GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavcodec/aacdec.c Lines: 198 263 75.3 %
Date: 2019-11-18 18:00:01 Branches: 74 129 57.4 %

Line Branch Exec Source
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
#define FFT_FLOAT 1
36
#define FFT_FIXED_32 0
37
#define USE_FIXED 0
38
39
#include "libavutil/float_dsp.h"
40
#include "libavutil/opt.h"
41
#include "avcodec.h"
42
#include "internal.h"
43
#include "get_bits.h"
44
#include "fft.h"
45
#include "mdct15.h"
46
#include "lpc.h"
47
#include "kbdwin.h"
48
#include "sinewin.h"
49
50
#include "aac.h"
51
#include "aactab.h"
52
#include "aacdectab.h"
53
#include "adts_header.h"
54
#include "cbrt_data.h"
55
#include "sbr.h"
56
#include "aacsbr.h"
57
#include "mpeg4audio.h"
58
#include "profiles.h"
59
#include "libavutil/intfloat.h"
60
61
#include <errno.h>
62
#include <math.h>
63
#include <stdint.h>
64
#include <string.h>
65
66
#if ARCH_ARM
67
#   include "arm/aac.h"
68
#elif ARCH_MIPS
69
#   include "mips/aacdec_mips.h"
70
#endif
71
72
38505
static av_always_inline void reset_predict_state(PredictorState *ps)
73
{
74
38505
    ps->r0   = 0.0f;
75
38505
    ps->r1   = 0.0f;
76
38505
    ps->cor0 = 0.0f;
77
38505
    ps->cor1 = 0.0f;
78
38505
    ps->var0 = 1.0f;
79
38505
    ps->var1 = 1.0f;
80
38505
}
81
82
#ifndef VMUL2
83
1836424
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
84
                           const float *scale)
85
{
86
1836424
    float s = *scale;
87
1836424
    *dst++ = v[idx    & 15] * s;
88
1836424
    *dst++ = v[idx>>4 & 15] * s;
89
1836424
    return dst;
90
}
91
#endif
92
93
#ifndef VMUL4
94
2256735
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
95
                           const float *scale)
96
{
97
2256735
    float s = *scale;
98
2256735
    *dst++ = v[idx    & 3] * s;
99
2256735
    *dst++ = v[idx>>2 & 3] * s;
100
2256735
    *dst++ = v[idx>>4 & 3] * s;
101
2256735
    *dst++ = v[idx>>6 & 3] * s;
102
2256735
    return dst;
103
}
104
#endif
105
106
#ifndef VMUL2S
107
2262756
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
108
                            unsigned sign, const float *scale)
109
{
110
    union av_intfloat32 s0, s1;
111
112
2262756
    s0.f = s1.f = *scale;
113
2262756
    s0.i ^= sign >> 1 << 31;
114
2262756
    s1.i ^= sign      << 31;
115
116
2262756
    *dst++ = v[idx    & 15] * s0.f;
117
2262756
    *dst++ = v[idx>>4 & 15] * s1.f;
118
119
2262756
    return dst;
120
}
121
#endif
122
123
#ifndef VMUL4S
124
1821575
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
125
                            unsigned sign, const float *scale)
126
{
127
1821575
    unsigned nz = idx >> 12;
128
1821575
    union av_intfloat32 s = { .f = *scale };
129
    union av_intfloat32 t;
130
131
1821575
    t.i = s.i ^ (sign & 1U<<31);
132
1821575
    *dst++ = v[idx    & 3] * t.f;
133
134
1821575
    sign <<= nz & 1; nz >>= 1;
135
1821575
    t.i = s.i ^ (sign & 1U<<31);
136
1821575
    *dst++ = v[idx>>2 & 3] * t.f;
137
138
1821575
    sign <<= nz & 1; nz >>= 1;
139
1821575
    t.i = s.i ^ (sign & 1U<<31);
140
1821575
    *dst++ = v[idx>>4 & 3] * t.f;
141
142
1821575
    sign <<= nz & 1;
143
1821575
    t.i = s.i ^ (sign & 1U<<31);
144
1821575
    *dst++ = v[idx>>6 & 3] * t.f;
145
146
1821575
    return dst;
147
}
148
#endif
149
150
4047936
static av_always_inline float flt16_round(float pf)
151
{
152
    union av_intfloat32 tmp;
153
4047936
    tmp.f = pf;
154
4047936
