FFmpeg: libavcodec/aacenc.c Source File

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 #include <float.h>
32 
33 #include "libavutil/channel_layout.h"
34 #include "libavutil/libm.h"
35 #include "libavutil/float_dsp.h"
36 #include "libavutil/opt.h"
37 #include "avcodec.h"
38 #include "encode.h"
39 #include "put_bits.h"
40 #include "internal.h"
41 #include "mpeg4audio.h"
42 #include "sinewin.h"
43 #include "profiles.h"
44 
45 #include "aac.h"
46 #include "aactab.h"
47 #include "aacenc.h"
48 #include "aacenctab.h"
49 #include "aacenc_utils.h"
50 
51 #include "psymodel.h"
52 
53  static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
54 {
55  int i, j;
56  AACEncContext *s = avctx->priv_data;
57  AACPCEInfo *pce = &s->pce;
58  const int bitexact = avctx->flags & AV_CODEC_FLAG_BITEXACT;
59  const char *aux_data = bitexact ? "Lavc" : LIBAVCODEC_IDENT;
60 
61  put_bits(pb, 4, 0);
62 
63  put_bits(pb, 2, avctx->profile);
64  put_bits(pb, 4, s->samplerate_index);
65 
66  put_bits(pb, 4, pce->num_ele[0]); /* Front */
67  put_bits(pb, 4, pce->num_ele[1]); /* Side */
68  put_bits(pb, 4, pce->num_ele[2]); /* Back */
69  put_bits(pb, 2, pce->num_ele[3]); /* LFE */
70  put_bits(pb, 3, 0); /* Assoc data */
71  put_bits(pb, 4, 0); /* CCs */
72 
73  put_bits(pb, 1, 0); /* Stereo mixdown */
74  put_bits(pb, 1, 0); /* Mono mixdown */
75  put_bits(pb, 1, 0); /* Something else */
76 
77  for (i = 0; i < 4; i++) {
78  for (j = 0; j < pce->num_ele[i]; j++) {
79  if (i < 3)
80  put_bits(pb, 1, pce->pairing[i][j]);
81  put_bits(pb, 4, pce->index[i][j]);
82  }
83  }
84 
85  align_put_bits(pb);
86  put_bits(pb, 8, strlen(aux_data));
87  ff_put_string(pb, aux_data, 0);
88 }
89 
90 /**
91  * Make AAC audio config object.
92  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
93  */
94  static int put_audio_specific_config(AVCodecContext *avctx)
95 {
96  PutBitContext pb;
97  AACEncContext *s = avctx->priv_data;
98  int channels = (!s->needs_pce)*(s->channels - (s->channels == 8 ? 1 : 0));
99  const int max_size = 32;
100 
101  avctx->extradata = av_mallocz(max_size);
102  if (!avctx->extradata)
103  return AVERROR(ENOMEM);
104 
105  init_put_bits(&pb, avctx->extradata, max_size);
106  put_bits(&pb, 5, s->profile+1); //profile
107  put_bits(&pb, 4, s->samplerate_index); //sample rate index
108  put_bits(&pb, 4, channels);
109  //GASpecificConfig
110  put_bits(&pb, 1, 0); //frame length - 1024 samples
111  put_bits(&pb, 1, 0); //does not depend on core coder
112  put_bits(&pb, 1, 0); //is not extension
113  if (s->needs_pce)
114  put_pce(&pb, avctx);
115 
116  //Explicitly Mark SBR absent
117  put_bits(&pb, 11, 0x2b7); //sync extension
118  put_bits(&pb, 5, AOT_SBR);
119  put_bits(&pb, 1, 0);
120  flush_put_bits(&pb);
121  avctx->extradata_size = put_bytes_output(&pb);
122 
123  return 0;
124 }
125 
126  void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
127 {
128  ++s->quantize_band_cost_cache_generation;
129  if (s->quantize_band_cost_cache_generation == 0) {
130  memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache));
131  s->quantize_band_cost_cache_generation = 1;
132  }
133 }
134 
135  #define WINDOW_FUNC(type) \
136 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
137  SingleChannelElement *sce, \
138  const float *audio)
139 
140  WINDOW_FUNC(only_long)
141 {
142  const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
143  const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
144  float *out = sce->ret_buf;
145 
146  fdsp->vector_fmul (out, audio, lwindow, 1024);
147  fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
148 }
149 
150  WINDOW_FUNC(long_start)
151 {
152  const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
153  const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
154  float *out = sce->ret_buf;
155 
156  fdsp->vector_fmul(out, audio, lwindow, 1024);
157  memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
158  fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
159  memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
160 }
161 
162  WINDOW_FUNC(long_stop)
163 {
164  const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
165  const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
166  float *out = sce->ret_buf;
167 
168  memset(out, 0, sizeof(out[0]) * 448);
169  fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
170  memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
171  fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
172 }
173 
174  WINDOW_FUNC(eight_short)
175 {
176  const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
177  const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
178  const float *in = audio + 448;
179  float *out = sce->ret_buf;
180  int w;
181 
182  for (w = 0; w < 8; w++) {
183  fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
184  out += 128;
185  in += 128;
186  fdsp->vector_fmul_reverse(out, in, swindow, 128);
187  out += 128;
188  }
189 }
190 
191  static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
192  SingleChannelElement *sce,
193  const float *audio) = {
194  [ONLY_LONG_SEQUENCE] = apply_only_long_window,
195  [LONG_START_SEQUENCE] = apply_long_start_window,
196  [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
197  [LONG_STOP_SEQUENCE] = apply_long_stop_window
198 };
199 
200  static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
201  float *audio)
202 {
203  int i;
204  const float *output = sce->ret_buf;
205 
206  apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
207 
208  if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
209  s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
210  else
211  for (i = 0; i < 1024; i += 128)
212  s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
213  memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
214  memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
215 }
216 
217 /**
218  * Encode ics_info element.
219  * @see Table 4.6 (syntax of ics_info)
220  */
221  static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
222 {
223  int w;
224 
225  put_bits(&s->pb, 1, 0); // ics_reserved bit
226  put_bits(&s->pb, 2, info->window_sequence[0]);
227  put_bits(&s->pb, 1, info->use_kb_window[0]);
228  if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
229  put_bits(&s->pb, 6, info->max_sfb);
230  put_bits(&s->pb, 1, !!info->predictor_present);
231  } else {
232  put_bits(&s->pb, 4, info->max_sfb);
233  for (w = 1; w < 8; w++)
234  put_bits(&s->pb, 1, !info->group_len[w]);
235  }
236 }
237 
238 /**
239  * Encode MS data.
240  * @see 4.6.8.1 "Joint Coding - M/S Stereo"
241  */
242  static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
243 {
244  int i, w;
245 
246  put_bits(pb, 2, cpe->ms_mode);
247  if (cpe->ms_mode == 1)
248  for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
249  for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
250  put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
251 }
252 
253 /**
254  * Produce integer coefficients from scalefactors provided by the model.
