FFmpeg  4.4
aacenc.c
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1 /*
2  * AAC encoder
3  * Copyright (C) 2008 Konstantin Shishkov
4  *
5  * This file is part of FFmpeg.
6  *
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.
11  *
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.
16  *
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
20  */
21 
22 /**
23  * @file
24  * AAC encoder
25  */
26 
27 /***********************************
28  * TODOs:
29  * add sane pulse detection
30  ***********************************/
31 
32 #include "libavutil/libm.h"
33 #include "libavutil/float_dsp.h"
34 #include "libavutil/opt.h"
35 #include "avcodec.h"
36 #include "put_bits.h"
37 #include "internal.h"
38 #include "mpeg4audio.h"
39 #include "sinewin.h"
40 #include "profiles.h"
41 
42 #include "aac.h"
43 #include "aactab.h"
44 #include "aacenc.h"
45 #include "aacenctab.h"
46 #include "aacenc_utils.h"
47 
48 #include "psymodel.h"
49 
50 static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
51 {
52  int i, j;
53  AACEncContext *s = avctx->priv_data;
54  AACPCEInfo *pce = &s->pce;
55  const int bitexact = avctx->flags & AV_CODEC_FLAG_BITEXACT;
56  const char *aux_data = bitexact ? "Lavc" : LIBAVCODEC_IDENT;
57 
58  put_bits(pb, 4, 0);
59 
60  put_bits(pb, 2, avctx->profile);
61  put_bits(pb, 4, s->samplerate_index);
62 
63  put_bits(pb, 4, pce->num_ele[0]); /* Front */
64  put_bits(pb, 4, pce->num_ele[1]); /* Side */
65  put_bits(pb, 4, pce->num_ele[2]); /* Back */
66  put_bits(pb, 2, pce->num_ele[3]); /* LFE */
67  put_bits(pb, 3, 0); /* Assoc data */
68  put_bits(pb, 4, 0); /* CCs */
69 
70  put_bits(pb, 1, 0); /* Stereo mixdown */
71  put_bits(pb, 1, 0); /* Mono mixdown */
72  put_bits(pb, 1, 0); /* Something else */
73 
74  for (i = 0; i < 4; i++) {
75  for (j = 0; j < pce->num_ele[i]; j++) {
76  if (i < 3)
77  put_bits(pb, 1, pce->pairing[i][j]);
78  put_bits(pb, 4, pce->index[i][j]);
79  }
80  }
81 
82  align_put_bits(pb);
83  put_bits(pb, 8, strlen(aux_data));
84  ff_put_string(pb, aux_data, 0);
85 }
86 
87 /**
88  * Make AAC audio config object.
89  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
90  */
92 {
93  PutBitContext pb;
94  AACEncContext *s = avctx->priv_data;
95  int channels = (!s->needs_pce)*(s->channels - (s->channels == 8 ? 1 : 0));
96  const int max_size = 32;
97 
98  avctx->extradata = av_mallocz(max_size);
99  if (!avctx->extradata)
100  return AVERROR(ENOMEM);
101 
102  init_put_bits(&pb, avctx->extradata, max_size);
103  put_bits(&pb, 5, s->profile+1); //profile
104  put_bits(&pb, 4, s->samplerate_index); //sample rate index
105  put_bits(&pb, 4, channels);
106  //GASpecificConfig
107  put_bits(&pb, 1, 0); //frame length - 1024 samples
108  put_bits(&pb, 1, 0); //does not depend on core coder
109  put_bits(&pb, 1, 0); //is not extension
110  if (s->needs_pce)
111  put_pce(&pb, avctx);
112 
113  //Explicitly Mark SBR absent
114  put_bits(&pb, 11, 0x2b7); //sync extension
115  put_bits(&pb, 5, AOT_SBR);
116  put_bits(&pb, 1, 0);
117  flush_put_bits(&pb);
118  avctx->extradata_size = put_bits_count(&pb) >> 3;
119 
120  return 0;
121 }
122 
124 {
125  ++s->quantize_band_cost_cache_generation;
126  if (s->quantize_band_cost_cache_generation == 0) {
127  memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache));
128  s->quantize_band_cost_cache_generation = 1;
129  }
130 }
131 
132 #define WINDOW_FUNC(type) \
133 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
134  SingleChannelElement *sce, \
135  const float *audio)
136 
137 WINDOW_FUNC(only_long)
138 {
139  const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
140  const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
141  float *out = sce->ret_buf;
142 
143  fdsp->vector_fmul (out, audio, lwindow, 1024);
144  fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
145 }
146 
147 WINDOW_FUNC(long_start)
148 {
149  const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
150  const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
151  float *out = sce->ret_buf;
152 
153  fdsp->vector_fmul(out, audio, lwindow, 1024);
154  memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
155  fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
156  memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
157 }
158 
159 WINDOW_FUNC(long_stop)
160 {
161  const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
162  const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
163  float *out = sce->ret_buf;
164 
165  memset(out, 0, sizeof(out[0]) * 448);
166  fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
167  memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
168  fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
169 }
170 
171 WINDOW_FUNC(eight_short)
172 {
173  const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
174  const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
175  const float *in = audio + 448;
176  float *out = sce->ret_buf;
177  int w;
178 
179  for (w = 0; w < 8; w++) {
180  fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
181  out += 128;
182  in += 128;
183  fdsp->vector_fmul_reverse(out, in, swindow, 128);
184  out += 128;
185  }
186 }
187 
188 static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
190  const float *audio) = {
191  [ONLY_LONG_SEQUENCE] = apply_only_long_window,
192  [LONG_START_SEQUENCE] = apply_long_start_window,
193  [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
194  [LONG_STOP_SEQUENCE] = apply_long_stop_window
195 };
196 
198  float *audio)
199 {
200  int i;
201  const float *output = sce->ret_buf;
202 
203  apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
204 
206  s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
207  else
208  for (i = 0; i < 1024; i += 128)
209  s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
210  memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
211  memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
212 }
213 
214 /**
215  * Encode ics_info element.
216  * @see Table 4.6 (syntax of ics_info)
217  */
219 {
220  int w;
221 
222  put_bits(&s->pb, 1, 0); // ics_reserved bit
223  put_bits(&s->pb, 2, info->window_sequence[0]);
224  put_bits(&s->pb, 1, info->use_kb_window[0]);
225  if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
226  put_bits(&s->pb, 6, info->max_sfb);
227  put_bits(&s->pb, 1, !!info->predictor_present);
228  } else {
229  put_bits(&s->pb, 4, info->max_sfb);
230  for (w = 1; w < 8; w++)
231  put_bits(&s->pb, 1, !info->group_len[w]);
232  }
233 }
234 
235 /**
236  * Encode MS data.
