FFmpeg
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 #include "profiles.h"
43 
44 #include "aac.h"
45 #include "aactab.h"
46 #include "aacenc.h"
47 #include "aacenctab.h"
48 #include "aacenc_utils.h"
49 
50 #include "psymodel.h"
51 
53 
54 static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
55 {
56  int i, j;
57  AACEncContext *s = avctx->priv_data;
58  AACPCEInfo *pce = &s->pce;
59  const int bitexact = avctx->flags & AV_CODEC_FLAG_BITEXACT;
60  const char *aux_data = bitexact ? "Lavc" : LIBAVCODEC_IDENT;
61 
62  put_bits(pb, 4, 0);
63 
64  put_bits(pb, 2, avctx->profile);
65  put_bits(pb, 4, s->samplerate_index);
66 
67  put_bits(pb, 4, pce->num_ele[0]); /* Front */
68  put_bits(pb, 4, pce->num_ele[1]); /* Side */
69  put_bits(pb, 4, pce->num_ele[2]); /* Back */
70  put_bits(pb, 2, pce->num_ele[3]); /* LFE */
71  put_bits(pb, 3, 0); /* Assoc data */
72  put_bits(pb, 4, 0); /* CCs */
73 
74  put_bits(pb, 1, 0); /* Stereo mixdown */
75  put_bits(pb, 1, 0); /* Mono mixdown */
76  put_bits(pb, 1, 0); /* Something else */
77 
78  for (i = 0; i < 4; i++) {
79  for (j = 0; j < pce->num_ele[i]; j++) {
80  if (i < 3)
81  put_bits(pb, 1, pce->pairing[i][j]);
82  put_bits(pb, 4, pce->index[i][j]);
83  }
84  }
85 
87  put_bits(pb, 8, strlen(aux_data));
88  avpriv_put_string(pb, aux_data, 0);
89 }
90 
91 /**
92  * Make AAC audio config object.
93  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
94  */
96 {
97  PutBitContext pb;
98  AACEncContext *s = avctx->priv_data;
99  int channels = (!s->needs_pce)*(s->channels - (s->channels == 8 ? 1 : 0));
100  const int max_size = 32;
101 
102  avctx->extradata = av_mallocz(max_size);
103  if (!avctx->extradata)
104  return AVERROR(ENOMEM);
105 
106  init_put_bits(&pb, avctx->extradata, max_size);
107  put_bits(&pb, 5, s->profile+1); //profile
108  put_bits(&pb, 4, s->samplerate_index); //sample rate index
109  put_bits(&pb, 4, channels);
110  //GASpecificConfig
111  put_bits(&pb, 1, 0); //frame length - 1024 samples
112  put_bits(&pb, 1, 0); //does not depend on core coder
113  put_bits(&pb, 1, 0); //is not extension
114  if (s->needs_pce)
115  put_pce(&pb, avctx);
116 
117  //Explicitly Mark SBR absent
118  put_bits(&pb, 11, 0x2b7); //sync extension
119  put_bits(&pb, 5, AOT_SBR);
120  put_bits(&pb, 1, 0);
121  flush_put_bits(&pb);
122  avctx->extradata_size = put_bits_count(&pb) >> 3;
123 
124  return 0;
125 }
126 
128 {
131  memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache));
133  }
134 }
135 
136 #define WINDOW_FUNC(type) \
137 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
138  SingleChannelElement *sce, \
139  const float *audio)
140 
141 WINDOW_FUNC(only_long)
142 {
143  const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
144  const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
145  float *out = sce->ret_buf;
146 
147  fdsp->vector_fmul (out, audio, lwindow, 1024);
148  fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
149 }
150 
151 WINDOW_FUNC(long_start)
152 {
153  const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
154  const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
155  float *out = sce->ret_buf;
156 
157  fdsp->vector_fmul(out, audio, lwindow, 1024);
158  memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
159  fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
160  memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
161 }
162 
163 WINDOW_FUNC(long_stop)
164 {
165  const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
166  const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
167  float *out = sce->ret_buf;
168 
169  memset(out, 0, sizeof(out[0]) * 448);
170  fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
171  memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
172  fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
173 }
174 
175 WINDOW_FUNC(eight_short)
176 {
177  const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
178  const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
179  const float *in = audio + 448;
180  float *out = sce->ret_buf;
181  int w;
182 
183  for (w = 0; w < 8; w++) {
184  fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
185  out += 128;
186  in += 128;
187  fdsp->vector_fmul_reverse(out, in, swindow, 128);
188  out += 128;
189  }
190 }
191 
192 static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
194  const float *audio) = {
195  [ONLY_LONG_SEQUENCE] = apply_only_long_window,
196  [LONG_START_SEQUENCE] = apply_long_start_window,
197  [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
198  [LONG_STOP_SEQUENCE] = apply_long_stop_window
199 };
200 
202  float *audio)
203 {
204  int i;
205  const float *output = sce->ret_buf;
206 
207  apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
208 
210  s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
211  else
212  for (i = 0; i < 1024; i += 128)
213  s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
214  memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
215  memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
216 }
217 
218 /**
219  * Encode ics_info element.
220  * @see Table 4.6 (syntax of ics_info)
221  */
223 {
224  int w;
225 
226  put_bits(&s->pb, 1, 0); // ics_reserved bit
227  put_bits(&s->pb, 2, info->window_sequence[0]);
228  put_bits(&s->pb, 1, info->use_kb_window[0]);
229  if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
230  put_bits(&s->pb, 6, info->max_sfb);
231  put_bits(&s->pb, 1, !!info->predictor_present);
232  } else {
233  put_bits(&s->pb, 4, info->max_sfb);
234  for (w = 1; w < 8; w++)
235  put_bits(&s->pb, 1, !info->group_len[w]);
236  }
237 }
238 
239 /**
240  * Encode MS data.
