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Imported Debian version 2.5.0~trusty1.1
[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)) {
f6fa7814
DM		 624 if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
625 av_log(ac->avctx, AV_LOG_WARNING,
626 "This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
627 type == TYPE_SCE ? "SCE" : "LFE", elem_id);
628 ac->warned_remapping_once++;
629 }
2ba45a60
DM		 630 ac->tags_mapped++;
631 return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
632 }
633 case 5:
634 if (ac->tags_mapped == 2 && type == TYPE_CPE) {
635 ac->tags_mapped++;
636 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
637 }
638 case 4:
f6fa7814
DM		 639 /* Some streams incorrectly code 4.0 audio as
640 * SCE[0] CPE[0] LFE[0]
641 * instead of
642 * SCE[0] CPE[0] SCE[1].
643 * If we seem to have encountered such a stream, transfer
644 * the SCE[1] element to the LFE[0]'s mapping */
645 if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
646 if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
647 av_log(ac->avctx, AV_LOG_WARNING,
648 "This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
649 type == TYPE_SCE ? "SCE" : "LFE", elem_id);
650 ac->warned_remapping_once++;
651 }
652 ac->tags_mapped++;
653 return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
654 }
2ba45a60
DM		 655 if (ac->tags_mapped == 2 &&
656 ac->oc[1].m4ac.chan_config == 4 &&
657 type == TYPE_SCE) {
658 ac->tags_mapped++;
659 return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
660 }
661 case 3:
662 case 2:
663 if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
664 type == TYPE_CPE) {
665 ac->tags_mapped++;
666 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
667 } else if (ac->oc[1].m4ac.chan_config == 2) {
668 return NULL;
669 }
670 case 1:
671 if (!ac->tags_mapped && type == TYPE_SCE) {
672 ac->tags_mapped++;
673 return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
674 }
675 default:
676 return NULL;
677 }
678}
679
680/**
681 * Decode an array of 4 bit element IDs, optionally interleaved with a
682 * stereo/mono switching bit.
683 *
684 * @param type speaker type/position for these channels
685 */
686static void decode_channel_map(uint8_t layout_map[][3],
687 enum ChannelPosition type,
688 GetBitContext *gb, int n)
689{
690 while (n--) {
691 enum RawDataBlockType syn_ele;
692 switch (type) {
693 case AAC_CHANNEL_FRONT:
694 case AAC_CHANNEL_BACK:
695 case AAC_CHANNEL_SIDE:
696 syn_ele = get_bits1(gb);
697 break;
698 case AAC_CHANNEL_CC:
699 skip_bits1(gb);
700 syn_ele = TYPE_CCE;
701 break;
702 case AAC_CHANNEL_LFE:
703 syn_ele = TYPE_LFE;
704 break;
705 default:
f6fa7814 706 // AAC_CHANNEL_OFF has no channel map
2ba45a60
DM		 707 av_assert0(0);
708 }
709 layout_map[0][0] = syn_ele;
710 layout_map[0][1] = get_bits(gb, 4);
711 layout_map[0][2] = type;
712 layout_map++;
713 }
714}
715
716/**
717 * Decode program configuration element; reference: table 4.2.
718 *
719 * @return Returns error status. 0 - OK, !0 - error
720 */
721static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
722 uint8_t (*layout_map)[3],
723 GetBitContext *gb)
724{
725 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
726 int sampling_index;
727 int comment_len;
728 int tags;
729
730 skip_bits(gb, 2); // object_type
731
732 sampling_index = get_bits(gb, 4);
733 if (m4ac->sampling_index != sampling_index)
734 av_log(avctx, AV_LOG_WARNING,
735 "Sample rate index in program config element does not "
736 "match the sample rate index configured by the container.\n");
737
738 num_front = get_bits(gb, 4);
739 num_side = get_bits(gb, 4);
740 num_back = get_bits(gb, 4);
741 num_lfe = get_bits(gb, 2);
742 num_assoc_data = get_bits(gb, 3);
743 num_cc = get_bits(gb, 4);
744
745 if (get_bits1(gb))
746 skip_bits(gb, 4); // mono_mixdown_tag
747 if (get_bits1(gb))
748 skip_bits(gb, 4); // stereo_mixdown_tag
749
750 if (get_bits1(gb))
751 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
752
753 if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
754 av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
755 return -1;
756 }
757 decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front);
758 tags = num_front;
759 decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side);
760 tags += num_side;
761 decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back);
762 tags += num_back;
763 decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe);
764 tags += num_lfe;
765
766 skip_bits_long(gb, 4 * num_assoc_data);
767
768 decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc);
769 tags += num_cc;
770
771 align_get_bits(gb);
772
773 /* comment field, first byte is length */
774 comment_len = get_bits(gb, 8) * 8;
775 if (get_bits_left(gb) < comment_len) {
776 av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
777 return AVERROR_INVALIDDATA;
778 }
779 skip_bits_long(gb, comment_len);
780 return tags;
781}
782
783/**
784 * Decode GA "General Audio" specific configuration; reference: table 4.1.
785 *
786 * @param ac pointer to AACContext, may be null
787 * @param avctx pointer to AVCCodecContext, used for logging
788 *
789 * @return Returns error status. 0 - OK, !0 - error
790 */
791static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
792 GetBitContext *gb,
793 MPEG4AudioConfig *m4ac,
794 int channel_config)
795{
796 int extension_flag, ret, ep_config, res_flags;
797 uint8_t layout_map[MAX_ELEM_ID*4][3];
798 int tags = 0;
799
800 if (get_bits1(gb)) { // frameLengthFlag
801 avpriv_request_sample(avctx, "960/120 MDCT window");
802 return AVERROR_PATCHWELCOME;
803 }
804
805 if (get_bits1(gb)) // dependsOnCoreCoder
806 skip_bits(gb, 14); // coreCoderDelay
807 extension_flag = get_bits1(gb);
808
809 if (m4ac->object_type == AOT_AAC_SCALABLE ||
810 m4ac->object_type == AOT_ER_AAC_SCALABLE)
811 skip_bits(gb, 3); // layerNr
812
813 if (channel_config == 0) {
814 skip_bits(gb, 4); // element_instance_tag
815 tags = decode_pce(avctx, m4ac, layout_map, gb);
816 if (tags < 0)
817 return tags;
818 } else {
819 if ((ret = set_default_channel_config(avctx, layout_map,
820 &tags, channel_config)))
821 return ret;
822 }
823
824 if (count_channels(layout_map, tags) > 1) {
825 m4ac->ps = 0;
826 } else if (m4ac->sbr == 1 && m4ac->ps == -1)
827 m4ac->ps = 1;
828
829 if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
830 return ret;
831
832 if (extension_flag) {
833 switch (m4ac->object_type) {
834 case AOT_ER_BSAC:
835 skip_bits(gb, 5); // numOfSubFrame
836 skip_bits(gb, 11); // layer_length
837 break;
838 case AOT_ER_AAC_LC:
839 case AOT_ER_AAC_LTP:
840 case AOT_ER_AAC_SCALABLE:
841 case AOT_ER_AAC_LD:
842 res_flags = get_bits(gb, 3);
843 if (res_flags) {
844 avpriv_report_missing_feature(avctx,
845 "AAC data resilience (flags %x)",
846 res_flags);
847 return AVERROR_PATCHWELCOME;
848 }
849 break;
850 }
851 skip_bits1(gb); // extensionFlag3 (TBD in version 3)
852 }
853 switch (m4ac->object_type) {
854 case AOT_ER_AAC_LC:
855 case AOT_ER_AAC_LTP:
856 case AOT_ER_AAC_SCALABLE:
857 case AOT_ER_AAC_LD:
858 ep_config = get_bits(gb, 2);
859 if (ep_config) {
860 avpriv_report_missing_feature(avctx,
861 "epConfig %d", ep_config);
862 return AVERROR_PATCHWELCOME;
863 }
864 }
865 return 0;
866}
867
868static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
869 GetBitContext *gb,
870 MPEG4AudioConfig *m4ac,
871 int channel_config)
872{
873 int ret, ep_config, res_flags;
874 uint8_t layout_map[MAX_ELEM_ID*4][3];
875 int tags = 0;
876 const int ELDEXT_TERM = 0;
877
878 m4ac->ps = 0;
879 m4ac->sbr = 0;
880
881 if (get_bits1(gb)) { // frameLengthFlag
882 avpriv_request_sample(avctx, "960/120 MDCT window");
883 return AVERROR_PATCHWELCOME;
884 }
885
886 res_flags = get_bits(gb, 3);
887 if (res_flags) {
888 avpriv_report_missing_feature(avctx,
889 "AAC data resilience (flags %x)",
890 res_flags);
891 return AVERROR_PATCHWELCOME;
892 }
893
894 if (get_bits1(gb)) { // ldSbrPresentFlag
895 avpriv_report_missing_feature(avctx,
896 "Low Delay SBR");
897 return AVERROR_PATCHWELCOME;
898 }
899
900 while (get_bits(gb, 4) != ELDEXT_TERM) {
901 int len = get_bits(gb, 4);
902 if (len == 15)
903 len += get_bits(gb, 8);
904 if (len == 15 + 255)
905 len += get_bits(gb, 16);
906 if (get_bits_left(gb) < len * 8 + 4) {
907 av_log(ac->avctx, AV_LOG_ERROR, overread_err);
908 return AVERROR_INVALIDDATA;
909 }
910 skip_bits_long(gb, 8 * len);
911 }
912
913 if ((ret = set_default_channel_config(avctx, layout_map,
914 &tags, channel_config)))
915 return ret;
916
917 if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
918 return ret;
919
920 ep_config = get_bits(gb, 2);
921 if (ep_config) {
922 avpriv_report_missing_feature(avctx,
923 "epConfig %d", ep_config);
924 return AVERROR_PATCHWELCOME;
925 }
926 return 0;
927}
928
929/**
930 * Decode audio specific configuration; reference: table 1.13.