237  * @see 4.6.8.1 "Joint Coding - M/S Stereo"
238  */
240 {
241  int i, w;
242 
243  put_bits(pb, 2, cpe->ms_mode);
244  if (cpe->ms_mode == 1)
245  for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
246  for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
247  put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
248 }
249 
250 /**
251  * Produce integer coefficients from scalefactors provided by the model.
252  */
253 static void adjust_frame_information(ChannelElement *cpe, int chans)
254 {
255  int i, w, w2, g, ch;
256  int maxsfb, cmaxsfb;
257 
258  for (ch = 0; ch < chans; ch++) {
259  IndividualChannelStream *ics = &cpe->ch[ch].ics;
260  maxsfb = 0;
261  cpe->ch[ch].pulse.num_pulse = 0;
262  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
263  for (w2 = 0; w2 < ics->group_len[w]; w2++) {
264  for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
265  ;
266  maxsfb = FFMAX(maxsfb, cmaxsfb);
267  }
268  }
269  ics->max_sfb = maxsfb;
270 
271  //adjust zero bands for window groups
272  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
273  for (g = 0; g < ics->max_sfb; g++) {
274  i = 1;
275  for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
276  if (!cpe->ch[ch].zeroes[w2*16 + g]) {
277  i = 0;
278  break;
279  }
280  }
281  cpe->ch[ch].zeroes[w*16 + g] = i;
282  }
283  }
284  }
285 
286  if (chans > 1 && cpe->common_window) {
287  IndividualChannelStream *ics0 = &cpe->ch[0].ics;
288  IndividualChannelStream *ics1 = &cpe->ch[1].ics;
289  int msc = 0;
290  ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
291  ics1->max_sfb = ics0->max_sfb;
292  for (w = 0; w < ics0->num_windows*16; w += 16)
293  for (i = 0; i < ics0->max_sfb; i++)
294  if (cpe->ms_mask[w+i])
295  msc++;
296  if (msc == 0 || ics0->max_sfb == 0)
297  cpe->ms_mode = 0;
298  else
299  cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
300  }
301 }
302 
304 {
305  int w, w2, g, i;
306  IndividualChannelStream *ics = &cpe->ch[0].ics;
307  if (!cpe->common_window)
308  return;
309  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
310  for (w2 = 0; w2 < ics->group_len[w]; w2++) {
311  int start = (w+w2) * 128;
312  for (g = 0; g < ics->num_swb; g++) {
313  int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
314  float scale = cpe->ch[0].is_ener[w*16+g];
315  if (!cpe->is_mask[w*16 + g]) {
316  start += ics->swb_sizes[g];
317  continue;
318  }
319  if (cpe->ms_mask[w*16 + g])
320  p *= -1;
321  for (i = 0; i < ics->swb_sizes[g]; i++) {
322  float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
323  cpe->ch[0].coeffs[start+i] = sum;
324  cpe->ch[1].coeffs[start+i] = 0.0f;
325  }
326  start += ics->swb_sizes[g];
327  }
328  }
329  }
330 }
331 
333 {
334  int w, w2, g, i;
335  IndividualChannelStream *ics = &cpe->ch[0].ics;
336  if (!cpe->common_window)
337  return;
338  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
339  for (w2 = 0; w2 < ics->group_len[w]; w2++) {
340  int start = (w+w2) * 128;
341  for (g = 0; g < ics->num_swb; g++) {
342  /* ms_mask can be used for other purposes in PNS and I/S,
343  * so must not apply M/S if any band uses either, even if
344  * ms_mask is set.
345  */
346  if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]
347  || cpe->ch[0].band_type[w*16 + g] >= NOISE_BT
348  || cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) {
349  start += ics->swb_sizes[g];
350  continue;
351  }
352  for (i = 0; i < ics->swb_sizes[g]; i++) {
353  float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
354  float R = L - cpe->ch[1].coeffs[start+i];
355  cpe->ch[0].coeffs[start+i] = L;
356  cpe->ch[1].coeffs[start+i] = R;
357  }
358  start += ics->swb_sizes[g];
359  }
360  }
361  }
362 }
363 
364 /**
365  * Encode scalefactor band coding type.
366  */
368 {
369  int w;
370 
371  if (s->coder->set_special_band_scalefactors)
372  s->coder->set_special_band_scalefactors(s, sce);
373 
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);
376 }
377 
378 /**
379  * Encode scalefactors.
380  */
383 {
384  int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
385  int off_is = 0, noise_flag = 1;
386  int i, w;
387 
388  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
389  for (i = 0; i < sce->ics.max_sfb; i++) {
390  if (!sce->zeroes[w*16 + i]) {
391  if (sce->band_type[w*16 + i] == NOISE_BT) {
392  diff = sce->sf_idx[w*16 + i] - off_pns;
393  off_pns = sce->sf_idx[w*16 + i];
394  if (noise_flag-- > 0) {
396  continue;
397  }
398  } else if (sce->band_type[w*16 + i] == INTENSITY_BT ||
399  sce->band_type[w*16 + i] == INTENSITY_BT2) {
400  diff = sce->sf_idx[w*16 + i] - off_is;
401  off_is = sce->sf_idx[w*16 + i];
402  } else {
403  diff = sce->sf_idx[w*16 + i] - off_sf;
404  off_sf = sce->sf_idx[w*16 + i];
405  }
407  av_assert0(diff >= 0 && diff <= 120);
409  }
410  }
411  }
412 }
413 
414 /**
415  * Encode pulse data.
416  */
417 static void encode_pulses(AACEncContext *s, Pulse *pulse)
418 {
419  int i;
420 
421  put_bits(&s->pb, 1, !!pulse->num_pulse);
422  if (!pulse->num_pulse)
423  return;
424 
425  put_bits(&s->pb, 2, pulse->num_pulse - 1);
426  put_bits(&s->pb, 6, pulse->start);
427  for (i = 0; i < pulse->num_pulse; i++) {
428  put_bits(&s->pb, 5, pulse->pos[i]);
429  put_bits(&s->pb, 4, pulse->amp[i]);
430  }
431 }
432 
433 /**
434  * Encode spectral coefficients processed by psychoacoustic model.
435  */
437 {
438  int start, i, w, w2;
439 
440  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
441  start = 0;
442  for (i = 0; i < sce->ics.max_sfb; i++) {
443  if (sce->zeroes[w*16 + i]) {
444  start += sce->ics.swb_sizes[i];
445  continue;
446  }
447  for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
448  s->coder->quantize_and_encode_band(s, &s->pb,
449  &sce->coeffs[start + w2*128],
450  NULL, sce->ics.swb_sizes[i],
451  sce->sf_idx[w*16 + i],
452  sce->band_type[w*16 + i],
453  s->lambda,
454  sce->ics.window_clipping[w]);
455  }
456  start += sce->ics.swb_sizes[i];
457  }
458  }
459 }
460 
461 /**
462  * Downscale spectral coefficients for near-clipping windows to avoid artifacts
463  */
465 {
466  int start, i, j, w;
467 
468  if (sce->ics.clip_avoidance_factor < 1.0f) {
469  for (w = 0; w < sce->ics.num_windows; w++) {
470  start = 0;
471  for (i = 0; i < sce->ics.max_sfb; i++) {
472  float *swb_coeffs = &sce->coeffs[start + w*128];
473  for (j = 0; j < sce->ics.swb_sizes[i]; j++)
474  swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
475  start += sce->ics.swb_sizes[i];
476  }
477  }
478  }
479 }
480 
481 /**
482  * Encode one channel of audio data.
