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Imported Debian version 2.4.3~trusty1
[deb_ffmpeg.git] / ffmpeg / libavcodec / aacdec.c
CommitLineData
2ba45a60
DM		 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
117static VLC vlc_scalefactors;
118static VLC vlc_spectral[11];
119
120static 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
126static 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 */
150static av_cold int che_configure(AACContext *ac,
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
181static int frame_configure_elements(AVCodecContext *avctx)
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
215struct elem_to_channel {
216 uint64_t av_position;
217 uint8_t syn_ele;
218 uint8_t elem_id;
219 uint8_t aac_position;
220};
221
222static 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
251static 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
283static 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) {
310 .av_position = AV_CH_FRONT_CENTER,
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,
320 AV_CH_FRONT_LEFT_OF_CENTER,
321 AV_CH_FRONT_RIGHT_OF_CENTER,
322 AAC_CHANNEL_FRONT);
323 num_front_channels -= 2;
324 }
325 if (num_front_channels >= 2) {
326 i += assign_pair(e2c_vec, layout_map, i,
327 AV_CH_FRONT_LEFT,
328 AV_CH_FRONT_RIGHT,
329 AAC_CHANNEL_FRONT);
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,
336 AAC_CHANNEL_FRONT);
337 num_front_channels -= 2;
338 }
339
340 if (num_side_channels >= 2) {
341 i += assign_pair(e2c_vec, layout_map, i,
342 AV_CH_SIDE_LEFT,
343 AV_CH_SIDE_RIGHT,
344 AAC_CHANNEL_FRONT);
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,
351 AAC_CHANNEL_SIDE);
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,
359 AAC_CHANNEL_BACK);
360 num_back_channels -= 2;
361 }
362 if (num_back_channels >= 2) {
363 i += assign_pair(e2c_vec, layout_map, i,
364 AV_CH_BACK_LEFT,
365 AV_CH_BACK_RIGHT,
366 AAC_CHANNEL_BACK);
367 num_back_channels -= 2;
368 }
369 if (num_back_channels) {
370 e2c_vec[i] = (struct elem_to_channel) {
371 .av_position = AV_CH_BACK_CENTER,
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) {
382 .av_position = AV_CH_LOW_FREQUENCY,
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 */
427static void push_output_configuration(AACContext *ac) {
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 */
438static void pop_output_configuration(AACContext *ac) {
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 */
454static int output_configure(AACContext *ac,
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.
469 if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
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) {
483 layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
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
503static 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 */
526static int set_default_channel_config(AVCodecContext *avctx,
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
562static 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;
574 push_output_configuration(ac);
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;
593 push_output_configuration(ac);
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 */
664static 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 */
698static 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 */
768static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
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 ||
787 m4ac->object_type == AOT_ER_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) {
821 avpriv_report_missing_feature(avctx,
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) {
837 avpriv_report_missing_feature(avctx,
838 "epConfig %d", ep_config);
839 return AVERROR_PATCHWELCOME;
840 }
841 }
842 return 0;
843}
844
845static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
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) {
865 avpriv_report_missing_feature(avctx,
866 "AAC data resilience (flags %x)",
867 res_flags);
868 return AVERROR_PATCHWELCOME;
869 }
870
871 if (get_bits1(gb)) { // ldSbrPresentFlag
872 avpriv_report_missing_feature(avctx,
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) {
884 av_log(ac->avctx, AV_LOG_ERROR, overread_err);
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) {
899 avpriv_report_missing_feature(avctx,
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 */
918static int decode_audio_specific_config(AACContext *ac,
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:
970 avpriv_report_missing_feature(avctx,
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 */
993static 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
999static av_always_inline void reset_predict_state(PredictorState *ps)
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
1009static void reset_all_predictors(PredictorState *ps)
1010{
1011 int i;
1012 for (i = 0; i < MAX_PREDICTORS; i++)
1013 reset_predict_state(&ps[i]);
1014}
1015
1016static 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
1032static 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
1047static void aacdec_init(AACContext *ac);
1048
1049static av_cold int aac_decode_init(AVCodecContext *avctx)
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;
1081 if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
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);
1117 avpriv_float_dsp_init(&ac->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
1118
1119 ac->random_state = 0x1f2e3d4c;
1120
1121 ff_aac_tableinit();
1122
1123 INIT_VLC_STATIC(&vlc_scalefactors, 7,
1124 FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
1125 ff_aac_scalefactor_bits,
1126 sizeof(ff_aac_scalefactor_bits[0]),
1127 sizeof(ff_aac_scalefactor_bits[0]),
1128 ff_aac_scalefactor_code,
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
1138 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
1139 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
1140 ff_init_ff_sine_windows(10);
1141 ff_init_ff_sine_windows( 9);
1142 ff_init_ff_sine_windows( 7);
1143
1144 cbrt_tableinit();
1145
1146 return 0;
1147}
1148
1149/**
1150 * Skip data_stream_element; reference: table 4.10.
