aacenc.c
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1 /*
2  * AAC encoder
3  * Copyright (C) 2008 Konstantin Shishkov
4  *
5  * This file is part of Libav.
6  *
7  * Libav is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * Libav is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with Libav; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
27 /***********************************
28  * TODOs:
29  * add sane pulse detection
30  * add temporal noise shaping
31  ***********************************/
32 
33 #include "libavutil/opt.h"
34 #include "avcodec.h"
35 #include "put_bits.h"
36 #include "dsputil.h"
37 #include "mpeg4audio.h"
38 #include "kbdwin.h"
39 #include "sinewin.h"
40 
41 #include "aac.h"
42 #include "aactab.h"
43 #include "aacenc.h"
44 
45 #include "psymodel.h"
46 
47 #define AAC_MAX_CHANNELS 6
48 
49 static const uint8_t swb_size_1024_96[] = {
50  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
51  12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
52  64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
53 };
54 
55 static const uint8_t swb_size_1024_64[] = {
56  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
57  12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
58  40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
59 };
60 
61 static const uint8_t swb_size_1024_48[] = {
62  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
63  12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
64  32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
65  96
66 };
67 
68 static const uint8_t swb_size_1024_32[] = {
69  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
70  12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
71  32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
72 };
73 
74 static const uint8_t swb_size_1024_24[] = {
75  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
76  12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
77  32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
78 };
79 
80 static const uint8_t swb_size_1024_16[] = {
81  8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
82  12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
83  32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
84 };
85 
86 static const uint8_t swb_size_1024_8[] = {
87  12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
88  16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
89  32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
90 };
91 
92 static const uint8_t *swb_size_1024[] = {
97 };
98 
99 static const uint8_t swb_size_128_96[] = {
100  4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
101 };
102 
103 static const uint8_t swb_size_128_48[] = {
104  4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
105 };
106 
107 static const uint8_t swb_size_128_24[] = {
108  4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
109 };
110 
111 static const uint8_t swb_size_128_16[] = {
112  4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
113 };
114 
115 static const uint8_t swb_size_128_8[] = {
116  4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
117 };
118 
119 static const uint8_t *swb_size_128[] = {
120  /* the last entry on the following row is swb_size_128_64 but is a
121  duplicate of swb_size_128_96 */
126 };
127 
129 static const uint8_t aac_chan_configs[6][5] = {
130  {1, TYPE_SCE}, // 1 channel - single channel element
131  {1, TYPE_CPE}, // 2 channels - channel pair
132  {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
133  {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
134  {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
135  {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
136 };
137 
143 {
144  PutBitContext pb;
145  AACEncContext *s = avctx->priv_data;
146 
147  init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
148  put_bits(&pb, 5, 2); //object type - AAC-LC
149  put_bits(&pb, 4, s->samplerate_index); //sample rate index
