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aacdec.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  * This file is part of FFmpeg.
12  *
13  * FFmpeg is free software; you can redistribute it and/or
14  * modify it under the terms of the GNU Lesser General Public
15  * License as published by the Free Software Foundation; either
16  * version 2.1 of the License, or (at your option) any later version.
17  *
18  * FFmpeg is distributed in the hope that it will be useful,
19  * but WITHOUT ANY WARRANTY; without even the implied warranty of
20  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
21  * Lesser General Public License for more details.
22  *
23  * You should have received a copy of the GNU Lesser General Public
24  * License along with FFmpeg; if not, write to the Free Software
25  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
26  */
27 
28 /**
29  * @file
30  * AAC decoder
31  * @author Oded Shimon ( ods15 ods15 dyndns org )
32  * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
33  */
34 
35 /*
36  * supported tools
37  *
38  * Support? Name
39  * N (code in SoC repo) gain control
40  * Y block switching
41  * Y window shapes - standard
42  * N window shapes - Low Delay
43  * Y filterbank - standard
44  * N (code in SoC repo) filterbank - Scalable Sample Rate
45  * Y Temporal Noise Shaping
46  * Y Long Term Prediction
47  * Y intensity stereo
48  * Y channel coupling
49  * Y frequency domain prediction
50  * Y Perceptual Noise Substitution
51  * Y Mid/Side stereo
52  * N Scalable Inverse AAC Quantization
53  * N Frequency Selective Switch
54  * N upsampling filter
55  * Y quantization & coding - AAC
56  * N quantization & coding - TwinVQ
57  * N quantization & coding - BSAC
58  * N AAC Error Resilience tools
59  * N Error Resilience payload syntax
60  * N Error Protection tool
61  * N CELP
62  * N Silence Compression
63  * N HVXC
64  * N HVXC 4kbits/s VR
65  * N Structured Audio tools
66  * N Structured Audio Sample Bank Format
67  * N MIDI
68  * N Harmonic and Individual Lines plus Noise
69  * N Text-To-Speech Interface
70  * Y Spectral Band Replication
71  * Y (not in this code) Layer-1
72  * Y (not in this code) Layer-2
73  * Y (not in this code) Layer-3
74  * N SinuSoidal Coding (Transient, Sinusoid, Noise)
75  * Y Parametric Stereo
76  * N Direct Stream Transfer
77  * Y Enhanced AAC Low Delay (ER AAC ELD)
78  *
79  * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
80  * - HE AAC v2 comprises LC AAC with Spectral Band Replication and
81  Parametric Stereo.
82  */
83 
84 #include "libavutil/float_dsp.h"
85 #include "libavutil/opt.h"
86 #include "avcodec.h"
87 #include "internal.h"
88 #include "get_bits.h"
89 #include "fft.h"
90 #include "fmtconvert.h"
91 #include "lpc.h"
92 #include "kbdwin.h"
93 #include "sinewin.h"
94 
95 #include "aac.h"
96 #include "aactab.h"
97 #include "aacdectab.h"
98 #include "cbrt_tablegen.h"
99 #include "sbr.h"
100 #include "aacsbr.h"
101 #include "mpeg4audio.h"
102 #include "aacadtsdec.h"
103 #include "libavutil/intfloat.h"
104 
105 #include <assert.h>
106 #include <errno.h>
107 #include <math.h>
108 #include <stdint.h>
109 #include <string.h>
110 
111 #if ARCH_ARM
112 # include "arm/aac.h"
113 #elif ARCH_MIPS
114 # include "mips/aacdec_mips.h"
115 #endif
116 
118 static VLC vlc_spectral[11];
119 
120 static int output_configure(AACContext *ac,
121  uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
122  enum OCStatus oc_type, int get_new_frame);
123 
124 #define overread_err "Input buffer exhausted before END element found\n"
125 
126 static int count_channels(uint8_t (*layout)[3], int tags)
127 {
128  int i, sum = 0;
129  for (i = 0; i < tags; i++) {
130  int syn_ele = layout[i][0];
131  int pos = layout[i][2];
132  sum += (1 + (syn_ele == TYPE_CPE)) *
133  (pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);
134  }
135  return sum;
136 }
137 
138 /**
139  * Check for the channel element in the current channel position configuration.
140  * If it exists, make sure the appropriate element is allocated and map the
141  * channel order to match the internal FFmpeg channel layout.
142  *
143  * @param che_pos current channel position configuration
144  * @param type channel element type
145  * @param id channel element id
146  * @param channels count of the number of channels in the configuration
147  *
148  * @return Returns error status. 0 - OK, !0 - error
149  */
151  enum ChannelPosition che_pos,
152  int type, int id, int *channels)
153 {
154  if (*channels >= MAX_CHANNELS)
155  return AVERROR_INVALIDDATA;
156  if (che_pos) {
157  if (!ac->che[type][id]) {
158  if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
159  return AVERROR(ENOMEM);
160  ff_aac_sbr_ctx_init(ac, &ac->che[type][id]->sbr);
161  }
162  if (type != TYPE_CCE) {
163  if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
164  av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
165  return AVERROR_INVALIDDATA;
166  }
167  ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
168  if (type == TYPE_CPE ||
169  (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
170  ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
171  }
172  }
173  } else {
174  if (ac->che[type][id])
175  ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
176  av_freep(&ac->che[type][id]);
177  }
178  return 0;
179 }
180 
182 {
183  AACContext *ac = avctx->priv_data;
184  int type, id, ch, ret;
185 
186  /* set channel pointers to internal buffers by default */
187  for (type = 0; type < 4; type++) {
188  for (id = 0; id < MAX_ELEM_ID; id++) {
189  ChannelElement *che = ac->che[type][id];
190  if (che) {
191  che->ch[0].ret = che->ch[0].ret_buf;
192  che->ch[1].ret = che->ch[1].ret_buf;
193  }
194  }
195  }
196 
197  /* get output buffer */
198  av_frame_unref(ac->frame);
199  if (!avctx->channels)
200  return 1;
201 
202  ac->frame->nb_samples = 2048;
203  if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
204  return ret;
205 
206  /* map output channel pointers to AVFrame data */
207  for (ch = 0; ch < avctx->channels; ch++) {
208  if (ac->output_element[ch])
209  ac->output_element[ch]->ret = (float *)ac->frame->extended_data[ch];
210  }
211 
212  return 0;
213 }
214 
216  uint64_t av_position;
220 };
221 
222 static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
223  uint8_t (*layout_map)[3], int offset, uint64_t left,
224  uint64_t right, int pos)
225 {
226  if (layout_map[offset][0] == TYPE_CPE) {
227  e2c_vec[offset] = (struct elem_to_channel) {
228  .av_position = left | right,
229  .syn_ele = TYPE_CPE,
230  .elem_id = layout_map[offset][1],
231  .aac_position = pos
232  };
233  return 1;
234  } else {
235  e2c_vec[offset] = (struct elem_to_channel) {
236  .av_position = left,
237  .syn_ele = TYPE_SCE,
238  .elem_id = layout_map[offset][1],
239  .aac_position = pos
240  };
241  e2c_vec[offset + 1] = (struct elem_to_channel) {
242  .av_position = right,
243  .syn_ele = TYPE_SCE,
244  .elem_id = layout_map[offset + 1][1],
245  .aac_position = pos
246  };
247  return 2;
248  }
249 }
250 
251 static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
252  int *current)
253 {
254  int num_pos_channels = 0;
255  int first_cpe = 0;
256  int sce_parity = 0;
257  int i;
258  for (i = *current; i < tags; i++) {
259  if (layout_map[i][2] != pos)
260  break;
261  if (layout_map[i][0] == TYPE_CPE) {
262  if (sce_parity) {
263  if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
264  sce_parity = 0;
265  } else {
266  return -1;
267  }
268  }
269  num_pos_channels += 2;
270  first_cpe = 1;
271  } else {
272  num_pos_channels++;
273  sce_parity ^= 1;
274  }
275  }
276  if (sce_parity &&
277  ((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
278  return -1;
279  *current = i;
280  return num_pos_channels;
281 }
282 
283 static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
284 {
285  int i, n, total_non_cc_elements;
286  struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
287  int num_front_channels, num_side_channels, num_back_channels;
288  uint64_t layout;
289 
290  if (FF_ARRAY_ELEMS(e2c_vec) < tags)
291  return 0;
292 
293  i = 0;
294  num_front_channels =
295  count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
296  if (num_front_channels < 0)
297  return 0;
298  num_side_channels =
299  count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
300  if (num_side_channels < 0)
301  return 0;
302  num_back_channels =
303  count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
304  if (num_back_channels < 0)
305  return 0;
306 
307  i = 0;
308  if (num_front_channels & 1) {
309  e2c_vec[i] = (struct elem_to_channel) {
311  .syn_ele = TYPE_SCE,
312  .elem_id = layout_map[i][1],
313  .aac_position = AAC_CHANNEL_FRONT
314  };
315  i++;
316  num_front_channels--;
317  }
318  if (num_front_channels >= 4) {
319  i += assign_pair(e2c_vec, layout_map, i,
323  num_front_channels -= 2;
324  }
325  if (num_front_channels >= 2) {
326  i += assign_pair(e2c_vec, layout_map, i,
330  num_front_channels -= 2;
331  }
332  while (num_front_channels >= 2) {
333  i += assign_pair(e2c_vec, layout_map, i,
334  UINT64_MAX,
335  UINT64_MAX,
337  num_front_channels -= 2;
338  }
339 
340  if (num_side_channels >= 2) {
341  i += assign_pair(e2c_vec, layout_map, i,
345  num_side_channels -= 2;
346  }
347  while (num_side_channels >= 2) {
348  i += assign_pair(e2c_vec, layout_map, i,
349  UINT64_MAX,
350  UINT64_MAX,
352  num_side_channels -= 2;
353  }
354 
355  while (num_back_channels >= 4) {
356  i += assign_pair(e2c_vec, layout_map, i,
357  UINT64_MAX,
358  UINT64_MAX,
360  num_back_channels -= 2;
361  }
362  if (num_back_channels >= 2) {
363  i += assign_pair(e2c_vec, layout_map, i,
367  num_back_channels -= 2;
368  }
369  if (num_back_channels) {
370  e2c_vec[i] = (struct elem_to_channel) {
372  .syn_ele = TYPE_SCE,
373  .elem_id = layout_map[i][1],
374  .aac_position = AAC_CHANNEL_BACK
375  };
376  i++;
377  num_back_channels--;
378  }
379 
380  if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
381  e2c_vec[i] = (struct elem_to_channel) {
383  .syn_ele = TYPE_LFE,
384  .elem_id = layout_map[i][1],
385  .aac_position = AAC_CHANNEL_LFE
386  };
387  i++;
388  }
389  while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
390  e2c_vec[i] = (struct elem_to_channel) {
391  .av_position = UINT64_MAX,
392  .syn_ele = TYPE_LFE,
393  .elem_id = layout_map[i][1],
394  .aac_position = AAC_CHANNEL_LFE
395  };
396  i++;
397  }
398 
399  // Must choose a stable sort
400  total_non_cc_elements = n = i;
401  do {
402  int next_n = 0;
403  for (i = 1; i < n; i++)
404  if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
405  FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
406  next_n = i;
407  }
408  n = next_n;
409  } while (n > 0);
410 
411  layout = 0;
412  for (i = 0; i < total_non_cc_elements; i++) {
413  layout_map[i][0] = e2c_vec[i].syn_ele;
414  layout_map[i][1] = e2c_vec[i].elem_id;
415  layout_map[i][2] = e2c_vec[i].aac_position;
416  if (e2c_vec[i].av_position != UINT64_MAX) {
417  layout |= e2c_vec[i].av_position;
418  }
419  }
420 
421  return layout;
422 }
423 
424 /**
425  * Save current output configuration if and only if it has been locked.
426  */
428  if (ac->oc[1].status == OC_LOCKED) {
429  ac->oc[0] = ac->oc[1];
430  }
431  ac->oc[1].status = OC_NONE;
432 }
433 
434 /**
435  * Restore the previous output configuration if and only if the current
436  * configuration is unlocked.
437  */
439  if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
440  ac->oc[1] = ac->oc[0];
441  ac->avctx->channels = ac->oc[1].channels;
442  ac->avctx->channel_layout = ac->oc[1].channel_layout;
443  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
444  ac->oc[1].status, 0);
445  }
446 }
447 
448 /**
449  * Configure output channel order based on the current program
450  * configuration element.
451  *
452  * @return Returns error status. 0 - OK, !0 - error
453  */
455  uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
456  enum OCStatus oc_type, int get_new_frame)
457 {
458  AVCodecContext *avctx = ac->avctx;
459  int i, channels = 0, ret;
460  uint64_t layout = 0;
461 
462  if (ac->oc[1].layout_map != layout_map) {
463  memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
464  ac->oc[1].layout_map_tags = tags;
465  }
466 
467  // Try to sniff a reasonable channel order, otherwise output the
468  // channels in the order the PCE declared them.
470  layout = sniff_channel_order(layout_map, tags);
471  for (i = 0; i < tags; i++) {
472  int type = layout_map[i][0];
473  int id = layout_map[i][1];
474  int position = layout_map[i][2];
475  // Allocate or free elements depending on if they are in the
476  // current program configuration.
