758 {
1759  int i, k, g, idx = 0;
1760  const int c = 1024 / ics->num_windows;
1761  const uint16_t *offsets = ics->swb_offset;
1762  INTFLOAT *coef_base = coef;
1763 
1764  for (g = 0; g < ics->num_windows; g++)
1765  memset(coef + g * 128 + offsets[ics->max_sfb], 0,
1766  sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
1767 
1768  for (g = 0; g < ics->num_window_groups; g++) {
1769  unsigned g_len = ics->group_len[g];
1770 
1771  for (i = 0; i < ics->max_sfb; i++, idx++) {
1772  const unsigned cbt_m1 = band_type[idx] - 1;
1773  INTFLOAT *cfo = coef + offsets[i];
1774  int off_len = offsets[i + 1] - offsets[i];
1775  int group;
1776 
1777  if (cbt_m1 >= INTENSITY_BT2 - 1) {
1778  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1779  memset(cfo, 0, off_len * sizeof(*cfo));
1780  }
1781  } else if (cbt_m1 == NOISE_BT - 1) {
1782  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1783  INTFLOAT band_energy;
1784 #if USE_FIXED
1785  for (k = 0; k < off_len; k++) {
1787  cfo[k] = ac->random_state >> 3;
1788  }
1789 
1790  band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
1791  band_energy = fixed_sqrt(band_energy, 31);
1792  noise_scale(cfo, sf[idx], band_energy, off_len);
1793 #else
1794  float scale;
1795 
1796  for (k = 0; k < off_len; k++) {
1798  cfo[k] = ac->random_state;
1799  }
1800 
1801  band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
1802  scale = sf[idx] / sqrtf(band_energy);
1803  ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
1804 #endif /* USE_FIXED */
1805  }
1806  } else {
1807 #if !USE_FIXED
1808  const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1809 #endif /* !USE_FIXED */
1810  VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1811  OPEN_READER(re, gb);
1812 
1813  switch (cbt_m1 >> 1) {
1814  case 0:
1815  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1816  INTFLOAT *cf = cfo;
1817  int len = off_len;
1818 
1819  do {
1820  int code;
1821  unsigned cb_idx;
1822 
1823  UPDATE_CACHE(re, gb);
1824  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1825  cb_idx = code;
1826 #if USE_FIXED
1827  cf = DEC_SQUAD(cf, cb_idx);
1828 #else
1829  cf = VMUL4(cf, vq, cb_idx, sf + idx);
1830 #endif /* USE_FIXED */
1831  } while (len -= 4);
1832  }
1833  break;
1834 
1835  case 1:
1836  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1837  INTFLOAT *cf = cfo;
1838  int len = off_len;
1839 
1840  do {
1841  int code;
1842  unsigned nnz;
1843  unsigned cb_idx;
1844  uint32_t bits;
1845 
1846  UPDATE_CACHE(re, gb);
1847  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1848  cb_idx = code;
1849  nnz = cb_idx >> 8 & 15;
1850  bits = nnz ? GET_CACHE(re, gb) : 0;
1851  LAST_SKIP_BITS(re, gb, nnz);
1852 #if USE_FIXED
1853  cf = DEC_UQUAD(cf, cb_idx, bits);
1854 #else
1855  cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1856 #endif /* USE_FIXED */
1857  } while (len -= 4);
1858  }
1859  break;
1860 
1861  case 2:
1862  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1863  INTFLOAT *cf = cfo;
1864  int len = off_len;
1865 
1866  do {
1867  int code;
1868  unsigned cb_idx;
1869 
1870  UPDATE_CACHE(re, gb);
1871  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1872  cb_idx = code;
1873 #if USE_FIXED
1874  cf = DEC_SPAIR(cf, cb_idx);
1875 #else
1876  cf = VMUL2(cf, vq, cb_idx, sf + idx);
1877 #endif /* USE_FIXED */
1878  } while (len -= 2);
1879  }
1880  break;
1881 
1882  case 3:
1883  case 4:
1884  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1885  INTFLOAT *cf = cfo;
1886  int len = off_len;
1887 
1888  do {
1889  int code;
1890  unsigned nnz;
1891  unsigned cb_idx;
1892  unsigned sign;
1893 
1894  UPDATE_CACHE(re, gb);
1895  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1896  cb_idx = code;
1897  nnz = cb_idx >> 8 & 15;
1898  sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
1899  LAST_SKIP_BITS(re, gb, nnz);
1900 #if USE_FIXED
1901  cf = DEC_UPAIR(cf, cb_idx, sign);
1902 #else
1903  cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1904 #endif /* USE_FIXED */
1905  } while (len -= 2);
1906  }
1907  break;
1908 
1909  default:
1910  for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1911 #if USE_FIXED
1912  int *icf = cfo;
1913  int v;
1914 #else
1915  float *cf = cfo;
1916  uint32_t *icf = (uint32_t *) cf;
1917 #endif /* USE_FIXED */
1918  int len = off_len;
1919 
1920  do {
1921  int code;
1922  unsigned nzt, nnz;
1923  unsigned cb_idx;
1924  uint32_t bits;
1925  int j;
1926 
1927  UPDATE_CACHE(re, gb);
1928  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1929  cb_idx = code;
1930 
1931  if (cb_idx == 0x0000) {
1932  *icf++ = 0;
1933  *icf++ = 0;
1934  continue;
1935  }
1936 
1937  nnz = cb_idx >> 12;
1938  nzt = cb_idx >> 8;
1939  bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1940  LAST_SKIP_BITS(re, gb, nnz);
1941 
1942  for (j = 0; j < 2; j++) {
1943  if (nzt & 1<<j) {
1944  uint32_t b;
1945  int n;
1946  /* The total length of escape_sequence must be < 22 bits according
1947  to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1948  UPDATE_CACHE(re, gb);
1949  b = GET_CACHE(re, gb);
1950  b = 31 - av_log2(~b);
1951 
1952  if (b > 8) {
1953  av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1954  return AVERROR_INVALIDDATA;
1955  }
1956 
1957  SKIP_BITS(re, gb, b + 1);
1958  b += 4;
1959  n = (1 << b) + SHOW_UBITS(re, gb, b);
1960  LAST_SKIP_BITS(re, gb, b);
1961 #if USE_FIXED
1962  v = n;
1963  if (bits & 1U<<31)
1964  v = -v;
1965  *icf++ = v;
1966 #else
1967  *icf++ = ff_cbrt_tab[n] | (bits & 1U<<31);
1968 #endif /* USE_FIXED */
1969  bits <<= 1;
1970  } else {
1971 #if USE_FIXED
1972  v = cb_idx & 15;
1973  if (bits & 1U<<31)
1974  v = -v;
1975  *icf++ = v;
1976 #else
1977  unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1978  *icf++ = (bits & 1U<<31) | v;
1979 #endif /* USE_FIXED */
1980  bits <<= !!v;
1981  }
1982  cb_idx >>= 4;
1983  }
1984  } while (len -= 2);
1985 #if !USE_FIXED
1986  ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1987 #endif /* !USE_FIXED */
1988  }
1989  }
1990 
1991  CLOSE_READER(re, gb);
1992  }
1993  }
1994  coef += g_len << 7;
1995  }
1996 
1997  if (pulse_present) {
1998  idx = 0;
1999  for (i = 0; i < pulse->num_pulse; i++) {
2000  INTFLOAT co = coef_base[ pulse->pos[i] ];
2001  while (offsets[idx + 1] <= pulse->pos[i])
2002  idx++;
2003  if (band_type[idx] != NOISE_BT && sf[idx]) {
2004  INTFLOAT ico = -pulse->amp[i];
2005 #if USE_FIXED
2006  if (co) {
2007  ico = co + (co > 0 ? -ico : ico);
2008  }
2009  coef_base[ pulse->pos[i] ] = ico;
2010 #else
2011  if (co) {
2012  co /= sf[idx];
2013  ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
2014  }
2015  coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
2016 #endif /* USE_FIXED */
2017  }
2018  }
2019  }
2020 #if USE_FIXED
2021  coef = coef_base;
2022  idx = 0;
2023  for (g = 0; g < ics->num_window_groups; g++) {
2024  unsigned g_len = ics->group_len[g];
2025 
2026  for (i = 0; i < ics->max_sfb; i++, idx++) {
2027  const unsigned cbt_m1 = band_type[idx] - 1;
2028  int *cfo = coef + offsets[i];
2029  int off_len = offsets[i + 1] - offsets[i];
2030  int group;
2031 
2032  if (cbt_m1 < NOISE_BT - 1) {
2033  for (group = 0; group < (int)g_len; group++, cfo+=128) {
2034  ac->vector_pow43(cfo, off_len);
2035  ac->subband_scale(cfo, cfo, sf[idx], 34, off_len, ac->avctx);
2036  }
2037  }
2038  }
2039  coef += g_len << 7;
2040  }
2041 #endif /* USE_FIXED */
2042  return 0;
2043 }
2044 
2045 /**
2046  * Apply AAC-Main style frequency domain prediction.
2047  */
2049 {
2050  int sfb, k;
2051 
2052  if (!sce->ics.predictor_initialized) {
2054  sce->ics.predictor_initialized = 1;
2055  }
2056 
2057  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
2058  for (sfb = 0;
2059  sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
2060  sfb++) {
2061  for (k = sce->ics.swb_offset[sfb];
2062  k < sce->ics.swb_offset[sfb + 1];
2063  k++) {
2064  predict(&sce->predictor_state[k], &sce->coeffs[k],
2065  sce->ics.predictor_present &&
2066  sce->ics.prediction_used[sfb]);
2067  }
2068  }
2069  if (sce->ics.predictor_reset_group)
2071  sce->ics.predictor_reset_group);
2072  } else
2074 }
2075 
2077 {
2078  // wd_num, wd_test, aloc_size
2079  static const uint8_t gain_mode[4][3] = {
2080  {1, 0, 5}, // ONLY_LONG_SEQUENCE = 0,
2081  {2, 1, 2}, // LONG_START_SEQUENCE,
2082  {8, 0, 2}, // EIGHT_SHORT_SEQUENCE,
2083  {2, 1, 5}, // LONG_STOP_SEQUENCE
2084  };
2085 
2086  const int mode = sce->ics.window_sequence[0];
2087  uint8_t bd, wd, ad;
2088 
2089  // FIXME: Store the gain control data on |sce| and do something with it.
2090  uint8_t max_band = get_bits(gb, 2);
2091  for (bd = 0; bd < max_band; bd++) {
2092  for (wd = 0; wd < gain_mode[mode][0]; wd++) {
2093  uint8_t adjust_num = get_bits(gb, 3);
2094  for (ad = 0; ad < adjust_num; ad++) {
2095  skip_bits(gb, 4 + ((wd == 0 && gain_mode[mode][1])
2096  ? 4
2097  : gain_mode[mode][2]));
2098  }
2099  }
2100  }
2101 }
2102 
2103 /**
2104  * Decode an individual_channel_stream payload; reference: table 4.44.
2105  *
2106  * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
2107  * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
2108  *
2109  * @return Returns error status. 0 - OK, !0 - error
2110  */
2112  GetBitContext *gb, int common_window, int scale_flag)
2113 {
2114  Pulse pulse;
2115  TemporalNoiseShaping *tns = &sce->tns;
2116  IndividualChannelStream *ics = &sce->ics;
2117  INTFLOAT *out = sce->coeffs;
2118  int global_gain, eld_syntax, er_syntax, pulse_present = 0;
2119  int ret;
2120 
2121  eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2122  er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
2123  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
2124  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
2125  ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2126 
2127  /* This assignment is to silence a GCC warning about the variable being used
2128  * uninitialized when in fact it always is.
