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 ***********************************/
32 #include "libavutil/libm.h"
33 #include "libavutil/thread.h"
34 #include "libavutil/float_dsp.h"
35 #include "libavutil/opt.h"
39 #include "mpeg4audio.h"
46 #include "aacenctab.h"
47 #include "aacenc_utils.h"
51 static AVOnce aac_table_init = AV_ONCE_INIT;
54 * Make AAC audio config object.
55 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
57 static void put_audio_specific_config(AVCodecContext *avctx)
60 AACEncContext *s = avctx->priv_data;
61 int channels = s->channels - (s->channels == 8 ? 1 : 0);
63 init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
64 put_bits(&pb, 5, s->profile+1); //profile
65 put_bits(&pb, 4, s->samplerate_index); //sample rate index
66 put_bits(&pb, 4, channels);
68 put_bits(&pb, 1, 0); //frame length - 1024 samples
69 put_bits(&pb, 1, 0); //does not depend on core coder
70 put_bits(&pb, 1, 0); //is not extension
72 //Explicitly Mark SBR absent
73 put_bits(&pb, 11, 0x2b7); //sync extension
74 put_bits(&pb, 5, AOT_SBR);
79 void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
82 for (sf = 0; sf < 256; sf++) {
83 for (g = 0; g < 128; g++) {
84 s->quantize_band_cost_cache[sf][g].bits = -1;
89 #define WINDOW_FUNC(type) \
90 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
91 SingleChannelElement *sce, \
94 WINDOW_FUNC(only_long)
96 const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
97 const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
98 float *out = sce->ret_buf;
100 fdsp->vector_fmul (out, audio, lwindow, 1024);
101 fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
104 WINDOW_FUNC(long_start)
106 const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
107 const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
108 float *out = sce->ret_buf;
110 fdsp->vector_fmul(out, audio, lwindow, 1024);
111 memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
112 fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
113 memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
116 WINDOW_FUNC(long_stop)
118 const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
119 const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
120 float *out = sce->ret_buf;
122 memset(out, 0, sizeof(out[0]) * 448);
123 fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
124 memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
125 fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
128 WINDOW_FUNC(eight_short)
130 const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
131 const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
132 const float *in = audio + 448;
133 float *out = sce->ret_buf;
136 for (w = 0; w < 8; w++) {
137 fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
140 fdsp->vector_fmul_reverse(out, in, swindow, 128);
145 static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
146 SingleChannelElement *sce,
147 const float *audio) = {
148 [ONLY_LONG_SEQUENCE] = apply_only_long_window,
149 [LONG_START_SEQUENCE] = apply_long_start_window,
150 [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
151 [LONG_STOP_SEQUENCE] = apply_long_stop_window
154 static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
158 const float *output = sce->ret_buf;
160 apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
162 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
163 s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
165 for (i = 0; i < 1024; i += 128)
166 s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
167 memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
168 memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
172 * Encode ics_info element.
173 * @see Table 4.6 (syntax of ics_info)
175 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
179 put_bits(&s->pb, 1, 0); // ics_reserved bit
180 put_bits(&s->pb, 2, info->window_sequence[0]);
181 put_bits(&s->pb, 1, info->use_kb_window[0]);
182 if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
183 put_bits(&s->pb, 6, info->max_sfb);
184 put_bits(&s->pb, 1, !!info->predictor_present);
186 put_bits(&s->pb, 4, info->max_sfb);
187 for (w = 1; w < 8; w++)
188 put_bits(&s->pb, 1, !info->group_len[w]);
194 * @see 4.6.8.1 "Joint Coding - M/S Stereo"
196 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
200 put_bits(pb, 2, cpe->ms_mode);
201 if (cpe->ms_mode == 1)
202 for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
203 for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
204 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
208 * Produce integer coefficients from scalefactors provided by the model.