931 *
932 * @param ac pointer to AACContext, may be null
933 * @param avctx pointer to AVCCodecContext, used for logging
934 * @param m4ac pointer to MPEG4AudioConfig, used for parsing
935 * @param data pointer to buffer holding an audio specific config
936 * @param bit_size size of audio specific config or data in bits
937 * @param sync_extension look for an appended sync extension
938 *
939 * @return Returns error status or number of consumed bits. <0 - error
940 */
941static int decode_audio_specific_config(AACContext *ac,
942 AVCodecContext *avctx,
943 MPEG4AudioConfig *m4ac,
944 const uint8_t *data, int bit_size,
945 int sync_extension)
946{
947 GetBitContext gb;
948 int i, ret;
949
950 av_dlog(avctx, "audio specific config size %d\n", bit_size >> 3);
951 for (i = 0; i < bit_size >> 3; i++)
952 av_dlog(avctx, "%02x ", data[i]);
953 av_dlog(avctx, "\n");
954
955 if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
956 return ret;
957
958 if ((i = avpriv_mpeg4audio_get_config(m4ac, data, bit_size,
959 sync_extension)) < 0)
960 return AVERROR_INVALIDDATA;
961 if (m4ac->sampling_index > 12) {
962 av_log(avctx, AV_LOG_ERROR,
963 "invalid sampling rate index %d\n",
964 m4ac->sampling_index);
965 return AVERROR_INVALIDDATA;
966 }
967 if (m4ac->object_type == AOT_ER_AAC_LD &&
968 (m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
969 av_log(avctx, AV_LOG_ERROR,
970 "invalid low delay sampling rate index %d\n",
971 m4ac->sampling_index);
972 return AVERROR_INVALIDDATA;
973 }
974
975 skip_bits_long(&gb, i);
976
977 switch (m4ac->object_type) {
978 case AOT_AAC_MAIN:
979 case AOT_AAC_LC:
980 case AOT_AAC_LTP:
981 case AOT_ER_AAC_LC:
982 case AOT_ER_AAC_LD:
983 if ((ret = decode_ga_specific_config(ac, avctx, &gb,
984 m4ac, m4ac->chan_config)) < 0)
985 return ret;
986 break;
987 case AOT_ER_AAC_ELD:
988 if ((ret = decode_eld_specific_config(ac, avctx, &gb,
989 m4ac, m4ac->chan_config)) < 0)
990 return ret;
991 break;
992 default:
993 avpriv_report_missing_feature(avctx,
994 "Audio object type %s%d",
995 m4ac->sbr == 1 ? "SBR+" : "",
996 m4ac->object_type);
997 return AVERROR(ENOSYS);
998 }
999
1000 av_dlog(avctx,
1001 "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
1002 m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
1003 m4ac->sample_rate, m4ac->sbr,
1004 m4ac->ps);
1005
1006 return get_bits_count(&gb);
1007}
1008
1009/**
1010 * linear congruential pseudorandom number generator
1011 *
1012 * @param previous_val pointer to the current state of the generator
1013 *
1014 * @return Returns a 32-bit pseudorandom integer
1015 */
1016static av_always_inline int lcg_random(unsigned previous_val)
1017{
1018 union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
1019 return v.s;
1020}
1021
1022static av_always_inline void reset_predict_state(PredictorState *ps)
1023{
1024 ps->r0 = 0.0f;
1025 ps->r1 = 0.0f;
1026 ps->cor0 = 0.0f;
1027 ps->cor1 = 0.0f;
1028 ps->var0 = 1.0f;
1029 ps->var1 = 1.0f;
1030}
1031
1032static void reset_all_predictors(PredictorState *ps)
1033{
1034 int i;
1035 for (i = 0; i < MAX_PREDICTORS; i++)
1036 reset_predict_state(&ps[i]);
1037}
1038
1039static int sample_rate_idx (int rate)
1040{
1041 if (92017 <= rate) return 0;
1042 else if (75132 <= rate) return 1;
1043 else if (55426 <= rate) return 2;
1044 else if (46009 <= rate) return 3;
1045 else if (37566 <= rate) return 4;
1046 else if (27713 <= rate) return 5;
1047 else if (23004 <= rate) return 6;
1048 else if (18783 <= rate) return 7;
1049 else if (13856 <= rate) return 8;
1050 else if (11502 <= rate) return 9;
1051 else if (9391 <= rate) return 10;
1052 else return 11;
1053}
1054
1055static void reset_predictor_group(PredictorState *ps, int group_num)
1056{
1057 int i;
1058 for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
1059 reset_predict_state(&ps[i]);
1060}
1061
1062#define AAC_INIT_VLC_STATIC(num, size) \
1063 INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
1064 ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \
1065 sizeof(ff_aac_spectral_bits[num][0]), \
1066 ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
1067 sizeof(ff_aac_spectral_codes[num][0]), \
1068 size);
1069
1070static void aacdec_init(AACContext *ac);
1071
1072static av_cold int aac_decode_init(AVCodecContext *avctx)
1073{
1074 AACContext *ac = avctx->priv_data;
1075 int ret;
1076
1077 ac->avctx = avctx;
1078 ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
1079
1080 aacdec_init(ac);
1081
1082 avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1083
1084 if (avctx->extradata_size > 0) {
1085 if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
1086 avctx->extradata,
1087 avctx->extradata_size * 8,
1088 1)) < 0)
1089 return ret;
1090 } else {
1091 int sr, i;
1092 uint8_t layout_map[MAX_ELEM_ID*4][3];
1093 int layout_map_tags;
1094
1095 sr = sample_rate_idx(avctx->sample_rate);
1096 ac->oc[1].m4ac.sampling_index = sr;
1097 ac->oc[1].m4ac.channels = avctx->channels;
1098 ac->oc[1].m4ac.sbr = -1;
1099 ac->oc[1].m4ac.ps = -1;
1100
1101 for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
1102 if (ff_mpeg4audio_channels[i] == avctx->channels)
1103 break;
1104 if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
1105 i = 0;
1106 }
1107 ac->oc[1].m4ac.chan_config = i;
1108
1109 if (ac->oc[1].m4ac.chan_config) {
1110 int ret = set_default_channel_config(avctx, layout_map,
1111 &layout_map_tags, ac->oc[1].m4ac.chan_config);
1112 if (!ret)
1113 output_configure(ac, layout_map, layout_map_tags,
1114 OC_GLOBAL_HDR, 0);
1115 else if (avctx->err_recognition & AV_EF_EXPLODE)
1116 return AVERROR_INVALIDDATA;
1117 }
1118 }
1119
1120 if (avctx->channels > MAX_CHANNELS) {
1121 av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
1122 return AVERROR_INVALIDDATA;
1123 }
1124
1125 AAC_INIT_VLC_STATIC( 0, 304);
1126 AAC_INIT_VLC_STATIC( 1, 270);
1127 AAC_INIT_VLC_STATIC( 2, 550);
1128 AAC_INIT_VLC_STATIC( 3, 300);
1129 AAC_INIT_VLC_STATIC( 4, 328);
1130 AAC_INIT_VLC_STATIC( 5, 294);
1131 AAC_INIT_VLC_STATIC( 6, 306);
1132 AAC_INIT_VLC_STATIC( 7, 268);
1133 AAC_INIT_VLC_STATIC( 8, 510);
1134 AAC_INIT_VLC_STATIC( 9, 366);
1135 AAC_INIT_VLC_STATIC(10, 462);
1136
1137 ff_aac_sbr_init();
1138
1139 ff_fmt_convert_init(&ac->fmt_conv, avctx);
f6fa7814
DM		 1140 ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
1141 if (!ac->fdsp) {
1142 return AVERROR(ENOMEM);
1143 }
2ba45a60
DM		 1144
1145 ac->random_state = 0x1f2e3d4c;
1146
1147 ff_aac_tableinit();
1148
1149 INIT_VLC_STATIC(&vlc_scalefactors, 7,
1150 FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
1151 ff_aac_scalefactor_bits,
1152 sizeof(ff_aac_scalefactor_bits[0]),
1153 sizeof(ff_aac_scalefactor_bits[0]),
1154 ff_aac_scalefactor_code,
1155 sizeof(ff_aac_scalefactor_code[0]),
1156 sizeof(ff_aac_scalefactor_code[0]),
1157 352);
1158
1159 ff_mdct_init(&ac->mdct, 11, 1, 1.0 / (32768.0 * 1024.0));
1160 ff_mdct_init(&ac->mdct_ld, 10, 1, 1.0 / (32768.0 * 512.0));
1161 ff_mdct_init(&ac->mdct_small, 8, 1, 1.0 / (32768.0 * 128.0));
1162 ff_mdct_init(&ac->mdct_ltp, 11, 0, -2.0 * 32768.0);
1163 // window initialization
1164 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
1165 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
1166 ff_init_ff_sine_windows(10);
1167 ff_init_ff_sine_windows( 9);
1168 ff_init_ff_sine_windows( 7);
1169
1170 cbrt_tableinit();
1171
1172 return 0;
1173}
1174
1175/**
1176 * Skip data_stream_element; reference: table 4.10.
1177 */
1178static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
1179{
1180 int byte_align = get_bits1(gb);
1181 int count = get_bits(gb, 8);
1182 if (count == 255)
1183 count += get_bits(gb, 8);
1184 if (byte_align)
1185 align_get_bits(gb);
1186
1187 if (get_bits_left(gb) < 8 * count) {
1188 av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
1189 return AVERROR_INVALIDDATA;
1190 }
1191 skip_bits_long(gb, 8 * count);
1192 return 0;
1193}
1194
1195static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
1196 GetBitContext *gb)
1197{
1198 int sfb;
1199 if (get_bits1(gb)) {
1200 ics->predictor_reset_group = get_bits(gb, 5);
1201 if (ics->predictor_reset_group == 0 ||
1202 ics->predictor_reset_group > 30) {
1203 av_log(ac->avctx, AV_LOG_ERROR,
1204 "Invalid Predictor Reset Group.\n");
1205 return AVERROR_INVALIDDATA;
1206 }
1207 }
1208 for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
1209 ics->prediction_used[sfb] = get_bits1(gb);
1210 }
1211 return 0;
1212}
1213
1214/**
1215 * Decode Long Term Prediction data; reference: table 4.xx.
1216 */
1217static void decode_ltp(LongTermPrediction *ltp,
1218 GetBitContext *gb, uint8_t max_sfb)
1219{
1220 int sfb;
1221
1222 ltp->lag = get_bits(gb, 11);
1223 ltp->coef = ltp_coef[get_bits(gb, 3)];
1224 for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
1225 ltp->used[sfb] = get_bits1(gb);
1226}
1227
1228/**
1229 * Decode Individual Channel Stream info; reference: table 4.6.
1230 */
1231static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
1232 GetBitContext *gb)
1233{
1234 int aot = ac->oc[1].m4ac.object_type;
1235 if (aot != AOT_ER_AAC_ELD) {
1236 if (get_bits1(gb)) {
1237 av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
1238 return AVERROR_INVALIDDATA;
1239 }
1240 ics->window_sequence[1] = ics->window_sequence[0];
1241 ics->window_sequence[0] = get_bits(gb, 2);
1242 if (aot == AOT_ER_AAC_LD &&
1243 ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
1244 av_log(ac->avctx, AV_LOG_ERROR,
1245 "AAC LD is only defined for ONLY_LONG_SEQUENCE but "
1246 "window sequence %d found.\n", ics->window_sequence[0]);
1247 ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
1248 return AVERROR_INVALIDDATA;
1249 }
1250 ics->use_kb_window[1] = ics->use_kb_window[0];
1251 ics->use_kb_window[0] = get_bits1(gb);
1252 }
1253 ics->num_window_groups = 1;
1254 ics->group_len[0] = 1;
1255 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1256 int i;
1257 ics->max_sfb = get_bits(gb, 4);
1258 for (i = 0; i < 7; i++) {
1259 if (get_bits1(gb)) {
1260 ics->group_len[ics->num_window_groups - 1]++;
1261 } else {
1262 ics->num_window_groups++;
1263 ics->group_len[ics->num_window_groups - 1] = 1;
1264 }
1265 }
1266 ics->num_windows = 8;
1267 ics->swb_offset = ff_swb_offset_128[ac->oc[1].m4ac.sampling_index];
1268 ics->num_swb = ff_aac_num_swb_128[ac->oc[1].m4ac.sampling_index];
1269 ics->tns_max_bands = ff_tns_max_bands_128[ac->oc[1].m4ac.sampling_index];
1270 ics->predictor_present = 0;
1271 } else {
1272 ics->max_sfb = get_bits(gb, 6);
1273 ics->num_windows = 1;
1274 if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
1275 ics->swb_offset = ff_swb_offset_512[ac->oc[1].m4ac.sampling_index];
1276 ics->num_swb = ff_aac_num_swb_512[ac->oc[1].m4ac.sampling_index];
1277 ics->tns_max_bands = ff_tns_max_bands_512[ac->oc[1].m4ac.sampling_index];
1278 if (!ics->num_swb || !ics->swb_offset)
1279 return AVERROR_BUG;
1280 } else {
1281 ics->swb_offset = ff_swb_offset_1024[ac->oc[1].m4ac.sampling_index];
1282 ics->num_swb = ff_aac_num_swb_1024[ac->oc[1].m4ac.sampling_index];
1283 ics->tns_max_bands = ff_tns_max_bands_1024[ac->oc[1].m4ac.sampling_index];
1284 }
1285 if (aot != AOT_ER_AAC_ELD) {
1286 ics->predictor_present = get_bits1(gb);
1287 ics->predictor_reset_group = 0;
1288 }
1289 if (ics->predictor_present) {
1290 if (aot == AOT_AAC_MAIN) {
1291 if (decode_prediction(ac, ics, gb)) {
1292 goto fail;
1293 }
1294 } else if (aot == AOT_AAC_LC ||
1295 aot == AOT_ER_AAC_LC) {
1296 av_log(ac->avctx, AV_LOG_ERROR,
1297 "Prediction is not allowed in AAC-LC.\n");
1298 goto fail;
1299 } else {
1300 if (aot == AOT_ER_AAC_LD) {
1301 av_log(ac->avctx, AV_LOG_ERROR,
1302 "LTP in ER AAC LD not yet implemented.\n");
1303 return AVERROR_PATCHWELCOME;
1304 }
1305 if ((ics->ltp.present = get_bits(gb, 1)))
1306 decode_ltp(&ics->ltp, gb, ics->max_sfb);
1307 }
1308 }
1309 }
1310
1311 if (ics->max_sfb > ics->num_swb) {
1312 av_log(ac->avctx, AV_LOG_ERROR,
1313 "Number of scalefactor bands in group (%d) "
1314 "exceeds limit (%d).\n",
1315 ics->max_sfb, ics->num_swb);
1316 goto fail;
1317 }
1318
1319 return 0;
1320fail:
1321 ics->max_sfb = 0;
1322 return AVERROR_INVALIDDATA;
1323}
1324
1325/**
1326 * Decode band types (section_data payload); reference: table 4.46.