255  */
256  static void adjust_frame_information(ChannelElement *cpe, int chans)
257 {
258  int i, w, w2, g, ch;
259  int maxsfb, cmaxsfb;
260 
261  for (ch = 0; ch < chans; ch++) {
262  IndividualChannelStream *ics = &cpe->ch[ch].ics;
263  maxsfb = 0;
264  cpe->ch[ch].pulse.num_pulse = 0;
265  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
266  for (w2 = 0; w2 < ics->group_len[w]; w2++) {
267  for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
268  ;
269  maxsfb = FFMAX(maxsfb, cmaxsfb);
270  }
271  }
272  ics->max_sfb = maxsfb;
273 
274  //adjust zero bands for window groups
275  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
276  for (g = 0; g < ics->max_sfb; g++) {
277  i = 1;
278  for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
279  if (!cpe->ch[ch].zeroes[w2*16 + g]) {
280  i = 0;
281  break;
282  }
283  }
284  cpe->ch[ch].zeroes[w*16 + g] = i;
285  }
286  }
287  }
288 
289  if (chans > 1 && cpe->common_window) {
290  IndividualChannelStream *ics0 = &cpe->ch[0].ics;
291  IndividualChannelStream *ics1 = &cpe->ch[1].ics;
292  int msc = 0;
293  ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
294  ics1->max_sfb = ics0->max_sfb;
295  for (w = 0; w < ics0->num_windows*16; w += 16)
296  for (i = 0; i < ics0->max_sfb; i++)
297  if (cpe->ms_mask[w+i])
298  msc++;
299  if (msc == 0 || ics0->max_sfb == 0)
300  cpe->ms_mode = 0;
301  else
302  cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
303  }
304 }
305 
306  static void apply_intensity_stereo(ChannelElement *cpe)
307 {
308  int w, w2, g, i;
309  IndividualChannelStream *ics = &cpe->ch[0].ics;
310  if (!cpe->common_window)
311  return;
312  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
313  for (w2 = 0; w2 < ics->group_len[w]; w2++) {
314  int start = (w+w2) * 128;
315  for (g = 0; g < ics->num_swb; g++) {
316  int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
317  float scale = cpe->ch[0].is_ener[w*16+g];
318  if (!cpe->is_mask[w*16 + g]) {
319  start += ics->swb_sizes[g];
320  continue;
321  }
322  if (cpe->ms_mask[w*16 + g])
323  p *= -1;
324  for (i = 0; i < ics->swb_sizes[g]; i++) {
325  float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
326  cpe->ch[0].coeffs[start+i] = sum;
327  cpe->ch[1].coeffs[start+i] = 0.0f;
328  }
329  start += ics->swb_sizes[g];
330  }
331  }
332  }
333 }
334 
335  static void apply_mid_side_stereo(ChannelElement *cpe)
336 {
337  int w, w2, g, i;
338  IndividualChannelStream *ics = &cpe->ch[0].ics;
339  if (!cpe->common_window)
340  return;
341  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
342  for (w2 = 0; w2 < ics->group_len[w]; w2++) {
343  int start = (w+w2) * 128;
344  for (g = 0; g < ics->num_swb; g++) {
345  /* ms_mask can be used for other purposes in PNS and I/S,
346  * so must not apply M/S if any band uses either, even if
347  * ms_mask is set.
348  */
349  if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]
350  || cpe->ch[0].band_type[w*16 + g] >= NOISE_BT
351  || cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) {
352  start += ics->swb_sizes[g];
353  continue;
354  }
355  for (i = 0; i < ics->swb_sizes[g]; i++) {
356  float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
357  float R = L - cpe->ch[1].coeffs[start+i];
358  cpe->ch[0].coeffs[start+i] = L;
359  cpe->ch[1].coeffs[start+i] = R;
360  }
361  start += ics->swb_sizes[g];
362  }
363  }
364  }
365 }
366 
367 /**
368  * Encode scalefactor band coding type.
369  */
370  static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
371 {
372  int w;
373 
374  if (s->coder->set_special_band_scalefactors)
375  s->coder->set_special_band_scalefactors(s, sce);
376 
377  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
378  s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
379 }
380 
381 /**
382  * Encode scalefactors.
383  */
384  static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
385  SingleChannelElement *sce)
386 {
387  int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
388  int off_is = 0, noise_flag = 1;
389  int i, w;
390 
391  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
392  for (i = 0; i < sce->ics.max_sfb; i++) {
393  if (!sce->zeroes[w*16 + i]) {
394  if (sce->band_type[w*16 + i] == NOISE_BT) {
395  diff = sce->sf_idx[w*16 + i] - off_pns;
396  off_pns = sce->sf_idx[w*16 + i];
397  if (noise_flag-- > 0) {
398  put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
399  continue;
400  }
401  } else if (sce->band_type[w*16 + i] == INTENSITY_BT ||
402  sce->band_type[w*16 + i] == INTENSITY_BT2) {
403  diff = sce->sf_idx[w*16 + i] - off_is;
404  off_is = sce->sf_idx[w*16 + i];
405  } else {
406  diff = sce->sf_idx[w*16 + i] - off_sf;
407  off_sf = sce->sf_idx[w*16 + i];
408  }
409  diff += SCALE_DIFF_ZERO;
410  av_assert0(diff >= 0 && diff <= 120);
411  put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
412  }
413  }
414  }
415 }
416 
417 /**
418  * Encode pulse data.
419  */
420  static void encode_pulses(AACEncContext *s, Pulse *pulse)
421 {
422  int i;
423 
424  put_bits(&s->pb, 1, !!pulse->num_pulse);
425  if (!pulse->num_pulse)
426  return;
427 
428  put_bits(&s->pb, 2, pulse->num_pulse - 1);
429  put_bits(&s->pb, 6, pulse->start);
430  for (i = 0; i < pulse->num_pulse; i++) {
431  put_bits(&s->pb, 5, pulse->pos[i]);
432  put_bits(&s->pb, 4, pulse->amp[i]);
433  }
434 }
435 
436 /**
437  * Encode spectral coefficients processed by psychoacoustic model.
438  */
439  static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
440 {
441  int start, i, w, w2;
442 
443  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
444  start = 0;
445  for (i = 0; i < sce->ics.max_sfb; i++) {
446  if (sce->zeroes[w*16 + i]) {
447  start += sce->ics.swb_sizes[i];
448  continue;
449  }
450  for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
451  s->coder->quantize_and_encode_band(s, &s->pb,
452  &sce->coeffs[start + w2*128],
453  NULL, sce->ics.swb_sizes[i],
454  sce->sf_idx[w*16 + i],
455  sce->band_type[w*16 + i],
456  s->lambda,
457  sce->ics.window_clipping[w]);
458  }
459  start += sce->ics.swb_sizes[i];
460  }
461  }
462 }
463 
464 /**
465  * Downscale spectral coefficients for near-clipping windows to avoid artifacts
466  */
467  static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
468 {
469  int start, i, j, w;
470 
471  if (sce->ics.clip_avoidance_factor < 1.0f) {
472  for (w = 0; w < sce->ics.num_windows; w++) {
473  start = 0;
474  for (i = 0; i < sce->ics.max_sfb; i++) {
475  float *swb_coeffs = &sce->coeffs[start + w*128];
476  for (j = 0; j < sce->ics.swb_sizes[i]; j++)
477  swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
478  start += sce->ics.swb_sizes[i];
479  }
480  }
481  }
482 }
483 
484 /**
485  * Encode one channel of audio data.
486  */
487  static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
488  SingleChannelElement *sce,
489  int common_window)
490 {
491  put_bits(&s->pb, 8, sce->sf_idx[0]);
492  if (!common_window) {
493  put_ics_info(s, &sce->ics);
494  if (s->coder->encode_main_pred)
495  s->coder->encode_main_pred(s, sce);
496  if (s->coder->encode_ltp_info)
497  s->coder->encode_ltp_info(s, sce, 0);
498  }
499  encode_band_info(s, sce);
500  encode_scale_factors(avctx, s, sce);
501  encode_pulses(s, &sce->pulse);
502  put_bits(&s->pb, 1, !!sce->tns.present);
503  if (s->coder->encode_tns_info)
504  s->coder->encode_tns_info(s, sce);
505  put_bits(&s->pb, 1, 0); //ssr
506  encode_spectral_coeffs(s, sce);
507  return 0;
508 }
509 
510 /**
511  * Write some auxiliary information about the created AAC file.
512  */
513  static void put_bitstream_info(AACEncContext *s, const char *name)
514 {
515  int i, namelen, padbits;
516 
517  namelen = strlen(name) + 2;
518  put_bits(&s->pb, 3, TYPE_FIL);
519  put_bits(&s->pb, 4, FFMIN(namelen, 15));
520  if (namelen >= 15)
521  put_bits(&s->pb, 8, namelen - 14);
522  put_bits(&s->pb, 4, 0); //extension type - filler
523  padbits = -put_bits_count(&s->pb) & 7;
524  align_put_bits(&s->pb);
525  for (i = 0; i < namelen - 2; i++)
526  put_bits(&s->pb, 8, name[i]);
527  put_bits(&s->pb, 12 - padbits, 0);
528 }
529 
530 /*
531  * Copy input samples.