241  * @see 4.6.8.1 "Joint Coding - M/S Stereo"
242  */
244 {
245  int i, w;
246 
247  put_bits(pb, 2, cpe->ms_mode);
248  if (cpe->ms_mode == 1)
249  for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
250  for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
251  put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
252 }
253 
254 /**
255  * Produce integer coefficients from scalefactors provided by the model.
256  */
257 static void adjust_frame_information(ChannelElement *cpe, int chans)
258 {
259  int i, w, w2, g, ch;
260  int maxsfb, cmaxsfb;
261 
262  for (ch = 0; ch < chans; ch++) {
263  IndividualChannelStream *ics = &cpe->ch[ch].ics;
264  maxsfb = 0;
265  cpe->ch[ch].pulse.num_pulse = 0;
266  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
267  for (w2 = 0; w2 < ics->group_len[w]; w2++) {
268  for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
269  ;
270  maxsfb = FFMAX(maxsfb, cmaxsfb);
271  }
272  }
273  ics->max_sfb = maxsfb;
274 
275  //adjust zero bands for window groups
276  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
277  for (g = 0; g < ics->max_sfb; g++) {
278  i = 1;
279  for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
280  if (!cpe->ch[ch].zeroes[w2*16 + g]) {
281  i = 0;
282  break;
283  }
284  }
285  cpe->ch[ch].zeroes[w*16 + g] = i;
286  }
287  }
288  }
289 
290  if (chans > 1 && cpe->common_window) {
291  IndividualChannelStream *ics0 = &cpe->ch[0].ics;
292  IndividualChannelStream *ics1 = &cpe->ch[1].ics;
293  int msc = 0;
294  ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
295  ics1->max_sfb = ics0->max_sfb;
296  for (w = 0; w < ics0->num_windows*16; w += 16)
297  for (i = 0; i < ics0->max_sfb; i++)
298  if (cpe->ms_mask[w+i])
299  msc++;
300  if (msc == 0 || ics0->max_sfb == 0)
301  cpe->ms_mode = 0;
302  else
303  cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
304  }
305 }
306 
308 {
309  int w, w2, g, i;
310  IndividualChannelStream *ics = &cpe->ch[0].ics;
311  if (!cpe->common_window)
312  return;
313  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
314  for (w2 = 0; w2 < ics->group_len[w]; w2++) {
315  int start = (w+w2) * 128;
316  for (g = 0; g < ics->num_swb; g++) {
317  int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
318  float scale = cpe->ch[0].is_ener[w*16+g];
319  if (!cpe->is_mask[w*16 + g]) {
320  start += ics->swb_sizes[g];
321  continue;
322  }
323  if (cpe->ms_mask[w*16 + g])
324  p *= -1;
325  for (i = 0; i < ics->swb_sizes[g]; i++) {
326  float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
327  cpe->ch[0].coeffs[start+i] = sum;
328  cpe->ch[1].coeffs[start+i] = 0.0f;
329  }
330  start += ics->swb_sizes[g];
331  }
332  }
333  }
334 }
335 
337 {
338  int w, w2, g, i;
339  IndividualChannelStream *ics = &cpe->ch[0].ics;
340  if (!cpe->common_window)
341  return;
342  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
343  for (w2 = 0; w2 < ics->group_len[w]; w2++) {
344  int start = (w+w2) * 128;
345  for (g = 0; g < ics->num_swb; g++) {
346  /* ms_mask can be used for other purposes in PNS and I/S,
347  * so must not apply M/S if any band uses either, even if
348  * ms_mask is set.
349  */
350  if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]
351  || cpe->ch[0].band_type[w*16 + g] >= NOISE_BT
352  || cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) {
353  start += ics->swb_sizes[g];
354  continue;
355  }
356  for (i = 0; i < ics->swb_sizes[g]; i++) {
357  float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
358  float R = L - cpe->ch[1].coeffs[start+i];
359  cpe->ch[0].coeffs[start+i] = L;
360  cpe->ch[1].coeffs[start+i] = R;
361  }
362  start += ics->swb_sizes[g];
363  }
364  }
365  }
366 }
367 
368 /**
369  * Encode scalefactor band coding type.
370  */
372 {
373  int w;
374 
377 
378  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
379  s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
380 }
381 
382 /**
383  * Encode scalefactors.
384  */
387 {
388  int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
389  int off_is = 0, noise_flag = 1;
390  int i, w;
391 
392  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
393  for (i = 0; i < sce->ics.max_sfb; i++) {
394  if (!sce->zeroes[w*16 + i]) {
395  if (sce->band_type[w*16 + i] == NOISE_BT) {
396  diff = sce->sf_idx[w*16 + i] - off_pns;
397  off_pns = sce->sf_idx[w*16 + i];
398  if (noise_flag-- > 0) {
399  put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
400  continue;
401  }
402  } else if (sce->band_type[w*16 + i] == INTENSITY_BT ||
403  sce->band_type[w*16 + i] == INTENSITY_BT2) {
404  diff = sce->sf_idx[w*16 + i] - off_is;
405  off_is = sce->sf_idx[w*16 + i];
406  } else {
407  diff = sce->sf_idx[w*16 + i] - off_sf;
408  off_sf = sce->sf_idx[w*16 + i];
409  }
410  diff += SCALE_DIFF_ZERO;
411  av_assert0(diff >= 0 && diff <= 120);
413  }
414  }
415  }
416 }
417 
418 /**
419  * Encode pulse data.
