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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/thread.h"
34 #include "libavutil/float_dsp.h"
35 #include "libavutil/opt.h"
36 #include "avcodec.h"
37 #include "put_bits.h"
38 #include "internal.h"
39 #include "mpeg4audio.h"
40 #include "kbdwin.h"
41 #include "sinewin.h"
42 
43 #include "aac.h"
44 #include "aactab.h"
45 #include "aacenc.h"
46 #include "aacenctab.h"
47 #include "aacenc_utils.h"
48 
49 #include "psymodel.h"
50 
52 
53 static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
54 {
55  int i, j;
56  AACEncContext *s = avctx->priv_data;
57  AACPCEInfo *pce = &s->pce;
58  const int bitexact = avctx->flags & AV_CODEC_FLAG_BITEXACT;
59  const char *aux_data = bitexact ? "Lavc" : LIBAVCODEC_IDENT;
60 
61  put_bits(pb, 4, 0);
62 
63  put_bits(pb, 2, avctx->profile);
64  put_bits(pb, 4, s->samplerate_index);
65 
66  put_bits(pb, 4, pce->num_ele[0]); /* Front */
67  put_bits(pb, 4, pce->num_ele[1]); /* Side */
68  put_bits(pb, 4, pce->num_ele[2]); /* Back */
69  put_bits(pb, 2, pce->num_ele[3]); /* LFE */
70  put_bits(pb, 3, 0); /* Assoc data */
71  put_bits(pb, 4, 0); /* CCs */
72 
73  put_bits(pb, 1, 0); /* Stereo mixdown */
74  put_bits(pb, 1, 0); /* Mono mixdown */
75  put_bits(pb, 1, 0); /* Something else */
76 
77  for (i = 0; i < 4; i++) {
78  for (j = 0; j < pce->num_ele[i]; j++) {
79  if (i < 3)
80  put_bits(pb, 1, pce->pairing[i][j]);
81  put_bits(pb, 4, pce->index[i][j]);
82  }
83  }
84 
86  put_bits(pb, 8, strlen(aux_data));
87  avpriv_put_string(pb, aux_data, 0);
88 }
89 
90 /**
91  * Make AAC audio config object.
92  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
93  */
95 {
96  PutBitContext pb;
97  AACEncContext *s = avctx->priv_data;
98  int channels = (!s->needs_pce)*(s->channels - (s->channels == 8 ? 1 : 0));
99  const int max_size = 32;
100 
101  avctx->extradata = av_mallocz(max_size);
102  if (!avctx->extradata)
103  return AVERROR(ENOMEM);
104 
105  init_put_bits(&pb, avctx->extradata, max_size);
106  put_bits(&pb, 5, s->profile+1); //profile
107  put_bits(&pb, 4, s->samplerate_index); //sample rate index
108  put_bits(&pb, 4, channels);
109  //GASpecificConfig
110  put_bits(&pb, 1, 0); //frame length - 1024 samples
111  put_bits(&pb, 1, 0); //does not depend on core coder
112  put_bits(&pb, 1, 0); //is not extension
113  if (s->needs_pce)
114  put_pce(&pb, avctx);
115 
116  //Explicitly Mark SBR absent
117  put_bits(&pb, 11, 0x2b7); //sync extension
118  put_bits(&pb, 5, AOT_SBR);
119  put_bits(&pb, 1, 0);
120  flush_put_bits(&pb);
121  avctx->extradata_size = put_bits_count(&pb) >> 3;
122 
123  return 0;
124 }
125 
127 {
130  memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache));
132  }
133 }
134 
135 #define WINDOW_FUNC(type) \
136 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
137  SingleChannelElement *sce, \
138  const float *audio)
139 
140 WINDOW_FUNC(only_long)
141 {
142  const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
143  const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
144  float *out = sce->ret_buf;
145 
146  fdsp->vector_fmul (out, audio, lwindow, 1024);
147  fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
148 }
149 
150 WINDOW_FUNC(long_start)
151 {
152  const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
153  const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
154  float *out = sce->ret_buf;
155 
156  fdsp->vector_fmul(out, audio, lwindow, 1024);
157  memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
158  fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
159  memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
160 }
161 
162 WINDOW_FUNC(long_stop)
163 {
164  const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
165  const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
166  float *out = sce->ret_buf;
167 
168  memset(out, 0, sizeof(out[0]) * 448);
169  fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
170  memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
171  fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
172 }
173 
174 WINDOW_FUNC(eight_short)
175 {
176  const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
177  const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
178  const float *in = audio + 448;
179  float *out = sce->ret_buf;
180  int w;
181 
182  for (w = 0; w < 8; w++) {
183  fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
184  out += 128;
185  in += 128;
186  fdsp->vector_fmul_reverse(out, in, swindow, 128);
187  out += 128;
188  }
189 }
190 
191 static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
193  const float *audio) = {
194  [ONLY_LONG_SEQUENCE] = apply_only_long_window,
195  [LONG_START_SEQUENCE] = apply_long_start_window,
196  [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
197  [LONG_STOP_SEQUENCE] = apply_long_stop_window
198 };
199 
201  float *audio)
202 {
203  int i;
204  const float *output = sce->ret_buf;
205 
206  apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
207 
209  s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
210  else
211  for (i = 0; i < 1024; i += 128)
212  s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
213  memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
214  memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
215 }
216 
217 /**
218  * Encode ics_info element.
219  * @see Table 4.6 (syntax of ics_info)
220  */
222 {
223  int w;
224 
225  put_bits(&s->pb, 1, 0); // ics_reserved bit
226  put_bits(&s->pb, 2, info->window_sequence[0]);
227  put_bits(&s->pb, 1, info->use_kb_window[0]);
228  if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
229  put_bits(&s->pb, 6, info->max_sfb);
230  put_bits(&s->pb, 1, !!info->predictor_present);
231  } else {
232  put_bits(&s->pb, 4, info->max_sfb);
233  for (w = 1; w < 8; w++)
234  put_bits(&s->pb, 1, !info->group_len[w]);
235  }
236 }
237 
238 /**
239  * Encode MS data.
240  * @see 4.6.8.1 "Joint Coding - M/S Stereo"
241  */
243 {
244  int i, w;
245 
246  put_bits(pb, 2, cpe->ms_mode);
247  if (cpe->ms_mode == 1)
248  for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
249  for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
250  put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
251 }
252 
253 /**
254  * Produce integer coefficients from scalefactors provided by the model.