477  ret = che_configure(ac, position, type, id, &channels);
478  if (ret < 0)
479  return ret;
480  }
481  if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
482  if (layout == AV_CH_FRONT_CENTER) {
484  } else {
485  layout = 0;
486  }
487  }
488 
489  memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
490  if (layout) avctx->channel_layout = layout;
491  ac->oc[1].channel_layout = layout;
492  avctx->channels = ac->oc[1].channels = channels;
493  ac->oc[1].status = oc_type;
494 
495  if (get_new_frame) {
496  if ((ret = frame_configure_elements(ac->avctx)) < 0)
497  return ret;
498  }
499 
500  return 0;
501 }
502 
503 static void flush(AVCodecContext *avctx)
504 {
505  AACContext *ac= avctx->priv_data;
506  int type, i, j;
507 
508  for (type = 3; type >= 0; type--) {
509  for (i = 0; i < MAX_ELEM_ID; i++) {
510  ChannelElement *che = ac->che[type][i];
511  if (che) {
512  for (j = 0; j <= 1; j++) {
513  memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
514  }
515  }
516  }
517  }
518 }
519 
520 /**
521  * Set up channel positions based on a default channel configuration
522  * as specified in table 1.17.
523  *
524  * @return Returns error status. 0 - OK, !0 - error
525  */
527  uint8_t (*layout_map)[3],
528  int *tags,
529  int channel_config)
530 {
531  if (channel_config < 1 || channel_config > 7) {
532  av_log(avctx, AV_LOG_ERROR,
533  "invalid default channel configuration (%d)\n",
534  channel_config);
535  return AVERROR_INVALIDDATA;
536  }
537  *tags = tags_per_config[channel_config];
538  memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
539  *tags * sizeof(*layout_map));
540 
541  /*
542  * AAC specification has 7.1(wide) as a default layout for 8-channel streams.
543  * However, at least Nero AAC encoder encodes 7.1 streams using the default
544  * channel config 7, mapping the side channels of the original audio stream
545  * to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
546  * decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
547  * the incorrect streams as if they were correct (and as the encoder intended).
548  *
549  * As actual intended 7.1(wide) streams are very rare, default to assuming a
550  * 7.1 layout was intended.
551  */
552  if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
553  av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
554  " instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
555  " according to the specification instead.\n", FF_COMPLIANCE_STRICT);
556  layout_map[2][2] = AAC_CHANNEL_SIDE;
557  }
558 
559  return 0;
560 }
561 
562 static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
563 {
564  /* For PCE based channel configurations map the channels solely based
565  * on tags. */
566  if (!ac->oc[1].m4ac.chan_config) {
567  return ac->tag_che_map[type][elem_id];
568  }
569  // Allow single CPE stereo files to be signalled with mono configuration.
570  if (!ac->tags_mapped && type == TYPE_CPE &&
571  ac->oc[1].m4ac.chan_config == 1) {
572  uint8_t layout_map[MAX_ELEM_ID*4][3];
573  int layout_map_tags;
575 
576  av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
577 
578  if (set_default_channel_config(ac->avctx, layout_map,
579  &layout_map_tags, 2) < 0)
580  return NULL;
581  if (output_configure(ac, layout_map, layout_map_tags,
582  OC_TRIAL_FRAME, 1) < 0)
583  return NULL;
584 
585  ac->oc[1].m4ac.chan_config = 2;
586  ac->oc[1].m4ac.ps = 0;
587  }
588  // And vice-versa
589  if (!ac->tags_mapped && type == TYPE_SCE &&
590  ac->oc[1].m4ac.chan_config == 2) {
591  uint8_t layout_map[MAX_ELEM_ID * 4][3];
592  int layout_map_tags;
594 
595  av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
596 
597  if (set_default_channel_config(ac->avctx, layout_map,
598  &layout_map_tags, 1) < 0)
599  return NULL;
600  if (output_configure(ac, layout_map, layout_map_tags,
601  OC_TRIAL_FRAME, 1) < 0)
602  return NULL;
603 
604  ac->oc[1].m4ac.chan_config = 1;
605  if (ac->oc[1].m4ac.sbr)
606  ac->oc[1].m4ac.ps = -1;
607  }
608  /* For indexed channel configurations map the channels solely based
609  * on position. */
610  switch (ac->oc[1].m4ac.chan_config) {
611  case 7:
612  if (ac->tags_mapped == 3 && type == TYPE_CPE) {
613  ac->tags_mapped++;
614  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
615  }
616  case 6:
617  /* Some streams incorrectly code 5.1 audio as
618  * SCE[0] CPE[0] CPE[1] SCE[1]
619  * instead of
620  * SCE[0] CPE[0] CPE[1] LFE[0].
621  * If we seem to have encountered such a stream, transfer
622  * the LFE[0] element to the SCE[1]'s mapping */
623  if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
624  ac->tags_mapped++;
625  return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
626  }
627  case 5:
628  if (ac->tags_mapped == 2 && type == TYPE_CPE) {
629  ac->tags_mapped++;
630  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
631  }
632  case 4:
633  if (ac->tags_mapped == 2 &&
634  ac->oc[1].m4ac.chan_config == 4 &&
635  type == TYPE_SCE) {
636  ac->tags_mapped++;
637  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
638  }
639  case 3:
640  case 2:
641  if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
642  type == TYPE_CPE) {
643  ac->tags_mapped++;
644  return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
645  } else if (ac->oc[1].m4ac.chan_config == 2) {
646  return NULL;
647  }
648  case 1:
649  if (!ac->tags_mapped && type == TYPE_SCE) {
650  ac->tags_mapped++;
651  return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
652  }
653  default:
654  return NULL;
655  }
656 }
657 
658 /**
659  * Decode an array of 4 bit element IDs, optionally interleaved with a
660  * stereo/mono switching bit.
661  *
662  * @param type speaker type/position for these channels
663  */
664 static void decode_channel_map(uint8_t layout_map[][3],
665  enum ChannelPosition type,
666  GetBitContext *gb, int n)
667 {
668  while (n--) {
669  enum RawDataBlockType syn_ele;
670  switch (type) {
671  case AAC_CHANNEL_FRONT:
672  case AAC_CHANNEL_BACK:
673  case AAC_CHANNEL_SIDE:
674  syn_ele = get_bits1(gb);
675  break;
676  case AAC_CHANNEL_CC:
677  skip_bits1(gb);
678  syn_ele = TYPE_CCE;
679  break;
680  case AAC_CHANNEL_LFE:
681  syn_ele = TYPE_LFE;
682  break;
683  default:
684  av_assert0(0);
685  }
686  layout_map[0][0] = syn_ele;
687  layout_map[0][1] = get_bits(gb, 4);
688  layout_map[0][2] = type;
689  layout_map++;
690  }
691 }
692 
693 /**
694  * Decode program configuration element; reference: table 4.2.
695  *
696  * @return Returns error status. 0 - OK, !0 - error
697  */
698 static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
699  uint8_t (*layout_map)[3],
700  GetBitContext *gb)
701 {
702  int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
703  int sampling_index;
704  int comment_len;
705  int tags;
706 
707  skip_bits(gb, 2); // object_type
708 
709  sampling_index = get_bits(gb, 4);
710  if (m4ac->sampling_index != sampling_index)
711  av_log(avctx, AV_LOG_WARNING,
712  "Sample rate index in program config element does not "
713  "match the sample rate index configured by the container.\n");
714 
715  num_front = get_bits(gb, 4);
716  num_side = get_bits(gb, 4);
717  num_back = get_bits(gb, 4);
718  num_lfe = get_bits(gb, 2);
719  num_assoc_data = get_bits(gb, 3);
720  num_cc = get_bits(gb, 4);
721 
722  if (get_bits1(gb))
723  skip_bits(gb, 4); // mono_mixdown_tag
724  if (get_bits1(gb))
725  skip_bits(gb, 4); // stereo_mixdown_tag
726 
727  if (get_bits1(gb))
728  skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
729 
730  if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
731  av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
732  return -1;
733  }
734  decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front);
735  tags = num_front;
736  decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side);
737  tags += num_side;
738  decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back);
739  tags += num_back;
740  decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe);
741  tags += num_lfe;
742 
743  skip_bits_long(gb, 4 * num_assoc_data);
744 
745  decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc);
746  tags += num_cc;
747 
748  align_get_bits(gb);
749 
750  /* comment field, first byte is length */
751  comment_len = get_bits(gb, 8) * 8;
752  if (get_bits_left(gb) < comment_len) {
753  av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
754  return AVERROR_INVALIDDATA;
755  }
756  skip_bits_long(gb, comment_len);
757  return tags;
758 }
759 
760 /**
761  * Decode GA "General Audio" specific configuration; reference: table 4.1.
762  *
763  * @param ac pointer to AACContext, may be null
764  * @param avctx pointer to AVCCodecContext, used for logging
765  *
766  * @return Returns error status. 0 - OK, !0 - error
767  */
769  GetBitContext *gb,
770  MPEG4AudioConfig *m4ac,
771  int channel_config)
772 {
773  int extension_flag, ret, ep_config, res_flags;
774  uint8_t layout_map[MAX_ELEM_ID*4][3];
775  int tags = 0;
776 
777  if (get_bits1(gb)) { // frameLengthFlag
778  avpriv_request_sample(avctx, "960/120 MDCT window");
779  return AVERROR_PATCHWELCOME;
780  }
781 
782  if (get_bits1(gb)) // dependsOnCoreCoder
783  skip_bits(gb, 14); // coreCoderDelay
784  extension_flag = get_bits1(gb);
785 
786  if (m4ac->object_type == AOT_AAC_SCALABLE ||
788  skip_bits(gb, 3); // layerNr
789 
790  if (channel_config == 0) {
791  skip_bits(gb, 4); // element_instance_tag
792  tags = decode_pce(avctx, m4ac, layout_map, gb);
793  if (tags < 0)
794  return tags;
795  } else {
796  if ((ret = set_default_channel_config(avctx, layout_map,
797  &tags, channel_config)))
798  return ret;
799  }
800 
801  if (count_channels(layout_map, tags) > 1) {
802  m4ac->ps = 0;
803  } else if (m4ac->sbr == 1 && m4ac->ps == -1)
804  m4ac->ps = 1;
805 
806  if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
807  return ret;
808 
809  if (extension_flag) {
810  switch (m4ac->object_type) {
811  case AOT_ER_BSAC:
812  skip_bits(gb, 5); // numOfSubFrame
813  skip_bits(gb, 11); // layer_length
814  break;
815  case AOT_ER_AAC_LC:
816  case AOT_ER_AAC_LTP:
817  case AOT_ER_AAC_SCALABLE:
818  case AOT_ER_AAC_LD:
819  res_flags = get_bits(gb, 3);
820  if (res_flags) {
822  "AAC data resilience (flags %x)",
823  res_flags);
824  return AVERROR_PATCHWELCOME;
825  }
826  break;
827  }
828  skip_bits1(gb); // extensionFlag3 (TBD in version 3)
829  }
830  switch (m4ac->object_type) {
831  case AOT_ER_AAC_LC:
832  case AOT_ER_AAC_LTP:
833  case AOT_ER_AAC_SCALABLE:
834  case AOT_ER_AAC_LD:
835  ep_config = get_bits(gb, 2);
836  if (ep_config) {
838  "epConfig %d", ep_config);
839  return AVERROR_PATCHWELCOME;
840  }
841  }
842  return 0;
843 }
844 
846  GetBitContext *gb,
847  MPEG4AudioConfig *m4ac,
848  int channel_config)
849 {
850  int ret, ep_config, res_flags;
851  uint8_t layout_map[MAX_ELEM_ID*4][3];
852  int tags = 0;
853  const int ELDEXT_TERM = 0;
854 
855  m4ac->ps = 0;
856  m4ac->sbr = 0;
857 
858  if (get_bits1(gb)) { // frameLengthFlag
859  avpriv_request_sample(avctx, "960/120 MDCT window");
860  return AVERROR_PATCHWELCOME;
861  }
862 
863  res_flags = get_bits(gb, 3);
864  if (res_flags) {
866  "AAC data resilience (flags %x)",
867  res_flags);
868  return AVERROR_PATCHWELCOME;
869  }
870 
871  if (get_bits1(gb)) { // ldSbrPresentFlag
873  "Low Delay SBR");
874  return AVERROR_PATCHWELCOME;
875  }
876 
877  while (get_bits(gb, 4) != ELDEXT_TERM) {
878  int len = get_bits(gb, 4);
879  if (len == 15)
880  len += get_bits(gb, 8);
881  if (len == 15 + 255)
882  len += get_bits(gb, 16);
883  if (get_bits_left(gb) < len * 8 + 4) {
885  return AVERROR_INVALIDDATA;
886  }
887  skip_bits_long(gb, 8 * len);
888  }
889 
890  if ((ret = set_default_channel_config(avctx, layout_map,
891  &tags, channel_config)))
892  return ret;
893 
894  if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
895  return ret;
896 
897  ep_config = get_bits(gb, 2);
898  if (ep_config) {
900  "epConfig %d", ep_config);
901  return AVERROR_PATCHWELCOME;
902  }
903  return 0;
904 }
905 
906 /**
907  * Decode audio specific configuration; reference: table 1.13.