483  */
486  int common_window)
487 {
488  put_bits(&s->pb, 8, sce->sf_idx[0]);
489  if (!common_window) {
490  put_ics_info(s, &sce->ics);
491  if (s->coder->encode_main_pred)
492  s->coder->encode_main_pred(s, sce);
493  if (s->coder->encode_ltp_info)
494  s->coder->encode_ltp_info(s, sce, 0);
495  }
496  encode_band_info(s, sce);
497  encode_scale_factors(avctx, s, sce);
498  encode_pulses(s, &sce->pulse);
499  put_bits(&s->pb, 1, !!sce->tns.present);
500  if (s->coder->encode_tns_info)
501  s->coder->encode_tns_info(s, sce);
502  put_bits(&s->pb, 1, 0); //ssr
504  return 0;
505 }
506 
507 /**
508  * Write some auxiliary information about the created AAC file.
509  */
510 static void put_bitstream_info(AACEncContext *s, const char *name)
511 {
512  int i, namelen, padbits;
513 
514  namelen = strlen(name) + 2;
515  put_bits(&s->pb, 3, TYPE_FIL);
516  put_bits(&s->pb, 4, FFMIN(namelen, 15));
517  if (namelen >= 15)
518  put_bits(&s->pb, 8, namelen - 14);
519  put_bits(&s->pb, 4, 0); //extension type - filler
520  padbits = -put_bits_count(&s->pb) & 7;
521  align_put_bits(&s->pb);
522  for (i = 0; i < namelen - 2; i++)
523  put_bits(&s->pb, 8, name[i]);
524  put_bits(&s->pb, 12 - padbits, 0);
525 }
526 
527 /*
528  * Copy input samples.
529  * Channels are reordered from libavcodec's default order to AAC order.
530  */
532 {
533  int ch;
534  int end = 2048 + (frame ? frame->nb_samples : 0);
535  const uint8_t *channel_map = s->reorder_map;
536 
537  /* copy and remap input samples */
538  for (ch = 0; ch < s->channels; ch++) {
539  /* copy last 1024 samples of previous frame to the start of the current frame */
540  memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
541 
542  /* copy new samples and zero any remaining samples */
543  if (frame) {
544  memcpy(&s->planar_samples[ch][2048],
545  frame->extended_data[channel_map[ch]],
546  frame->nb_samples * sizeof(s->planar_samples[0][0]));
547  }
548  memset(&s->planar_samples[ch][end], 0,
549  (3072 - end) * sizeof(s->planar_samples[0][0]));
550  }
551 }
552 
553 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
554  const AVFrame *frame, int *got_packet_ptr)
555 {
556  AACEncContext *s = avctx->priv_data;
557  float **samples = s->planar_samples, *samples2, *la, *overlap;
558  ChannelElement *cpe;
561  int i, its, ch, w, chans, tag, start_ch, ret, frame_bits;
562  int target_bits, rate_bits, too_many_bits, too_few_bits;
563  int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
564  int chan_el_counter[4];
566 
567  /* add current frame to queue */
568  if (frame) {
569  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
570  return ret;
571  } else {
572  if (!s->afq.remaining_samples || (!s->afq.frame_alloc && !s->afq.frame_count))
573  return 0;
574  }
575 
577  if (s->psypp)
578  ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
579 
580  if (!avctx->frame_number)
581  return 0;
582 
583  start_ch = 0;
584  for (i = 0; i < s->chan_map[0]; i++) {
585  FFPsyWindowInfo* wi = windows + start_ch;
586  tag = s->chan_map[i+1];
587  chans = tag == TYPE_CPE ? 2 : 1;
588  cpe = &s->cpe[i];
589  for (ch = 0; ch < chans; ch++) {
590  int k;
591  float clip_avoidance_factor;
592  sce = &cpe->ch[ch];
593  ics = &sce->ics;
594  s->cur_channel = start_ch + ch;
595  overlap = &samples[s->cur_channel][0];
596  samples2 = overlap + 1024;
597  la = samples2 + (448+64);
598  if (!frame)
599  la = NULL;
600  if (tag == TYPE_LFE) {
601  wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE;
602  wi[ch].window_shape = 0;
603  wi[ch].num_windows = 1;
604  wi[ch].grouping[0] = 1;
605  wi[ch].clipping[0] = 0;
606 
607  /* Only the lowest 12 coefficients are used in a LFE channel.
608  * The expression below results in only the bottom 8 coefficients
609  * being used for 11.025kHz to 16kHz sample rates.