1327 *
1328 * @param band_type array of the used band type
1329 * @param band_type_run_end array of the last scalefactor band of a band type run
1330 *
1331 * @return Returns error status. 0 - OK, !0 - error
1332 */
1333static int decode_band_types(AACContext *ac, enum BandType band_type[120],
1334 int band_type_run_end[120], GetBitContext *gb,
1335 IndividualChannelStream *ics)
1336{
1337 int g, idx = 0;
1338 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
1339 for (g = 0; g < ics->num_window_groups; g++) {
1340 int k = 0;
1341 while (k < ics->max_sfb) {
1342 uint8_t sect_end = k;
1343 int sect_len_incr;
1344 int sect_band_type = get_bits(gb, 4);
1345 if (sect_band_type == 12) {
1346 av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
1347 return AVERROR_INVALIDDATA;
1348 }
1349 do {
1350 sect_len_incr = get_bits(gb, bits);
1351 sect_end += sect_len_incr;
1352 if (get_bits_left(gb) < 0) {
1353 av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
1354 return AVERROR_INVALIDDATA;
1355 }
1356 if (sect_end > ics->max_sfb) {
1357 av_log(ac->avctx, AV_LOG_ERROR,
1358 "Number of bands (%d) exceeds limit (%d).\n",
1359 sect_end, ics->max_sfb);
1360 return AVERROR_INVALIDDATA;
1361 }
1362 } while (sect_len_incr == (1 << bits) - 1);
1363 for (; k < sect_end; k++) {
1364 band_type [idx] = sect_band_type;
1365 band_type_run_end[idx++] = sect_end;
1366 }
1367 }
1368 }
1369 return 0;
1370}
1371
1372/**
1373 * Decode scalefactors; reference: table 4.47.
1374 *
1375 * @param global_gain first scalefactor value as scalefactors are differentially coded
1376 * @param band_type array of the used band type
1377 * @param band_type_run_end array of the last scalefactor band of a band type run
1378 * @param sf array of scalefactors or intensity stereo positions
1379 *
1380 * @return Returns error status. 0 - OK, !0 - error
1381 */
1382static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
1383 unsigned int global_gain,
1384 IndividualChannelStream *ics,
1385 enum BandType band_type[120],
1386 int band_type_run_end[120])
1387{
1388 int g, i, idx = 0;
1389 int offset[3] = { global_gain, global_gain - 90, 0 };
1390 int clipped_offset;
1391 int noise_flag = 1;
1392 for (g = 0; g < ics->num_window_groups; g++) {
1393 for (i = 0; i < ics->max_sfb;) {
1394 int run_end = band_type_run_end[idx];
1395 if (band_type[idx] == ZERO_BT) {
1396 for (; i < run_end; i++, idx++)
1397 sf[idx] = 0.0;
1398 } else if ((band_type[idx] == INTENSITY_BT) ||
1399 (band_type[idx] == INTENSITY_BT2)) {
1400 for (; i < run_end; i++, idx++) {
1401 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1402 clipped_offset = av_clip(offset[2], -155, 100);
1403 if (offset[2] != clipped_offset) {
1404 avpriv_request_sample(ac->avctx,
1405 "If you heard an audible artifact, there may be a bug in the decoder. "
1406 "Clipped intensity stereo position (%d -> %d)",
1407 offset[2], clipped_offset);
1408 }
1409 sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
1410 }
1411 } else if (band_type[idx] == NOISE_BT) {
1412 for (; i < run_end; i++, idx++) {
1413 if (noise_flag-- > 0)
1414 offset[1] += get_bits(gb, 9) - 256;
1415 else
1416 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1417 clipped_offset = av_clip(offset[1], -100, 155);
1418 if (offset[1] != clipped_offset) {
1419 avpriv_request_sample(ac->avctx,
1420 "If you heard an audible artifact, there may be a bug in the decoder. "
1421 "Clipped noise gain (%d -> %d)",
1422 offset[1], clipped_offset);
1423 }
1424 sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
1425 }
1426 } else {
1427 for (; i < run_end; i++, idx++) {
1428 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1429 if (offset[0] > 255U) {
1430 av_log(ac->avctx, AV_LOG_ERROR,
1431 "Scalefactor (%d) out of range.\n", offset[0]);
1432 return AVERROR_INVALIDDATA;
1433 }
1434 sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
1435 }
1436 }
1437 }
1438 }
1439 return 0;
1440}
1441
1442/**
1443 * Decode pulse data; reference: table 4.7.
1444 */
1445static int decode_pulses(Pulse *pulse, GetBitContext *gb,
1446 const uint16_t *swb_offset, int num_swb)
1447{
1448 int i, pulse_swb;
1449 pulse->num_pulse = get_bits(gb, 2) + 1;
1450 pulse_swb = get_bits(gb, 6);
1451 if (pulse_swb >= num_swb)
1452 return -1;
1453 pulse->pos[0] = swb_offset[pulse_swb];
1454 pulse->pos[0] += get_bits(gb, 5);
1455 if (pulse->pos[0] >= swb_offset[num_swb])
1456 return -1;
1457 pulse->amp[0] = get_bits(gb, 4);
1458 for (i = 1; i < pulse->num_pulse; i++) {
1459 pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
1460 if (pulse->pos[i] >= swb_offset[num_swb])
1461 return -1;
1462 pulse->amp[i] = get_bits(gb, 4);
1463 }
1464 return 0;
1465}
1466
1467/**
1468 * Decode Temporal Noise Shaping data; reference: table 4.48.
1469 *
1470 * @return Returns error status. 0 - OK, !0 - error
1471 */
1472static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
1473 GetBitContext *gb, const IndividualChannelStream *ics)
1474{
1475 int w, filt, i, coef_len, coef_res, coef_compress;
1476 const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
1477 const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
1478 for (w = 0; w < ics->num_windows; w++) {
1479 if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
1480 coef_res = get_bits1(gb);
1481
1482 for (filt = 0; filt < tns->n_filt[w]; filt++) {
1483 int tmp2_idx;
1484 tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
1485
1486 if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
1487 av_log(ac->avctx, AV_LOG_ERROR,
1488 "TNS filter order %d is greater than maximum %d.\n",
1489 tns->order[w][filt], tns_max_order);
1490 tns->order[w][filt] = 0;
1491 return AVERROR_INVALIDDATA;
1492 }
1493 if (tns->order[w][filt]) {
1494 tns->direction[w][filt] = get_bits1(gb);
1495 coef_compress = get_bits1(gb);
1496 coef_len = coef_res + 3 - coef_compress;
1497 tmp2_idx = 2 * coef_compress + coef_res;
1498
1499 for (i = 0; i < tns->order[w][filt]; i++)
1500 tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
1501 }
1502 }
1503 }
1504 }
1505 return 0;
1506}
1507
1508/**
1509 * Decode Mid/Side data; reference: table 4.54.
1510 *
1511 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
1512 * [1] mask is decoded from bitstream; [2] mask is all 1s;
1513 * [3] reserved for scalable AAC
1514 */
1515static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
1516 int ms_present)
1517{
1518 int idx;
f6fa7814 1519 int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
2ba45a60 1520 if (ms_present == 1) {
f6fa7814 1521 for (idx = 0; idx < max_idx; idx++)
2ba45a60
DM		 1522 cpe->ms_mask[idx] = get_bits1(gb);
1523 } else if (ms_present == 2) {
f6fa7814 1524 memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
2ba45a60
DM		 1525 }
1526}
1527
1528#ifndef VMUL2
1529static inline float *VMUL2(float *dst, const float *v, unsigned idx,
1530 const float *scale)
1531{
1532 float s = *scale;
1533 *dst++ = v[idx & 15] * s;
1534 *dst++ = v[idx>>4 & 15] * s;
1535 return dst;
1536}
1537#endif
1538
1539#ifndef VMUL4
1540static inline float *VMUL4(float *dst, const float *v, unsigned idx,
1541 const float *scale)
1542{
1543 float s = *scale;
1544 *dst++ = v[idx & 3] * s;
1545 *dst++ = v[idx>>2 & 3] * s;
1546 *dst++ = v[idx>>4 & 3] * s;
1547 *dst++ = v[idx>>6 & 3] * s;
1548 return dst;
1549}
1550#endif
1551
1552#ifndef VMUL2S
1553static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
1554 unsigned sign, const float *scale)
1555{
1556 union av_intfloat32 s0, s1;
1557
1558 s0.f = s1.f = *scale;
1559 s0.i ^= sign >> 1 << 31;
1560 s1.i ^= sign << 31;
1561
1562 *dst++ = v[idx & 15] * s0.f;
1563 *dst++ = v[idx>>4 & 15] * s1.f;
1564
1565 return dst;
1566}
1567#endif
1568
1569#ifndef VMUL4S
1570static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
1571 unsigned sign, const float *scale)
1572{
1573 unsigned nz = idx >> 12;
1574 union av_intfloat32 s = { .f = *scale };
1575 union av_intfloat32 t;
1576
1577 t.i = s.i ^ (sign & 1U<<31);
1578 *dst++ = v[idx & 3] * t.f;
1579
1580 sign <<= nz & 1; nz >>= 1;
1581 t.i = s.i ^ (sign & 1U<<31);
1582 *dst++ = v[idx>>2 & 3] * t.f;
1583
1584 sign <<= nz & 1; nz >>= 1;
1585 t.i = s.i ^ (sign & 1U<<31);
1586 *dst++ = v[idx>>4 & 3] * t.f;
1587
1588 sign <<= nz & 1;
1589 t.i = s.i ^ (sign & 1U<<31);
1590 *dst++ = v[idx>>6 & 3] * t.f;
1591
1592 return dst;
1593}
1594#endif
1595
1596/**
1597 * Decode spectral data; reference: table 4.50.
1598 * Dequantize and scale spectral data; reference: 4.6.3.3.