255  */
256 static void adjust_frame_information(ChannelElement *cpe, int chans)
257 {
258  int i, w, w2, g, ch;
259  int maxsfb, cmaxsfb;
260 
261  for (ch = 0; ch < chans; ch++) {
262  IndividualChannelStream *ics = &cpe->ch[ch].ics;
263  maxsfb = 0;
264  cpe->ch[ch].pulse.num_pulse = 0;
265  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
266  for (w2 = 0; w2 < ics->group_len[w]; w2++) {
267  for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
268  ;
269  maxsfb = FFMAX(maxsfb, cmaxsfb);
270  }
271  }
272  ics->max_sfb = maxsfb;
273 
274  //adjust zero bands for window groups
275  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
276  for (g = 0; g < ics->max_sfb; g++) {
277  i = 1;
278  for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
279  if (!cpe->ch[ch].zeroes[w2*16 + g]) {
280  i = 0;
281  break;
282  }
283  }
284  cpe->ch[ch].zeroes[w*16 + g] = i;
285  }
286  }
287  }
288 
289  if (chans > 1 && cpe->common_window) {
290  IndividualChannelStream *ics0 = &cpe->ch[0].ics;
291  IndividualChannelStream *ics1 = &cpe->ch[1].ics;
292  int msc = 0;
293  ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
294  ics1->max_sfb = ics0->max_sfb;
295  for (w = 0; w < ics0->num_windows*16; w += 16)
296  for (i = 0; i < ics0->max_sfb; i++)
297  if (cpe->ms_mask[w+i])
298  msc++;
299  if (msc == 0 || ics0->max_sfb == 0)
300  cpe->ms_mode = 0;
301  else
302  cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
303  }
304 }
305 
307 {
308  int w, w2, g, i;
309  IndividualChannelStream *ics = &cpe->ch[0].ics;
310  if (!cpe->common_window)
311  return;
312  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
313  for (w2 = 0; w2 < ics->group_len[w]; w2++) {
314  int start = (w+w2) * 128;
315  for (g = 0; g < ics->num_swb; g++) {
316  int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
317  float scale = cpe->ch[0].is_ener[w*16+g];
318  if (!cpe->is_mask[w*16 + g]) {
319  start += ics->swb_sizes[g];
320  continue;
321  }
322  if (cpe->ms_mask[w*16 + g])
323  p *= -1;
324  for (i = 0; i < ics->swb_sizes[g]; i++) {
325  float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
326  cpe->ch[0].coeffs[start+i] = sum;
327  cpe->ch[1].coeffs[start+i] = 0.0f;
328  }
329  start += ics->swb_sizes[g];
330  }
331  }
332  }
333 }
334 
336 {
337  int w, w2, g, i;
338  IndividualChannelStream *ics = &cpe->ch[0].ics;
339  if (!cpe->common_window)
340  return;
341  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
342  for (w2 = 0; w2 < ics->group_len[w]; w2++) {
343  int start = (w+w2) * 128;
344  for (g = 0; g < ics->num_swb; g++) {
345  /* ms_mask can be used for other purposes in PNS and I/S,
346  * so must not apply M/S if any band uses either, even if
347  * ms_mask is set.
348  */
349  if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]
350  || cpe->ch[0].band_type[w*16 + g] >= NOISE_BT
351  || cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) {
352  start += ics->swb_sizes[g];
353  continue;
354  }
355  for (i = 0; i < ics->swb_sizes[g]; i++) {
356  float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
357  float R = L - cpe->ch[1].coeffs[start+i];
358  cpe->ch[0].coeffs[start+i] = L;
359  cpe->ch[1].coeffs[start+i] = R;
360  }
361  start += ics->swb_sizes[g];
362  }
363  }
364  }
365 }
366 
367 /**
368  * Encode scalefactor band coding type.
369  */
371 {
372  int w;
373 
376 
377  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
378  s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
379 }
380 
381 /**
382  * Encode scalefactors.
383  */
386 {
387  int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
388  int off_is = 0, noise_flag = 1;
389  int i, w;
390 
391  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
392  for (i = 0; i < sce->ics.max_sfb; i++) {
393  if (!sce->zeroes[w*16 + i]) {
394  if (sce->band_type[w*16 + i] == NOISE_BT) {
395  diff = sce->sf_idx[w*16 + i] - off_pns;
396  off_pns = sce->sf_idx[w*16 + i];
397  if (noise_flag-- > 0) {
398  put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
399  continue;
400  }
401  } else if (sce->band_type[w*16 + i] == INTENSITY_BT ||
402  sce->band_type[w*16 + i] == INTENSITY_BT2) {
403  diff = sce->sf_idx[w*16 + i] - off_is;
404  off_is = sce->sf_idx[w*16 + i];
405  } else {
406  diff = sce->sf_idx[w*16 + i] - off_sf;
407  off_sf = sce->sf_idx[w*16 + i];
408  }
409  diff += SCALE_DIFF_ZERO;
410  av_assert0(diff >= 0 && diff <= 120);
412  }
413  }
414  }
415 }
416 
417 /**
418  * Encode pulse data.
419  */
420 static void encode_pulses(AACEncContext *s, Pulse *pulse)
421 {
422  int i;
423 
424  put_bits(&s->pb, 1, !!pulse->num_pulse);
425  if (!pulse->num_pulse)
426  return;
427 
428  put_bits(&s->pb, 2, pulse->num_pulse - 1);
429  put_bits(&s->pb, 6, pulse->start);
430  for (i = 0; i < pulse->num_pulse; i++) {
431  put_bits(&s->pb, 5, pulse->pos[i]);
432  put_bits(&s->pb, 4, pulse->amp[i]);
433  }
434 }
435 
436 /**
437  * Encode spectral coefficients processed by psychoacoustic model.
438  */
440 {
441  int start, i, w, w2;
442 
443  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
444  start = 0;
445  for (i = 0; i < sce->ics.max_sfb; i++) {
446  if (sce->zeroes[w*16 + i]) {
447  start += sce->ics.swb_sizes[i];
448  continue;
449  }
450  for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
451  s->coder->quantize_and_encode_band(s, &s->pb,
452  &sce->coeffs[start + w2*128],
453  NULL, sce->ics.swb_sizes[i],
454  sce->sf_idx[w*16 + i],
455  sce->band_type[w*16 + i],
456  s->lambda,
457  sce->ics.window_clipping[w]);
458  }
459  start += sce->ics.swb_sizes[i];
460  }
461  }
462 }
463 
464 /**
465  * Downscale spectral coefficients for near-clipping windows to avoid artifacts
466  */
468 {
469  int start, i, j, w;
470 
471  if (sce->ics.clip_avoidance_factor < 1.0f) {
472  for (w = 0; w < sce->ics.num_windows; w++) {
473  start = 0;
474  for (i = 0; i < sce->ics.max_sfb; i++) {
475  float *swb_coeffs = &sce->coeffs[start + w*128];
476  for (j = 0; j < sce->ics.swb_sizes[i]; j++)
477  swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
478  start += sce->ics.swb_sizes[i];
479  }
480  }
481  }
482 }
483 
484 /**
485  * Encode one channel of audio data.
486  */
489  int common_window)
490 {
491  put_bits(&s->pb, 8, sce->sf_idx[0]);
492  if (!common_window) {
493  put_ics_info(s, &sce->ics);
494  if (s->coder->encode_main_pred)
495  s->coder->encode_main_pred(s, sce);
496  if (s->coder->encode_ltp_info)
497  s->coder->encode_ltp_info(s, sce, 0);
498  }
499  encode_band_info(s, sce);
500  encode_scale_factors(avctx, s, sce);
501  encode_pulses(s, &sce->pulse);
502  put_bits(&s->pb, 1, !!sce->tns.present);
503  if (s->coder->encode_tns_info)
504  s->coder->encode_tns_info(s, sce);
505  put_bits(&s->pb, 1, 0); //ssr
506  encode_spectral_coeffs(s, sce);
507  return 0;
508 }
509 
510 /**
511  * Write some auxiliary information about the created AAC file.