610  */
611  ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
612  } else {
613  wi[ch] = s->psy.model->window(&s->psy, samples2, la, s->cur_channel,
614  ics->window_sequence[0]);
615  }
616  ics->window_sequence[1] = ics->window_sequence[0];
617  ics->window_sequence[0] = wi[ch].window_type[0];
618  ics->use_kb_window[1] = ics->use_kb_window[0];
619  ics->use_kb_window[0] = wi[ch].window_shape;
620  ics->num_windows = wi[ch].num_windows;
621  ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
622  ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
623  ics->max_sfb = FFMIN(ics->max_sfb, ics->num_swb);
624  ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
625  ff_swb_offset_128 [s->samplerate_index]:
626  ff_swb_offset_1024[s->samplerate_index];
627  ics->tns_max_bands = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
628  ff_tns_max_bands_128 [s->samplerate_index]:
629  ff_tns_max_bands_1024[s->samplerate_index];
630 
631  for (w = 0; w < ics->num_windows; w++)
632  ics->group_len[w] = wi[ch].grouping[w];
633 
634  /* Calculate input sample maximums and evaluate clipping risk */
635  clip_avoidance_factor = 0.0f;
636  for (w = 0; w < ics->num_windows; w++) {
637  const float *wbuf = overlap + w * 128;
638  const int wlen = 2048 / ics->num_windows;
639  float max = 0;
640  int j;
641  /* mdct input is 2 * output */
642  for (j = 0; j < wlen; j++)
643  max = FFMAX(max, fabsf(wbuf[j]));
644  wi[ch].clipping[w] = max;
645  }
646  for (w = 0; w < ics->num_windows; w++) {
647  if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
648  ics->window_clipping[w] = 1;
649  clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
650  } else {
651  ics->window_clipping[w] = 0;
652  }
653  }
654  if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
655  ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
656  } else {
657  ics->clip_avoidance_factor = 1.0f;
658  }
659 
660  apply_window_and_mdct(s, sce, overlap);
661 
662  if (s->options.ltp && s->coder->update_ltp) {
663  s->coder->update_ltp(s, sce);
664  apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]);
665  s->mdct1024.mdct_calc(&s->mdct1024, sce->lcoeffs, sce->ret_buf);
666  }
667 
668  for (k = 0; k < 1024; k++) {
669  if (!(fabs(cpe->ch[ch].coeffs[k]) < 1E16)) { // Ensure headroom for energy calculation
670  av_log(avctx, AV_LOG_ERROR, "Input contains (near) NaN/+-Inf\n");
671  return AVERROR(EINVAL);
672  }
673  }
674  avoid_clipping(s, sce);
675  }
676  start_ch += chans;
677  }
678  if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0)
679  return ret;
680  frame_bits = its = 0;
681  do {
682  init_put_bits(&s->pb, avpkt->data, avpkt->size);
683 
684  if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
686  start_ch = 0;
687  target_bits = 0;
688  memset(chan_el_counter, 0, sizeof(chan_el_counter));
689  for (i = 0; i < s->chan_map[0]; i++) {
690  FFPsyWindowInfo* wi = windows + start_ch;
691  const float *coeffs[2];
692  tag = s->chan_map[i+1];
693  chans = tag == TYPE_CPE ? 2 : 1;
694  cpe = &s->cpe[i];
695  cpe->common_window = 0;
696  memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
697  memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
698  put_bits(&s->pb, 3, tag);
699  put_bits(&s->pb, 4, chan_el_counter[tag]++);
700  for (ch = 0; ch < chans; ch++) {
701  sce = &cpe->ch[ch];
702  coeffs[ch] = sce->coeffs;
703  sce->ics.predictor_present = 0;
704  sce->ics.ltp.present = 0;
705  memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used));
706  memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
707  memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
708  for (w = 0; w < 128; w++)
709  if (sce->band_type[w] > RESERVED_BT)
710  sce->band_type[w] = 0;
711  }
712  s->psy.bitres.alloc = -1;
713  s->psy.bitres.bits = s->last_frame_pb_count / s->channels;
714  s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
715  if (s->psy.bitres.alloc > 0) {
716  /* Lambda unused here on purpose, we need to take psy's unscaled allocation */
717  target_bits += s->psy.bitres.alloc
718  * (s->lambda / (avctx->global_quality ? avctx->global_quality : 120));
719  s->psy.bitres.alloc /= chans;
720  }
721  s->cur_type = tag;
722  for (ch = 0; ch < chans; ch++) {
723  s->cur_channel = start_ch + ch;
724  if (s->options.pns && s->coder->mark_pns)
725  s->coder->mark_pns(s, avctx, &cpe->ch[ch]);
726  s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
727  }
728  if (chans > 1
729  && wi[0].window_type[0] == wi[1].window_type[0]
730  && wi[0].window_shape == wi[1].window_shape) {
731 
732  cpe->common_window = 1;
733  for (w = 0; w < wi[0].num_windows; w++) {
734  if (wi[0].grouping[w] != wi[1].grouping[w]) {
735  cpe->common_window = 0;
736  break;
737  }
738  }
739  }
740  for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
741  sce = &cpe->ch[ch];
742  s->cur_channel = start_ch + ch;
743  if (s->options.tns && s->coder->search_for_tns)
744  s->coder->search_for_tns(s, sce);
745  if (s->options.tns && s->coder->apply_tns_filt)
746  s->coder->apply_tns_filt(s, sce);
747  if (sce->tns.present)
748  tns_mode = 1;
749  if (s->options.pns && s->coder->search_for_pns)
750  s->coder->search_for_pns(s, avctx, sce);
751  }
752  s->cur_channel = start_ch;
753  if (s->options.intensity_stereo) { /* Intensity Stereo */
754  if (s->coder->search_for_is)
755  s->coder->search_for_is(s, avctx, cpe);
756  if (cpe->is_mode) is_mode = 1;
758  }
759  if (s->options.pred) { /* Prediction */
760  for (ch = 0; ch < chans; ch++) {
761  sce = &cpe->ch[ch];
762  s->cur_channel = start_ch + ch;
763  if (s->options.pred && s->coder->search_for_pred)
764  s->coder->search_for_pred(s, sce);
765  if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
766  }
767  if (s->coder->adjust_common_pred)
768  s->coder->adjust_common_pred(s, cpe);
769  for (ch = 0; ch < chans; ch++) {
770  sce = &cpe->ch[ch];
771  s->cur_channel = start_ch + ch;
772  if (s->options.pred && s->coder->apply_main_pred)
773  s->coder->apply_main_pred(s, sce);
774  }
775  s->cur_channel = start_ch;
776  }
777  if (s->options.mid_side) { /* Mid/Side stereo */
778  if (s->options.mid_side == -1 && s->coder->search_for_ms)
779  s->coder->search_for_ms(s, cpe);
780  else if (cpe->common_window)
781  memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
783  }
784  adjust_frame_information(cpe, chans);
785  if (s->options.ltp) { /* LTP */
786  for (ch = 0; ch < chans; ch++) {
787  sce = &cpe->ch[ch];
788  s->cur_channel = start_ch + ch;
789  if (s->coder->search_for_ltp)
790  s->coder->search_for_ltp(s, sce, cpe->common_window);
791  if (sce->ics.ltp.present) pred_mode = 1;
792  }
793  s->cur_channel = start_ch;
794  if (s->coder->adjust_common_ltp)
795  s->coder->adjust_common_ltp(s, cpe);
796  }
797  if (chans == 2) {
798  put_bits(&s->pb, 1, cpe->common_window);