420  */
421 static void encode_pulses(AACEncContext *s, Pulse *pulse)
422 {
423  int i;
424 
425  put_bits(&s->pb, 1, !!pulse->num_pulse);
426  if (!pulse->num_pulse)
427  return;
428 
429  put_bits(&s->pb, 2, pulse->num_pulse - 1);
430  put_bits(&s->pb, 6, pulse->start);
431  for (i = 0; i < pulse->num_pulse; i++) {
432  put_bits(&s->pb, 5, pulse->pos[i]);
433  put_bits(&s->pb, 4, pulse->amp[i]);
434  }
435 }
436 
437 /**
438  * Encode spectral coefficients processed by psychoacoustic model.
439  */
441 {
442  int start, i, w, w2;
443 
444  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
445  start = 0;
446  for (i = 0; i < sce->ics.max_sfb; i++) {
447  if (sce->zeroes[w*16 + i]) {
448  start += sce->ics.swb_sizes[i];
449  continue;
450  }
451  for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
452  s->coder->quantize_and_encode_band(s, &s->pb,
453  &sce->coeffs[start + w2*128],
454  NULL, sce->ics.swb_sizes[i],
455  sce->sf_idx[w*16 + i],
456  sce->band_type[w*16 + i],
457  s->lambda,
458  sce->ics.window_clipping[w]);
459  }
460  start += sce->ics.swb_sizes[i];
461  }
462  }
463 }
464 
465 /**
466  * Downscale spectral coefficients for near-clipping windows to avoid artifacts
467  */
469 {
470  int start, i, j, w;
471 
472  if (sce->ics.clip_avoidance_factor < 1.0f) {
473  for (w = 0; w < sce->ics.num_windows; w++) {
474  start = 0;
475  for (i = 0; i < sce->ics.max_sfb; i++) {
476  float *swb_coeffs = &sce->coeffs[start + w*128];
477  for (j = 0; j < sce->ics.swb_sizes[i]; j++)
478  swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
479  start += sce->ics.swb_sizes[i];
480  }
481  }
482  }
483 }
484 
485 /**
486  * Encode one channel of audio data.
487  */
490  int common_window)
491 {
492  put_bits(&s->pb, 8, sce->sf_idx[0]);
493  if (!common_window) {
494  put_ics_info(s, &sce->ics);
495  if (s->coder->encode_main_pred)
496  s->coder->encode_main_pred(s, sce);
497  if (s->coder->encode_ltp_info)
498  s->coder->encode_ltp_info(s, sce, 0);
499  }
500  encode_band_info(s, sce);
501  encode_scale_factors(avctx, s, sce);
502  encode_pulses(s, &sce->pulse);
503  put_bits(&s->pb, 1, !!sce->tns.present);
504  if (s->coder->encode_tns_info)
505  s->coder->encode_tns_info(s, sce);
506  put_bits(&s->pb, 1, 0); //ssr
507  encode_spectral_coeffs(s, sce);
508  return 0;
509 }
510 
511 /**
512  * Write some auxiliary information about the created AAC file.
513  */
514 static void put_bitstream_info(AACEncContext *s, const char *name)
515 {
516  int i, namelen, padbits;
517 
518  namelen = strlen(name) + 2;
519  put_bits(&s->pb, 3, TYPE_FIL);
520  put_bits(&s->pb, 4, FFMIN(namelen, 15));
521  if (namelen >= 15)
522  put_bits(&s->pb, 8, namelen - 14);
523  put_bits(&s->pb, 4, 0); //extension type - filler
524  padbits = -put_bits_count(&s->pb) & 7;
526  for (i = 0; i < namelen - 2; i++)
527  put_bits(&s->pb, 8, name[i]);
528  put_bits(&s->pb, 12 - padbits, 0);
529 }
530 
531 /*
532  * Copy input samples.
533  * Channels are reordered from libavcodec's default order to AAC order.
534  */
536 {
537  int ch;
538  int end = 2048 + (frame ? frame->nb_samples : 0);
539  const uint8_t *channel_map = s->reorder_map;
540 
541  /* copy and remap input samples */
542  for (ch = 0; ch < s->channels; ch++) {
543  /* copy last 1024 samples of previous frame to the start of the current frame */
544  memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
545 
546  /* copy new samples and zero any remaining samples */
547  if (frame) {
548  memcpy(&s->planar_samples[ch][2048],
549  frame->extended_data[channel_map[ch]],
550  frame->nb_samples * sizeof(s->planar_samples[0][0]));
551  }
552  memset(&s->planar_samples[ch][end], 0,
553  (3072 - end) * sizeof(s->planar_samples[0][0]));
554  }
555 }
556 
557 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
558  const AVFrame *frame, int *got_packet_ptr)
559 {
560  AACEncContext *s = avctx->priv_data;
561  float **samples = s->planar_samples, *samples2, *la, *overlap;
562  ChannelElement *cpe;
565  int i, its, ch, w, chans, tag, start_ch, ret, frame_bits;
566  int target_bits, rate_bits, too_many_bits, too_few_bits;
567  int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
568  int chan_el_counter[4];
570 
571  /* add current frame to queue */
572  if (frame) {
573  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
574  return ret;
575  } else {
576  if (!s->afq.remaining_samples || (!s->afq.frame_alloc && !s->afq.frame_count))
577  return 0;
578  }
579 
580  copy_input_samples(s, frame);
581  if (s->psypp)
583 
584  if (!avctx->frame_number)
585  return 0;
586 
587  start_ch = 0;
588  for (i = 0; i < s->chan_map[0]; i++) {
589  FFPsyWindowInfo* wi = windows + start_ch;
590  tag = s->chan_map[i+1];
591  chans = tag == TYPE_CPE ? 2 : 1;
592  cpe = &s->cpe[i];
593  for (ch = 0; ch < chans; ch++) {
594  int k;
595  float clip_avoidance_factor;
596  sce = &cpe->ch[ch];
597  ics = &sce->ics;
598  s->cur_channel = start_ch + ch;
599  overlap = &samples[s->cur_channel][0];
600  samples2 = overlap + 1024;
601  la = samples2 + (448+64);
602  if (!frame)
603  la = NULL;
604  if (tag == TYPE_LFE) {
605  wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE;
606  wi[ch].window_shape = 0;
607  wi[ch].num_windows = 1;
608  wi[ch].grouping[0] = 1;
609  wi[ch].clipping[0] = 0;
610 
611  /* Only the lowest 12 coefficients are used in a LFE channel.