512  */
513 static void put_bitstream_info(AACEncContext *s, const char *name)
514 {
515  int i, namelen, padbits;
516 
517  namelen = strlen(name) + 2;
518  put_bits(&s->pb, 3, TYPE_FIL);
519  put_bits(&s->pb, 4, FFMIN(namelen, 15));
520  if (namelen >= 15)
521  put_bits(&s->pb, 8, namelen - 14);
522  put_bits(&s->pb, 4, 0); //extension type - filler
523  padbits = -put_bits_count(&s->pb) & 7;
525  for (i = 0; i < namelen - 2; i++)
526  put_bits(&s->pb, 8, name[i]);
527  put_bits(&s->pb, 12 - padbits, 0);
528 }
529 
530 /*
531  * Copy input samples.
532  * Channels are reordered from libavcodec's default order to AAC order.
533  */
535 {
536  int ch;
537  int end = 2048 + (frame ? frame->nb_samples : 0);
538  const uint8_t *channel_map = s->reorder_map;
539 
540  /* copy and remap input samples */
541  for (ch = 0; ch < s->channels; ch++) {
542  /* copy last 1024 samples of previous frame to the start of the current frame */
543  memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
544 
545  /* copy new samples and zero any remaining samples */
546  if (frame) {
547  memcpy(&s->planar_samples[ch][2048],
548  frame->extended_data[channel_map[ch]],
549  frame->nb_samples * sizeof(s->planar_samples[0][0]));
550  }
551  memset(&s->planar_samples[ch][end], 0,
552  (3072 - end) * sizeof(s->planar_samples[0][0]));
553  }
554 }
555 
556 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
557  const AVFrame *frame, int *got_packet_ptr)
558 {
559  AACEncContext *s = avctx->priv_data;
560  float **samples = s->planar_samples, *samples2, *la, *overlap;
561  ChannelElement *cpe;
564  int i, its, ch, w, chans, tag, start_ch, ret, frame_bits;
565  int target_bits, rate_bits, too_many_bits, too_few_bits;
566  int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
567  int chan_el_counter[4];
569 
570  /* add current frame to queue */
571  if (frame) {
572  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
573  return ret;
574  } else {
575  if (!s->afq.remaining_samples || (!s->afq.frame_alloc && !s->afq.frame_count))
576  return 0;
577  }
578 
579  copy_input_samples(s, frame);
580  if (s->psypp)
582 
583  if (!avctx->frame_number)
584  return 0;
585 
586  start_ch = 0;
587  for (i = 0; i < s->chan_map[0]; i++) {
588  FFPsyWindowInfo* wi = windows + start_ch;
589  tag = s->chan_map[i+1];
590  chans = tag == TYPE_CPE ? 2 : 1;
591  cpe = &s->cpe[i];
592  for (ch = 0; ch < chans; ch++) {
593  int k;
594  float clip_avoidance_factor;
595  sce = &cpe->ch[ch];
596  ics = &sce->ics;
597  s->cur_channel = start_ch + ch;
598  overlap = &samples[s->cur_channel][0];
599  samples2 = overlap + 1024;
600  la = samples2 + (448+64);
601  if (!frame)
602  la = NULL;
603  if (tag == TYPE_LFE) {
604  wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE;
605  wi[ch].window_shape = 0;
606  wi[ch].num_windows = 1;
607  wi[ch].grouping[0] = 1;
608  wi[ch].clipping[0] = 0;
609 
610  /* Only the lowest 12 coefficients are used in a LFE channel.
611  * The expression below results in only the bottom 8 coefficients
612  * being used for 11.025kHz to 16kHz sample rates.
613  */
614  ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
615  } else {
616  wi[ch] = s->psy.model->window(&s->psy, samples2, la, s->cur_channel,
617  ics->window_sequence[0]);
618  }
619  ics->window_sequence[1] = ics->window_sequence[0];
620  ics->window_sequence[0] = wi[ch].window_type[0];
621  ics->use_kb_window[1] = ics->use_kb_window[0];
622  ics->use_kb_window[0] = wi[ch].window_shape;
623  ics->num_windows = wi[ch].num_windows;
624  ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
625  ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
626  ics->max_sfb = FFMIN(ics->max_sfb, ics->num_swb);
627  ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
633 
634  for (w = 0; w < ics->num_windows; w++)
635  ics->group_len[w] = wi[ch].grouping[w];
636 
637  /* Calculate input sample maximums and evaluate clipping risk */
638  clip_avoidance_factor = 0.0f;
639  for (w = 0; w < ics->num_windows; w++) {
640  const float *wbuf = overlap + w * 128;
641  const int wlen = 2048 / ics->num_windows;
642  float max = 0;
643  int j;
644  /* mdct input is 2 * output */
645  for (j = 0; j < wlen; j++)
646  max = FFMAX(max, fabsf(wbuf[j]));
647  wi[ch].clipping[w] = max;
648  }
649  for (w = 0; w < ics->num_windows; w++) {
650  if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
651  ics->window_clipping[w] = 1;
652  clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
653  } else {
654  ics->window_clipping[w] = 0;
655  }
656  }
657  if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
658  ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
659  } else {
660  ics->clip_avoidance_factor = 1.0f;
661  }
662 
663  apply_window_and_mdct(s, sce, overlap);
664 
665  if (s->options.ltp && s->coder->update_ltp) {
666  s->coder->update_ltp(s, sce);
667  apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]);
668  s->mdct1024.mdct_calc(&s->mdct1024, sce->lcoeffs, sce->ret_buf);
669  }
670 
671  for (k = 0; k < 1024; k++) {
672  if (!(fabs(cpe->ch[ch].coeffs[k]) < 1E16)) { // Ensure headroom for energy calculation
673  av_log(avctx, AV_LOG_ERROR, "Input contains (near) NaN/+-Inf\n");
674  return AVERROR(EINVAL);
675  }
676  }
677  avoid_clipping(s, sce);
678  }
679  start_ch += chans;
680  }
681  if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0)
682  return ret;
683  frame_bits = its = 0;
684  do {
685  init_put_bits(&s->pb, avpkt->data, avpkt->size);
686 
687  if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
689  start_ch = 0;
690  target_bits = 0;
691  memset(chan_el_counter, 0, sizeof(chan_el_counter));
692  for (i = 0; i < s->chan_map[0]; i++) {
693  FFPsyWindowInfo* wi = windows + start_ch;
694  const float *coeffs[2];
695  tag = s->chan_map[i+1];
696  chans = tag == TYPE_CPE ? 2 : 1;
697  cpe = &s->cpe[i];
698  cpe->common_window = 0;
699  memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
700  memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
701  put_bits(&s->pb, 3, tag);
702  put_bits(&s->pb, 4, chan_el_counter[tag]++);
703  for (ch = 0; ch < chans; ch++) {
704  sce = &cpe->ch[ch];
705  coeffs[ch] = sce->coeffs;
706  sce->ics.predictor_present = 0;
707  sce->ics.ltp.present = 0;
708  memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used));
709  memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
710  memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
711  for (w = 0; w < 128; w++)
712  if (sce->band_type[w] > RESERVED_BT)
713  sce->band_type[w] = 0;
714  }
715  s->psy.bitres.alloc = -1;
717  s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
718  if (s->psy.bitres.alloc > 0) {
719  /* Lambda unused here on purpose, we need to take psy's unscaled allocation */
720  target_bits += s->psy.bitres.alloc
721  * (s->lambda / (avctx->global_quality ? avctx->global_quality : 120));
722  s->psy.bitres.alloc /= chans;
723  }
724  s->cur_type = tag;
725  for (ch = 0; ch < chans; ch++) {