799  if (cpe->common_window) {
800  put_ics_info(s, &cpe->ch[0].ics);
801  if (s->coder->encode_main_pred)
802  s->coder->encode_main_pred(s, &cpe->ch[0]);
803  if (s->coder->encode_ltp_info)
804  s->coder->encode_ltp_info(s, &cpe->ch[0], 1);
805  encode_ms_info(&s->pb, cpe);
806  if (cpe->ms_mode) ms_mode = 1;
807  }
808  }
809  for (ch = 0; ch < chans; ch++) {
810  s->cur_channel = start_ch + ch;
811  encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
812  }
813  start_ch += chans;
814  }
815 
816  if (avctx->flags & AV_CODEC_FLAG_QSCALE) {
817  /* When using a constant Q-scale, don't mess with lambda */
818  break;
819  }
820 
821  /* rate control stuff
822  * allow between the nominal bitrate, and what psy's bit reservoir says to target
823  * but drift towards the nominal bitrate always
824  */
825  frame_bits = put_bits_count(&s->pb);
826  rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate;
827  rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3);
828  too_many_bits = FFMAX(target_bits, rate_bits);
829  too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3);
830  too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits);
831 
832  /* When using ABR, be strict (but only for increasing) */
833  too_few_bits = too_few_bits - too_few_bits/8;
834  too_many_bits = too_many_bits + too_many_bits/2;
835 
836  if ( its == 0 /* for steady-state Q-scale tracking */
837  || (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits))
838  || frame_bits >= 6144 * s->channels - 3 )
839  {
840  float ratio = ((float)rate_bits) / frame_bits;
841 
842  if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) {
843  /*
844  * This path is for steady-state Q-scale tracking
845  * When frame bits fall within the stable range, we still need to adjust
846  * lambda to maintain it like so in a stable fashion (large jumps in lambda
847  * create artifacts and should be avoided), but slowly
848  */
849  ratio = sqrtf(sqrtf(ratio));
850  ratio = av_clipf(ratio, 0.9f, 1.1f);
851  } else {
852  /* Not so fast though */
853  ratio = sqrtf(ratio);
854  }
855  s->lambda = FFMIN(s->lambda * ratio, 65536.f);
856 
857  /* Keep iterating if we must reduce and lambda is in the sky */
858  if (ratio > 0.9f && ratio < 1.1f) {
859  break;
860  } else {
861  if (is_mode || ms_mode || tns_mode || pred_mode) {
862  for (i = 0; i < s->chan_map[0]; i++) {
863  // Must restore coeffs
864  chans = tag == TYPE_CPE ? 2 : 1;
865  cpe = &s->cpe[i];
866  for (ch = 0; ch < chans; ch++)
867  memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
868  }
869  }
870  its++;
871  }
872  } else {
873  break;
874  }
875  } while (1);
876 
877  if (s->options.ltp && s->coder->ltp_insert_new_frame)
878  s->coder->ltp_insert_new_frame(s);
879 
880  put_bits(&s->pb, 3, TYPE_END);
881  flush_put_bits(&s->pb);
882 
883  s->last_frame_pb_count = put_bits_count(&s->pb);
884 
885  s->lambda_sum += s->lambda;
886  s->lambda_count++;
887 
888  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
889  &avpkt->duration);
890 
891  avpkt->size = put_bits_count(&s->pb) >> 3;
892  *got_packet_ptr = 1;
893  return 0;
894 }
895 
897 {
898  AACEncContext *s = avctx->priv_data;
899 
900  av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_sum / s->lambda_count);
901 
902  ff_mdct_end(&s->mdct1024);
903  ff_mdct_end(&s->mdct128);
904  ff_psy_end(&s->psy);
905  ff_lpc_end(&s->lpc);
906  if (s->psypp)
907  ff_psy_preprocess_end(s->psypp);
908  av_freep(&s->buffer.samples);
909  av_freep(&s->cpe);
910  av_freep(&s->fdsp);
911  ff_af_queue_close(&s->afq);
912  return 0;
913 }
914 
916 {
917  int ret = 0;
918 
920  if (!s->fdsp)
921  return AVERROR(ENOMEM);
922 
923  // window init
925 
926  if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
927  return ret;
928  if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0)
929  return ret;
930 
931  return 0;
932 }
933 
935 {
936  int ch;
937  if (!FF_ALLOCZ_TYPED_ARRAY(s->buffer.samples, s->channels * 3 * 1024) ||
938  !FF_ALLOCZ_TYPED_ARRAY(s->cpe, s->chan_map[0]))
939  return AVERROR(ENOMEM);
940 
941  for(ch = 0; ch < s->channels; ch++)
942  s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
943 
944  return 0;
945 }
946 
948 {
949  AACEncContext *s = avctx->priv_data;
950  int i, ret = 0;
951  const uint8_t *sizes[2];
952  uint8_t grouping[AAC_MAX_CHANNELS];
953  int lengths[2];
954 
955  /* Constants */
956  s->last_frame_pb_count = 0;
957  avctx->frame_size = 1024;
958  avctx->initial_padding = 1024;
959  s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
960 
961  /* Channel map and unspecified bitrate guessing */
962  s->channels = avctx->channels;
963 
964  s->needs_pce = 1;
965  for (i = 0; i < FF_ARRAY_ELEMS(aac_normal_chan_layouts); i++) {
966  if (avctx->channel_layout == aac_normal_chan_layouts[i]) {
967  s->needs_pce = s->options.pce;
968  break;
969  }
970  }
971 
972  if (s->needs_pce) {
973  char buf[64];
974  for (i = 0; i < FF_ARRAY_ELEMS(aac_pce_configs); i++)
975  if (avctx->channel_layout == aac_pce_configs[i].layout)
976  break;
977  av_get_channel_layout_string(buf, sizeof(buf), -1, avctx->channel_layout);
978  ERROR_IF(i == FF_ARRAY_ELEMS(aac_pce_configs), "Unsupported channel layout \"%s\"\n", buf);
979  av_log(avctx, AV_LOG_INFO, "Using a PCE to encode channel layout \"%s\"\n", buf);
980  s->pce = aac_pce_configs[i];
981  s->reorder_map = s->pce.reorder_map;
982  s->chan_map = s->pce.config_map;
983  } else {
984  s->reorder_map = aac_chan_maps[s->channels - 1];
985  s->chan_map = aac_chan_configs[s->channels - 1];
986  }
987 
988  if (!avctx->bit_rate) {
989  for (i = 1; i <= s->chan_map[0]; i++) {
990  avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */
991  s->chan_map[i] == TYPE_LFE ? 16000 : /* LFE */
992  69000 ; /* SCE */
993  }
994  }
995 
996  /* Samplerate */
997  for (i = 0; i < 16; i++)
999  break;
1000  s->samplerate_index = i;
1001  ERROR_IF(s->samplerate_index == 16 ||
1002  s->samplerate_index >= ff_aac_swb_size_1024_len ||
1003  s->samplerate_index >= ff_aac_swb_size_128_len,
1004  "Unsupported sample rate %d\n", avctx->sample_rate);
1005 
1006  /* Bitrate limiting */
1007  WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
1008  "Too many bits %f > %d per frame requested, clamping to max\n",
1009  1024.0 * avctx->bit_rate / avctx->sample_rate,
1010  6144 * s->channels);
1011  avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate,
1012  avctx->bit_rate);
1013 
1014  /* Profile and option setting */
1015  avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
1016  avctx->profile;
1017  for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++)
1018  if (avctx->profile == aacenc_profiles[i])
1019  break;
1020  if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) {
1021  avctx->profile = FF_PROFILE_AAC_LOW;
1022  ERROR_IF(s->options.pred,