612  * The expression below results in only the bottom 8 coefficients
613  * being used for 11.025kHz to 16kHz sample rates.
614  */
615  ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
616  } else {
617  wi[ch] = s->psy.model->window(&s->psy, samples2, la, s->cur_channel,
618  ics->window_sequence[0]);
619  }
620  ics->window_sequence[1] = ics->window_sequence[0];
621  ics->window_sequence[0] = wi[ch].window_type[0];
622  ics->use_kb_window[1] = ics->use_kb_window[0];
623  ics->use_kb_window[0] = wi[ch].window_shape;
624  ics->num_windows = wi[ch].num_windows;
625  ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
626  ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
627  ics->max_sfb = FFMIN(ics->max_sfb, ics->num_swb);
628  ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
634 
635  for (w = 0; w < ics->num_windows; w++)
636  ics->group_len[w] = wi[ch].grouping[w];
637 
638  /* Calculate input sample maximums and evaluate clipping risk */
639  clip_avoidance_factor = 0.0f;
640  for (w = 0; w < ics->num_windows; w++) {
641  const float *wbuf = overlap + w * 128;
642  const int wlen = 2048 / ics->num_windows;
643  float max = 0;
644  int j;
645  /* mdct input is 2 * output */
646  for (j = 0; j < wlen; j++)
647  max = FFMAX(max, fabsf(wbuf[j]));
648  wi[ch].clipping[w] = max;
649  }
650  for (w = 0; w < ics->num_windows; w++) {
651  if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
652  ics->window_clipping[w] = 1;
653  clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
654  } else {
655  ics->window_clipping[w] = 0;
656  }
657  }
658  if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
659  ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
660  } else {
661  ics->clip_avoidance_factor = 1.0f;
662  }
663 
664  apply_window_and_mdct(s, sce, overlap);
665 
666  if (s->options.ltp && s->coder->update_ltp) {
667  s->coder->update_ltp(s, sce);
668  apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]);
669  s->mdct1024.mdct_calc(&s->mdct1024, sce->lcoeffs, sce->ret_buf);
670  }
671 
672  for (k = 0; k < 1024; k++) {
673  if (!(fabs(cpe->ch[ch].coeffs[k]) < 1E16)) { // Ensure headroom for energy calculation
674  av_log(avctx, AV_LOG_ERROR, "Input contains (near) NaN/+-Inf\n");
675  return AVERROR(EINVAL);
676  }
677  }
678  avoid_clipping(s, sce);
679  }
680  start_ch += chans;
681  }
682  if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0)
683  return ret;
684  frame_bits = its = 0;
685  do {
686  init_put_bits(&s->pb, avpkt->data, avpkt->size);
687 
688  if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
690  start_ch = 0;
691  target_bits = 0;
692  memset(chan_el_counter, 0, sizeof(chan_el_counter));
693  for (i = 0; i < s->chan_map[0]; i++) {
694  FFPsyWindowInfo* wi = windows + start_ch;
695  const float *coeffs[2];
696  tag = s->chan_map[i+1];
697  chans = tag == TYPE_CPE ? 2 : 1;
698  cpe = &s->cpe[i];
699  cpe->common_window = 0;
700  memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
701  memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
702  put_bits(&s->pb, 3, tag);
703  put_bits(&s->pb, 4, chan_el_counter[tag]++);
704  for (ch = 0; ch < chans; ch++) {
705  sce = &cpe->ch[ch];
706  coeffs[ch] = sce->coeffs;
707  sce->ics.predictor_present = 0;
708  sce->ics.ltp.present = 0;
709  memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used));
710  memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
711  memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
712  for (w = 0; w < 128; w++)
713  if (sce->band_type[w] > RESERVED_BT)
714  sce->band_type[w] = 0;
715  }
716  s->psy.bitres.alloc = -1;
718  s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
719  if (s->psy.bitres.alloc > 0) {
720  /* Lambda unused here on purpose, we need to take psy's unscaled allocation */
721  target_bits += s->psy.bitres.alloc
722  * (s->lambda / (avctx->global_quality ? avctx->global_quality : 120));
723  s->psy.bitres.alloc /= chans;
724  }
725  s->cur_type = tag;
726  for (ch = 0; ch < chans; ch++) {
727  s->cur_channel = start_ch + ch;
728  if (s->options.pns && s->coder->mark_pns)
729  s->coder->mark_pns(s, avctx, &cpe->ch[ch]);
730  s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
731  }
732  if (chans > 1
733  && wi[0].window_type[0] == wi[1].window_type[0]
734  && wi[0].window_shape == wi[1].window_shape) {
735 
736  cpe->common_window = 1;
737  for (w = 0; w < wi[0].num_windows; w++) {
738  if (wi[0].grouping[w] != wi[1].grouping[w]) {
739  cpe->common_window = 0;
740  break;
741  }
742  }
743  }
744  for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
745  sce = &cpe->ch[ch];
746  s->cur_channel = start_ch + ch;
747  if (s->options.tns && s->coder->search_for_tns)
748  s->coder->search_for_tns(s, sce);
749  if (s->options.tns && s->coder->apply_tns_filt)
750  s->coder->apply_tns_filt(s, sce);
751  if (sce->tns.present)
752  tns_mode = 1;
753  if (s->options.pns && s->coder->search_for_pns)
754  s->coder->search_for_pns(s, avctx, sce);
755  }
756  s->cur_channel = start_ch;
757  if (s->options.intensity_stereo) { /* Intensity Stereo */
758  if (s->coder->search_for_is)
759  s->coder->search_for_is(s, avctx, cpe);