2129  */
2130  pulse.num_pulse = 0;
2131 
2132  global_gain = get_bits(gb, 8);
2133 
2134  if (!common_window && !scale_flag) {
2135  ret = decode_ics_info(ac, ics, gb);
2136  if (ret < 0)
2137  goto fail;
2138  }
2139 
2140  if ((ret = decode_band_types(ac, sce->band_type,
2141  sce->band_type_run_end, gb, ics)) < 0)
2142  goto fail;
2143  if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
2144  sce->band_type, sce->band_type_run_end)) < 0)
2145  goto fail;
2146 
2147  pulse_present = 0;
2148  if (!scale_flag) {
2149  if (!eld_syntax && (pulse_present = get_bits1(gb))) {
2150  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2151  av_log(ac->avctx, AV_LOG_ERROR,
2152  "Pulse tool not allowed in eight short sequence.\n");
2153  ret = AVERROR_INVALIDDATA;
2154  goto fail;
2155  }
2156  if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
2157  av_log(ac->avctx, AV_LOG_ERROR,
2158  "Pulse data corrupt or invalid.\n");
2159  ret = AVERROR_INVALIDDATA;
2160  goto fail;
2161  }
2162  }
2163  tns->present = get_bits1(gb);
2164  if (tns->present && !er_syntax) {
2165  ret = decode_tns(ac, tns, gb, ics);
2166  if (ret < 0)
2167  goto fail;
2168  }
2169  if (!eld_syntax && get_bits1(gb)) {
2170  decode_gain_control(sce, gb);
2171  if (!ac->warned_gain_control) {
2172  avpriv_report_missing_feature(ac->avctx, "Gain control");
2173  ac->warned_gain_control = 1;
2174  }
2175  }
2176  // I see no textual basis in the spec for this occurring after SSR gain
2177  // control, but this is what both reference and real implmentations do
2178  if (tns->present && er_syntax) {
2179  ret = decode_tns(ac, tns, gb, ics);
2180  if (ret < 0)
2181  goto fail;
2182  }
2183  }
2184 
2185  ret = decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
2186  &pulse, ics, sce->band_type);
2187  if (ret < 0)
2188  goto fail;
2189 
2190  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
2191  apply_prediction(ac, sce);
2192 
2193  return 0;
2194 fail:
2195  tns->present = 0;
2196  return ret;
2197 }
2198 
2199 /**
2200  * Mid/Side stereo decoding; reference: 4.6.8.1.3.
2201  */
2203 {
2204  const IndividualChannelStream *ics = &cpe->ch[0].ics;
2205  INTFLOAT *ch0 = cpe->ch[0].coeffs;
2206  INTFLOAT *ch1 = cpe->ch[1].coeffs;
2207  int g, i, group, idx = 0;
2208  const uint16_t *offsets = ics->swb_offset;
2209  for (g = 0; g < ics->num_window_groups; g++) {
2210  for (i = 0; i < ics->max_sfb; i++, idx++) {
2211  if (cpe->ms_mask[idx] &&
2212  cpe->ch[0].band_type[idx] < NOISE_BT &&
2213  cpe->ch[1].band_type[idx] < NOISE_BT) {
2214 #if USE_FIXED
2215  for (group = 0; group < ics->group_len[g]; group++) {
2216  ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
2217  ch1 + group * 128 + offsets[i],
2218  offsets[i+1] - offsets[i]);
2219 #else
2220  for (group = 0; group < ics->group_len[g]; group++) {
2221  ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
2222  ch1 + group * 128 + offsets[i],
2223  offsets[i+1] - offsets[i]);
2224 #endif /* USE_FIXED */
2225  }
2226  }
2227  }
2228  ch0 += ics->group_len[g] * 128;
2229  ch1 += ics->group_len[g] * 128;
2230  }
2231 }
2232 
2233 /**
2234  * intensity stereo decoding; reference: 4.6.8.2.3
2235  *
2236  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
2237  * [1] mask is decoded from bitstream; [2] mask is all 1s;
2238  * [3] reserved for scalable AAC
2239  */
2241  ChannelElement *cpe, int ms_present)
2242 {
2243  const IndividualChannelStream *ics = &cpe->ch[1].ics;
2244  SingleChannelElement *sce1 = &cpe->ch[1];
2245  INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
2246  const uint16_t *offsets = ics->swb_offset;
2247  int g, group, i, idx = 0;
2248  int c;
2249  INTFLOAT scale;
2250  for (g = 0; g < ics->num_window_groups; g++) {
2251  for (i = 0; i < ics->max_sfb;) {
2252  if (sce1->band_type[idx] == INTENSITY_BT ||
2253  sce1->band_type[idx] == INTENSITY_BT2) {
2254  const int bt_run_end = sce1->band_type_run_end[idx];
2255  for (; i < bt_run_end; i++, idx++) {
2256  c = -1 + 2 * (sce1->band_type[idx] - 14);
2257  if (ms_present)
2258  c *= 1 - 2 * cpe->ms_mask[idx];
2259  scale = c * sce1->sf[idx];
2260  for (group = 0; group < ics->group_len[g]; group++)
2261 #if USE_FIXED
2262  ac->subband_scale(coef1 + group * 128 + offsets[i],
2263  coef0 + group * 128 + offsets[i],
2264  scale,
2265  23,
2266  offsets[i + 1] - offsets[i] ,ac->avctx);
2267 #else
2268  ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
2269  coef0 + group * 128 + offsets[i],
2270  scale,
2271  offsets[i + 1] - offsets[i]);
2272 #endif /* USE_FIXED */
2273  }
2274  } else {
2275  int bt_run_end = sce1->band_type_run_end[idx];
2276  idx += bt_run_end - i;
2277  i = bt_run_end;
2278  }
2279  }
2280  coef0 += ics->group_len[g] * 128;
2281  coef1 += ics->group_len[g] * 128;
2282  }
2283 }
2284 
2285 /**
2286  * Decode a channel_pair_element; reference: table 4.4.
2287  *
2288  * @return Returns error status. 0 - OK, !0 - error
2289  */
2291 {
2292  int i, ret, common_window, ms_present = 0;
2293  int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2294 
2295  common_window = eld_syntax || get_bits1(gb);
2296  if (common_window) {
2297  if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
2298  return AVERROR_INVALIDDATA;
2299  i = cpe->ch[1].ics.use_kb_window[0];
2300  cpe->ch[1].ics = cpe->ch[0].ics;
2301  cpe->ch[1].ics.use_kb_window[1] = i;
2302  if (cpe->ch[1].ics.predictor_present &&
2303  (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
2304  if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
2305  decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
2306  ms_present = get_bits(gb, 2);
2307  if (ms_present == 3) {
2308  av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
2309  return AVERROR_INVALIDDATA;
2310  } else if (ms_present)
2311  decode_mid_side_stereo(cpe, gb, ms_present);
2312  }
2313  if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
2314  return ret;
2315  if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
2316  return ret;
2317 
2318  if (common_window) {
2319  if (ms_present)
2320  apply_mid_side_stereo(ac, cpe);
2321  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
2322  apply_prediction(ac, &cpe->ch[0]);
2323  apply_prediction(ac, &cpe->ch[1]);
2324  }
2325  }
2326 
2327  apply_intensity_stereo(ac, cpe, ms_present);
2328  return 0;
2329 }
2330 
2331 static const float cce_scale[] = {
2332  1.09050773266525765921, //2^(1/8)
2333  1.18920711500272106672, //2^(1/4)
2334  M_SQRT2,
2335  2,
2336 };
2337 
2338 /**
2339  * Decode coupling_channel_element; reference: table 4.8.
2340  *
2341  * @return Returns error status. 0 - OK, !0 - error
2342  */
2344 {
2345  int num_gain = 0;
2346  int c, g, sfb, ret;
2347  int sign;
2348  INTFLOAT scale;
2349  SingleChannelElement *sce = &che->ch[0];
2350  ChannelCoupling *coup = &che->coup;
2351 
2352  coup->coupling_point = 2 * get_bits1(gb);
2353  coup->num_coupled = get_bits(gb, 3);
2354  for (c = 0; c <= coup->num_coupled; c++) {
2355  num_gain++;
2356  coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
2357  coup->id_select[c] = get_bits(gb, 4);
2358  if (coup->type[c] == TYPE_CPE) {
2359  coup->ch_select[c] = get_bits(gb, 2);
2360  if (coup->ch_select[c] == 3)
2361  num_gain++;
2362  } else
2363  coup->ch_select[c] = 2;
2364  }
2365  coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
2366 
2367  sign = get_bits(gb, 1);
2368 #if USE_FIXED
2369  scale = get_bits(gb, 2);
2370 #else
2371  scale = cce_scale[get_bits(gb, 2)];
2372 #endif
2373 
2374  if ((ret = decode_ics(ac, sce, gb, 0, 0)))
2375  return ret;
2376 
2377  for (c = 0; c < num_gain; c++) {
2378  int idx = 0;
2379  int cge = 1;
2380  int gain = 0;
2381  INTFLOAT gain_cache = FIXR10(1.);
2382  if (c) {
2383  cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
2384  gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
2385  gain_cache = GET_GAIN(scale, gain);
2386 #if USE_FIXED
2387  if ((abs(gain_cache)-1024) >> 3 > 30)
2388  return AVERROR(ERANGE);
2389 #endif
2390  }
2391  if (coup->coupling_point == AFTER_IMDCT) {
2392  coup->gain[c][0] = gain_cache;
2393  } else {
2394  for (g = 0; g < sce->ics.num_window_groups; g++) {
2395  for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
2396  if (sce->band_type[idx] != ZERO_BT) {
2397  if (!cge) {
2398  int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
2399  if (t) {
2400  int s = 1;
2401  t = gain += t;
2402  if (sign) {
2403  s -= 2 * (t & 0x1);
2404  t >>= 1;
2405  }
2406  gain_cache = GET_GAIN(scale, t) * s;
2407 #if USE_FIXED
2408  if ((abs(gain_cache)-1024) >> 3 > 30)
2409  return AVERROR(ERANGE);
2410 #endif
2411  }
2412  }
2413  coup->gain[c][idx] = gain_cache;
2414  }
2415  }
2416  }
2417  }
2418  }
2419  return 0;
2420 }
2421 
2422 /**
2423  * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
2424  *
2425  * @return Returns number of bytes consumed.
2426  */
2428  GetBitContext *gb)
2429 {
2430  int i;
2431  int num_excl_chan = 0;
2432 
2433  do {
2434  for (i = 0; i < 7; i++)
2435  che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
2436  } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
2437 
2438  return num_excl_chan / 7;
2439 }
2440 
2441 /**
2442  * Decode dynamic range information; reference: table 4.52.
2443  *
2444  * @return Returns number of bytes consumed.