532  * Channels are reordered from libavcodec's default order to AAC order.
533  */
534  static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
535 {
536  int ch;
537  int end = 2048 + (frame ? frame->nb_samples : 0);
538  const uint8_t *channel_map = s->reorder_map;
539 
540  /* copy and remap input samples */
541  for (ch = 0; ch < s->channels; ch++) {
542  /* copy last 1024 samples of previous frame to the start of the current frame */
543  memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
544 
545  /* copy new samples and zero any remaining samples */
546  if (frame) {
547  memcpy(&s->planar_samples[ch][2048],
548  frame->extended_data[channel_map[ch]],
549  frame->nb_samples * sizeof(s->planar_samples[0][0]));
550  }
551  memset(&s->planar_samples[ch][end], 0,
552  (3072 - end) * sizeof(s->planar_samples[0][0]));
553  }
554 }
555 
556  static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
557  const AVFrame *frame, int *got_packet_ptr)
558 {
559  AACEncContext *s = avctx->priv_data;
560  float **samples = s->planar_samples, *samples2, *la, *overlap;
561  ChannelElement *cpe;
562  SingleChannelElement *sce;
563  IndividualChannelStream *ics;
564  int i, its, ch, w, chans, tag, start_ch, ret, frame_bits;
565  int target_bits, rate_bits, too_many_bits, too_few_bits;
566  int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
567  int chan_el_counter[4];
568  FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
569 
570  /* add current frame to queue */
571  if (frame) {
572  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
573  return ret;
574  } else {
575  if (!s->afq.remaining_samples || (!s->afq.frame_alloc && !s->afq.frame_count))
576  return 0;
577  }
578 
579  copy_input_samples(s, frame);
580  if (s->psypp)
581  ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
582 
583  if (!avctx->frame_number)
584  return 0;
585 
586  start_ch = 0;
587  for (i = 0; i < s->chan_map[0]; i++) {
588  FFPsyWindowInfo* wi = windows + start_ch;
589  tag = s->chan_map[i+1];
590  chans = tag == TYPE_CPE ? 2 : 1;
591  cpe = &s->cpe[i];
592  for (ch = 0; ch < chans; ch++) {
593  int k;
594  float clip_avoidance_factor;
595  sce = &cpe->ch[ch];
596  ics = &sce->ics;
597  s->cur_channel = start_ch + ch;
598  overlap = &samples[s->cur_channel][0];
599  samples2 = overlap + 1024;
600  la = samples2 + (448+64);
601  if (!frame)
602  la = NULL;
603  if (tag == TYPE_LFE) {
604  wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE;
605  wi[ch].window_shape = 0;
606  wi[ch].num_windows = 1;
607  wi[ch].grouping[0] = 1;
608  wi[ch].clipping[0] = 0;
609 
610  /* Only the lowest 12 coefficients are used in a LFE channel.
611  * The expression below results in only the bottom 8 coefficients
612  * being used for 11.025kHz to 16kHz sample rates.
613  */
614  ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
615  } else {
616  wi[ch] = s->psy.model->window(&s->psy, samples2, la, s->cur_channel,
617  ics->window_sequence[0]);
618  }
619  ics->window_sequence[1] = ics->window_sequence[0];
620  ics->window_sequence[0] = wi[ch].window_type[0];
621  ics->use_kb_window[1] = ics->use_kb_window[0];
622  ics->use_kb_window[0] = wi[ch].window_shape;
623  ics->num_windows = wi[ch].num_windows;
624  ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
625  ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
626  ics->max_sfb = FFMIN(ics->max_sfb, ics->num_swb);
627  ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
628  ff_swb_offset_128 [s->samplerate_index]:
629  ff_swb_offset_1024[s->samplerate_index];
630  ics->tns_max_bands = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
631  ff_tns_max_bands_128 [s->samplerate_index]:
632  ff_tns_max_bands_1024[s->samplerate_index];
633 
634  for (w = 0; w < ics->num_windows; w++)
635  ics->group_len[w] = wi[ch].grouping[w];
636 
637  /* Calculate input sample maximums and evaluate clipping risk */
638  clip_avoidance_factor = 0.0f;
639  for (w = 0; w < ics->num_windows; w++) {
640  const float *wbuf = overlap + w * 128;
641  const int wlen = 2048 / ics->num_windows;
642  float max = 0;
643  int j;
644  /* mdct input is 2 * output */
645  for (j = 0; j < wlen; j++)
646  max = FFMAX(max, fabsf(wbuf[j]));
647  wi[ch].clipping[w] = max;
648  }
649  for (w = 0; w < ics->num_windows; w++) {
650  if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
651  ics->window_clipping[w] = 1;
652  clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
653  } else {
654  ics->window_clipping[w] = 0;
655  }
656  }
657  if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
658  ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
659  } else {
660  ics->clip_avoidance_factor = 1.0f;
661  }
662 
663  apply_window_and_mdct(s, sce, overlap);
664 
665  if (s->options.ltp && s->coder->update_ltp) {
666  s->coder->update_ltp(s, sce);
667  apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]);
668  s->mdct1024.mdct_calc(&s->mdct1024, sce->lcoeffs, sce->ret_buf);
669  }
670 
671  for (k = 0; k < 1024; k++) {
672  if (!(fabs(cpe->ch[ch].coeffs[k]) < 1E16)) { // Ensure headroom for energy calculation
673  av_log(avctx, AV_LOG_ERROR, "Input contains (near) NaN/+-Inf\n");
674  return AVERROR(EINVAL);
675  }
676  }
677  avoid_clipping(s, sce);
678  }
679  start_ch += chans;
680  }
681  if ((ret = ff_alloc_packet(avctx, avpkt, 8192 * s->channels)) < 0)
682  return ret;
683  frame_bits = its = 0;
684  do {
685  init_put_bits(&s->pb, avpkt->data, avpkt->size);
686 
687  if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
688  put_bitstream_info(s, LIBAVCODEC_IDENT);
689  start_ch = 0;
690  target_bits = 0;
691  memset(chan_el_counter, 0, sizeof(chan_el_counter));
692  for (i = 0; i < s->chan_map[0]; i++) {
693  FFPsyWindowInfo* wi = windows + start_ch;
694  const float *coeffs[2];
695  tag = s->chan_map[i+1];
696  chans = tag == TYPE_CPE ? 2 : 1;
697  cpe = &s->cpe[i];
698  cpe->common_window = 0;
699  memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
700  memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
701  put_bits(&s->pb, 3, tag);
702  put_bits(&s->pb, 4, chan_el_counter[tag]++);
703  for (ch = 0; ch < chans; ch++) {
704  sce = &cpe->ch[ch];
705  coeffs[ch] = sce->coeffs;
706  sce->ics.predictor_present = 0;
707  sce->ics.ltp.present = 0;
708  memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used));
709  memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
710  memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
711  for (w = 0; w < 128; w++)
712  if (sce->band_type[w] > RESERVED_BT)
713  sce->band_type[w] = 0;
714  }
715  s->psy.bitres.alloc = -1;
716  s->psy.bitres.bits = s->last_frame_pb_count / s->channels;
717  s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
718  if (s->psy.bitres.alloc > 0) {
719  /* Lambda unused here on purpose, we need to take psy's unscaled allocation */
720  target_bits += s->psy.bitres.alloc
721  * (s->lambda / (avctx->global_quality ? avctx->global_quality : 120));
722  s->psy.bitres.alloc /= chans;
723  }
724  s->cur_type = tag;
725  for (ch = 0; ch < chans; ch++) {
726  s->cur_channel = start_ch + ch;
727  if (s->options.pns && s->coder->mark_pns)
728  s->coder->mark_pns(s, avctx, &cpe->ch[ch]);
729  s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
730  }
731  if (chans > 1
732  && wi[0].window_type[0] == wi[1].window_type[0]
