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, 0x27b); //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;
229 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
230 memcpy(s->output, sce->saved, sizeof(float)*1024);
231 if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
232 memset(s->output, 0, sizeof(s->output[0]) * 448);
233 for (i = 448; i < 576; i++)
234 s->output[i] = sce->saved[i] * pwindow[i - 448];
235 for (i = 576; i < 704; i++)
236 s->output[i] = sce->saved[i];
238 if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
239 for (i = 0; i < 1024; i++) {
240 s->output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1];
241 sce->saved[i] = audio[i * chans] * lwindow[i];
244 for (i = 0; i < 448; i++)
245 s->output[i+1024] = audio[i * chans];
247 s->output[i+1024] = audio[i * chans] * swindow[576 - i - 1];
248 memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
249 for (i = 0; i < 1024; i++)
250 sce->saved[i] = audio[i * chans];
252 ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
254 for (k = 0; k < 1024; k += 128) {
255 for (i = 448 + k; i < 448 + k + 256; i++)
256 s->output[i - 448 - k] = (i < 1024)
258 : audio[(i-1024)*chans];
259 s->dsp.vector_fmul (s->output, s->output, k ? swindow : pwindow, 128);
260 s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
261 ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
263 for (i = 0; i < 1024; i++)
264 sce->saved[i] = audio[i * chans];
269 * Encode ics_info element.
270 * @see Table 4.6 (syntax of ics_info)
272 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
276 put_bits(&s->pb, 1, 0); // ics_reserved bit
277 put_bits(&s->pb, 2, info->window_sequence[0]);
278 put_bits(&s->pb, 1, info->use_kb_window[0]);
279 if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
280 put_bits(&s->pb, 6, info->max_sfb);
281 put_bits(&s->pb, 1, 0); // no prediction
283 put_bits(&s->pb, 4, info->max_sfb);
284 for (w = 1; w < 8; w++)
285 put_bits(&s->pb, 1, !info->group_len[w]);
291 * @see 4.6.8.1 "Joint Coding - M/S Stereo"
293 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
297 put_bits(pb, 2, cpe->ms_mode);
298 if (cpe->ms_mode == 1)
299 for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
300 for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
301 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
305 * Produce integer coefficients from scalefactors provided by the model.
307 static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
310 int start, maxsfb, cmaxsfb;
312 for (ch = 0; ch < chans; ch++) {
313 IndividualChannelStream *ics = &cpe->ch[ch].ics;
316 cpe->ch[ch].pulse.num_pulse = 0;
317 for (w = 0; w < ics->num_windows*16; w += 16) {
318 for (g = 0; g < ics->num_swb; g++) {
320 if (cpe->common_window && !ch && cpe->ms_mask[w + g]) {
321 for (i = 0; i < ics->swb_sizes[g]; i++) {
322 cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
323 cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
326 start += ics->swb_sizes[g];
328 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
330 maxsfb = FFMAX(maxsfb, cmaxsfb);
332 ics->max_sfb = maxsfb;
334 //adjust zero bands for window groups
335 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
336 for (g = 0; g < ics->max_sfb; g++) {
338 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
339 if (!cpe->ch[ch].zeroes[w2*16 + g]) {
344 cpe->ch[ch].zeroes[w*16 + g] = i;
349 if (chans > 1 && cpe->common_window) {
350 IndividualChannelStream *ics0 = &cpe->ch[0].ics;
351 IndividualChannelStream *ics1 = &cpe->ch[1].ics;
353 ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
354 ics1->max_sfb = ics0->max_sfb;
355 for (w = 0; w < ics0->num_windows*16; w += 16)
356 for (i = 0; i < ics0->max_sfb; i++)
357 if (cpe->ms_mask[w+i])
359 if (msc == 0 || ics0->max_sfb == 0)
362 cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
367 * Encode scalefactor band coding type.
369 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
373 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
374 s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
378 * Encode scalefactors.
380 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
381 SingleChannelElement *sce)
383 int off = sce->sf_idx[0], diff;
386 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
387 for (i = 0; i < sce->ics.max_sfb; i++) {
388 if (!sce->zeroes[w*16 + i]) {
389 diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
390 if (diff < 0 || diff > 120)
391 av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
392 off = sce->sf_idx[w*16 + i];
393 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
402 static void encode_pulses(AACEncContext *s, Pulse *pulse)
406 put_bits(&s->pb, 1, !!pulse->num_pulse);
407 if (!pulse->num_pulse)
410 put_bits(&s->pb, 2, pulse->num_pulse - 1);
411 put_bits(&s->pb, 6, pulse->start);
412 for (i = 0; i < pulse->num_pulse; i++) {
413 put_bits(&s->pb, 5, pulse->pos[i]);
414 put_bits(&s->pb, 4, pulse->amp[i]);
419 * Encode spectral coefficients processed by psychoacoustic model.
421 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
425 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
427 for (i = 0; i < sce->ics.max_sfb; i++) {
428 if (sce->zeroes[w*16 + i]) {
429 start += sce->ics.swb_sizes[i];
432 for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
433 s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
434 sce->ics.swb_sizes[i],
435 sce->sf_idx[w*16 + i],
436 sce->band_type[w*16 + i],
438 start += sce->ics.swb_sizes[i];
444 * Encode one channel of audio data.
446 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
447 SingleChannelElement *sce,
450 put_bits(&s->pb, 8, sce->sf_idx[0]);
452 put_ics_info(s, &sce->ics);
453 encode_band_info(s, sce);
454 encode_scale_factors(avctx, s, sce);
455 encode_pulses(s, &sce->pulse);
456 put_bits(&s->pb, 1, 0); //tns
457 put_bits(&s->pb, 1, 0); //ssr
458 encode_spectral_coeffs(s, sce);
463 * Write some auxiliary information about the created AAC file.