150  put_bits(&pb, 4, avctx->channels);
151  //GASpecificConfig
152  put_bits(&pb, 1, 0); //frame length - 1024 samples
153  put_bits(&pb, 1, 0); //does not depend on core coder
154  put_bits(&pb, 1, 0); //is not extension
155 
156  //Explicitly Mark SBR absent
157  put_bits(&pb, 11, 0x2b7); //sync extension
158  put_bits(&pb, 5, AOT_SBR);
159  put_bits(&pb, 1, 0);
160  flush_put_bits(&pb);
161 }
162 
164 {
165  AACEncContext *s = avctx->priv_data;
166  int i;
167  const uint8_t *sizes[2];
168  uint8_t grouping[AAC_MAX_CHANNELS];
169  int lengths[2];
170 
171  avctx->frame_size = 1024;
172 
173  for (i = 0; i < 16; i++)
175  break;
176  if (i == 16) {
177  av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
178  return -1;
179  }
180  if (avctx->channels > AAC_MAX_CHANNELS) {
181  av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
182  return -1;
183  }
184  if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) {
185  av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile);
186  return -1;
187  }
188  if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) {
189  av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n");
190  return -1;
191  }
192  s->samplerate_index = i;
193 
194  dsputil_init(&s->dsp, avctx);
195  ff_mdct_init(&s->mdct1024, 11, 0, 1.0);
196  ff_mdct_init(&s->mdct128, 8, 0, 1.0);
197  // window init
202 
203  s->chan_map = aac_chan_configs[avctx->channels-1];
204  s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
205  s->cpe = av_mallocz(sizeof(ChannelElement) * s->chan_map[0]);
207  avctx->extradata_size = 5;
209 
210  sizes[0] = swb_size_1024[i];
211  sizes[1] = swb_size_128[i];
212  lengths[0] = ff_aac_num_swb_1024[i];
213  lengths[1] = ff_aac_num_swb_128[i];
214  for (i = 0; i < s->chan_map[0]; i++)
215  grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
216  ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping);
217  s->psypp = ff_psy_preprocess_init(avctx);
218  s->coder = &ff_aac_coders[2];
219 
220  s->lambda = avctx->global_quality ? avctx->global_quality : 120;
221 
223 
224  return 0;
225 }
226 
228  SingleChannelElement *sce, short *audio)
229 {
230  int i, k;
231  const int chans = avctx->channels;
232  const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
233  const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
234  const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
235  float *output = sce->ret;
236 
237  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
238  memcpy(output, sce->saved, sizeof(float)*1024);
239  if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
240  memset(output, 0, sizeof(output[0]) * 448);
241  for (i = 448; i < 576; i++)
242  output[i] = sce->saved[i] * pwindow[i - 448];
243  for (i = 576; i < 704; i++)
244  output[i] = sce->saved[i];
245  }
246  if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
247  for (i = 0; i < 1024; i++) {
248  output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1];
249  sce->saved[i] = audio[i * chans] * lwindow[i];
250  }
251  } else {
252  for (i = 0; i < 448; i++)
253  output[i+1024] = audio[i * chans];
254  for (; i < 576; i++)
255  output[i+1024] = audio[i * chans] * swindow[576 - i - 1];
256  memset(output+1024+576, 0, sizeof(output[0]) * 448);
257  for (i = 0; i < 1024; i++)
258  sce->saved[i] = audio[i * chans];
259  }
260  s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
261  } else {
262  for (k = 0; k < 1024; k += 128) {
263  for (i = 448 + k; i < 448 + k + 256; i++)
264  output[i - 448 - k] = (i < 1024)
265  ? sce->saved[i]
266  : audio[(i-1024)*chans];
267  s->dsp.vector_fmul (output, output, k ? swindow : pwindow, 128);
268  s->dsp.vector_fmul_reverse(output+128, output+128, swindow, 128);
269  s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + k, output);
270  }
271  for (i = 0; i < 1024; i++)
272  sce->saved[i] = audio[i * chans];
273  }
274 }
275 
281 {
282  int w;
283 
284  put_bits(&s->pb, 1, 0); // ics_reserved bit