1599 *
1600 * @param coef array of dequantized, scaled spectral data
1601 * @param sf array of scalefactors or intensity stereo positions
1602 * @param pulse_present set if pulses are present
1603 * @param pulse pointer to pulse data struct
1604 * @param band_type array of the used band type
1605 *
1606 * @return Returns error status. 0 - OK, !0 - error
1607 */
1608static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
1609 GetBitContext *gb, const float sf[120],
1610 int pulse_present, const Pulse *pulse,
1611 const IndividualChannelStream *ics,
1612 enum BandType band_type[120])
1613{
1614 int i, k, g, idx = 0;
1615 const int c = 1024 / ics->num_windows;
1616 const uint16_t *offsets = ics->swb_offset;
1617 float *coef_base = coef;
1618
1619 for (g = 0; g < ics->num_windows; g++)
1620 memset(coef + g * 128 + offsets[ics->max_sfb], 0,
1621 sizeof(float) * (c - offsets[ics->max_sfb]));
1622
1623 for (g = 0; g < ics->num_window_groups; g++) {
1624 unsigned g_len = ics->group_len[g];
1625
1626 for (i = 0; i < ics->max_sfb; i++, idx++) {
1627 const unsigned cbt_m1 = band_type[idx] - 1;
1628 float *cfo = coef + offsets[i];
1629 int off_len = offsets[i + 1] - offsets[i];
1630 int group;
1631
1632 if (cbt_m1 >= INTENSITY_BT2 - 1) {
1633 for (group = 0; group < g_len; group++, cfo+=128) {
1634 memset(cfo, 0, off_len * sizeof(float));
1635 }
1636 } else if (cbt_m1 == NOISE_BT - 1) {
1637 for (group = 0; group < g_len; group++, cfo+=128) {
1638 float scale;
1639 float band_energy;
1640
1641 for (k = 0; k < off_len; k++) {
1642 ac->random_state = lcg_random(ac->random_state);
1643 cfo[k] = ac->random_state;
1644 }
1645
f6fa7814 1646 band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
2ba45a60 1647 scale = sf[idx] / sqrtf(band_energy);
f6fa7814 1648 ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
2ba45a60
DM		 1649 }
1650 } else {
1651 const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1652 const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1653 VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1654 OPEN_READER(re, gb);
1655
1656 switch (cbt_m1 >> 1) {
1657 case 0:
1658 for (group = 0; group < g_len; group++, cfo+=128) {
1659 float *cf = cfo;
1660 int len = off_len;
1661
1662 do {
1663 int code;
1664 unsigned cb_idx;
1665
1666 UPDATE_CACHE(re, gb);
1667 GET_VLC(code, re, gb, vlc_tab, 8, 2);
1668 cb_idx = cb_vector_idx[code];
1669 cf = VMUL4(cf, vq, cb_idx, sf + idx);
1670 } while (len -= 4);
1671 }
1672 break;
1673
1674 case 1:
1675 for (group = 0; group < g_len; group++, cfo+=128) {
1676 float *cf = cfo;
1677 int len = off_len;
1678
1679 do {
1680 int code;
1681 unsigned nnz;
1682 unsigned cb_idx;
1683 uint32_t bits;
1684
1685 UPDATE_CACHE(re, gb);
1686 GET_VLC(code, re, gb, vlc_tab, 8, 2);
1687 cb_idx = cb_vector_idx[code];
1688 nnz = cb_idx >> 8 & 15;
1689 bits = nnz ? GET_CACHE(re, gb) : 0;
1690 LAST_SKIP_BITS(re, gb, nnz);
1691 cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1692 } while (len -= 4);
1693 }
1694 break;
1695
1696 case 2:
1697 for (group = 0; group < g_len; group++, cfo+=128) {
1698 float *cf = cfo;
1699 int len = off_len;
1700
1701 do {
1702 int code;
1703 unsigned cb_idx;
1704
1705 UPDATE_CACHE(re, gb);
1706 GET_VLC(code, re, gb, vlc_tab, 8, 2);
1707 cb_idx = cb_vector_idx[code];
1708 cf = VMUL2(cf, vq, cb_idx, sf + idx);
1709 } while (len -= 2);
1710 }
1711 break;
1712
1713 case 3:
1714 case 4:
1715 for (group = 0; group < g_len; group++, cfo+=128) {
1716 float *cf = cfo;
1717 int len = off_len;
1718
1719 do {
1720 int code;
1721 unsigned nnz;
1722 unsigned cb_idx;
1723 unsigned sign;
1724
1725 UPDATE_CACHE(re, gb);
1726 GET_VLC(code, re, gb, vlc_tab, 8, 2);
1727 cb_idx = cb_vector_idx[code];
1728 nnz = cb_idx >> 8 & 15;
1729 sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
1730 LAST_SKIP_BITS(re, gb, nnz);
1731 cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1732 } while (len -= 2);
1733 }
1734 break;
1735
1736 default:
1737 for (group = 0; group < g_len; group++, cfo+=128) {
1738 float *cf = cfo;
1739 uint32_t *icf = (uint32_t *) cf;
1740 int len = off_len;
1741
1742 do {
1743 int code;
1744 unsigned nzt, nnz;
1745 unsigned cb_idx;
1746 uint32_t bits;
1747 int j;
1748
1749 UPDATE_CACHE(re, gb);
1750 GET_VLC(code, re, gb, vlc_tab, 8, 2);
1751
1752 if (!code) {
1753 *icf++ = 0;
1754 *icf++ = 0;
1755 continue;
1756 }
1757
1758 cb_idx = cb_vector_idx[code];
1759 nnz = cb_idx >> 12;
1760 nzt = cb_idx >> 8;
1761 bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1762 LAST_SKIP_BITS(re, gb, nnz);
1763
1764 for (j = 0; j < 2; j++) {
1765 if (nzt & 1<<j) {
1766 uint32_t b;
1767 int n;
1768 /* The total length of escape_sequence must be < 22 bits according
1769 to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1770 UPDATE_CACHE(re, gb);
1771 b = GET_CACHE(re, gb);
1772 b = 31 - av_log2(~b);
1773
1774 if (b > 8) {
1775 av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1776 return AVERROR_INVALIDDATA;
1777 }
1778
1779 SKIP_BITS(re, gb, b + 1);
1780 b += 4;
1781 n = (1 << b) + SHOW_UBITS(re, gb, b);
1782 LAST_SKIP_BITS(re, gb, b);
1783 *icf++ = cbrt_tab[n] | (bits & 1U<<31);
1784 bits <<= 1;
1785 } else {
1786 unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1787 *icf++ = (bits & 1U<<31) | v;
1788 bits <<= !!v;
1789 }
1790 cb_idx >>= 4;
1791 }
1792 } while (len -= 2);
1793
f6fa7814 1794 ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
2ba45a60
DM		 1795 }
1796 }
1797
1798 CLOSE_READER(re, gb);
1799 }
1800 }
1801 coef += g_len << 7;
1802 }
1803
1804 if (pulse_present) {
1805 idx = 0;
1806 for (i = 0; i < pulse->num_pulse; i++) {
1807 float co = coef_base[ pulse->pos[i] ];
1808 while (offsets[idx + 1] <= pulse->pos[i])
1809 idx++;
1810 if (band_type[idx] != NOISE_BT && sf[idx]) {
1811 float ico = -pulse->amp[i];
1812 if (co) {
1813 co /= sf[idx];
1814 ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1815 }
1816 coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1817 }
1818 }
1819 }
1820 return 0;
1821}
1822
1823static av_always_inline float flt16_round(float pf)
1824{
1825 union av_intfloat32 tmp;
1826 tmp.f = pf;
1827 tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
1828 return tmp.f;
1829}
1830
1831static av_always_inline float flt16_even(float pf)
1832{
1833 union av_intfloat32 tmp;
1834 tmp.f = pf;
1835 tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
1836 return tmp.f;
1837}
1838
1839static av_always_inline float flt16_trunc(float pf)
1840{
1841 union av_intfloat32 pun;
1842 pun.f = pf;
1843 pun.i &= 0xFFFF0000U;
1844 return pun.f;
1845}
1846
1847static av_always_inline void predict(PredictorState *ps, float *coef,
1848 int output_enable)
1849{
1850 const float a = 0.953125; // 61.0 / 64
1851 const float alpha = 0.90625; // 29.0 / 32
1852 float e0, e1;
1853 float pv;
1854 float k1, k2;
1855 float r0 = ps->r0, r1 = ps->r1;
1856 float cor0 = ps->cor0, cor1 = ps->cor1;
1857 float var0 = ps->var0, var1 = ps->var1;
1858
1859 k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
1860 k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
1861
1862 pv = flt16_round(k1 * r0 + k2 * r1);
1863 if (output_enable)
1864 *coef += pv;
1865
1866 e0 = *coef;
1867 e1 = e0 - k1 * r0;
1868
1869 ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
1870 ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
1871 ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
1872 ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
1873
1874 ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
1875 ps->r0 = flt16_trunc(a * e0);
1876}
1877
1878/**
1879 * Apply AAC-Main style frequency domain prediction.
1880 */
1881static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1882{
1883 int sfb, k;
1884
1885 if (!sce->ics.predictor_initialized) {
1886 reset_all_predictors(sce->predictor_state);
1887 sce->ics.predictor_initialized = 1;
1888 }
1889
1890 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1891 for (sfb = 0;
1892 sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
1893 sfb++) {
1894 for (k = sce->ics.swb_offset[sfb];
1895 k < sce->ics.swb_offset[sfb + 1];
1896 k++) {
1897 predict(&sce->predictor_state[k], &sce->coeffs[k],
1898 sce->ics.predictor_present &&
1899 sce->ics.prediction_used[sfb]);
1900 }
1901 }
1902 if (sce->ics.predictor_reset_group)
1903 reset_predictor_group(sce->predictor_state,
1904 sce->ics.predictor_reset_group);
1905 } else
1906 reset_all_predictors(sce->predictor_state);
1907}
1908
1909/**
1910 * Decode an individual_channel_stream payload; reference: table 4.44.
1911 *
1912 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
1913 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1914 *
1915 * @return Returns error status. 0 - OK, !0 - error
1916 */
1917static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1918 GetBitContext *gb, int common_window, int scale_flag)
1919{
1920 Pulse pulse;
1921 TemporalNoiseShaping *tns = &sce->tns;
1922 IndividualChannelStream *ics = &sce->ics;
1923 float *out = sce->coeffs;
1924 int global_gain, eld_syntax, er_syntax, pulse_present = 0;
1925 int ret;
1926
1927 eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1928 er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
1929 ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
1930 ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
1931 ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1932
1933 /* This assignment is to silence a GCC warning about the variable being used
1934 * uninitialized when in fact it always is.
1935 */
1936 pulse.num_pulse = 0;
1937
1938 global_gain = get_bits(gb, 8);
1939
1940 if (!common_window && !scale_flag) {
1941 if (decode_ics_info(ac, ics, gb) < 0)
1942 return AVERROR_INVALIDDATA;
1943 }
1944
1945 if ((ret = decode_band_types(ac, sce->band_type,
1946 sce->band_type_run_end, gb, ics)) < 0)
1947 return ret;
1948 if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
1949 sce->band_type, sce->band_type_run_end)) < 0)
1950 return ret;
1951
1952 pulse_present = 0;
1953 if (!scale_flag) {
1954 if (!eld_syntax && (pulse_present = get_bits1(gb))) {
1955 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1956 av_log(ac->avctx, AV_LOG_ERROR,
1957 "Pulse tool not allowed in eight short sequence.\n");
1958 return AVERROR_INVALIDDATA;
1959 }
1960 if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1961 av_log(ac->avctx, AV_LOG_ERROR,
1962 "Pulse data corrupt or invalid.\n");
1963 return AVERROR_INVALIDDATA;
1964 }
1965 }
1966 tns->present = get_bits1(gb);
1967 if (tns->present && !er_syntax)
1968 if (decode_tns(ac, tns, gb, ics) < 0)
1969 return AVERROR_INVALIDDATA;
1970 if (!eld_syntax && get_bits1(gb)) {
1971 avpriv_request_sample(ac->avctx, "SSR");
1972 return AVERROR_PATCHWELCOME;
1973 }
1974 // I see no textual basis in the spec for this occurring after SSR gain
1975 // control, but this is what both reference and real implmentations do
1976 if (tns->present && er_syntax)
1977 if (decode_tns(ac, tns, gb, ics) < 0)
1978 return AVERROR_INVALIDDATA;
1979 }
1980
1981 if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
1982 &pulse, ics, sce->band_type) < 0)
1983 return AVERROR_INVALIDDATA;
1984
1985 if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
1986 apply_prediction(ac, sce);
1987
1988 return 0;
1989}
1990
1991/**
1992 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1993 */
1994static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1995{
1996 const IndividualChannelStream *ics = &cpe->ch[0].ics;
1997 float *ch0 = cpe->ch[0].coeffs;
1998 float *ch1 = cpe->ch[1].coeffs;
1999 int g, i, group, idx = 0;
2000 const uint16_t *offsets = ics->swb_offset;
2001 for (g = 0; g < ics->num_window_groups; g++) {
2002 for (i = 0; i < ics->max_sfb; i++, idx++) {
2003 if (cpe->ms_mask[idx] &&
2004 cpe->ch[0].band_type[idx] < NOISE_BT &&
2005 cpe->ch[1].band_type[idx] < NOISE_BT) {
2006 for (group = 0; group < ics->group_len[g]; group++) {
f6fa7814 2007 ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
2ba45a60
DM		 2008 ch1 + group * 128 + offsets[i],
2009 offsets[i+1] - offsets[i]);
2010 }
2011 }
2012 }
2013 ch0 += ics->group_len[g] * 128;
2014 ch1 += ics->group_len[g] * 128;
2015 }
2016}
2017
2018/**
2019 * intensity stereo decoding; reference: 4.6.8.2.3
2020 *
2021 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
2022 * [1] mask is decoded from bitstream; [2] mask is all 1s;
2023 * [3] reserved for scalable AAC
2024 */
2025static void apply_intensity_stereo(AACContext *ac,
2026 ChannelElement *cpe, int ms_present)
2027{
2028 const IndividualChannelStream *ics = &cpe->ch[1].ics;
2029 SingleChannelElement *sce1 = &cpe->ch[1];
2030 float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
2031 const uint16_t *offsets = ics->swb_offset;
2032 int g, group, i, idx = 0;
2033 int c;
2034 float scale;
2035 for (g = 0; g < ics->num_window_groups; g++) {
2036 for (i = 0; i < ics->max_sfb;) {
2037 if (sce1->band_type[idx] == INTENSITY_BT ||
2038 sce1->band_type[idx] == INTENSITY_BT2) {
2039 const int bt_run_end = sce1->band_type_run_end[idx];
2040 for (; i < bt_run_end; i++, idx++) {
2041 c = -1 + 2 * (sce1->band_type[idx] - 14);
2042 if (ms_present)
2043 c *= 1 - 2 * cpe->ms_mask[idx];
2044 scale = c * sce1->sf[idx];
2045 for (group = 0; group < ics->group_len[g]; group++)
f6fa7814 2046 ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
2ba45a60
DM		 2047 coef0 + group * 128 + offsets[i],
2048 scale,
2049 offsets[i + 1] - offsets[i]);
2050 }
2051 } else {
2052 int bt_run_end = sce1->band_type_run_end[idx];
2053 idx += bt_run_end - i;
2054 i = bt_run_end;
2055 }
2056 }
2057 coef0 += ics->group_len[g] * 128;
2058 coef1 += ics->group_len[g] * 128;
2059 }
2060}
2061
2062/**
2063 * Decode a channel_pair_element; reference: table 4.4.