726  s->cur_channel = start_ch + ch;
727  if (s->options.pns && s->coder->mark_pns)
728  s->coder->mark_pns(s, avctx, &cpe->ch[ch]);
729  s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
730  }
731  if (chans > 1
732  && wi[0].window_type[0] == wi[1].window_type[0]
733  && wi[0].window_shape == wi[1].window_shape) {
734 
735  cpe->common_window = 1;
736  for (w = 0; w < wi[0].num_windows; w++) {
737  if (wi[0].grouping[w] != wi[1].grouping[w]) {
738  cpe->common_window = 0;
739  break;
740  }
741  }
742  }
743  for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
744  sce = &cpe->ch[ch];
745  s->cur_channel = start_ch + ch;
746  if (s->options.tns && s->coder->search_for_tns)
747  s->coder->search_for_tns(s, sce);
748  if (s->options.tns && s->coder->apply_tns_filt)
749  s->coder->apply_tns_filt(s, sce);
750  if (sce->tns.present)
751  tns_mode = 1;
752  if (s->options.pns && s->coder->search_for_pns)
753  s->coder->search_for_pns(s, avctx, sce);
754  }
755  s->cur_channel = start_ch;
756  if (s->options.intensity_stereo) { /* Intensity Stereo */
757  if (s->coder->search_for_is)
758  s->coder->search_for_is(s, avctx, cpe);
759  if (cpe->is_mode) is_mode = 1;
761  }
762  if (s->options.pred) { /* Prediction */
763  for (ch = 0; ch < chans; ch++) {
764  sce = &cpe->ch[ch];
765  s->cur_channel = start_ch + ch;
766  if (s->options.pred && s->coder->search_for_pred)
767  s->coder->search_for_pred(s, sce);
768  if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
769  }
770  if (s->coder->adjust_common_pred)
771  s->coder->adjust_common_pred(s, cpe);
772  for (ch = 0; ch < chans; ch++) {
773  sce = &cpe->ch[ch];
774  s->cur_channel = start_ch + ch;
775  if (s->options.pred && s->coder->apply_main_pred)
776  s->coder->apply_main_pred(s, sce);
777  }
778  s->cur_channel = start_ch;
779  }
780  if (s->options.mid_side) { /* Mid/Side stereo */
781  if (s->options.mid_side == -1 && s->coder->search_for_ms)
782  s->coder->search_for_ms(s, cpe);
783  else if (cpe->common_window)
784  memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
786  }
787  adjust_frame_information(cpe, chans);
788  if (s->options.ltp) { /* LTP */
789  for (ch = 0; ch < chans; ch++) {
790  sce = &cpe->ch[ch];
791  s->cur_channel = start_ch + ch;
792  if (s->coder->search_for_ltp)
793  s->coder->search_for_ltp(s, sce, cpe->common_window);
794  if (sce->ics.ltp.present) pred_mode = 1;
795  }
796  s->cur_channel = start_ch;
797  if (s->coder->adjust_common_ltp)
798  s->coder->adjust_common_ltp(s, cpe);
799  }
800  if (chans == 2) {
801  put_bits(&s->pb, 1, cpe->common_window);
802  if (cpe->common_window) {
803  put_ics_info(s, &cpe->ch[0].ics);
804  if (s->coder->encode_main_pred)
805  s->coder->encode_main_pred(s, &cpe->ch[0]);
806  if (s->coder->encode_ltp_info)
807  s->coder->encode_ltp_info(s, &cpe->ch[0], 1);
808  encode_ms_info(&s->pb, cpe);
809  if (cpe->ms_mode) ms_mode = 1;
810  }
811  }
812  for (ch = 0; ch < chans; ch++) {
813  s->cur_channel = start_ch + ch;
814  encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
815  }
816  start_ch += chans;
817  }
818 
819  if (avctx->flags & AV_CODEC_FLAG_QSCALE) {
820  /* When using a constant Q-scale, don't mess with lambda */
821  break;
822  }
823 
824  /* rate control stuff
825  * allow between the nominal bitrate, and what psy's bit reservoir says to target
826  * but drift towards the nominal bitrate always
827  */
828  frame_bits = put_bits_count(&s->pb);
829  rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate;
830  rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3);
831  too_many_bits = FFMAX(target_bits, rate_bits);
832  too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3);
833  too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits);
834 
835  /* When using ABR, be strict (but only for increasing) */
836  too_few_bits = too_few_bits - too_few_bits/8;
837  too_many_bits = too_many_bits + too_many_bits/2;
838 
839  if ( its == 0 /* for steady-state Q-scale tracking */
840  || (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits))
841  || frame_bits >= 6144 * s->channels - 3 )
842  {
843  float ratio = ((float)rate_bits) / frame_bits;
844 
845  if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) {
846  /*
847  * This path is for steady-state Q-scale tracking
848  * When frame bits fall within the stable range, we still need to adjust
849  * lambda to maintain it like so in a stable fashion (large jumps in lambda
850  * create artifacts and should be avoided), but slowly
851  */
852  ratio = sqrtf(sqrtf(ratio));
853  ratio = av_clipf(ratio, 0.9f, 1.1f);
854  } else {
855  /* Not so fast though */
856  ratio = sqrtf(ratio);
857  }
858  s->lambda = FFMIN(s->lambda * ratio, 65536.f);
859 
860  /* Keep iterating if we must reduce and lambda is in the sky */
861  if (ratio > 0.9f && ratio < 1.1f) {
862  break;
863  } else {
864  if (is_mode || ms_mode || tns_mode || pred_mode) {
865  for (i = 0; i < s->chan_map[0]; i++) {
866  // Must restore coeffs
867  chans = tag == TYPE_CPE ? 2 : 1;
868  cpe = &s->cpe[i];
869  for (ch = 0; ch < chans; ch++)
870  memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
871  }
872  }
873  its++;
874  }
875  } else {
876  break;
877  }
878  } while (1);
879 
880  if (s->options.ltp && s->coder->ltp_insert_new_frame)
882 
883  put_bits(&s->pb, 3, TYPE_END);
884  flush_put_bits(&s->pb);
885 
887 
888  s->lambda_sum += s->lambda;
889  s->lambda_count++;
890 
891  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
892  &avpkt->duration);
893 
894  avpkt->size = put_bits_count(&s->pb) >> 3;
895  *got_packet_ptr = 1;
896  return 0;
897 }
898 
900 {
901  AACEncContext *s = avctx->priv_data;
902 
903  av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_sum / s->lambda_count);
904 
905  ff_mdct_end(&s->mdct1024);
906  ff_mdct_end(&s->mdct128);
907  ff_psy_end(&s->psy);
908  ff_lpc_end(&s->lpc);
909  if (s->psypp)
911  av_freep(&s->buffer.samples);
912  av_freep(&s->cpe);
913  av_freep(&s->fdsp);
914  ff_af_queue_close(&s->afq);
915  return 0;
916 }
917 
919 {
920  int ret = 0;
921 
923  if (!s->fdsp)
924  return AVERROR(ENOMEM);
925 
926  // window init
931 
932  if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
933  return ret;
934  if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0)
935  return ret;
936 
937  return 0;
938 }
939 
941 {
942  int ch;
943  FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);
944  FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);
945 
946  for(ch = 0; ch < s->channels; ch++)
947  s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
948 
949  return 0;
950 alloc_fail:
951  return AVERROR(ENOMEM);
952 }
953 
955 {
957 }
958 
960 {
961  AACEncContext *s = avctx->priv_data;
962  int i, ret = 0;
963  const uint8_t *sizes[2];
964  uint8_t grouping[AAC_MAX_CHANNELS];
965  int lengths[2];
966 
967  /* Constants */
968  s->last_frame_pb_count = 0;
969  avctx->frame_size = 1024;
970  avctx->initial_padding = 1024;
