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 USE_FIXED 0
37 
38 #include "libavutil/float_dsp.h"
39 #include "libavutil/opt.h"
40 #include "avcodec.h"
41 #include "codec_internal.h"
42 #include "get_bits.h"
43 #include "fft.h"
44 #include "mdct15.h"
45 #include "lpc.h"
46 #include "kbdwin.h"
47 #include "sinewin.h"
48 
49 #include "aac.h"
50 #include "aactab.h"
51 #include "aacdectab.h"
52 #include "adts_header.h"
53 #include "cbrt_data.h"
54 #include "sbr.h"
55 #include "aacsbr.h"
56 #include "mpeg4audio.h"
57 #include "profiles.h"
58 #include "libavutil/intfloat.h"
59 
60 #include <errno.h>
61 #include <math.h>
62 #include <stdint.h>
63 #include <string.h>
64 
65 #if ARCH_ARM
66 #   include "arm/aac.h"
67 #elif ARCH_MIPS
68 #   include "mips/aacdec_mips.h"
69 #endif
70 
71 DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(sine_120))[120];
72 DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(sine_960))[960];
73 DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(aac_kbd_long_960))[960];
74 DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(aac_kbd_short_120))[120];
75 
reset_predict_state(PredictorState *ps)76 static av_always_inline void reset_predict_state(PredictorState *ps)
77 {
78     ps->r0   = 0.0f;
79     ps->r1   = 0.0f;
80     ps->cor0 = 0.0f;
81     ps->cor1 = 0.0f;
82     ps->var0 = 1.0f;
83     ps->var1 = 1.0f;
84 }
85 
86 #ifndef VMUL2
VMUL2(float *dst, const float *v, unsigned idx, const float *scale)87 static inline float *VMUL2(float *dst, const float *v, unsigned idx,
88                            const float *scale)
89 {
90     float s = *scale;
91     *dst++ = v[idx    & 15] * s;
92     *dst++ = v[idx>>4 & 15] * s;
93     return dst;
94 }
95 #endif
96 
97 #ifndef VMUL4
VMUL4(float *dst, const float *v, unsigned idx, const float *scale)98 static inline float *VMUL4(float *dst, const float *v, unsigned idx,
99                            const float *scale)
100 {
101     float s = *scale;
102     *dst++ = v[idx    & 3] * s;
103     *dst++ = v[idx>>2 & 3] * s;
104     *dst++ = v[idx>>4 & 3] * s;
105     *dst++ = v[idx>>6 & 3] * s;
106     return dst;
107 }
108 #endif
109 
110 #ifndef VMUL2S
VMUL2S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)111 static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
112                             unsigned sign, const float *scale)
113 {
114     union av_intfloat32 s0, s1;
115 
116     s0.f = s1.f = *scale;
117     s0.i ^= sign >> 1 << 31;
118     s1.i ^= sign      << 31;
119 
120     *dst++ = v[idx    & 15] * s0.f;
121     *dst++ = v[idx>>4 & 15] * s1.f;
122 
123     return dst;
124 }
125 #endif
126 
127 #ifndef VMUL4S
VMUL4S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)128 static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
129                             unsigned sign, const float *scale)
130 {
131     unsigned nz = idx >> 12;
132     union av_intfloat32 s = { .f = *scale };
133     union av_intfloat32 t;
134 
135     t.i = s.i ^ (sign & 1U<<31);
136     *dst++ = v[idx    & 3] * t.f;
137 
138     sign <<= nz & 1; nz >>= 1;
139     t.i = s.i ^ (sign & 1U<<31);
140     *dst++ = v[idx>>2 & 3] * t.f;
141 
142     sign <<= nz & 1; nz >>= 1;
143     t.i = s.i ^ (sign & 1U<<31);
144     *dst++ = v[idx>>4 & 3] * t.f;
145 
146     sign <<= nz & 1;
147     t.i = s.i ^ (sign & 1U<<31);
148     *dst++ = v[idx>>6 & 3] * t.f;
149 
150     return dst;
151 }
152 #endif
153 
flt16_round(float pf)154 static av_always_inline float flt16_round(float pf)
155 {
156     union av_intfloat32 tmp;
157     tmp.f = pf;
158     tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
159     return tmp.f;
160 }
161 
flt16_even(float pf)162 static av_always_inline float flt16_even(float pf)
163 {
164     union av_intfloat32 tmp;
165     tmp.f = pf;
166     tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
167     return tmp.f;
168 }
169 
flt16_trunc(float pf)170 static av_always_inline float flt16_trunc(float pf)
171 {
172     union av_intfloat32 pun;
173     pun.f = pf;
174     pun.i &= 0xFFFF0000U;
175     return pun.f;
176 }
177 
predict(PredictorState *ps, float *coef, int output_enable)178 static av_always_inline void predict(PredictorState *ps, float *coef,
179                                      int output_enable)
180 {
181     const float a     = 0.953125; // 61.0 / 64
182     const float alpha = 0.90625;  // 29.0 / 32
183     float e0, e1;
184     float pv;
185     float k1, k2;
186     float   r0 = ps->r0,     r1 = ps->r1;
187     float cor0 = ps->cor0, cor1 = ps->cor1;
188     float var0 = ps->var0, var1 = ps->var1;
189 
190     k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
191     k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
192 
193     pv = flt16_round(k1 * r0 + k2 * r1);
194     if (output_enable)
195         *coef += pv;
196 
197     e0 = *coef;
198     e1 = e0 - k1 * r0;
199 
200     ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
201     ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
202     ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
203     ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
204 
205     ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
206     ps->r0 = flt16_trunc(a * e0);
207 }
208 
209 /**
210  * Apply dependent channel coupling (applied before IMDCT).