2445  */
2447  GetBitContext *gb)
2448 {
2449  int n = 1;
2450  int drc_num_bands = 1;
2451  int i;
2452 
2453  /* pce_tag_present? */
2454  if (get_bits1(gb)) {
2455  che_drc->pce_instance_tag = get_bits(gb, 4);
2456  skip_bits(gb, 4); // tag_reserved_bits
2457  n++;
2458  }
2459 
2460  /* excluded_chns_present? */
2461  if (get_bits1(gb)) {
2462  n += decode_drc_channel_exclusions(che_drc, gb);
2463  }
2464 
2465  /* drc_bands_present? */
2466  if (get_bits1(gb)) {
2467  che_drc->band_incr = get_bits(gb, 4);
2468  che_drc->interpolation_scheme = get_bits(gb, 4);
2469  n++;
2470  drc_num_bands += che_drc->band_incr;
2471  for (i = 0; i < drc_num_bands; i++) {
2472  che_drc->band_top[i] = get_bits(gb, 8);
2473  n++;
2474  }
2475  }
2476 
2477  /* prog_ref_level_present? */
2478  if (get_bits1(gb)) {
2479  che_drc->prog_ref_level = get_bits(gb, 7);
2480  skip_bits1(gb); // prog_ref_level_reserved_bits
2481  n++;
2482  }
2483 
2484  for (i = 0; i < drc_num_bands; i++) {
2485  che_drc->dyn_rng_sgn[i] = get_bits1(gb);
2486  che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
2487  n++;
2488  }
2489 
2490  return n;
2491 }
2492 
2493 static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
2494  uint8_t buf[256];
2495  int i, major, minor;
2496 
2497  if (len < 13+7*8)
2498  goto unknown;
2499 
2500  get_bits(gb, 13); len -= 13;
2501 
2502  for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
2503  buf[i] = get_bits(gb, 8);
2504 
2505  buf[i] = 0;
2506  if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
2507  av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
2508 
2509  if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
2510  ac->avctx->internal->skip_samples = 1024;
2511  }
2512 
2513 unknown:
2514  skip_bits_long(gb, len);
2515 
2516  return 0;
2517 }
2518 
2519 /**
2520  * Decode extension data (incomplete); reference: table 4.51.
2521  *
2522  * @param cnt length of TYPE_FIL syntactic element in bytes
2523  *
2524  * @return Returns number of bytes consumed
2525  */
2527  ChannelElement *che, enum RawDataBlockType elem_type)
2528 {
2529  int crc_flag = 0;
2530  int res = cnt;
2531  int type = get_bits(gb, 4);
2532 
2533  if (ac->avctx->debug & FF_DEBUG_STARTCODE)
2534  av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
2535 
2536  switch (type) { // extension type
2537  case EXT_SBR_DATA_CRC:
2538  crc_flag++;
2539  case EXT_SBR_DATA:
2540  if (!che) {
2541  av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
2542  return res;
2543  } else if (ac->oc[1].m4ac.frame_length_short) {
2544  if (!ac->warned_960_sbr)
2546  "SBR with 960 frame length");
2547  ac->warned_960_sbr = 1;
2548  skip_bits_long(gb, 8 * cnt - 4);
2549  return res;
2550  } else if (!ac->oc[1].m4ac.sbr) {
2551  av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
2552  skip_bits_long(gb, 8 * cnt - 4);
2553  return res;
2554  } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
2555  av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
2556  skip_bits_long(gb, 8 * cnt - 4);
2557  return res;
2558  } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
2559  ac->oc[1].m4ac.sbr = 1;
2560  ac->oc[1].m4ac.ps = 1;
2562  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
2563  ac->oc[1].status, 1);
2564  } else {
2565  ac->oc[1].m4ac.sbr = 1;
2567  }
2568  res = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
2569  break;
2570  case EXT_DYNAMIC_RANGE:
2571  res = decode_dynamic_range(&ac->che_drc, gb);
2572  break;
2573  case EXT_FILL:
2574  decode_fill(ac, gb, 8 * cnt - 4);
2575  break;
2576  case EXT_FILL_DATA:
2577  case EXT_DATA_ELEMENT:
2578  default:
2579  skip_bits_long(gb, 8 * cnt - 4);
2580  break;
2581  };
2582  return res;
2583 }
2584 
2585 /**
2586  * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
2587  *
2588  * @param decode 1 if tool is used normally, 0 if tool is used in LTP.
2589  * @param coef spectral coefficients
2590  */
2591 static void apply_tns(INTFLOAT coef_param[1024], TemporalNoiseShaping *tns,
2592  IndividualChannelStream *ics, int decode)
2593 {
2594  const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
2595  int w, filt, m, i;
2596  int bottom, top, order, start, end, size, inc;
2597  INTFLOAT lpc[TNS_MAX_ORDER];
2599  UINTFLOAT *coef = coef_param;
2600 
2601  if(!mmm)
2602  return;
2603 
2604  for (w = 0; w < ics->num_windows; w++) {
2605  bottom = ics->num_swb;
2606  for (filt = 0; filt < tns->n_filt[w]; filt++) {
2607  top = bottom;
2608  bottom = FFMAX(0, top - tns->length[w][filt]);
2609  order = tns->order[w][filt];
2610  if (order == 0)
2611  continue;
2612 
2613  // tns_decode_coef
2614  AAC_RENAME(compute_lpc_coefs)(tns->coef[w][filt], order, lpc, 0, 0, 0);
2615 
2616  start = ics->swb_offset[FFMIN(bottom, mmm)];
2617  end = ics->swb_offset[FFMIN( top, mmm)];
2618  if ((size = end - start) <= 0)
2619  continue;
2620  if (tns->direction[w][filt]) {
2621  inc = -1;
2622  start = end - 1;
2623  } else {
2624  inc = 1;
2625  }
2626  start += w * 128;
2627 
2628  if (decode) {
2629  // ar filter
2630  for (m = 0; m < size; m++, start += inc)
2631  for (i = 1; i <= FFMIN(m, order); i++)
2632  coef[start] -= AAC_MUL26((INTFLOAT)coef[start - i * inc], lpc[i - 1]);
2633  } else {
2634  // ma filter
2635  for (m = 0; m < size; m++, start += inc) {
2636  tmp[0] = coef[start];
2637  for (i = 1; i <= FFMIN(m, order); i++)
2638  coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
2639  for (i = order; i > 0; i--)
2640  tmp[i] = tmp[i - 1];
2641  }
2642  }
2643  }
2644  }
2645 }
2646 
2647 /**
2648  * Apply windowing and MDCT to obtain the spectral
2649  * coefficient from the predicted sample by LTP.
2650  */
2653 {
2654  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME2(aac_kbd_long_1024) : AAC_RENAME2(sine_1024);
2655  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME2(aac_kbd_short_128) : AAC_RENAME2(sine_128);
2656  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME2(aac_kbd_long_1024) : AAC_RENAME2(sine_1024);
2657  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME2(aac_kbd_short_128) : AAC_RENAME2(sine_128);
2658 
2659  if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
2660  ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
2661  } else {
2662  memset(in, 0, 448 * sizeof(*in));
2663  ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
2664  }
2665  if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
2666  ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
2667  } else {
2668  ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
2669  memset(in + 1024 + 576, 0, 448 * sizeof(*in));
2670  }
2671  ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
2672 }
2673 
2674 /**
2675  * Apply the long term prediction
2676  */
2678 {
2679  const LongTermPrediction *ltp = &sce->ics.ltp;
2680  const uint16_t *offsets = sce->ics.swb_offset;
2681  int i, sfb;
2682 
2683  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
2684  INTFLOAT *predTime = sce->ret;
2685  INTFLOAT *predFreq = ac->buf_mdct;
2686  int16_t num_samples = 2048;
2687 
2688  if (ltp->lag < 1024)
2689  num_samples = ltp->lag + 1024;
2690  for (i = 0; i < num_samples; i++)
2691  predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
2692  memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));
2693 
2694  ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
2695 
2696  if (sce->tns.present)
2697  ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
2698 
2699  for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
2700  if (ltp->used[sfb])
2701  for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
2702  sce->coeffs[i] += (UINTFLOAT)predFreq[i];
2703  }
2704 }
2705 
2706 /**
2707  * Update the LTP buffer for next frame
2708  */
2710 {
2711  IndividualChannelStream *ics = &sce->ics;
2712  INTFLOAT *saved = sce->saved;
2713  INTFLOAT *saved_ltp = sce->coeffs;
2714  const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME2(aac_kbd_long_1024) : AAC_RENAME2(sine_1024);
2715  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME2(aac_kbd_short_128) : AAC_RENAME2(sine_128);
2716  int i;
2717 
2718  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2719  memcpy(saved_ltp, saved, 512 * sizeof(*saved_ltp));
2720  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2721  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2722 
2723  for (i = 0; i < 64; i++)
2724  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2725  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2726  memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(*saved_ltp));
2727  memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2728  ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2729 
2730  for (i = 0; i < 64; i++)
2731  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2732  } else { // LONG_STOP or ONLY_LONG
2733  ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
2734 
2735  for (i = 0; i < 512; i++)
2736  saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
2737  }
2738 
2739  memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
2740  memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
2741  memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
2742 }
2743 
2744 /**
2745  * Conduct IMDCT and windowing.
2746  */
2748 {
2749  IndividualChannelStream *ics = &sce->ics;
2750  INTFLOAT *in = sce->coeffs;
2751  INTFLOAT *out = sce->ret;
2752  INTFLOAT *saved = sce->saved;
2753  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME2(aac_kbd_short_128) : AAC_RENAME2(sine_128);
2754  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME2(aac_kbd_long_1024) : AAC_RENAME2(sine_1024);
2755  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME2(aac_kbd_short_128) : AAC_RENAME2(sine_128);
2756  INTFLOAT *buf = ac->buf_mdct;
2757  INTFLOAT *temp = ac->temp;
2758  int i;
2759 
2760  // imdct
2761  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2762  for (i = 0; i < 1024; i += 128)
2763  ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
2764  } else {
2765  ac->mdct.imdct_half(&ac->mdct, buf, in);
2766 #if USE_FIXED
2767  for (i=0; i<1024; i++)
2768  buf[i] = (buf[i] + 4LL) >> 3;
2769 #endif /* USE_FIXED */
2770  }
2771 
2772  /* window overlapping
2773  * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2774  * and long to short transitions are considered to be short to short
2775  * transitions. This leaves just two cases (long to long and short to short)
2776  * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2777  */
2778  if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2780  ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
2781  } else {
2782  memcpy( out, saved, 448 * sizeof(*out));
2783 
2784  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2785  ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
2786  ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
2787  ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
2788  ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
2789  ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
2790  memcpy( out + 448 + 4*128, temp, 64 * sizeof(*out));
2791  } else {
2792  ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
2793  memcpy( out + 576, buf + 64, 448 * sizeof(*out));
2794  }
2795  }
2796 
2797  // buffer update
2798  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2799  memcpy( saved, temp + 64, 64 * sizeof(*saved));
2800  ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
2801  ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
2802  ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
2803  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2804  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2805  memcpy( saved, buf + 512, 448 * sizeof(*saved));
2806  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2807  } else { // LONG_STOP or ONLY_LONG
2808  memcpy( saved, buf + 512, 512 * sizeof(*saved));
2809  }
2810 }
2811 
2812 /**
2813  * Conduct IMDCT and windowing.