2064 *
2065 * @return Returns error status. 0 - OK, !0 - error
2066 */
2067static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
2068{
2069 int i, ret, common_window, ms_present = 0;
2070 int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2071
2072 common_window = eld_syntax || get_bits1(gb);
2073 if (common_window) {
2074 if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
2075 return AVERROR_INVALIDDATA;
2076 i = cpe->ch[1].ics.use_kb_window[0];
2077 cpe->ch[1].ics = cpe->ch[0].ics;
2078 cpe->ch[1].ics.use_kb_window[1] = i;
2079 if (cpe->ch[1].ics.predictor_present &&
2080 (ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
2081 if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
2082 decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
2083 ms_present = get_bits(gb, 2);
2084 if (ms_present == 3) {
2085 av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
2086 return AVERROR_INVALIDDATA;
2087 } else if (ms_present)
2088 decode_mid_side_stereo(cpe, gb, ms_present);
2089 }
2090 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
2091 return ret;
2092 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
2093 return ret;
2094
2095 if (common_window) {
2096 if (ms_present)
2097 apply_mid_side_stereo(ac, cpe);
2098 if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
2099 apply_prediction(ac, &cpe->ch[0]);
2100 apply_prediction(ac, &cpe->ch[1]);
2101 }
2102 }
2103
2104 apply_intensity_stereo(ac, cpe, ms_present);
2105 return 0;
2106}
2107
2108static const float cce_scale[] = {
2109 1.09050773266525765921, //2^(1/8)
2110 1.18920711500272106672, //2^(1/4)
2111 M_SQRT2,
2112 2,
2113};
2114
2115/**
2116 * Decode coupling_channel_element; reference: table 4.8.
2117 *
2118 * @return Returns error status. 0 - OK, !0 - error
2119 */
2120static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
2121{
2122 int num_gain = 0;
2123 int c, g, sfb, ret;
2124 int sign;
2125 float scale;
2126 SingleChannelElement *sce = &che->ch[0];
2127 ChannelCoupling *coup = &che->coup;
2128
2129 coup->coupling_point = 2 * get_bits1(gb);
2130 coup->num_coupled = get_bits(gb, 3);
2131 for (c = 0; c <= coup->num_coupled; c++) {
2132 num_gain++;
2133 coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
2134 coup->id_select[c] = get_bits(gb, 4);
2135 if (coup->type[c] == TYPE_CPE) {
2136 coup->ch_select[c] = get_bits(gb, 2);
2137 if (coup->ch_select[c] == 3)
2138 num_gain++;
2139 } else
2140 coup->ch_select[c] = 2;
2141 }
2142 coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
2143
2144 sign = get_bits(gb, 1);
2145 scale = cce_scale[get_bits(gb, 2)];
2146
2147 if ((ret = decode_ics(ac, sce, gb, 0, 0)))
2148 return ret;
2149
2150 for (c = 0; c < num_gain; c++) {
2151 int idx = 0;
2152 int cge = 1;
2153 int gain = 0;
2154 float gain_cache = 1.0;
2155 if (c) {
2156 cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
2157 gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
2158 gain_cache = powf(scale, -gain);
2159 }
2160 if (coup->coupling_point == AFTER_IMDCT) {
2161 coup->gain[c][0] = gain_cache;
2162 } else {
2163 for (g = 0; g < sce->ics.num_window_groups; g++) {
2164 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
2165 if (sce->band_type[idx] != ZERO_BT) {
2166 if (!cge) {
2167 int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
2168 if (t) {
2169 int s = 1;
2170 t = gain += t;
2171 if (sign) {
2172 s -= 2 * (t & 0x1);
2173 t >>= 1;
2174 }
2175 gain_cache = powf(scale, -t) * s;
2176 }
2177 }
2178 coup->gain[c][idx] = gain_cache;
2179 }
2180 }
2181 }
2182 }
2183 }
2184 return 0;
2185}
2186
2187/**
2188 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
2189 *
2190 * @return Returns number of bytes consumed.
2191 */
2192static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
2193 GetBitContext *gb)
2194{
2195 int i;
2196 int num_excl_chan = 0;
2197
2198 do {
2199 for (i = 0; i < 7; i++)
2200 che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
2201 } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
2202
2203 return num_excl_chan / 7;
2204}
2205
2206/**
2207 * Decode dynamic range information; reference: table 4.52.
2208 *
2209 * @return Returns number of bytes consumed.
2210 */
2211static int decode_dynamic_range(DynamicRangeControl *che_drc,
2212 GetBitContext *gb)
2213{
2214 int n = 1;
2215 int drc_num_bands = 1;
2216 int i;
2217
2218 /* pce_tag_present? */
2219 if (get_bits1(gb)) {
2220 che_drc->pce_instance_tag = get_bits(gb, 4);
2221 skip_bits(gb, 4); // tag_reserved_bits
2222 n++;
2223 }
2224
2225 /* excluded_chns_present? */
2226 if (get_bits1(gb)) {
2227 n += decode_drc_channel_exclusions(che_drc, gb);
2228 }
2229
2230 /* drc_bands_present? */
2231 if (get_bits1(gb)) {
2232 che_drc->band_incr = get_bits(gb, 4);
2233 che_drc->interpolation_scheme = get_bits(gb, 4);
2234 n++;
2235 drc_num_bands += che_drc->band_incr;
2236 for (i = 0; i < drc_num_bands; i++) {
2237 che_drc->band_top[i] = get_bits(gb, 8);
2238 n++;
2239 }
2240 }
2241
2242 /* prog_ref_level_present? */
2243 if (get_bits1(gb)) {
2244 che_drc->prog_ref_level = get_bits(gb, 7);
2245 skip_bits1(gb); // prog_ref_level_reserved_bits
2246 n++;
2247 }
2248
2249 for (i = 0; i < drc_num_bands; i++) {
2250 che_drc->dyn_rng_sgn[i] = get_bits1(gb);
2251 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
2252 n++;
2253 }
2254
2255 return n;
2256}
2257
2258static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
2259 uint8_t buf[256];
2260 int i, major, minor;
2261
2262 if (len < 13+7*8)
2263 goto unknown;
2264
2265 get_bits(gb, 13); len -= 13;
2266
2267 for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
2268 buf[i] = get_bits(gb, 8);
2269
2270 buf[i] = 0;
2271 if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
2272 av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
2273
2274 if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
2275 ac->avctx->internal->skip_samples = 1024;
2276 }
2277
2278unknown:
2279 skip_bits_long(gb, len);
2280
2281 return 0;
2282}
2283
2284/**
2285 * Decode extension data (incomplete); reference: table 4.51.
2286 *
2287 * @param cnt length of TYPE_FIL syntactic element in bytes
2288 *
2289 * @return Returns number of bytes consumed
2290 */
2291static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
2292 ChannelElement *che, enum RawDataBlockType elem_type)
2293{
2294 int crc_flag = 0;
2295 int res = cnt;
f6fa7814
DM		 2296 int type = get_bits(gb, 4);
2297
2298 if (ac->avctx->debug & FF_DEBUG_STARTCODE)
2299 av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
2300
2301 switch (type) { // extension type
2ba45a60
DM		 2302 case EXT_SBR_DATA_CRC:
2303 crc_flag++;
2304 case EXT_SBR_DATA:
2305 if (!che) {
2306 av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
2307 return res;
2308 } else if (!ac->oc[1].m4ac.sbr) {
2309 av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
2310 skip_bits_long(gb, 8 * cnt - 4);
2311 return res;
2312 } else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
2313 av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
2314 skip_bits_long(gb, 8 * cnt - 4);
2315 return res;
2316 } else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
2317 ac->oc[1].m4ac.sbr = 1;
2318 ac->oc[1].m4ac.ps = 1;
2319 ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
2320 output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
2321 ac->oc[1].status, 1);
2322 } else {
2323 ac->oc[1].m4ac.sbr = 1;
2324 ac->avctx->profile = FF_PROFILE_AAC_HE;
2325 }
2326 res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
2327 break;
2328 case EXT_DYNAMIC_RANGE:
2329 res = decode_dynamic_range(&ac->che_drc, gb);
2330 break;
2331 case EXT_FILL:
2332 decode_fill(ac, gb, 8 * cnt - 4);
2333 break;
2334 case EXT_FILL_DATA:
2335 case EXT_DATA_ELEMENT:
2336 default:
2337 skip_bits_long(gb, 8 * cnt - 4);
2338 break;
2339 };
2340 return res;
2341}
2342
2343/**
2344 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
2345 *
2346 * @param decode 1 if tool is used normally, 0 if tool is used in LTP.
2347 * @param coef spectral coefficients
2348 */
2349static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
2350 IndividualChannelStream *ics, int decode)
2351{
2352 const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
2353 int w, filt, m, i;
2354 int bottom, top, order, start, end, size, inc;
2355 float lpc[TNS_MAX_ORDER];
2356 float tmp[TNS_MAX_ORDER+1];
2357
2358 for (w = 0; w < ics->num_windows; w++) {
2359 bottom = ics->num_swb;
2360 for (filt = 0; filt < tns->n_filt[w]; filt++) {
2361 top = bottom;
2362 bottom = FFMAX(0, top - tns->length[w][filt]);
2363 order = tns->order[w][filt];
2364 if (order == 0)
2365 continue;
2366
2367 // tns_decode_coef
2368 compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
2369
2370 start = ics->swb_offset[FFMIN(bottom, mmm)];
2371 end = ics->swb_offset[FFMIN( top, mmm)];
2372 if ((size = end - start) <= 0)
2373 continue;
2374 if (tns->direction[w][filt]) {
2375 inc = -1;
2376 start = end - 1;
2377 } else {
2378 inc = 1;
2379 }
2380 start += w * 128;
2381
2382 if (decode) {
2383 // ar filter
2384 for (m = 0; m < size; m++, start += inc)
2385 for (i = 1; i <= FFMIN(m, order); i++)
2386 coef[start] -= coef[start - i * inc] * lpc[i - 1];
2387 } else {
2388 // ma filter
2389 for (m = 0; m < size; m++, start += inc) {
2390 tmp[0] = coef[start];
2391 for (i = 1; i <= FFMIN(m, order); i++)
2392 coef[start] += tmp[i] * lpc[i - 1];
2393 for (i = order; i > 0; i--)
2394 tmp[i] = tmp[i - 1];
2395 }
2396 }
2397 }
2398 }
2399}
2400
2401/**
2402 * Apply windowing and MDCT to obtain the spectral
2403 * coefficient from the predicted sample by LTP.