    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
155
4047936
    return tmp.f;
156
}
157
158
5083656
static av_always_inline float flt16_even(float pf)
159
{
160
    union av_intfloat32 tmp;
161
5083656
    tmp.f = pf;
162
5083656
    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
163
5083656
    return tmp.f;
164
}
165
166
24287616
static av_always_inline float flt16_trunc(float pf)
167
{
168
    union av_intfloat32 pun;
169
24287616
    pun.f = pf;
170
24287616
    pun.i &= 0xFFFF0000U;
171
24287616
    return pun.f;
172
}
173
174
4047936
static av_always_inline void predict(PredictorState *ps, float *coef,
175
                                     int output_enable)
176
{
177
4047936
    const float a     = 0.953125; // 61.0 / 64
178
4047936
    const float alpha = 0.90625;  // 29.0 / 32
179
    float e0, e1;
180
    float pv;
181
    float k1, k2;
182
4047936
    float   r0 = ps->r0,     r1 = ps->r1;
183
4047936
    float cor0 = ps->cor0, cor1 = ps->cor1;
184
4047936
    float var0 = ps->var0, var1 = ps->var1;
185
186
4047936
    k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
187
4047936
    k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
188
189
4047936
    pv = flt16_round(k1 * r0 + k2 * r1);
190
4047936
    if (output_enable)
191
272888
        *coef += pv;
192
193
4047936
    e0 = *coef;
194
4047936
    e1 = e0 - k1 * r0;
195
196
4047936
    ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
197
4047936
    ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
198
4047936
    ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
199
4047936
    ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
200
201
4047936
    ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
202
4047936
    ps->r0 = flt16_trunc(a * e0);
203
4047936
}
204
205
/**
206
 * Apply dependent channel coupling (applied before IMDCT).
207
 *
208
 * @param   index   index into coupling gain array
209
 */
210
2743
static void apply_dependent_coupling(AACContext *ac,
211
                                     SingleChannelElement *target,
212
                                     ChannelElement *cce, int index)
213
{
214
2743
    IndividualChannelStream *ics = &cce->ch[0].ics;
215
2743
    const uint16_t *offsets = ics->swb_offset;
216
2743
    float *dest = target->coeffs;
217
2743
    const float *src = cce->ch[0].coeffs;
218
2743
    int g, i, group, k, idx = 0;
219
2743
    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
220
        av_log(ac->avctx, AV_LOG_ERROR,
221
               "Dependent coupling is not supported together with LTP\n");
222
        return;
223
    }
224
5486
    for (g = 0; g < ics->num_window_groups; g++) {
225
112463
        for (i = 0; i < ics->max_sfb; i++, idx++) {
226
109720
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
227
5196
                const float gain = cce->coup.gain[index][idx];
228
10392
                for (group = 0; group < ics->group_len[g]; group++) {
229
81964
                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
230
                        // FIXME: SIMDify
231
76768
                        dest[group * 128 + k] += gain * src[group * 128 + k];
232
                    }
233
                }
234
            }
235
        }
236
2743
        dest += ics->group_len[g] * 128;
237
2743
        src  += ics->group_len[g] * 128;
238
    }
239
}
240
241
/**
242
 * Apply independent channel coupling (applied after IMDCT).
