3 * Copyright (C) 2008 Konstantin Shishkov
5 * This file is part of Libav.
7 * Libav 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 * Libav 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 Libav; 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 ***********************************/
33 #include "libavutil/opt.h"
37 #include "mpeg4audio.h"
47 #define AAC_MAX_CHANNELS 6
49 static const uint8_t swb_size_1024_96[] = {
50 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
51 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
52 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
55 static const uint8_t swb_size_1024_64[] = {
56 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
57 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
58 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
61 static const uint8_t swb_size_1024_48[] = {
62 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
63 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
64 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
68 static const uint8_t swb_size_1024_32[] = {
69 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
70 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
71 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
74 static const uint8_t swb_size_1024_24[] = {
75 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
76 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
77 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
80 static const uint8_t swb_size_1024_16[] = {
81 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
82 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
83 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
86 static const uint8_t swb_size_1024_8[] = {
87 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
88 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
89 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
92 static const uint8_t *swb_size_1024[] = {
93 swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
94 swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
95 swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
96 swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
99 static const uint8_t swb_size_128_96[] = {
100 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
103 static const uint8_t swb_size_128_48[] = {
104 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
107 static const uint8_t swb_size_128_24[] = {
108 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
111 static const uint8_t swb_size_128_16[] = {
112 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
115 static const uint8_t swb_size_128_8[] = {
116 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
119 static const uint8_t *swb_size_128[] = {
120 /* the last entry on the following row is swb_size_128_64 but is a
121 duplicate of swb_size_128_96 */
122 swb_size_128_96, swb_size_128_96, swb_size_128_96,
123 swb_size_128_48, swb_size_128_48, swb_size_128_48,
124 swb_size_128_24, swb_size_128_24, swb_size_128_16,
125 swb_size_128_16, swb_size_128_16, swb_size_128_8
128 /** default channel configurations */
129 static const uint8_t aac_chan_configs[6][5] = {
130 {1, TYPE_SCE}, // 1 channel - single channel element
131 {1, TYPE_CPE}, // 2 channels - channel pair
132 {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
133 {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
134 {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
135 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
139 * Make AAC audio config object.
140 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
142 static void put_audio_specific_config(AVCodecContext *avctx)
145 AACEncContext *s = avctx->priv_data;
147 init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
148 put_bits(&pb, 5, 2); //object type - AAC-LC
149 put_bits(&pb, 4, s->samplerate_index); //sample rate index
150 put_bits(&pb, 4, avctx->channels);
152 put_bits(&pb, 1, 0); //frame length - 1024 samples
153 put_bits(&pb, 1, 0); //does not depend on core coder
154 put_bits(&pb, 1, 0); //is not extension
156 //Explicitly Mark SBR absent
157 put_bits(&pb, 11, 0x2b7); //sync extension
158 put_bits(&pb, 5, AOT_SBR);
163 static av_cold int aac_encode_init(AVCodecContext *avctx)
165 AACEncContext *s = avctx->priv_data;
167 const uint8_t *sizes[2];
168 uint8_t grouping[AAC_MAX_CHANNELS];
171 avctx->frame_size = 1024;
173 for (i = 0; i < 16; i++)
174 if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
177 av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
180 if (avctx->channels > AAC_MAX_CHANNELS) {
181 av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
184 if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) {
185 av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile);
188 if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) {
189 av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n");
192 s->samplerate_index = i;
194 dsputil_init(&s->dsp, avctx);
195 ff_mdct_init(&s->mdct1024, 11, 0, 1.0);
196 ff_mdct_init(&s->mdct128, 8, 0, 1.0);
198 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
199 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