2404 */
2405static void windowing_and_mdct_ltp(AACContext *ac, float *out,
2406 float *in, IndividualChannelStream *ics)
2407{
2408 const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
2409 const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
2410 const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
2411 const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
2412
2413 if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
f6fa7814 2414 ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
2ba45a60
DM		 2415 } else {
2416 memset(in, 0, 448 * sizeof(float));
f6fa7814 2417 ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
2ba45a60
DM		 2418 }
2419 if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
f6fa7814 2420 ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
2ba45a60 2421 } else {
f6fa7814 2422 ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
2ba45a60
DM		 2423 memset(in + 1024 + 576, 0, 448 * sizeof(float));
2424 }
2425 ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
2426}
2427
2428/**
2429 * Apply the long term prediction
2430 */
2431static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
2432{
2433 const LongTermPrediction *ltp = &sce->ics.ltp;
2434 const uint16_t *offsets = sce->ics.swb_offset;
2435 int i, sfb;
2436
2437 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
2438 float *predTime = sce->ret;
2439 float *predFreq = ac->buf_mdct;
2440 int16_t num_samples = 2048;
2441
2442 if (ltp->lag < 1024)
2443 num_samples = ltp->lag + 1024;
2444 for (i = 0; i < num_samples; i++)
2445 predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
2446 memset(&predTime[i], 0, (2048 - i) * sizeof(float));
2447
2448 ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
2449
2450 if (sce->tns.present)
2451 ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
2452
2453 for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
2454 if (ltp->used[sfb])
2455 for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
2456 sce->coeffs[i] += predFreq[i];
2457 }
2458}
2459
2460/**
2461 * Update the LTP buffer for next frame
2462 */
2463static void update_ltp(AACContext *ac, SingleChannelElement *sce)
2464{
2465 IndividualChannelStream *ics = &sce->ics;
2466 float *saved = sce->saved;
2467 float *saved_ltp = sce->coeffs;
2468 const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
2469 const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
2470 int i;
2471
2472 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2473 memcpy(saved_ltp, saved, 512 * sizeof(float));
2474 memset(saved_ltp + 576, 0, 448 * sizeof(float));
f6fa7814 2475 ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2ba45a60
DM		 2476 for (i = 0; i < 64; i++)
2477 saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
2478 } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2479 memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(float));
2480 memset(saved_ltp + 576, 0, 448 * sizeof(float));
f6fa7814 2481 ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2ba45a60
DM		 2482 for (i = 0; i < 64; i++)
2483 saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
2484 } else { // LONG_STOP or ONLY_LONG
f6fa7814 2485 ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
2ba45a60
DM		 2486 for (i = 0; i < 512; i++)
2487 saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
2488 }
2489
2490 memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
2491 memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
2492 memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
2493}
2494
2495/**
2496 * Conduct IMDCT and windowing.
2497 */
2498static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
2499{
2500 IndividualChannelStream *ics = &sce->ics;
2501 float *in = sce->coeffs;
2502 float *out = sce->ret;
2503 float *saved = sce->saved;
2504 const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
2505 const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
2506 const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
2507 float *buf = ac->buf_mdct;
2508 float *temp = ac->temp;
2509 int i;
2510
2511 // imdct
2512 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2513 for (i = 0; i < 1024; i += 128)
2514 ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
2515 } else
2516 ac->mdct.imdct_half(&ac->mdct, buf, in);
2517
2518 /* window overlapping
2519 * NOTE: To simplify the overlapping code, all 'meaningless' short to long
2520 * and long to short transitions are considered to be short to short
2521 * transitions. This leaves just two cases (long to long and short to short)
2522 * with a little special sauce for EIGHT_SHORT_SEQUENCE.
2523 */
2524 if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2525 (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
f6fa7814 2526 ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
2ba45a60
DM		 2527 } else {
2528 memcpy( out, saved, 448 * sizeof(float));
2529
2530 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
f6fa7814
DM		 2531 ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
2532 ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
2533 ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
2534 ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
2535 ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
2ba45a60
DM		 2536 memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
2537 } else {
f6fa7814 2538 ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
2ba45a60
DM		 2539 memcpy( out + 576, buf + 64, 448 * sizeof(float));
2540 }
2541 }
2542
2543 // buffer update
2544 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2545 memcpy( saved, temp + 64, 64 * sizeof(float));
f6fa7814
DM		 2546 ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
2547 ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
2548 ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
2ba45a60
DM		 2549 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
2550 } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2551 memcpy( saved, buf + 512, 448 * sizeof(float));
2552 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
2553 } else { // LONG_STOP or ONLY_LONG
2554 memcpy( saved, buf + 512, 512 * sizeof(float));
2555 }
2556}
2557
2558static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
2559{
2560 IndividualChannelStream *ics = &sce->ics;
2561 float *in = sce->coeffs;
2562 float *out = sce->ret;
2563 float *saved = sce->saved;
2564 float *buf = ac->buf_mdct;
2565
2566 // imdct
2567 ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2568
2569 // window overlapping
2570 if (ics->use_kb_window[1]) {
2571 // AAC LD uses a low overlap sine window instead of a KBD window
2572 memcpy(out, saved, 192 * sizeof(float));
f6fa7814 2573 ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, ff_sine_128, 64);
2ba45a60
DM		 2574 memcpy( out + 320, buf + 64, 192 * sizeof(float));
2575 } else {
f6fa7814 2576 ac->fdsp->vector_fmul_window(out, saved, buf, ff_sine_512, 256);
2ba45a60
DM		 2577 }
2578
2579 // buffer update
2580 memcpy(saved, buf + 256, 256 * sizeof(float));
2581}
2582
2583static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
2584{
2585 float *in = sce->coeffs;
2586 float *out = sce->ret;
2587 float *saved = sce->saved;
2588 const float *const window = ff_aac_eld_window;
2589 float *buf = ac->buf_mdct;
2590 int i;
2591 const int n = 512;
2592 const int n2 = n >> 1;
2593 const int n4 = n >> 2;
2594
2595 // Inverse transform, mapped to the conventional IMDCT by
2596 // Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
2597 // "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
2598 // International Conference on Audio, Language and Image Processing, ICALIP 2008.
2599 // URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
2600 for (i = 0; i < n2; i+=2) {
2601 float temp;
2602 temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
2603 temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
2604 }
2605 ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2606 for (i = 0; i < n; i+=2) {
2607 buf[i] = -buf[i];
2608 }
2609 // Like with the regular IMDCT at this point we still have the middle half
2610 // of a transform but with even symmetry on the left and odd symmetry on
2611 // the right
2612
2613 // window overlapping
2614 // The spec says to use samples [0..511] but the reference decoder uses
2615 // samples [128..639].
2616 for (i = n4; i < n2; i ++) {
2617 out[i - n4] = buf[n2 - 1 - i] * window[i - n4] +
2618 saved[ i + n2] * window[i + n - n4] +
2619 -saved[ n + n2 - 1 - i] * window[i + 2*n - n4] +
2620 -saved[2*n + n2 + i] * window[i + 3*n - n4];
2621 }
2622 for (i = 0; i < n2; i ++) {
2623 out[n4 + i] = buf[i] * window[i + n2 - n4] +
2624 -saved[ n - 1 - i] * window[i + n2 + n - n4] +
2625 -saved[ n + i] * window[i + n2 + 2*n - n4] +
2626 saved[2*n + n - 1 - i] * window[i + n2 + 3*n - n4];
2627 }
2628 for (i = 0; i < n4; i ++) {
2629 out[n2 + n4 + i] = buf[ i + n2] * window[i + n - n4] +
2630 -saved[ n2 - 1 - i] * window[i + 2*n - n4] +
2631 -saved[ n + n2 + i] * window[i + 3*n - n4];
2632 }
2633
2634 // buffer update
2635 memmove(saved + n, saved, 2 * n * sizeof(float));
2636 memcpy( saved, buf, n * sizeof(float));
2637}
2638
2639/**
2640 * Apply dependent channel coupling (applied before IMDCT).
2641 *
2642 * @param index index into coupling gain array
2643 */
2644static void apply_dependent_coupling(AACContext *ac,
2645 SingleChannelElement *target,
2646 ChannelElement *cce, int index)
2647{
2648 IndividualChannelStream *ics = &cce->ch[0].ics;
2649 const uint16_t *offsets = ics->swb_offset;
2650 float *dest = target->coeffs;
2651 const float *src = cce->ch[0].coeffs;
2652 int g, i, group, k, idx = 0;
2653 if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2654 av_log(ac->avctx, AV_LOG_ERROR,
2655 "Dependent coupling is not supported together with LTP\n");
2656 return;
2657 }
2658 for (g = 0; g < ics->num_window_groups; g++) {
2659 for (i = 0; i < ics->max_sfb; i++, idx++) {
2660 if (cce->ch[0].band_type[idx] != ZERO_BT) {
2661 const float gain = cce->coup.gain[index][idx];
2662 for (group = 0; group < ics->group_len[g]; group++) {
2663 for (k = offsets[i]; k < offsets[i + 1]; k++) {
2664 // FIXME: SIMDify
2665 dest[group * 128 + k] += gain * src[group * 128 + k];
2666 }
2667 }
2668 }
2669 }
2670 dest += ics->group_len[g] * 128;
2671 src += ics->group_len[g] * 128;
2672 }
2673}
2674
2675/**
2676 * Apply independent channel coupling (applied after IMDCT).
2677 *
2678 * @param index index into coupling gain array
2679 */
2680static void apply_independent_coupling(AACContext *ac,
2681 SingleChannelElement *target,
2682 ChannelElement *cce, int index)
2683{
2684 int i;
2685 const float gain = cce->coup.gain[index][0];
2686 const float *src = cce->ch[0].ret;
2687 float *dest = target->ret;
2688 const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
2689
2690 for (i = 0; i < len; i++)
2691 dest[i] += gain * src[i];
2692}
2693
2694/**
2695 * channel coupling transformation interface
2696 *
2697 * @param apply_coupling_method pointer to (in)dependent coupling function
2698 */
2699static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
2700 enum RawDataBlockType type, int elem_id,
2701 enum CouplingPoint coupling_point,
2702 void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
2703{
2704 int i, c;
2705
2706 for (i = 0; i < MAX_ELEM_ID; i++) {
2707 ChannelElement *cce = ac->che[TYPE_CCE][i];
2708 int index = 0;
2709
2710 if (cce && cce->coup.coupling_point == coupling_point) {
2711 ChannelCoupling *coup = &cce->coup;
2712
2713 for (c = 0; c <= coup->num_coupled; c++) {
2714 if (coup->type[c] == type && coup->id_select[c] == elem_id) {
2715 if (coup->ch_select[c] != 1) {
2716 apply_coupling_method(ac, &cc->ch[0], cce, index);
2717 if (coup->ch_select[c] != 0)
2718 index++;
2719 }
2720 if (coup->ch_select[c] != 2)
2721 apply_coupling_method(ac, &cc->ch[1], cce, index++);
2722 } else
2723 index += 1 + (coup->ch_select[c] == 3);
2724 }
2725 }
2726 }
2727}
2728
2729/**
2730 * Convert spectral data to float samples, applying all supported tools as appropriate.
2731 */
2732static void spectral_to_sample(AACContext *ac)
2733{
2734 int i, type;
2735 void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
2736 switch (ac->oc[1].m4ac.object_type) {
2737 case AOT_ER_AAC_LD:
2738 imdct_and_window = imdct_and_windowing_ld;
2739 break;
2740 case AOT_ER_AAC_ELD:
2741 imdct_and_window = imdct_and_windowing_eld;
2742 break;
2743 default:
2744 imdct_and_window = ac->imdct_and_windowing;
2745 }
2746 for (type = 3; type >= 0; type--) {
2747 for (i = 0; i < MAX_ELEM_ID; i++) {
2748 ChannelElement *che = ac->che[type][i];
f6fa7814 2749 if (che && che->present) {
2ba45a60
DM		 2750 if (type <= TYPE_CPE)
2751 apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
2752 if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2753 if (che->ch[0].ics.predictor_present) {
2754 if (che->ch[0].ics.ltp.present)
2755 ac->apply_ltp(ac, &che->ch[0]);
2756 if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
2757 ac->apply_ltp(ac, &che->ch[1]);
2758 }
2759 }
2760 if (che->ch[0].tns.present)
2761 ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2762 if (che->ch[1].tns.present)
2763 ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2764 if (type <= TYPE_CPE)
2765 apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
2766 if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2767 imdct_and_window(ac, &che->ch[0]);
2768 if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2769 ac->update_ltp(ac, &che->ch[0]);
2770 if (type == TYPE_CPE) {
2771 imdct_and_window(ac, &che->ch[1]);
2772 if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2773 ac->update_ltp(ac, &che->ch[1]);
2774 }
2775 if (ac->oc[1].m4ac.sbr > 0) {
2776 ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2777 }
2778 }
2779 if (type <= TYPE_CCE)
2780 apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
f6fa7814
DM		 2781 che->present = 0;
2782 } else if (che) {
2783 av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
2ba45a60
DM		 2784 }
2785 }
2786 }
2787}
2788
2789static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
2790{
2791 int size;
2792 AACADTSHeaderInfo hdr_info;
2793 uint8_t layout_map[MAX_ELEM_ID*4][3];
2794 int layout_map_tags, ret;
2795
2796 size = avpriv_aac_parse_header(gb, &hdr_info);
2797 if (size > 0) {
2798 if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
2799 // This is 2 for "VLB " audio in NSV files.