243
 *
244
 * @param   index   index into coupling gain array
245
 */
246
1315
static void apply_independent_coupling(AACContext *ac,
247
                                       SingleChannelElement *target,
248
                                       ChannelElement *cce, int index)
249
{
250
1315
    const float gain = cce->coup.gain[index][0];
251
1315
    const float *src = cce->ch[0].ret;
252
1315
    float *dest = target->ret;
253
1315
    const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
254
255
1315
    ac->fdsp->vector_fmac_scalar(dest, src, gain, len);
256
1315
}
257
258
#include "aacdec_template.c"
259
260
#define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
261
262
struct LATMContext {
263
    AACContext aac_ctx;     ///< containing AACContext
264
    int initialized;        ///< initialized after a valid extradata was seen
265
266
    // parser data
267
    int audio_mux_version_A; ///< LATM syntax version
268
    int frame_length_type;   ///< 0/1 variable/fixed frame length
269
    int frame_length;        ///< frame length for fixed frame length
270
};
271
272
static inline uint32_t latm_get_value(GetBitContext *b)
273
{
274
    int length = get_bits(b, 2);
275
276
    return get_bits_long(b, (length+1)*8);
277
}
278
279
321
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
280
                                             GetBitContext *gb, int asclen)
281
{
282
321
    AACContext *ac        = &latmctx->aac_ctx;
283
321
    AVCodecContext *avctx = ac->avctx;
284
321
    MPEG4AudioConfig m4ac = { 0 };
285
    GetBitContext gbc;
286
321
    int config_start_bit  = get_bits_count(gb);
287
321
    int sync_extension    = 0;
288
    int bits_consumed, esize, i;
289
290
321
    if (asclen > 0) {
291
        sync_extension = 1;
292
        asclen         = FFMIN(asclen, get_bits_left(gb));
293
        init_get_bits(&gbc, gb->buffer, config_start_bit + asclen);
294
        skip_bits_long(&gbc, config_start_bit);
295
321
    } else if (asclen == 0) {
296
321
        gbc = *gb;
297
    } else {
298
        return AVERROR_INVALIDDATA;
299
    }
300
301
321
    if (get_bits_left(gb) <= 0)
302
        return AVERROR_INVALIDDATA;
303
304
321
    bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac,
305
                                                    &gbc, config_start_bit,
306
                                                    sync_extension);
307
308
321
    if (bits_consumed < config_start_bit)
309
        return AVERROR_INVALIDDATA;
310
321
    bits_consumed -= config_start_bit;
311
312
321
    if (asclen == 0)
313
321
      asclen = bits_consumed;
314
315
321
    if (!latmctx->initialized ||
316
316
        ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
317
316
        ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
318
319
6
        if (latmctx->initialized) {
320
1
            av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config);
321
        } else {
322
5
            av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
323
        }
324
6
        latmctx->initialized = 0;
325
326
6
        esize = (asclen + 7) / 8;
327
328
6
        if (avctx->extradata_size < esize) {
329
5
            av_free(avctx->extradata);
330
5
            avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE);
331
5
            if (!avctx->extradata)
332
                return AVERROR(ENOMEM);
333
        }
334
335
6
        avctx->extradata_size = esize;
336
6
        gbc = *gb;
337
25
        for (i = 0; i < esize; i++) {
338
19
          avctx->extradata[i] = get_bits(&gbc, 8);
339
        }
340
6
        memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE);
341
    }
342
321
    skip_bits_long(gb, asclen);
343
344
321
    return 0;
345
}
346
347
321
static int read_stream_mux_config(struct LATMContext *latmctx,
348
                                  GetBitContext *gb)
349
{
350
321
    int ret, audio_mux_version = get_bits(gb, 1);
351
352
321
    latmctx->audio_mux_version_A = 0;
353
321
    if (audio_mux_version)
354
        latmctx->audio_mux_version_A = get_bits(gb, 1);
355
356
321
    if (!latmctx->audio_mux_version_A) {
357
358
321
        if (audio_mux_version)
359
            latm_get_value(gb);                 // taraFullness
360
361
321
        skip_bits(gb, 1);                       // allStreamSameTimeFraming
362
321
        skip_bits(gb, 6);                       // numSubFrames
363