760  if (cpe->is_mode) is_mode = 1;
762  }
763  if (s->options.pred) { /* Prediction */
764  for (ch = 0; ch < chans; ch++) {
765  sce = &cpe->ch[ch];
766  s->cur_channel = start_ch + ch;
767  if (s->options.pred && s->coder->search_for_pred)
768  s->coder->search_for_pred(s, sce);
769  if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
770  }
771  if (s->coder->adjust_common_pred)
772  s->coder->adjust_common_pred(s, cpe);
773  for (ch = 0; ch < chans; ch++) {
774  sce = &cpe->ch[ch];
775  s->cur_channel = start_ch + ch;
776  if (s->options.pred && s->coder->apply_main_pred)
777  s->coder->apply_main_pred(s, sce);
778  }
779  s->cur_channel = start_ch;
780  }
781  if (s->options.mid_side) { /* Mid/Side stereo */
782  if (s->options.mid_side == -1 && s->coder->search_for_ms)
783  s->coder->search_for_ms(s, cpe);
784  else if (cpe->common_window)
785  memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
787  }
788  adjust_frame_information(cpe, chans);
789  if (s->options.ltp) { /* LTP */
790  for (ch = 0; ch < chans; ch++) {
791  sce = &cpe->ch[ch];
792  s->cur_channel = start_ch + ch;
793  if (s->coder->search_for_ltp)
794  s->coder->search_for_ltp(s, sce, cpe->common_window);
795  if (sce->ics.ltp.present) pred_mode = 1;
796  }
797  s->cur_channel = start_ch;
798  if (s->coder->adjust_common_ltp)
799  s->coder->adjust_common_ltp(s, cpe);
800  }
801  if (chans == 2) {
802  put_bits(&s->pb, 1, cpe->common_window);
803  if (cpe->common_window) {
804  put_ics_info(s, &cpe->ch[0].ics);
805  if (s->coder->encode_main_pred)
806  s->coder->encode_main_pred(s, &cpe->ch[0]);
807  if (s->coder->encode_ltp_info)
808  s->coder->encode_ltp_info(s, &cpe->ch[0], 1);
809  encode_ms_info(&s->pb, cpe);
810  if (cpe->ms_mode) ms_mode = 1;
811  }
812  }
813  for (ch = 0; ch < chans; ch++) {
814  s->cur_channel = start_ch + ch;
815  encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
816  }
817  start_ch += chans;
818  }
819 
820  if (avctx->flags & AV_CODEC_FLAG_QSCALE) {
821  /* When using a constant Q-scale, don't mess with lambda */
822  break;
823  }
824 
825  /* rate control stuff
826  * allow between the nominal bitrate, and what psy's bit reservoir says to target
827  * but drift towards the nominal bitrate always
828  */
829  frame_bits = put_bits_count(&s->pb);
830  rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate;
831  rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3);
832  too_many_bits = FFMAX(target_bits, rate_bits);
833  too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3);
834  too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits);
835 
836  /* When using ABR, be strict (but only for increasing) */
837  too_few_bits = too_few_bits - too_few_bits/8;
838  too_many_bits = too_many_bits + too_many_bits/2;
839 
840  if ( its == 0 /* for steady-state Q-scale tracking */
841  || (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits))
842  || frame_bits >= 6144 * s->channels - 3 )
843  {
844  float ratio = ((float)rate_bits) / frame_bits;
845 
846  if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) {
847  /*
848  * This path is for steady-state Q-scale tracking
849  * When frame bits fall within the stable range, we still need to adjust
850  * lambda to maintain it like so in a stable fashion (large jumps in lambda
851  * create artifacts and should be avoided), but slowly
852  */
853  ratio = sqrtf(sqrtf(ratio));
854  ratio = av_clipf(ratio, 0.9f, 1.1f);
855  } else {
856  /* Not so fast though */
857  ratio = sqrtf(ratio);
858  }
859  s->lambda = FFMIN(s->lambda * ratio, 65536.f);
860 
861  /* Keep iterating if we must reduce and lambda is in the sky */
862  if (ratio > 0.9f && ratio < 1.1f) {
863  break;
864  } else {
865  if (is_mode || ms_mode || tns_mode || pred_mode) {
866  for (i = 0; i < s->chan_map[0]; i++) {
867  // Must restore coeffs
868  chans = tag == TYPE_CPE ? 2 : 1;
869  cpe = &s->cpe[i];
870  for (ch = 0; ch < chans; ch++)
871  memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
872  }
873  }
874  its++;
875  }
876  } else {
877  break;
878  }
879  } while (1);
880 
881  if (s->options.ltp && s->coder->ltp_insert_new_frame)
883 
884  put_bits(&s->pb, 3, TYPE_END);
885  flush_put_bits(&s->pb);
886 
888 
889  s->lambda_sum += s->lambda;
890  s->lambda_count++;
891 
892  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
893  &avpkt->duration);
894 
895  avpkt->size = put_bits_count(&s->pb) >> 3;
896  *got_packet_ptr = 1;
897  return 0;
898 }
899 
901 {
902  AACEncContext *s = avctx->priv_data;
903 
904  av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_sum / s->lambda_count);
905 
906  ff_mdct_end(&s->mdct1024);
907  ff_mdct_end(&s->mdct128);
908  ff_psy_end(&s->psy);
909  ff_lpc_end(&s->lpc);
910  if (s->psypp)
912  av_freep(&s->buffer.samples);
913  av_freep(&s->cpe);
914  av_freep(&s->fdsp);
915  ff_af_queue_close(&s->afq);
916  return 0;
917 }
918 
920 {
921  int ret = 0;
922 
924  if (!s->fdsp)
925  return AVERROR(ENOMEM);
926 
927  // window init
932 
933  if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
934  return ret;
935  if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0)
936  return ret;