210 static void adjust_frame_information(ChannelElement *cpe, int chans)
215 for (ch = 0; ch < chans; ch++) {
216 IndividualChannelStream *ics = &cpe->ch[ch].ics;
218 cpe->ch[ch].pulse.num_pulse = 0;
219 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
220 for (w2 = 0; w2 < ics->group_len[w]; w2++) {
221 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
223 maxsfb = FFMAX(maxsfb, cmaxsfb);
226 ics->max_sfb = maxsfb;
228 //adjust zero bands for window groups
229 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
230 for (g = 0; g < ics->max_sfb; g++) {
232 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
233 if (!cpe->ch[ch].zeroes[w2*16 + g]) {
238 cpe->ch[ch].zeroes[w*16 + g] = i;
243 if (chans > 1 && cpe->common_window) {
244 IndividualChannelStream *ics0 = &cpe->ch[0].ics;
245 IndividualChannelStream *ics1 = &cpe->ch[1].ics;
247 ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
248 ics1->max_sfb = ics0->max_sfb;
249 for (w = 0; w < ics0->num_windows*16; w += 16)
250 for (i = 0; i < ics0->max_sfb; i++)
251 if (cpe->ms_mask[w+i])
253 if (msc == 0 || ics0->max_sfb == 0)
256 cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
260 static void apply_intensity_stereo(ChannelElement *cpe)
263 IndividualChannelStream *ics = &cpe->ch[0].ics;
264 if (!cpe->common_window)
266 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
267 for (w2 = 0; w2 < ics->group_len[w]; w2++) {
268 int start = (w+w2) * 128;
269 for (g = 0; g < ics->num_swb; g++) {
270 int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
271 float scale = cpe->ch[0].is_ener[w*16+g];
272 if (!cpe->is_mask[w*16 + g]) {
273 start += ics->swb_sizes[g];
276 if (cpe->ms_mask[w*16 + g])
278 for (i = 0; i < ics->swb_sizes[g]; i++) {
279 float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
280 cpe->ch[0].coeffs[start+i] = sum;
281 cpe->ch[1].coeffs[start+i] = 0.0f;
283 start += ics->swb_sizes[g];
289 static void apply_mid_side_stereo(ChannelElement *cpe)
292 IndividualChannelStream *ics = &cpe->ch[0].ics;
293 if (!cpe->common_window)
295 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
296 for (w2 = 0; w2 < ics->group_len[w]; w2++) {
297 int start = (w+w2) * 128;
298 for (g = 0; g < ics->num_swb; g++) {
299 /* ms_mask can be used for other purposes in PNS and I/S,
300 * so must not apply M/S if any band uses either, even if
303 if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]
304 || cpe->ch[0].band_type[w*16 + g] >= NOISE_BT
305 || cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) {
306 start += ics->swb_sizes[g];
309 for (i = 0; i < ics->swb_sizes[g]; i++) {
310 float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
311 float R = L - cpe->ch[1].coeffs[start+i];
312 cpe->ch[0].coeffs[start+i] = L;
313 cpe->ch[1].coeffs[start+i] = R;
315 start += ics->swb_sizes[g];
322 * Encode scalefactor band coding type.
324 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
328 if (s->coder->set_special_band_scalefactors)
329 s->coder->set_special_band_scalefactors(s, sce);
331 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
332 s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
336 * Encode scalefactors.
338 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
339 SingleChannelElement *sce)
341 int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
342 int off_is = 0, noise_flag = 1;
345 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
346 for (i = 0; i < sce->ics.max_sfb; i++) {
347 if (!sce->zeroes[w*16 + i]) {
348 if (sce->band_type[w*16 + i] == NOISE_BT) {
349 diff = sce->sf_idx[w*16 + i] - off_pns;
350 off_pns = sce->sf_idx[w*16 + i];
351 if (noise_flag-- > 0) {
352 put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
355 } else if (sce->band_type[w*16 + i] == INTENSITY_BT ||
356 sce->band_type[w*16 + i] == INTENSITY_BT2) {
357 diff = sce->sf_idx[w*16 + i] - off_is;
358 off_is = sce->sf_idx[w*16 + i];
360 diff = sce->sf_idx[w*16 + i] - off_sf;
361 off_sf = sce->sf_idx[w*16 + i];
363 diff += SCALE_DIFF_ZERO;
364 av_assert0(diff >= 0 && diff <= 120);
365 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
374 static void encode_pulses(AACEncContext *s, Pulse *pulse)
378 put_bits(&s->pb, 1, !!pulse->num_pulse);
379 if (!pulse->num_pulse)
382 put_bits(&s->pb, 2, pulse->num_pulse - 1);
383 put_bits(&s->pb, 6, pulse->start);
384 for (i = 0; i < pulse->num_pulse; i++) {
385 put_bits(&s->pb, 5, pulse->pos[i]);
386 put_bits(&s->pb, 4, pulse->amp[i]);
391 * Encode spectral coefficients processed by psychoacoustic model.