2814  */
2816 {
2817 #if !USE_FIXED
2818  IndividualChannelStream *ics = &sce->ics;
2819  INTFLOAT *in = sce->coeffs;
2820  INTFLOAT *out = sce->ret;
2821  INTFLOAT *saved = sce->saved;
2822  const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(aac_kbd_short_120) : AAC_RENAME(sine_120);
2823  const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(aac_kbd_long_960) : AAC_RENAME(sine_960);
2824  const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(aac_kbd_short_120) : AAC_RENAME(sine_120);
2825  INTFLOAT *buf = ac->buf_mdct;
2826  INTFLOAT *temp = ac->temp;
2827  int i;
2828 
2829  // imdct
2830  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2831  for (i = 0; i < 8; i++)
2832  ac->mdct120->imdct_half(ac->mdct120, buf + i * 120, in + i * 128, 1);
2833  } else {
2834  ac->mdct960->imdct_half(ac->mdct960, buf, in, 1);
2835  }
2836 
2837  /* window overlapping
2838  * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2839  * and long to short transitions are considered to be short to short
2840  * transitions. This leaves just two cases (long to long and short to short)
2841  * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2842  */
2843 
2844  if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2846  ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 480);
2847  } else {
2848  memcpy( out, saved, 420 * sizeof(*out));
2849 
2850  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2851  ac->fdsp->vector_fmul_window(out + 420 + 0*120, saved + 420, buf + 0*120, swindow_prev, 60);
2852  ac->fdsp->vector_fmul_window(out + 420 + 1*120, buf + 0*120 + 60, buf + 1*120, swindow, 60);
2853  ac->fdsp->vector_fmul_window(out + 420 + 2*120, buf + 1*120 + 60, buf + 2*120, swindow, 60);
2854  ac->fdsp->vector_fmul_window(out + 420 + 3*120, buf + 2*120 + 60, buf + 3*120, swindow, 60);
2855  ac->fdsp->vector_fmul_window(temp, buf + 3*120 + 60, buf + 4*120, swindow, 60);
2856  memcpy( out + 420 + 4*120, temp, 60 * sizeof(*out));
2857  } else {
2858  ac->fdsp->vector_fmul_window(out + 420, saved + 420, buf, swindow_prev, 60);
2859  memcpy( out + 540, buf + 60, 420 * sizeof(*out));
2860  }
2861  }
2862 
2863  // buffer update
2864  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2865  memcpy( saved, temp + 60, 60 * sizeof(*saved));
2866  ac->fdsp->vector_fmul_window(saved + 60, buf + 4*120 + 60, buf + 5*120, swindow, 60);
2867  ac->fdsp->vector_fmul_window(saved + 180, buf + 5*120 + 60, buf + 6*120, swindow, 60);
2868  ac->fdsp->vector_fmul_window(saved + 300, buf + 6*120 + 60, buf + 7*120, swindow, 60);
2869  memcpy( saved + 420, buf + 7*120 + 60, 60 * sizeof(*saved));
2870  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2871  memcpy( saved, buf + 480, 420 * sizeof(*saved));
2872  memcpy( saved + 420, buf + 7*120 + 60, 60 * sizeof(*saved));
2873  } else { // LONG_STOP or ONLY_LONG
2874  memcpy( saved, buf + 480, 480 * sizeof(*saved));
2875  }
2876 #endif
2877 }
2879 {
2880  IndividualChannelStream *ics = &sce->ics;
2881  INTFLOAT *in = sce->coeffs;
2882  INTFLOAT *out = sce->ret;
2883  INTFLOAT *saved = sce->saved;
2884  INTFLOAT *buf = ac->buf_mdct;
2885 #if USE_FIXED
2886  int i;
2887 #endif /* USE_FIXED */
2888 
2889  // imdct
2890  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2891 
2892 #if USE_FIXED
2893  for (i = 0; i < 1024; i++)
2894  buf[i] = (buf[i] + 2) >> 2;
2895 #endif /* USE_FIXED */
2896 
2897  // window overlapping
2898  if (ics->use_kb_window[1]) {
2899  // AAC LD uses a low overlap sine window instead of a KBD window
2900  memcpy(out, saved, 192 * sizeof(*out));
2901  ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME2(sine_128), 64);
2902  memcpy( out + 320, buf + 64, 192 * sizeof(*out));
2903  } else {
2904  ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME2(sine_512), 256);
2905  }
2906 
2907  // buffer update
2908  memcpy(saved, buf + 256, 256 * sizeof(*saved));
2909 }
2910 
2912 {
2913  UINTFLOAT *in = sce->coeffs;
2914  INTFLOAT *out = sce->ret;
2915  INTFLOAT *saved = sce->saved;
2916  INTFLOAT *buf = ac->buf_mdct;
2917  int i;
2918  const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
2919  const int n2 = n >> 1;
2920  const int n4 = n >> 2;
2921  const INTFLOAT *const window = n == 480 ? AAC_RENAME(ff_aac_eld_window_480) :
2923 
2924  // Inverse transform, mapped to the conventional IMDCT by
2925  // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
2926  // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
2927  // International Conference on Audio, Language and Image Processing, ICALIP 2008.
2928  // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
2929  for (i = 0; i < n2; i+=2) {
2930  INTFLOAT temp;
2931  temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
2932  temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
2933  }
2934 #if !USE_FIXED
2935  if (n == 480)
2936  ac->mdct480->imdct_half(ac->mdct480, buf, in, 1);
2937  else
2938 #endif
2939  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2940 
2941 #if USE_FIXED
2942  for (i = 0; i < 1024; i++)
2943  buf[i] = (buf[i] + 1) >> 1;
2944 #endif /* USE_FIXED */
2945 
2946  for (i = 0; i < n; i+=2) {
2947  buf[i] = -buf[i];
2948  }
2949  // Like with the regular IMDCT at this point we still have the middle half
2950  // of a transform but with even symmetry on the left and odd symmetry on
2951  // the right
2952 
2953  // window overlapping
2954  // The spec says to use samples [0..511] but the reference decoder uses
2955  // samples [128..639].
2956  for (i = n4; i < n2; i ++) {
2957  out[i - n4] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) +
2958  AAC_MUL31( saved[ i + n2] , window[i + n - n4]) +
2959  AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
2960  AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]);
2961  }
2962  for (i = 0; i < n2; i ++) {
2963  out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) +
2964  AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) +
2965  AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) +
2966  AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
2967  }
2968  for (i = 0; i < n4; i ++) {
2969  out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) +
2970  AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
2971  AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
2972  }
2973 
2974  // buffer update
2975  memmove(saved + n, saved, 2 * n * sizeof(*saved));
2976  memcpy( saved, buf, n * sizeof(*saved));
2977 }
2978 
2979 /**
2980  * channel coupling transformation interface
2981  *
2982  * @param apply_coupling_method pointer to (in)dependent coupling function
2983  */
2985  enum RawDataBlockType type, int elem_id,
2986  enum CouplingPoint coupling_point,
2987  void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
2988 {
2989  int i, c;
2990 
2991  for (i = 0; i < MAX_ELEM_ID; i++) {
2992  ChannelElement *cce = ac->che[TYPE_CCE][i];
2993  int index = 0;
2994 
2995  if (cce && cce->coup.coupling_point == coupling_point) {
2996  ChannelCoupling *coup = &cce->coup;
2997 
2998  for (c = 0; c <= coup->num_coupled; c++) {
2999  if (coup->type[c] == type && coup->id_select[c] == elem_id) {
3000  if (coup->ch_select[c] != 1) {
3001  apply_coupling_method(ac, &cc->ch[0], cce, index);
3002  if (coup->ch_select[c] != 0)
3003  index++;
3004  }
3005  if (coup->ch_select[c] != 2)
3006  apply_coupling_method(ac, &cc->ch[1], cce, index++);
3007  } else
3008  index += 1 + (coup->ch_select[c] == 3);
3009  }
3010  }
3011  }
3012 }
3013 
3014 /**
3015  * Convert spectral data to samples, applying all supported tools as appropriate.
3016  */
3017 static void spectral_to_sample(AACContext *ac, int samples)
3018 {
3019  int i, type;
3021  switch (ac->oc[1].m4ac.object_type) {
3022  case AOT_ER_AAC_LD:
3024  break;
3025  case AOT_ER_AAC_ELD:
3027  break;
3028  default:
3029  if (ac->oc[1].m4ac.frame_length_short)
3031  else
3033  }
3034  for (type = 3; type >= 0; type--) {
3035  for (i = 0; i < MAX_ELEM_ID; i++) {
3036  ChannelElement *che = ac->che[type][i];
3037  if (che && che->present) {
3038  if (type <= TYPE_CPE)
3040  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
3041  if (che->ch[0].ics.predictor_present) {
3042  if (che->ch[0].ics.ltp.present)
3043  ac->apply_ltp(ac, &che->ch[0]);
3044  if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
3045  ac->apply_ltp(ac, &che->ch[1]);
3046  }
3047  }
3048  if (che->ch[0].tns.present)
3049  ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
3050  if (che->ch[1].tns.present)
3051  ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
3052  if (type <= TYPE_CPE)
3054  if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
3055  imdct_and_window(ac, &che->ch[0]);
3056  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
3057  ac->update_ltp(ac, &che->ch[0]);
3058  if (type == TYPE_CPE) {
3059  imdct_and_window(ac, &che->ch[1]);
3060  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
3061  ac->update_ltp(ac, &che->ch[1]);
3062  }
3063  if (ac->oc[1].m4ac.sbr > 0) {
3064  AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
3065  }
3066  }
3067  if (type <= TYPE_CCE)
3069 
3070 #if USE_FIXED
3071  {
3072  int j;
3073  /* preparation for resampler */
3074  for(j = 0; j<samples; j++){
3075  che->ch[0].ret[j] = (int32_t)av_clip64((int64_t)che->ch[0].ret[j]*128, INT32_MIN, INT32_MAX-0x8000)+0x8000;
3076  if(type == TYPE_CPE)
3077  che->ch[1].ret[j] = (int32_t)av_clip64((int64_t)che->ch[1].ret[j]*128, INT32_MIN, INT32_MAX-0x8000)+0x8000;
3078  }
3079  }
3080 #endif /* USE_FIXED */
3081  che->present = 0;
3082  } else if (che) {
3083  av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
3084  }
3085  }
3086  }
3087 }
3088 
3090 {
3091  int size;
3092  AACADTSHeaderInfo hdr_info;
3093  uint8_t layout_map[MAX_ELEM_ID*4][3];
3094  int layout_map_tags, ret;
3095 
3096  size = ff_adts_header_parse(gb, &hdr_info);
3097  if (size > 0) {
3098  if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
3099  // This is 2 for "VLB " audio in NSV files.
3100  // See samples/nsv/vlb_audio.
3102  "More than one AAC RDB per ADTS frame");
3103  ac->warned_num_aac_frames = 1;
3104  }
3106  if (hdr_info.chan_config) {
3107  ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
3108  if ((ret = set_default_channel_config(ac, ac->avctx,
3109  layout_map,
3110  &layout_map_tags,
3111  hdr_info.chan_config)) < 0)
3112  return ret;
3113  if ((ret = output_configure(ac, layout_map, layout_map_tags,
3114  FFMAX(ac->oc[1].status,
3115  OC_TRIAL_FRAME), 0)) < 0)
3116  return ret;
3117  } else {
3118  ac->oc[1].m4ac.chan_config = 0;
3119  /**
3120  * dual mono frames in Japanese DTV can have chan_config 0
3121  * WITHOUT specifying PCE.
3122  * thus, set dual mono as default.