937 
938  return 0;
939 }
940 
942 {
943  int ch;
944  if (!FF_ALLOCZ_TYPED_ARRAY(s->buffer.samples, s->channels * 3 * 1024) ||
945  !FF_ALLOCZ_TYPED_ARRAY(s->cpe, s->chan_map[0]))
946  return AVERROR(ENOMEM);
947 
948  for(ch = 0; ch < s->channels; ch++)
949  s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
950 
951  return 0;
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  char buf[64];
986  for (i = 0; i < FF_ARRAY_ELEMS(aac_pce_configs); i++)
987  if (avctx->channel_layout == aac_pce_configs[i].layout)
988  break;
989  av_get_channel_layout_string(buf, sizeof(buf), -1, avctx->channel_layout);
990  ERROR_IF(i == FF_ARRAY_ELEMS(aac_pce_configs), "Unsupported channel layout \"%s\"\n", buf);
991  av_log(avctx, AV_LOG_INFO, "Using a PCE to encode channel layout \"%s\"\n", buf);
992  s->pce = aac_pce_configs[i];
993  s->reorder_map = s->pce.reorder_map;
994  s->chan_map = s->pce.config_map;
995  } else {
996  s->reorder_map = aac_chan_maps[s->channels - 1];
997  s->chan_map = aac_chan_configs[s->channels - 1];
998  }
999 
1000  if (!avctx->bit_rate) {
1001  for (i = 1; i <= s->chan_map[0]; i++) {
1002  avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */
1003  s->chan_map[i] == TYPE_LFE ? 16000 : /* LFE */
1004  69000 ; /* SCE */
1005  }
1006  }
1007 
1008  /* Samplerate */
1009  for (i = 0; i < 16; i++)
1011  break;
1012  s->samplerate_index = i;
1013  ERROR_IF(s->samplerate_index == 16 ||
1016  "Unsupported sample rate %d\n", avctx->sample_rate);
1017 
1018  /* Bitrate limiting */
1019  WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
1020  "Too many bits %f > %d per frame requested, clamping to max\n",
1021  1024.0 * avctx->bit_rate / avctx->sample_rate,
1022  6144 * s->channels);
1023  avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate,
1024  avctx->bit_rate);
1025 
1026  /* Profile and option setting */
1027  avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
1028  avctx->profile;
1029  for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++)
1030  if (avctx->profile == aacenc_profiles[i])
1031  break;
1032  if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) {
1033  avctx->profile = FF_PROFILE_AAC_LOW;
1034  ERROR_IF(s->options.pred,
1035  "Main prediction unavailable in the \"mpeg2_aac_low\" profile\n");
1036  ERROR_IF(s->options.ltp,
1037  "LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n");
1038  WARN_IF(s->options.pns,
1039  "PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n");
1040  s->options.pns = 0;
1041  } else if (avctx->profile == FF_PROFILE_AAC_LTP) {
1042  s->options.ltp = 1;
1043  ERROR_IF(s->options.pred,
1044  "Main prediction unavailable in the \"aac_ltp\" profile\n");
1045  } else if (avctx->profile == FF_PROFILE_AAC_MAIN) {
1046  s->options.pred = 1;
1047  ERROR_IF(s->options.ltp,
1048  "LTP prediction unavailable in the \"aac_main\" profile\n");
1049  } else if (s->options.ltp) {
1050  avctx->profile = FF_PROFILE_AAC_LTP;
1051  WARN_IF(1,
1052  "Chainging profile to \"aac_ltp\"\n");
1053  ERROR_IF(s->options.pred,
1054  "Main prediction unavailable in the \"aac_ltp\" profile\n");
1055  } else if (s->options.pred) {
1056  avctx->profile = FF_PROFILE_AAC_MAIN;
1057  WARN_IF(1,
1058  "Chainging profile to \"aac_main\"\n");
1059  ERROR_IF(s->options.ltp,
1060  "LTP prediction unavailable in the \"aac_main\" profile\n");
1061  }
1062  s->profile = avctx->profile;
1063 
1064  /* Coder limitations */
1065  s->coder = &ff_aac_coders[s->options.coder];
1066  if (s->options.coder == AAC_CODER_ANMR) {
1068  "The ANMR coder is considered experimental, add -strict -2 to enable!\n");
1069  s->options.intensity_stereo = 0;
1070  s->options.pns = 0;
1071  }
1073  "The LPT profile requires experimental compliance, add -strict -2 to enable!\n");
1074 
1075  /* M/S introduces horrible artifacts with multichannel files, this is temporary */
1076  if (s->channels > 3)
1077  s->options.mid_side = 0;
1078 
1079  if ((ret = dsp_init(avctx, s)) < 0)
1080  return ret;
1081 
1082  if ((ret = alloc_buffers(avctx, s)) < 0)
1083  return ret;
1084 
1085  if ((ret = put_audio_specific_config(avctx)))
1086  return ret;
1087 
1088  sizes[0] = ff_aac_swb_size_1024[s->samplerate_index];
1089  sizes[1] = ff_aac_swb_size_128[s->samplerate_index];
1090  lengths[0] = ff_aac_num_swb_1024[s->samplerate_index];
1091  lengths[1] = ff_aac_num_swb_128[s->samplerate_index];
1092  for (i = 0; i < s->chan_map[0]; i++)
1093  grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
1094  if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
1095  s->chan_map[0], grouping)) < 0)
1096  return ret;
1097  s->psypp = ff_psy_preprocess_init(avctx);
1099  s->random_state = 0x1f2e3d4c;
1100 
1101  s->abs_pow34 = abs_pow34_v;
1103 
1104  if (ARCH_X86)
1106 
1107  if (HAVE_MIPSDSP)
1109 
1111  return AVERROR_UNKNOWN;
1112 
1113  ff_af_queue_init(avctx, &s->afq);
1114 
1115  return 0;
1116 }
1117 
1118 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1119 static const AVOption aacenc_options[] = {
1120  {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_FAST}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