465 static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
468 int i, namelen, padbits;
470 namelen = strlen(name) + 2;
471 put_bits(&s->pb, 3, TYPE_FIL);
472 put_bits(&s->pb, 4, FFMIN(namelen, 15));
474 put_bits(&s->pb, 8, namelen - 16);
475 put_bits(&s->pb, 4, 0); //extension type - filler
476 padbits = 8 - (put_bits_count(&s->pb) & 7);
477 align_put_bits(&s->pb);
478 for (i = 0; i < namelen - 2; i++)
479 put_bits(&s->pb, 8, name[i]);
480 put_bits(&s->pb, 12 - padbits, 0);
483 static int aac_encode_frame(AVCodecContext *avctx,
484 uint8_t *frame, int buf_size, void *data)
486 AACEncContext *s = avctx->priv_data;
487 int16_t *samples = s->samples, *samples2, *la;
489 int i, j, chans, tag, start_ch;
490 const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
491 int chan_el_counter[4];
492 FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
498 memcpy(s->samples + 1024 * avctx->channels, data,
499 1024 * avctx->channels * sizeof(s->samples[0]));
502 samples2 = s->samples + 1024 * avctx->channels;
503 for (i = 0; i < chan_map[0]; i++) {
505 chans = tag == TYPE_CPE ? 2 : 1;
506 ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
507 samples2 + start_ch, start_ch, chans);
512 if (!avctx->frame_number) {
513 memcpy(s->samples, s->samples + 1024 * avctx->channels,
514 1024 * avctx->channels * sizeof(s->samples[0]));
519 for (i = 0; i < chan_map[0]; i++) {
520 FFPsyWindowInfo* wi = windows + start_ch;
522 chans = tag == TYPE_CPE ? 2 : 1;
524 for (j = 0; j < chans; j++) {
525 IndividualChannelStream *ics = &cpe->ch[j].ics;
527 int cur_channel = start_ch + j;
528 samples2 = samples + cur_channel;
529 la = samples2 + (448+64) * avctx->channels;
532 if (tag == TYPE_LFE) {
533 wi[j].window_type[0] = ONLY_LONG_SEQUENCE;
534 wi[j].window_shape = 0;
535 wi[j].num_windows = 1;
536 wi[j].grouping[0] = 1;
538 wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, cur_channel,
539 ics->window_sequence[0]);
541 ics->window_sequence[1] = ics->window_sequence[0];
542 ics->window_sequence[0] = wi[j].window_type[0];
543 ics->use_kb_window[1] = ics->use_kb_window[0];
544 ics->use_kb_window[0] = wi[j].window_shape;
545 ics->num_windows = wi[j].num_windows;
546 ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
547 ics->num_swb = tag == TYPE_LFE ? 12 : s->psy.num_bands[ics->num_windows == 8];
548 for (k = 0; k < ics->num_windows; k++)
549 ics->group_len[k] = wi[j].grouping[k];
551 apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2);
557 init_put_bits(&s->pb, frame, buf_size*8);
558 if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
559 put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
561 memset(chan_el_counter, 0, sizeof(chan_el_counter));
562 for (i = 0; i < chan_map[0]; i++) {
563 FFPsyWindowInfo* wi = windows + start_ch;
565 chans = tag == TYPE_CPE ? 2 : 1;
567 put_bits(&s->pb, 3, tag);
568 put_bits(&s->pb, 4, chan_el_counter[tag]++);
569 for (j = 0; j < chans; j++) {
570 s->cur_channel = start_ch + j;
571 ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
572 s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
574 cpe->common_window = 0;
576 && wi[0].window_type[0] == wi[1].window_type[0]
577 && wi[0].window_shape == wi[1].window_shape) {
579 cpe->common_window = 1;
580 for (j = 0; j < wi[0].num_windows; j++) {
581 if (wi[0].grouping[j] != wi[1].grouping[j]) {
582 cpe->common_window = 0;
587 s->cur_channel = start_ch;
588 if (cpe->common_window && s->coder->search_for_ms)
589 s->coder->search_for_ms(s, cpe, s->lambda);
590 adjust_frame_information(s, cpe, chans);
592 put_bits(&s->pb, 1, cpe->common_window);
593 if (cpe->common_window) {
594 put_ics_info(s, &cpe->ch[0].ics);
595 encode_ms_info(&s->pb, cpe);
598 for (j = 0; j < chans; j++) {
599 s->cur_channel = start_ch + j;
600 encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
605 frame_bits = put_bits_count(&s->pb);
606 if (frame_bits <= 6144 * avctx->channels - 3)
609 s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
613 put_bits(&s->pb, 3, TYPE_END);
614 flush_put_bits(&s->pb);
615 avctx->frame_bits = put_bits_count(&s->pb);
617 // rate control stuff
618 if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
619 float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
621 s->lambda = FFMIN(s->lambda, 65536.f);
626 memcpy(s->samples, s->samples + 1024 * avctx->channels,
627 1024 * avctx->channels * sizeof(s->samples[0]));
628 return put_bits_count(&s->pb)>>3;
631 static av_cold int aac_encode_end(AVCodecContext *avctx)
633 AACEncContext *s = avctx->priv_data;
635 ff_mdct_end(&s->mdct1024);
636 ff_mdct_end(&s->mdct128);
638 ff_psy_preprocess_end(s->psypp);
639 av_freep(&s->samples);
644 AVCodec aac_encoder = {
648 sizeof(AACEncContext),
652 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
653 .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
654 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),