3123  */
3124  if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
3125  layout_map_tags = 2;
3126  layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
3127  layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
3128  layout_map[0][1] = 0;
3129  layout_map[1][1] = 1;
3130  if (output_configure(ac, layout_map, layout_map_tags,
3131  OC_TRIAL_FRAME, 0))
3132  return -7;
3133  }
3134  }
3135  ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
3136  ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
3137  ac->oc[1].m4ac.object_type = hdr_info.object_type;
3138  ac->oc[1].m4ac.frame_length_short = 0;
3139  if (ac->oc[0].status != OC_LOCKED ||
3140  ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
3141  ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
3142  ac->oc[1].m4ac.sbr = -1;
3143  ac->oc[1].m4ac.ps = -1;
3144  }
3145  if (!hdr_info.crc_absent)
3146  skip_bits(gb, 16);
3147  }
3148  return size;
3149 }
3150 
3151 static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
3152  int *got_frame_ptr, GetBitContext *gb)
3153 {
3154  AACContext *ac = avctx->priv_data;
3155  const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
3156  ChannelElement *che;
3157  int err, i;
3158  int samples = m4ac->frame_length_short ? 960 : 1024;
3159  int chan_config = m4ac->chan_config;
3160  int aot = m4ac->object_type;
3161 
3162  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
3163  samples >>= 1;
3164 
3165  ac->frame = data;
3166 
3167  if ((err = frame_configure_elements(avctx)) < 0)
3168  return err;
3169 
3170  // The FF_PROFILE_AAC_* defines are all object_type - 1
3171  // This may lead to an undefined profile being signaled
3172  ac->avctx->profile = aot - 1;
3173 
3174  ac->tags_mapped = 0;
3175 
3176  if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
3177  avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
3178  chan_config);
3179  return AVERROR_INVALIDDATA;
3180  }
3181  for (i = 0; i < tags_per_config[chan_config]; i++) {
3182  const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
3183  const int elem_id = aac_channel_layout_map[chan_config-1][i][1];
3184  if (!(che=get_che(ac, elem_type, elem_id))) {
3185  av_log(ac->avctx, AV_LOG_ERROR,
3186  "channel element %d.%d is not allocated\n",
3187  elem_type, elem_id);
3188  return AVERROR_INVALIDDATA;
3189  }
3190  che->present = 1;
3191  if (aot != AOT_ER_AAC_ELD)
3192  skip_bits(gb, 4);
3193  switch (elem_type) {
3194  case TYPE_SCE:
3195  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3196  break;
3197  case TYPE_CPE:
3198  err = decode_cpe(ac, gb, che);
3199  break;
3200  case TYPE_LFE:
3201  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3202  break;
3203  }
3204  if (err < 0)
3205  return err;
3206  }
3207 
3208  spectral_to_sample(ac, samples);
3209 
3210  if (!ac->frame->data[0] && samples) {
3211  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
3212  return AVERROR_INVALIDDATA;
3213  }
3214 
3215  ac->frame->nb_samples = samples;
3216  ac->frame->sample_rate = avctx->sample_rate;
3217  *got_frame_ptr = 1;
3218 
3219  skip_bits_long(gb, get_bits_left(gb));
3220  return 0;
3221 }
3222 
3223 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
3224  int *got_frame_ptr, GetBitContext *gb,
3225  const AVPacket *avpkt)
3226 {
3227  AACContext *ac = avctx->priv_data;
3228  ChannelElement *che = NULL, *che_prev = NULL;
3229  enum RawDataBlockType elem_type, che_prev_type = TYPE_END;
3230  int err, elem_id;
3231  int samples = 0, multiplier, audio_found = 0, pce_found = 0;
3232  int is_dmono, sce_count = 0;
3233  int payload_alignment;
3234  uint8_t che_presence[4][MAX_ELEM_ID] = {{0}};
3235 
3236  ac->frame = data;
3237 
3238  if (show_bits(gb, 12) == 0xfff) {
3239  if ((err = parse_adts_frame_header(ac, gb)) < 0) {
3240  av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
3241  goto fail;
3242  }
3243  if (ac->oc[1].m4ac.sampling_index > 12) {
3244  av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
3245  err = AVERROR_INVALIDDATA;
3246  goto fail;
3247  }
3248  }
3249 
3250  if ((err = frame_configure_elements(avctx)) < 0)
3251  goto fail;
3252 
3253  // The FF_PROFILE_AAC_* defines are all object_type - 1
3254  // This may lead to an undefined profile being signaled
3255  ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
3256 
3257  payload_alignment = get_bits_count(gb);
3258  ac->tags_mapped = 0;
3259  // parse
3260  while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
3261  elem_id = get_bits(gb, 4);
3262 
3263  if (avctx->debug & FF_DEBUG_STARTCODE)
3264  av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
3265 
3266  if (!avctx->channels && elem_type != TYPE_PCE) {
3267  err = AVERROR_INVALIDDATA;
3268  goto fail;
3269  }
3270 
3271  if (elem_type < TYPE_DSE) {
3272  if (che_presence[elem_type][elem_id]) {
3273  int error = che_presence[elem_type][elem_id] > 1;
3274  av_log(ac->avctx, error ? AV_LOG_ERROR : AV_LOG_DEBUG, "channel element %d.%d duplicate\n",
3275  elem_type, elem_id);
3276  if (error) {
3277  err = AVERROR_INVALIDDATA;
3278  goto fail;
3279  }
3280  }
3281  che_presence[elem_type][elem_id]++;
3282 
3283  if (!(che=get_che(ac, elem_type, elem_id))) {
3284  av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
3285  elem_type, elem_id);
3286  err = AVERROR_INVALIDDATA;
3287  goto fail;
3288  }
3289  samples = ac->oc[1].m4ac.frame_length_short ? 960 : 1024;
3290  che->present = 1;
3291  }
3292 
3293  switch (elem_type) {
3294 
3295  case TYPE_SCE:
3296  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3297  audio_found = 1;
3298  sce_count++;
3299  break;
3300 
3301  case TYPE_CPE:
3302  err = decode_cpe(ac, gb, che);
3303  audio_found = 1;
3304  break;
3305 
3306  case TYPE_CCE:
3307  err = decode_cce(ac, gb, che);
3308  break;
3309 
3310  case TYPE_LFE:
3311  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3312  audio_found = 1;
3313  break;
3314 
3315  case TYPE_DSE:
3316  err = skip_data_stream_element(ac, gb);
3317  break;
3318 
3319  case TYPE_PCE: {
3320  uint8_t layout_map[MAX_ELEM_ID*4][3] = {{0}};
3321  int tags;
3322 
3323  int pushed = push_output_configuration(ac);
3324  if (pce_found && !pushed) {
3325  err = AVERROR_INVALIDDATA;
3326  goto fail;
3327  }
3328 
3329  tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb,
3330  payload_alignment);
3331  if (tags < 0) {
3332  err = tags;
3333  break;
3334  }
3335  if (pce_found) {
3336  av_log(avctx, AV_LOG_ERROR,
3337  "Not evaluating a further program_config_element as this construct is dubious at best.\n");
3339  } else {
3340  err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
3341  if (!err)
3342  ac->oc[1].m4ac.chan_config = 0;
3343  pce_found = 1;
3344  }
3345  break;
3346  }
3347 
3348  case TYPE_FIL:
3349  if (elem_id == 15)
3350  elem_id += get_bits(gb, 8) - 1;
3351  if (get_bits_left(gb) < 8 * elem_id) {
3352  av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
3353  err = AVERROR_INVALIDDATA;
3354  goto fail;
3355  }
3356  err = 0;
3357  while (elem_id > 0) {
3358  int ret = decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type);
3359  if (ret < 0) {
3360  err = ret;
3361  break;
3362  }
3363  elem_id -= ret;
3364  }
3365  break;
3366 
3367  default:
3368  err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
3369  break;
3370  }
3371 
3372  if (elem_type < TYPE_DSE) {
3373  che_prev = che;
3374  che_prev_type = elem_type;
3375  }
3376 
3377  if (err)
3378  goto fail;
3379 
3380  if (get_bits_left(gb) < 3) {
3381  av_log(avctx, AV_LOG_ERROR, overread_err);
3382  err = AVERROR_INVALIDDATA;
3383  goto fail;
3384  }
3385  }
3386 
3387  if (!avctx->channels) {
3388  *got_frame_ptr = 0;
3389  return 0;
3390  }
3391 
3392  multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
3393  samples <<= multiplier;
3394 
3395  spectral_to_sample(ac, samples);
3396 
3397  if (ac->oc[1].status && audio_found) {
3398  avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
3399  avctx->frame_size = samples;
3400  ac->oc[1].status = OC_LOCKED;
3401  }
3402 
3403  if (multiplier)
3404  avctx->internal->skip_samples_multiplier = 2;
3405 
3406  if (!ac->frame->data[0] && samples) {
3407  av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
3408  err = AVERROR_INVALIDDATA;
3409  goto fail;
3410  }
3411 
3412  if (samples) {
3413  ac->frame->nb_samples = samples;
3414  ac->frame->sample_rate = avctx->sample_rate;
3415  } else
3416  av_frame_unref(ac->frame);
3417  *got_frame_ptr = !!samples;
3418 
3419  /* for dual-mono audio (SCE + SCE) */
3420  is_dmono = ac->dmono_mode && sce_count == 2 &&
3422  if (is_dmono) {
3423  if (ac->dmono_mode == 1)
3424  ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
3425  else if (ac->dmono_mode == 2)
3426  ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
3427  }
3428 
3429  return 0;
3430 fail:
3432  return err;
3433 }
3434 
3435 static int aac_decode_frame(AVCodecContext *avctx, void *data,
3436  int *got_frame_ptr, AVPacket *avpkt)
3437 {
3438  AACContext *ac = avctx->priv_data;
3439  const uint8_t *buf = avpkt->data;
3440  int buf_size = avpkt->size;
3441  GetBitContext gb;
3442  int buf_consumed;
3443  int buf_offset;
3444  int err;
3445  buffer_size_t new_extradata_size;
3446  const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
3448  &new_extradata_size);
3449  buffer_size_t jp_dualmono_size;
3450  const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
3452  &jp_dualmono_size);
3453 
3454  if (new_extradata) {
3455  /* discard previous configuration */
3456  ac->oc[1].status = OC_NONE;
3457  err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
3458  new_extradata,
3459  new_extradata_size * 8LL, 1);
3460  if (err < 0) {
3461  return err;
3462  }
3463  }
3464 
3465  ac->dmono_mode = 0;
3466  if (jp_dualmono && jp_dualmono_size > 0)
3467  ac->dmono_mode = 1 + *jp_dualmono;
3468  if (ac->force_dmono_mode >= 0)
3469  ac->dmono_mode = ac->force_dmono_mode;
3470 
3471  if (INT_MAX / 8 <= buf_size)
3472  return AVERROR_INVALIDDATA;
3473 
3474  if ((err = init_get_bits8(&gb, buf, buf_size)) < 0)
3475  return err;
3476 
3477  switch (ac->oc[1].m4ac.object_type) {
3478  case AOT_ER_AAC_LC:
3479  case AOT_ER_AAC_LTP:
3480  case AOT_ER_AAC_LD:
3481  case AOT_ER_AAC_ELD:
3482  err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
3483  break;
3484  default:
3485  err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
3486  }
3487  if (err < 0)
3488  return err;
3489 
3490  buf_consumed = (get_bits_count(&gb) + 7) >> 3;
3491  for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
3492  if (buf[buf_offset])
3493  break;
3494 
3495  return buf_size > buf_offset ? buf_consumed : buf_size;
3496 }
3497 
3499 {
3500  AACContext *ac = avctx->priv_data;
3501  int i, type;
3502 
3503  for (i = 0; i < MAX_ELEM_ID; i++) {
3504  for (type = 0; type < 4; type++) {
3505  if (ac->che[type][i])
3507  av_freep(&ac->che[type][i]);
3508  }
3509  }
3510 
3511  ff_mdct_end(&ac->mdct);
3512  ff_mdct_end(&ac->mdct_small);
3513  ff_mdct_end(&ac->mdct_ld);
3514  ff_mdct_end(&ac->mdct_ltp);
3515 #if !USE_FIXED
3516  ff_mdct15_uninit(&ac->mdct120);
3517  ff_mdct15_uninit(&ac->mdct480);
3518  ff_mdct15_uninit(&ac->mdct960);
3519 #endif
3520  av_freep(&ac->fdsp);
3521  return 0;
3522 }
3523 
3524 static void aacdec_init(AACContext *c)
3525 {
3526  c->imdct_and_windowing = imdct_and_windowing;
3527  c->apply_ltp = apply_ltp;
3528  c->apply_tns = apply_tns;
3529  c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
3530  c->update_ltp = update_ltp;
3531 #if USE_FIXED
3532  c->vector_pow43 = vector_pow43;
3533  c->subband_scale = subband_scale;
3534 #endif
3535 
3536 #if !USE_FIXED
3537  if(ARCH_MIPS)
3539 #endif /* !USE_FIXED */
3540 }
3541 /**
3542  * AVOptions for Japanese DTV specific extensions (ADTS only)
3543  */
3544 #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
3545 static const AVOption options[] = {
3546  {"dual_mono_mode", "Select the channel to decode for dual mono",
3547  offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
3548  AACDEC_FLAGS, "dual_mono_mode"},
3549 
3550  {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3551  {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3552  {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3553  {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3554 
3555  {NULL},
3556 };
3557 
3558 static const AVClass aac_decoder_class = {
3559  .class_name = "AAC decoder",
3560  .item_name = av_default_item_name,
3561  .option = options,
3562  .version = LIBAVUTIL_VERSION_INT,
3563 };
@ EIGHT_SHORT_SEQUENCE
Definition: aac.h:79
@ LONG_STOP_SEQUENCE
Definition: aac.h:80
@ ONLY_LONG_SEQUENCE
Definition: aac.h:77
@ LONG_START_SEQUENCE
Definition: aac.h:78
@ EXT_DATA_ELEMENT
Definition: aac.h:70
@ EXT_SBR_DATA_CRC
Definition: aac.h:73
@ EXT_DYNAMIC_RANGE
Definition: aac.h:71
@ EXT_FILL_DATA
Definition: aac.h:69
@ EXT_FILL
Definition: aac.h:68
@ EXT_SBR_DATA
Definition: aac.h:72
#define NOISE_PRE
preamble for NOISE_BT, put in bitstream with the first noise band
Definition: aac.h:157
BandType
Definition: aac.h:83
@ INTENSITY_BT
Scalefactor data are intensity stereo positions (in phase).