971  s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
972 
973  /* Channel map and unspecified bitrate guessing */
974  s->channels = avctx->channels;
975 
976  s->needs_pce = 1;
977  for (i = 0; i < FF_ARRAY_ELEMS(aac_normal_chan_layouts); i++) {
978  if (avctx->channel_layout == aac_normal_chan_layouts[i]) {
979  s->needs_pce = s->options.pce;
980  break;
981  }
982  }
983 
984  if (s->needs_pce) {
985  for (i = 0; i < FF_ARRAY_ELEMS(aac_pce_configs); i++)
986  if (avctx->channel_layout == aac_pce_configs[i].layout)
987  break;
988  ERROR_IF(i == FF_ARRAY_ELEMS(aac_pce_configs), "Unsupported channel layout\n");
989  av_log(avctx, AV_LOG_INFO, "Using a PCE to encode channel layout\n");
990  s->pce = aac_pce_configs[i];
991  s->reorder_map = s->pce.reorder_map;
992  s->chan_map = s->pce.config_map;
993  } else {
994  s->reorder_map = aac_chan_maps[s->channels - 1];
995  s->chan_map = aac_chan_configs[s->channels - 1];
996  }
997 
998  if (!avctx->bit_rate) {
999  for (i = 1; i <= s->chan_map[0]; i++) {
1000  avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */
1001  s->chan_map[i] == TYPE_LFE ? 16000 : /* LFE */
1002  69000 ; /* SCE */
1003  }
1004  }
1005 
1006  /* Samplerate */
1007  for (i = 0; i < 16; i++)
1009  break;
1010  s->samplerate_index = i;
1011  ERROR_IF(s->samplerate_index == 16 ||
1014  "Unsupported sample rate %d\n", avctx->sample_rate);
1015 
1016  /* Bitrate limiting */
1017  WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
1018  "Too many bits %f > %d per frame requested, clamping to max\n",
1019  1024.0 * avctx->bit_rate / avctx->sample_rate,
1020  6144 * s->channels);
1021  avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate,
1022  avctx->bit_rate);
1023 
1024  /* Profile and option setting */
1025  avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
1026  avctx->profile;
1027  for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++)
1028  if (avctx->profile == aacenc_profiles[i])
1029  break;
1030  if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) {
1031  avctx->profile = FF_PROFILE_AAC_LOW;
1032  ERROR_IF(s->options.pred,
1033  "Main prediction unavailable in the \"mpeg2_aac_low\" profile\n");
1034  ERROR_IF(s->options.ltp,
1035  "LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n");
1036  WARN_IF(s->options.pns,
1037  "PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n");
1038  s->options.pns = 0;
1039  } else if (avctx->profile == FF_PROFILE_AAC_LTP) {
1040  s->options.ltp = 1;
1041  ERROR_IF(s->options.pred,
1042  "Main prediction unavailable in the \"aac_ltp\" profile\n");
1043  } else if (avctx->profile == FF_PROFILE_AAC_MAIN) {
1044  s->options.pred = 1;
1045  ERROR_IF(s->options.ltp,
1046  "LTP prediction unavailable in the \"aac_main\" profile\n");
1047  } else if (s->options.ltp) {
1048  avctx->profile = FF_PROFILE_AAC_LTP;
1049  WARN_IF(1,
1050  "Chainging profile to \"aac_ltp\"\n");
1051  ERROR_IF(s->options.pred,
1052  "Main prediction unavailable in the \"aac_ltp\" profile\n");
1053  } else if (s->options.pred) {
1054  avctx->profile = FF_PROFILE_AAC_MAIN;
1055  WARN_IF(1,
1056  "Chainging profile to \"aac_main\"\n");
1057  ERROR_IF(s->options.ltp,
1058  "LTP prediction unavailable in the \"aac_main\" profile\n");
1059  }
1060  s->profile = avctx->profile;
1061 
1062  /* Coder limitations */
1063  s->coder = &ff_aac_coders[s->options.coder];
1064  if (s->options.coder == AAC_CODER_ANMR) {
1066  "The ANMR coder is considered experimental, add -strict -2 to enable!\n");
1067  s->options.intensity_stereo = 0;
1068  s->options.pns = 0;
1069  }
1071  "The LPT profile requires experimental compliance, add -strict -2 to enable!\n");
1072 
1073  /* M/S introduces horrible artifacts with multichannel files, this is temporary */
1074  if (s->channels > 3)
1075  s->options.mid_side = 0;
1076 
1077  if ((ret = dsp_init(avctx, s)) < 0)
1078  goto fail;
1079 
1080  if ((ret = alloc_buffers(avctx, s)) < 0)
1081  goto fail;
1082 
1083  if ((ret = put_audio_specific_config(avctx)))
1084  goto fail;
1085 
1086  sizes[0] = ff_aac_swb_size_1024[s->samplerate_index];
1087  sizes[1] = ff_aac_swb_size_128[s->samplerate_index];
1088  lengths[0] = ff_aac_num_swb_1024[s->samplerate_index];
1089  lengths[1] = ff_aac_num_swb_128[s->samplerate_index];
1090  for (i = 0; i < s->chan_map[0]; i++)
1091  grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
1092  if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
1093  s->chan_map[0], grouping)) < 0)
1094  goto fail;
1095  s->psypp = ff_psy_preprocess_init(avctx);
1097  s->random_state = 0x1f2e3d4c;
1098 
1099  s->abs_pow34 = abs_pow34_v;
1101 
1102  if (ARCH_X86)
1104 
1105  if (HAVE_MIPSDSP)
1107 
1109  return AVERROR_UNKNOWN;
1110 
1111  ff_af_queue_init(avctx, &s->afq);
1112 
1113  return 0;
1114 fail:
1115  aac_encode_end(avctx);
1116  return ret;
1117 }
1118 
1119 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1120 static const AVOption aacenc_options[] = {
1121  {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
1122  {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1123  {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1124  {"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1125  {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
1126  {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1127  {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1128  {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1129  {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1130  {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1131  {"aac_pce", "Forces the use of PCEs", offsetof(AACEncContext, options.pce), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1132  {NULL}
1133 };
1134 
1135 static const AVClass aacenc_class = {
1136  .class_name = "AAC encoder",
1137  .item_name = av_default_item_name,
1138  .option = aacenc_options,
1139  .version = LIBAVUTIL_VERSION_INT,
1140 };
1141 
1143  { "b", "0" },
1144  { NULL }
1145 };
1146 
1148  .name = "aac",
1149  .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
1150  .type = AVMEDIA_TYPE_AUDIO,
1151  .id = AV_CODEC_ID_AAC,
1152  .priv_data_size = sizeof(AACEncContext),
1153  .init = aac_encode_init,
1154  .encode2 = aac_encode_frame,
1155  .close = aac_encode_end,
1156  .defaults = aac_encode_defaults,
1157  .supported_samplerates = mpeg4audio_sample_rates,
1158  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
1160  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
1162  .priv_class = &aacenc_class,
1163 };
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
Definition: avcodec.h:2551
float, planar
Definition: samplefmt.h:69
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
Definition: aacenc.c:126
#define NULL
Definition: coverity.c:32
const char * s
Definition: avisynth_c.h:768
void(* search_for_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce)
Definition: aacenc.h:73
const AACCoefficientsEncoder * coder
Definition: aacenc.h:397
Band types following are encoded differently from others.
