GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavcodec/aacdec.c Lines: 198 263 75.3 %
Date: 2021-04-18 10:33:33 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
DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(sine_120))[120];
73
DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(sine_960))[960];
74
DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(aac_kbd_long_960))[960];
75
DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(aac_kbd_short_120))[120];
76
77
38542
static av_always_inline void reset_predict_state(PredictorState *ps)
78
{
79
38542
    ps->r0   = 0.0f;
80
38542
    ps->r1   = 0.0f;
81
38542
    ps->cor0 = 0.0f;
82
38542
    ps->cor1 = 0.0f;
83
38542
    ps->var0 = 1.0f;
84
38542
    ps->var1 = 1.0f;
85
38542
}
86
87
#ifndef VMUL2
88
1839110
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
89
                           const float *scale)
90
{
91
1839110
    float s = *scale;
92
1839110
    *dst++ = v[idx    & 15] * s;
93
1839110
    *dst++ = v[idx>>4 & 15] * s;
94
1839110
    return dst;
95
}
96
#endif
97
98
#ifndef VMUL4
99
2262052
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
100
                           const float *scale)
101
{
102
2262052
    float s = *scale;
103
2262052
    *dst++ = v[idx    & 3] * s;
104
2262052
    *dst++ = v[idx>>2 & 3] * s;
105
2262052
    *dst++ = v[idx>>4 & 3] * s;
106
2262052
    *dst++ = v[idx>>6 & 3] * s;
107
2262052
    return dst;
108
}
109
#endif
110
111
#ifndef VMUL2S
112
2265052
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
113
                            unsigned sign, const float *scale)
114
{
115
    union av_intfloat32 s0, s1;
116
117
2265052
    s0.f = s1.f = *scale;
118
2265052
    s0.i ^= sign >> 1 << 31;
119
2265052
    s1.i ^= sign      << 31;
120
121
2265052
    *dst++ = v[idx    & 15] * s0.f;
122
2265052
    *dst++ = v[idx>>4 & 15] * s1.f;
123
124
2265052
    return dst;
125
}
126
#endif
127
128
#ifndef VMUL4S
129
1810354
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
130
                            unsigned sign, const float *scale)
131
{
132
1810354
    unsigned nz = idx >> 12;
133
1810354
    union av_intfloat32 s = { .f = *scale };
134
    union av_intfloat32 t;
135
136
1810354
    t.i = s.i ^ (sign & 1U<<31);
137
1810354
    *dst++ = v[idx    & 3] * t.f;
138
139
1810354
    sign <<= nz & 1; nz >>= 1;
140
1810354
    t.i = s.i ^ (sign & 1U<<31);
141
1810354
    *dst++ = v[idx>>2 & 3] * t.f;
142
143
1810354
    sign <<= nz & 1; nz >>= 1;
144
1810354
    t.i = s.i ^ (sign & 1U<<31);
145
1810354
    *dst++ = v[idx>>4 & 3] * t.f;
146
147
1810354
    sign <<= nz & 1;
148
1810354
    t.i = s.i ^ (sign & 1U<<31);
149
1810354
    *dst++ = v[idx>>6 & 3] * t.f;
150
151
1810354
    return dst;
152
}
153
#endif
154
155
4047936
static av_always_inline float flt16_round(float pf)
156
{
157
    union av_intfloat32 tmp;
158
4047936
    tmp.f = pf;
159
4047936
    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
160
4047936
    return tmp.f;
161
}
162
163
5085872
static av_always_inline float flt16_even(float pf)
164
{
165
    union av_intfloat32 tmp;
166
5085872
    tmp.f = pf;
167
5085872
    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
168
5085872
    return tmp.f;
169
}
170
171
24287616
static av_always_inline float flt16_trunc(float pf)
172
{
173
    union av_intfloat32 pun;
174
24287616
    pun.f = pf;
175
24287616
    pun.i &= 0xFFFF0000U;
176
24287616
    return pun.f;
177
}
178
179
4047936
static av_always_inline void predict(PredictorState *ps, float *coef,
180
                                     int output_enable)
181
{
182
4047936
    const float a     = 0.953125; // 61.0 / 64
183
4047936
    const float alpha = 0.90625;  // 29.0 / 32
184
    float e0, e1;
185
    float pv;
186
    float k1, k2;
187
4047936
    float   r0 = ps->r0,     r1 = ps->r1;
188
4047936
    float cor0 = ps->cor0, cor1 = ps->cor1;
189
4047936
    float var0 = ps->var0, var1 = ps->var1;
190
191
4047936
    k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
192
4047936
    k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
193
194
4047936
    pv = flt16_round(k1 * r0 + k2 * r1);
195
4047936
    if (output_enable)
196
271896
        *coef += pv;
197
198
4047936
    e0 = *coef;
199
4047936
    e1 = e0 - k1 * r0;
200
201
4047936
    ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
202
4047936
    ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
203
4047936
    ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
204
4047936
    ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
205
206
4047936
    ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
207
4047936
    ps->r0 = flt16_trunc(a * e0);
208
4047936
}
209
210
/**
211
 * Apply dependent channel coupling (applied before IMDCT).