908  *
909  * @param ac pointer to AACContext, may be null
910  * @param avctx pointer to AVCCodecContext, used for logging
911  * @param m4ac pointer to MPEG4AudioConfig, used for parsing
912  * @param data pointer to buffer holding an audio specific config
913  * @param bit_size size of audio specific config or data in bits
914  * @param sync_extension look for an appended sync extension
915  *
916  * @return Returns error status or number of consumed bits. <0 - error
917  */
919  AVCodecContext *avctx,
920  MPEG4AudioConfig *m4ac,
921  const uint8_t *data, int bit_size,
922  int sync_extension)
923 {
924  GetBitContext gb;
925  int i, ret;
926 
927  av_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
928  for (i = 0; i < bit_size >> 3; i++)
929  av_dlog(avctx, "%02x ", data[i]);
930  av_dlog(avctx, "\n");
931 
932  if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
933  return ret;
934 
935  if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size,
936  sync_extension)) < 0)
937  return AVERROR_INVALIDDATA;
938  if (m4ac->sampling_index > 12) {
939  av_log(avctx, AV_LOG_ERROR,
940  "invalid sampling rate index %d\n",
941  m4ac->sampling_index);
942  return AVERROR_INVALIDDATA;
943  }
944  if (m4ac->object_type == AOT_ER_AAC_LD &&
945  (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
946  av_log(avctx, AV_LOG_ERROR,
947  "invalid low delay sampling rate index %d\n",
948  m4ac->sampling_index);
949  return AVERROR_INVALIDDATA;
950  }
951 
952  skip_bits_long(&gb, i);
953 
954  switch (m4ac->object_type) {
955  case AOT_AAC_MAIN:
956  case AOT_AAC_LC:
957  case AOT_AAC_LTP:
958  case AOT_ER_AAC_LC:
959  case AOT_ER_AAC_LD:
960  if ((ret = decode_ga_specific_config(ac, avctx, &gb,
961  m4ac, m4ac->chan_config)) < 0)
962  return ret;
963  break;
964  case AOT_ER_AAC_ELD:
965  if ((ret = decode_eld_specific_config(ac, avctx, &gb,
966  m4ac, m4ac->chan_config)) < 0)
967  return ret;
968  break;
969  default:
971  "Audio object type %s%d",
972  m4ac->sbr == 1 ? "SBR+" : "",
973  m4ac->object_type);
974  return AVERROR(ENOSYS);
975  }
976 
977  av_dlog(avctx,
978  "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
979  m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
980  m4ac->sample_rate, m4ac->sbr,
981  m4ac->ps);
982 
983  return get_bits_count(&gb);
984 }
985 
986 /**
987  * linear congruential pseudorandom number generator
988  *
989  * @param previous_val pointer to the current state of the generator
990  *
991  * @return Returns a 32-bit pseudorandom integer
992  */
993 static av_always_inline int lcg_random(unsigned previous_val)
994 {
995  union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
996  return v.s;
997 }
998 
1000 {
1001  ps->r0 = 0.0f;
1002  ps->r1 = 0.0f;
1003  ps->cor0 = 0.0f;
1004  ps->cor1 = 0.0f;
1005  ps->var0 = 1.0f;
1006  ps->var1 = 1.0f;
1007 }
1008 
1010 {
1011  int i;
1012  for (i = 0; i < MAX_PREDICTORS; i++)
1013  reset_predict_state(&ps[i]);
1014 }
1015 
1016 static int sample_rate_idx (int rate)
1017 {
1018  if (92017 <= rate) return 0;
1019  else if (75132 <= rate) return 1;
1020  else if (55426 <= rate) return 2;
1021  else if (46009 <= rate) return 3;
1022  else if (37566 <= rate) return 4;
1023  else if (27713 <= rate) return 5;
1024  else if (23004 <= rate) return 6;
1025  else if (18783 <= rate) return 7;
1026  else if (13856 <= rate) return 8;
1027  else if (11502 <= rate) return 9;
1028  else if (9391 <= rate) return 10;
1029  else return 11;
1030 }
1031 
1032 static void reset_predictor_group(PredictorState *ps, int group_num)
1033 {
1034  int i;
1035  for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
1036  reset_predict_state(&ps[i]);
1037 }
1038 
1039 #define AAC_INIT_VLC_STATIC(num, size) \
1040  INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
1041  ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \
1042  sizeof(ff_aac_spectral_bits[num][0]), \
1043  ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
1044  sizeof(ff_aac_spectral_codes[num][0]), \
1045  size);
1046 
1047 static void aacdec_init(AACContext *ac);
1048 
1050 {
1051  AACContext *ac = avctx->priv_data;
1052  int ret;
1053 
1054  ac->avctx = avctx;
1055  ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
1056 
1057  aacdec_init(ac);
1058 
1059  avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1060 
1061  if (avctx->extradata_size > 0) {
1062  if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
1063  avctx->extradata,
1064  avctx->extradata_size * 8,
1065  1)) < 0)
1066  return ret;
1067  } else {
1068  int sr, i;
1069  uint8_t layout_map[MAX_ELEM_ID*4][3];
1070  int layout_map_tags;
1071 
1072  sr = sample_rate_idx(avctx->sample_rate);
1073  ac->oc[1].m4ac.sampling_index = sr;
1074  ac->oc[1].m4ac.channels = avctx->channels;
1075  ac->oc[1].m4ac.sbr = -1;
1076  ac->oc[1].m4ac.ps = -1;
1077 
1078  for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
1079  if (ff_mpeg4audio_channels[i] == avctx->channels)
1080  break;
1082  i = 0;
1083  }
1084  ac->oc[1].m4ac.chan_config = i;
1085 
1086  if (ac->oc[1].m4ac.chan_config) {
1087  int ret = set_default_channel_config(avctx, layout_map,
1088  &layout_map_tags, ac->oc[1].m4ac.chan_config);
1089  if (!ret)
1090  output_configure(ac, layout_map, layout_map_tags,
1091  OC_GLOBAL_HDR, 0);
1092  else if (avctx->err_recognition & AV_EF_EXPLODE)
1093  return AVERROR_INVALIDDATA;
1094  }
1095  }
1096 
1097  if (avctx->channels > MAX_CHANNELS) {
1098  av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
1099  return AVERROR_INVALIDDATA;
1100  }
1101 
1102  AAC_INIT_VLC_STATIC( 0, 304);
1103  AAC_INIT_VLC_STATIC( 1, 270);
1104  AAC_INIT_VLC_STATIC( 2, 550);
1105  AAC_INIT_VLC_STATIC( 3, 300);
1106  AAC_INIT_VLC_STATIC( 4, 328);
1107  AAC_INIT_VLC_STATIC( 5, 294);
1108  AAC_INIT_VLC_STATIC( 6, 306);
1109  AAC_INIT_VLC_STATIC( 7, 268);
1110  AAC_INIT_VLC_STATIC( 8, 510);
1111  AAC_INIT_VLC_STATIC( 9, 366);
1112  AAC_INIT_VLC_STATIC(10, 462);
1113 
1114  ff_aac_sbr_init();
1115 
1116  ff_fmt_convert_init(&ac->fmt_conv, avctx);
1118 
1119  ac->random_state = 0x1f2e3d4c;
1120 
1121  ff_aac_tableinit();
1122 
1123  INIT_VLC_STATIC(&vlc_scalefactors, 7,
1126  sizeof(ff_aac_scalefactor_bits[0]),
1127  sizeof(ff_aac_scalefactor_bits[0]),
1129  sizeof(ff_aac_scalefactor_code[0]),
1130  sizeof(ff_aac_scalefactor_code[0]),
1131  352);
1132 
1133  ff_mdct_init(&ac->mdct, 11, 1, 1.0 / (32768.0 * 1024.0));
1134  ff_mdct_init(&ac->mdct_ld, 10, 1, 1.0 / (32768.0 * 512.0));
1135  ff_mdct_init(&ac->mdct_small, 8, 1, 1.0 / (32768.0 * 128.0));
1136  ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0 * 32768.0);
1137  // window initialization
1143 
1144  cbrt_tableinit();
1145 
1146  return 0;
1147 }
1148 
1149 /**
1150  * Skip data_stream_element; reference: table 4.10.
1151  */
1153 {
1154  int byte_align = get_bits1(gb);
1155  int count = get_bits(gb, 8);
1156  if (count == 255)
1157  count += get_bits(gb, 8);
1158  if (byte_align)
1159  align_get_bits(gb);
1160 
1161  if (get_bits_left(gb) < 8 * count) {
1162  av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
1163  return AVERROR_INVALIDDATA;
1164  }
1165  skip_bits_long(gb, 8 * count);
1166  return 0;
1167 }
1168 
1170  GetBitContext *gb)
1171 {
1172  int sfb;
1173  if (get_bits1(gb)) {
1174  ics->predictor_reset_group = get_bits(gb, 5);
1175  if (ics->predictor_reset_group == 0 ||
1176  ics->predictor_reset_group > 30) {
1177  av_log(ac->avctx, AV_LOG_ERROR,
1178  "Invalid Predictor Reset Group.\n");
1179  return AVERROR_INVALIDDATA;
1180  }
1181  }
1182  for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
1183  ics->prediction_used[sfb] = get_bits1(gb);
1184  }
1185  return 0;
1186 }
1187 
1188 /**
1189  * Decode Long Term Prediction data; reference: table 4.xx.
1190  */
1192  GetBitContext *gb, uint8_t max_sfb)
1193 {
1194  int sfb;
1195 
1196  ltp->lag = get_bits(gb, 11);
1197  ltp->coef = ltp_coef[get_bits(gb, 3)];
1198  for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
1199  ltp->used[sfb] = get_bits1(gb);
1200 }
1201 
1202 /**
1203  * Decode Individual Channel Stream info; reference: table 4.6.
1204  */
1206  GetBitContext *gb)
1207 {
1208  int aot = ac->oc[1].m4ac.object_type;
1209  if (aot != AOT_ER_AAC_ELD) {
1210  if (get_bits1(gb)) {
1211  av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
1212  return AVERROR_INVALIDDATA;
1213  }
1214  ics->window_sequence[1] = ics->window_sequence[0];
1215  ics->window_sequence[0] = get_bits(gb, 2);
1216  if (aot == AOT_ER_AAC_LD &&
1217  ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
1218  av_log(ac->avctx, AV_LOG_ERROR,
1219  "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
1220  "window sequence %d found.\n", ics->window_sequence[0]);
1222  return AVERROR_INVALIDDATA;
1223  }
1224  ics->use_kb_window[1] = ics->use_kb_window[0];
1225  ics->use_kb_window[0] = get_bits1(gb);
1226  }
1227  ics->num_window_groups = 1;
1228  ics->group_len[0] = 1;
1229  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1230  int i;
1231  ics->max_sfb = get_bits(gb, 4);
1232  for (i = 0; i < 7; i++) {
1233  if (get_bits1(gb)) {
1234  ics->group_len[ics->num_window_groups - 1]++;
1235  } else {
1236  ics->num_window_groups++;
1237  ics->group_len[ics->num_window_groups - 1] = 1;
1238  }
1239  }
1240  ics->num_windows = 8;
1244  ics->predictor_present = 0;
1245  } else {
1246  ics->max_sfb = get_bits(gb, 6);
1247  ics->num_windows = 1;
1248  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
1252  if (!ics->num_swb || !ics->swb_offset)
1253  return AVERROR_BUG;
1254  } else {
1258  }
1259  if (aot != AOT_ER_AAC_ELD) {
1260  ics->predictor_present = get_bits1(gb);
1261  ics->predictor_reset_group = 0;
1262  }
1263  if (ics->predictor_present) {
1264  if (aot == AOT_AAC_MAIN) {
1265  if (decode_prediction(ac, ics, gb)) {
1266  goto fail;
1267  }
1268  } else if (aot == AOT_AAC_LC ||
1269  aot == AOT_ER_AAC_LC) {
1270  av_log(ac->avctx, AV_LOG_ERROR,
1271  "Prediction is not allowed in AAC-LC.\n");
1272  goto fail;
1273  } else {
1274  if (aot == AOT_ER_AAC_LD) {
1275  av_log(ac->avctx, AV_LOG_ERROR,
1276  "LTP in ER AAC LD not yet implemented.\n");
1277  return AVERROR_PATCHWELCOME;
1278  }
1279  if ((ics->ltp.present = get_bits(gb, 1)))
1280  decode_ltp(&ics->ltp, gb, ics->max_sfb);
1281  }
1282  }
1283  }
1284 
1285  if (ics->max_sfb > ics->num_swb) {
1286  av_log(ac->avctx, AV_LOG_ERROR,
1287  "Number of scalefactor bands in group (%d) "
1288  "exceeds limit (%d).\n",
1289  ics->max_sfb, ics->num_swb);
1290  goto fail;
1291  }
1292 
1293  return 0;
1294 fail:
1295  ics->max_sfb = 0;
1296  return AVERROR_INVALIDDATA;
1297 }
1298 
1299 /**
1300  * Decode band types (section_data payload); reference: table 4.46.
1301  *
1302  * @param band_type array of the used band type
1303  * @param band_type_run_end array of the last scalefactor band of a band type run
1304  *
1305  * @return Returns error status. 0 - OK, !0 - error
1306  */
1307 static int decode_band_types(AACContext *ac, enum BandType band_type[120],
1308  int band_type_run_end[120], GetBitContext *gb,
1310 {
1311  int g, idx = 0;
1312  const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
1313  for (g = 0; g < ics->num_window_groups; g++) {
1314  int k = 0;
1315  while (k < ics->max_sfb) {
1316  uint8_t sect_end = k;
1317  int sect_len_incr;
1318  int sect_band_type = get_bits(gb, 4);
1319  if (sect_band_type == 12) {
1320  av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
1321  return AVERROR_INVALIDDATA;
1322  }
1323  do {
1324  sect_len_incr = get_bits(gb, bits);
1325  sect_end += sect_len_incr;
1326  if (get_bits_left(gb) < 0) {
1327  av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
1328  return AVERROR_INVALIDDATA;
1329  }
1330  if (sect_end > ics->max_sfb) {
1331  av_log(ac->avctx, AV_LOG_ERROR,
1332  "Number of bands (%d) exceeds limit (%d).\n",
1333  sect_end, ics->max_sfb);
1334  return AVERROR_INVALIDDATA;
1335  }
1336  } while (sect_len_incr == (1 << bits) - 1);
1337  for (; k < sect_end; k++) {
1338  band_type [idx] = sect_band_type;
1339  band_type_run_end[idx++] = sect_end;
1340  }
1341  }
1342  }
1343  return 0;
1344 }
1345 
1346 /**
1347  * Decode scalefactors; reference: table 4.47.