733  && wi[0].window_shape == wi[1].window_shape) {
734 
735  cpe->common_window = 1;
736  for (w = 0; w < wi[0].num_windows; w++) {
737  if (wi[0].grouping[w] != wi[1].grouping[w]) {
738  cpe->common_window = 0;
739  break;
740  }
741  }
742  }
743  for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
744  sce = &cpe->ch[ch];
745  s->cur_channel = start_ch + ch;
746  if (s->options.tns && s->coder->search_for_tns)
747  s->coder->search_for_tns(s, sce);
748  if (s->options.tns && s->coder->apply_tns_filt)
749  s->coder->apply_tns_filt(s, sce);
750  if (sce->tns.present)
751  tns_mode = 1;
752  if (s->options.pns && s->coder->search_for_pns)
753  s->coder->search_for_pns(s, avctx, sce);
754  }
755  s->cur_channel = start_ch;
756  if (s->options.intensity_stereo) { /* Intensity Stereo */
757  if (s->coder->search_for_is)
758  s->coder->search_for_is(s, avctx, cpe);
759  if (cpe->is_mode) is_mode = 1;
760  apply_intensity_stereo(cpe);
761  }
762  if (s->options.pred) { /* Prediction */
763  for (ch = 0; ch < chans; ch++) {
764  sce = &cpe->ch[ch];
765  s->cur_channel = start_ch + ch;
766  if (s->options.pred && s->coder->search_for_pred)
767  s->coder->search_for_pred(s, sce);
768  if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
769  }
770  if (s->coder->adjust_common_pred)
771  s->coder->adjust_common_pred(s, cpe);
772  for (ch = 0; ch < chans; ch++) {
773  sce = &cpe->ch[ch];
774  s->cur_channel = start_ch + ch;
775  if (s->options.pred && s->coder->apply_main_pred)
776  s->coder->apply_main_pred(s, sce);
777  }
778  s->cur_channel = start_ch;
779  }
780  if (s->options.mid_side) { /* Mid/Side stereo */
781  if (s->options.mid_side == -1 && s->coder->search_for_ms)
782  s->coder->search_for_ms(s, cpe);
783  else if (cpe->common_window)
784  memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
785  apply_mid_side_stereo(cpe);
786  }
787  adjust_frame_information(cpe, chans);
788  if (s->options.ltp) { /* LTP */
789  for (ch = 0; ch < chans; ch++) {
790  sce = &cpe->ch[ch];
791  s->cur_channel = start_ch + ch;
792  if (s->coder->search_for_ltp)
793  s->coder->search_for_ltp(s, sce, cpe->common_window);
794  if (sce->ics.ltp.present) pred_mode = 1;
795  }
796  s->cur_channel = start_ch;
797  if (s->coder->adjust_common_ltp)
798  s->coder->adjust_common_ltp(s, cpe);
799  }
800  if (chans == 2) {
801  put_bits(&s->pb, 1, cpe->common_window);
802  if (cpe->common_window) {
803  put_ics_info(s, &cpe->ch[0].ics);
804  if (s->coder->encode_main_pred)
805  s->coder->encode_main_pred(s, &cpe->ch[0]);
806  if (s->coder->encode_ltp_info)
807  s->coder->encode_ltp_info(s, &cpe->ch[0], 1);
808  encode_ms_info(&s->pb, cpe);
809  if (cpe->ms_mode) ms_mode = 1;
810  }
811  }
812  for (ch = 0; ch < chans; ch++) {
813  s->cur_channel = start_ch + ch;
814  encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
815  }
816  start_ch += chans;
817  }
818 
819  if (avctx->flags & AV_CODEC_FLAG_QSCALE) {
820  /* When using a constant Q-scale, don't mess with lambda */
821  break;
822  }
823 
824  /* rate control stuff
825  * allow between the nominal bitrate, and what psy's bit reservoir says to target
826  * but drift towards the nominal bitrate always
827  */
828  frame_bits = put_bits_count(&s->pb);
829  rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate;
830  rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3);
831  too_many_bits = FFMAX(target_bits, rate_bits);
832  too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3);
833  too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits);
834 
835  /* When using ABR, be strict (but only for increasing) */
836  too_few_bits = too_few_bits - too_few_bits/8;
837  too_many_bits = too_many_bits + too_many_bits/2;
838 
839  if ( its == 0 /* for steady-state Q-scale tracking */
840  || (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits))
841  || frame_bits >= 6144 * s->channels - 3 )
842  {
843  float ratio = ((float)rate_bits) / frame_bits;
844 
845  if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) {
846  /*
847  * This path is for steady-state Q-scale tracking
848  * When frame bits fall within the stable range, we still need to adjust
849  * lambda to maintain it like so in a stable fashion (large jumps in lambda
850  * create artifacts and should be avoided), but slowly
851  */
852  ratio = sqrtf(sqrtf(ratio));
853  ratio = av_clipf(ratio, 0.9f, 1.1f);
854  } else {
855  /* Not so fast though */
856  ratio = sqrtf(ratio);
857  }
858  s->lambda = av_clipf(s->lambda * ratio, FLT_EPSILON, 65536.f);
859 
860  /* Keep iterating if we must reduce and lambda is in the sky */
861  if (ratio > 0.9f && ratio < 1.1f) {
862  break;
863  } else {
864  if (is_mode || ms_mode || tns_mode || pred_mode) {
865  for (i = 0; i < s->chan_map[0]; i++) {
866  // Must restore coeffs
867  chans = tag == TYPE_CPE ? 2 : 1;
868  cpe = &s->cpe[i];
869  for (ch = 0; ch < chans; ch++)
870  memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
871  }
872  }
873  its++;
874  }
875  } else {
876  break;
877  }
878  } while (1);
879 
880  if (s->options.ltp && s->coder->ltp_insert_new_frame)
881  s->coder->ltp_insert_new_frame(s);
882 
883  put_bits(&s->pb, 3, TYPE_END);
884  flush_put_bits(&s->pb);
885 
886  s->last_frame_pb_count = put_bits_count(&s->pb);
887  avpkt->size = put_bytes_output(&s->pb);
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  *got_packet_ptr = 1;
896  return 0;
897 }
898 
899  static av_cold int aac_encode_end(AVCodecContext *avctx)
900 {
901  AACEncContext *s = avctx->priv_data;
902 
903  av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_count ? s->lambda_sum / s->lambda_count : NAN);
904 
905  ff_mdct_end(&s->mdct1024);
906  ff_mdct_end(&s->mdct128);
907  ff_psy_end(&s->psy);
908  ff_lpc_end(&s->lpc);
909  if (s->psypp)
910  ff_psy_preprocess_end(s->psypp);
911  av_freep(&s->buffer.samples);
912  av_freep(&s->cpe);
913  av_freep(&s->fdsp);
914  ff_af_queue_close(&s->afq);
915  return 0;
916 }
917 
918  static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
919 {
920  int ret = 0;
921 
922  s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
923  if (!s->fdsp)
924  return AVERROR(ENOMEM);
925 
926  // window init
927  ff_aac_float_common_init();
928 
929  if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
930  return ret;
931  if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0)
932  return ret;
933 
934  return 0;
935 }
936 
937  static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
938 {
939  int ch;
940  if (!FF_ALLOCZ_TYPED_ARRAY(s->buffer.samples, s->channels * 3 * 1024) ||
941  !FF_ALLOCZ_TYPED_ARRAY(s->cpe, s->chan_map[0]))
942  return AVERROR(ENOMEM);
943 
944  for(ch = 0; ch < s->channels; ch++)
945  s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
946 
947  return 0;
948 }
949 
950  static av_cold int aac_encode_init(AVCodecContext *avctx)
951 {
952  AACEncContext *s = avctx->priv_data;
953  int i, ret = 0;
954  const uint8_t *sizes[2];
955  uint8_t grouping[AAC_MAX_CHANNELS];
956  int lengths[2];
957 
958  /* Constants */
959  s->last_frame_pb_count = 0;
960  avctx->frame_size = 1024;
961  avctx->initial_padding = 1024;
962  s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
963 
964  /* Channel map and unspecified bitrate guessing */
965  s->channels = avctx->channels;
966 
