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
153 //Explicitly Mark SBR absent
154 put_bits(&pb, 11, 0x2b7); //sync extension
155 put_bits(&pb, 5, AOT_SBR);
160 static av_cold int aac_encode_init(AVCodecContext *avctx)
162 AACEncContext *s = avctx->priv_data;
164 const uint8_t *sizes[2];
167 avctx->frame_size = 1024;
169 for (i = 0; i < 16; i++)
170 if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
173 av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
176 if (avctx->channels > AAC_MAX_CHANNELS) {
177 av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
180 if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) {
181 av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile);
184 if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) {
185 av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n");
188 s->samplerate_index = i;
190 dsputil_init(&s->dsp, avctx);
191 ff_mdct_init(&s->mdct1024, 11, 0, 1.0);
192 ff_mdct_init(&s->mdct128, 8, 0, 1.0);
194 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
195 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
196 ff_init_ff_sine_windows(10);
197 ff_init_ff_sine_windows(7);
199 s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
200 s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
201 avctx->extradata = av_mallocz(5 + FF_INPUT_BUFFER_PADDING_SIZE);
202 avctx->extradata_size = 5;
203 put_audio_specific_config(avctx);
205 sizes[0] = swb_size_1024[i];
206 sizes[1] = swb_size_128[i];
207 lengths[0] = ff_aac_num_swb_1024[i];
208 lengths[1] = ff_aac_num_swb_128[i];
209 ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
210 s->psypp = ff_psy_preprocess_init(avctx);
211 s->coder = &ff_aac_coders[2];
213 s->lambda = avctx->global_quality ? avctx->global_quality : 120;
220 static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
221 SingleChannelElement *sce, short *audio)
224 const int chans = avctx->channels;
225 const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
226 const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
227 const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
228 float *output = sce->ret;
230 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
231 memcpy(output, sce->saved, sizeof(float)*1024);
232 if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
233 memset(output, 0, sizeof(output[0]) * 448);
234 for (i = 448; i < 576; i++)
235 output[i] = sce->saved[i] * pwindow[i - 448];
236 for (i = 576; i < 704; i++)
237 output[i] = sce->saved[i];
239 if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
240 for (i = 0; i < 1024; i++) {
241 output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1];
242 sce->saved[i] = audio[i * chans] * lwindow[i];
245 for (i = 0; i < 448; i++)
246 output[i+1024] = audio[i * chans];
248 output[i+1024] = audio[i * chans] * swindow[576 - i - 1];
249 memset(output+1024+576, 0, sizeof(output[0]) * 448);
250 for (i = 0; i < 1024; i++)
251 sce->saved[i] = audio[i * chans];
253 ff_mdct_calc(&s->mdct1024, sce->coeffs, output);
255 for (k = 0; k < 1024; k += 128) {
256 for (i = 448 + k; i < 448 + k + 256; i++)
257 output[i - 448 - k] = (i < 1024)
259 : audio[(i-1024)*chans];
260 s->dsp.vector_fmul (output, output, k ? swindow : pwindow, 128);
261 s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
262 ff_mdct_calc(&s->mdct128, sce->coeffs + k, output);
264 for (i = 0; i < 1024; i++)
265 sce->saved[i] = audio[i * chans];
270 * Encode ics_info element.
271 * @see Table 4.6 (syntax of ics_info)
273 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
277 put_bits(&s->pb, 1, 0); // ics_reserved bit
278 put_bits(&s->pb, 2, info->window_sequence[0]);
279 put_bits(&s->pb, 1, info->use_kb_window[0]);
280 if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
281 put_bits(&s->pb, 6, info->max_sfb);
282 put_bits(&s->pb, 1, 0); // no prediction
284 put_bits(&s->pb, 4, info->max_sfb);
285 for (w = 1; w < 8; w++)
286 put_bits(&s->pb, 1, !info->group_len[w]);
292 * @see 4.6.8.1 "Joint Coding - M/S Stereo"
294 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
298 put_bits(pb, 2, cpe->ms_mode);
299 if (cpe->ms_mode == 1)
300 for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
301 for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
302 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
306 * Produce integer coefficients from scalefactors provided by the model.