1348  *
1349  * @param global_gain first scalefactor value as scalefactors are differentially coded
1350  * @param band_type array of the used band type
1351  * @param band_type_run_end array of the last scalefactor band of a band type run
1352  * @param sf array of scalefactors or intensity stereo positions
1353  *
1354  * @return Returns error status. 0 - OK, !0 - error
1355  */
1356 static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
1357  unsigned int global_gain,
1359  enum BandType band_type[120],
1360  int band_type_run_end[120])
1361 {
1362  int g, i, idx = 0;
1363  int offset[3] = { global_gain, global_gain - 90, 0 };
1364  int clipped_offset;
1365  int noise_flag = 1;
1366  for (g = 0; g < ics->num_window_groups; g++) {
1367  for (i = 0; i < ics->max_sfb;) {
1368  int run_end = band_type_run_end[idx];
1369  if (band_type[idx] == ZERO_BT) {
1370  for (; i < run_end; i++, idx++)
1371  sf[idx] = 0.0;
1372  } else if ((band_type[idx] == INTENSITY_BT) ||
1373  (band_type[idx] == INTENSITY_BT2)) {
1374  for (; i < run_end; i++, idx++) {
1375  offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1376  clipped_offset = av_clip(offset[2], -155, 100);
1377  if (offset[2] != clipped_offset) {
1379  "If you heard an audible artifact, there may be a bug in the decoder. "
1380  "Clipped intensity stereo position (%d -> %d)",
1381  offset[2], clipped_offset);
1382  }
1383  sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
1384  }
1385  } else if (band_type[idx] == NOISE_BT) {
1386  for (; i < run_end; i++, idx++) {
1387  if (noise_flag-- > 0)
1388  offset[1] += get_bits(gb, 9) - 256;
1389  else
1390  offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1391  clipped_offset = av_clip(offset[1], -100, 155);
1392  if (offset[1] != clipped_offset) {
1394  "If you heard an audible artifact, there may be a bug in the decoder. "
1395  "Clipped noise gain (%d -> %d)",
1396  offset[1], clipped_offset);
1397  }
1398  sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
1399  }
1400  } else {
1401  for (; i < run_end; i++, idx++) {
1402  offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1403  if (offset[0] > 255U) {
1404  av_log(ac->avctx, AV_LOG_ERROR,
1405  "Scalefactor (%d) out of range.\n", offset[0]);
1406  return AVERROR_INVALIDDATA;
1407  }
1408  sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
1409  }
1410  }
1411  }
1412  }
1413  return 0;
1414 }
1415 
1416 /**
1417  * Decode pulse data; reference: table 4.7.
1418  */
1419 static int decode_pulses(Pulse *pulse, GetBitContext *gb,
1420  const uint16_t *swb_offset, int num_swb)
1421 {
1422  int i, pulse_swb;
1423  pulse->num_pulse = get_bits(gb, 2) + 1;
1424  pulse_swb = get_bits(gb, 6);
1425  if (pulse_swb >= num_swb)
1426  return -1;
1427  pulse->pos[0] = swb_offset[pulse_swb];
1428  pulse->pos[0] += get_bits(gb, 5);
1429  if (pulse->pos[0] >= swb_offset[num_swb])
1430  return -1;
1431  pulse->amp[0] = get_bits(gb, 4);
1432  for (i = 1; i < pulse->num_pulse; i++) {
1433  pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
1434  if (pulse->pos[i] >= swb_offset[num_swb])
1435  return -1;
1436  pulse->amp[i] = get_bits(gb, 4);
1437  }
1438  return 0;
1439 }
1440 
1441 /**
1442  * Decode Temporal Noise Shaping data; reference: table 4.48.
1443  *
1444  * @return Returns error status. 0 - OK, !0 - error
1445  */
1447  GetBitContext *gb, const IndividualChannelStream *ics)
1448 {
1449  int w, filt, i, coef_len, coef_res, coef_compress;
1450  const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
1451  const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
1452  for (w = 0; w < ics->num_windows; w++) {
1453  if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
1454  coef_res = get_bits1(gb);
1455 
1456  for (filt = 0; filt < tns->n_filt[w]; filt++) {
1457  int tmp2_idx;
1458  tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
1459 
1460  if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
1461  av_log(ac->avctx, AV_LOG_ERROR,
1462  "TNS filter order %d is greater than maximum %d.\n",
1463  tns->order[w][filt], tns_max_order);
1464  tns->order[w][filt] = 0;
1465  return AVERROR_INVALIDDATA;
1466  }
1467  if (tns->order[w][filt]) {
1468  tns->direction[w][filt] = get_bits1(gb);
1469  coef_compress = get_bits1(gb);
1470  coef_len = coef_res + 3 - coef_compress;
1471  tmp2_idx = 2 * coef_compress + coef_res;
1472 
1473  for (i = 0; i < tns->order[w][filt]; i++)
1474  tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
1475  }
1476  }
1477  }
1478  }
1479  return 0;
1480 }
1481 
1482 /**
1483  * Decode Mid/Side data; reference: table 4.54.
1484  *
1485  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
1486  * [1] mask is decoded from bitstream; [2] mask is all 1s;
1487  * [3] reserved for scalable AAC
1488  */
1490  int ms_present)
1491 {
1492  int idx;
1493  if (ms_present == 1) {
1494  for (idx = 0;
1495  idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
1496  idx++)
1497  cpe->ms_mask[idx] = get_bits1(gb);
1498  } else if (ms_present == 2) {
1499  memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask[0]) * cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb);
1500  }
1501 }
1502 
1503 #ifndef VMUL2
1504 static inline float *VMUL2(float *dst, const float *v, unsigned idx,
1505  const float *scale)
1506 {
1507  float s = *scale;
1508  *dst++ = v[idx & 15] * s;
1509  *dst++ = v[idx>>4 & 15] * s;
1510  return dst;
1511 }
1512 #endif
1513 
1514 #ifndef VMUL4
1515 static inline float *VMUL4(float *dst, const float *v, unsigned idx,
1516  const float *scale)
1517 {
1518  float s = *scale;
1519  *dst++ = v[idx & 3] * s;
1520  *dst++ = v[idx>>2 & 3] * s;
1521  *dst++ = v[idx>>4 & 3] * s;
1522  *dst++ = v[idx>>6 & 3] * s;
1523  return dst;
1524 }
1525 #endif
1526 
1527 #ifndef VMUL2S
1528 static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
1529  unsigned sign, const float *scale)
1530 {
1531  union av_intfloat32 s0, s1;
1532 
1533  s0.f = s1.f = *scale;
1534  s0.i ^= sign >> 1 << 31;
1535  s1.i ^= sign << 31;
1536 
1537  *dst++ = v[idx & 15] * s0.f;
1538  *dst++ = v[idx>>4 & 15] * s1.f;
1539 
1540  return dst;
1541 }
1542 #endif
1543 
1544 #ifndef VMUL4S
1545 static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
1546  unsigned sign, const float *scale)
1547 {
1548  unsigned nz = idx >> 12;
1549  union av_intfloat32 s = { .f = *scale };
1550  union av_intfloat32 t;
1551 
1552  t.i = s.i ^ (sign & 1U<<31);
1553  *dst++ = v[idx & 3] * t.f;
1554 
1555  sign <<= nz & 1; nz >>= 1;
1556  t.i = s.i ^ (sign & 1U<<31);
1557  *dst++ = v[idx>>2 & 3] * t.f;
1558 
1559  sign <<= nz & 1; nz >>= 1;
1560  t.i = s.i ^ (sign & 1U<<31);
1561  *dst++ = v[idx>>4 & 3] * t.f;
1562 
1563  sign <<= nz & 1;
1564  t.i = s.i ^ (sign & 1U<<31);
1565  *dst++ = v[idx>>6 & 3] * t.f;
1566 
1567  return dst;
1568 }
1569 #endif
1570 
1571 /**
1572  * Decode spectral data; reference: table 4.50.
1573  * Dequantize and scale spectral data; reference: 4.6.3.3.
1574  *
1575  * @param coef array of dequantized, scaled spectral data
1576  * @param sf array of scalefactors or intensity stereo positions
1577  * @param pulse_present set if pulses are present
1578  * @param pulse pointer to pulse data struct
1579  * @param band_type array of the used band type
1580  *
1581  * @return Returns error status. 0 - OK, !0 - error
1582  */
1583 static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
1584  GetBitContext *gb, const float sf[120],
1585  int pulse_present, const Pulse *pulse,
1586  const IndividualChannelStream *ics,
1587  enum BandType band_type[120])
1588 {
1589  int i, k, g, idx = 0;
1590  const int c = 1024 / ics->num_windows;
1591  const uint16_t *offsets = ics->swb_offset;
1592  float *coef_base = coef;
1593 
1594  for (g = 0; g < ics->num_windows; g++)
1595  memset(coef + g * 128 + offsets[ics->max_sfb], 0,
1596  sizeof(float) * (c - offsets[ics->max_sfb]));
1597 
1598  for (g = 0; g < ics->num_window_groups; g++) {
1599  unsigned g_len = ics->group_len[g];
1600 
1601  for (i = 0; i < ics->max_sfb; i++, idx++) {
1602  const unsigned cbt_m1 = band_type[idx] - 1;
1603  float *cfo = coef + offsets[i];
1604  int off_len = offsets[i + 1] - offsets[i];
1605  int group;
1606 
1607  if (cbt_m1 >= INTENSITY_BT2 - 1) {
1608  for (group = 0; group < g_len; group++, cfo+=128) {
1609  memset(cfo, 0, off_len * sizeof(float));
1610  }
1611  } else if (cbt_m1 == NOISE_BT - 1) {
1612  for (group = 0; group < g_len; group++, cfo+=128) {
1613  float scale;
1614  float band_energy;
1615 
1616  for (k = 0; k < off_len; k++) {
1618  cfo[k] = ac->random_state;
1619  }
1620 
1621  band_energy = ac->fdsp.scalarproduct_float(cfo, cfo, off_len);
1622  scale = sf[idx] / sqrtf(band_energy);
1623  ac->fdsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
1624  }
1625  } else {
1626  const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1627  const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1628  VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1629  OPEN_READER(re, gb);
1630 
1631  switch (cbt_m1 >> 1) {
1632  case 0:
1633  for (group = 0; group < g_len; group++, cfo+=128) {
1634  float *cf = cfo;
1635  int len = off_len;
1636 
1637  do {
1638  int code;
1639  unsigned cb_idx;
1640 
1641  UPDATE_CACHE(re, gb);
1642  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1643  cb_idx = cb_vector_idx[code];
1644  cf = VMUL4(cf, vq, cb_idx, sf + idx);
1645  } while (len -= 4);
1646  }
1647  break;
1648 
1649  case 1:
1650  for (group = 0; group < g_len; group++, cfo+=128) {
1651  float *cf = cfo;
1652  int len = off_len;
1653 
1654  do {
1655  int code;
1656  unsigned nnz;
1657  unsigned cb_idx;
1658  uint32_t bits;
1659 
1660  UPDATE_CACHE(re, gb);
1661  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1662  cb_idx = cb_vector_idx[code];
1663  nnz = cb_idx >> 8 & 15;
1664  bits = nnz ? GET_CACHE(re, gb) : 0;
1665  LAST_SKIP_BITS(re, gb, nnz);
1666  cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1667  } while (len -= 4);
1668  }
1669  break;
1670 
1671  case 2:
1672  for (group = 0; group < g_len; group++, cfo+=128) {
1673  float *cf = cfo;
1674  int len = off_len;
1675 
1676  do {
1677  int code;
1678  unsigned cb_idx;
1679 
1680  UPDATE_CACHE(re, gb);
1681  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1682  cb_idx = cb_vector_idx[code];
1683  cf = VMUL2(cf, vq, cb_idx, sf + idx);
1684  } while (len -= 2);
1685  }
1686  break;
1687 
1688  case 3:
1689  case 4:
1690  for (group = 0; group < g_len; group++, cfo+=128) {
1691  float *cf = cfo;
1692  int len = off_len;
1693 
1694  do {
1695  int code;
1696  unsigned nnz;
1697  unsigned cb_idx;
1698  unsigned sign;
1699 
1700  UPDATE_CACHE(re, gb);
1701  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1702  cb_idx = cb_vector_idx[code];
1703  nnz = cb_idx >> 8 & 15;
1704  sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
1705  LAST_SKIP_BITS(re, gb, nnz);
1706  cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1707  } while (len -= 2);
1708  }
1709  break;
1710 
1711  default:
1712  for (group = 0; group < g_len; group++, cfo+=128) {
1713  float *cf = cfo;
1714  uint32_t *icf = (uint32_t *) cf;
1715  int len = off_len;
1716 
1717  do {
1718  int code;
1719  unsigned nzt, nnz;
1720  unsigned cb_idx;
1721  uint32_t bits;
1722  int j;
1723 
1724  UPDATE_CACHE(re, gb);
1725  GET_VLC(code, re, gb, vlc_tab, 8, 2);
1726 
1727  if (!code) {
1728  *icf++ = 0;
1729  *icf++ = 0;
1730  continue;
1731  }
1732 
1733  cb_idx = cb_vector_idx[code];
1734  nnz = cb_idx >> 12;
1735  nzt = cb_idx >> 8;
1736  bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1737  LAST_SKIP_BITS(re, gb, nnz);
1738 
1739  for (j = 0; j < 2; j++) {
1740  if (nzt & 1<<j) {
1741  uint32_t b;
1742  int n;
1743  /* The total length of escape_sequence must be < 22 bits according
1744  to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1745  UPDATE_CACHE(re, gb);
1746  b = GET_CACHE(re, gb);
1747  b = 31 - av_log2(~b);
1748 
1749  if (b > 8) {
1750  av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1751  return AVERROR_INVALIDDATA;
1752  }
1753 
1754  SKIP_BITS(re, gb, b + 1);
1755  b += 4;
1756  n = (1 << b) + SHOW_UBITS(re, gb, b);
1757  LAST_SKIP_BITS(re, gb, b);
1758  *icf++ = cbrt_tab[n] | (bits & 1U<<31);
1759  bits <<= 1;
1760  } else {
1761  unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1762  *icf++ = (bits & 1U<<31) | v;
1763  bits <<= !!v;
1764  }
1765  cb_idx >>= 4;
1766  }
1767  } while (len -= 2);
1768 
1769  ac->fdsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1770  }
1771  }
1772 
1773  CLOSE_READER(re, gb);
1774  }
1775  }
1776  coef += g_len << 7;
1777  }
1778 
1779  if (pulse_present) {
1780  idx = 0;
1781  for (i = 0; i < pulse->num_pulse; i++) {
1782  float co = coef_base[ pulse->pos[i] ];
1783  while (offsets[idx + 1] <= pulse->pos[i])
1784  idx++;
1785  if (band_type[idx] != NOISE_BT && sf[idx]) {
1786  float ico = -pulse->amp[i];
1787  if (co) {
1788  co /= sf[idx];
1789  ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1790  }
1791  coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1792  }
1793  }
1794  }
1795  return 0;
1796 }
1797 
1798 static av_always_inline float flt16_round(float pf)
1799 {
1800  union av_intfloat32 tmp;
1801  tmp.f = pf;
1802  tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
1803  return tmp.f;
1804 }
1805 
1806 static av_always_inline float flt16_even(float pf)
1807 {
1808  union av_intfloat32 tmp;
1809  tmp.f = pf;
1810  tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
1811  return tmp.f;
1812 }
1813 
1814 static av_always_inline float flt16_trunc(float pf)
1815 {
1816  union av_intfloat32 pun;
1817  pun.f = pf;
1818  pun.i &= 0xFFFF0000U;
1819  return pun.f;
1820 }
1821 
1822 static av_always_inline void predict(PredictorState *ps, float *coef,
1823  int output_enable)
1824 {
1825  const float a = 0.953125; // 61.0 / 64
1826  const float alpha = 0.90625; // 29.0 / 32
1827  float e0, e1;
1828  float pv;
1829  float k1, k2;
1830  float r0 = ps->r0, r1 = ps->r1;
1831  float cor0 = ps->cor0, cor1 = ps->cor1;
1832  float var0 = ps->var0, var1 = ps->var1;
1833 
1834  k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
1835  k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
1836 
1837  pv = flt16_round(k1 * r0 + k2 * r1);
1838  if (output_enable)
1839  *coef += pv;
1840 
1841  e0 = *coef;
1842  e1 = e0 - k1 * r0;
1843 
1844  ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
1845  ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
1846  ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
1847  ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
1848 
1849  ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
1850  ps->r0 = flt16_trunc(a * e0);
1851 }
1852 
1853 /**
1854  * Apply AAC-Main style frequency domain prediction.