285  put_bits(&s->pb, 2, info->window_sequence[0]);
286  put_bits(&s->pb, 1, info->use_kb_window[0]);
287  if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
288  put_bits(&s->pb, 6, info->max_sfb);
289  put_bits(&s->pb, 1, 0); // no prediction
290  } else {
291  put_bits(&s->pb, 4, info->max_sfb);
292  for (w = 1; w < 8; w++)
293  put_bits(&s->pb, 1, !info->group_len[w]);
294  }
295 }
296 
302 {
303  int i, w;
304 
305  put_bits(pb, 2, cpe->ms_mode);
306  if (cpe->ms_mode == 1)
307  for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
308  for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
309  put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
310 }
311 
315 static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
316 {
317  int i, w, w2, g, ch;
318  int start, maxsfb, cmaxsfb;
319 
320  for (ch = 0; ch < chans; ch++) {
321  IndividualChannelStream *ics = &cpe->ch[ch].ics;
322  start = 0;
323  maxsfb = 0;
324  cpe->ch[ch].pulse.num_pulse = 0;
325  for (w = 0; w < ics->num_windows*16; w += 16) {
326  for (g = 0; g < ics->num_swb; g++) {
327  //apply M/S
328  if (cpe->common_window && !ch && cpe->ms_mask[w + g]) {
329  for (i = 0; i < ics->swb_sizes[g]; i++) {
330  cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
331  cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
332  }
333  }
334  start += ics->swb_sizes[g];
335  }
336  for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
337  ;
338  maxsfb = FFMAX(maxsfb, cmaxsfb);
339  }
340  ics->max_sfb = maxsfb;
341 
342  //adjust zero bands for window groups
343  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
344  for (g = 0; g < ics->max_sfb; g++) {
345  i = 1;
346  for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
347  if (!cpe->ch[ch].zeroes[w2*16 + g]) {
348  i = 0;
349  break;
350  }
351  }
352  cpe->ch[ch].zeroes[w*16 + g] = i;
353  }
354  }
355  }
356 
357  if (chans > 1 && cpe->common_window) {
358  IndividualChannelStream *ics0 = &cpe->ch[0].ics;
359  IndividualChannelStream *ics1 = &cpe->ch[1].ics;
360  int msc = 0;
361  ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
362  ics1->max_sfb = ics0->max_sfb;
363  for (w = 0; w < ics0->num_windows*16; w += 16)
364  for (i = 0; i < ics0->max_sfb; i++)
365  if (cpe->ms_mask[w+i])
366  msc++;
367  if (msc == 0 || ics0->max_sfb == 0)
368  cpe->ms_mode = 0;
369  else
370  cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
371  }
372 }
373 
378 {
379  int w;
380 
381  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
382  s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
383 }
384 
390 {
391  int off = sce->sf_idx[0], diff;
392  int i, w;
393 
394  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
395  for (i = 0; i < sce->ics.max_sfb; i++) {
396  if (!sce->zeroes[w*16 + i]) {
397  diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
398  if (diff < 0 || diff > 120)
399  av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
400  off = sce->sf_idx[w*16 + i];
402  }
403  }
404  }
405 }
406 
410 static void encode_pulses(AACEncContext *s, Pulse *pulse)
411 {
412  int i;
413 
414  put_bits(&s->pb, 1, !!pulse->num_pulse);
415  if (!pulse->num_pulse)
416  return;
417 
418  put_bits(&s->pb, 2, pulse->num_pulse - 1);
419  put_bits(&s->pb, 6, pulse->start);
420  for (i = 0; i < pulse->num_pulse; i++) {
421  put_bits(&s->pb, 5, pulse->pos[i]);
422  put_bits(&s->pb, 4, pulse->amp[i]);
423  }
424 }
425 
430 {
431  int start, i, w, w2;
432 
433  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
434  start = 0;
435  for (i = 0; i < sce->ics.max_sfb; i++) {
436  if (sce->zeroes[w*16 + i]) {
437  start += sce->ics.swb_sizes[i];
438  continue;
439  }
440  for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
441  s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
442  sce->ics.swb_sizes[i],
443  sce->sf_idx[w*16 + i],
444  sce->band_type[w*16 + i],
445  s->lambda);
446  start += sce->ics.swb_sizes[i];
447  }
448  }
449 }
450 
456  int common_window)
457 {
