3 * Copyright (C) 2008 Konstantin Shishkov
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
27 /***********************************
29 * add sane pulse detection
30 * add temporal noise shaping
31 ***********************************/
36 #include "mpeg4audio.h"
44 #define AAC_MAX_CHANNELS 6
46 static const uint8_t swb_size_1024_96[] = {
47 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
48 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
49 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
52 static const uint8_t swb_size_1024_64[] = {
53 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
54 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
55 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
58 static const uint8_t swb_size_1024_48[] = {
59 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
60 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
61 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
65 static const uint8_t swb_size_1024_32[] = {
66 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
67 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
68 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
71 static const uint8_t swb_size_1024_24[] = {
72 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
73 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
74 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
77 static const uint8_t swb_size_1024_16[] = {
78 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
79 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
80 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
83 static const uint8_t swb_size_1024_8[] = {
84 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
85 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
86 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
89 static const uint8_t *swb_size_1024[] = {
90 swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
91 swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
92 swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
93 swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
96 static const uint8_t swb_size_128_96[] = {
97 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
100 static const uint8_t swb_size_128_48[] = {
101 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
104 static const uint8_t swb_size_128_24[] = {
105 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
108 static const uint8_t swb_size_128_16[] = {
109 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
112 static const uint8_t swb_size_128_8[] = {
113 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
116 static const uint8_t *swb_size_128[] = {
117 /* the last entry on the following row is swb_size_128_64 but is a
118 duplicate of swb_size_128_96 */
119 swb_size_128_96, swb_size_128_96, swb_size_128_96,
120 swb_size_128_48, swb_size_128_48, swb_size_128_48,
121 swb_size_128_24, swb_size_128_24, swb_size_128_16,
122 swb_size_128_16, swb_size_128_16, swb_size_128_8
125 /** default channel configurations */
126 static const uint8_t aac_chan_configs[6][5] = {
127 {1, TYPE_SCE}, // 1 channel - single channel element
128 {1, TYPE_CPE}, // 2 channels - channel pair
129 {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
130 {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
131 {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
132 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
136 * Make AAC audio config object.
137 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
139 static void put_audio_specific_config(AVCodecContext *avctx)
142 AACEncContext *s = avctx->priv_data;
144 init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
145 put_bits(&pb, 5, 2); //object type - AAC-LC
146 put_bits(&pb, 4, s->samplerate_index); //sample rate index
147 put_bits(&pb, 4, avctx->channels);
149 put_bits(&pb, 1, 0); //frame length - 1024 samples
150 put_bits(&pb, 1, 0); //does not depend on core coder
151 put_bits(&pb, 1, 0); //is not extension
155 static av_cold int aac_encode_init(AVCodecContext *avctx)
157 AACEncContext *s = avctx->priv_data;
159 const uint8_t *sizes[2];
162 avctx->frame_size = 1024;
164 for (i = 0; i < 16; i++)
165 if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
168 av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
171 if (avctx->channels > AAC_MAX_CHANNELS) {
172 av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
175 if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) {
176 av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile);
179 if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) {
180 av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n");
183 s->samplerate_index = i;
185 dsputil_init(&s->dsp, avctx);
186 ff_mdct_init(&s->mdct1024, 11, 0, 1.0);
187 ff_mdct_init(&s->mdct128, 8, 0, 1.0);