211  *
212  * @param   index   index into coupling gain array
213  */
apply_dependent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)214 static void apply_dependent_coupling(AACContext *ac,
215                                      SingleChannelElement *target,
216                                      ChannelElement *cce, int index)
217 {
218     IndividualChannelStream *ics = &cce->ch[0].ics;
219     const uint16_t *offsets = ics->swb_offset;
220     float *dest = target->coeffs;
221     const float *src = cce->ch[0].coeffs;
222     int g, i, group, k, idx = 0;
223     if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
224         av_log(ac->avctx, AV_LOG_ERROR,
225                "Dependent coupling is not supported together with LTP\n");
226         return;
227     }
228     for (g = 0; g < ics->num_window_groups; g++) {
229         for (i = 0; i < ics->max_sfb; i++, idx++) {
230             if (cce->ch[0].band_type[idx] != ZERO_BT) {
231                 const float gain = cce->coup.gain[index][idx];
232                 for (group = 0; group < ics->group_len[g]; group++) {
233                     for (k = offsets[i]; k < offsets[i + 1]; k++) {
234                         // FIXME: SIMDify
235                         dest[group * 128 + k] += gain * src[group * 128 + k];
236                     }
237                 }
238             }
239         }
240         dest += ics->group_len[g] * 128;
241         src  += ics->group_len[g] * 128;
242     }
243 }
244 
245 /**
246  * Apply independent channel coupling (applied after IMDCT).
247  *
248  * @param   index   index into coupling gain array
249  */
apply_independent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)250 static void apply_independent_coupling(AACContext *ac,
251                                        SingleChannelElement *target,
252                                        ChannelElement *cce, int index)
253 {
254     const float gain = cce->coup.gain[index][0];
255     const float *src = cce->ch[0].ret;
256     float *dest = target->ret;
257     const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
258 
259     ac->fdsp->vector_fmac_scalar(dest, src, gain, len);
260 }
261 
262 #include "aacdec_template.c"
263 
264 #define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
265 
266 struct LATMContext {
267     AACContext aac_ctx;     ///< containing AACContext
268     int initialized;        ///< initialized after a valid extradata was seen
269 
270     // parser data
271     int audio_mux_version_A; ///< LATM syntax version
272     int frame_length_type;   ///< 0/1 variable/fixed frame length
273     int frame_length;        ///< frame length for fixed frame length
274 };
275 
latm_get_value(GetBitContext *b)276 static inline uint32_t latm_get_value(GetBitContext *b)
277 {
278     int length = get_bits(b, 2);
279 
280     return get_bits_long(b, (length+1)*8);
281 }
282 
latm_decode_audio_specific_config(struct LATMContext *latmctx, GetBitContext *gb, int asclen)283 static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
284                                              GetBitContext *gb, int asclen)
285 {
286     AACContext *ac        = &latmctx->aac_ctx;
287     AVCodecContext *avctx = ac->avctx;
288     MPEG4AudioConfig m4ac = { 0 };
289     GetBitContext gbc;
290     int config_start_bit  = get_bits_count(gb);
291     int sync_extension    = 0;
292     int bits_consumed, esize, i;
293 
294     if (asclen > 0) {
295         sync_extension = 1;
296         asclen         = FFMIN(asclen, get_bits_left(gb));
297         init_get_bits(&gbc, gb->buffer, config_start_bit + asclen);
298         skip_bits_long(&gbc, config_start_bit);
299     } else if (asclen == 0) {
300         gbc = *gb;
301     } else {
302         return AVERROR_INVALIDDATA;
303     }
304 
305     if (get_bits_left(gb) <= 0)
306         return AVERROR_INVALIDDATA;
307 
308     bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac,
309                                                     &gbc, config_start_bit,
310                                                     sync_extension);
311 
312     if (bits_consumed < config_start_bit)
313         return AVERROR_INVALIDDATA;
314     bits_consumed -= config_start_bit;
315 
316     if (asclen == 0)
317       asclen = bits_consumed;
318 
319     if (!latmctx->initialized ||
320         ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
321         ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
322 
323         if (latmctx->initialized) {
324             av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config);
325         } else {
326             av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
327         }
328         latmctx->initialized = 0;
329 
330         esize = (asclen + 7) / 8;
331 
332         if (avctx->extradata_size < esize) {
333             av_free(avctx->extradata);
334             avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE);
335             if (!avctx->extradata)
336                 return AVERROR(ENOMEM);
337         }
338 
339         avctx->extradata_size = esize;