308 static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
311 int start, maxsfb, cmaxsfb;
313 for (ch = 0; ch < chans; ch++) {
314 IndividualChannelStream *ics = &cpe->ch[ch].ics;
317 cpe->ch[ch].pulse.num_pulse = 0;
318 for (w = 0; w < ics->num_windows*16; w += 16) {
319 for (g = 0; g < ics->num_swb; g++) {
321 if (cpe->common_window && !ch && cpe->ms_mask[w + g]) {
322 for (i = 0; i < ics->swb_sizes[g]; i++) {
323 cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
324 cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
327 start += ics->swb_sizes[g];
329 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
331 maxsfb = FFMAX(maxsfb, cmaxsfb);
333 ics->max_sfb = maxsfb;
335 //adjust zero bands for window groups
336 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
337 for (g = 0; g < ics->max_sfb; g++) {
339 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
340 if (!cpe->ch[ch].zeroes[w2*16 + g]) {
345 cpe->ch[ch].zeroes[w*16 + g] = i;
350 if (chans > 1 && cpe->common_window) {
351 IndividualChannelStream *ics0 = &cpe->ch[0].ics;
352 IndividualChannelStream *ics1 = &cpe->ch[1].ics;
354 ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
355 ics1->max_sfb = ics0->max_sfb;
356 for (w = 0; w < ics0->num_windows*16; w += 16)
357 for (i = 0; i < ics0->max_sfb; i++)
358 if (cpe->ms_mask[w+i])
360 if (msc == 0 || ics0->max_sfb == 0)
363 cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
368 * Encode scalefactor band coding type.
370 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
374 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
375 s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
379 * Encode scalefactors.
381 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
382 SingleChannelElement *sce)
384 int off = sce->sf_idx[0], diff;
387 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
388 for (i = 0; i < sce->ics.max_sfb; i++) {
389 if (!sce->zeroes[w*16 + i]) {
390 diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
391 if (diff < 0 || diff > 120)
392 av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
393 off = sce->sf_idx[w*16 + i];
394 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
403 static void encode_pulses(AACEncContext *s, Pulse *pulse)
407 put_bits(&s->pb, 1, !!pulse->num_pulse);
408 if (!pulse->num_pulse)
411 put_bits(&s->pb, 2, pulse->num_pulse - 1);
412 put_bits(&s->pb, 6, pulse->start);
413 for (i = 0; i < pulse->num_pulse; i++) {
414 put_bits(&s->pb, 5, pulse->pos[i]);
415 put_bits(&s->pb, 4, pulse->amp[i]);
420 * Encode spectral coefficients processed by psychoacoustic model.
422 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
426 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
428 for (i = 0; i < sce->ics.max_sfb; i++) {
429 if (sce->zeroes[w*16 + i]) {
430 start += sce->ics.swb_sizes[i];
433 for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
434 s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
435 sce->ics.swb_sizes[i],
436 sce->sf_idx[w*16 + i],
437 sce->band_type[w*16 + i],
439 start += sce->ics.swb_sizes[i];
445 * Encode one channel of audio data.
447 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
448 SingleChannelElement *sce,
451 put_bits(&s->pb, 8, sce->sf_idx[0]);
453 put_ics_info(s, &sce->ics);
454 encode_band_info(s, sce);
455 encode_scale_factors(avctx, s, sce);
456 encode_pulses(s, &sce->pulse);
457 put_bits(&s->pb, 1, 0); //tns
458 put_bits(&s->pb, 1, 0); //ssr
459 encode_spectral_coeffs(s, sce);
464 * Write some auxiliary information about the created AAC file.