189 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
190 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
191 ff_init_ff_sine_windows(10);
192 ff_init_ff_sine_windows(7);
194 s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
195 s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
196 avctx->extradata = av_mallocz(2 + FF_INPUT_BUFFER_PADDING_SIZE);
197 avctx->extradata_size = 2;
198 put_audio_specific_config(avctx);
200 sizes[0] = swb_size_1024[i];
201 sizes[1] = swb_size_128[i];
202 lengths[0] = ff_aac_num_swb_1024[i];
203 lengths[1] = ff_aac_num_swb_128[i];
204 ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
205 s->psypp = ff_psy_preprocess_init(avctx);
206 s->coder = &ff_aac_coders[2];
208 s->lambda = avctx->global_quality ? avctx->global_quality : 120;
215 static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
216 SingleChannelElement *sce, short *audio)
219 const int chans = avctx->channels;
220 const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
221 const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
222 const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
224 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
225 memcpy(s->output, sce->saved, sizeof(float)*1024);
226 if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
227 memset(s->output, 0, sizeof(s->output[0]) * 448);
228 for (i = 448; i < 576; i++)
229 s->output[i] = sce->saved[i] * pwindow[i - 448];
230 for (i = 576; i < 704; i++)
231 s->output[i] = sce->saved[i];
233 if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
234 for (i = 0; i < 1024; i++) {
235 s->output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1];
236 sce->saved[i] = audio[i * chans] * lwindow[i];
239 for (i = 0; i < 448; i++)
240 s->output[i+1024] = audio[i * chans];
242 s->output[i+1024] = audio[i * chans] * swindow[576 - i - 1];
243 memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
244 for (i = 0; i < 1024; i++)
245 sce->saved[i] = audio[i * chans];
247 ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
249 for (k = 0; k < 1024; k += 128) {
250 for (i = 448 + k; i < 448 + k + 256; i++)
251 s->output[i - 448 - k] = (i < 1024)
253 : audio[(i-1024)*chans];
254 s->dsp.vector_fmul (s->output, k ? swindow : pwindow, 128);
255 s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
256 ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
258 for (i = 0; i < 1024; i++)
259 sce->saved[i] = audio[i * chans];
264 * Encode ics_info element.
265 * @see Table 4.6 (syntax of ics_info)
267 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
271 put_bits(&s->pb, 1, 0); // ics_reserved bit
272 put_bits(&s->pb, 2, info->window_sequence[0]);
273 put_bits(&s->pb, 1, info->use_kb_window[0]);
274 if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
275 put_bits(&s->pb, 6, info->max_sfb);
276 put_bits(&s->pb, 1, 0); // no prediction
278 put_bits(&s->pb, 4, info->max_sfb);
279 for (w = 1; w < 8; w++)
280 put_bits(&s->pb, 1, !info->group_len[w]);
286 * @see 4.6.8.1 "Joint Coding - M/S Stereo"
288 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
292 put_bits(pb, 2, cpe->ms_mode);
293 if (cpe->ms_mode == 1)
294 for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
295 for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
296 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
300 * Produce integer coefficients from scalefactors provided by the model.
302 static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
305 int start, sum, maxsfb, cmaxsfb;
307 for (ch = 0; ch < chans; ch++) {
308 IndividualChannelStream *ics = &cpe->ch[ch].ics;
311 cpe->ch[ch].pulse.num_pulse = 0;
312 for (w = 0; w < ics->num_windows*16; w += 16) {
313 for (g = 0; g < ics->num_swb; g++) {
316 if (!ch && cpe->ms_mask[w + g]) {
317 for (i = 0; i < ics->swb_sizes[g]; i++) {
318 cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
319 cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
322 start += ics->swb_sizes[g];
324 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
326 maxsfb = FFMAX(maxsfb, cmaxsfb);
328 ics->max_sfb = maxsfb;
330 //adjust zero bands for window groups
331 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
332 for (g = 0; g < ics->max_sfb; g++) {
334 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
335 if (!cpe->ch[ch].zeroes[w2*16 + g]) {
340 cpe->ch[ch].zeroes[w*16 + g] = i;
345 if (chans > 1 && cpe->common_window) {
346 IndividualChannelStream *ics0 = &cpe->ch[0].ics;
347 IndividualChannelStream *ics1 = &cpe->ch[1].ics;
349 ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
350 ics1->max_sfb = ics0->max_sfb;
351 for (w = 0; w < ics0->num_windows*16; w += 16)
352 for (i = 0; i < ics0->max_sfb; i++)
353 if (cpe->ms_mask[w+i])
355 if (msc == 0 || ics0->max_sfb == 0)
358 cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
363 * Encode scalefactor band coding type.