Definition: aac.h:90
@ ZERO_BT
Scalefactors and spectral data are all zero.
Definition: aac.h:84
@ INTENSITY_BT2
Scalefactor data are intensity stereo positions (out of phase).
Definition: aac.h:89
@ NOISE_BT
Spectral data are scaled white noise not coded in the bitstream.
Definition: aac.h:88
CouplingPoint
The point during decoding at which channel coupling is applied.
Definition: aac.h:107
@ BETWEEN_TNS_AND_IMDCT
Definition: aac.h:109
@ AFTER_IMDCT
Definition: aac.h:110
@ BEFORE_TNS
Definition: aac.h:108
ChannelPosition
Definition: aac.h:95
@ AAC_CHANNEL_LFE
Definition: aac.h:100
@ AAC_CHANNEL_SIDE
Definition: aac.h:98
@ AAC_CHANNEL_OFF
Definition: aac.h:96
@ AAC_CHANNEL_CC
Definition: aac.h:101
@ AAC_CHANNEL_BACK
Definition: aac.h:99
@ AAC_CHANNEL_FRONT
Definition: aac.h:97
RawDataBlockType
Definition: aac.h:56
@ TYPE_CCE
Definition: aac.h:59
@ TYPE_PCE
Definition: aac.h:62
@ TYPE_CPE
Definition: aac.h:58
@ TYPE_SCE
Definition: aac.h:57
@ TYPE_FIL
Definition: aac.h:63
@ TYPE_LFE
Definition: aac.h:60
@ TYPE_END
Definition: aac.h:64
@ TYPE_DSE
Definition: aac.h:61
#define MAX_LTP_LONG_SFB
Definition: aac.h:52
#define TNS_MAX_ORDER
Definition: aac.h:51
#define NOISE_PRE_BITS
length of preamble
Definition: aac.h:158
#define SCALE_DIFF_ZERO
codebook index corresponding to zero scalefactor indices difference
Definition: aac.h:153
#define MAX_CHANNELS
Definition: aac.h:48
OCStatus
Output configuration status.
Definition: aac.h:116
@ OC_TRIAL_FRAME
Output configuration under trial specified by a frame header.
Definition: aac.h:119
@ OC_TRIAL_PCE
Output configuration under trial specified by an inband PCE.
Definition: aac.h:118
@ OC_LOCKED
Output configuration locked in place.
Definition: aac.h:121
@ OC_GLOBAL_HDR
Output configuration set in a global header but not yet locked.
Definition: aac.h:120
@ OC_NONE
Output unconfigured.
Definition: aac.h:117
void ff_aacdec_init_mips(AACContext *c)
Definition: aacdec_mips.c:433
#define MAX_PREDICTORS
Definition: aac.h:147
#define NOISE_OFFSET
subtracted from global gain, used as offset for the preamble
Definition: aac.h:159
#define MAX_ELEM_ID
Definition: aac.h:49
#define POW_SF2_ZERO
ff_aac_pow2sf_tab index corresponding to pow(2, 0);
Definition: aac.h:155
#define AAC_RENAME(x)
Definition: aac_defines.h:85
#define RANGE15(x)
Definition: aac_defines.h:99
#define FIXR(x)
Definition: aac_defines.h:94
#define AAC_MUL31(x, y)
Definition: aac_defines.h:104
#define AAC_MUL30(x, y)
Definition: aac_defines.h:103
float INTFLOAT
Definition: aac_defines.h:88
#define USE_FIXED
Definition: aac_defines.h:25
#define AAC_MUL26(x, y)
Definition: aac_defines.h:102
#define AAC_RENAME2(x)
Definition: aac_defines.h:87
unsigned AAC_SIGNE
Definition: aac_defines.h:93
#define AAC_RENAME_32(x)
Definition: aac_defines.h:86
#define FIXR10(x)
Definition: aac_defines.h:95
#define GET_GAIN(x, y)
Definition: aac_defines.h:100
float UINTFLOAT
Definition: aac_defines.h:89
static void subband_scale(int *dst, int *src, int scale, int offset, int len, void *log_context)
Definition: aacdec_fixed.c:168
static int * DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
Definition: aacdec_fixed.c:136
static int * DEC_SQUAD(int *dst, unsigned idx)
Definition: aacdec_fixed.c:118
static int * DEC_SPAIR(int *dst, unsigned idx)
Definition: aacdec_fixed.c:110
static void vector_pow43(int *coefs, int len)
Definition: aacdec_fixed.c:154
static int * DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
Definition: aacdec_fixed.c:128
static void noise_scale(int *coefs, int scale, int band_energy, int len)
Definition: aacdec_fixed.c:199
static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx, GetBitContext *gb, MPEG4AudioConfig *m4ac, int channel_config)
static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
static void relative_align_get_bits(GetBitContext *gb, int reference_position)
#define AACDEC_FLAGS
AVOptions for Japanese DTV specific extensions (ADTS only)
static uint64_t sniff_channel_order(uint8_t(*layout_map)[3], int tags)
static void decode_channel_map(uint8_t layout_map[][3], enum ChannelPosition type, GetBitContext *gb, int n)
Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
static int aac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static int aac_decode_er_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb)
static int sample_rate_idx(int rate)
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
Apply AAC-Main style frequency domain prediction.
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.
static void spectral_to_sample(AACContext *ac, int samples)
Convert spectral data to samples, applying all supported tools as appropriate.
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, GetBitContext *gb, const IndividualChannelStream *ics)
Decode Temporal Noise Shaping data; reference: table 4.48.
static const AVOption options[]
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
Mid/Side stereo decoding; reference: 4.6.8.1.3.
static int decode_audio_specific_config_gb(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, GetBitContext *gb, int get_bit_alignment, int sync_extension)
Decode audio specific configuration; reference: table 1.13.
static void imdct_and_windowing_960(AACContext *ac, SingleChannelElement *sce)
Conduct IMDCT and windowing.
static VLC vlc_scalefactors
#define overread_err
static av_cold void aac_static_table_init(void)
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, int ms_present)
Decode Mid/Side data; reference: table 4.54.
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
Decode a channel_pair_element; reference: table 4.4.
static int aac_decode_frame_int(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb, const AVPacket *avpkt)
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.
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
Apply the long term prediction.
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
Skip data_stream_element; reference: table 4.10.
static VLC vlc_spectral[11]
static int frame_configure_elements(AVCodecContext *avctx)
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.
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....
static int count_paired_channels(uint8_t(*layout_map)[3], int tags, int pos, int *current)
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.
static int decode_pulses(Pulse *pulse, GetBitContext *gb, const uint16_t *swb_offset, int num_swb)
Decode pulse data; reference: table 4.7.
static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext *gb)
Decode dynamic range information; reference: table 4.52.
static void decode_gain_control(SingleChannelElement *sce, GetBitContext *gb)
static av_cold int che_configure(AACContext *ac, enum ChannelPosition che_pos, int type, int id, int *channels)
Check for the channel element in the current channel position configuration.
static int decode_fill(AACContext *ac, GetBitContext *gb, int len)
static av_cold int aac_decode_close(AVCodecContext *avctx)
static int count_channels(uint8_t(*layout)[3], int tags)
static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac, uint8_t(*layout_map)[3], GetBitContext *gb, int byte_align_ref)
Decode program configuration element; reference: table 4.2.
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
Conduct IMDCT and windowing.
static av_cold int aac_decode_init(AVCodecContext *avctx)
static ChannelElement * get_che(AACContext *ac, int type, int elem_id)
static int decode_audio_specific_config(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, const uint8_t *data, int64_t bit_size, int sync_extension)
static void aacdec_init(AACContext *ac)
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.
static int set_default_channel_config(AACContext *ac, 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....
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
Update the LTP buffer for next frame.
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
Decode coupling_channel_element; reference: table 4.8.
static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
intensity stereo decoding; reference: 4.6.8.2.3
static void apply_tns(INTFLOAT coef_param[1024], TemporalNoiseShaping *tns, IndividualChannelStream *ics, int decode)
Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4....
static AVOnce aac_table_init
static void decode_ltp(LongTermPrediction *ltp, GetBitContext *gb, uint8_t max_sfb)
Decode Long Term Prediction data; reference: table 4.xx.
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, uint64_t *layout)
static void reset_all_predictors(PredictorState *ps)
static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx, GetBitContext *gb, int get_bit_alignment, MPEG4AudioConfig *m4ac, int channel_config)
Decode GA "General Audio" specific configuration; reference: table 4.1.
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
static void reset_predictor_group(PredictorState *ps, int group_num)
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.
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb)
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, GetBitContext *gb)
Decode Individual Channel Stream info; reference: table 4.6.
static av_always_inline int lcg_random(unsigned previous_val)
linear congruential pseudorandom number generator
static int push_output_configuration(AACContext *ac)
Save current output configuration if and only if it has been locked.
static void pop_output_configuration(AACContext *ac)
Restore the previous output configuration if and only if the current configuration is unlocked.
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
#define PREFIX_FOR_22POINT2
static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
static const AVClass aac_decoder_class
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.
static const float cce_scale[]
static void flush(AVCodecContext *avctx)
static const uint8_t aac_channel_layout_map[16][16][3]
Definition: aacdectab.h:40
static const int8_t tags_per_config[16]
Definition: aacdectab.h:38
void AAC_RENAME() ff_aac_sbr_ctx_close(SpectralBandReplication *sbr)
Close one SBR context.
void AAC_RENAME() ff_aac_sbr_ctx_init(AACContext *ac, SpectralBandReplication *sbr, int id_aac)
Initialize one SBR context.