1855  */
1857 {
1858  int sfb, k;
1859 
1860  if (!sce->ics.predictor_initialized) {
1862  sce->ics.predictor_initialized = 1;
1863  }
1864 
1865  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1866  for (sfb = 0;
1867  sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
1868  sfb++) {
1869  for (k = sce->ics.swb_offset[sfb];
1870  k < sce->ics.swb_offset[sfb + 1];
1871  k++) {
1872  predict(&sce->predictor_state[k], &sce->coeffs[k],
1873  sce->ics.predictor_present &&
1874  sce->ics.prediction_used[sfb]);
1875  }
1876  }
1877  if (sce->ics.predictor_reset_group)
1879  sce->ics.predictor_reset_group);
1880  } else
1882 }
1883 
1884 /**
1885  * Decode an individual_channel_stream payload; reference: table 4.44.
1886  *
1887  * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
1888  * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1889  *
1890  * @return Returns error status. 0 - OK, !0 - error
1891  */
1893  GetBitContext *gb, int common_window, int scale_flag)
1894 {
1895  Pulse pulse;
1896  TemporalNoiseShaping *tns = &sce->tns;
1897  IndividualChannelStream *ics = &sce->ics;
1898  float *out = sce->coeffs;
1899  int global_gain, eld_syntax, er_syntax, pulse_present = 0;
1900  int ret;
1901 
1902  eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1903  er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
1904  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
1905  ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
1906  ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1907 
1908  /* This assignment is to silence a GCC warning about the variable being used
1909  * uninitialized when in fact it always is.
1910  */
1911  pulse.num_pulse = 0;
1912 
1913  global_gain = get_bits(gb, 8);
1914 
1915  if (!common_window && !scale_flag) {
1916  if (decode_ics_info(ac, ics, gb) < 0)
1917  return AVERROR_INVALIDDATA;
1918  }
1919 
1920  if ((ret = decode_band_types(ac, sce->band_type,
1921  sce->band_type_run_end, gb, ics)) < 0)
1922  return ret;
1923  if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
1924  sce->band_type, sce->band_type_run_end)) < 0)
1925  return ret;
1926 
1927  pulse_present = 0;
1928  if (!scale_flag) {
1929  if (!eld_syntax && (pulse_present = get_bits1(gb))) {
1930  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1931  av_log(ac->avctx, AV_LOG_ERROR,
1932  "Pulse tool not allowed in eight short sequence.\n");
1933  return AVERROR_INVALIDDATA;
1934  }
1935  if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1936  av_log(ac->avctx, AV_LOG_ERROR,
1937  "Pulse data corrupt or invalid.\n");
1938  return AVERROR_INVALIDDATA;
1939  }
1940  }
1941  tns->present = get_bits1(gb);
1942  if (tns->present && !er_syntax)
1943  if (decode_tns(ac, tns, gb, ics) < 0)
1944  return AVERROR_INVALIDDATA;
1945  if (!eld_syntax && get_bits1(gb)) {
1946  avpriv_request_sample(ac->avctx, "SSR");
1947  return AVERROR_PATCHWELCOME;
1948  }
1949  // I see no textual basis in the spec for this occurring after SSR gain
1950  // control, but this is what both reference and real implmentations do
1951  if (tns->present && er_syntax)
1952  if (decode_tns(ac, tns, gb, ics) < 0)
1953  return AVERROR_INVALIDDATA;
1954  }
1955 
1956  if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
1957  &pulse, ics, sce->band_type) < 0)
1958  return AVERROR_INVALIDDATA;
1959 
1960  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
1961  apply_prediction(ac, sce);
1962 
1963  return 0;
1964 }
1965 
1966 /**
1967  * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1968  */
1970 {
1971  const IndividualChannelStream *ics = &cpe->ch[0].ics;
1972  float *ch0 = cpe->ch[0].coeffs;
1973  float *ch1 = cpe->ch[1].coeffs;
1974  int g, i, group, idx = 0;
1975  const uint16_t *offsets = ics->swb_offset;
1976  for (g = 0; g < ics->num_window_groups; g++) {
1977  for (i = 0; i < ics->max_sfb; i++, idx++) {
1978  if (cpe->ms_mask[idx] &&
1979  cpe->ch[0].band_type[idx] < NOISE_BT &&
1980  cpe->ch[1].band_type[idx] < NOISE_BT) {
1981  for (group = 0; group < ics->group_len[g]; group++) {
1982  ac->fdsp.butterflies_float(ch0 + group * 128 + offsets[i],
1983  ch1 + group * 128 + offsets[i],
1984  offsets[i+1] - offsets[i]);
1985  }
1986  }
1987  }
1988  ch0 += ics->group_len[g] * 128;
1989  ch1 += ics->group_len[g] * 128;
1990  }
1991 }
1992 
1993 /**
1994  * intensity stereo decoding; reference: 4.6.8.2.3
1995  *
1996  * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
1997  * [1] mask is decoded from bitstream; [2] mask is all 1s;
1998  * [3] reserved for scalable AAC
1999  */
2001  ChannelElement *cpe, int ms_present)
2002 {
2003  const IndividualChannelStream *ics = &cpe->ch[1].ics;
2004  SingleChannelElement *sce1 = &cpe->ch[1];
2005  float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
2006  const uint16_t *offsets = ics->swb_offset;
2007  int g, group, i, idx = 0;
2008  int c;
2009  float scale;
2010  for (g = 0; g < ics->num_window_groups; g++) {
2011  for (i = 0; i < ics->max_sfb;) {
2012  if (sce1->band_type[idx] == INTENSITY_BT ||
2013  sce1->band_type[idx] == INTENSITY_BT2) {
2014  const int bt_run_end = sce1->band_type_run_end[idx];
2015  for (; i < bt_run_end; i++, idx++) {
2016  c = -1 + 2 * (sce1->band_type[idx] - 14);
2017  if (ms_present)
2018  c *= 1 - 2 * cpe->ms_mask[idx];
2019  scale = c * sce1->sf[idx];
2020  for (group = 0; group < ics->group_len[g]; group++)
2021  ac->fdsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
2022  coef0 + group * 128 + offsets[i],
2023  scale,
2024  offsets[i + 1] - offsets[i]);
2025  }
2026  } else {
2027  int bt_run_end = sce1->band_type_run_end[idx];
2028  idx += bt_run_end - i;
2029  i = bt_run_end;
2030  }
2031  }
2032  coef0 += ics->group_len[g] * 128;
2033  coef1 += ics->group_len[g] * 128;
2034  }
2035 }
2036 
2037 /**
2038  * Decode a channel_pair_element; reference: table 4.4.
2039  *
2040  * @return Returns error status. 0 - OK, !0 - error
2041  */
2043 {
2044  int i, ret, common_window, ms_present = 0;
2045  int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2046 
2047  common_window = eld_syntax || get_bits1(gb);
2048  if (common_window) {
2049  if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
2050  return AVERROR_INVALIDDATA;
2051  i = cpe->ch[1].ics.use_kb_window[0];
2052  cpe->ch[1].ics = cpe->ch[0].ics;
2053  cpe->ch[1].ics.use_kb_window[1] = i;
2054  if (cpe->ch[1].ics.predictor_present &&
2055  (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
2056  if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
2057  decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
2058  ms_present = get_bits(gb, 2);
2059  if (ms_present == 3) {
2060  av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
2061  return AVERROR_INVALIDDATA;
2062  } else if (ms_present)
2063  decode_mid_side_stereo(cpe, gb, ms_present);
2064  }
2065  if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
2066  return ret;
2067  if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
2068  return ret;
2069 
2070  if (common_window) {
2071  if (ms_present)
2072  apply_mid_side_stereo(ac, cpe);
2073  if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
2074  apply_prediction(ac, &cpe->ch[0]);
2075  apply_prediction(ac, &cpe->ch[1]);
2076  }
2077  }
2078 
2079  apply_intensity_stereo(ac, cpe, ms_present);
2080  return 0;
2081 }
2082 
2083 static const float cce_scale[] = {
2084  1.09050773266525765921, //2^(1/8)
2085  1.18920711500272106672, //2^(1/4)
2086  M_SQRT2,
2087  2,
2088 };
2089 
2090 /**
2091  * Decode coupling_channel_element; reference: table 4.8.
2092  *
2093  * @return Returns error status. 0 - OK, !0 - error
2094  */
2096 {
2097  int num_gain = 0;
2098  int c, g, sfb, ret;
2099  int sign;
2100  float scale;
2101  SingleChannelElement *sce = &che->ch[0];
2102  ChannelCoupling *coup = &che->coup;
2103 
2104  coup->coupling_point = 2 * get_bits1(gb);
2105  coup->num_coupled = get_bits(gb, 3);
2106  for (c = 0; c <= coup->num_coupled; c++) {
2107  num_gain++;
2108  coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
2109  coup->id_select[c] = get_bits(gb, 4);
2110  if (coup->type[c] == TYPE_CPE) {
2111  coup->ch_select[c] = get_bits(gb, 2);
2112  if (coup->ch_select[c] == 3)
2113  num_gain++;
2114  } else
2115  coup->ch_select[c] = 2;
2116  }
2117  coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
2118 
2119  sign = get_bits(gb, 1);
2120  scale = cce_scale[get_bits(gb, 2)];
2121 
2122  if ((ret = decode_ics(ac, sce, gb, 0, 0)))
2123  return ret;
2124 
2125  for (c = 0; c < num_gain; c++) {
2126  int idx = 0;
2127  int cge = 1;
2128  int gain = 0;
2129  float gain_cache = 1.0;
2130  if (c) {
2131  cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
2132  gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
2133  gain_cache = powf(scale, -gain);
2134  }
2135  if (coup->coupling_point == AFTER_IMDCT) {
2136  coup->gain[c][0] = gain_cache;
2137  } else {
2138  for (g = 0; g < sce->ics.num_window_groups; g++) {
2139  for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
2140  if (sce->band_type[idx] != ZERO_BT) {
2141  if (!cge) {
2142  int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
2143  if (t) {
2144  int s = 1;
2145  t = gain += t;
2146  if (sign) {
2147  s -= 2 * (t & 0x1);
2148  t >>= 1;
2149  }
2150  gain_cache = powf(scale, -t) * s;
2151  }
2152  }
2153  coup->gain[c][idx] = gain_cache;
2154  }
2155  }
2156  }
2157  }
2158  }
2159  return 0;
2160 }
2161 
2162 /**
2163  * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
2164  *
2165  * @return Returns number of bytes consumed.