Definition: aac.h:86
static const uint8_t aac_chan_configs[AAC_MAX_CHANNELS][6]
default channel configurations
Definition: aacenctab.h:58
uint8_t use_kb_window[2]
If set, use Kaiser-Bessel window, otherwise use a sine window.
Definition: aac.h:177
int coder
Definition: aacenc.h:44
This structure describes decoded (raw) audio or video data.
Definition: frame.h:201
#define FF_ALLOCZ_ARRAY_OR_GOTO(ctx, p, nelem, elsize, label)
Definition: internal.h:167
int grouping[8]
window grouping (for e.g. AAC)
Definition: psymodel.h:81
AVOption.
Definition: opt.h:246
enum RawDataBlockType cur_type
channel group type cur_channel belongs to
Definition: aacenc.h:404
uint8_t ** bands
scalefactor band sizes for possible frame sizes
Definition: psymodel.h:98
Definition: aac.h:224
void(* mdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:109
struct FFPsyContext::@118 bitres
AACQuantizeBandCostCacheEntry quantize_band_cost_cache[256][128]
memoization area for quantize_band_cost
Definition: aacenc.h:411
static void abs_pow34_v(float *out, const float *in, const int size)
Definition: aacenc_utils.h:40
static const AVClass aacenc_class
Definition: aacenc.c:1135
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:207
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
Generate a Kaiser-Bessel Derived Window.
Definition: kbdwin.c:26
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1538
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
#define SCALE_DIFF_ZERO
codebook index corresponding to zero scalefactor indices difference
Definition: aac.h:152
uint8_t window_clipping[8]
set if a certain window is near clipping
Definition: aac.h:191
Definition: aac.h:63
const char * g
Definition: vf_curves.c:112
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
void(* encode_ltp_info)(struct AACEncContext *s, SingleChannelElement *sce, int common_window)
Definition: aacenc.h:64
Definition: aac.h:57
channels
Definition: aptx.c:30
av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
Cleanup audio preprocessing module.
Definition: psymodel.c:152
#define WARN_IF(cond,...)
Definition: aacenc_utils.h:274
int size
Definition: avcodec.h:1401
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:193
const int ff_aac_swb_size_1024_len
Definition: aacenctab.c:108
void avpriv_align_put_bits(PutBitContext *s)
Pad the bitstream with zeros up to the next byte boundary.
Definition: bitstream.c:48
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
Encode ics_info element.
Definition: aacenc.c:221
int common_window
Set if channels share a common 'IndividualChannelStream' in bitstream.
Definition: aac.h:278
int alloc
number of bits allocated by the psy, or -1 if no allocation was done
Definition: psymodel.h:105
const uint8_t * ff_aac_swb_size_1024[]
Definition: aacenctab.c:99
#define FF_PROFILE_AAC_MAIN
Definition: avcodec.h:2817
int lambda_count
count(lambda), for Qvg reporting
Definition: aacenc.h:403
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
uint8_t ms_mask[128]
Set if mid/side stereo is used for each scalefactor window band.
Definition: aac.h:281
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:236
float lambda
Definition: aacenc.h:400
#define NOISE_PRE
preamble for NOISE_BT, put in bitstream with the first noise band
Definition: aac.h:156
int profile
profile
Definition: avcodec.h:2813
AVCodec.
Definition: avcodec.h:3351
void(* search_for_tns)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:75
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
Encode spectral coefficients processed by psychoacoustic model.
Definition: aacenc.c:439
Spectral data are scaled white noise not coded in the bitstream.
Definition: aac.h:87
int * num_bands
number of scalefactor bands for possible frame sizes
Definition: psymodel.h:99
static AVOnce aac_table_init
Definition: aacenc.c:51
static int put_audio_specific_config(AVCodecContext *avctx)
Make AAC audio config object.
Definition: aacenc.c:94
const uint8_t ff_aac_num_swb_128[]
Definition: aactab.c:61
INTFLOAT pcoeffs[1024]
coefficients for IMDCT, pristine
Definition: aac.h:261
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:181
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
#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: avcodec.h:981
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
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:32
AACEncOptions options
encoding options
Definition: aacenc.h:378
#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:40
AAC encoder context.
Definition: aacenc.h:376
int num_ele[4]
front, side, back, lfe
Definition: aacenc.h:95
uint8_t
#define av_cold
Definition: attributes.h:82
void(* quant_bands)(int *out, const float *in, const float *scaled, int size, int is_signed, int maxval, const float Q34, const float rounding)
Definition: aacenc.h:414
AVOptions.
int intensity_stereo
Definition: aacenc.h:51
#define WINDOW_FUNC(type)
Definition: aacenc.c:135
LPCContext lpc
used by TNS
Definition: aacenc.h:388
void ff_aac_coder_init_mips(AACEncContext *c)
SingleChannelElement ch[2]
Definition: aac.h:284
int samplerate_index
MPEG-4 samplerate index.
Definition: aacenc.h:389
Definition: aac.h:59
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: avcodec.h:1418
const uint8_t * chan_map
channel configuration map
Definition: aacenc.h:392
TemporalNoiseShaping tns
Definition: aac.h:250
const uint8_t ff_aac_scalefactor_bits[121]
Definition: aactab.c:92
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1588
AudioFrameQueue afq
Definition: aacenc.h:406
const AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB]
Definition: aaccoder.c:897
static AVFrame * frame
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
#define FF_PROFILE_AAC_LTP
Definition: avcodec.h:2820
uint8_t * data
Definition: avcodec.h:1400
static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
Definition: aacenc.c:53
const uint8_t * ff_aac_swb_size_128[]
Definition: aacenctab.c:91
uint32_t tag
Definition: movenc.c:1448
Scalefactor data are intensity stereo positions (in phase).