393 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
397 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
399 for (i = 0; i < sce->ics.max_sfb; i++) {
400 if (sce->zeroes[w*16 + i]) {
401 start += sce->ics.swb_sizes[i];
404 for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
405 s->coder->quantize_and_encode_band(s, &s->pb,
406 &sce->coeffs[start + w2*128],
407 NULL, sce->ics.swb_sizes[i],
408 sce->sf_idx[w*16 + i],
409 sce->band_type[w*16 + i],
411 sce->ics.window_clipping[w]);
413 start += sce->ics.swb_sizes[i];
419 * Downscale spectral coefficients for near-clipping windows to avoid artifacts
421 static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
425 if (sce->ics.clip_avoidance_factor < 1.0f) {
426 for (w = 0; w < sce->ics.num_windows; w++) {
428 for (i = 0; i < sce->ics.max_sfb; i++) {
429 float *swb_coeffs = &sce->coeffs[start + w*128];
430 for (j = 0; j < sce->ics.swb_sizes[i]; j++)
431 swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
432 start += sce->ics.swb_sizes[i];
439 * Encode one channel of audio data.
441 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
442 SingleChannelElement *sce,
445 put_bits(&s->pb, 8, sce->sf_idx[0]);
446 if (!common_window) {
447 put_ics_info(s, &sce->ics);
448 if (s->coder->encode_main_pred)
449 s->coder->encode_main_pred(s, sce);
450 if (s->coder->encode_ltp_info)
451 s->coder->encode_ltp_info(s, sce, 0);
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, !!sce->tns.present);
457 if (s->coder->encode_tns_info)
458 s->coder->encode_tns_info(s, sce);
459 put_bits(&s->pb, 1, 0); //ssr
460 encode_spectral_coeffs(s, sce);
465 * Write some auxiliary information about the created AAC file.
467 static void put_bitstream_info(AACEncContext *s, const char *name)
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 - 14);
476 put_bits(&s->pb, 4, 0); //extension type - filler
477 padbits = -put_bits_count(&s->pb) & 7;
478 avpriv_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);
485 * Copy input samples.
486 * Channels are reordered from libavcodec's default order to AAC order.
488 static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
491 int end = 2048 + (frame ? frame->nb_samples : 0);
492 const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
494 /* copy and remap input samples */
495 for (ch = 0; ch < s->channels; ch++) {
496 /* copy last 1024 samples of previous frame to the start of the current frame */
497 memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
499 /* copy new samples and zero any remaining samples */
501 memcpy(&s->planar_samples[ch][2048],
502 frame->extended_data[channel_map[ch]],
503 frame->nb_samples * sizeof(s->planar_samples[0][0]));
505 memset(&s->planar_samples[ch][end], 0,
506 (3072 - end) * sizeof(s->planar_samples[0][0]));
510 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
511 const AVFrame *frame, int *got_packet_ptr)
513 AACEncContext *s = avctx->priv_data;
514 float **samples = s->planar_samples, *samples2, *la, *overlap;
516 SingleChannelElement *sce;
517 IndividualChannelStream *ics;
518 int i, its, ch, w, chans, tag, start_ch, ret, frame_bits;
519 int target_bits, rate_bits, too_many_bits, too_few_bits;
520 int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
521 int chan_el_counter[4];
522 FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
524 if (s->last_frame == 2)
527 /* add current frame to queue */
529 if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
533 copy_input_samples(s, frame);
535 ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
537 if (!avctx->frame_number)
541 for (i = 0; i < s->chan_map[0]; i++) {
542 FFPsyWindowInfo* wi = windows + start_ch;
543 tag = s->chan_map[i+1];
544 chans = tag == TYPE_CPE ? 2 : 1;
546 for (ch = 0; ch < chans; ch++) {
548 float clip_avoidance_factor;
551 s->cur_channel = start_ch + ch;
552 overlap = &samples[s->cur_channel][0];
553 samples2 = overlap + 1024;
554 la = samples2 + (448+64);
557 if (tag == TYPE_LFE) {
558 wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
559 wi[ch].window_shape = 0;
560 wi[ch].num_windows = 1;
561 wi[ch].grouping[0] = 1;
563 /* Only the lowest 12 coefficients are used in a LFE channel.