340         gbc = *gb;
341         for (i = 0; i < esize; i++) {
342           avctx->extradata[i] = get_bits(&gbc, 8);
343         }
344         memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE);
345     }
346     skip_bits_long(gb, asclen);
347 
348     return 0;
349 }
350 
read_stream_mux_config(struct LATMContext *latmctx, GetBitContext *gb)351 static int read_stream_mux_config(struct LATMContext *latmctx,
352                                   GetBitContext *gb)
353 {
354     int ret, audio_mux_version = get_bits(gb, 1);
355 
356     latmctx->audio_mux_version_A = 0;
357     if (audio_mux_version)
358         latmctx->audio_mux_version_A = get_bits(gb, 1);
359 
360     if (!latmctx->audio_mux_version_A) {
361 
362         if (audio_mux_version)
363             latm_get_value(gb);                 // taraFullness
364 
365         skip_bits(gb, 1);                       // allStreamSameTimeFraming
366         skip_bits(gb, 6);                       // numSubFrames
367         // numPrograms
368         if (get_bits(gb, 4)) {                  // numPrograms
369             avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
370             return AVERROR_PATCHWELCOME;
371         }
372 
373         // for each program (which there is only one in DVB)
374 
375         // for each layer (which there is only one in DVB)
376         if (get_bits(gb, 3)) {                   // numLayer
377             avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
378             return AVERROR_PATCHWELCOME;
379         }
380 
381         // for all but first stream: use_same_config = get_bits(gb, 1);
382         if (!audio_mux_version) {
383             if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
384                 return ret;
385         } else {
386             int ascLen = latm_get_value(gb);
387             if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
388                 return ret;
389         }
390 
391         latmctx->frame_length_type = get_bits(gb, 3);
392         switch (latmctx->frame_length_type) {
393         case 0:
394             skip_bits(gb, 8);       // latmBufferFullness
395             break;
396         case 1:
397             latmctx->frame_length = get_bits(gb, 9);
398             break;
399         case 3:
400         case 4:
401         case 5:
402             skip_bits(gb, 6);       // CELP frame length table index
403             break;
404         case 6:
405         case 7:
406             skip_bits(gb, 1);       // HVXC frame length table index
407             break;
408         }
409 
410         if (get_bits(gb, 1)) {                  // other data
411             if (audio_mux_version) {
412                 latm_get_value(gb);             // other_data_bits
413             } else {
414                 int esc;
415                 do {
416                     if (get_bits_left(gb) < 9)
417                         return AVERROR_INVALIDDATA;
418                     esc = get_bits(gb, 1);
419                     skip_bits(gb, 8);
420                 } while (esc);
421             }
422         }
423 
424         if (get_bits(gb, 1))                     // crc present
425             skip_bits(gb, 8);                    // config_crc
426     }
427 
428     return 0;
429 }
430 
read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)431 static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
432 {
433     uint8_t tmp;
434 
435     if (ctx->frame_length_type == 0) {
436         int mux_slot_length = 0;
437         do {
438             if (get_bits_left(gb) < 8)
439                 return AVERROR_INVALIDDATA;
440             tmp = get_bits(gb, 8);
441             mux_slot_length += tmp;
442         } while (tmp == 255);
443         return mux_slot_length;
444     } else if (ctx->frame_length_type == 1) {
445         return ctx->frame_length;
446     } else if (ctx->frame_length_type == 3 ||
447                ctx->frame_length_type == 5 ||
448                ctx->frame_length_type == 7) {
449         skip_bits(gb, 2);          // mux_slot_length_coded
450     }
451     return 0;
452 }
453 
read_audio_mux_element(struct LATMContext *latmctx, GetBitContext *gb)454 static int read_audio_mux_element(struct LATMContext *latmctx,
455                                   GetBitContext *gb)
456 {
457     int err;
458     uint8_t use_same_mux = get_bits(gb, 1);
459     if (!use_same_mux) {
460         if ((err = read_stream_mux_config(latmctx, gb)) < 0)
461             return err;
462     } else if (!latmctx->aac_ctx.avctx->extradata) {
463         av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
464                "no decoder config found\n");
465         return 1;
466     }
467     if (latmctx->audio_mux_version_A == 0) {
468         int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
469         if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) {
470             av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
471             return AVERROR_INVALIDDATA;
472         } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