458  put_bits(&s->pb, 8, sce->sf_idx[0]);
459  if (!common_window)
460  put_ics_info(s, &sce->ics);
461  encode_band_info(s, sce);
462  encode_scale_factors(avctx, s, sce);
463  encode_pulses(s, &sce->pulse);
464  put_bits(&s->pb, 1, 0); //tns
465  put_bits(&s->pb, 1, 0); //ssr
466  encode_spectral_coeffs(s, sce);
467  return 0;
468 }
469 
474  const char *name)
475 {
476  int i, namelen, padbits;
477 
478  namelen = strlen(name) + 2;
479  put_bits(&s->pb, 3, TYPE_FIL);
480  put_bits(&s->pb, 4, FFMIN(namelen, 15));
481  if (namelen >= 15)
482  put_bits(&s->pb, 8, namelen - 16);
483  put_bits(&s->pb, 4, 0); //extension type - filler
484  padbits = 8 - (put_bits_count(&s->pb) & 7);
486  for (i = 0; i < namelen - 2; i++)
487  put_bits(&s->pb, 8, name[i]);
488  put_bits(&s->pb, 12 - padbits, 0);
489 }
490 
492  uint8_t *frame, int buf_size, void *data)
493 {
494  AACEncContext *s = avctx->priv_data;
495  int16_t *samples = s->samples, *samples2, *la;
496  ChannelElement *cpe;
497  int i, ch, w, g, chans, tag, start_ch;
498  int chan_el_counter[4];
500 
501  if (s->last_frame)
502  return 0;
503  if (data) {
504  if (!s->psypp) {
505  memcpy(s->samples + 1024 * avctx->channels, data,
506  1024 * avctx->channels * sizeof(s->samples[0]));
507  } else {
508  start_ch = 0;
509  samples2 = s->samples + 1024 * avctx->channels;
510  for (i = 0; i < s->chan_map[0]; i++) {
511  tag = s->chan_map[i+1];
512  chans = tag == TYPE_CPE ? 2 : 1;
513  ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
514  samples2 + start_ch, start_ch, chans);
515  start_ch += chans;
516  }
517  }
518  }
519  if (!avctx->frame_number) {
520  memcpy(s->samples, s->samples + 1024 * avctx->channels,
521  1024 * avctx->channels * sizeof(s->samples[0]));
522  return 0;
523  }
524 
525  start_ch = 0;
526  for (i = 0; i < s->chan_map[0]; i++) {
527  FFPsyWindowInfo* wi = windows + start_ch;
528  tag = s->chan_map[i+1];
529  chans = tag == TYPE_CPE ? 2 : 1;
530  cpe = &s->cpe[i];
531  for (ch = 0; ch < chans; ch++) {
532  IndividualChannelStream *ics = &cpe->ch[ch].ics;
533  int cur_channel = start_ch + ch;
534  samples2 = samples + cur_channel;
535  la = samples2 + (448+64) * avctx->channels;
536  if (!data)
537  la = NULL;
538  if (tag == TYPE_LFE) {
539  wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
540  wi[ch].window_shape = 0;
541  wi[ch].num_windows = 1;
542  wi[ch].grouping[0] = 1;
543 
544  /* Only the lowest 12 coefficients are used in a LFE channel.
545  * The expression below results in only the bottom 8 coefficients
546  * being used for 11.025kHz to 16kHz sample rates.
547  */
548  ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
549  } else {
550  wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
551  ics->window_sequence[0]);
552  }
553  ics->window_sequence[1] = ics->window_sequence[0];
554  ics->window_sequence[0] = wi[ch].window_type[0];
555  ics->use_kb_window[1] = ics->use_kb_window[0];
556  ics->use_kb_window[0] = wi[ch].window_shape;
557  ics->num_windows = wi[ch].num_windows;
558  ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
559  ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
560  for (w = 0; w < ics->num_windows; w++)
561  ics->group_len[w] = wi[ch].grouping[w];
562 
563  apply_window_and_mdct(avctx, s, &cpe->ch[ch], samples2);
564  }
565  start_ch += chans;
566  }
567  do {
568  int frame_bits;
569  init_put_bits(&s->pb, frame, buf_size*8);
570  if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
572  start_ch = 0;
573  memset(chan_el_counter, 0, sizeof(chan_el_counter));
574  for (i = 0; i < s->chan_map[0]; i++) {
575  FFPsyWindowInfo* wi = windows + start_ch;
576  const float *coeffs[2];
577  tag = s->chan_map[i+1];
578  chans = tag == TYPE_CPE ? 2 : 1;
579  cpe = &s->cpe[i];
580  put_bits(&s->pb, 3, tag);
581  put_bits(&s->pb, 4, chan_el_counter[tag]++);
582  for (ch = 0; ch < chans; ch++)
583  coeffs[ch] = cpe->ch[ch].coeffs;
584  s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