564 * The expression below results in only the bottom 8 coefficients
565 * being used for 11.025kHz to 16kHz sample rates.
567 ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
569 wi[ch] = s->psy.model->window(&s->psy, samples2, la, s->cur_channel,
570 ics->window_sequence[0]);
572 ics->window_sequence[1] = ics->window_sequence[0];
573 ics->window_sequence[0] = wi[ch].window_type[0];
574 ics->use_kb_window[1] = ics->use_kb_window[0];
575 ics->use_kb_window[0] = wi[ch].window_shape;
576 ics->num_windows = wi[ch].num_windows;
577 ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
578 ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
579 ics->max_sfb = FFMIN(ics->max_sfb, ics->num_swb);
580 ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
581 ff_swb_offset_128 [s->samplerate_index]:
582 ff_swb_offset_1024[s->samplerate_index];
583 ics->tns_max_bands = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
584 ff_tns_max_bands_128 [s->samplerate_index]:
585 ff_tns_max_bands_1024[s->samplerate_index];
586 clip_avoidance_factor = 0.0f;
587 for (w = 0; w < ics->num_windows; w++)
588 ics->group_len[w] = wi[ch].grouping[w];
589 for (w = 0; w < ics->num_windows; w++) {
590 if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
591 ics->window_clipping[w] = 1;
592 clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
594 ics->window_clipping[w] = 0;
597 if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
598 ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
600 ics->clip_avoidance_factor = 1.0f;
603 apply_window_and_mdct(s, sce, overlap);
605 if (s->options.ltp && s->coder->update_ltp) {
606 s->coder->update_ltp(s, sce);
607 apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]);
608 s->mdct1024.mdct_calc(&s->mdct1024, sce->lcoeffs, sce->ret_buf);
611 for (k = 0; k < 1024; k++) {
612 if (!isfinite(cpe->ch[ch].coeffs[k])) {
613 av_log(avctx, AV_LOG_ERROR, "Input contains NaN/+-Inf\n");
614 return AVERROR(EINVAL);
617 avoid_clipping(s, sce);
621 if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0)
623 frame_bits = its = 0;
625 init_put_bits(&s->pb, avpkt->data, avpkt->size);
627 if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
628 put_bitstream_info(s, LIBAVCODEC_IDENT);
631 memset(chan_el_counter, 0, sizeof(chan_el_counter));
632 for (i = 0; i < s->chan_map[0]; i++) {
633 FFPsyWindowInfo* wi = windows + start_ch;
634 const float *coeffs[2];
635 tag = s->chan_map[i+1];
636 chans = tag == TYPE_CPE ? 2 : 1;
638 cpe->common_window = 0;
639 memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
640 memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
641 put_bits(&s->pb, 3, tag);
642 put_bits(&s->pb, 4, chan_el_counter[tag]++);
643 for (ch = 0; ch < chans; ch++) {
645 coeffs[ch] = sce->coeffs;
646 sce->ics.predictor_present = 0;
647 sce->ics.ltp.present = 0;
648 memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used));
649 memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
650 memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
651 for (w = 0; w < 128; w++)
652 if (sce->band_type[w] > RESERVED_BT)
653 sce->band_type[w] = 0;
655 s->psy.bitres.alloc = -1;
656 s->psy.bitres.bits = s->last_frame_pb_count / s->channels;
657 s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
658 if (s->psy.bitres.alloc > 0) {
659 /* Lambda unused here on purpose, we need to take psy's unscaled allocation */
660 target_bits += s->psy.bitres.alloc
661 * (s->lambda / (avctx->global_quality ? avctx->global_quality : 120));
662 s->psy.bitres.alloc /= chans;
665 for (ch = 0; ch < chans; ch++) {
666 s->cur_channel = start_ch + ch;
667 if (s->options.pns && s->coder->mark_pns)
668 s->coder->mark_pns(s, avctx, &cpe->ch[ch]);
669 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