1023  "Main prediction unavailable in the \"mpeg2_aac_low\" profile\n");
1024  ERROR_IF(s->options.ltp,
1025  "LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n");
1026  WARN_IF(s->options.pns,
1027  "PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n");
1028  s->options.pns = 0;
1029  } else if (avctx->profile == FF_PROFILE_AAC_LTP) {
1030  s->options.ltp = 1;
1031  ERROR_IF(s->options.pred,
1032  "Main prediction unavailable in the \"aac_ltp\" profile\n");
1033  } else if (avctx->profile == FF_PROFILE_AAC_MAIN) {
1034  s->options.pred = 1;
1035  ERROR_IF(s->options.ltp,
1036  "LTP prediction unavailable in the \"aac_main\" profile\n");
1037  } else if (s->options.ltp) {
1038  avctx->profile = FF_PROFILE_AAC_LTP;
1039  WARN_IF(1,
1040  "Chainging profile to \"aac_ltp\"\n");
1041  ERROR_IF(s->options.pred,
1042  "Main prediction unavailable in the \"aac_ltp\" profile\n");
1043  } else if (s->options.pred) {
1044  avctx->profile = FF_PROFILE_AAC_MAIN;
1045  WARN_IF(1,
1046  "Chainging profile to \"aac_main\"\n");
1047  ERROR_IF(s->options.ltp,
1048  "LTP prediction unavailable in the \"aac_main\" profile\n");
1049  }
1050  s->profile = avctx->profile;
1051 
1052  /* Coder limitations */
1053  s->coder = &ff_aac_coders[s->options.coder];
1054  if (s->options.coder == AAC_CODER_ANMR) {
1056  "The ANMR coder is considered experimental, add -strict -2 to enable!\n");
1057  s->options.intensity_stereo = 0;
1058  s->options.pns = 0;
1059  }
1061  "The LPT profile requires experimental compliance, add -strict -2 to enable!\n");
1062 
1063  /* M/S introduces horrible artifacts with multichannel files, this is temporary */
1064  if (s->channels > 3)
1065  s->options.mid_side = 0;
1066 
1067  if ((ret = dsp_init(avctx, s)) < 0)
1068  return ret;
1069 
1070  if ((ret = alloc_buffers(avctx, s)) < 0)
1071  return ret;
1072 
1073  if ((ret = put_audio_specific_config(avctx)))
1074  return ret;
1075 
1076  sizes[0] = ff_aac_swb_size_1024[s->samplerate_index];
1077  sizes[1] = ff_aac_swb_size_128[s->samplerate_index];
1078  lengths[0] = ff_aac_num_swb_1024[s->samplerate_index];
1079  lengths[1] = ff_aac_num_swb_128[s->samplerate_index];
1080  for (i = 0; i < s->chan_map[0]; i++)
1081  grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
1082  if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
1083  s->chan_map[0], grouping)) < 0)
1084  return ret;
1085  s->psypp = ff_psy_preprocess_init(avctx);
1087  s->random_state = 0x1f2e3d4c;
1088 
1089  s->abs_pow34 = abs_pow34_v;
1090  s->quant_bands = quantize_bands;
1091 
1092  if (ARCH_X86)
1094 
1095  if (HAVE_MIPSDSP)
1097 
1098  ff_af_queue_init(avctx, &s->afq);
1099  ff_aac_tableinit();
1100 
1101  return 0;
1102 }
1103 
1104 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1105 static const AVOption aacenc_options[] = {
1106  {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_FAST}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
1107  {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1108  {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1109  {"fast", "Default fast search", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1110  {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
1111  {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1112  {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1113  {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1114  {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1115  {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1116  {"aac_pce", "Forces the use of PCEs", offsetof(AACEncContext, options.pce), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1118  {NULL}
1119 };
1120 
1121 static const AVClass aacenc_class = {
1122  .class_name = "AAC encoder",
1123  .item_name = av_default_item_name,
1124  .option = aacenc_options,
1125  .version = LIBAVUTIL_VERSION_INT,
1126 };
1127 
1129  { "b", "0" },
1130  { NULL }
1131 };
1132 
1134  .name = "aac",
1135  .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
1136  .type = AVMEDIA_TYPE_AUDIO,
1137  .id = AV_CODEC_ID_AAC,
1138  .priv_data_size = sizeof(AACEncContext),
1139  .init = aac_encode_init,
1140  .encode2 = aac_encode_frame,
1141  .close = aac_encode_end,
1143  .supported_samplerates = mpeg4audio_sample_rates,
1146  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
1148  .priv_class = &aacenc_class,
1149 };
@ EIGHT_SHORT_SEQUENCE
Definition: aac.h:79
@ LONG_STOP_SEQUENCE
Definition: aac.h:80
@ ONLY_LONG_SEQUENCE
Definition: aac.h:77
@ LONG_START_SEQUENCE
Definition: aac.h:78
#define NOISE_PRE
preamble for NOISE_BT, put in bitstream with the first noise band
Definition: aac.h:157
@ INTENSITY_BT
Scalefactor data are intensity stereo positions (in phase).
Definition: aac.h:90
@ INTENSITY_BT2
Scalefactor data are intensity stereo positions (out of phase).
Definition: aac.h:89
@ RESERVED_BT
Band types following are encoded differently from others.
Definition: aac.h:87
@ NOISE_BT
Spectral data are scaled white noise not coded in the bitstream.
Definition: aac.h:88
@ TYPE_CPE
Definition: aac.h:58
@ TYPE_FIL
Definition: aac.h:63
@ TYPE_LFE
Definition: aac.h:60
@ TYPE_END
Definition: aac.h:64
#define CLIP_AVOIDANCE_FACTOR
Definition: aac.h:54
#define TNS_MAX_ORDER
Definition: aac.h:51
#define NOISE_PRE_BITS
length of preamble
Definition: aac.h:158
#define SCALE_DIFF_ZERO
codebook index corresponding to zero scalefactor indices difference
Definition: aac.h:153
#define NOISE_OFFSET
subtracted from global gain, used as offset for the preamble
Definition: aac.h:159
const AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB]
Definition: aaccoder.c:897
static void put_bitstream_info(AACEncContext *s, const char *name)
Write some auxiliary information about the created AAC file.
Definition: aacenc.c:510
static const AVCodecDefault aac_encode_defaults[]
Definition: aacenc.c:1128
static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
Downscale spectral coefficients for near-clipping windows to avoid artifacts.
Definition: aacenc.c:464
static void apply_mid_side_stereo(ChannelElement *cpe)
Definition: aacenc.c:332
static void adjust_frame_information(ChannelElement *cpe, int chans)
Produce integer coefficients from scalefactors provided by the model.
Definition: aacenc.c:253
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, int common_window)
Encode one channel of audio data.
Definition: aacenc.c:484
static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
Definition: aacenc.c:915
static const AVOption aacenc_options[]
Definition: aacenc.c:1105
static int put_audio_specific_config(AVCodecContext *avctx)
Make AAC audio config object.
Definition: aacenc.c:91
static void(*const apply_window[4])(AVFloatDSPContext *fdsp, SingleChannelElement *sce, const float *audio)
Definition: aacenc.c:188
static void encode_pulses(AACEncContext *s, Pulse *pulse)
Encode pulse data.