967  s->needs_pce = 1;
968  for (i = 0; i < FF_ARRAY_ELEMS(aac_normal_chan_layouts); i++) {
969  if (avctx->channel_layout == aac_normal_chan_layouts[i]) {
970  s->needs_pce = s->options.pce;
971  break;
972  }
973  }
974 
975  if (s->needs_pce) {
976  char buf[64];
977  for (i = 0; i < FF_ARRAY_ELEMS(aac_pce_configs); i++)
978  if (avctx->channel_layout == aac_pce_configs[i].layout)
979  break;
980  av_get_channel_layout_string(buf, sizeof(buf), -1, avctx->channel_layout);
981  ERROR_IF(i == FF_ARRAY_ELEMS(aac_pce_configs), "Unsupported channel layout \"%s\"\n", buf);
982  av_log(avctx, AV_LOG_INFO, "Using a PCE to encode channel layout \"%s\"\n", buf);
983  s->pce = aac_pce_configs[i];
984  s->reorder_map = s->pce.reorder_map;
985  s->chan_map = s->pce.config_map;
986  } else {
987  s->reorder_map = aac_chan_maps[s->channels - 1];
988  s->chan_map = aac_chan_configs[s->channels - 1];
989  }
990 
991  if (!avctx->bit_rate) {
992  for (i = 1; i <= s->chan_map[0]; i++) {
993  avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */
994  s->chan_map[i] == TYPE_LFE ? 16000 : /* LFE */
995  69000 ; /* SCE */
996  }
997  }
998 
999  /* Samplerate */
1000  for (i = 0; i < 16; i++)
1001  if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
1002  break;
1003  s->samplerate_index = i;
1004  ERROR_IF(s->samplerate_index == 16 ||
1005  s->samplerate_index >= ff_aac_swb_size_1024_len ||
1006  s->samplerate_index >= ff_aac_swb_size_128_len,
1007  "Unsupported sample rate %d\n", avctx->sample_rate);
1008 
1009  /* Bitrate limiting */
1010  WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
1011  "Too many bits %f > %d per frame requested, clamping to max\n",
1012  1024.0 * avctx->bit_rate / avctx->sample_rate,
1013  6144 * s->channels);
1014  avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate,
1015  avctx->bit_rate);
1016 
1017  /* Profile and option setting */
1018  avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
1019  avctx->profile;
1020  for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++)
1021  if (avctx->profile == aacenc_profiles[i])
1022  break;
1023  if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) {
1024  avctx->profile = FF_PROFILE_AAC_LOW;
1025  ERROR_IF(s->options.pred,
1026  "Main prediction unavailable in the \"mpeg2_aac_low\" profile\n");
1027  ERROR_IF(s->options.ltp,
1028  "LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n");
1029  WARN_IF(s->options.pns,
1030  "PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n");
1031  s->options.pns = 0;
1032  } else if (avctx->profile == FF_PROFILE_AAC_LTP) {
1033  s->options.ltp = 1;
1034  ERROR_IF(s->options.pred,
1035  "Main prediction unavailable in the \"aac_ltp\" profile\n");
1036  } else if (avctx->profile == FF_PROFILE_AAC_MAIN) {
1037  s->options.pred = 1;
1038  ERROR_IF(s->options.ltp,
1039  "LTP prediction unavailable in the \"aac_main\" profile\n");
1040  } else if (s->options.ltp) {
1041  avctx->profile = FF_PROFILE_AAC_LTP;
1042  WARN_IF(1,
1043  "Chainging profile to \"aac_ltp\"\n");
1044  ERROR_IF(s->options.pred,
1045  "Main prediction unavailable in the \"aac_ltp\" profile\n");
1046  } else if (s->options.pred) {
1047  avctx->profile = FF_PROFILE_AAC_MAIN;
1048  WARN_IF(1,
1049  "Chainging profile to \"aac_main\"\n");
1050  ERROR_IF(s->options.ltp,
1051  "LTP prediction unavailable in the \"aac_main\" profile\n");
1052  }
1053  s->profile = avctx->profile;
1054 
1055  /* Coder limitations */
1056  s->coder = &ff_aac_coders[s->options.coder];
1057  if (s->options.coder == AAC_CODER_ANMR) {
1058  ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
1059  "The ANMR coder is considered experimental, add -strict -2 to enable!\n");
1060  s->options.intensity_stereo = 0;
1061  s->options.pns = 0;
1062  }
1063  ERROR_IF(s->options.ltp && avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
1064  "The LPT profile requires experimental compliance, add -strict -2 to enable!\n");
1065 
1066  /* M/S introduces horrible artifacts with multichannel files, this is temporary */
1067  if (s->channels > 3)
1068  s->options.mid_side = 0;
1069 
1070  if ((ret = dsp_init(avctx, s)) < 0)
1071  return ret;
1072 
1073  if ((ret = alloc_buffers(avctx, s)) < 0)
1074  return ret;
1075 
1076  if ((ret = put_audio_specific_config(avctx)))
1077  return ret;
1078 
1079  sizes[0] = ff_aac_swb_size_1024[s->samplerate_index];
1080  sizes[1] = ff_aac_swb_size_128[s->samplerate_index];
1081  lengths[0] = ff_aac_num_swb_1024[s->samplerate_index];
1082  lengths[1] = ff_aac_num_swb_128[s->samplerate_index];
1083  for (i = 0; i < s->chan_map[0]; i++)
1084  grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
1085  if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
1086  s->chan_map[0], grouping)) < 0)
1087  return ret;
1088  s->psypp = ff_psy_preprocess_init(avctx);
1089  ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);
1090  s->random_state = 0x1f2e3d4c;
1091 
1092  s->abs_pow34 = abs_pow34_v;
1093  s->quant_bands = quantize_bands;
1094 
1095  if (ARCH_X86)
1096  ff_aac_dsp_init_x86(s);
1097 
1098  if (HAVE_MIPSDSP)
1099  ff_aac_coder_init_mips(s);
1100 
1101  ff_af_queue_init(avctx, &s->afq);
1102  ff_aac_tableinit();
1103 
1104  return 0;
1105 }
1106 
1107  #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1108  static const AVOption aacenc_options[] = {
1109  {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
1110  {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1111  {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1112  {"fast", "Default fast search", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1113  {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
1114  {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1115  {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1116  {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1117  {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1118  {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1119  {"aac_pce", "Forces the use of PCEs", offsetof(AACEncContext, options.pce), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1120  FF_AAC_PROFILE_OPTS
1121  {NULL}
1122 };
1123 
1124  static const AVClass aacenc_class = {
1125  .class_name = "AAC encoder",
1126  .item_name = av_default_item_name,
1127  .option = aacenc_options,
1128  .version = LIBAVUTIL_VERSION_INT,
1129 };
1130 
1131  static const AVCodecDefault aac_encode_defaults[] = {
1132  { "b", "0" },
1133  { NULL }
1134 };
1135 
1136  const AVCodec ff_aac_encoder = {
1137  .name = "aac",
1138  .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
1139  .type = AVMEDIA_TYPE_AUDIO,
1140  .id = AV_CODEC_ID_AAC,
1141  .priv_data_size = sizeof(AACEncContext),
1142  .init = aac_encode_init,
1143  .encode2 = aac_encode_frame,
1144  .close = aac_encode_end,
1145  .defaults = aac_encode_defaults,
1146  .supported_samplerates = ff_mpeg4audio_sample_rates,
1147  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
1148  .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
1149  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
1150  AV_SAMPLE_FMT_NONE },
1151  .priv_class = &aacenc_class,
1152 };
FF_ALLOCZ_TYPED_ARRAY
#define FF_ALLOCZ_TYPED_ARRAY(p, nelem)
Definition: internal.h:98
AVCodecContext::frame_size
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:1012
AVCodec
AVCodec.