2166  */
2168  GetBitContext *gb)
2169 {
2170  int i;
2171  int num_excl_chan = 0;
2172 
2173  do {
2174  for (i = 0; i < 7; i++)
2175  che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
2176  } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
2177 
2178  return num_excl_chan / 7;
2179 }
2180 
2181 /**
2182  * Decode dynamic range information; reference: table 4.52.
2183  *
2184  * @return Returns number of bytes consumed.
2185  */
2187  GetBitContext *gb)
2188 {
2189  int n = 1;
2190  int drc_num_bands = 1;
2191  int i;
2192 
2193  /* pce_tag_present? */
2194  if (get_bits1(gb)) {
2195  che_drc->pce_instance_tag = get_bits(gb, 4);
2196  skip_bits(gb, 4); // tag_reserved_bits
2197  n++;
2198  }
2199 
2200  /* excluded_chns_present? */
2201  if (get_bits1(gb)) {
2202  n += decode_drc_channel_exclusions(che_drc, gb);
2203  }
2204 
2205  /* drc_bands_present? */
2206  if (get_bits1(gb)) {
2207  che_drc->band_incr = get_bits(gb, 4);
2208  che_drc->interpolation_scheme = get_bits(gb, 4);
2209  n++;
2210  drc_num_bands += che_drc->band_incr;
2211  for (i = 0; i < drc_num_bands; i++) {
2212  che_drc->band_top[i] = get_bits(gb, 8);
2213  n++;
2214  }
2215  }
2216 
2217  /* prog_ref_level_present? */
2218  if (get_bits1(gb)) {
2219  che_drc->prog_ref_level = get_bits(gb, 7);
2220  skip_bits1(gb); // prog_ref_level_reserved_bits
2221  n++;
2222  }
2223 
2224  for (i = 0; i < drc_num_bands; i++) {
2225  che_drc->dyn_rng_sgn[i] = get_bits1(gb);
2226  che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
2227  n++;
2228  }
2229 
2230  return n;
2231 }
2232 
2233 static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
2234  uint8_t buf[256];
2235  int i, major, minor;
2236 
2237  if (len < 13+7*8)
2238  goto unknown;
2239 
2240  get_bits(gb, 13); len -= 13;
2241 
2242  for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
2243  buf[i] = get_bits(gb, 8);
2244 
2245  buf[i] = 0;
2246  if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
2247  av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
2248 
2249  if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
2250  ac->avctx->internal->skip_samples = 1024;
2251  }
2252 
2253 unknown:
2254  skip_bits_long(gb, len);
2255 
2256  return 0;
2257 }
2258 
2259 /**
2260  * Decode extension data (incomplete); reference: table 4.51.
2261  *
2262  * @param cnt length of TYPE_FIL syntactic element in bytes
2263  *
2264  * @return Returns number of bytes consumed
2265  */
2267  ChannelElement *che, enum RawDataBlockType elem_type)
2268 {
2269  int crc_flag = 0;
2270  int res = cnt;
2271  switch (get_bits(gb, 4)) { // extension type
2272  case EXT_SBR_DATA_CRC:
2273  crc_flag++;
2274  case EXT_SBR_DATA:
2275  if (!che) {
2276  av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
2277  return res;
2278  } else if (!ac->oc[1].m4ac.sbr) {
2279  av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
2280  skip_bits_long(gb, 8 * cnt - 4);
2281  return res;
2282  } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
2283  av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
2284  skip_bits_long(gb, 8 * cnt - 4);
2285  return res;
2286  } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
2287  ac->oc[1].m4ac.sbr = 1;
2288  ac->oc[1].m4ac.ps = 1;
2290  output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
2291  ac->oc[1].status, 1);
2292  } else {
2293  ac->oc[1].m4ac.sbr = 1;
2295  }
2296  res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
2297  break;
2298  case EXT_DYNAMIC_RANGE:
2299  res = decode_dynamic_range(&ac->che_drc, gb);
2300  break;
2301  case EXT_FILL:
2302  decode_fill(ac, gb, 8 * cnt - 4);
2303  break;
2304  case EXT_FILL_DATA:
2305  case EXT_DATA_ELEMENT:
2306  default:
2307  skip_bits_long(gb, 8 * cnt - 4);
2308  break;
2309  };
2310  return res;
2311 }
2312 
2313 /**
2314  * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
2315  *
2316  * @param decode 1 if tool is used normally, 0 if tool is used in LTP.
2317  * @param coef spectral coefficients
2318  */
2319 static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
2320  IndividualChannelStream *ics, int decode)
2321 {
2322  const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
2323  int w, filt, m, i;
2324  int bottom, top, order, start, end, size, inc;
2325  float lpc[TNS_MAX_ORDER];
2326  float tmp[TNS_MAX_ORDER+1];
2327 
2328  for (w = 0; w < ics->num_windows; w++) {
2329  bottom = ics->num_swb;
2330  for (filt = 0; filt < tns->n_filt[w]; filt++) {
2331  top = bottom;
2332  bottom = FFMAX(0, top - tns->length[w][filt]);
2333  order = tns->order[w][filt];
2334  if (order == 0)
2335  continue;
2336 
2337  // tns_decode_coef
2338  compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
2339 
2340  start = ics->swb_offset[FFMIN(bottom, mmm)];
2341  end = ics->swb_offset[FFMIN( top, mmm)];
2342  if ((size = end - start) <= 0)
2343  continue;
2344  if (tns->direction[w][filt]) {
2345  inc = -1;
2346  start = end - 1;
2347  } else {
2348  inc = 1;
2349  }
2350  start += w * 128;
2351 
2352  if (decode) {
2353  // ar filter
2354  for (m = 0; m < size; m++, start += inc)
2355  for (i = 1; i <= FFMIN(m, order); i++)
2356  coef[start] -= coef[start - i * inc] * lpc[i - 1];
2357  } else {
2358  // ma filter
2359  for (m = 0; m < size; m++, start += inc) {
2360  tmp[0] = coef[start];
2361  for (i = 1; i <= FFMIN(m, order); i++)
2362  coef[start] += tmp[i] * lpc[i - 1];
2363  for (i = order; i > 0; i--)
2364  tmp[i] = tmp[i - 1];
2365  }
2366  }
2367  }
2368  }
2369 }
2370 
2371 /**
2372  * Apply windowing and MDCT to obtain the spectral
2373  * coefficient from the predicted sample by LTP.
2374  */
2375 static void windowing_and_mdct_ltp(AACContext *ac, float *out,
2376  float *in, IndividualChannelStream *ics)
2377 {
2378  const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
2379  const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
2380  const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
2381  const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
2382 
2383  if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
2384  ac->fdsp.vector_fmul(in, in, lwindow_prev, 1024);
2385  } else {
2386  memset(in, 0, 448 * sizeof(float));
2387  ac->fdsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
2388  }
2389  if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
2390  ac->fdsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
2391  } else {
2392  ac->fdsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
2393  memset(in + 1024 + 576, 0, 448 * sizeof(float));
2394  }
2395  ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
2396 }
2397 
2398 /**
2399  * Apply the long term prediction
2400  */
2402 {
2403  const LongTermPrediction *ltp = &sce->ics.ltp;
2404  const uint16_t *offsets = sce->ics.swb_offset;
2405  int i, sfb;
2406 
2407  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
2408  float *predTime = sce->ret;
2409  float *predFreq = ac->buf_mdct;
2410  int16_t num_samples = 2048;
2411 
2412  if (ltp->lag < 1024)
2413  num_samples = ltp->lag + 1024;
2414  for (i = 0; i < num_samples; i++)
2415  predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
2416  memset(&predTime[i], 0, (2048 - i) * sizeof(float));
2417 
2418  ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
2419 
2420  if (sce->tns.present)
2421  ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
2422 
2423  for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
2424  if (ltp->used[sfb])
2425  for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
2426  sce->coeffs[i] += predFreq[i];
2427  }
2428 }
2429 
2430 /**
2431  * Update the LTP buffer for next frame
2432  */
2434 {
2435  IndividualChannelStream *ics = &sce->ics;
2436  float *saved = sce->saved;
2437  float *saved_ltp = sce->coeffs;
2438  const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
2439  const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
2440  int i;
2441 
2442  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2443  memcpy(saved_ltp, saved, 512 * sizeof(float));
2444  memset(saved_ltp + 576, 0, 448 * sizeof(float));
2445  ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2446  for (i = 0; i < 64; i++)
2447  saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
2448  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2449  memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(float));
2450  memset(saved_ltp + 576, 0, 448 * sizeof(float));
2451  ac->fdsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2452  for (i = 0; i < 64; i++)
2453  saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
2454  } else { // LONG_STOP or ONLY_LONG
2455  ac->fdsp.vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
2456  for (i = 0; i < 512; i++)
2457  saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
2458  }
2459 
2460  memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
2461  memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
2462  memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
2463 }
2464 
2465 /**
2466  * Conduct IMDCT and windowing.
2467  */
2469 {
2470  IndividualChannelStream *ics = &sce->ics;
2471  float *in = sce->coeffs;
2472  float *out = sce->ret;
2473  float *saved = sce->saved;
2474  const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
2475  const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
2476  const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
2477  float *buf = ac->buf_mdct;
2478  float *temp = ac->temp;
2479  int i;
2480 
2481  // imdct
2482  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2483  for (i = 0; i < 1024; i += 128)
2484  ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
2485  } else
2486  ac->mdct.imdct_half(&ac->mdct, buf, in);
2487 
2488  /* window overlapping
2489  * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2490  * and long to short transitions are considered to be short to short
2491  * transitions. This leaves just two cases (long to long and short to short)
2492  * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2493  */
2494  if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2496  ac->fdsp.vector_fmul_window( out, saved, buf, lwindow_prev, 512);
2497  } else {
2498  memcpy( out, saved, 448 * sizeof(float));
2499 
2500  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2501  ac->fdsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
2502  ac->fdsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
2503  ac->fdsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
2504  ac->fdsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
2505  ac->fdsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
2506  memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
2507  } else {
2508  ac->fdsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
2509  memcpy( out + 576, buf + 64, 448 * sizeof(float));
2510  }
2511  }
2512 
2513  // buffer update
2514  if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2515  memcpy( saved, temp + 64, 64 * sizeof(float));
2516  ac->fdsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
2517  ac->fdsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
2518  ac->fdsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
2519  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
2520  } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2521  memcpy( saved, buf + 512, 448 * sizeof(float));
2522  memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
2523  } else { // LONG_STOP or ONLY_LONG
2524  memcpy( saved, buf + 512, 512 * sizeof(float));
2525  }
2526 }
2527 
2529 {
2530  IndividualChannelStream *ics = &sce->ics;
2531  float *in = sce->coeffs;
2532  float *out = sce->ret;
2533  float *saved = sce->saved;
2534  float *buf = ac->buf_mdct;
2535 
2536  // imdct
2537  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2538 
2539  // window overlapping
2540  if (ics->use_kb_window[1]) {
2541  // AAC LD uses a low overlap sine window instead of a KBD window
2542  memcpy(out, saved, 192 * sizeof(float));
2543  ac->fdsp.vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
2544  memcpy( out + 320, buf + 64, 192 * sizeof(float));
2545  } else {
2546  ac->fdsp.vector_fmul_window(out, saved, buf, ff_sine_512, 256);
2547  }
2548 
2549  // buffer update
2550  memcpy(saved, buf + 256, 256 * sizeof(float));
2551 }
2552 
2554 {
2555  float *in = sce->coeffs;
2556  float *out = sce->ret;
2557  float *saved = sce->saved;
2558  const float *const window = ff_aac_eld_window;
2559  float *buf = ac->buf_mdct;
2560  int i;
2561  const int n = 512;
2562  const int n2 = n >> 1;
2563  const int n4 = n >> 2;
2564 
2565  // Inverse transform, mapped to the conventional IMDCT by
2566  // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
2567  // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
2568  // International Conference on Audio, Language and Image Processing, ICALIP 2008.
2569  // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
2570  for (i = 0; i < n2; i+=2) {
2571  float temp;
2572  temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
2573  temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
2574  }
2575  ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2576  for (i = 0; i < n; i+=2) {
2577  buf[i] = -buf[i];
2578  }
2579  // Like with the regular IMDCT at this point we still have the middle half
2580  // of a transform but with even symmetry on the left and odd symmetry on
2581  // the right
2582 
2583  // window overlapping
2584  // The spec says to use samples [0..511] but the reference decoder uses
2585  // samples [128..639].
2586  for (i = n4; i < n2; i ++) {
2587  out[i - n4] = buf[n2 - 1 - i] * window[i - n4] +
2588  saved[ i + n2] * window[i + n - n4] +
2589  -saved[ n + n2 - 1 - i] * window[i + 2*n - n4] +
2590  -saved[2*n + n2 + i] * window[i + 3*n - n4];
2591  }
2592  for (i = 0; i < n2; i ++) {
2593  out[n4 + i] = buf[i] * window[i + n2 - n4] +
2594  -saved[ n - 1 - i] * window[i + n2 + n - n4] +
2595  -saved[ n + i] * window[i + n2 + 2*n - n4] +
2596  saved[2*n + n - 1 - i] * window[i + n2 + 3*n - n4];
2597  }
2598  for (i = 0; i < n4; i ++) {
2599  out[n2 + n4 + i] = buf[ i + n2] * window[i + n - n4] +
2600  -saved[ n2 - 1 - i] * window[i + 2*n - n4] +
2601  -saved[ n + n2 + i] * window[i + 3*n - n4];
2602  }
2603 
2604  // buffer update
2605  memmove(saved + n, saved, 2 * n * sizeof(float));
2606  memcpy( saved, buf, n * sizeof(float));
2607 }
2608 
2609 /**
2610  * Apply dependent channel coupling (applied before IMDCT).