2800 // See samples/nsv/vlb_audio.
2801 avpriv_report_missing_feature(ac->avctx,
2802 "More than one AAC RDB per ADTS frame");
2803 ac->warned_num_aac_frames = 1;
2804 }
2805 push_output_configuration(ac);
2806 if (hdr_info.chan_config) {
2807 ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
2808 if ((ret = set_default_channel_config(ac->avctx,
2809 layout_map,
2810 &layout_map_tags,
2811 hdr_info.chan_config)) < 0)
2812 return ret;
2813 if ((ret = output_configure(ac, layout_map, layout_map_tags,
2814 FFMAX(ac->oc[1].status,
2815 OC_TRIAL_FRAME), 0)) < 0)
2816 return ret;
2817 } else {
2818 ac->oc[1].m4ac.chan_config = 0;
2819 /**
2820 * dual mono frames in Japanese DTV can have chan_config 0
2821 * WITHOUT specifying PCE.
2822 * thus, set dual mono as default.
2823 */
2824 if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
2825 layout_map_tags = 2;
2826 layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
2827 layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
2828 layout_map[0][1] = 0;
2829 layout_map[1][1] = 1;
2830 if (output_configure(ac, layout_map, layout_map_tags,
2831 OC_TRIAL_FRAME, 0))
2832 return -7;
2833 }
2834 }
2835 ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
2836 ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
2837 ac->oc[1].m4ac.object_type = hdr_info.object_type;
2838 if (ac->oc[0].status != OC_LOCKED ||
2839 ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
2840 ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
2841 ac->oc[1].m4ac.sbr = -1;
2842 ac->oc[1].m4ac.ps = -1;
2843 }
2844 if (!hdr_info.crc_absent)
2845 skip_bits(gb, 16);
2846 }
2847 return size;
2848}
2849
2850static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
2851 int *got_frame_ptr, GetBitContext *gb)
2852{
2853 AACContext *ac = avctx->priv_data;
2854 ChannelElement *che;
2855 int err, i;
2856 int samples = 1024;
2857 int chan_config = ac->oc[1].m4ac.chan_config;
2858 int aot = ac->oc[1].m4ac.object_type;
2859
2860 if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
2861 samples >>= 1;
2862
2863 ac->frame = data;
2864
2865 if ((err = frame_configure_elements(avctx)) < 0)
2866 return err;
2867
2868 // The FF_PROFILE_AAC_* defines are all object_type - 1
2869 // This may lead to an undefined profile being signaled
2870 ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
2871
2872 ac->tags_mapped = 0;
2873
2874 if (chan_config < 0 || chan_config >= 8) {
2875 avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
2876 ac->oc[1].m4ac.chan_config);
2877 return AVERROR_INVALIDDATA;
2878 }
2879 for (i = 0; i < tags_per_config[chan_config]; i++) {
2880 const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
2881 const int elem_id = aac_channel_layout_map[chan_config-1][i][1];
2882 if (!(che=get_che(ac, elem_type, elem_id))) {
2883 av_log(ac->avctx, AV_LOG_ERROR,
2884 "channel element %d.%d is not allocated\n",
2885 elem_type, elem_id);
2886 return AVERROR_INVALIDDATA;
2887 }
f6fa7814 2888 che->present = 1;
2ba45a60
DM		 2889 if (aot != AOT_ER_AAC_ELD)
2890 skip_bits(gb, 4);
2891 switch (elem_type) {
2892 case TYPE_SCE:
2893 err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2894 break;
2895 case TYPE_CPE:
2896 err = decode_cpe(ac, gb, che);
2897 break;
2898 case TYPE_LFE:
2899 err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2900 break;
2901 }
2902 if (err < 0)
2903 return err;
2904 }
2905
2906 spectral_to_sample(ac);
2907
2908 ac->frame->nb_samples = samples;
2909 ac->frame->sample_rate = avctx->sample_rate;
2910 *got_frame_ptr = 1;
2911
2912 skip_bits_long(gb, get_bits_left(gb));
2913 return 0;
2914}
2915
2916static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
2917 int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
2918{
2919 AACContext *ac = avctx->priv_data;
2920 ChannelElement *che = NULL, *che_prev = NULL;
2921 enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
2922 int err, elem_id;
2923 int samples = 0, multiplier, audio_found = 0, pce_found = 0;
2924 int is_dmono, sce_count = 0;
2925
2926 ac->frame = data;
2927
2928 if (show_bits(gb, 12) == 0xfff) {
2929 if ((err = parse_adts_frame_header(ac, gb)) < 0) {
2930 av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
2931 goto fail;
2932 }
2933 if (ac->oc[1].m4ac.sampling_index > 12) {
2934 av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
2935 err = AVERROR_INVALIDDATA;
2936 goto fail;
2937 }
2938 }
2939
2940 if ((err = frame_configure_elements(avctx)) < 0)
2941 goto fail;
2942
2943 // The FF_PROFILE_AAC_* defines are all object_type - 1
2944 // This may lead to an undefined profile being signaled
2945 ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
2946
2947 ac->tags_mapped = 0;
2948 // parse
2949 while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
2950 elem_id = get_bits(gb, 4);
2951
f6fa7814
DM		 2952 if (avctx->debug & FF_DEBUG_STARTCODE)
2953 av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
2954
2ba45a60
DM		 2955 if (elem_type < TYPE_DSE) {
2956 if (!(che=get_che(ac, elem_type, elem_id))) {
2957 av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
2958 elem_type, elem_id);
2959 err = AVERROR_INVALIDDATA;
2960 goto fail;
2961 }
2962 samples = 1024;
f6fa7814 2963 che->present = 1;
2ba45a60
DM		 2964 }
2965
2966 switch (elem_type) {
2967
2968 case TYPE_SCE:
2969 err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2970 audio_found = 1;
2971 sce_count++;
2972 break;
2973
2974 case TYPE_CPE:
2975 err = decode_cpe(ac, gb, che);
2976 audio_found = 1;
2977 break;
2978
2979 case TYPE_CCE:
2980 err = decode_cce(ac, gb, che);
2981 break;
2982
2983 case TYPE_LFE:
2984 err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2985 audio_found = 1;
2986 break;
2987
2988 case TYPE_DSE:
2989 err = skip_data_stream_element(ac, gb);
2990 break;
2991
2992 case TYPE_PCE: {
2993 uint8_t layout_map[MAX_ELEM_ID*4][3];
2994 int tags;
2995 push_output_configuration(ac);
2996 tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb);
2997 if (tags < 0) {
2998 err = tags;
2999 break;
3000 }
3001 if (pce_found) {
3002 av_log(avctx, AV_LOG_ERROR,
3003 "Not evaluating a further program_config_element as this construct is dubious at best.\n");
3004 } else {
3005 err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
3006 if (!err)
3007 ac->oc[1].m4ac.chan_config = 0;
3008 pce_found = 1;
3009 }
3010 break;
3011 }
3012
3013 case TYPE_FIL:
3014 if (elem_id == 15)
3015 elem_id += get_bits(gb, 8) - 1;
3016 if (get_bits_left(gb) < 8 * elem_id) {
3017 av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
3018 err = AVERROR_INVALIDDATA;
3019 goto fail;
3020 }
3021 while (elem_id > 0)
3022 elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
3023 err = 0; /* FIXME */
3024 break;
3025
3026 default:
3027 err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
3028 break;
3029 }
3030
3031 che_prev = che;
3032 elem_type_prev = elem_type;
3033
3034 if (err)
3035 goto fail;
3036
3037 if (get_bits_left(gb) < 3) {
3038 av_log(avctx, AV_LOG_ERROR, overread_err);
3039 err = AVERROR_INVALIDDATA;
3040 goto fail;
3041 }
3042 }
3043
3044 spectral_to_sample(ac);
3045
3046 multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
3047 samples <<= multiplier;
3048
3049 if (ac->oc[1].status && audio_found) {
3050 avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
3051 avctx->frame_size = samples;
3052 ac->oc[1].status = OC_LOCKED;
3053 }
3054
3055 if (multiplier) {
3056 int side_size;
3057 const uint8_t *side = av_packet_get_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, &side_size);
3058 if (side && side_size>=4)
3059 AV_WL32(side, 2*AV_RL32(side));
3060 }
3061
3062 *got_frame_ptr = !!samples;
3063 if (samples) {
3064 ac->frame->nb_samples = samples;
3065 ac->frame->sample_rate = avctx->sample_rate;
3066 } else
3067 av_frame_unref(ac->frame);
3068 *got_frame_ptr = !!samples;
3069
3070 /* for dual-mono audio (SCE + SCE) */
3071 is_dmono = ac->dmono_mode && sce_count == 2 &&
3072 ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
3073 if (is_dmono) {
3074 if (ac->dmono_mode == 1)
3075 ((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
3076 else if (ac->dmono_mode == 2)
3077 ((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
3078 }
3079
3080 return 0;
3081fail:
3082 pop_output_configuration(ac);
3083 return err;
3084}
3085
3086static int aac_decode_frame(AVCodecContext *avctx, void *data,
3087 int *got_frame_ptr, AVPacket *avpkt)
3088{
3089 AACContext *ac = avctx->priv_data;
3090 const uint8_t *buf = avpkt->data;
3091 int buf_size = avpkt->size;
3092 GetBitContext gb;
3093 int buf_consumed;
3094 int buf_offset;
3095 int err;
3096 int new_extradata_size;
3097 const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
3098 AV_PKT_DATA_NEW_EXTRADATA,
3099 &new_extradata_size);
3100 int jp_dualmono_size;
3101 const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
3102 AV_PKT_DATA_JP_DUALMONO,
3103 &jp_dualmono_size);
3104
3105 if (new_extradata && 0) {
3106 av_free(avctx->extradata);
3107 avctx->extradata = av_mallocz(new_extradata_size +
3108 FF_INPUT_BUFFER_PADDING_SIZE);
3109 if (!avctx->extradata)
3110 return AVERROR(ENOMEM);
3111 avctx->extradata_size = new_extradata_size;
3112 memcpy(avctx->extradata, new_extradata, new_extradata_size);
3113 push_output_configuration(ac);
3114 if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
3115 avctx->extradata,
3116 avctx->extradata_size*8, 1) < 0) {
3117 pop_output_configuration(ac);
3118 return AVERROR_INVALIDDATA;
3119 }
3120 }
3121
3122 ac->dmono_mode = 0;
3123 if (jp_dualmono && jp_dualmono_size > 0)
3124 ac->dmono_mode = 1 + *jp_dualmono;
3125 if (ac->force_dmono_mode >= 0)
3126 ac->dmono_mode = ac->force_dmono_mode;
3127
3128 if (INT_MAX / 8 <= buf_size)
3129 return AVERROR_INVALIDDATA;
3130
3131 if ((err = init_get_bits(&gb, buf, buf_size * 8)) < 0)
3132 return err;
3133
3134 switch (ac->oc[1].m4ac.object_type) {
3135 case AOT_ER_AAC_LC:
3136 case AOT_ER_AAC_LTP:
3137 case AOT_ER_AAC_LD:
3138 case AOT_ER_AAC_ELD:
3139 err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
3140 break;
3141 default:
3142 err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
3143 }
3144 if (err < 0)
3145 return err;
3146
3147 buf_consumed = (get_bits_count(&gb) + 7) >> 3;
3148 for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
3149 if (buf[buf_offset])
3150 break;
3151
3152 return buf_size > buf_offset ? buf_consumed : buf_size;
3153}
3154
3155static av_cold int aac_decode_close(AVCodecContext *avctx)
3156{
3157 AACContext *ac = avctx->priv_data;
3158 int i, type;
3159
3160 for (i = 0; i < MAX_ELEM_ID; i++) {
3161 for (type = 0; type < 4; type++) {
3162 if (ac->che[type][i])
3163 ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
3164 av_freep(&ac->che[type][i]);
3165 }
3166 }
3167
3168 ff_mdct_end(&ac->mdct);
3169 ff_mdct_end(&ac->mdct_small);
3170 ff_mdct_end(&ac->mdct_ld);
3171 ff_mdct_end(&ac->mdct_ltp);
f6fa7814 3172 av_freep(&ac->fdsp);
2ba45a60
DM		 3173 return 0;
3174}