585  for (ch = 0; ch < chans; ch++) {
586  s->cur_channel = start_ch * 2 + ch;
587  s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
588  }
589  cpe->common_window = 0;
590  if (chans > 1
591  && wi[0].window_type[0] == wi[1].window_type[0]
592  && wi[0].window_shape == wi[1].window_shape) {
593 
594  cpe->common_window = 1;
595  for (w = 0; w < wi[0].num_windows; w++) {
596  if (wi[0].grouping[w] != wi[1].grouping[w]) {
597  cpe->common_window = 0;
598  break;
599  }
600  }
601  }
602  s->cur_channel = start_ch * 2;
603  if (s->options.stereo_mode && cpe->common_window) {
604  if (s->options.stereo_mode > 0) {
605  IndividualChannelStream *ics = &cpe->ch[0].ics;
606  for (w = 0; w < ics->num_windows; w += ics->group_len[w])
607  for (g = 0; g < ics->num_swb; g++)
608  cpe->ms_mask[w*16+g] = 1;
609  } else if (s->coder->search_for_ms) {
610  s->coder->search_for_ms(s, cpe, s->lambda);
611  }
612  }
613  adjust_frame_information(s, cpe, chans);
614  if (chans == 2) {
615  put_bits(&s->pb, 1, cpe->common_window);
616  if (cpe->common_window) {
617  put_ics_info(s, &cpe->ch[0].ics);
618  encode_ms_info(&s->pb, cpe);
619  }
620  }
621  for (ch = 0; ch < chans; ch++) {
622  s->cur_channel = start_ch + ch;
623  encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
624  }
625  start_ch += chans;
626  }
627 
628  frame_bits = put_bits_count(&s->pb);
629  if (frame_bits <= 6144 * avctx->channels - 3) {
630  s->psy.bitres.bits = frame_bits / avctx->channels;
631  break;
632  }
633 
634  s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
635 
636  } while (1);
637 
638  put_bits(&s->pb, 3, TYPE_END);
639  flush_put_bits(&s->pb);
640  avctx->frame_bits = put_bits_count(&s->pb);
641 
642  // rate control stuff
643  if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
644  float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
645  s->lambda *= ratio;
646  s->lambda = FFMIN(s->lambda, 65536.f);
647  }
648 
649  if (!data)
650  s->last_frame = 1;
651  memcpy(s->samples, s->samples + 1024 * avctx->channels,
652  1024 * avctx->channels * sizeof(s->samples[0]));
653  return put_bits_count(&s->pb)>>3;
654 }
655 
657 {
658  AACEncContext *s = avctx->priv_data;
659 
660  ff_mdct_end(&s->mdct1024);
661  ff_mdct_end(&s->mdct128);
662  ff_psy_end(&s->psy);
664  av_freep(&s->samples);
665  av_freep(&s->cpe);
666  return 0;
667 }
668 
669 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
670 static const AVOption aacenc_options[] = {
671  {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.dbl = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
672  {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.dbl = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
673  {"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.dbl = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
674  {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.dbl = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
675  {NULL}
676 };
677 
678 static const AVClass aacenc_class = {
679  "AAC encoder",
683 };
684 
686  .name = "aac",
687  .type = AVMEDIA_TYPE_AUDIO,
688  .id = CODEC_ID_AAC,
689  .priv_data_size = sizeof(AACEncContext),
691  .encode = aac_encode_frame,
694  .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
695  .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
696  .priv_class = &aacenc_class,
697 };
void * av_mallocz(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:154
static const int16_t coeffs[28]
static const uint8_t aac_chan_configs[6][5]
default channel configurations
Definition: aacenc.c:129
void(* search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda)
Definition: aacenc.h:46
uint8_t use_kb_window[2]
If set, use Kaiser-Bessel window, otherwise use a sinus window.
Definition: aac.h:152
static const uint8_t swb_size_1024_64[]
Definition: aacenc.c:55
static short * samples
Definition: ffmpeg.c:233
int grouping[8]
window grouping (for e.g. AAC)
Definition: psymodel.h:67
AVOption.