365 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
369 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
370 s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
374 * Encode scalefactors.
376 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
377 SingleChannelElement *sce)
379 int off = sce->sf_idx[0], diff;
382 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
383 for (i = 0; i < sce->ics.max_sfb; i++) {
384 if (!sce->zeroes[w*16 + i]) {
385 diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
386 if (diff < 0 || diff > 120)
387 av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
388 off = sce->sf_idx[w*16 + i];
389 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
398 static void encode_pulses(AACEncContext *s, Pulse *pulse)
402 put_bits(&s->pb, 1, !!pulse->num_pulse);
403 if (!pulse->num_pulse)
406 put_bits(&s->pb, 2, pulse->num_pulse - 1);
407 put_bits(&s->pb, 6, pulse->start);
408 for (i = 0; i < pulse->num_pulse; i++) {
409 put_bits(&s->pb, 5, pulse->pos[i]);
410 put_bits(&s->pb, 4, pulse->amp[i]);
415 * Encode spectral coefficients processed by psychoacoustic model.
417 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
421 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
423 for (i = 0; i < sce->ics.max_sfb; i++) {
424 if (sce->zeroes[w*16 + i]) {
425 start += sce->ics.swb_sizes[i];
428 for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
429 s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
430 sce->ics.swb_sizes[i],
431 sce->sf_idx[w*16 + i],
432 sce->band_type[w*16 + i],
434 start += sce->ics.swb_sizes[i];
440 * Encode one channel of audio data.
442 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
443 SingleChannelElement *sce,
446 put_bits(&s->pb, 8, sce->sf_idx[0]);
448 put_ics_info(s, &sce->ics);
449 encode_band_info(s, sce);
450 encode_scale_factors(avctx, s, sce);
451 encode_pulses(s, &sce->pulse);
452 put_bits(&s->pb, 1, 0); //tns
453 put_bits(&s->pb, 1, 0); //ssr
454 encode_spectral_coeffs(s, sce);
459 * Write some auxiliary information about the created AAC file.
461 static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
464 int i, namelen, padbits;
466 namelen = strlen(name) + 2;
467 put_bits(&s->pb, 3, TYPE_FIL);
468 put_bits(&s->pb, 4, FFMIN(namelen, 15));
470 put_bits(&s->pb, 8, namelen - 16);
471 put_bits(&s->pb, 4, 0); //extension type - filler
472 padbits = 8 - (put_bits_count(&s->pb) & 7);
473 align_put_bits(&s->pb);
474 for (i = 0; i < namelen - 2; i++)
475 put_bits(&s->pb, 8, name[i]);
476 put_bits(&s->pb, 12 - padbits, 0);
479 static int aac_encode_frame(AVCodecContext *avctx,
480 uint8_t *frame, int buf_size, void *data)
482 AACEncContext *s = avctx->priv_data;
483 int16_t *samples = s->samples, *samples2, *la;
485 int i, j, chans, tag, start_ch;
486 const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
487 int chan_el_counter[4];
488 FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
494 memcpy(s->samples + 1024 * avctx->channels, data,
495 1024 * avctx->channels * sizeof(s->samples[0]));
498 samples2 = s->samples + 1024 * avctx->channels;
499 for (i = 0; i < chan_map[0]; i++) {
501 chans = tag == TYPE_CPE ? 2 : 1;
502 ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
503 samples2 + start_ch, start_ch, chans);
508 if (!avctx->frame_number) {
509 memcpy(s->samples, s->samples + 1024 * avctx->channels,
510 1024 * avctx->channels * sizeof(s->samples[0]));
515 for (i = 0; i < chan_map[0]; i++) {
516 FFPsyWindowInfo* wi = windows + start_ch;