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.
void AAC_RENAME() ff_aac_sbr_init(void)
Initialize SBR.
int AAC_RENAME() ff_decode_sbr_extension(AACContext *ac, SpectralBandReplication *sbr, GetBitContext *gb, int crc, int cnt, int id_aac)
Decode one SBR element.
const uint32_t ff_aac_scalefactor_code[121]
Definition: aactab.c:92
const uint8_t ff_aac_num_swb_120[]
Definition: aactab.c:84
const uint8_t ff_aac_num_swb_960[]
Definition: aactab.c:68
const float ff_aac_eld_window_480[1800]
Definition: aactab.c:2397
const uint8_t ff_tns_max_bands_512[]
Definition: aactab.c:1417
const uint8_t ff_tns_max_bands_1024[]
Definition: aactab.c:1413
const uint16_t *const ff_swb_offset_128[]
Definition: aactab.c:1387
const uint16_t *const ff_swb_offset_120[]
Definition: aactab.c:1397
const uint8_t ff_aac_num_swb_480[]
Definition: aactab.c:76
const uint8_t ff_aac_pred_sfb_max[]
Definition: aactab.c:88
const uint16_t *const ff_swb_offset_1024[]
Definition: aactab.c:1355
const uint8_t ff_aac_scalefactor_bits[121]
Definition: aactab.c:111
const uint16_t *const ff_aac_codebook_vector_idx[]
Definition: aactab.c:1102
void ff_aac_tableinit(void)
Definition: aactab.c:3347
float ff_aac_pow2sf_tab[428]
Definition: aactab.c:39
const uint16_t *const ff_swb_offset_960[]
Definition: aactab.c:1363
const uint16_t *const ff_swb_offset_480[]
Definition: aactab.c:1379
const uint8_t ff_aac_num_swb_1024[]
Definition: aactab.c:64
const uint8_t ff_aac_num_swb_128[]
Definition: aactab.c:80
const float ff_aac_eld_window_512[1920]
Definition: aactab.c:1430
const uint8_t ff_tns_max_bands_480[]
Definition: aactab.c:1421
const uint8_t *const ff_aac_spectral_bits[11]
Definition: aactab.c:441
const uint8_t ff_aac_num_swb_512[]
Definition: aactab.c:72
const uint16_t ff_aac_spectral_sizes[11]
Definition: aactab.c:446
const uint16_t *const ff_aac_spectral_codes[11]
Definition: aactab.c:436
const uint8_t ff_tns_max_bands_128[]
Definition: aactab.c:1425
const float *const ff_aac_codebook_vector_vals[]
Definition: aactab.c:1093
const uint16_t *const ff_swb_offset_512[]
Definition: aactab.c:1371
static const INTFLOAT *const tns_tmp2_map[4]
Definition: aactab.h:82
void ff_aac_float_common_init(void)
static const INTFLOAT ltp_coef[8]
Definition: aactab.h:50
int ff_adts_header_parse(GetBitContext *gbc, AACADTSHeaderInfo *hdr)
Parse the ADTS frame header to the end of the variable header, which is the first 54 bits.
Definition: adts_header.c:30
static const int8_t filt[NUMTAPS *2]
Definition: af_earwax.c:39
channels
Definition: aptx.h:33
#define av_always_inline
Definition: attributes.h:45
#define av_cold
Definition: attributes.h:88
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
uint8_t
int32_t
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
#define FF_DEBUG_STARTCODE
Definition: avcodec.h:1631
#define AV_EF_BITSTREAM
detect bitstream specification deviations
Definition: avcodec.h:1654
#define FF_PROFILE_AAC_HE
Definition: avcodec.h:1866
#define FF_PROFILE_AAC_HE_V2
Definition: avcodec.h:1867
#define FF_COMPLIANCE_STRICT
Strictly conform to all the things in the spec no matter what consequences.
Definition: avcodec.h:1603
#define FF_DEBUG_PICT_INFO
Definition: avcodec.h:1624
#define AV_EF_EXPLODE
abort decoding on minor error detection
Definition: avcodec.h:1656
uint8_t * av_packet_get_side_data(const AVPacket *pkt, enum AVPacketSideDataType type, buffer_size_t *size)
Definition: avpacket.c:368
int ff_init_vlc_sparse(VLC *vlc_arg, int nb_bits, int nb_codes, const void *bits, int bits_wrap, int bits_size, const void *codes, int codes_wrap, int codes_size, const void *symbols, int symbols_wrap, int symbols_size, int flags)
Definition: bitstream.c:323
void ff_cbrt_tableinit(void)
Definition: cbrt_tablegen.h:40
uint32_t ff_cbrt_tab[1<< 13]
#define u(width, name, range_min, range_max)
Definition: cbs_h2645.c:264
#define s(width, name)
Definition: cbs_vp9.c:257
uint64_t layout
#define fail()
Definition: checkasm.h:133
static VLC_TYPE vlc_buf[16716][2]
Definition: clearvideo.c:86
#define FFSWAP(type, a, b)
Definition: common.h:108
#define FFMIN(a, b)
Definition: common.h:105
#define av_clip
Definition: common.h:122
#define FFMAX(a, b)
Definition: common.h:103
#define av_clip64
Definition: common.h:125
#define ARCH_MIPS
Definition: config.h:27
#define NULL
Definition: coverity.c:32
long long int64_t
Definition: coverity.c:34
#define abs(x)
Definition: cuda_runtime.h:35
static __device__ float fabsf(float a)
Definition: cuda_runtime.h:181
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1900
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
mode
Use these values in ebur128_init (or'ed).
Definition: ebur128.h:83
enum AVCodecID id
int
static SDL_Window * window
Definition: ffplay.c:366
float re
Definition: fft.c:82
#define ff_mdct_init
Definition: fft.h:161
#define ff_mdct_end
Definition: fft.h:162
static av_always_inline int fixed_sqrt(int x, int bits)
Calculate the square root.
Definition: fixed_dsp.h:176
#define GET_VLC(code, name, gb, table, bits, max_depth)
If the vlc code is invalid and max_depth=1, then no bits will be removed.
Definition: get_bits.h:706
#define GET_CACHE(name, gb)
Definition: get_bits.h:215
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
Definition: get_bits.h:797
#define CLOSE_READER(name, gb)
Definition: get_bits.h:149
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:291
#define SHOW_UBITS(name, gb, num)
Definition: get_bits.h:211
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
#define OPEN_READER(name, gb)
Definition: get_bits.h:138
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
#define SKIP_BITS(name, gb, num)
Definition: get_bits.h:193
#define UPDATE_CACHE(name, gb)
Definition: get_bits.h:178
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
#define LAST_SKIP_BITS(name, gb, num)
Definition: get_bits.h:199
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
static void skip_bits1(GetBitContext *s)
Definition: get_bits.h:538
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
static const uint8_t * align_get_bits(GetBitContext *s)
Definition: get_bits.h:693
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
Definition: get_bits.h:446
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:659
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
@ AV_OPT_TYPE_INT
Definition: opt.h:225
#define AV_CH_LAYOUT_22POINT2
#define AV_CH_SIDE_LEFT
#define AV_CH_TOP_FRONT_LEFT
#define AV_CH_FRONT_RIGHT
#define AV_CH_TOP_BACK_CENTER
#define AV_CH_BOTTOM_FRONT_CENTER
#define AV_CH_FRONT_RIGHT_OF_CENTER
#define AV_CH_BACK_CENTER
#define AV_CH_TOP_FRONT_CENTER
#define AV_CH_FRONT_LEFT_OF_CENTER
#define AV_CH_LOW_FREQUENCY_2
#define AV_CH_BACK_RIGHT
#define AV_CH_LAYOUT_NATIVE
Channel mask value used for AVCodecContext.request_channel_layout to indicate that the user requests ...
#define AV_CH_TOP_SIDE_RIGHT
#define AV_CH_FRONT_CENTER
#define AV_CH_TOP_BACK_RIGHT
#define AV_CH_TOP_CENTER
#define AV_CH_SIDE_RIGHT
#define AV_CH_BACK_LEFT
#define AV_CH_TOP_SIDE_LEFT
#define AV_CH_TOP_BACK_LEFT
#define AV_CH_LOW_FREQUENCY
#define AV_CH_BOTTOM_FRONT_RIGHT
#define AV_CH_TOP_FRONT_RIGHT
#define AV_CH_BOTTOM_FRONT_LEFT
#define AV_CH_FRONT_LEFT
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:333
@ 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: packet.h:166
@ 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: packet.h:55
#define AVERROR_UNKNOWN
Unknown error, typically from an external library.
Definition: error.h:71
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
#define AVERROR_BUG
Internal bug, also see AVERROR_BUG2.
Definition: error.h:50
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define AVERROR(e)
Definition: error.h:43
void av_frame_unref(AVFrame *frame)
Unreference all the buffers referenced by frame and reset the frame fields.
Definition: frame.c:553
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:215
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:200
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:210
#define AV_LOG_INFO
Standard information.
Definition: log.h:205
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:237
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:68
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
int index
Definition: gxfenc.c:89
static const int offsets[]
Definition: hevc_pel.c:34
cl_device_type type
int i
Definition: input.c:407
#define av_log2
Definition: intmath.h:83
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
Generate a Kaiser-Bessel Derived Window.
Definition: kbdwin.c:26
static av_always_inline void reset_predict_state(PredictorState *ps)
Definition: aacdec.c:77
static float * VMUL4S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
Definition: aacdec.c:129
static void apply_dependent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply dependent channel coupling (applied before IMDCT).
Definition: aacdec.c:215
static float * VMUL4(float *dst, const float *v, unsigned idx, const float *scale)
Definition: aacdec.c:99
static INTFLOAT aac_kbd_short_120[120]
Definition: aacdec.c:75
static av_always_inline void predict(PredictorState *ps, float *coef, int output_enable)
Definition: aacdec.c:179
static void apply_independent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply independent channel coupling (applied after IMDCT).
Definition: aacdec.c:251
static INTFLOAT aac_kbd_long_960[960]
Definition: aacdec.c:74
static INTFLOAT sine_960[960]
Definition: aacdec.c:73
static INTFLOAT sine_120[120]
Definition: aacdec.c:72
static float * VMUL2(float *dst, const float *v, unsigned idx, const float *scale)
Definition: aacdec.c:88
static float * VMUL2S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
Definition: aacdec.c:112
AVFixedDSPContext * avpriv_alloc_fixed_dsp(int bit_exact)
Allocate and initialize a fixed DSP context.
Definition: fixed_dsp.c:149
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
int buffer_size_t
Definition: internal.h:306
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
#define AVOnce
Definition: thread.h:172
static int ff_thread_once(char *control, void(*routine)(void))
Definition: thread.h:175
#define AV_ONCE_INIT
Definition: thread.h:173
static av_always_inline float cbrtf(float x)
Definition: libm.h:61
uint8_t w
Definition: llviddspenc.c:39
static int AAC_RENAME() compute_lpc_coefs(const LPC_TYPE *autoc, int max_order, LPC_TYPE *lpc, int lpc_stride, int fail, int normalize)
Levinson-Durbin recursion.
Definition: lpc.h:166
#define M_SQRT2
Definition: mathematics.h:61
av_cold void ff_mdct15_uninit(MDCT15Context **ps)
Definition: mdct15.c:43
av_cold int ff_mdct15_init(MDCT15Context **ps, int inverse, int N, double scale)
Definition: mdct15.c:247
int ff_mpeg4audio_get_config_gb(MPEG4AudioConfig *c, GetBitContext *gb, int sync_extension, void *logctx)
Parse MPEG-4 systems extradata from a potentially unaligned GetBitContext to retrieve audio configura...
Definition: mpeg4audio.c:99
const uint8_t ff_mpeg4audio_channels[14]
Definition: mpeg4audio.c:67
@ AOT_AAC_LTP
Y Long Term Prediction.
Definition: mpeg4audio.h:93
@ AOT_ER_AAC_LD
N Error Resilient Low Delay.
Definition: mpeg4audio.h:109
@ AOT_ER_AAC_ELD
N Error Resilient Enhanced Low Delay.
Definition: mpeg4audio.h:125
@ AOT_ER_AAC_LC
N Error Resilient Low Complexity.