212
 *
213
 * @param   index   index into coupling gain array
214
 */
215
2743
static void apply_dependent_coupling(AACContext *ac,
216
                                     SingleChannelElement *target,
217
                                     ChannelElement *cce, int index)
218
{
219
2743
    IndividualChannelStream *ics = &cce->ch[0].ics;
220
2743
    const uint16_t *offsets = ics->swb_offset;
221
2743
    float *dest = target->coeffs;
222
2743
    const float *src = cce->ch[0].coeffs;
223
2743
    int g, i, group, k, idx = 0;
224
2743
    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
225
        av_log(ac->avctx, AV_LOG_ERROR,
226
               "Dependent coupling is not supported together with LTP\n");
227
        return;
228
    }
229
5486
    for (g = 0; g < ics->num_window_groups; g++) {
230
112463
        for (i = 0; i < ics->max_sfb; i++, idx++) {
231
109720
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
232
5196
                const float gain = cce->coup.gain[index][idx];
233
10392
                for (group = 0; group < ics->group_len[g]; group++) {
234
81964
                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
235
                        // FIXME: SIMDify
236
76768
                        dest[group * 128 + k] += gain * src[group * 128 + k];
237
                    }
238
                }
239
            }
240
        }
241
2743
        dest += ics->group_len[g] * 128;
242
2743
        src  += ics->group_len[g] * 128;
243
    }
244
}
245
246
/**
247
 * Apply independent channel coupling (applied after IMDCT).
248
 *
249
 * @param   index   index into coupling gain array
250
 */
251
1315
static void apply_independent_coupling(AACContext *ac,
252
                                       SingleChannelElement *target,
253
                                       ChannelElement *cce, int index)
254
{
255
1315
    const float gain = cce->coup.gain[index][0];
256
1315
    const float *src = cce->ch[0].ret;
257
1315
    float *dest = target->ret;
258
1315
    const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
259
260
1315
    ac->fdsp->vector_fmac_scalar(dest, src, gain, len);
261
1315
}
262
263
#include "aacdec_template.c"
264
265
#define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
266
267
struct LATMContext {
268
    AACContext aac_ctx;     ///< containing AACContext
269
    int initialized;        ///< initialized after a valid extradata was seen
270
271
    // parser data
272
    int audio_mux_version_A; ///< LATM syntax version
273
    int frame_length_type;   ///< 0/1 variable/fixed frame length
274
    int frame_length;        ///< frame length for fixed frame length
275
};
276
277
static inline uint32_t latm_get_value(GetBitContext *b)
278
{
279
    int length = get_bits(b, 2);
280
281
    return get_bits_long(b, (length+1)*8);
282
}
283
284
321
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
285
                                             GetBitContext *gb, int asclen)
286
{
287
321
    AACContext *ac        = &latmctx->aac_ctx;
288
321
    AVCodecContext *avctx = ac->avctx;
289
321
    MPEG4AudioConfig m4ac = { 0 };
290
    GetBitContext gbc;
291
321
    int config_start_bit  = get_bits_count(gb);
292
321
    int sync_extension    = 0;
293
    int bits_consumed, esize, i;
294
295
321
    if (asclen > 0) {
296
        sync_extension = 1;
297
        asclen         = FFMIN(asclen, get_bits_left(gb));
298
        init_get_bits(&gbc, gb->buffer, config_start_bit + asclen);
299
        skip_bits_long(&gbc, config_start_bit);
300
321
    } else if (asclen == 0) {
301
321
        gbc = *gb;
302
    } else {
303
        return AVERROR_INVALIDDATA;
304
    }
305
306
321
    if (get_bits_left(gb) <= 0)
307
        return AVERROR_INVALIDDATA;
308
309
321
    bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac,
310
                                                    &gbc, config_start_bit,
311
                                                    sync_extension);
312
313
321
    if (bits_consumed < config_start_bit)
314
        return AVERROR_INVALIDDATA;
315
321
    bits_consumed -= config_start_bit;
316
317
321
    if (asclen == 0)
318
321
      asclen = bits_consumed;
319
320
321
    if (!latmctx->initialized ||
321
316
        ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
322
316
        ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
323
324
6
        if (latmctx->initialized) {
325
1
            av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config);
326
        } else {
327
5
            av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
328
        }
329
6
        latmctx->initialized = 0;
330
331
6
        esize = (asclen + 7) / 8;
332
333
6
        if (avctx->extradata_size < esize) {
334
5
            av_free(avctx->extradata);
335
5
            avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE);
336
5
            if (!avctx->extradata)
337
                return AVERROR(ENOMEM);