2611  *
2612  * @param index index into coupling gain array
2613  */
2615  SingleChannelElement *target,
2616  ChannelElement *cce, int index)
2617 {
2618  IndividualChannelStream *ics = &cce->ch[0].ics;
2619  const uint16_t *offsets = ics->swb_offset;
2620  float *dest = target->coeffs;
2621  const float *src = cce->ch[0].coeffs;
2622  int g, i, group, k, idx = 0;
2623  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2624  av_log(ac->avctx, AV_LOG_ERROR,
2625  "Dependent coupling is not supported together with LTP\n");
2626  return;
2627  }
2628  for (g = 0; g < ics->num_window_groups; g++) {
2629  for (i = 0; i < ics->max_sfb; i++, idx++) {
2630  if (cce->ch[0].band_type[idx] != ZERO_BT) {
2631  const float gain = cce->coup.gain[index][idx];
2632  for (group = 0; group < ics->group_len[g]; group++) {
2633  for (k = offsets[i]; k < offsets[i + 1]; k++) {
2634  // FIXME: SIMDify
2635  dest[group * 128 + k] += gain * src[group * 128 + k];
2636  }
2637  }
2638  }
2639  }
2640  dest += ics->group_len[g] * 128;
2641  src += ics->group_len[g] * 128;
2642  }
2643 }
2644 
2645 /**
2646  * Apply independent channel coupling (applied after IMDCT).
2647  *
2648  * @param index index into coupling gain array
2649  */
2651  SingleChannelElement *target,
2652  ChannelElement *cce, int index)
2653 {
2654  int i;
2655  const float gain = cce->coup.gain[index][0];
2656  const float *src = cce->ch[0].ret;
2657  float *dest = target->ret;
2658  const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
2659 
2660  for (i = 0; i < len; i++)
2661  dest[i] += gain * src[i];
2662 }
2663 
2664 /**
2665  * channel coupling transformation interface
2666  *
2667  * @param apply_coupling_method pointer to (in)dependent coupling function
2668  */
2670  enum RawDataBlockType type, int elem_id,
2671  enum CouplingPoint coupling_point,
2672  void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
2673 {
2674  int i, c;
2675 
2676  for (i = 0; i < MAX_ELEM_ID; i++) {
2677  ChannelElement *cce = ac->che[TYPE_CCE][i];
2678  int index = 0;
2679 
2680  if (cce && cce->coup.coupling_point == coupling_point) {
2681  ChannelCoupling *coup = &cce->coup;
2682 
2683  for (c = 0; c <= coup->num_coupled; c++) {
2684  if (coup->type[c] == type && coup->id_select[c] == elem_id) {
2685  if (coup->ch_select[c] != 1) {
2686  apply_coupling_method(ac, &cc->ch[0], cce, index);
2687  if (coup->ch_select[c] != 0)
2688  index++;
2689  }
2690  if (coup->ch_select[c] != 2)
2691  apply_coupling_method(ac, &cc->ch[1], cce, index++);
2692  } else
2693  index += 1 + (coup->ch_select[c] == 3);
2694  }
2695  }
2696  }
2697 }
2698 
2699 /**
2700  * Convert spectral data to float samples, applying all supported tools as appropriate.
2701  */
2703 {
2704  int i, type;
2706  switch (ac->oc[1].m4ac.object_type) {
2707  case AOT_ER_AAC_LD:
2709  break;
2710  case AOT_ER_AAC_ELD:
2712  break;
2713  default:
2715  }
2716  for (type = 3; type >= 0; type--) {
2717  for (i = 0; i < MAX_ELEM_ID; i++) {
2718  ChannelElement *che = ac->che[type][i];
2719  if (che) {
2720  if (type <= TYPE_CPE)
2722  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2723  if (che->ch[0].ics.predictor_present) {
2724  if (che->ch[0].ics.ltp.present)
2725  ac->apply_ltp(ac, &che->ch[0]);
2726  if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
2727  ac->apply_ltp(ac, &che->ch[1]);
2728  }
2729  }
2730  if (che->ch[0].tns.present)
2731  ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2732  if (che->ch[1].tns.present)
2733  ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2734  if (type <= TYPE_CPE)
2736  if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2737  imdct_and_window(ac, &che->ch[0]);
2738  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2739  ac->update_ltp(ac, &che->ch[0]);
2740  if (type == TYPE_CPE) {
2741  imdct_and_window(ac, &che->ch[1]);
2742  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2743  ac->update_ltp(ac, &che->ch[1]);
2744  }
2745  if (ac->oc[1].m4ac.sbr > 0) {
2746  ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2747  }
2748  }
2749  if (type <= TYPE_CCE)
2751  }
2752  }
2753  }
2754 }
2755 
2757 {
2758  int size;
2759  AACADTSHeaderInfo hdr_info;
2760  uint8_t layout_map[MAX_ELEM_ID*4][3];
2761  int layout_map_tags, ret;
2762 
2763  size = avpriv_aac_parse_header(gb, &hdr_info);
2764  if (size > 0) {
2765  if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
2766  // This is 2 for "VLB " audio in NSV files.
2767  // See samples/nsv/vlb_audio.
2769  "More than one AAC RDB per ADTS frame");
2770  ac->warned_num_aac_frames = 1;
2771  }
2773  if (hdr_info.chan_config) {
2774  ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
2775  if ((ret = set_default_channel_config(ac->avctx,
2776  layout_map,
2777  &layout_map_tags,
2778  hdr_info.chan_config)) < 0)
2779  return ret;
2780  if ((ret = output_configure(ac, layout_map, layout_map_tags,
2781  FFMAX(ac->oc[1].status,
2782  OC_TRIAL_FRAME), 0)) < 0)
2783  return ret;
2784  } else {
2785  ac->oc[1].m4ac.chan_config = 0;
2786  /**
2787  * dual mono frames in Japanese DTV can have chan_config 0
2788  * WITHOUT specifying PCE.
2789  * thus, set dual mono as default.
2790  */
2791  if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
2792  layout_map_tags = 2;
2793  layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
2794  layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
2795  layout_map[0][1] = 0;
2796  layout_map[1][1] = 1;
2797  if (output_configure(ac, layout_map, layout_map_tags,
2798  OC_TRIAL_FRAME, 0))
2799  return -7;
2800  }
2801  }
2802  ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
2803  ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
2804  ac->oc[1].m4ac.object_type = hdr_info.object_type;
2805  if (ac->oc[0].status != OC_LOCKED ||
2806  ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
2807  ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
2808  ac->oc[1].m4ac.sbr = -1;
2809  ac->oc[1].m4ac.ps = -1;
2810  }
2811  if (!hdr_info.crc_absent)
2812  skip_bits(gb, 16);
2813  }
2814  return size;
2815 }
2816 
2817 static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
2818  int *got_frame_ptr, GetBitContext *gb)
2819 {
2820  AACContext *ac = avctx->priv_data;
2821  ChannelElement *che;
2822  int err, i;
2823  int samples = 1024;
2824  int chan_config = ac->oc[1].m4ac.chan_config;
2825  int aot = ac->oc[1].m4ac.object_type;
2826 
2827  if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
2828  samples >>= 1;
2829 
2830  ac->frame = data;
2831 
2832  if ((err = frame_configure_elements(avctx)) < 0)
2833  return err;
2834 
2835  // The FF_PROFILE_AAC_* defines are all object_type - 1
2836  // This may lead to an undefined profile being signaled
2837  ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
2838 
2839  ac->tags_mapped = 0;
2840 
2841  if (chan_config < 0 || chan_config >= 8) {
2842  avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
2843  ac->oc[1].m4ac.chan_config);
2844  return AVERROR_INVALIDDATA;
2845  }
2846  for (i = 0; i < tags_per_config[chan_config]; i++) {
2847  const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
2848  const int elem_id = aac_channel_layout_map[chan_config-1][i][1];
2849  if (!(che=get_che(ac, elem_type, elem_id))) {
2850  av_log(ac->avctx, AV_LOG_ERROR,
2851  "channel element %d.%d is not allocated\n",
2852  elem_type, elem_id);
2853  return AVERROR_INVALIDDATA;
2854  }
2855  if (aot != AOT_ER_AAC_ELD)
2856  skip_bits(gb, 4);
2857  switch (elem_type) {
2858  case TYPE_SCE:
2859  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2860  break;
2861  case TYPE_CPE:
2862  err = decode_cpe(ac, gb, che);
2863  break;
2864  case TYPE_LFE:
2865  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2866  break;
2867  }
2868  if (err < 0)
2869  return err;
2870  }
2871 
2872  spectral_to_sample(ac);
2873 
2874  ac->frame->nb_samples = samples;
2875  ac->frame->sample_rate = avctx->sample_rate;
2876  *got_frame_ptr = 1;
2877 
2878  skip_bits_long(gb, get_bits_left(gb));
2879  return 0;
2880 }
2881 
2882 static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
2883  int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
2884 {
2885  AACContext *ac = avctx->priv_data;
2886  ChannelElement *che = NULL, *che_prev = NULL;
2887  enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
2888  int err, elem_id;
2889  int samples = 0, multiplier, audio_found = 0, pce_found = 0;
2890  int is_dmono, sce_count = 0;
2891 
2892  ac->frame = data;
2893 
2894  if (show_bits(gb, 12) == 0xfff) {
2895  if ((err = parse_adts_frame_header(ac, gb)) < 0) {
2896  av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
2897  goto fail;
2898  }
2899  if (ac->oc[1].m4ac.sampling_index > 12) {
2900  av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
2901  err = AVERROR_INVALIDDATA;
2902  goto fail;
2903  }
2904  }
2905 
2906  if ((err = frame_configure_elements(avctx)) < 0)
2907  goto fail;
2908 
2909  // The FF_PROFILE_AAC_* defines are all object_type - 1
2910  // This may lead to an undefined profile being signaled
2911  ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
2912 
2913  ac->tags_mapped = 0;
2914  // parse
2915  while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
2916  elem_id = get_bits(gb, 4);
2917 
2918  if (elem_type < TYPE_DSE) {
2919  if (!(che=get_che(ac, elem_type, elem_id))) {
2920  av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
2921  elem_type, elem_id);
2922  err = AVERROR_INVALIDDATA;
2923  goto fail;
2924  }
2925  samples = 1024;
2926  }
2927 
2928  switch (elem_type) {
2929 
2930  case TYPE_SCE:
2931  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2932  audio_found = 1;
2933  sce_count++;
2934  break;
2935 
2936  case TYPE_CPE:
2937  err = decode_cpe(ac, gb, che);
2938  audio_found = 1;
2939  break;
2940 
2941  case TYPE_CCE:
2942  err = decode_cce(ac, gb, che);
2943  break;
2944 
2945  case TYPE_LFE:
2946  err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2947  audio_found = 1;
2948  break;
2949 
2950  case TYPE_DSE:
2951  err = skip_data_stream_element(ac, gb);
2952  break;
2953 
2954  case TYPE_PCE: {
2955  uint8_t layout_map[MAX_ELEM_ID*4][3];
2956  int tags;
2958  tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);
2959  if (tags < 0) {
2960  err = tags;
2961  break;
2962  }
2963  if (pce_found) {
2964  av_log(avctx, AV_LOG_ERROR,
2965  "Not evaluating a further program_config_element as this construct is dubious at best.\n");
2966  } else {
2967  err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
2968  if (!err)
2969  ac->oc[1].m4ac.chan_config = 0;
2970  pce_found = 1;
2971  }
2972  break;
2973  }
2974 
2975  case TYPE_FIL:
2976  if (elem_id == 15)
2977  elem_id += get_bits(gb, 8) - 1;
2978  if (get_bits_left(gb) < 8 * elem_id) {
2979  av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
2980  err = AVERROR_INVALIDDATA;
2981  goto fail;
2982  }
2983  while (elem_id > 0)
2984  elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
2985  err = 0; /* FIXME */
2986  break;
2987 
2988  default:
2989  err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
2990  break;
2991  }
2992 
2993  che_prev = che;
2994  elem_type_prev = elem_type;
2995 
2996  if (err)
2997  goto fail;
2998 
2999  if (get_bits_left(gb) < 3) {
3000  av_log(avctx, AV_LOG_ERROR, overread_err);
3001  err = AVERROR_INVALIDDATA;
3002  goto fail;
3003  }
3004  }
3005 
3006  spectral_to_sample(ac);
3007 
3008  multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
3009  samples <<= multiplier;
3010 
3011  if (ac->oc[1].status && audio_found) {
3012  avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
3013  avctx->frame_size = samples;
3014  ac->oc[1].status = OC_LOCKED;
3015  }
3016 
3017  if (multiplier) {
3018  int side_size;
3019  const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
3020  if (side && side_size>=4)
3021  AV_WL32(side, 2*AV_RL32(side));
3022  }
3023 
3024  *got_frame_ptr = !!samples;
3025  if (samples) {
3026  ac->frame->nb_samples = samples;
3027  ac->frame->sample_rate = avctx->sample_rate;
3028  } else
3029  av_frame_unref(ac->frame);
3030  *got_frame_ptr = !!samples;
3031 
3032  /* for dual-mono audio (SCE + SCE) */
3033  is_dmono = ac->dmono_mode && sce_count == 2 &&
3035  if (is_dmono) {
3036  if (ac->dmono_mode == 1)
3037  ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
3038  else if (ac->dmono_mode == 2)
3039  ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
3040  }
3041 