200 ff_init_ff_sine_windows(10);
201 ff_init_ff_sine_windows(7);
203 s->chan_map = aac_chan_configs[avctx->channels-1];
204 s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
205 s->cpe = av_mallocz(sizeof(ChannelElement) * s->chan_map[0]);
206 avctx->extradata = av_mallocz(5 + FF_INPUT_BUFFER_PADDING_SIZE);
207 avctx->extradata_size = 5;
208 put_audio_specific_config(avctx);
210 sizes[0] = swb_size_1024[i];
211 sizes[1] = swb_size_128[i];
212 lengths[0] = ff_aac_num_swb_1024[i];
213 lengths[1] = ff_aac_num_swb_128[i];
214 for (i = 0; i < s->chan_map[0]; i++)
215 grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
216 ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping);
217 s->psypp = ff_psy_preprocess_init(avctx);
218 s->coder = &ff_aac_coders[2];
220 s->lambda = avctx->global_quality ? avctx->global_quality : 120;
227 static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
228 SingleChannelElement *sce, short *audio)
231 const int chans = avctx->channels;
232 const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
233 const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
234 const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
235 float *output = sce->ret;
237 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
238 memcpy(output, sce->saved, sizeof(float)*1024);
239 if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
240 memset(output, 0, sizeof(output[0]) * 448);
241 for (i = 448; i < 576; i++)
242 output[i] = sce->saved[i] * pwindow[i - 448];
243 for (i = 576; i < 704; i++)
244 output[i] = sce->saved[i];
246 if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
247 for (i = 0; i < 1024; i++) {
248 output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1];
249 sce->saved[i] = audio[i * chans] * lwindow[i];
252 for (i = 0; i < 448; i++)
253 output[i+1024] = audio[i * chans];
255 output[i+1024] = audio[i * chans] * swindow[576 - i - 1];
256 memset(output+1024+576, 0, sizeof(output[0]) * 448);
257 for (i = 0; i < 1024; i++)
258 sce->saved[i] = audio[i * chans];
260 s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
262 for (k = 0; k < 1024; k += 128) {
263 for (i = 448 + k; i < 448 + k + 256; i++)
264 output[i - 448 - k] = (i < 1024)
266 : audio[(i-1024)*chans];
267 s->dsp.vector_fmul (output, output, k ? swindow : pwindow, 128);
268 s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
269 s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + k, output);
271 for (i = 0; i < 1024; i++)
272 sce->saved[i] = audio[i * chans];
277 * Encode ics_info element.
278 * @see Table 4.6 (syntax of ics_info)
280 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
284 put_bits(&s->pb, 1, 0); // ics_reserved bit
285 put_bits(&s->pb, 2, info->window_sequence[0]);
286 put_bits(&s->pb, 1, info->use_kb_window[0]);
287 if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
288 put_bits(&s->pb, 6, info->max_sfb);
289 put_bits(&s->pb, 1, 0); // no prediction
291 put_bits(&s->pb, 4, info->max_sfb);
292 for (w = 1; w < 8; w++)
293 put_bits(&s->pb, 1, !info->group_len[w]);
299 * @see 4.6.8.1 "Joint Coding - M/S Stereo"
301 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
305 put_bits(pb, 2, cpe->ms_mode);
306 if (cpe->ms_mode == 1)
307 for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
308 for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
309 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
313 * Produce integer coefficients from scalefactors provided by the model.
315 static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
318 int start, maxsfb, cmaxsfb;
320 for (ch = 0; ch < chans; ch++) {
321 IndividualChannelStream *ics = &cpe->ch[ch].ics;
324 cpe->ch[ch].pulse.num_pulse = 0;
325 for (w = 0; w < ics->num_windows*16; w += 16) {
326 for (g = 0; g < ics->num_swb; g++) {
328 if (cpe->common_window && !ch && cpe->ms_mask[w + g]) {
329 for (i = 0; i < ics->swb_sizes[g]; i++) {
330 cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
331 cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
334 start += ics->swb_sizes[g];
336 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
338 maxsfb = FFMAX(maxsfb, cmaxsfb);
340 ics->max_sfb = maxsfb;
342 //adjust zero bands for window groups
343 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
344 for (g = 0; g < ics->max_sfb; g++) {
346 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
347 if (!cpe->ch[ch].zeroes[w2*16 + g]) {
352 cpe->ch[ch].zeroes[w*16 + g] = i;
357 if (chans > 1 && cpe->common_window) {
358 IndividualChannelStream *ics0 = &cpe->ch[0].ics;
359 IndividualChannelStream *ics1 = &cpe->ch[1].ics;
361 ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
362 ics1->max_sfb = ics0->max_sfb;
363 for (w = 0; w < ics0->num_windows*16; w += 16)
364 for (i = 0; i < ics0->max_sfb; i++)
365 if (cpe->ms_mask[w+i])
367 if (msc == 0 || ics0->max_sfb == 0)
370 cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
375 * Encode scalefactor band coding type.
377 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
381 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
382 s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
386 * Encode scalefactors.