1151 */
1152static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
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
1169static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
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 */
1191static void decode_ltp(LongTermPrediction *ltp,
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 */
1205static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
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]);
1221 ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
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;
1241 ics->swb_offset = ff_swb_offset_128[ac->oc[1].m4ac.sampling_index];
1242 ics->num_swb = ff_aac_num_swb_128[ac->oc[1].m4ac.sampling_index];
1243 ics->tns_max_bands = ff_tns_max_bands_128[ac->oc[1].m4ac.sampling_index];
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) {
1249 ics->swb_offset = ff_swb_offset_512[ac->oc[1].m4ac.sampling_index];
1250 ics->num_swb = ff_aac_num_swb_512[ac->oc[1].m4ac.sampling_index];
1251 ics->tns_max_bands = ff_tns_max_bands_512[ac->oc[1].m4ac.sampling_index];
1252 if (!ics->num_swb || !ics->swb_offset)
1253 return AVERROR_BUG;
1254 } else {
1255 ics->swb_offset = ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
1256 ics->num_swb = ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
1257 ics->tns_max_bands = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
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;
1294fail:
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 */
1307static int decode_band_types(AACContext *ac, enum BandType band_type[120],
1308 int band_type_run_end[120], GetBitContext *gb,
1309 IndividualChannelStream *ics)
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 */
1356static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
1357 unsigned int global_gain,
1358 IndividualChannelStream *ics,
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) {
1378 avpriv_request_sample(ac->avctx,
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) {
1393 avpriv_request_sample(ac->avctx,
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 */
1419static 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 */
1446static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
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 */
1489static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
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
1504static 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
1515static 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
1528static 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
1545static 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 */
1583static 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++) {
1617 ac->random_state = lcg_random(ac->random_state);
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
1798static 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
1806static 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
1814static 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
1822static 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 */
1856static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1857{
1858 int sfb, k;
1859
1860 if (!sce->ics.predictor_initialized) {
1861 reset_all_predictors(sce->predictor_state);
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)
1878 reset_predictor_group(sce->predictor_state,
1879 sce->ics.predictor_reset_group);
1880 } else
1881 reset_all_predictors(sce->predictor_state);
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 */
1892static int decode_ics(AACContext *ac, SingleChannelElement *sce,
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 */
1969static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
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 */
2000static void apply_intensity_stereo(AACContext *ac,
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 */
2042static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
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
2083static 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 */
2095static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
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 */
2167static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
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 */
2186static int decode_dynamic_range(DynamicRangeControl *che_drc,
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
2233static 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
2253unknown:
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 */
2266static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
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;
2289 ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
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;
2294 ac->avctx->profile = FF_PROFILE_AAC_HE;
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 */
2319static 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 */
2375static 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 */
2401static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
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 */
2433static void update_ltp(AACContext *ac, SingleChannelElement *sce)
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 */
2468static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
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) &&
2495 (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_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
2528static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
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
2553static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
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 */
2614static void apply_dependent_coupling(AACContext *ac,
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 */
2650static void apply_independent_coupling(AACContext *ac,
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 */
2669static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
2670 enum RawDataBlockType type, int elem_id,
2671 enum CouplingPoint coupling_point,
2672 void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
2673{
2674 int i, c;
2675
2676 for (i = 0; i < MAX_ELEM_ID; i++) {
2677 ChannelElement *cce = ac->che[TYPE_CCE][i];
2678 int index = 0;
2679
2680 if (cce && cce->coup.coupling_point == coupling_point) {
2681 ChannelCoupling *coup = &cce->coup;
2682
2683 for (c = 0; c <= coup->num_coupled; c++) {
2684 if (coup->type[c] == type && coup->id_select[c] == elem_id) {
2685 if (coup->ch_select[c] != 1) {
2686 apply_coupling_method(ac, &cc->ch[0], cce, index);
2687 if (coup->ch_select[c] != 0)
2688 index++;
2689 }
2690 if (coup->ch_select[c] != 2)
2691 apply_coupling_method(ac, &cc->ch[1], cce, index++);
2692 } else
2693 index += 1 + (coup->ch_select[c] == 3);
2694 }
2695 }
2696 }
2697}
2698
2699/**
2700 * Convert spectral data to float samples, applying all supported tools as appropriate.