Definition: aac.h:89
int profile
copied from avctx
Definition: aacenc.h:386
#define AVOnce
Definition: thread.h:157
const OptionDef options[]
Definition: ffserver.c:3948
uint8_t reorder_map[16]
maps channels from lavc to aac order
Definition: aacenc.h:99
static void adjust_frame_information(ChannelElement *cpe, int chans)
Produce integer coefficients from scalefactors provided by the model.
Definition: aacenc.c:256
#define av_log(a,...)
static const AVOption aacenc_options[]
Definition: aacenc.c:1120
int64_t layout
Definition: aacenc.h:94
const uint8_t * reorder_map
lavc to aac reorder map
Definition: aacenc.h:391
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define R
Definition: huffyuvdsp.h:34
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:127
static const int sizes[][2]
Definition: img2dec.c:51
#define AVERROR(e)
Definition: error.h:43
const uint8_t ff_aac_num_swb_1024[]
Definition: aactab.c:45
#define FF_PROFILE_MPEG2_AAC_LOW
Definition: avcodec.h:2825
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
float is_ener[128]
Intensity stereo pos (used by encoder)
Definition: aac.h:259
int initial_padding
Audio only.
Definition: avcodec.h:2989
static const AACPCEInfo aac_pce_configs[]
List of PCE (Program Configuration Element) for the channel layouts listed in channel_layout.h.
Definition: aacenc.h:137
float ff_aac_kbd_long_1024[1024]
Definition: aactab.c:38
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1568
static const int mpeg4audio_sample_rates[16]
Definition: aacenctab.h:85
int amp[4]
Definition: aac.h:228
const char * name
Name of the codec implementation.
Definition: avcodec.h:3358
int num_windows
number of windows in a frame
Definition: psymodel.h:80
static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
Definition: aacenc.c:534
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:175
static const AVCodecDefault defaults[]
Definition: amfenc_h264.c:361
struct AACEncContext::@42 buffer
#define ff_mdct_init
Definition: fft.h:169
Definition: aac.h:62
void(* set_special_band_scalefactors)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:72
int num_swb
number of scalefactor window bands
Definition: aac.h:183
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
#define FFMAX(a, b)
Definition: common.h:94
#define fail()
Definition: checkasm.h:112
void(* search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s, SingleChannelElement *sce, const float lambda)
Definition: aacenc.h:57
int index[4][8]
front, side, back, lfe
Definition: aacenc.h:97
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2194
static int put_bits_count(PutBitContext *s)
Definition: put_bits.h:85
#define AACENC_FLAGS
Definition: aacenc.c:1119
INTFLOAT ret_buf[2048]
PCM output buffer.
Definition: aac.h:264
void(* search_for_is)(struct AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe)
Definition: aacenc.h:78
void(* abs_pow34)(float *out, const float *in, const int size)
Definition: aacenc.h:413
enum WindowSequence window_sequence[2]
Definition: aac.h:176
INTFLOAT ltp_state[3072]
time signal for LTP
Definition: aac.h:265
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:883
av_cold void ff_lpc_end(LPCContext *s)
Uninitialize LPCContext.
Definition: lpc.c:322
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:830
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:986
int cur_channel
current channel for coder context
Definition: aacenc.h:398
int last_frame_pb_count
number of bits for the previous frame
Definition: aacenc.h:401
#define FFMIN(a, b)
Definition: common.h:96
static void apply_intensity_stereo(ChannelElement *cpe)
Definition: aacenc.c:306
static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: aacenc.c:556
void(* apply_tns_filt)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:69
void(* analyze)(FFPsyContext *ctx, int channel, const float **coeffs, const FFPsyWindowInfo *wi)
Perform psychoacoustic analysis and set band info (threshold, energy) for a group of channels...
Definition: psymodel.h:139
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
#define FF_PROFILE_AAC_LOW
Definition: avcodec.h:2818
static const AVCodecDefault aac_encode_defaults[]
Definition: aacenc.c:1142
#define FF_PROFILE_UNKNOWN
Definition: avcodec.h:2814
int pos[4]
Definition: aac.h:227
void(* quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, int size, int scale_idx, int cb, const float lambda, int rtz)
Definition: aacenc.h:61
int channels
channel count
Definition: aacenc.h:390
void(* encode_main_pred)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:65
AAC definitions and structures.
void(* search_for_ltp)(struct AACEncContext *s, SingleChannelElement *sce, int common_window)
Definition: aacenc.h:76
const uint8_t ff_tns_max_bands_1024[]
Definition: aactab.c:1394
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
FFTContext mdct128
short (128 samples) frame transform context
Definition: aacenc.h:381
PutBitContext pb
Definition: aacenc.h:379
static void(*const apply_window[4])(AVFloatDSPContext *fdsp, SingleChannelElement *sce, const float *audio)
Definition: aacenc.c:191
void(* search_for_pred)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:79
#define L(x)
Definition: vp56_arith.h:36
AVFloatDSPContext * fdsp
Definition: aacenc.h:382
int mid_side
Definition: aacenc.h:50
#define FF_ARRAY_ELEMS(a)
void(* adjust_common_pred)(struct AACEncContext *s, ChannelElement *cpe)
Definition: aacenc.h:66
static av_cold int aac_encode_end(AVCodecContext *avctx)
Definition: aacenc.c:899
void ff_aac_dsp_init_x86(AACEncContext *s)
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2163
#define AV_LOG_INFO
Standard information.
Definition: log.h:187
#define AV_ONCE_INIT
Definition: thread.h:158
#define CLIP_AVOIDANCE_FACTOR
Definition: aac.h:53
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
void(* search_for_ms)(struct AACEncContext *s, ChannelElement *cpe)
Definition: aacenc.h:77
Temporal Noise Shaping.
Definition: aac.h:198
int sample_rate
samples per second
Definition: avcodec.h:2143
float ff_aac_kbd_short_128[128]
Definition: aactab.c:39
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
Encode MS data.
Definition: aacenc.c:242
FFPsyWindowInfo(* window)(FFPsyContext *ctx, const float *audio, const float *la, int channel, int prev_type)
Suggest window sequence for channel.
Definition: psymodel.h:129
void(* apply_main_pred)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:68
main external API structure.
Definition: avcodec.h:1488
int pairing[3][8]
front, side, back
Definition: aacenc.h:96
int bits
number of bits used in the bitresevoir
Definition: psymodel.h:104
#define NOISE_PRE_BITS
length of preamble
Definition: aac.h:157
Levinson-Durbin recursion.
Definition: lpc.h:47
IndividualChannelStream ics
Definition: aac.h:249
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
void(* encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce, int win, int group_len, const float lambda)
Definition: aacenc.h:59
void(* mark_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce)
Definition: aacenc.h:74
int extradata_size
Definition: avcodec.h:1589
uint8_t group_len[8]
Definition: aac.h:179
Replacements for frequently missing libm functions.
float lambda_sum
sum(lambda), for Qvg reporting
Definition: aacenc.h:402
Describe the class of an AVClass context structure.