Definition: codec.h:202
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
FF_CODEC_CAP_INIT_THREADSAFE
#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:42
name
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default minimum maximum flags name is the option name
Definition: writing_filters.txt:88
ff_tns_max_bands_128
const uint8_t ff_tns_max_bands_128[]
Definition: aactab.c:1424
aacenc_class
static const AVClass aacenc_class
Definition: aacenc.c:1124
AVERROR
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
opt.h
LIBAVCODEC_IDENT
#define LIBAVCODEC_IDENT
Definition: version.h:42
AVCodecContext::channel_layout
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1043
put_bitstream_info
static void put_bitstream_info(AACEncContext *s, const char *name)
Write some auxiliary information about the created AAC file.
Definition: aacenc.c:513
ff_aac_kbd_short_128
float ff_aac_kbd_short_128[128]
libm.h
SingleChannelElement::pulse
Pulse pulse
Definition: aac.h:252
align_put_bits
static void align_put_bits(PutBitContext *s)
Pad the bitstream with zeros up to the next byte boundary.
Definition: put_bits.h:412
TYPE_FIL
@ TYPE_FIL
Definition: aac.h:63
ff_af_queue_remove
void ff_af_queue_remove(AudioFrameQueue *afq, int nb_samples, int64_t *pts, int64_t *duration)
Remove frame(s) from the queue.
Definition: audio_frame_queue.c:75
out
FILE * out
Definition: movenc.c:54
FF_COMPLIANCE_EXPERIMENTAL
#define FF_COMPLIANCE_EXPERIMENTAL
Allow nonstandardized experimental things.
Definition: avcodec.h:1285
put_bytes_output
static int put_bytes_output(const PutBitContext *s)
Definition: put_bits.h:88
AVCodecContext::sample_rate
int sample_rate
samples per second
Definition: avcodec.h:992
ff_af_queue_close
void ff_af_queue_close(AudioFrameQueue *afq)
Close AudioFrameQueue.
Definition: audio_frame_queue.c:36
aacenctab.h
abs_pow34_v
static void abs_pow34_v(float *out, const float *in, const int size)
Definition: aacenc_utils.h:40
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:948
copy_input_samples
static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
Definition: aacenc.c:534
aac_encode_init
static av_cold int aac_encode_init(AVCodecContext *avctx)
Definition: aacenc.c:950
aacenc_profiles
static const int aacenc_profiles[]
Definition: aacenctab.h:126
Pulse::num_pulse
int num_pulse
Definition: aac.h:226
AV_CODEC_FLAG_QSCALE
#define AV_CODEC_FLAG_QSCALE
Use fixed qscale.
Definition: avcodec.h:210
output
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
Definition: filter_design.txt:225
FF_PROFILE_AAC_MAIN
#define FF_PROFILE_AAC_MAIN
Definition: avcodec.h:1529
SingleChannelElement::zeroes
uint8_t zeroes[128]
band is not coded (used by encoder)
Definition: aac.h:258
init_put_bits
static void init_put_bits(PutBitContext *s, uint8_t *buffer, int buffer_size)
Initialize the PutBitContext s.
Definition: put_bits.h:61
ff_af_queue_init
av_cold void ff_af_queue_init(AVCodecContext *avctx, AudioFrameQueue *afq)
Initialize AudioFrameQueue.
Definition: audio_frame_queue.c:28
av_get_channel_layout_string
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.
Definition: channel_layout.c:217
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:317
put_bits
static void put_bits(Jpeg2000EncoderContext *s, int val, int n)
put n times val bit
Definition: j2kenc.c:220
WARN_IF
#define WARN_IF(cond,...)
Definition: aacenc_utils.h:274
w
uint8_t w
Definition: llviddspenc.c:38
R
#define R
Definition: huffyuvdsp.h:34
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:373
ff_aac_coders
const AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB]
Definition: aaccoder.c:896
AVOption
AVOption.
Definition: opt.h:247
encode.h
ff_mdct_init
#define ff_mdct_init
Definition: fft.h:153
FF_PROFILE_AAC_LTP
#define FF_PROFILE_AAC_LTP
Definition: avcodec.h:1532
encode_band_info
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
Encode scalefactor band coding type.
Definition: aacenc.c:370
TemporalNoiseShaping::present
int present
Definition: aac.h:200
FFPsyWindowInfo::window_shape
int window_shape
window shape (sine/KBD/whatever)
Definition: psymodel.h:79
float.h
aac_chan_configs
static const uint8_t aac_chan_configs[AAC_MAX_CHANNELS][6]
default channel configurations
Definition: aacenctab.h:59
AAC_CODER_NB
@ AAC_CODER_NB
Definition: aacenc.h:43
LongTermPrediction::used
int8_t used[MAX_LTP_LONG_SFB]
Definition: aac.h:169
AVPacket::duration
int64_t duration
Duration of this packet in AVStream->time_base units, 0 if unknown.
Definition: packet.h:391
SingleChannelElement::pcoeffs
INTFLOAT pcoeffs[1024]
coefficients for IMDCT, pristine
Definition: aac.h:262
max
#define max(a, b)
Definition: cuda_runtime.h:33
FFMAX
#define FFMAX(a, b)
Definition: macros.h:47
ff_swb_offset_128
const uint16_t *const ff_swb_offset_128[]
Definition: aactab.c:1386
AACPCEInfo::layout
int64_t layout
Definition: aacenc.h:97
encode_spectral_coeffs
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
Encode spectral coefficients processed by psychoacoustic model.
Definition: aacenc.c:439
ff_tns_max_bands_1024
const uint8_t ff_tns_max_bands_1024[]
Definition: aactab.c:1412
AAC_CODER_FAST
@ AAC_CODER_FAST
Definition: aacenc.h:41
init
static int init
Definition: av_tx.c:47
IndividualChannelStream::num_swb
int num_swb
number of scalefactor window bands
Definition: aac.h:184
WINDOW_FUNC
#define WINDOW_FUNC(type)
Definition: aacenc.c:135
avoid_clipping
static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
Downscale spectral coefficients for near-clipping windows to avoid artifacts.
Definition: aacenc.c:467
mpeg4audio.h
SingleChannelElement::ret_buf
INTFLOAT ret_buf[2048]
PCM output buffer.
Definition: aac.h:265
FF_PROFILE_MPEG2_AAC_LOW
#define FF_PROFILE_MPEG2_AAC_LOW
Definition: avcodec.h:1537
apply_mid_side_stereo
static void apply_mid_side_stereo(ChannelElement *cpe)
Definition: aacenc.c:335
TYPE_CPE
@ TYPE_CPE
Definition: aac.h:58
ChannelElement::ms_mode
int ms_mode
Signals mid/side stereo flags coding mode (used by encoder)
Definition: aac.h:280
AVCodecContext::initial_padding
int initial_padding
Audio only.
Definition: avcodec.h:1701
defaults
static const AVCodecDefault defaults[]
Definition: amfenc_h264.c:361
AVCodecContext::flags
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:463
Pulse::amp
int amp[4]
Definition: aac.h:229
Pulse::pos
int pos[4]
Definition: aac.h:228
put_pce
static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
Definition: aacenc.c:53
ff_psy_end
av_cold void ff_psy_end(FFPsyContext *ctx)
Cleanup model context at the end.
Definition: psymodel.c:83
Pulse::start
int start
Definition: aac.h:227
scale
static av_always_inline float scale(float x, float s)
Definition: vf_v360.c:1388
fabsf
static __device__ float fabsf(float a)
Definition: cuda_runtime.h:181
ff_af_queue_add
int ff_af_queue_add(AudioFrameQueue *afq, const AVFrame *f)
Add a frame to the queue.
Definition: audio_frame_queue.c:44
IndividualChannelStream::prediction_used
uint8_t prediction_used[41]
Definition: aac.h:191
AACPCEInfo::num_ele
int num_ele[4]
front, side, back, lfe
Definition: aacenc.h:98
SingleChannelElement::ics
IndividualChannelStream ics
Definition: aac.h:250
FFPsyWindowInfo
windowing related information
Definition: psymodel.h:77
adjust_frame_information
static void adjust_frame_information(ChannelElement *cpe, int chans)
Produce integer coefficients from scalefactors provided by the model.