1121  {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1122  {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1123  {"fast", "Default fast search", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1124  {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
1125  {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1126  {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1127  {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1128  {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1129  {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1130  {"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,
1160  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
1162  .priv_class = &aacenc_class,
1163 };
#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:48
struct FFPsyContext::@114 bitres
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
Definition: avcodec.h:1594
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:127
#define NULL
Definition: coverity.c:32
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:308
int grouping[8]
window grouping (for e.g. AAC)
Definition: psymodel.h:81
void(* search_for_ms)(struct AACEncContext *s, ChannelElement *cpe)
Definition: aacenc.h:77
AVOption.
Definition: opt.h:248
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
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:208
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:576
#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:115
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
Definition: aac.h:57
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: packet.h:356
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
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:222
void(* search_for_tns)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:75
int common_window
Set if channels share a common &#39;IndividualChannelStream&#39; 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:1863
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:237
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:1859
AVCodec.
Definition: codec.h:190
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
Encode spectral coefficients processed by psychoacoustic model.
Definition: aacenc.c:440
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:52
static int put_audio_specific_config(AVCodecContext *avctx)
Make AAC audio config object.
Definition: aacenc.c:95
struct AACEncContext::@6 buffer
void(* apply_tns_filt)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:69
const uint8_t ff_aac_num_swb_128[]
Definition: aactab.c:61
void(* search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s, SingleChannelElement *sce, const float lambda)
Definition: aacenc.h:57
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: codec.h:75
#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:33
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:88
void(* search_for_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce)
Definition: aacenc.h:73
AVOptions.
int intensity_stereo
Definition: aacenc.h:51
#define WINDOW_FUNC(type)
Definition: aacenc.c:136
void(* update_ltp)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:70
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
#define f(width, name)
Definition: cbs_vp9.c:255
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: packet.h:373
const uint8_t * chan_map
channel configuration map
Definition: aacenc.h:392
TemporalNoiseShaping tns
Definition: aac.h:250
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce output
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:627
AudioFrameQueue afq
Definition: aacenc.h:406
const AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB]
Definition: aaccoder.c:897
static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS]
Table to remap channels from libavcodec&#39;s default order to AAC order.
Definition: aacenctab.h:72
#define FF_PROFILE_AAC_LTP
Definition: avcodec.h:1866
uint8_t * data
Definition: packet.h:355
static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
Definition: aacenc.c:54
const uint8_t * ff_aac_swb_size_128[]
Definition: aacenctab.c:91
uint32_t tag
Definition: movenc.c:1532
Scalefactor data are intensity stereo positions (in phase).
Definition: aac.h:89
#define max(a, b)
Definition: cuda_runtime.h:33
int profile
copied from avctx
Definition: aacenc.h:386
channels
Definition: aptx.h:33
#define AVOnce
Definition: thread.h:172
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:257
#define av_log(a,...)
static const AVOption aacenc_options[]
Definition: aacenc.c:1119
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:194
#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:135
void(* encode_ltp_info)(struct AACEncContext *s, SingleChannelElement *sce, int common_window)
Definition: aacenc.h:64
static const int sizes[][2]
Definition: img2dec.c:53
const uint8_t ff_aac_num_swb_1024[]
Definition: aactab.c:45
#define FF_PROFILE_MPEG2_AAC_LOW
Definition: avcodec.h:1871
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:153
#define FF_AAC_PROFILE_OPTS
Definition: profiles.h:28
float is_ener[128]
Intensity stereo pos (used by encoder)
Definition: aac.h:259
int initial_padding
Audio only.