Definition: mpeg4audio.h:104
@ AOT_AAC_SCALABLE
N Scalable.
Definition: mpeg4audio.h:95
@ AOT_ER_BSAC
N Error Resilient Bit-Sliced Arithmetic Coding.
Definition: mpeg4audio.h:108
@ AOT_AAC_SSR
N (code in SoC repo) Scalable Sample Rate.
Definition: mpeg4audio.h:92
@ AOT_AAC_LC
Y Low Complexity.
Definition: mpeg4audio.h:91
@ AOT_ER_AAC_LTP
N Error Resilient Long Term Prediction.
Definition: mpeg4audio.h:105
@ AOT_ER_AAC_SCALABLE
N Error Resilient Scalable.
Definition: mpeg4audio.h:106
@ AOT_AAC_MAIN
Y Main.
Definition: mpeg4audio.h:90
const char data[16]
Definition: mxf.c:142
typedef void(RENAME(mix_any_func_type))
void ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
void ff_sine_window_init(float *window, int n)
Generate a sine window.
static av_cold void init_sine_windows_fixed(void)
#define FF_ARRAY_ELEMS(a)
const uint8_t * code
Definition: spdifenc.c:413
unsigned int pos
Definition: spdifenc.c:412
uint8_t num_aac_frames
Definition: adts_header.h:36
uint32_t sample_rate
Definition: adts_header.h:29
uint8_t object_type
Definition: adts_header.h:33
uint8_t chan_config
Definition: adts_header.h:35
uint8_t crc_absent
Definition: adts_header.h:32
uint8_t sampling_index
Definition: adts_header.h:34
main AAC context
Definition: aac.h:294
int warned_960_sbr
Definition: aac.h:359
FFTContext mdct
Definition: aac.h:324
SingleChannelElement * output_element[MAX_CHANNELS]
Points to each SingleChannelElement.
Definition: aac.h:343
FFTContext mdct_ld
Definition: aac.h:326
int tags_mapped
Definition: aac.h:308
AVCodecContext * avctx
Definition: aac.h:296
void(* update_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:370
int dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
Definition: aac.h:352
DynamicRangeControl che_drc
Definition: aac.h:300
void(* vector_pow43)(int *coefs, int len)
Definition: aac.h:371
OutputConfiguration oc[2]
Definition: aac.h:357
MDCT15Context * mdct120
Definition: aac.h:331
FFTContext mdct_ltp
Definition: aac.h:327
int force_dmono_mode
0->not dmono, 1->use first channel, 2->use second channel
Definition: aac.h:351
MDCT15Context * mdct480
Definition: aac.h:332
void(* apply_ltp)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:365
FFTContext mdct_small
Definition: aac.h:325
int warned_num_aac_frames
Definition: aac.h:358
MDCT15Context * mdct960
Definition: aac.h:333
AVFloatDSPContext * fdsp
Definition: aac.h:334
INTFLOAT buf_mdct[1024]
Definition: aac.h:317
void(* imdct_and_windowing)(AACContext *ac, SingleChannelElement *sce)
Definition: aac.h:364
int warned_gain_control
Definition: aac.h:361
void(* subband_scale)(int *dst, int *src, int scale, int offset, int len, void *log_context)
Definition: aac.h:372
int warned_remapping_once
Definition: aac.h:309
AVFrame * frame
Definition: aac.h:297
ChannelElement * tag_che_map[4][MAX_ELEM_ID]
Definition: aac.h:307
ChannelElement * che[4][MAX_ELEM_ID]
Definition: aac.h:306
int random_state
Definition: aac.h:336
void(* apply_tns)(INTFLOAT coef[1024], TemporalNoiseShaping *tns, IndividualChannelStream *ics, int decode)
Definition: aac.h:366
unsigned warned_71_wide
Definition: aac.h:360
INTFLOAT temp[128]
Definition: aac.h:355
void(* windowing_and_mdct_ltp)(AACContext *ac, INTFLOAT *out, INTFLOAT *in, IndividualChannelStream *ics)
Definition: aac.h:368
Describe the class of an AVClass context structure.
Definition: log.h:67
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
main external API structure.
Definition: avcodec.h:536
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
int debug
debug
Definition: avcodec.h:1623
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:1601
int profile
profile
Definition: avcodec.h:1858
int sample_rate
samples per second
Definition: avcodec.h:1196
uint64_t request_channel_layout
Request decoder to use this channel layout if it can (0 for default)
Definition: avcodec.h:1254
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:616
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:637
int channels
number of audio channels
Definition: avcodec.h:1197
int extradata_size
Definition: avcodec.h:638
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1247
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1216
struct AVCodecInternal * internal
Private context used for internal data.
Definition: avcodec.h:571
void * priv_data
Definition: avcodec.h:563
int err_recognition
Error recognition; may misdetect some more or less valid parts as errors.
Definition: avcodec.h:1645
int skip_samples_multiplier
Definition: internal.h:208
int skip_samples
Number of audio samples to skip at the start of the next decoded frame.
Definition: internal.h:175
void(* vector_fmul_reverse)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats, and store the result in a vector of floats...
Definition: float_dsp.h:154
void(* butterflies_float)(float *av_restrict v1, float *av_restrict v2, int len)
Calculate the sum and difference of two vectors of floats.
Definition: float_dsp.h:164
float(* scalarproduct_float)(const float *v1, const float *v2, int len)
Calculate the scalar product of two vectors of floats.
Definition: float_dsp.h:175
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
Definition: float_dsp.h:85
void(* vector_fmul_window)(float *dst, const float *src0, const float *src1, const float *win, int len)
Overlap/add with window function.
Definition: float_dsp.h:119
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
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:332
int sample_rate
Sample rate of the audio data.
Definition: frame.h:490
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
AVOption.
Definition: opt.h:248
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
uint8_t * data
Definition: packet.h:369
coupling parameters
Definition: aac.h:235
int id_select[8]
element id
Definition: aac.h:239
enum CouplingPoint coupling_point
The point during decoding at which coupling is applied.
Definition: aac.h:236
int num_coupled
number of target elements
Definition: aac.h:237
INTFLOAT gain[16][120]
Definition: aac.h:243
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:240
enum RawDataBlockType type[8]
Type of channel element to be coupled - SCE or CPE.
Definition: aac.h:238
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:276
int present
Definition: aac.h:277
ChannelCoupling coup
Definition: aac.h:287
uint8_t ms_mask[128]
Set if mid/side stereo is used for each scalefactor window band.
Definition: aac.h:282
SpectralBandReplication sbr
Definition: aac.h:288
SingleChannelElement ch[2]
Definition: aac.h:285
Dynamic Range Control - decoded from the bitstream but not processed further.
Definition: aac.h:212
int interpolation_scheme
Indicates the interpolation scheme used in the SBR QMF domain.
Definition: aac.h:218
int exclude_mask[MAX_CHANNELS]
Channels to be excluded from DRC processing.
Definition: aac.h:216
int band_incr
Number of DRC bands greater than 1 having DRC info.
Definition: aac.h:217
int dyn_rng_ctl[17]
DRC magnitude information.
Definition: aac.h:215
int prog_ref_level
A reference level for the long-term program audio level for all channels combined.
Definition: aac.h:220
int pce_instance_tag
Indicates with which program the DRC info is associated.
Definition: aac.h:213
int band_top[17]
Indicates the top of the i-th DRC band in units of 4 spectral lines.
Definition: aac.h:219
int dyn_rng_sgn[17]
DRC sign information; 0 - positive, 1 - negative.
Definition: aac.h:214
void(* imdct_half)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:103
void(* mdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:104
Individual Channel Stream.
Definition: aac.h:175
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:176
uint8_t group_len[8]
Definition: aac.h:180
int num_swb
number of scalefactor window bands
Definition: aac.h:184
LongTermPrediction ltp
Definition: aac.h:181
uint8_t use_kb_window[2]
If set, use Kaiser-Bessel window, otherwise use a sine window.
Definition: aac.h:178
int predictor_reset_group
Definition: aac.h:189
uint8_t prediction_used[41]
Definition: aac.h:191
int predictor_initialized
Definition: aac.h:188
enum WindowSequence window_sequence[2]
Definition: aac.h:177
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:182
Long Term Prediction.
Definition: aac.h:164
int8_t present
Definition: aac.h:165
int16_t lag
Definition: aac.h:166
INTFLOAT coef
Definition: aac.h:168
int8_t used[MAX_LTP_LONG_SFB]
Definition: aac.h:169
void(* imdct_half)(struct MDCT15Context *s, float *dst, const float *src, ptrdiff_t stride)
Definition: mdct15.h:54
int sbr
-1 implicit, 1 presence
Definition: mpeg4audio.h:38
int frame_length_short
Definition: mpeg4audio.h:45
int ps
-1 implicit, 1 presence
Definition: mpeg4audio.h:44
int layout_map_tags
Definition: aac.h:127
enum OCStatus status
Definition: aac.h:130
MPEG4AudioConfig m4ac
Definition: aac.h:125
uint8_t layout_map[MAX_ELEM_ID *4][3]
Definition: aac.h:126
uint64_t channel_layout
Definition: aac.h:129
Predictor State.
Definition: aac.h:136
Definition: aac.h:225
int pos[4]
Definition: aac.h:228
int amp[4]
Definition: aac.h:229
int num_pulse
Definition: aac.h:226
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:249
INTFLOAT * ret
PCM output.
Definition: aac.h:270
enum BandType band_type[128]
band types
Definition: aac.h:253
PredictorState predictor_state[MAX_PREDICTORS]
Definition: aac.h:269
int band_type_run_end[120]
band type run end points
Definition: aac.h:255
INTFLOAT ret_buf[2048]
PCM output buffer.
Definition: aac.h:265
TemporalNoiseShaping tns
Definition: aac.h:251
INTFLOAT sf[120]
scalefactors
Definition: aac.h:256
INTFLOAT coeffs[1024]
coefficients for IMDCT, maybe processed
Definition: aac.h:263
INTFLOAT saved[1536]
overlap
Definition: aac.h:264
INTFLOAT ltp_state[3072]
time signal for LTP
Definition: aac.h:266
IndividualChannelStream ics
Definition: aac.h:250
Temporal Noise Shaping.
Definition: aac.h:199
INTFLOAT coef[8][4][TNS_MAX_ORDER]
Definition: aac.h:206
int direction[8][4]
Definition: aac.h:203
int order[8][4]
Definition: aac.h:204
int length[8][4]
Definition: aac.h:202
int n_filt[8]
Definition: aac.h:201
Definition: vlc.h:26
int table_size
Definition: vlc.h:29
int table_allocated
Definition: vlc.h:29
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
#define ff_dlog(a,...)
#define avpriv_request_sample(...)
#define av_freep(p)
#define av_log(a,...)
static void error(const char *err)
static uint8_t tmp[11]
Definition: aes_ctr.c:27
FILE * out
Definition: movenc.c:54
static void imdct_and_window(TwinVQContext *tctx, enum TwinVQFrameType ftype, int wtype, float *in, float *prev, int ch)
Definition: twinvq.c:327
int size
const char * b
Definition: vf_curves.c:118
const char * g
Definition: vf_curves.c:117
else temp
Definition: vf_mcdeint.c:259
static const uint8_t offset[127][2]
Definition: vf_spp.c:107
#define INIT_VLC_STATIC(vlc, bits, a, b, c, d, e, f, g, static_size)
Definition: vlc.h:120
#define INIT_VLC_STATIC_OVERLONG
Definition: vlc.h:96
#define VLC_TYPE
Definition: vlc.h:24
int len
uint8_t bits
Definition: vp3data.h:141
static double c[64]