338
        }
339
340
6
        avctx->extradata_size = esize;
341
6
        gbc = *gb;
342
25
        for (i = 0; i < esize; i++) {
343
19
          avctx->extradata[i] = get_bits(&gbc, 8);
344
        }
345
6
        memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE);
346
    }
347
321
    skip_bits_long(gb, asclen);
348
349
321
    return 0;
350
}
351
352
321
static int read_stream_mux_config(struct LATMContext *latmctx,
353
                                  GetBitContext *gb)
354
{
355
321
    int ret, audio_mux_version = get_bits(gb, 1);
356
357
321
    latmctx->audio_mux_version_A = 0;
358
321
    if (audio_mux_version)
359
        latmctx->audio_mux_version_A = get_bits(gb, 1);
360
361
321
    if (!latmctx->audio_mux_version_A) {
362
363
321
        if (audio_mux_version)
364
            latm_get_value(gb);                 // taraFullness
365
366
321
        skip_bits(gb, 1);                       // allStreamSameTimeFraming
367
321
        skip_bits(gb, 6);                       // numSubFrames
368
        // numPrograms
369
321
        if (get_bits(gb, 4)) {                  // numPrograms
370
            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
371
            return AVERROR_PATCHWELCOME;
372
        }
373
374
        // for each program (which there is only one in DVB)
375
376
        // for each layer (which there is only one in DVB)
377
321
        if (get_bits(gb, 3)) {                   // numLayer
378
            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
379
            return AVERROR_PATCHWELCOME;
380
        }
381
382
        // for all but first stream: use_same_config = get_bits(gb, 1);
383
321
        if (!audio_mux_version) {
384
321
            if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
385
                return ret;
386
        } else {
387
            int ascLen = latm_get_value(gb);
388
            if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
389
                return ret;
390
        }
391
392
321
        latmctx->frame_length_type = get_bits(gb, 3);
393

321
        switch (latmctx->frame_length_type) {
394
321
        case 0:
395
321
            skip_bits(gb, 8);       // latmBufferFullness
396
321
            break;
397
        case 1:
398
            latmctx->frame_length = get_bits(gb, 9);
399
            break;
400
        case 3:
401
        case 4:
402
        case 5:
403
            skip_bits(gb, 6);       // CELP frame length table index
404
            break;
405
        case 6:
406
        case 7:
407
            skip_bits(gb, 1);       // HVXC frame length table index
408
            break;
409
        }
410
411
321
        if (get_bits(gb, 1)) {                  // other data
412
            if (audio_mux_version) {
413
                latm_get_value(gb);             // other_data_bits
414
            } else {
415
                int esc;
416
                do {
417
                    if (get_bits_left(gb) < 9)
418
                        return AVERROR_INVALIDDATA;
419
                    esc = get_bits(gb, 1);
420
                    skip_bits(gb, 8);
421
                } while (esc);
422
            }
423
        }
424
425
321
        if (get_bits(gb, 1))                     // crc present
426
            skip_bits(gb, 8);                    // config_crc
427
    }
428
429
321
    return 0;
430
}
431
432
514
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
433
{
434
    uint8_t tmp;
435
436
514
    if (ctx->frame_length_type == 0) {
437
514
        int mux_slot_length = 0;
438
        do {
439
1501
            if (get_bits_left(gb) < 8)
440
                return AVERROR_INVALIDDATA;
441
1501
            tmp = get_bits(gb, 8);
442
1501
            mux_slot_length += tmp;
443
1501
        } while (tmp == 255);
444
514
        return mux_slot_length;
445
    } else if (ctx->frame_length_type == 1) {
446
        return ctx->frame_length;
447
    } else if (ctx->frame_length_type == 3 ||
448
               ctx->frame_length_type == 5 ||
449
               ctx->frame_length_type == 7) {
450
        skip_bits(gb, 2);          // mux_slot_length_coded
451
    }
452
    return 0;
453
}
454
455
518
static int read_audio_mux_element(struct LATMContext *latmctx,
456
                                  GetBitContext *gb)
457
{
458
    int err;
459
518
    uint8_t use_same_mux = get_bits(gb, 1);
460
518
    if (!use_same_mux) {
461
321
        if ((err = read_stream_mux_config(latmctx, gb)) < 0)
462
            return err;
463
197
    } else if (!latmctx->aac_ctx.avctx->extradata) {
464
4
        av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
465
               "no decoder config found\n");
466
4
        return 1;
467
    }
468
514
    if (latmctx->audio_mux_version_A == 0) {
469
514
        int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
470

514
        if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) {
471