672 && wi[0].window_type[0] == wi[1].window_type[0]
673 && wi[0].window_shape == wi[1].window_shape) {
675 cpe->common_window = 1;
676 for (w = 0; w < wi[0].num_windows; w++) {
677 if (wi[0].grouping[w] != wi[1].grouping[w]) {
678 cpe->common_window = 0;
683 for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
685 s->cur_channel = start_ch + ch;
686 if (s->options.tns && s->coder->search_for_tns)
687 s->coder->search_for_tns(s, sce);
688 if (s->options.tns && s->coder->apply_tns_filt)
689 s->coder->apply_tns_filt(s, sce);
690 if (sce->tns.present)
692 if (s->options.pns && s->coder->search_for_pns)
693 s->coder->search_for_pns(s, avctx, sce);
695 s->cur_channel = start_ch;
696 if (s->options.intensity_stereo) { /* Intensity Stereo */
697 if (s->coder->search_for_is)
698 s->coder->search_for_is(s, avctx, cpe);
699 if (cpe->is_mode) is_mode = 1;
700 apply_intensity_stereo(cpe);
702 if (s->options.pred) { /* Prediction */
703 for (ch = 0; ch < chans; ch++) {
705 s->cur_channel = start_ch + ch;
706 if (s->options.pred && s->coder->search_for_pred)
707 s->coder->search_for_pred(s, sce);
708 if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
710 if (s->coder->adjust_common_pred)
711 s->coder->adjust_common_pred(s, cpe);
712 for (ch = 0; ch < chans; ch++) {
714 s->cur_channel = start_ch + ch;
715 if (s->options.pred && s->coder->apply_main_pred)
716 s->coder->apply_main_pred(s, sce);
718 s->cur_channel = start_ch;
720 if (s->options.mid_side) { /* Mid/Side stereo */
721 if (s->options.mid_side == -1 && s->coder->search_for_ms)
722 s->coder->search_for_ms(s, cpe);
723 else if (cpe->common_window)
724 memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
725 apply_mid_side_stereo(cpe);
727 adjust_frame_information(cpe, chans);
728 if (s->options.ltp) { /* LTP */
729 for (ch = 0; ch < chans; ch++) {
731 s->cur_channel = start_ch + ch;
732 if (s->coder->search_for_ltp)
733 s->coder->search_for_ltp(s, sce, cpe->common_window);
734 if (sce->ics.ltp.present) pred_mode = 1;
736 s->cur_channel = start_ch;
737 if (s->coder->adjust_common_ltp)
738 s->coder->adjust_common_ltp(s, cpe);
741 put_bits(&s->pb, 1, cpe->common_window);
742 if (cpe->common_window) {
743 put_ics_info(s, &cpe->ch[0].ics);
744 if (s->coder->encode_main_pred)
745 s->coder->encode_main_pred(s, &cpe->ch[0]);
746 if (s->coder->encode_ltp_info)
747 s->coder->encode_ltp_info(s, &cpe->ch[0], 1);
748 encode_ms_info(&s->pb, cpe);
749 if (cpe->ms_mode) ms_mode = 1;
752 for (ch = 0; ch < chans; ch++) {
753 s->cur_channel = start_ch + ch;
754 encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
759 if (avctx->flags & CODEC_FLAG_QSCALE) {
760 /* When using a constant Q-scale, don't mess with lambda */
764 /* rate control stuff
765 * allow between the nominal bitrate, and what psy's bit reservoir says to target
766 * but drift towards the nominal bitrate always
768 frame_bits = put_bits_count(&s->pb);
769 rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate;
770 rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3);
771 too_many_bits = FFMAX(target_bits, rate_bits);
772 too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3);
773 too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits);
775 /* When using ABR, be strict (but only for increasing) */
776 too_few_bits = too_few_bits - too_few_bits/8;
777 too_many_bits = too_many_bits + too_many_bits/2;
779 if ( its == 0 /* for steady-state Q-scale tracking */
780 || (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits))
781 || frame_bits >= 6144 * s->channels - 3 )
783 float ratio = ((float)rate_bits) / frame_bits;
785 if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) {
787 * This path is for steady-state Q-scale tracking
788 * When frame bits fall within the stable range, we still need to adjust