473             av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
474                    "frame length mismatch %d << %d\n",
475                    mux_slot_length_bytes * 8, get_bits_left(gb));
476             return AVERROR_INVALIDDATA;
477         }
478     }
479     return 0;
480 }
481 
482 
latm_decode_frame(AVCodecContext *avctx, AVFrame *out, int *got_frame_ptr, AVPacket *avpkt)483 static int latm_decode_frame(AVCodecContext *avctx, AVFrame *out,
484                              int *got_frame_ptr, AVPacket *avpkt)
485 {
486     struct LATMContext *latmctx = avctx->priv_data;
487     int                 muxlength, err;
488     GetBitContext       gb;
489 
490     if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
491         return err;
492 
493     // check for LOAS sync word
494     if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
495         return AVERROR_INVALIDDATA;
496 
497     muxlength = get_bits(&gb, 13) + 3;
498     // not enough data, the parser should have sorted this out
499     if (muxlength > avpkt->size)
500         return AVERROR_INVALIDDATA;
501 
502     if ((err = read_audio_mux_element(latmctx, &gb)))
503         return (err < 0) ? err : avpkt->size;
504 
505     if (!latmctx->initialized) {
506         if (!avctx->extradata) {
507             *got_frame_ptr = 0;
508             return avpkt->size;
509         } else {
510             push_output_configuration(&latmctx->aac_ctx);
511             if ((err = decode_audio_specific_config(
512                     &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
513                     avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) {
514                 pop_output_configuration(&latmctx->aac_ctx);
515                 return err;
516             }
517             latmctx->initialized = 1;
518         }
519     }
520 
521     if (show_bits(&gb, 12) == 0xfff) {
522         av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
523                "ADTS header detected, probably as result of configuration "
524                "misparsing\n");
525         return AVERROR_INVALIDDATA;
526     }
527 
528     switch (latmctx->aac_ctx.oc[1].m4ac.object_type) {
529     case AOT_ER_AAC_LC:
530     case AOT_ER_AAC_LTP:
531     case AOT_ER_AAC_LD:
532     case AOT_ER_AAC_ELD:
533         err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb);
534         break;
535     default:
536         err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt);
537     }
538     if (err < 0)
539         return err;
540 
541     return muxlength;
542 }
543 
latm_decode_init(AVCodecContext *avctx)544 static av_cold int latm_decode_init(AVCodecContext *avctx)
545 {
546     struct LATMContext *latmctx = avctx->priv_data;
547     int ret = aac_decode_init(avctx);
548 
549     if (avctx->extradata_size > 0)
550         latmctx->initialized = !ret;
551 
552     return ret;
553 }
554 
555 const FFCodec ff_aac_decoder = {
556     .p.name          = "aac",
557     .p.long_name     = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
558     .p.type          = AVMEDIA_TYPE_AUDIO,
559     .p.id            = AV_CODEC_ID_AAC,
560     .priv_data_size  = sizeof(AACContext),
561     .init            = aac_decode_init,
562     .close           = aac_decode_close,
563     FF_CODEC_DECODE_CB(aac_decode_frame),
564     .p.sample_fmts   = (const enum AVSampleFormat[]) {
565         AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
566     },
567     .p.capabilities  = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
568     .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
569 #if FF_API_OLD_CHANNEL_LAYOUT
570     .p.channel_layouts = aac_channel_layout,
571 #endif
572     .p.ch_layouts    = aac_ch_layout,
573     .flush = flush,
574     .p.priv_class    = &aac_decoder_class,
575     .p.profiles      = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
576 };
577 
578 /*
579     Note: This decoder filter is intended to decode LATM streams transferred
580     in MPEG transport streams which only contain one program.
581     To do a more complex LATM demuxing a separate LATM demuxer should be used.
582 */
583 const FFCodec ff_aac_latm_decoder = {
584     .p.name          = "aac_latm",
585     .p.long_name     = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
586     .p.type          = AVMEDIA_TYPE_AUDIO,
587     .p.id            = AV_CODEC_ID_AAC_LATM,
588     .priv_data_size  = sizeof(struct LATMContext),
589     .init            = latm_decode_init,
590     .close           = aac_decode_close,
591     FF_CODEC_DECODE_CB(latm_decode_frame),
592     .p.sample_fmts   = (const enum AVSampleFormat[]) {
593         AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
594     },
595     .p.capabilities  = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
596     .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
597 #if FF_API_OLD_CHANNEL_LAYOUT
598     .p.channel_layouts = aac_channel_layout,
599 #endif
600     .p.ch_layouts    = aac_ch_layout,
601     .flush = flush,
602     .p.profiles      = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
603 };
604