Definition: opt.h:244
uint8_t ** bands
scalefactor band sizes for possible frame sizes
Definition: psymodel.h:82
Definition: aac.h:195
void(* mdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:83
static const AVClass aacenc_class
Definition: aacenc.c:678
av_cold void ff_kbd_window_init(float *window, float alpha, int n)
Generate a Kaiser-Bessel Derived Window.
Definition: kbdwin.c:26
#define SCALE_DIFF_ZERO
codebook index corresponding to zero scalefactor indices difference
Definition: aac.h:133
Definition: aac.h:56
Definition: aac.h:49
Definition: aac.h:50
av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
Cleanup audio preprocessing module.
Definition: psymodel.c:132
AVOptions.
AACCoefficientsEncoder * coder
Definition: aacenc.h:69
void avpriv_align_put_bits(PutBitContext *s)
Pad the bitstream with zeros up to the next byte boundary.
Definition: bitstream.c:44
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
Encode ics_info element.
Definition: aacenc.c:280
#define AAC_MAX_CHANNELS
Definition: aacenc.c:47
int common_window
Set if channels share a common 'IndividualChannelStream' in bitstream.
Definition: aac.h:240
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:28
uint8_t ms_mask[128]
Set if mid/side stereo is used for each scalefactor window band.
Definition: aac.h:242
static const uint8_t swb_size_128_8[]
Definition: aacenc.c:115
float lambda
Definition: aacenc.h:72
static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, short *audio)
Definition: aacenc.c:227
signed 16 bits
Definition: samplefmt.h:30
int profile
profile
Definition: avcodec.h:2462
AVCodec.
Definition: avcodec.h:3189
static const uint8_t swb_size_1024_8[]
Definition: aacenc.c:86
static const uint8_t swb_size_128_96[]
Definition: aacenc.c:99
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
Encode spectral coefficients processed by psychoacoustic model.
Definition: aacenc.c:429
int * num_bands
number of scalefactor bands for possible frame sizes
Definition: psymodel.h:83
void av_freep(void *arg)
Free a memory block which has been allocated with av_malloc(z)() or av_realloc() and set the pointer ...
Definition: mem.c:147
const uint8_t ff_aac_num_swb_128[]
Definition: aactab.c:43
int16_t * samples
saved preprocessed input
Definition: aacenc.h:61
AACEncOptions options
encoding options
Definition: aacenc.h:56
AAC encoder context.
Definition: aacenc.h:54
#define av_cold
Definition: attributes.h:71
void(* quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size, int scale_idx, int cb, const float lambda)
Definition: aacenc.h:44
SingleChannelElement ch[2]
Definition: aac.h:244
int samplerate_index
MPEG-4 samplerate index.
Definition: aacenc.h:63
Definition: aac.h:52
const uint8_t * chan_map
channel configuration map
Definition: aacenc.h:64
const uint8_t ff_aac_scalefactor_bits[121]
Definition: aactab.c:70
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1387
const char data[16]
Definition: mxf.c:60
static const uint8_t swb_size_1024_48[]
Definition: aacenc.c:61
uint32_t tag
Definition: movenc.c:670
float saved[1024]
overlap
Definition: aac.h:229
static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
Produce integer coefficients from scalefactors provided by the model.
Definition: aacenc.c:315
av_cold void dsputil_init(DSPContext *c, AVCodecContext *avctx)
Definition: dsputil.c:2789
static int init(AVCodecParserContext *s)
Definition: h264_parser.c:336
#define CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: avcodec.h:755
static const AVOption aacenc_options[]
Definition: aacenc.c:670
static const uint8_t swb_size_1024_24[]
Definition: aacenc.c:74
float coeffs[1024]
coefficients for IMDCT
Definition: aac.h:228
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, const int16_t *audio, int16_t *dest, int tag, int channels)
Preprocess several channel in audio frame in order to compress it better.
Definition: psymodel.c:115
const uint8_t ff_aac_num_swb_1024[]
Definition: aactab.c:39
int last_frame
Definition: aacenc.h:71
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:191
int stereo_mode
Definition: aacenc.h:34
g
Definition: yuv2rgb.c:481
FFPsyWindowInfo(* window)(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type)
Suggest window sequence for channel.