Definition: aacenc.c:256
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
FF_ARRAY_ELEMS
#define FF_ARRAY_ELEMS(a)
Definition: sinewin_tablegen.c:29
av_cold
#define av_cold
Definition: attributes.h:90
IndividualChannelStream::clip_avoidance_factor
float clip_avoidance_factor
set if any window is near clipping to the necessary atennuation factor to avoid it
Definition: aac.h:193
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:485
NOISE_BT
@ NOISE_BT
Spectral data are scaled white noise not coded in the bitstream.
Definition: aac.h:88
s
#define s(width, name)
Definition: cbs_vp9.c:257
SingleChannelElement::coeffs
INTFLOAT coeffs[1024]
coefficients for IMDCT, maybe processed
Definition: aac.h:263
AVCodecContext::global_quality
int global_quality
Global quality for codecs which cannot change it per frame.
Definition: avcodec.h:449
IndividualChannelStream::swb_sizes
const uint8_t * swb_sizes
table of scalefactor band sizes for a particular window
Definition: aac.h:183
g
const char * g
Definition: vf_curves.c:117
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
EIGHT_SHORT_SEQUENCE
@ EIGHT_SHORT_SEQUENCE
Definition: aac.h:79
info
MIPS optimizations info
Definition: mips.txt:2
INTENSITY_BT2
@ INTENSITY_BT2
Scalefactor data are intensity stereo positions (out of phase).
Definition: aac.h:89
IndividualChannelStream::group_len
uint8_t group_len[8]
Definition: aac.h:180
FF_PROFILE_UNKNOWN
#define FF_PROFILE_UNKNOWN
Definition: avcodec.h:1526
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
aac_normal_chan_layouts
static const int64_t aac_normal_chan_layouts[7]
Definition: aacenctab.h:48
alloc_buffers
static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
Definition: aacenc.c:937
ff_aac_swb_size_128_len
const int ff_aac_swb_size_128_len
Definition: aacenctab.c:107
channels
channels
Definition: aptx.h:33
LongTermPrediction::present
int8_t present
Definition: aac.h:165
ff_put_string
void ff_put_string(PutBitContext *pb, const char *string, int terminate_string)
Put the string string in the bitstream.
Definition: bitstream.c:59
IndividualChannelStream
Individual Channel Stream.
Definition: aac.h:175
SCALE_DIFF_ZERO
#define SCALE_DIFF_ZERO
codebook index corresponding to zero scalefactor indices difference
Definition: aac.h:153
NAN
#define NAN
Definition: mathematics.h:64
f
#define f(width, name)
Definition: cbs_vp9.c:255
NOISE_PRE
#define NOISE_PRE
preamble for NOISE_BT, put in bitstream with the first noise band
Definition: aac.h:157
PutBitContext
Definition: put_bits.h:49
IndividualChannelStream::swb_offset
const uint16_t * swb_offset
table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular wind...
Definition: aac.h:182
ff_aac_tableinit
void ff_aac_tableinit(void)
Definition: aactab.c:3346
aac_chan_maps
static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS]
Table to remap channels from libavcodec's default order to AAC order.
Definition: aacenctab.h:73
if
if(ret)
Definition: filter_design.txt:179
AVCodecDefault
Definition: internal.h:215
INTENSITY_BT
@ INTENSITY_BT
Scalefactor data are intensity stereo positions (in phase).
Definition: aac.h:90
FFPsyWindowInfo::window_type
int window_type[3]
window type (short/long/transitional, etc.) - current, previous and next
Definition: psymodel.h:78
AAC_MAX_CHANNELS
#define AAC_MAX_CHANNELS
Definition: aacenctab.h:40
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
aac_pce_configs
static const AACPCEInfo aac_pce_configs[]
List of PCE (Program Configuration Element) for the channel layouts listed in channel_layout....
Definition: aacenc.h:140
fabs
static __device__ float fabs(float a)
Definition: cuda_runtime.h:182
ChannelElement::is_mask
uint8_t is_mask[128]
Set if intensity stereo is used (used by encoder)
Definition: aac.h:283
NULL
#define NULL
Definition: coverity.c:32
sizes
static const int sizes[][2]
Definition: img2dec.c:53
ff_aac_swb_size_1024_len
const int ff_aac_swb_size_1024_len
Definition: aacenctab.c:108
encode_pulses
static void encode_pulses(AACEncContext *s, Pulse *pulse)
Encode pulse data.
Definition: aacenc.c:420
SingleChannelElement::is_ener
float is_ener[128]
Intensity stereo pos (used by encoder)
Definition: aac.h:260
ff_aac_num_swb_128
const uint8_t ff_aac_num_swb_128[]
Definition: aactab.c:79
AVCodecContext::bit_rate
int64_t bit_rate
the average bitrate
Definition: avcodec.h:433
av_clipf
#define av_clipf
Definition: common.h:144
av_default_item_name
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
profiles.h
aac.h
aactab.h
IndividualChannelStream::predictor_present
int predictor_present
Definition: aac.h:187
FFPsyWindowInfo::grouping
int grouping[8]
window grouping (for e.g. AAC)
Definition: psymodel.h:81
TNS_MAX_ORDER
#define TNS_MAX_ORDER
Definition: aac.h:51
SingleChannelElement::sf_idx
int sf_idx[128]
scalefactor indices (used by encoder)
Definition: aac.h:257
float_dsp.h
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
AV_CODEC_ID_AAC
@ AV_CODEC_ID_AAC
Definition: codec_id.h:425
aac_encode_frame
static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, const AVFrame *frame, int *got_packet_ptr)
Definition: aacenc.c:556
ff_aac_scalefactor_bits
const uint8_t ff_aac_scalefactor_bits[121]
Definition: aactab.c:110
AACPCEInfo
Definition: aacenc.h:96
FFPsyWindowInfo::clipping
float clipping[8]
maximum absolute normalized intensity in the given window for clip avoidance
Definition: psymodel.h:82
options
const OptionDef options[]
SingleChannelElement::lcoeffs
AAC_FLOAT lcoeffs[1024]
MDCT of LTP coefficients (used by encoder)
Definition: aac.h:267
AAC_CODER_ANMR
@ AAC_CODER_ANMR
Definition: aacenc.h:39
ChannelElement::ch
SingleChannelElement ch[2]
Definition: aac.h:285
ff_swb_offset_1024
const uint16_t *const ff_swb_offset_1024[]
Definition: aactab.c:1354
AVPacket::size
int size
Definition: packet.h:374
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
ONLY_LONG_SEQUENCE
@ ONLY_LONG_SEQUENCE
Definition: aac.h:77
TYPE_END
@ TYPE_END
Definition: aac.h:64
quantize_bands
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
ff_aac_float_common_init
void ff_aac_float_common_init(void)
FF_PROFILE_AAC_LOW
#define FF_PROFILE_AAC_LOW
Definition: avcodec.h:1530
encode_scale_factors
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce)
Encode scalefactors.
Definition: aacenc.c:384
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
ff_mdct_end
#define ff_mdct_end
Definition: fft.h:154
apply_window_and_mdct
static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, float *audio)
Definition: aacenc.c:200
AVFloatDSPContext
Definition: float_dsp.h:24
AAC_CODER_TWOLOOP
@ AAC_CODER_TWOLOOP
Definition: aacenc.h:40
ff_aac_coder_init_mips
void ff_aac_coder_init_mips(AACEncContext *c)
Definition: aaccoder_mips.c:2484
ChannelElement::common_window
int common_window
Set if channels share a common 'IndividualChannelStream' in bitstream.
Definition: aac.h:279
sinewin.h
apply_intensity_stereo
static void apply_intensity_stereo(ChannelElement *cpe)
Definition: aacenc.c:306
AACPCEInfo::index
int index[4][8]
front, side, back, lfe
Definition: aacenc.h:100
ChannelElement::ms_mask
uint8_t ms_mask[128]
Set if mid/side stereo is used for each scalefactor window band.