Definition: aacenc.c:417
#define WINDOW_FUNC(type)
Definition: aacenc.c:132
static const AVClass aacenc_class
Definition: aacenc.c:1121
static av_cold int aac_encode_end(AVCodecContext *avctx)
Definition: aacenc.c:896
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
Encode scalefactor band coding type.
Definition: aacenc.c:367
static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, float *audio)
Definition: aacenc.c:197
static av_cold int aac_encode_init(AVCodecContext *avctx)
Definition: aacenc.c:947
static void apply_intensity_stereo(ChannelElement *cpe)
Definition: aacenc.c:303
static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
Definition: aacenc.c:50
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
Encode MS data.
Definition: aacenc.c:239
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
Encode ics_info element.
Definition: aacenc.c:218
#define AACENC_FLAGS
Definition: aacenc.c:1104
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
Encode spectral coefficients processed by psychoacoustic model.
Definition: aacenc.c:436
static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
Definition: aacenc.c:934
static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: aacenc.c:553
static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
Definition: aacenc.c:531
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce)
Encode scalefactors.
Definition: aacenc.c:381
void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
Definition: aacenc.c:123
AVCodec ff_aac_encoder
Definition: aacenc.c:1133
void ff_aac_coder_init_mips(AACEncContext *c)
static const AACPCEInfo aac_pce_configs[]
List of PCE (Program Configuration Element) for the channel layouts listed in channel_layout....
Definition: aacenc.h:139
@ AAC_CODER_ANMR
Definition: aacenc.h:38
@ AAC_CODER_FAST
Definition: aacenc.h:40
@ AAC_CODER_NB
Definition: aacenc.h:42
@ AAC_CODER_TWOLOOP
Definition: aacenc.h:39
void ff_aac_dsp_init_x86(AACEncContext *s)
AAC encoder utilities.
#define WARN_IF(cond,...)
Definition: aacenc_utils.h:274
static void quantize_bands(int *out, const float *in, const float *scaled, int size, int is_signed, int maxval, const float Q34, const float rounding)
Definition: aacenc_utils.h:65
#define ERROR_IF(cond,...)
Definition: aacenc_utils.h:268
static void abs_pow34_v(float *out, const float *in, const int size)
Definition: aacenc_utils.h:40
const uint8_t *const ff_aac_swb_size_1024[]
Definition: aacenctab.c:99
const int ff_aac_swb_size_128_len
Definition: aacenctab.c:107
const uint8_t *const ff_aac_swb_size_128[]
Definition: aacenctab.c:91
const int ff_aac_swb_size_1024_len
Definition: aacenctab.c:108
AAC encoder data.
static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS]
Table to remap channels from libavcodec's default order to AAC order.
Definition: aacenctab.h:72
static const int64_t aac_normal_chan_layouts[7]
Definition: aacenctab.h:47
#define AAC_MAX_CHANNELS
Definition: aacenctab.h:39
static const uint8_t aac_chan_configs[AAC_MAX_CHANNELS][6]
default channel configurations
Definition: aacenctab.h:58
static const int aacenc_profiles[]
Definition: aacenctab.h:132
static const int mpeg4audio_sample_rates[16]
Definition: aacenctab.h:85
const uint32_t ff_aac_scalefactor_code[121]
Definition: aactab.c:92
const uint8_t ff_tns_max_bands_1024[]
Definition: aactab.c:1413
const uint16_t *const ff_swb_offset_128[]
Definition: aactab.c:1387
const uint16_t *const ff_swb_offset_1024[]
Definition: aactab.c:1355
const uint8_t ff_aac_scalefactor_bits[121]
Definition: aactab.c:111
void ff_aac_tableinit(void)
Definition: aactab.c:3347
const uint8_t ff_aac_num_swb_1024[]
Definition: aactab.c:64
const uint8_t ff_aac_num_swb_128[]
Definition: aactab.c:80
const uint8_t ff_tns_max_bands_128[]
Definition: aactab.c:1425
AAC data declarations.
float ff_aac_kbd_long_1024[1024]
void ff_aac_float_common_init(void)
float ff_aac_kbd_short_128[128]
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
static const AVCodecDefault defaults[]
Definition: amfenc_h264.c:361
channels
Definition: aptx.h:33
#define L(x)
Definition: vp56_arith.h:36
#define av_cold
Definition: attributes.h:88
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
uint8_t
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
Libavcodec external API header.
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
Definition: avcodec.h:1606
#define FF_PROFILE_UNKNOWN
Definition: avcodec.h:1859
#define FF_PROFILE_MPEG2_AAC_LOW
Definition: avcodec.h:1870
#define FF_PROFILE_AAC_MAIN
Definition: avcodec.h:1862
#define FF_PROFILE_AAC_LOW
Definition: avcodec.h:1863
#define FF_PROFILE_AAC_LTP
Definition: avcodec.h:1865
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
void ff_put_string(PutBitContext *pb, const char *string, int terminate_string)
Put the string string in the bitstream.
Definition: bitstream.c:59
#define s(width, name)
Definition: cbs_vp9.c:257
#define FFMIN(a, b)
Definition: common.h:105
#define FFMAX(a, b)
Definition: common.h:103
#define av_clipf
Definition: common.h:170
#define HAVE_MIPSDSP
Definition: config.h:81
#define ARCH_X86
Definition: config.h:39
#define NULL
Definition: coverity.c:32
static __device__ float fabsf(float a)
Definition: cuda_runtime.h:181
static __device__ float fabs(float a)
Definition: cuda_runtime.h:182
#define max(a, b)
Definition: cuda_runtime.h:33
static AVFrame * frame
int ff_alloc_packet2(AVCodecContext *avctx, AVPacket *avpkt, int64_t size, int64_t min_size)
Check AVPacket size and/or allocate data.
Definition: encode.c:33
const OptionDef options[]
#define ff_mdct_init
Definition: fft.h:161
#define ff_mdct_end
Definition: fft.h:162
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
@ AV_OPT_TYPE_INT
Definition: opt.h:225
@ AV_OPT_TYPE_BOOL
Definition: opt.h:242
void av_get_channel_layout_string(char *buf, int buf_size, int nb_channels, uint64_t channel_layout)
Return a description of a channel layout.
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:333
#define AV_CODEC_CAP_DELAY
Encoder or decoder requires flushing with NULL input at the end in order to give the complete and cor...
Definition: codec.h:77
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:275
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: codec.h:82
@ AV_CODEC_ID_AAC
Definition: codec_id.h:426
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_INFO
Standard information.