388 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
389 SingleChannelElement *sce)
391 int off = sce->sf_idx[0], diff;
394 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
395 for (i = 0; i < sce->ics.max_sfb; i++) {
396 if (!sce->zeroes[w*16 + i]) {
397 diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
398 if (diff < 0 || diff > 120)
399 av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
400 off = sce->sf_idx[w*16 + i];
401 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
410 static void encode_pulses(AACEncContext *s, Pulse *pulse)
414 put_bits(&s->pb, 1, !!pulse->num_pulse);
415 if (!pulse->num_pulse)
418 put_bits(&s->pb, 2, pulse->num_pulse - 1);
419 put_bits(&s->pb, 6, pulse->start);
420 for (i = 0; i < pulse->num_pulse; i++) {
421 put_bits(&s->pb, 5, pulse->pos[i]);
422 put_bits(&s->pb, 4, pulse->amp[i]);
427 * Encode spectral coefficients processed by psychoacoustic model.
429 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
433 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
435 for (i = 0; i < sce->ics.max_sfb; i++) {
436 if (sce->zeroes[w*16 + i]) {
437 start += sce->ics.swb_sizes[i];
440 for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
441 s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
442 sce->ics.swb_sizes[i],
443 sce->sf_idx[w*16 + i],
444 sce->band_type[w*16 + i],
446 start += sce->ics.swb_sizes[i];
452 * Encode one channel of audio data.
454 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
455 SingleChannelElement *sce,
458 put_bits(&s->pb, 8, sce->sf_idx[0]);
460 put_ics_info(s, &sce->ics);
461 encode_band_info(s, sce);
462 encode_scale_factors(avctx, s, sce);
463 encode_pulses(s, &sce->pulse);
464 put_bits(&s->pb, 1, 0); //tns
465 put_bits(&s->pb, 1, 0); //ssr
466 encode_spectral_coeffs(s, sce);
471 * Write some auxiliary information about the created AAC file.
473 static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
476 int i, namelen, padbits;
478 namelen = strlen(name) + 2;
479 put_bits(&s->pb, 3, TYPE_FIL);
480 put_bits(&s->pb, 4, FFMIN(namelen, 15));
482 put_bits(&s->pb, 8, namelen - 16);
483 put_bits(&s->pb, 4, 0); //extension type - filler
484 padbits = 8 - (put_bits_count(&s->pb) & 7);
485 align_put_bits(&s->pb);
486 for (i = 0; i < namelen - 2; i++)
487 put_bits(&s->pb, 8, name[i]);
488 put_bits(&s->pb, 12 - padbits, 0);
491 static int aac_encode_frame(AVCodecContext *avctx,
492 uint8_t *frame, int buf_size, void *data)
494 AACEncContext *s = avctx->priv_data;
495 int16_t *samples = s->samples, *samples2, *la;
497 int i, ch, w, g, chans, tag, start_ch;
498 int chan_el_counter[4];
499 FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
505 memcpy(s->samples + 1024 * avctx->channels, data,
506 1024 * avctx->channels * sizeof(s->samples[0]));
509 samples2 = s->samples + 1024 * avctx->channels;
510 for (i = 0; i < s->chan_map[0]; i++) {
511 tag = s->chan_map[i+1];
512 chans = tag == TYPE_CPE ? 2 : 1;
513 ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
514 samples2 + start_ch, start_ch, chans);
519 if (!avctx->frame_number) {
520 memcpy(s->samples, s->samples + 1024 * avctx->channels,
521 1024 * avctx->channels * sizeof(s->samples[0]));
526 for (i = 0; i < s->chan_map[0]; i++) {
527 FFPsyWindowInfo* wi = windows + start_ch;
528 tag = s->chan_map[i+1];
529 chans = tag == TYPE_CPE ? 2 : 1;
531 for (ch = 0; ch < chans; ch++) {
532 IndividualChannelStream *ics = &cpe->ch[ch].ics;
533 int cur_channel = start_ch + ch;
534 samples2 = samples + cur_channel;
535 la = samples2 + (448+64) * avctx->channels;
538 if (tag == TYPE_LFE) {
539 wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
540 wi[ch].window_shape = 0;
541 wi[ch].num_windows = 1;
542 wi[ch].grouping[0] = 1;
544 /* Only the lowest 12 coefficients are used in a LFE channel.
545 * The expression below results in only the bottom 8 coefficients
546 * being used for 11.025kHz to 16kHz sample rates.