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
472
            return AVERROR_INVALIDDATA;
473
514
        } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
474
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
475
                   "frame length mismatch %d << %d\n",
476
                   mux_slot_length_bytes * 8, get_bits_left(gb));
477
            return AVERROR_INVALIDDATA;
478
        }
479
    }
480
514
    return 0;
481
}
482
483
484
520
static int latm_decode_frame(AVCodecContext *avctx, void *out,
485
                             int *got_frame_ptr, AVPacket *avpkt)
486
{
487
520
    struct LATMContext *latmctx = avctx->priv_data;
488
    int                 muxlength, err;
489
    GetBitContext       gb;
490
491
520
    if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
492
        return err;
493
494
    // check for LOAS sync word
495
520
    if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
496
        return AVERROR_INVALIDDATA;
497
498
520
    muxlength = get_bits(&gb, 13) + 3;
499
    // not enough data, the parser should have sorted this out
500
520
    if (muxlength > avpkt->size)
501
2
        return AVERROR_INVALIDDATA;
502
503
518
    if ((err = read_audio_mux_element(latmctx, &gb)))
504
4
        return (err < 0) ? err : avpkt->size;
505
506
514
    if (!latmctx->initialized) {
507
6
        if (!avctx->extradata) {
508
            *got_frame_ptr = 0;
509
            return avpkt->size;
510
        } else {
511
6
            push_output_configuration(&latmctx->aac_ctx);
512
6
            if ((err = decode_audio_specific_config(
513
                    &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
514
6
                    avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) {
515
                pop_output_configuration(&latmctx->aac_ctx);
516
                return err;
517
            }
518
6
            latmctx->initialized = 1;
519
        }
520
    }
521
522
514
    if (show_bits(&gb, 12) == 0xfff) {
523
        av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
524
               "ADTS header detected, probably as result of configuration "
525
               "misparsing\n");
526
        return AVERROR_INVALIDDATA;
527
    }
528
529
514
    switch (latmctx->aac_ctx.oc[1].m4ac.object_type) {
530
    case AOT_ER_AAC_LC:
531
    case AOT_ER_AAC_LTP:
532
    case AOT_ER_AAC_LD:
533
    case AOT_ER_AAC_ELD:
534
        err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb);
535
        break;
536
514
    default:
537
514
        err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt);
538
    }
539
514
    if (err < 0)
540
        return err;
541
542
514
    return muxlength;
543
}
544
545
9
static av_cold int latm_decode_init(AVCodecContext *avctx)
546
{
547
9
    struct LATMContext *latmctx = avctx->priv_data;
548
9
    int ret = aac_decode_init(avctx);
549
550
9
    if (avctx->extradata_size > 0)
551
4
        latmctx->initialized = !ret;
552
553
9
    return ret;
554
}
555
556
AVCodec ff_aac_decoder = {
557
    .name            = "aac",
558
    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
559
    .type            = AVMEDIA_TYPE_AUDIO,
560
    .id              = AV_CODEC_ID_AAC,
561
    .priv_data_size  = sizeof(AACContext),
562
    .init            = aac_decode_init,
563
    .close           = aac_decode_close,
564
    .decode          = aac_decode_frame,
565
    .sample_fmts     = (const enum AVSampleFormat[]) {
566
        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
567
    },
568
    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
569
    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
570
    .channel_layouts = aac_channel_layout,
571
    .flush = flush,
572
    .priv_class      = &aac_decoder_class,
573
    .profiles        = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
574
};
575
576
/*
577
    Note: This decoder filter is intended to decode LATM streams transferred
578
    in MPEG transport streams which only contain one program.
579
    To do a more complex LATM demuxing a separate LATM demuxer should be used.
580
*/
581
AVCodec ff_aac_latm_decoder = {
582
    .name            = "aac_latm",
583
    .long_name       = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
584
    .type            = AVMEDIA_TYPE_AUDIO,
585
    .id              = AV_CODEC_ID_AAC_LATM,
586
    .priv_data_size  = sizeof(struct LATMContext),
587
    .init            = latm_decode_init,
588
    .close           = aac_decode_close,
589
    .decode          = latm_decode_frame,
590
    .sample_fmts     = (const enum AVSampleFormat[]) {
591
        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
592
    },
593
    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
594
    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
595
    .channel_layouts = aac_channel_layout,
596
    .flush = flush,
597
    .profiles        = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
598
};