2701 */
2702static void spectral_to_sample(AACContext *ac)
2703{
2704 int i, type;
2705 void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
2706 switch (ac->oc[1].m4ac.object_type) {
2707 case AOT_ER_AAC_LD:
2708 imdct_and_window = imdct_and_windowing_ld;
2709 break;
2710 case AOT_ER_AAC_ELD:
2711 imdct_and_window = imdct_and_windowing_eld;
2712 break;
2713 default:
2714 imdct_and_window = ac->imdct_and_windowing;
2715 }
2716 for (type = 3; type >= 0; type--) {
2717 for (i = 0; i < MAX_ELEM_ID; i++) {
2718 ChannelElement *che = ac->che[type][i];
2719 if (che) {
2720 if (type <= TYPE_CPE)
2721 apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
2722 if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2723 if (che->ch[0].ics.predictor_present) {
2724 if (che->ch[0].ics.ltp.present)
2725 ac->apply_ltp(ac, &che->ch[0]);
2726 if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
2727 ac->apply_ltp(ac, &che->ch[1]);
2728 }
2729 }
2730 if (che->ch[0].tns.present)
2731 ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2732 if (che->ch[1].tns.present)
2733 ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2734 if (type <= TYPE_CPE)
2735 apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
2736 if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2737 imdct_and_window(ac, &che->ch[0]);
2738 if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2739 ac->update_ltp(ac, &che->ch[0]);
2740 if (type == TYPE_CPE) {
2741 imdct_and_window(ac, &che->ch[1]);
2742 if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2743 ac->update_ltp(ac, &che->ch[1]);
2744 }
2745 if (ac->oc[1].m4ac.sbr > 0) {
2746 ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2747 }
2748 }
2749 if (type <= TYPE_CCE)
2750 apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
2751 }
2752 }
2753 }
2754}
2755
2756static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
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.
2768 avpriv_report_missing_feature(ac->avctx,
2769 "More than one AAC RDB per ADTS frame");
2770 ac->warned_num_aac_frames = 1;
2771 }
2772 push_output_configuration(ac);
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
2817static 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
2882static 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;
2957 push_output_configuration(ac);
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 &&
3034 ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
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;
3043fail:
3044 pop_output_configuration(ac);
3045 return err;
3046}
3047
3048static 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,
3060 AV_PKT_DATA_NEW_EXTRADATA,
3061 &new_extradata_size);
3062 int jp_dualmono_size;
3063 const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
3064 AV_PKT_DATA_JP_DUALMONO,
3065 &jp_dualmono_size);
3066
3067 if (new_extradata && 0) {
3068 av_free(avctx->extradata);
3069 avctx->extradata = av_mallocz(new_extradata_size +
3070 FF_INPUT_BUFFER_PADDING_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);
3075 push_output_configuration(ac);
3076 if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
3077 avctx->extradata,
3078 avctx->extradata_size*8, 1) < 0) {
3079 pop_output_configuration(ac);
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
3117static av_cold int aac_decode_close(AVCodecContext *avctx)
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
3140struct 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
3150static 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
3157static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
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) {
3174 avpriv_request_sample(latmctx->aac_ctx.avctx,
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);
3202 avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
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
3216static 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
3296static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
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
3317static 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
3346static 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 {
3373 push_output_configuration(&latmctx->aac_ctx);
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
3397static av_cold int latm_decode_init(AVCodecContext *avctx)
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
3408static void aacdec_init(AACContext *c)
3409{
3410 c->imdct_and_windowing = imdct_and_windowing;
3411 c->apply_ltp = apply_ltp;
3412 c->apply_tns = apply_tns;
3413 c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
3414 c->update_ltp = update_ltp;
3415
3416 if(ARCH_MIPS)
3417 ff_aacdec_init_mips(c);
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
3423static 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
3436static 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
3443AVCodec ff_aac_decoder = {
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,
3450 .close = aac_decode_close,
3451 .decode = aac_decode_frame,
3452 .sample_fmts = (const enum AVSampleFormat[]) {
3453 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
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*/
3466AVCodec ff_aac_latm_decoder = {
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[]) {
3476 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
3477 },
3478 .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
3479 .channel_layouts = aac_channel_layout,
3480 .flush = flush,
3481};
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