3175
3176
3177#define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
3178
3179struct LATMContext {
3180 AACContext aac_ctx; ///< containing AACContext
3181 int initialized; ///< initialized after a valid extradata was seen
3182
3183 // parser data
3184 int audio_mux_version_A; ///< LATM syntax version
3185 int frame_length_type; ///< 0/1 variable/fixed frame length
3186 int frame_length; ///< frame length for fixed frame length
3187};
3188
3189static inline uint32_t latm_get_value(GetBitContext *b)
3190{
3191 int length = get_bits(b, 2);
3192
3193 return get_bits_long(b, (length+1)*8);
3194}
3195
3196static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
3197 GetBitContext *gb, int asclen)
3198{
3199 AACContext *ac = &latmctx->aac_ctx;
3200 AVCodecContext *avctx = ac->avctx;
3201 MPEG4AudioConfig m4ac = { 0 };
3202 int config_start_bit = get_bits_count(gb);
3203 int sync_extension = 0;
3204 int bits_consumed, esize;
3205
3206 if (asclen) {
3207 sync_extension = 1;
3208 asclen = FFMIN(asclen, get_bits_left(gb));
3209 } else
3210 asclen = get_bits_left(gb);
3211
3212 if (config_start_bit % 8) {
3213 avpriv_request_sample(latmctx->aac_ctx.avctx,
3214 "Non-byte-aligned audio-specific config");
3215 return AVERROR_PATCHWELCOME;
3216 }
3217 if (asclen <= 0)
3218 return AVERROR_INVALIDDATA;
3219 bits_consumed = decode_audio_specific_config(NULL, avctx, &m4ac,
3220 gb->buffer + (config_start_bit / 8),
3221 asclen, sync_extension);
3222
3223 if (bits_consumed < 0)
3224 return AVERROR_INVALIDDATA;
3225
3226 if (!latmctx->initialized ||
3227 ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
3228 ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
3229
3230 if(latmctx->initialized) {
3231 av_log(avctx, AV_LOG_INFO, "audio config changed\n");
3232 } else {
3233 av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
3234 }
3235 latmctx->initialized = 0;
3236
3237 esize = (bits_consumed+7) / 8;
3238
3239 if (avctx->extradata_size < esize) {
3240 av_free(avctx->extradata);
3241 avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
3242 if (!avctx->extradata)
3243 return AVERROR(ENOMEM);
3244 }
3245
3246 avctx->extradata_size = esize;
3247 memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
3248 memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
3249 }
3250 skip_bits_long(gb, bits_consumed);
3251
3252 return bits_consumed;
3253}
3254
3255static int read_stream_mux_config(struct LATMContext *latmctx,
3256 GetBitContext *gb)
3257{
3258 int ret, audio_mux_version = get_bits(gb, 1);
3259
3260 latmctx->audio_mux_version_A = 0;
3261 if (audio_mux_version)
3262 latmctx->audio_mux_version_A = get_bits(gb, 1);
3263
3264 if (!latmctx->audio_mux_version_A) {
3265
3266 if (audio_mux_version)
3267 latm_get_value(gb); // taraFullness
3268
3269 skip_bits(gb, 1); // allStreamSameTimeFraming
3270 skip_bits(gb, 6); // numSubFrames
3271 // numPrograms
3272 if (get_bits(gb, 4)) { // numPrograms
3273 avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
3274 return AVERROR_PATCHWELCOME;
3275 }
3276
3277 // for each program (which there is only one in DVB)
3278
3279 // for each layer (which there is only one in DVB)
3280 if (get_bits(gb, 3)) { // numLayer
3281 avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
3282 return AVERROR_PATCHWELCOME;
3283 }
3284
3285 // for all but first stream: use_same_config = get_bits(gb, 1);
3286 if (!audio_mux_version) {
3287 if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
3288 return ret;
3289 } else {
3290 int ascLen = latm_get_value(gb);
3291 if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
3292 return ret;
3293 ascLen -= ret;
3294 skip_bits_long(gb, ascLen);
3295 }
3296
3297 latmctx->frame_length_type = get_bits(gb, 3);
3298 switch (latmctx->frame_length_type) {
3299 case 0:
3300 skip_bits(gb, 8); // latmBufferFullness
3301 break;
3302 case 1:
3303 latmctx->frame_length = get_bits(gb, 9);
3304 break;
3305 case 3:
3306 case 4:
3307 case 5:
3308 skip_bits(gb, 6); // CELP frame length table index
3309 break;
3310 case 6:
3311 case 7:
3312 skip_bits(gb, 1); // HVXC frame length table index
3313 break;
3314 }
3315
3316 if (get_bits(gb, 1)) { // other data
3317 if (audio_mux_version) {
3318 latm_get_value(gb); // other_data_bits
3319 } else {
3320 int esc;
3321 do {
3322 esc = get_bits(gb, 1);
3323 skip_bits(gb, 8);
3324 } while (esc);
3325 }
3326 }
3327
3328 if (get_bits(gb, 1)) // crc present
3329 skip_bits(gb, 8); // config_crc
3330 }
3331
3332 return 0;
3333}
3334
3335static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
3336{
3337 uint8_t tmp;
3338
3339 if (ctx->frame_length_type == 0) {
3340 int mux_slot_length = 0;
3341 do {
3342 tmp = get_bits(gb, 8);
3343 mux_slot_length += tmp;
3344 } while (tmp == 255);
3345 return mux_slot_length;
3346 } else if (ctx->frame_length_type == 1) {
3347 return ctx->frame_length;
3348 } else if (ctx->frame_length_type == 3 ||
3349 ctx->frame_length_type == 5 ||
3350 ctx->frame_length_type == 7) {
3351 skip_bits(gb, 2); // mux_slot_length_coded
3352 }
3353 return 0;
3354}
3355
3356static int read_audio_mux_element(struct LATMContext *latmctx,
3357 GetBitContext *gb)
3358{
3359 int err;
3360 uint8_t use_same_mux = get_bits(gb, 1);
3361 if (!use_same_mux) {
3362 if ((err = read_stream_mux_config(latmctx, gb)) < 0)
3363 return err;
3364 } else if (!latmctx->aac_ctx.avctx->extradata) {
3365 av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
3366 "no decoder config found\n");
3367 return AVERROR(EAGAIN);
3368 }
3369 if (latmctx->audio_mux_version_A == 0) {
3370 int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
3371 if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
3372 av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
3373 return AVERROR_INVALIDDATA;
3374 } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
3375 av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
3376 "frame length mismatch %d << %d\n",
3377 mux_slot_length_bytes * 8, get_bits_left(gb));
3378 return AVERROR_INVALIDDATA;
3379 }
3380 }
3381 return 0;
3382}
3383
3384
3385static int latm_decode_frame(AVCodecContext *avctx, void *out,
3386 int *got_frame_ptr, AVPacket *avpkt)
3387{
3388 struct LATMContext *latmctx = avctx->priv_data;
3389 int muxlength, err;
3390 GetBitContext gb;
3391
3392 if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
3393 return err;
3394
3395 // check for LOAS sync word
3396 if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
3397 return AVERROR_INVALIDDATA;
3398
3399 muxlength = get_bits(&gb, 13) + 3;
3400 // not enough data, the parser should have sorted this out
3401 if (muxlength > avpkt->size)
3402 return AVERROR_INVALIDDATA;
3403
3404 if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
3405 return err;
3406
3407 if (!latmctx->initialized) {
3408 if (!avctx->extradata) {
3409 *got_frame_ptr = 0;
3410 return avpkt->size;
3411 } else {
3412 push_output_configuration(&latmctx->aac_ctx);
3413 if ((err = decode_audio_specific_config(
3414 &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
3415 avctx->extradata, avctx->extradata_size*8, 1)) < 0) {
3416 pop_output_configuration(&latmctx->aac_ctx);
3417 return err;
3418 }
3419 latmctx->initialized = 1;
3420 }
3421 }
3422
3423 if (show_bits(&gb, 12) == 0xfff) {
3424 av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
3425 "ADTS header detected, probably as result of configuration "
3426 "misparsing\n");
3427 return AVERROR_INVALIDDATA;
3428 }
3429
3430 if ((err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt)) < 0)
3431 return err;
3432
3433 return muxlength;
3434}
3435
3436static av_cold int latm_decode_init(AVCodecContext *avctx)
3437{
3438 struct LATMContext *latmctx = avctx->priv_data;
3439 int ret = aac_decode_init(avctx);
3440
3441 if (avctx->extradata_size > 0)
3442 latmctx->initialized = !ret;
3443
3444 return ret;
3445}
3446
3447static void aacdec_init(AACContext *c)
3448{
3449 c->imdct_and_windowing = imdct_and_windowing;
3450 c->apply_ltp = apply_ltp;
3451 c->apply_tns = apply_tns;
3452 c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
3453 c->update_ltp = update_ltp;
3454
3455 if(ARCH_MIPS)
3456 ff_aacdec_init_mips(c);
3457}
3458/**
3459 * AVOptions for Japanese DTV specific extensions (ADTS only)
3460 */
3461#define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
3462static const AVOption options[] = {
3463 {"dual_mono_mode", "Select the channel to decode for dual mono",
3464 offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
3465 AACDEC_FLAGS, "dual_mono_mode"},
3466
3467 {"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3468 {"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3469 {"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3470 {"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3471
3472 {NULL},
3473};
3474
3475static const AVClass aac_decoder_class = {
3476 .class_name = "AAC decoder",
3477 .item_name = av_default_item_name,
3478 .option = options,
3479 .version = LIBAVUTIL_VERSION_INT,
3480};
3481
f6fa7814
DM		 3482static const AVProfile profiles[] = {
3483 { FF_PROFILE_AAC_MAIN, "Main" },
3484 { FF_PROFILE_AAC_LOW, "LC" },
3485 { FF_PROFILE_AAC_SSR, "SSR" },
3486 { FF_PROFILE_AAC_LTP, "LTP" },
3487 { FF_PROFILE_AAC_HE, "HE-AAC" },
3488 { FF_PROFILE_AAC_HE_V2, "HE-AACv2" },
3489 { FF_PROFILE_AAC_LD, "LD" },
3490 { FF_PROFILE_AAC_ELD, "ELD" },
3491 { FF_PROFILE_UNKNOWN },
3492};
3493
2ba45a60
DM		 3494AVCodec ff_aac_decoder = {
3495 .name = "aac",
3496 .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
3497 .type = AVMEDIA_TYPE_AUDIO,
3498 .id = AV_CODEC_ID_AAC,
3499 .priv_data_size = sizeof(AACContext),
3500 .init = aac_decode_init,
3501 .close = aac_decode_close,
3502 .decode = aac_decode_frame,
3503 .sample_fmts = (const enum AVSampleFormat[]) {
3504 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
3505 },
3506 .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
3507 .channel_layouts = aac_channel_layout,
3508 .flush = flush,
3509 .priv_class = &aac_decoder_class,
f6fa7814 3510 .profiles = profiles,
2ba45a60
DM		 3511};
3512
3513/*
3514 Note: This decoder filter is intended to decode LATM streams transferred
3515 in MPEG transport streams which only contain one program.
3516 To do a more complex LATM demuxing a separate LATM demuxer should be used.
3517*/
3518AVCodec ff_aac_latm_decoder = {
3519 .name = "aac_latm",
3520 .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
3521 .type = AVMEDIA_TYPE_AUDIO,
3522 .id = AV_CODEC_ID_AAC_LATM,
3523 .priv_data_size = sizeof(struct LATMContext),
3524 .init = latm_decode_init,
3525 .close = aac_decode_close,
3526 .decode = latm_decode_frame,
3527 .sample_fmts = (const enum AVSampleFormat[]) {
3528 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
3529 },
3530 .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1,
3531 .channel_layouts = aac_channel_layout,
3532 .flush = flush,
f6fa7814 3533 .profiles = profiles,
2ba45a60 3534};
ffmpeg Piment Noir Debian package repository