548 ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
550 wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
551 ics->window_sequence[0]);
553 ics->window_sequence[1] = ics->window_sequence[0];
554 ics->window_sequence[0] = wi[ch].window_type[0];
555 ics->use_kb_window[1] = ics->use_kb_window[0];
556 ics->use_kb_window[0] = wi[ch].window_shape;
557 ics->num_windows = wi[ch].num_windows;
558 ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
559 ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
560 for (w = 0; w < ics->num_windows; w++)
561 ics->group_len[w] = wi[ch].grouping[w];
563 apply_window_and_mdct(avctx, s, &cpe->ch[ch], samples2);
569 init_put_bits(&s->pb, frame, buf_size*8);
570 if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
571 put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
573 memset(chan_el_counter, 0, sizeof(chan_el_counter));
574 for (i = 0; i < s->chan_map[0]; i++) {
575 FFPsyWindowInfo* wi = windows + start_ch;
576 const float *coeffs[2];
577 tag = s->chan_map[i+1];
578 chans = tag == TYPE_CPE ? 2 : 1;
580 put_bits(&s->pb, 3, tag);
581 put_bits(&s->pb, 4, chan_el_counter[tag]++);
582 for (ch = 0; ch < chans; ch++)
583 coeffs[ch] = cpe->ch[ch].coeffs;
584 s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
585 for (ch = 0; ch < chans; ch++) {
586 s->cur_channel = start_ch * 2 + ch;
587 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
589 cpe->common_window = 0;
591 && wi[0].window_type[0] == wi[1].window_type[0]
592 && wi[0].window_shape == wi[1].window_shape) {
594 cpe->common_window = 1;
595 for (w = 0; w < wi[0].num_windows; w++) {
596 if (wi[0].grouping[w] != wi[1].grouping[w]) {
597 cpe->common_window = 0;
602 s->cur_channel = start_ch * 2;
603 if (s->options.stereo_mode && cpe->common_window) {
604 if (s->options.stereo_mode > 0) {
605 IndividualChannelStream *ics = &cpe->ch[0].ics;
606 for (w = 0; w < ics->num_windows; w += ics->group_len[w])
607 for (g = 0; g < ics->num_swb; g++)
608 cpe->ms_mask[w*16+g] = 1;
609 } else if (s->coder->search_for_ms) {
610 s->coder->search_for_ms(s, cpe, s->lambda);
613 adjust_frame_information(s, cpe, chans);
615 put_bits(&s->pb, 1, cpe->common_window);
616 if (cpe->common_window) {
617 put_ics_info(s, &cpe->ch[0].ics);
618 encode_ms_info(&s->pb, cpe);
621 for (ch = 0; ch < chans; ch++) {
622 s->cur_channel = start_ch + ch;
623 encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
628 frame_bits = put_bits_count(&s->pb);
629 if (frame_bits <= 6144 * avctx->channels - 3) {
630 s->psy.bitres.bits = frame_bits / avctx->channels;
634 s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
638 put_bits(&s->pb, 3, TYPE_END);
639 flush_put_bits(&s->pb);
640 avctx->frame_bits = put_bits_count(&s->pb);
642 // rate control stuff
643 if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
644 float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
646 s->lambda = FFMIN(s->lambda, 65536.f);
651 memcpy(s->samples, s->samples + 1024 * avctx->channels,
652 1024 * avctx->channels * sizeof(s->samples[0]));
653 return put_bits_count(&s->pb)>>3;
656 static av_cold int aac_encode_end(AVCodecContext *avctx)
658 AACEncContext *s = avctx->priv_data;
660 ff_mdct_end(&s->mdct1024);
661 ff_mdct_end(&s->mdct128);
663 ff_psy_preprocess_end(s->psypp);
664 av_freep(&s->samples);
669 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
670 static const AVOption aacenc_options[] = {
671 {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), FF_OPT_TYPE_INT, {.dbl = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
672 {"auto", "Selected by the Encoder", 0, FF_OPT_TYPE_CONST, {.dbl = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
673 {"ms_off", "Disable Mid/Side coding", 0, FF_OPT_TYPE_CONST, {.dbl = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
674 {"ms_force", "Force Mid/Side for the whole frame if possible", 0, FF_OPT_TYPE_CONST, {.dbl = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
678 static const AVClass aacenc_class = {
680 av_default_item_name,
682 LIBAVUTIL_VERSION_INT,
685 AVCodec ff_aac_encoder = {
687 .type = AVMEDIA_TYPE_AUDIO,
689 .priv_data_size = sizeof(AACEncContext),
690 .init = aac_encode_init,
691 .encode = aac_encode_frame,
692 .close = aac_encode_end,
693 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
694 .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
695 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
696 .priv_class = &aacenc_class,