3042  return 0;
3043 fail:
3045  return err;
3046 }
3047 
3048 static int aac_decode_frame(AVCodecContext *avctx, void *data,
3049  int *got_frame_ptr, AVPacket *avpkt)
3050 {
3051  AACContext *ac = avctx->priv_data;
3052  const uint8_t *buf = avpkt->data;
3053  int buf_size = avpkt->size;
3054  GetBitContext gb;
3055  int buf_consumed;
3056  int buf_offset;
3057  int err;
3058  int new_extradata_size;
3059  const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
3061  &new_extradata_size);
3062  int jp_dualmono_size;
3063  const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
3065  &jp_dualmono_size);
3066 
3067  if (new_extradata && 0) {
3068  av_free(avctx->extradata);
3069  avctx->extradata = av_mallocz(new_extradata_size +
3071  if (!avctx->extradata)
3072  return AVERROR(ENOMEM);
3073  avctx->extradata_size = new_extradata_size;
3074  memcpy(avctx->extradata, new_extradata, new_extradata_size);
3076  if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
3077  avctx->extradata,
3078  avctx->extradata_size*8, 1) < 0) {
3080  return AVERROR_INVALIDDATA;
3081  }
3082  }
3083 
3084  ac->dmono_mode = 0;
3085  if (jp_dualmono && jp_dualmono_size > 0)
3086  ac->dmono_mode = 1 + *jp_dualmono;
3087  if (ac->force_dmono_mode >= 0)
3088  ac->dmono_mode = ac->force_dmono_mode;
3089 
3090  if (INT_MAX / 8 <= buf_size)
3091  return AVERROR_INVALIDDATA;
3092 
3093  if ((err = init_get_bits(&gb, buf, buf_size * 8)) < 0)
3094  return err;
3095 
3096  switch (ac->oc[1].m4ac.object_type) {
3097  case AOT_ER_AAC_LC:
3098  case AOT_ER_AAC_LTP:
3099  case AOT_ER_AAC_LD:
3100  case AOT_ER_AAC_ELD:
3101  err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
3102  break;
3103  default:
3104  err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
3105  }
3106  if (err < 0)
3107  return err;
3108 
3109  buf_consumed = (get_bits_count(&gb) + 7) >> 3;
3110  for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
3111  if (buf[buf_offset])
3112  break;
3113 
3114  return buf_size > buf_offset ? buf_consumed : buf_size;
3115 }
3116 
3118 {
3119  AACContext *ac = avctx->priv_data;
3120  int i, type;
3121 
3122  for (i = 0; i < MAX_ELEM_ID; i++) {
3123  for (type = 0; type < 4; type++) {
3124  if (ac->che[type][i])
3125  ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
3126  av_freep(&ac->che[type][i]);
3127  }
3128  }
3129 
3130  ff_mdct_end(&ac->mdct);
3131  ff_mdct_end(&ac->mdct_small);
3132  ff_mdct_end(&ac->mdct_ld);
3133  ff_mdct_end(&ac->mdct_ltp);
3134  return 0;
3135 }
3136 
3137 
3138 #define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
3139 
3140 struct LATMContext {
3141  AACContext aac_ctx; ///< containing AACContext
3142  int initialized; ///< initialized after a valid extradata was seen
3143 
3144  // parser data
3145  int audio_mux_version_A; ///< LATM syntax version
3146  int frame_length_type; ///< 0/1 variable/fixed frame length
3147  int frame_length; ///< frame length for fixed frame length
3148 };
3149 
3150 static inline uint32_t latm_get_value(GetBitContext *b)
3151 {
3152  int length = get_bits(b, 2);
3153 
3154  return get_bits_long(b, (length+1)*8);
3155 }
3156 
3158  GetBitContext *gb, int asclen)
3159 {
3160  AACContext *ac = &latmctx->aac_ctx;
3161  AVCodecContext *avctx = ac->avctx;
3162  MPEG4AudioConfig m4ac = { 0 };
3163  int config_start_bit = get_bits_count(gb);
3164  int sync_extension = 0;
3165  int bits_consumed, esize;
3166 
3167  if (asclen) {
3168  sync_extension = 1;
3169  asclen = FFMIN(asclen, get_bits_left(gb));
3170  } else
3171  asclen = get_bits_left(gb);
3172 
3173  if (config_start_bit % 8) {
3175  "Non-byte-aligned audio-specific config");
3176  return AVERROR_PATCHWELCOME;
3177  }
3178  if (asclen <= 0)
3179  return AVERROR_INVALIDDATA;
3180  bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
3181  gb->buffer + (config_start_bit / 8),
3182  asclen, sync_extension);
3183 
3184  if (bits_consumed < 0)
3185  return AVERROR_INVALIDDATA;
3186 
3187  if (!latmctx->initialized ||
3188  ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
3189  ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
3190 
3191  if(latmctx->initialized) {
3192  av_log(avctx, AV_LOG_INFO, "audio config changed\n");
3193  } else {
3194  av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
3195  }
3196  latmctx->initialized = 0;
3197 
3198  esize = (bits_consumed+7) / 8;
3199 
3200  if (avctx->extradata_size < esize) {
3201  av_free(avctx->extradata);
3203  if (!avctx->extradata)
3204  return AVERROR(ENOMEM);
3205  }
3206 
3207  avctx->extradata_size = esize;
3208  memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
3209  memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
3210  }
3211  skip_bits_long(gb, bits_consumed);
3212 
3213  return bits_consumed;
3214 }
3215 
3216 static int read_stream_mux_config(struct LATMContext *latmctx,
3217  GetBitContext *gb)
3218 {
3219  int ret, audio_mux_version = get_bits(gb, 1);
3220 
3221  latmctx->audio_mux_version_A = 0;
3222  if (audio_mux_version)
3223  latmctx->audio_mux_version_A = get_bits(gb, 1);
3224 
3225  if (!latmctx->audio_mux_version_A) {
3226 
3227  if (audio_mux_version)
3228  latm_get_value(gb); // taraFullness
3229 
3230  skip_bits(gb, 1); // allStreamSameTimeFraming
3231  skip_bits(gb, 6); // numSubFrames
3232  // numPrograms
3233  if (get_bits(gb, 4)) { // numPrograms
3234  avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
3235  return AVERROR_PATCHWELCOME;
3236  }
3237 
3238  // for each program (which there is only one in DVB)
3239 
3240  // for each layer (which there is only one in DVB)
3241  if (get_bits(gb, 3)) { // numLayer
3242  avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
3243  return AVERROR_PATCHWELCOME;
3244  }
3245 
3246  // for all but first stream: use_same_config = get_bits(gb, 1);
3247  if (!audio_mux_version) {
3248  if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
3249  return ret;
3250  } else {
3251  int ascLen = latm_get_value(gb);
3252  if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
3253  return ret;
3254  ascLen -= ret;
3255  skip_bits_long(gb, ascLen);
3256  }
3257 
3258  latmctx->frame_length_type = get_bits(gb, 3);
3259  switch (latmctx->frame_length_type) {
3260  case 0:
3261  skip_bits(gb, 8); // latmBufferFullness
3262  break;
3263  case 1:
3264  latmctx->frame_length = get_bits(gb, 9);
3265  break;
3266  case 3:
3267  case 4:
3268  case 5:
3269  skip_bits(gb, 6); // CELP frame length table index
3270  break;
3271  case 6:
3272  case 7:
3273  skip_bits(gb, 1); // HVXC frame length table index
3274  break;
3275  }
3276 
3277  if (get_bits(gb, 1)) { // other data
3278  if (audio_mux_version) {
3279  latm_get_value(gb); // other_data_bits
3280  } else {
3281  int esc;
3282  do {
3283  esc = get_bits(gb, 1);
3284  skip_bits(gb, 8);
3285  } while (esc);
3286  }
3287  }
3288 
3289  if (get_bits(gb, 1)) // crc present
3290  skip_bits(gb, 8); // config_crc
3291  }
3292 
3293  return 0;
3294 }
3295 
3297 {
3298  uint8_t tmp;
3299 
3300  if (ctx->frame_length_type == 0) {
3301  int mux_slot_length = 0;
3302  do {
3303  tmp = get_bits(gb, 8);
3304  mux_slot_length += tmp;
3305  } while (tmp == 255);
3306  return mux_slot_length;
3307  } else if (ctx->frame_length_type == 1) {
3308  return ctx->frame_length;
3309  } else if (ctx->frame_length_type == 3 ||
3310  ctx->frame_length_type == 5 ||
3311  ctx->frame_length_type == 7) {
3312  skip_bits(gb, 2); // mux_slot_length_coded
3313  }
3314  return 0;
3315 }
3316 
3317 static int read_audio_mux_element(struct LATMContext *latmctx,
3318  GetBitContext *gb)
3319 {
3320  int err;
3321  uint8_t use_same_mux = get_bits(gb, 1);
3322  if (!use_same_mux) {
3323  if ((err = read_stream_mux_config(latmctx, gb)) < 0)
3324  return err;
3325  } else if (!latmctx->aac_ctx.avctx->extradata) {
3326  av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
3327  "no decoder config found\n");
3328  return AVERROR(EAGAIN);
3329  }
3330  if (latmctx->audio_mux_version_A == 0) {
3331  int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
3332  if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
3333  av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
3334  return AVERROR_INVALIDDATA;
3335  } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
3336  av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
3337  "frame length mismatch %d << %d\n",
3338  mux_slot_length_bytes * 8, get_bits_left(gb));
3339  return AVERROR_INVALIDDATA;
3340  }
3341  }
3342  return 0;
3343 }
3344 
3345 
3346 static int latm_decode_frame(AVCodecContext *avctx, void *out,
3347  int *got_frame_ptr, AVPacket *avpkt)
3348 {
3349  struct LATMContext *latmctx = avctx->priv_data;
3350  int muxlength, err;
3351  GetBitContext gb;
3352 
3353  if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
3354  return err;
3355 
3356  // check for LOAS sync word
3357  if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
3358  return AVERROR_INVALIDDATA;
3359 
3360  muxlength = get_bits(&gb, 13) + 3;
3361  // not enough data, the parser should have sorted this out
3362  if (muxlength > avpkt->size)
3363  return AVERROR_INVALIDDATA;
3364 
3365  if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
3366  return err;
3367 
3368  if (!latmctx->initialized) {
3369  if (!avctx->extradata) {
3370  *got_frame_ptr = 0;
3371  return avpkt->size;
3372  } else {
3374  if ((err = decode_audio_specific_config(
3375  &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
3376  avctx->extradata, avctx->extradata_size*8, 1)) < 0) {
3377  pop_output_configuration(&latmctx->aac_ctx);
3378  return err;
3379  }
3380  latmctx->initialized = 1;
3381  }
3382  }
3383 
3384  if (show_bits(&gb, 12) == 0xfff) {
3385  av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
3386  "ADTS header detected, probably as result of configuration "
3387  "misparsing\n");
3388  return AVERROR_INVALIDDATA;
3389  }
3390 
3391  if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt)) < 0)
3392  return err;
3393 
3394  return muxlength;
3395 }
3396 
3398 {
3399  struct LATMContext *latmctx = avctx->priv_data;
3400  int ret = aac_decode_init(avctx);
3401 
3402  if (avctx->extradata_size > 0)
3403  latmctx->initialized = !ret;
3404 
3405  return ret;
3406 }
3407 
3408 static void aacdec_init(AACContext *c)
3409 {
3411  c->apply_ltp = apply_ltp;
3412  c->apply_tns = apply_tns;
3414  c->update_ltp = update_ltp;
3415 
3416  if(ARCH_MIPS)
3418 }
3419 /**
3420  * AVOptions for Japanese DTV specific extensions (ADTS only)
3421  */
3422 #define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
3423 static const AVOption options[] = {
3424  {"dual_mono_mode", "Select the channel to decode for dual mono",
3425  offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
3426  AACDEC_FLAGS, "dual_mono_mode"},
3427 
3428  {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3429  {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3430  {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3431  {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3432 
3433  {NULL},
3434 };
3435 
3436 static const AVClass aac_decoder_class = {
3437  .class_name = "AAC decoder",
3438  .item_name = av_default_item_name,
3439  .option = options,
3440  .version = LIBAVUTIL_VERSION_INT,
3441 };
3442 
3444  .name = "aac",
3445  .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
3446  .type = AVMEDIA_TYPE_AUDIO,
3447  .id = AV_CODEC_ID_AAC,
3448  .priv_data_size = sizeof(AACContext),
3449  .init = aac_decode_init,
3452  .sample_fmts = (const enum AVSampleFormat[]) {
3454  },
3455  .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
3456  .channel_layouts = aac_channel_layout,
3457  .flush = flush,
3458  .priv_class = &aac_decoder_class,
3459 };
3460 
3461 /*
3462  Note: This decoder filter is intended to decode LATM streams transferred
3463  in MPEG transport streams which only contain one program.
3464  To do a more complex LATM demuxing a separate LATM demuxer should be used.
3465 */
3467  .name = "aac_latm",
3468  .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
3469  .type = AVMEDIA_TYPE_AUDIO,
3470  .id = AV_CODEC_ID_AAC_LATM,
3471  .priv_data_size = sizeof(struct LATMContext),
3472  .init = latm_decode_init,
3473  .close = aac_decode_close,
3474  .decode = latm_decode_frame,
3475  .sample_fmts = (const enum AVSampleFormat[]) {
3477  },
3478  .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
3479  .channel_layouts = aac_channel_layout,
3480  .flush = flush,
3481 };