518 chans = tag == TYPE_CPE ? 2 : 1;
520 for (j = 0; j < chans; j++) {
521 IndividualChannelStream *ics = &cpe->ch[j].ics;
523 int cur_channel = start_ch + j;
524 samples2 = samples + cur_channel;
525 la = samples2 + (448+64) * avctx->channels;
528 if (tag == TYPE_LFE) {
529 wi[j].window_type[0] = ONLY_LONG_SEQUENCE;
530 wi[j].window_shape = 0;
531 wi[j].num_windows = 1;
532 wi[j].grouping[0] = 1;
534 wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, cur_channel,
535 ics->window_sequence[0]);
537 ics->window_sequence[1] = ics->window_sequence[0];
538 ics->window_sequence[0] = wi[j].window_type[0];
539 ics->use_kb_window[1] = ics->use_kb_window[0];
540 ics->use_kb_window[0] = wi[j].window_shape;
541 ics->num_windows = wi[j].num_windows;
542 ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
543 ics->num_swb = tag == TYPE_LFE ? 12 : s->psy.num_bands[ics->num_windows == 8];
544 for (k = 0; k < ics->num_windows; k++)
545 ics->group_len[k] = wi[j].grouping[k];
547 s->cur_channel = cur_channel;
548 apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2);
554 init_put_bits(&s->pb, frame, buf_size*8);
555 if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
556 put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
558 memset(chan_el_counter, 0, sizeof(chan_el_counter));
559 for (i = 0; i < chan_map[0]; i++) {
560 FFPsyWindowInfo* wi = windows + start_ch;
562 chans = tag == TYPE_CPE ? 2 : 1;
564 for (j = 0; j < chans; j++) {
565 s->cur_channel = start_ch + j;
566 ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
567 s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
569 cpe->common_window = 0;
571 && wi[0].window_type[0] == wi[1].window_type[0]
572 && wi[0].window_shape == wi[1].window_shape) {
574 cpe->common_window = 1;
575 for (j = 0; j < wi[0].num_windows; j++) {
576 if (wi[0].grouping[j] != wi[1].grouping[j]) {
577 cpe->common_window = 0;
582 s->cur_channel = start_ch;
583 if (cpe->common_window && s->coder->search_for_ms)
584 s->coder->search_for_ms(s, cpe, s->lambda);
585 adjust_frame_information(s, cpe, chans);
586 put_bits(&s->pb, 3, tag);
587 put_bits(&s->pb, 4, chan_el_counter[tag]++);
589 put_bits(&s->pb, 1, cpe->common_window);
590 if (cpe->common_window) {
591 put_ics_info(s, &cpe->ch[0].ics);
592 encode_ms_info(&s->pb, cpe);
595 for (j = 0; j < chans; j++) {
596 s->cur_channel = start_ch + j;
597 encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
602 frame_bits = put_bits_count(&s->pb);
603 if (frame_bits <= 6144 * avctx->channels - 3)
606 s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
610 put_bits(&s->pb, 3, TYPE_END);
611 flush_put_bits(&s->pb);
612 avctx->frame_bits = put_bits_count(&s->pb);
614 // rate control stuff
615 if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
616 float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
618 s->lambda = FFMIN(s->lambda, 65536.f);
623 memcpy(s->samples, s->samples + 1024 * avctx->channels,
624 1024 * avctx->channels * sizeof(s->samples[0]));
625 return put_bits_count(&s->pb)>>3;
628 static av_cold int aac_encode_end(AVCodecContext *avctx)
630 AACEncContext *s = avctx->priv_data;
632 ff_mdct_end(&s->mdct1024);
633 ff_mdct_end(&s->mdct128);
635 ff_psy_preprocess_end(s->psypp);
636 av_freep(&s->samples);
641 AVCodec aac_encoder = {
645 sizeof(AACEncContext),
649 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
650 .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
651 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),