Definition: psymodel.h:112
float ff_aac_kbd_long_1024[1024]
Definition: aactab.c:36
int flags
CODEC_FLAG_*.
Definition: avcodec.h:1355
int amp[4]
Definition: aac.h:199
void av_log(void *avcl, int level, const char *fmt,...)
Definition: log.c:140
const char * name
Name of the codec implementation.
Definition: avcodec.h:3196
int num_windows
number of windows in a frame
Definition: psymodel.h:66
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:150
static void put_bits(PutBitContext *s, int n, unsigned int value)
Write up to 31 bits into a bitstream.
Definition: put_bits.h:136
#define ff_mdct_init
Definition: fft.h:146
Definition: aac.h:55
int num_swb
number of scalefactor window bands
Definition: aac.h:158
#define FFMAX(a, b)
Definition: common.h:53
void(* search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s, SingleChannelElement *sce, const float lambda)
Definition: aacenc.h:40
int off
Definition: dsputil_bfin.c:28
static int put_bits_count(PutBitContext *s)
Definition: put_bits.h:70
#define AACENC_FLAGS
Definition: aacenc.c:669
struct AACEncContext AACEncContext
AAC encoder context.
int bit_rate
the average bitrate
Definition: avcodec.h:1340
enum WindowSequence window_sequence[2]
Definition: aac.h:151
int cur_channel
Definition: aacenc.h:70
#define FFMIN(a, b)
Definition: common.h:55
static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
Write some auxiliary information about the created AAC file.
Definition: aacenc.c:473
float ret[2048]
PCM output.
Definition: aac.h:230
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:122
#define FF_PROFILE_AAC_LOW
Definition: avcodec.h:2467
#define FF_PROFILE_UNKNOWN
Definition: avcodec.h:2463
int pos[4]
Definition: aac.h:198
AAC definitions and structures.
FFTContext mdct128
short (128 samples) frame transform context
Definition: aacenc.h:59
PutBitContext pb
Definition: aacenc.h:57
static const uint8_t swb_size_128_48[]
Definition: aacenc.c:103
static const uint8_t swb_size_128_24[]
Definition: aacenc.c:107
LIBAVUTIL_VERSION_INT
Definition: eval.c:50
struct FFPsyContext::@42 bitres
static const uint8_t swb_size_1024_16[]
Definition: aacenc.c:80
static av_cold int aac_encode_end(AVCodecContext *avctx)
Definition: aacenc.c:656
int frame_size
Samples per packet, initialized when calling 'init'.
Definition: avcodec.h:1470
static const uint8_t swb_size_1024_32[]
Definition: aacenc.c:68
NULL
Definition: eval.c:50
external API header
static void put_audio_specific_config(AVCodecContext *avctx)
Make AAC audio config object.
Definition: aacenc.c:142
int sample_rate
samples per second
Definition: avcodec.h:1456
float ff_aac_kbd_short_128[128]
Definition: aactab.c:37
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
Encode MS data.
Definition: aacenc.c:301
av_default_item_name
Definition: dnxhdenc.c:43
int frame_bits
number of bits used for the previously encoded frame
Definition: avcodec.h:1564
static const OptionDef options[]
Definition: avconv.c:105
main external API structure.
Definition: avcodec.h:1329
static void close(AVCodecParserContext *s)
Definition: h264_parser.c:327
int bits
number of bits used in the bitresevoir
Definition: psymodel.h:88
DSPContext dsp
Definition: aacenc.h:60
#define CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:627
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:111
IndividualChannelStream ics
Definition: aac.h:220
void(* encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce, int win, int group_len, const float lambda)
Definition: aacenc.h:42
int extradata_size
Definition: avcodec.h:1388
uint8_t group_len[8]
Definition: aac.h:154
void * av_malloc(size_t size)
Allocate a block of size bytes with alignment suitable for all memory accesses (including vectors if ...
Definition: mem.c:64
Describe the class of an AVClass context structure.