Definition: avcodec.h:2060
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:606
void(* mdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:109
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: codec.h:197
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:535
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:175
static const AVCodecDefault defaults[]
Definition: amfenc_h264.c:361
void(* adjust_common_ltp)(struct AACEncContext *s, ChannelElement *cpe)
Definition: aacenc.h:67
#define ff_mdct_init
Definition: fft.h:169
Definition: aac.h:62
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
void(* mark_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce)
Definition: aacenc.h:74
int index[4][8]
front, side, back, lfe
Definition: aacenc.h:97
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1237
static int put_bits_count(PutBitContext *s)
Definition: put_bits.h:67
#define AACENC_FLAGS
Definition: aacenc.c:1118
INTFLOAT ret_buf[2048]
PCM output buffer.
Definition: aac.h:264
void(* set_special_band_scalefactors)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:72
MIPS optimizations info
Definition: mips.txt:2
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:333
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:275
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: codec.h:80
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:307
static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: aacenc.c:557
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
uint8_t w
Definition: llviddspenc.c:38
typedef void(APIENTRY *FF_PFNGLACTIVETEXTUREPROC)(GLenum texture)
#define FF_PROFILE_AAC_LOW
Definition: avcodec.h:1864
static const AVCodecDefault aac_encode_defaults[]
Definition: aacenc.c:1142
#define FF_PROFILE_UNKNOWN
Definition: avcodec.h:1860
int pos[4]
Definition: aac.h:227
int channels
channel count
Definition: aacenc.h:390
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
#define s(width, name)
Definition: cbs_vp9.c:257
AAC definitions and structures.
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:192
#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)
if(ret)
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.
static av_cold int aac_encode_end(AVCodecContext *avctx)
Definition: aacenc.c:900
void(* search_for_is)(struct AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe)
Definition: aacenc.h:78
void ff_aac_dsp_init_x86(AACEncContext *s)
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1206
void(* search_for_ltp)(struct AACEncContext *s, SingleChannelElement *sce, int common_window)
Definition: aacenc.h:76
#define AV_LOG_INFO
Standard information.
Definition: log.h:205
#define AV_ONCE_INIT
Definition: thread.h:173
#define CLIP_AVOIDANCE_FACTOR
Definition: aac.h:53
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
Temporal Noise Shaping.
Definition: aac.h:198
int sample_rate
samples per second
Definition: avcodec.h:1186
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:243
void(* ltp_insert_new_frame)(struct AACEncContext *s)
Definition: aacenc.h:71
void(* search_for_pred)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:79
main external API structure.
Definition: avcodec.h:526
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
void(* apply_main_pred)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:68
IndividualChannelStream ics
Definition: aac.h:249
int extradata_size
Definition: avcodec.h:628
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
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
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:514
const int ff_aac_swb_size_128_len
Definition: aacenctab.c:107
void(* encode_main_pred)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:65
void(* adjust_common_pred)(struct AACEncContext *s, ChannelElement *cpe)
Definition: aacenc.h:66
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:421
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:941
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
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0f/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(UINT64_C(1)<< 63)))#define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_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),};static void cpy1(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, len);}static void cpy2(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 2 *len);}static void cpy4(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 4 *len);}static void cpy8(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags){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) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;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)){case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8: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);return ctx;}void swri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len){int ch;int off=0;const int os=(out->planar?1:out->ch_count)*out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){int planes=in->planar?in->ch_count:1;unsigned m=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){int planes=out->planar?out->ch_count:1;unsigned m=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){int planes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
FFPsyWindowInfo(* window)(FFPsyContext *ctx, const float *audio, const float *la, int channel, int prev_type)
Suggest window sequence for channel.
Definition: psymodel.h:129
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
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:592
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
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
Scalefactor data are intensity stereo positions (out of phase).
Definition: aac.h:88
Y Spectral Band Replication.
Definition: mpeg4audio.h:94
const OptionDef options[]
Definition: ffmpeg_opt.c:3391
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
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&#39;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:385
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
#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:553
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)
int channels
number of audio channels
Definition: avcodec.h:1187
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:175
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
Encode scalefactor band coding type.
Definition: aacenc.c:371
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
static void apply_mid_side_stereo(ChannelElement *cpe)
Definition: aacenc.c:336
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:723
int frame_number
Frame counter, set by libavcodec.
Definition: avcodec.h:1217
FILE * out
Definition: movenc.c:54
Filter the word “frame” indicates either a video frame or a group of audio samples
#define av_freep(p)
void(* encode_tns_info)(struct AACEncContext *s, SingleChannelElement *sce)
Definition: aacenc.h:63
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, int common_window)
Encode one channel of audio data.
Definition: aacenc.c:488
#define FF_ALLOCZ_TYPED_ARRAY(p, nelem)
Definition: internal.h:141
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:201
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
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
static const int aacenc_profiles[]
Definition: aacenctab.h:132
void(* abs_pow34)(float *out, const float *in, const int size)
Definition: aacenc.h:413
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:355
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: packet.h:332
static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
Downscale spectral coefficients for near-clipping windows to avoid artifacts.
Definition: aacenc.c:468
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:374
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:1589
static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
Definition: aacenc.c:919
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:348
for(j=16;j >0;--j)
int pred
Definition: aacenc.h:49
int i
Definition: input.c:406
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
const char * name
Definition: opengl_enc.c:102
void(* encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce, int win, int group_len, const float lambda)
Definition: aacenc.h:59
bitstream writer API