Definition: log.h:205
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:237
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
for(j=16;j >0;--j)
#define R
Definition: huffyuvdsp.h:34
static const int sizes[][2]
Definition: img2dec.c:53
int i
Definition: input.c:407
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:218
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
Definition: internal.h:41
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
Definition: internal.h:49
#define LIBAVCODEC_IDENT
Definition: version.h:42
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
#define FF_ALLOCZ_TYPED_ARRAY(p, nelem)
Definition: internal.h:103
Replacements for frequently missing libm functions.
uint8_t w
Definition: llviddspenc.c:39
av_cold void ff_lpc_end(LPCContext *s)
Uninitialize LPCContext.
Definition: lpc.c:323
av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order, enum FFLPCType lpc_type)
Initialize LPCContext.
Definition: lpc.c:301
@ FF_LPC_TYPE_LEVINSON
Levinson-Durbin recursion.
Definition: lpc.h:47
uint32_t tag
Definition: movenc.c:1600
const int avpriv_mpeg4audio_sample_rates[16]
Definition: mpeg4audio.c:62
@ AOT_SBR
Y Spectral Band Replication.
Definition: mpeg4audio.h:94
AVOptions.
#define FF_AAC_PROFILE_OPTS
Definition: profiles.h:28
av_cold struct FFPsyPreprocessContext * ff_psy_preprocess_init(AVCodecContext *avctx)
psychoacoustic model audio preprocessing initialization
Definition: psymodel.c:103
av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
Cleanup audio preprocessing module.
Definition: psymodel.c:152
av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens, const uint8_t **bands, const int *num_bands, int num_groups, const uint8_t *group_map)
Initialize psychoacoustic model.
Definition: psymodel.c:31
av_cold void ff_psy_end(FFPsyContext *ctx)
Cleanup model context at the end.
Definition: psymodel.c:83
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)
Preprocess several channel in audio frame in order to compress it better.
Definition: psymodel.c:139
bitstream writer API
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:57
static int put_bits_count(PutBitContext *s)
Definition: put_bits.h:76
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:110
static void align_put_bits(PutBitContext *s)
Pad the bitstream with zeros up to the next byte boundary.
Definition: put_bits.h:386
const char * name
Definition: qsvenc.c:46
typedef void(RENAME(mix_any_func_type))
#define FF_ARRAY_ELEMS(a)
AAC encoder context.
Definition: aacenc.h:378
int num_ele[4]
front, side, back, lfe
Definition: aacenc.h:97
int index[4][8]
front, side, back, lfe
Definition: aacenc.h:99
int pairing[3][8]
front, side, back
Definition: aacenc.h:98
int64_t layout
Definition: aacenc.h:96
Describe the class of an AVClass context structure.
Definition: log.h:67
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
main external API structure.
Definition: avcodec.h:536
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:602
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:1601
int64_t bit_rate
the average bitrate
Definition: avcodec.h:586
int profile
profile
Definition: avcodec.h:1858
int initial_padding
Audio only.
Definition: avcodec.h:2062
int sample_rate
samples per second
Definition: avcodec.h:1196
int frame_number
Frame counter, set by libavcodec.
Definition: avcodec.h:1227
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:616
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:637
int channels
number of audio channels
Definition: avcodec.h:1197
int extradata_size
Definition: avcodec.h:638
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1247
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1216
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
AVOption.
Definition: opt.h:248
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:387
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:362
uint8_t * data
Definition: packet.h:369
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:276
uint8_t is_mask[128]
Set if intensity stereo is used (used by encoder)
Definition: aac.h:283
int ms_mode
Signals mid/side stereo flags coding mode (used by encoder)
Definition: aac.h:280
int common_window
Set if channels share a common 'IndividualChannelStream' in bitstream.
Definition: aac.h:279
uint8_t is_mode
Set if any bands have been encoded using intensity stereo (used by encoder)
Definition: aac.h:281
uint8_t ms_mask[128]
Set if mid/side stereo is used for each scalefactor window band.
Definition: aac.h:282
SingleChannelElement ch[2]
Definition: aac.h:285
windowing related information
Definition: psymodel.h:77
int num_windows
number of windows in a frame
Definition: psymodel.h:80
int grouping[8]
window grouping (for e.g. AAC)
Definition: psymodel.h:81
float clipping[8]
maximum absolute normalized intensity in the given window for clip avoidance
Definition: psymodel.h:82
int window_shape
window shape (sine/KBD/whatever)
Definition: psymodel.h:79
int window_type[3]
window type (short/long/transitional, etc.) - current, previous and next
Definition: psymodel.h:78
Individual Channel Stream.
Definition: aac.h:175
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:176
uint8_t group_len[8]
Definition: aac.h:180
int num_swb
number of scalefactor window bands
Definition: aac.h:184
LongTermPrediction ltp
Definition: aac.h:181
uint8_t use_kb_window[2]
If set, use Kaiser-Bessel window, otherwise use a sine window.
Definition: aac.h:178
uint8_t prediction_used[41]
Definition: aac.h:191
float clip_avoidance_factor
set if any window is near clipping to the necessary atennuation factor to avoid it
Definition: aac.h:193
const uint8_t * swb_sizes
table of scalefactor band sizes for a particular window
Definition: aac.h:183
enum WindowSequence window_sequence[2]
Definition: aac.h:177
uint8_t window_clipping[8]
set if a certain window is near clipping
Definition: aac.h:192
const uint16_t * swb_offset
table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular wind...
Definition: aac.h:182
int8_t present
Definition: aac.h:165
int8_t used[MAX_LTP_LONG_SFB]
Definition: aac.h:169
Definition: aac.h:225
int pos[4]
Definition: aac.h:228
int start
Definition: aac.h:227
int amp[4]
Definition: aac.h:229
int num_pulse
Definition: aac.h:226
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:249
uint8_t zeroes[128]
band is not coded (used by encoder)
Definition: aac.h:258
enum BandType band_type[128]
band types
Definition: aac.h:253
float is_ener[128]
Intensity stereo pos (used by encoder)
Definition: aac.h:260
INTFLOAT pcoeffs[1024]
coefficients for IMDCT, pristine
Definition: aac.h:262
INTFLOAT ret_buf[2048]
PCM output buffer.
Definition: aac.h:265
TemporalNoiseShaping tns
Definition: aac.h:251
INTFLOAT coeffs[1024]
coefficients for IMDCT, maybe processed
Definition: aac.h:263
INTFLOAT ltp_state[3072]
time signal for LTP
Definition: aac.h:266
AAC_FLOAT lcoeffs[1024]
MDCT of LTP coefficients (used by encoder)
Definition: aac.h:267
IndividualChannelStream ics
Definition: aac.h:250
int sf_idx[128]
scalefactor indices (used by encoder)
Definition: aac.h:257
Temporal Noise Shaping.
Definition: aac.h:199
#define av_freep(p)
#define av_log(a,...)
FILE * out
Definition: movenc.c:54
const char * g
Definition: vf_curves.c:117
if(ret< 0)
Definition: vf_mcdeint.c:282
static av_always_inline int diff(const uint32_t a, const uint32_t b)