Definition: log.h:33
static const uint8_t swb_size_1024_96[]
Definition: aacenc.c:49
int window_shape
window shape (sine/KBD/whatever)
Definition: psymodel.h:65
static void encode_pulses(AACEncContext *s, Pulse *pulse)
Encode pulse data.
Definition: aacenc.c:410
static const uint8_t swb_size_128_16[]
Definition: aacenc.c:111
const uint8_t * swb_sizes
table of scalefactor band sizes for a particular window
Definition: aac.h:157
#define FF_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:497
FFPsyContext psy
Definition: aacenc.h:67
const uint32_t ff_aac_scalefactor_code[121]
Definition: aactab.c:51
const struct FFPsyModel * model
encoder-specific model functions
Definition: psymodel.h:76
int ms_mode
Signals mid/side stereo flags coding mode (used by encoder)
Definition: aac.h:241
void(* vector_fmul_reverse)(float *dst, const float *src0, const float *src1, int len)
Definition: dsputil.h:413
static const uint8_t * swb_size_1024[]
Definition: aacenc.c:92
struct FFPsyPreprocessContext * psypp
Definition: aacenc.h:68
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:2208
uint8_t zeroes[128]
band is not coded (used by encoder)
Definition: aac.h:227
#define CODEC_CAP_EXPERIMENTAL
Codec is experimental and is thus avoided in favor of non experimental encoders.
Definition: avcodec.h:776
int sf_idx[128]
scalefactor indices (used by encoder)
Definition: aac.h:226
AVCodec ff_aac_encoder
Definition: aacenc.c:685
const int avpriv_mpeg4audio_sample_rates[16]
Definition: mpeg4audio.c:55
const char * name
Definition: audioconvert.c:61
Y Spectral Band Replication.
Definition: mpeg4audio.h:64
static av_cold int aac_encode_init(AVCodecContext *avctx)
Definition: aacenc.c:163
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:86
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:219
windowing related information
Definition: psymodel.h:63
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Definition: dsputil.h:412
#define ff_mdct_end
Definition: fft.h:147
av_cold struct FFPsyPreprocessContext * ff_psy_preprocess_init(AVCodecContext *avctx)
psychoacoustic model audio preprocessing initialization
Definition: psymodel.c:92
AACCoefficientsEncoder ff_aac_coders[]
Definition: aaccoder.c:1115
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce)
Encode scalefactors.
Definition: aacenc.c:388
AVSampleFormat
all in native-endian format
Definition: samplefmt.h:27
ChannelElement * cpe
channel elements
Definition: aacenc.h:66
Individual Channel Stream.
Definition: aac.h:149
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:52
static const uint8_t * swb_size_128[]
Definition: aacenc.c:119
DSP utils.
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:238
void * priv_data
Definition: avcodec.h:1531
int start
Definition: aac.h:197
FFTContext mdct1024
long (1024 samples) frame transform context
Definition: aacenc.h:58
int channels
number of audio channels
Definition: avcodec.h:1457
int num_pulse
Definition: aac.h:196
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
Encode scalefactor band coding type.
Definition: aacenc.c:377
static int aac_encode_frame(AVCodecContext *avctx, uint8_t *frame, int buf_size, void *data)
Definition: aacenc.c:491
enum BandType band_type[128]
band types
Definition: aac.h:223
#define LIBAVCODEC_IDENT
Definition: version.h:35
int frame_number
audio or video frame number
Definition: avcodec.h:1471
void ff_aac_tableinit(void)
Definition: aac_tablegen.h:35
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, int common_window)
Encode one channel of audio data.
Definition: aacenc.c:454
#define 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:750
static const int sizes[][2]
Definition: img2.c:91
#define CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:649
AAC data declarations.
av_cold void ff_psy_end(FFPsyContext *ctx)
Cleanup model context at the end.
Definition: psymodel.c:73
int window_type[3]
window type (short/long/transitional, etc.) - current, previous and next
Definition: psymodel.h:64
for(j=16;j >0;--j)
void ff_init_ff_sine_windows(int index)
initialize the specified entry of ff_sine_windows
bitstream writer API