Definition: log.h:67
static void put_bitstream_info(AACEncContext *s, const char *name)
Write some auxiliary information about the created AAC file.
Definition: aacenc.c:513
const int ff_aac_swb_size_128_len
Definition: aacenctab.c:107
int window_shape
window shape (sine/KBD/whatever)
Definition: psymodel.h:79
static void encode_pulses(AACEncContext *s, Pulse *pulse)
Encode pulse data.
Definition: aacenc.c:420
uint16_t quantize_band_cost_cache_generation
Definition: aacenc.h:410
static av_cold void aac_encode_init_tables(void)
Definition: aacenc.c:954
const uint8_t * swb_sizes
table of scalefactor band sizes for a particular window
Definition: aac.h:182
#define TNS_MAX_ORDER
Definition: aac.h:50
FFPsyContext psy
Definition: aacenc.h:395
const uint32_t ff_aac_scalefactor_code[121]
Definition: aactab.c:73
LongTermPrediction ltp
Definition: aac.h:180
static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
Definition: aacenc.c:940
const struct FFPsyModel * model
encoder-specific model functions
Definition: psymodel.h:91
av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order, enum FFLPCType lpc_type)
Initialize LPCContext.
Definition: lpc.c:300
#define AAC_MAX_CHANNELS
Definition: aacenctab.h:39
int needs_pce
flag for non-standard layout
Definition: aacenc.h:387
int ms_mode
Signals mid/side stereo flags coding mode (used by encoder)
Definition: aac.h:279
AAC encoder data.
const uint8_t ff_tns_max_bands_128[]
Definition: aactab.c:1406
struct FFPsyPreprocessContext * psypp
Definition: aacenc.h:396
#define NOISE_OFFSET
subtracted from global gain, used as offset for the preamble
Definition: aac.h:158
void(* encode_tns_info)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:63
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:1554
uint8_t zeroes[128]
band is not coded (used by encoder)
Definition: aac.h:257
int sf_idx[128]
scalefactor indices (used by encoder)
Definition: aac.h:256
AVCodec ff_aac_encoder
Definition: aacenc.c:1147
uint8_t is_mode
Set if any bands have been encoded using intensity stereo (used by encoder)
Definition: aac.h:280
INTFLOAT coeffs[1024]
coefficients for IMDCT, maybe processed
Definition: aac.h:262
const int avpriv_mpeg4audio_sample_rates[16]
Definition: mpeg4audio.c:62
Scalefactor data are intensity stereo positions (out of phase).
Definition: aac.h:88
Y Spectral Band Replication.
Definition: mpeg4audio.h:79
float * samples
Definition: aacenc.h:419
uint8_t prediction_used[41]
Definition: aac.h:190
static av_cold int aac_encode_init(AVCodecContext *avctx)
Definition: aacenc.c:959
common internal api header.
AACPCEInfo pce
PCE data, if needed.
Definition: aacenc.h:383
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:101
if(ret< 0)
Definition: vf_mcdeint.c:279
AAC encoder utilities.
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:248
windowing related information
Definition: psymodel.h:77
#define ff_mdct_end
Definition: fft.h:170
av_cold struct FFPsyPreprocessContext * ff_psy_preprocess_init(AVCodecContext *avctx)
psychoacoustic model audio preprocessing initialization
Definition: psymodel.c:103
const uint16_t *const ff_swb_offset_1024[]
Definition: aactab.c:1336
uint8_t config_map[16]
configs the encoder's channel specific settings
Definition: aacenc.h:98
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
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce)
Encode scalefactors.
Definition: aacenc.c:384
float * planar_samples[16]
saved preprocessed input
Definition: aacenc.h:384
ChannelElement * cpe
channel elements
Definition: aacenc.h:394
Individual Channel Stream.
Definition: aac.h:174
float clip_avoidance_factor
set if any window is near clipping to the necessary atennuation factor to avoid it ...
Definition: aac.h:192
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:48
#define AVERROR_UNKNOWN
Unknown error, typically from an external library.
Definition: error.h:71
void(* ltp_insert_new_frame)(struct AACEncContext *s)
Definition: aacenc.h:71
#define ERROR_IF(cond,...)
Definition: aacenc_utils.h:268
static void ff_aac_tableinit(void)
Definition: aactab.h:45
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:275
void * priv_data
Definition: avcodec.h:1515
int start
Definition: aac.h:226
FFTContext mdct1024
long (1024 samples) frame transform context
Definition: aacenc.h:380
int random_state
Definition: aacenc.h:399
static av_always_inline int diff(const uint32_t a, const uint32_t b)
static const int16_t coeffs[]
int channels
number of audio channels
Definition: avcodec.h:2144
int num_pulse
Definition: aac.h:225
AAC_FLOAT lcoeffs[1024]
MDCT of LTP coefficients (used by encoder)
Definition: aac.h:266
static int ff_thread_once(char *control, void(*routine)(void))
Definition: thread.h:160
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
Encode scalefactor band coding type.
Definition: aacenc.c:370
static void apply_mid_side_stereo(ChannelElement *cpe)
Definition: aacenc.c:335
static const int64_t aac_normal_chan_layouts[7]
Definition: aacenctab.h:47
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
enum BandType band_type[128]
band types
Definition: aac.h:252
#define LIBAVCODEC_IDENT
Definition: version.h:42
void avpriv_put_string(PutBitContext *pb, const char *string, int terminate_string)
Put the string string in the bitstream.
Definition: bitstream.c:53
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
int frame_number
Frame counter, set by libavcodec.
Definition: avcodec.h:2174
FILE * out
Definition: movenc.c:54
#define av_freep(p)
void INT64 start
Definition: avisynth_c.h:690
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, int common_window)
Encode one channel of audio data.
Definition: aacenc.c:487
void(* adjust_common_ltp)(struct AACEncContext *s, ChannelElement *cpe)
Definition: aacenc.h:67
int8_t used[MAX_LTP_LONG_SFB]
Definition: aac.h:168
static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, float *audio)
Definition: aacenc.c:200
const uint16_t *const ff_swb_offset_128[]
Definition: aactab.c:1368
int8_t present
Definition: aac.h:164
uint8_t is_mask[128]
Set if intensity stereo is used (used by encoder)
Definition: aac.h:282
static const int aacenc_profiles[]
Definition: aacenctab.h:132
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:248
AAC data declarations.
av_cold void ff_psy_end(FFPsyContext *ctx)
Cleanup model context at the end.
Definition: psymodel.c:83
This structure stores compressed data.
Definition: avcodec.h:1377
static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
Downscale spectral coefficients for near-clipping windows to avoid artifacts.
Definition: aacenc.c:467
int window_type[3]
window type (short/long/transitional, etc.) - current, previous and next
Definition: psymodel.h:78
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:267
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:2546
static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
Definition: aacenc.c:918
void(* update_ltp)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:70
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1393
for(j=16;j >0;--j)
int pred
Definition: aacenc.h:49
void AAC_RENAME() ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
float clipping[8]
maximum absolute normalized intensity in the given window for clip avoidance
Definition: psymodel.h:82
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
const char * name
Definition: opengl_enc.c:103
bitstream writer API