        // numPrograms
364
321
        if (get_bits(gb, 4)) {                  // numPrograms
365
            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
366
            return AVERROR_PATCHWELCOME;
367
        }
368
369
        // for each program (which there is only one in DVB)
370
371
        // for each layer (which there is only one in DVB)
372
321
        if (get_bits(gb, 3)) {                   // numLayer
373
            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
374
            return AVERROR_PATCHWELCOME;
375
        }
376
377
        // for all but first stream: use_same_config = get_bits(gb, 1);
378
321
        if (!audio_mux_version) {
379
321
            if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
380
                return ret;
381
        } else {
382
            int ascLen = latm_get_value(gb);
383
            if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
384
                return ret;
385
        }
386
387
321
        latmctx->frame_length_type = get_bits(gb, 3);
388

321
        switch (latmctx->frame_length_type) {
389
321
        case 0:
390
321
            skip_bits(gb, 8);       // latmBufferFullness
391
321
            break;
392
        case 1:
393
            latmctx->frame_length = get_bits(gb, 9);
394
            break;
395
        case 3:
396
        case 4:
397
        case 5:
398
            skip_bits(gb, 6);       // CELP frame length table index
399
            break;
400
        case 6:
401
        case 7:
402
            skip_bits(gb, 1);       // HVXC frame length table index
403
            break;
404
        }
405
406
321
        if (get_bits(gb, 1)) {                  // other data
407
            if (audio_mux_version) {
408
                latm_get_value(gb);             // other_data_bits
409
            } else {
410
                int esc;
411
                do {
412
                    if (get_bits_left(gb) < 9)
413
                        return AVERROR_INVALIDDATA;
414
                    esc = get_bits(gb, 1);
415
                    skip_bits(gb, 8);
416
                } while (esc);
417
            }
418
        }
419
420
321
        if (get_bits(gb, 1))                     // crc present
421
            skip_bits(gb, 8);                    // config_crc
422
    }
423
424
321
    return 0;
425
}
426
427
514
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
428
{
429
    uint8_t tmp;
430
431
514
    if (ctx->frame_length_type == 0) {
432
514
        int mux_slot_length = 0;
433
        do {
434
1501
            if (get_bits_left(gb) < 8)
435
                return AVERROR_INVALIDDATA;
436
1501
            tmp = get_bits(gb, 8);
437
1501
            mux_slot_length += tmp;
438
1501
        } while (tmp == 255);
439
514
        return mux_slot_length;
440
    } else if (ctx->frame_length_type == 1) {
441
        return ctx->frame_length;
442
    } else if (ctx->frame_length_type == 3 ||
443
               ctx->frame_length_type == 5 ||
444
               ctx->frame_length_type == 7) {
445
        skip_bits(gb, 2);          // mux_slot_length_coded
446
    }
447
    return 0;
448
}
449
450
518
static int read_audio_mux_element(struct LATMContext *latmctx,
451
                                  GetBitContext *gb)
452
{
453
    int err;
454
518
    uint8_t use_same_mux = get_bits(gb, 1);
455
518
    if (!use_same_mux) {
456
321
        if ((err = read_stream_mux_config(latmctx, gb)) < 0)
457
            return err;
458
197
    } else if (!latmctx->aac_ctx.avctx->extradata) {
459
4
        av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
460
               "no decoder config found\n");
461
4
        return 1;
462
    }
463
514
    if (latmctx->audio_mux_version_A == 0) {
464
514
        int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
465

514
        if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) {
466
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
467
            return AVERROR_INVALIDDATA;
468
514
        } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
469
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
470
                   "frame length mismatch %d << %d\n",
471
                   mux_slot_length_bytes * 8, get_bits_left(gb));
472
            return AVERROR_INVALIDDATA;
473
        }
474
    }
475
514
    return 0;
476
}
477
478
479
520
static int latm_decode_frame(AVCodecContext *avctx, void *out,
480