789 * lambda to maintain it like so in a stable fashion (large jumps in lambda
790 * create artifacts and should be avoided), but slowly
792 ratio = sqrtf(sqrtf(ratio));
793 ratio = av_clipf(ratio, 0.9f, 1.1f);
795 /* Not so fast though */
796 ratio = sqrtf(ratio);
798 s->lambda = FFMIN(s->lambda * ratio, 65536.f);
800 /* Keep iterating if we must reduce and lambda is in the sky */
801 if (ratio > 0.9f && ratio < 1.1f) {
804 if (is_mode || ms_mode || tns_mode || pred_mode) {
805 for (i = 0; i < s->chan_map[0]; i++) {
806 // Must restore coeffs
807 chans = tag == TYPE_CPE ? 2 : 1;
809 for (ch = 0; ch < chans; ch++)
810 memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
820 if (s->options.ltp && s->coder->ltp_insert_new_frame)
821 s->coder->ltp_insert_new_frame(s);
823 put_bits(&s->pb, 3, TYPE_END);
824 flush_put_bits(&s->pb);
826 s->last_frame_pb_count = put_bits_count(&s->pb);
828 s->lambda_sum += s->lambda;
834 ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
837 avpkt->size = put_bits_count(&s->pb) >> 3;
842 static av_cold int aac_encode_end(AVCodecContext *avctx)
844 AACEncContext *s = avctx->priv_data;
846 av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_sum / s->lambda_count);
848 ff_mdct_end(&s->mdct1024);
849 ff_mdct_end(&s->mdct128);
853 ff_psy_preprocess_end(s->psypp);
854 av_freep(&s->buffer.samples);
857 ff_af_queue_close(&s->afq);
861 static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
865 s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
867 return AVERROR(ENOMEM);
870 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
871 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
872 ff_init_ff_sine_windows(10);
873 ff_init_ff_sine_windows(7);
875 if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
877 if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0)
883 static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
886 FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);
887 FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);
888 FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
890 for(ch = 0; ch < s->channels; ch++)
891 s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
895 return AVERROR(ENOMEM);
898 static av_cold void aac_encode_init_tables(void)
903 static av_cold int aac_encode_init(AVCodecContext *avctx)
905 AACEncContext *s = avctx->priv_data;
907 const uint8_t *sizes[2];
908 uint8_t grouping[AAC_MAX_CHANNELS];
911 s->channels = avctx->channels;
912 s->chan_map = aac_chan_configs[s->channels-1];
913 s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
914 s->last_frame_pb_count = 0;
915 avctx->extradata_size = 5;
916 avctx->frame_size = 1024;
917 avctx->initial_padding = 1024;
918 avctx->bit_rate = (int)FFMIN(
919 6144 * s->channels / 1024.0 * avctx->sample_rate,
921 avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
924 for (i = 0; i < 16; i++)
925 if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
927 s->samplerate_index = i;
929 ERROR_IF(s->samplerate_index == 16 ||
930 s->samplerate_index >= ff_aac_swb_size_1024_len ||
931 s->samplerate_index >= ff_aac_swb_size_128_len,
932 "Unsupported sample rate %d\n", avctx->sample_rate);
933 ERROR_IF(s->channels > AAC_MAX_CHANNELS || s->channels == 7,
934 "Unsupported number of channels: %d\n", s->channels);
935 WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
936 "Too many bits %f > %d per frame requested, clamping to max\n",
937 1024.0 * avctx->bit_rate / avctx->sample_rate,
940 for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++)
941 if (avctx->profile == aacenc_profiles[i])
943 ERROR_IF(i == FF_ARRAY_ELEMS(aacenc_profiles),
944 "Unsupported encoding profile: %d\n", avctx->profile);
945 if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) {
946 avctx->profile = FF_PROFILE_AAC_LOW;