466 static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
469 int i, namelen, padbits;
471 namelen = strlen(name) + 2;
472 put_bits(&s->pb, 3, TYPE_FIL);
473 put_bits(&s->pb, 4, FFMIN(namelen, 15));
475 put_bits(&s->pb, 8, namelen - 16);
476 put_bits(&s->pb, 4, 0); //extension type - filler
477 padbits = 8 - (put_bits_count(&s->pb) & 7);
478 align_put_bits(&s->pb);
479 for (i = 0; i < namelen - 2; i++)
480 put_bits(&s->pb, 8, name[i]);
481 put_bits(&s->pb, 12 - padbits, 0);
484 static int aac_encode_frame(AVCodecContext *avctx,
485 uint8_t *frame, int buf_size, void *data)
487 AACEncContext *s = avctx->priv_data;
488 int16_t *samples = s->samples, *samples2, *la;
490 int i, j, chans, tag, start_ch;
491 const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
492 int chan_el_counter[4];
493 FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
499 memcpy(s->samples + 1024 * avctx->channels, data,
500 1024 * avctx->channels * sizeof(s->samples[0]));
503 samples2 = s->samples + 1024 * avctx->channels;
504 for (i = 0; i < chan_map[0]; i++) {
506 chans = tag == TYPE_CPE ? 2 : 1;
507 ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
508 samples2 + start_ch, start_ch, chans);
513 if (!avctx->frame_number) {
514 memcpy(s->samples, s->samples + 1024 * avctx->channels,
515 1024 * avctx->channels * sizeof(s->samples[0]));
520 for (i = 0; i < chan_map[0]; i++) {
521 FFPsyWindowInfo* wi = windows + start_ch;
523 chans = tag == TYPE_CPE ? 2 : 1;
525 for (j = 0; j < chans; j++) {
526 IndividualChannelStream *ics = &cpe->ch[j].ics;
528 int cur_channel = start_ch + j;
529 samples2 = samples + cur_channel;
530 la = samples2 + (448+64) * avctx->channels;
533 if (tag == TYPE_LFE) {
534 wi[j].window_type[0] = ONLY_LONG_SEQUENCE;
535 wi[j].window_shape = 0;
536 wi[j].num_windows = 1;
537 wi[j].grouping[0] = 1;
539 wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, cur_channel,
540 ics->window_sequence[0]);
542 ics->window_sequence[1] = ics->window_sequence[0];
543 ics->window_sequence[0] = wi[j].window_type[0];
544 ics->use_kb_window[1] = ics->use_kb_window[0];
545 ics->use_kb_window[0] = wi[j].window_shape;
546 ics->num_windows = wi[j].num_windows;
547 ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
548 ics->num_swb = tag == TYPE_LFE ? 12 : s->psy.num_bands[ics->num_windows == 8];
549 for (k = 0; k < ics->num_windows; k++)
550 ics->group_len[k] = wi[j].grouping[k];
552 apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2);
558 init_put_bits(&s->pb, frame, buf_size*8);
559 if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
560 put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
562 memset(chan_el_counter, 0, sizeof(chan_el_counter));
563 for (i = 0; i < chan_map[0]; i++) {
564 FFPsyWindowInfo* wi = windows + start_ch;
566 chans = tag == TYPE_CPE ? 2 : 1;
568 put_bits(&s->pb, 3, tag);
569 put_bits(&s->pb, 4, chan_el_counter[tag]++);
570 for (j = 0; j < chans; j++) {
571 s->cur_channel = start_ch + j;
572 ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
573 s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
575 cpe->common_window = 0;
577 && wi[0].window_type[0] == wi[1].window_type[0]
578 && wi[0].window_shape == wi[1].window_shape) {
580 cpe->common_window = 1;
581 for (j = 0; j < wi[0].num_windows; j++) {
582 if (wi[0].grouping[j] != wi[1].grouping[j]) {
583 cpe->common_window = 0;
588 s->cur_channel = start_ch;
589 if (cpe->common_window && s->coder->search_for_ms)
590 s->coder->search_for_ms(s, cpe, s->lambda);
591 adjust_frame_information(s, cpe, chans);
593 put_bits(&s->pb, 1, cpe->common_window);
594 if (cpe->common_window) {
595 put_ics_info(s, &cpe->ch[0].ics);
596 encode_ms_info(&s->pb, cpe);
599 for (j = 0; j < chans; j++) {
600 s->cur_channel = start_ch + j;
601 encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
606 frame_bits = put_bits_count(&s->pb);
607 if (frame_bits <= 6144 * avctx->channels - 3)
610 s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
614 put_bits(&s->pb, 3, TYPE_END);
615 flush_put_bits(&s->pb);
616 avctx->frame_bits = put_bits_count(&s->pb);
618 // rate control stuff
619 if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
620 float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
622 s->lambda = FFMIN(s->lambda, 65536.f);
627 memcpy(s->samples, s->samples + 1024 * avctx->channels,
628 1024 * avctx->channels * sizeof(s->samples[0]));
629 return put_bits_count(&s->pb)>>3;
632 static av_cold int aac_encode_end(AVCodecContext *avctx)
634 AACEncContext *s = avctx->priv_data;
636 ff_mdct_end(&s->mdct1024);
637 ff_mdct_end(&s->mdct128);
639 ff_psy_preprocess_end(s->psypp);
640 av_freep(&s->samples);
645 AVCodec ff_aac_encoder = {
649 sizeof(AACEncContext),
653 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
654 .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
655 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),