                             int *got_frame_ptr, AVPacket *avpkt)
481
{
482
520
    struct LATMContext *latmctx = avctx->priv_data;
483
    int                 muxlength, err;
484
    GetBitContext       gb;
485
486
520
    if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
487
        return err;
488
489
    // check for LOAS sync word
490
520
    if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
491
        return AVERROR_INVALIDDATA;
492
493
520
    muxlength = get_bits(&gb, 13) + 3;
494
    // not enough data, the parser should have sorted this out
495
520
    if (muxlength > avpkt->size)
496
2
        return AVERROR_INVALIDDATA;
497
498
518
    if ((err = read_audio_mux_element(latmctx, &gb)))
499
4
        return (err < 0) ? err : avpkt->size;
500
501
514
    if (!latmctx->initialized) {
502
6
        if (!avctx->extradata) {
503
            *got_frame_ptr = 0;
504
            return avpkt->size;
505
        } else {
506
6
            push_output_configuration(&latmctx->aac_ctx);
507
6
            if ((err = decode_audio_specific_config(
508
                    &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
509
6
                    avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) {
510
                pop_output_configuration(&latmctx->aac_ctx);
511
                return err;
512
            }
513
6
            latmctx->initialized = 1;
514
        }
515
    }
516
517
514
    if (show_bits(&gb, 12) == 0xfff) {
518
        av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
519
               "ADTS header detected, probably as result of configuration "
520
               "misparsing\n");
521
        return AVERROR_INVALIDDATA;
522
    }
523
524
514
    switch (latmctx->aac_ctx.oc[1].m4ac.object_type) {
525
    case AOT_ER_AAC_LC:
526
    case AOT_ER_AAC_LTP:
527
    case AOT_ER_AAC_LD:
528
    case AOT_ER_AAC_ELD:
529
        err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb);
530
        break;
531
514
    default:
532
514
        err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt);
533
    }
534
514
    if (err < 0)
535
        return err;
536
537
514
    return muxlength;
538
}
539
540
9
static av_cold int latm_decode_init(AVCodecContext *avctx)
541
{
542
9
    struct LATMContext *latmctx = avctx->priv_data;
543
9
    int ret = aac_decode_init(avctx);
544
545
9
    if (avctx->extradata_size > 0)
546
4
        latmctx->initialized = !ret;
547
548
9
    return ret;
549
}
550
551
AVCodec ff_aac_decoder = {
552
    .name            = "aac",
553
    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
554
    .type            = AVMEDIA_TYPE_AUDIO,
555
    .id              = AV_CODEC_ID_AAC,
556
    .priv_data_size  = sizeof(AACContext),
557
    .init            = aac_decode_init,
558
    .close           = aac_decode_close,
559
    .decode          = aac_decode_frame,
560
    .sample_fmts     = (const enum AVSampleFormat[]) {
561
        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
562
    },
563
    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
564
    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
565
    .channel_layouts = aac_channel_layout,
566
    .flush = flush,
567
    .priv_class      = &aac_decoder_class,
568
    .profiles        = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
569
};
570
571
/*
572
    Note: This decoder filter is intended to decode LATM streams transferred
573
    in MPEG transport streams which only contain one program.
574
    To do a more complex LATM demuxing a separate LATM demuxer should be used.
575
*/
576
AVCodec ff_aac_latm_decoder = {
577
    .name            = "aac_latm",
578
    .long_name       = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
579
    .type            = AVMEDIA_TYPE_AUDIO,
580
    .id              = AV_CODEC_ID_AAC_LATM,
581
    .priv_data_size  = sizeof(struct LATMContext),
582
    .init            = latm_decode_init,
583
    .close           = aac_decode_close,
584
    .decode          = latm_decode_frame,
585
    .sample_fmts     = (const enum AVSampleFormat[]) {
586
        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
587
    },
588
    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
589
    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
590
    .channel_layouts = aac_channel_layout,
591
    .flush = flush,
592
    .profiles        = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
593
};