947 ERROR_IF(s->options.pred,
948 "Main prediction unavailable in the \"mpeg2_aac_low\" profile\n");
949 ERROR_IF(s->options.ltp,
950 "LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n");
951 WARN_IF(s->options.pns,
952 "PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n");
954 } else if (avctx->profile == FF_PROFILE_AAC_LTP) {
956 ERROR_IF(s->options.pred,
957 "Main prediction unavailable in the \"aac_ltp\" profile\n");
958 } else if (avctx->profile == FF_PROFILE_AAC_MAIN) {
960 ERROR_IF(s->options.ltp,
961 "LTP prediction unavailable in the \"aac_main\" profile\n");
962 } else if (s->options.ltp) {
963 avctx->profile = FF_PROFILE_AAC_LTP;
965 "Chainging profile to \"aac_ltp\"\n");
966 ERROR_IF(s->options.pred,
967 "Main prediction unavailable in the \"aac_ltp\" profile\n");
968 } else if (s->options.pred) {
969 avctx->profile = FF_PROFILE_AAC_MAIN;
971 "Chainging profile to \"aac_main\"\n");
972 ERROR_IF(s->options.ltp,
973 "LTP prediction unavailable in the \"aac_main\" profile\n");
975 s->profile = avctx->profile;
976 s->coder = &ff_aac_coders[s->options.coder];
978 if (s->options.coder != AAC_CODER_TWOLOOP) {
979 ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
980 "Coders other than twoloop require -strict -2 and some may be removed in the future\n");
981 s->options.intensity_stereo = 0;
985 ERROR_IF(s->options.ltp && avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
986 "The LPT profile requires experimental compliance, add -strict -2 to enable!\n");
988 if ((ret = dsp_init(avctx, s)) < 0)
991 if ((ret = alloc_buffers(avctx, s)) < 0)
994 put_audio_specific_config(avctx);
996 sizes[0] = ff_aac_swb_size_1024[s->samplerate_index];
997 sizes[1] = ff_aac_swb_size_128[s->samplerate_index];
998 lengths[0] = ff_aac_num_swb_1024[s->samplerate_index];
999 lengths[1] = ff_aac_num_swb_128[s->samplerate_index];
1000 for (i = 0; i < s->chan_map[0]; i++)
1001 grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
1002 if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
1003 s->chan_map[0], grouping)) < 0)
1005 s->psypp = ff_psy_preprocess_init(avctx);
1006 ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);
1007 av_lfg_init(&s->lfg, 0x72adca55);
1010 ff_aac_coder_init_mips(s);
1012 if ((ret = ff_thread_once(&aac_table_init, &aac_encode_init_tables)) != 0)
1013 return AVERROR_UNKNOWN;
1015 ff_af_queue_init(avctx, &s->afq);
1019 aac_encode_end(avctx);
1023 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1024 static const AVOption aacenc_options[] = {
1025 {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
1026 {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1027 {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1028 {"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1029 {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
1030 {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1031 {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1032 {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1033 {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1034 {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1038 static const AVClass aacenc_class = {
1040 av_default_item_name,
1042 LIBAVUTIL_VERSION_INT,
1045 AVCodec ff_aac_encoder = {
1047 .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
1048 .type = AVMEDIA_TYPE_AUDIO,
1049 .id = AV_CODEC_ID_AAC,
1050 .priv_data_size = sizeof(AACEncContext),
1051 .init = aac_encode_init,
1052 .encode2 = aac_encode_frame,
1053 .close = aac_encode_end,
1054 .supported_samplerates = mpeg4audio_sample_rates,
1055 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
1056 .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
1057 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
1058 AV_SAMPLE_FMT_NONE },
1059 .priv_class = &aacenc_class,