Definition: aac.h:282
ff_lpc_end
av_cold void ff_lpc_end(LPCContext *s)
Uninitialize LPCContext.
Definition: lpc.c:323
AV_LOG_INFO
#define AV_LOG_INFO
Standard information.
Definition: log.h:191
AVCodecContext::channels
int channels
number of audio channels
Definition: avcodec.h:993
ff_psy_init
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
SingleChannelElement
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:249
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:271
AVPacket::pts
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:366
put_bits_count
static int put_bits_count(PutBitContext *s)
Definition: put_bits.h:79
IndividualChannelStream::num_windows
int num_windows
Definition: aac.h:185
AVCodecContext::extradata
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:484
FF_CODEC_CAP_INIT_CLEANUP
#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:50
aacenc_options
static const AVOption aacenc_options[]
Definition: aacenc.c:1108
LONG_STOP_SEQUENCE
@ LONG_STOP_SEQUENCE
Definition: aac.h:80
ChannelElement
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:276
apply_window
static void(*const apply_window[4])(AVFloatDSPContext *fdsp, SingleChannelElement *sce, const float *audio)
Definition: aacenc.c:191
AACPCEInfo::pairing
int pairing[3][8]
front, side, back
Definition: aacenc.h:99
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
NOISE_PRE_BITS
#define NOISE_PRE_BITS
length of preamble
Definition: aac.h:158
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
TYPE_LFE
@ TYPE_LFE
Definition: aac.h:60
ff_psy_preprocess_init
av_cold struct FFPsyPreprocessContext * ff_psy_preprocess_init(AVCodecContext *avctx)
psychoacoustic model audio preprocessing initialization
Definition: psymodel.c:103
ff_aac_kbd_long_1024
float ff_aac_kbd_long_1024[1024]
av_mallocz
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:263
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:209
ff_aac_dsp_init_x86
void ff_aac_dsp_init_x86(AACEncContext *s)
Definition: aacencdsp_init.c:35
AACENC_FLAGS
#define AACENC_FLAGS
Definition: aacenc.c:1107
IndividualChannelStream::tns_max_bands
int tns_max_bands
Definition: aac.h:186
avcodec.h
tag
uint32_t tag
Definition: movenc.c:1596
ret
ret
Definition: filter_design.txt:187
ff_aac_num_swb_1024
const uint8_t ff_aac_num_swb_1024[]
Definition: aactab.c:63
AVClass::class_name
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:71
frame
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
Definition: filter_design.txt:264
AVCodecContext::strict_std_compliance
int strict_std_compliance
strictly follow the standard (MPEG-4, ...).
Definition: avcodec.h:1280
encode_ms_info
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
Encode MS data.
Definition: aacenc.c:242
RESERVED_BT
@ RESERVED_BT
Band types following are encoded differently from others.
Definition: aac.h:87
LONG_START_SEQUENCE
@ LONG_START_SEQUENCE
Definition: aac.h:78
ff_psy_preprocess
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
CLIP_AVOIDANCE_FACTOR
#define CLIP_AVOIDANCE_FACTOR
Definition: aac.h:54
SingleChannelElement::tns
TemporalNoiseShaping tns
Definition: aac.h:251
AACEncContext
AAC encoder context.
Definition: aacenc.h:379
L
#define L(x)
Definition: vp56_arith.h:36
AVCodecContext
main external API structure.
Definition: avcodec.h:383
channel_layout.h
encode_individual_channel
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, SingleChannelElement *sce, int common_window)
Encode one channel of audio data.
Definition: aacenc.c:487
NOISE_OFFSET
#define NOISE_OFFSET
subtracted from global gain, used as offset for the preamble
Definition: aac.h:159
ERROR_IF
#define ERROR_IF(cond,...)
Definition: aacenc_utils.h:268
ff_aac_swb_size_1024
const uint8_t *const ff_aac_swb_size_1024[]
Definition: aacenctab.c:99
IndividualChannelStream::window_sequence
enum WindowSequence window_sequence[2]
Definition: aac.h:177
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:224
TemporalNoiseShaping
Temporal Noise Shaping.
Definition: aac.h:199
AVCodecContext::profile
int profile
profile
Definition: avcodec.h:1525
AOT_SBR
@ AOT_SBR
Y Spectral Band Replication.
Definition: mpeg4audio.h:80
put_audio_specific_config
static int put_audio_specific_config(AVCodecContext *avctx)
Make AAC audio config object.
Definition: aacenc.c:94
AV_CODEC_CAP_DELAY
#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:82
samples
Filter the word "frame" indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
ChannelElement::is_mode
uint8_t is_mode
Set if any bands have been encoded using intensity stereo (used by encoder)
Definition: aac.h:281
put_ics_info
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
Encode ics_info element.
Definition: aacenc.c:221
ff_mpeg4audio_sample_rates
const int ff_mpeg4audio_sample_rates[16]
Definition: mpeg4audio_sample_rates.h:26
ff_aac_swb_size_128
const uint8_t *const ff_aac_swb_size_128[]
Definition: aacenctab.c:91
aac_encode_defaults
static const AVCodecDefault aac_encode_defaults[]
Definition: aacenc.c:1131
AV_CODEC_FLAG_BITEXACT
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:272
aac_encode_end
static av_cold int aac_encode_end(AVCodecContext *avctx)
Definition: aacenc.c:899
AVCodecContext::frame_number
int frame_number
Frame counter, set by libavcodec.
Definition: avcodec.h:1023
flush_put_bits
static void flush_put_bits(PutBitContext *s)
Pad the end of the output stream with zeros.
Definition: put_bits.h:142
diff
static av_always_inline int diff(const uint32_t a, const uint32_t b)
Definition: vf_palettegen.c:139
FF_AAC_PROFILE_OPTS
#define FF_AAC_PROFILE_OPTS
Definition: profiles.h:28
AVPacket
This structure stores compressed data.
Definition: packet.h:350
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:410
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:241
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
avpriv_float_dsp_alloc
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
IndividualChannelStream::window_clipping
uint8_t window_clipping[8]
set if a certain window is near clipping
Definition: aac.h:192
IndividualChannelStream::max_sfb
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:176
Pulse
Definition: aac.h:225
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
SingleChannelElement::ltp_state
INTFLOAT ltp_state[3072]
time signal for LTP
Definition: aac.h:266
dsp_init
static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
Definition: aacenc.c:918
IndividualChannelStream::ltp
LongTermPrediction ltp
Definition: aac.h:181
ff_psy_preprocess_end
av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
Cleanup audio preprocessing module.
Definition: psymodel.c:152
aacenc_utils.h
AV_CODEC_CAP_SMALL_LAST_FRAME
#define AV_CODEC_CAP_SMALL_LAST_FRAME
Codec can be fed a final frame with a smaller size.
Definition: codec.h:87
put_bits.h
SingleChannelElement::band_type
enum BandType band_type[128]
band types
Definition: aac.h:253
psymodel.h
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:233
IndividualChannelStream::use_kb_window
uint8_t use_kb_window[2]
If set, use Kaiser-Bessel window, otherwise use a sine window.
Definition: aac.h:178
ff_alloc_packet
int ff_alloc_packet(AVCodecContext *avctx, AVPacket *avpkt, int64_t size)
Check AVPacket size and allocate data.
Definition: encode.c:34
FF_LPC_TYPE_LEVINSON
@ FF_LPC_TYPE_LEVINSON
Levinson-Durbin recursion.
Definition: lpc.h:47
ff_aac_encoder
const AVCodec ff_aac_encoder
Definition: aacenc.c:1136
FFPsyWindowInfo::num_windows
int num_windows
number of windows in a frame
Definition: psymodel.h:80
ff_lpc_init
av_cold int ff_lpc_init(LPCContext *s, int blocksize, int max_order, enum FFLPCType lpc_type)
Initialize LPCContext.
Definition: lpc.c:301
ff_aac_scalefactor_code
const uint32_t ff_aac_scalefactor_code[121]
Definition: aactab.c:91
ff_quantize_band_cost_cache_init
void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
Definition: aacenc.c:126
aacenc.h
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