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aacenc.c
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1 /*
2  * AAC encoder
3  * Copyright (C) 2008 Konstantin Shishkov
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * AAC encoder
25  */
26 
27 /***********************************
28  * TODOs:
29  * add sane pulse detection
30  * add temporal noise shaping
31  ***********************************/
32 
33 #include "libavutil/float_dsp.h"
34 #include "libavutil/opt.h"
35 #include "avcodec.h"
36 #include "put_bits.h"
37 #include "internal.h"
38 #include "mpeg4audio.h"
39 #include "kbdwin.h"
40 #include "sinewin.h"
41 
42 #include "aac.h"
43 #include "aactab.h"
44 #include "aacenc.h"
45 
46 #include "psymodel.h"
47 
48 #define AAC_MAX_CHANNELS 6
49 
50 #define ERROR_IF(cond, ...) \
51  if (cond) { \
52  av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
53  return AVERROR(EINVAL); \
54  }
55 
56 float ff_aac_pow34sf_tab[428];
57 
58 static const uint8_t swb_size_1024_96[] = {
59  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
60  12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
61  64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
62 };
63 
64 static const uint8_t swb_size_1024_64[] = {
65  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
66  12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
67  40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
68 };
69 
70 static const uint8_t swb_size_1024_48[] = {
71  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
72  12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
73  32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
74  96
75 };
76 
77 static const uint8_t swb_size_1024_32[] = {
78  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
79  12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
80  32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
81 };
82 
83 static const uint8_t swb_size_1024_24[] = {
84  4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
85  12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
86  32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
87 };
88 
89 static const uint8_t swb_size_1024_16[] = {
90  8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
91  12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
92  32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
93 };
94 
95 static const uint8_t swb_size_1024_8[] = {
96  12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
97  16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
98  32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
99 };
100 
101 static const uint8_t *swb_size_1024[] = {
106 };
107 
108 static const uint8_t swb_size_128_96[] = {
109  4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
110 };
111 
112 static const uint8_t swb_size_128_48[] = {
113  4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
114 };
115 
116 static const uint8_t swb_size_128_24[] = {
117  4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
118 };
119 
120 static const uint8_t swb_size_128_16[] = {
121  4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
122 };
123 
124 static const uint8_t swb_size_128_8[] = {
125  4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
126 };
127 
128 static const uint8_t *swb_size_128[] = {
129  /* the last entry on the following row is swb_size_128_64 but is a
130  duplicate of swb_size_128_96 */
135 };
136 
137 /** default channel configurations */
138 static const uint8_t aac_chan_configs[6][5] = {
139  {1, TYPE_SCE}, // 1 channel - single channel element
140  {1, TYPE_CPE}, // 2 channels - channel pair
141  {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
142  {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
143  {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
144  {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
145 };
146 
147 /**
148  * Table to remap channels from libavcodec's default order to AAC order.
149  */
151  { 0 },
152  { 0, 1 },
153  { 2, 0, 1 },
154  { 2, 0, 1, 3 },
155  { 2, 0, 1, 3, 4 },
156  { 2, 0, 1, 4, 5, 3 },
157 };
158 
159 /**
160  * Make AAC audio config object.
161  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
162  */
164 {
165  PutBitContext pb;
166  AACEncContext *s = avctx->priv_data;
167 
168  init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
169  put_bits(&pb, 5, 2); //object type - AAC-LC
170  put_bits(&pb, 4, s->samplerate_index); //sample rate index
171  put_bits(&pb, 4, s->channels);
172  //GASpecificConfig
173  put_bits(&pb, 1, 0); //frame length - 1024 samples
174  put_bits(&pb, 1, 0); //does not depend on core coder
175  put_bits(&pb, 1, 0); //is not extension
176 
177  //Explicitly Mark SBR absent
178  put_bits(&pb, 11, 0x2b7); //sync extension
179  put_bits(&pb, 5, AOT_SBR);
180  put_bits(&pb, 1, 0);
181  flush_put_bits(&pb);
182 }
183 
184 #define WINDOW_FUNC(type) \
185 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
186  SingleChannelElement *sce, \
187  const float *audio)
188 
189 WINDOW_FUNC(only_long)
190 {
191  const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
192  const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
193  float *out = sce->ret_buf;
194 
195  fdsp->vector_fmul (out, audio, lwindow, 1024);
196  fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
197 }
198 
199 WINDOW_FUNC(long_start)
200 {
201  const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
202  const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
203  float *out = sce->ret_buf;
204 
205  fdsp->vector_fmul(out, audio, lwindow, 1024);
206  memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
207  fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
208  memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
209 }
210 
211 WINDOW_FUNC(long_stop)
212 {
213  const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
214  const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
215  float *out = sce->ret_buf;
216 
217  memset(out, 0, sizeof(out[0]) * 448);
218  fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
219  memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
220  fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
221 }
222 
223 WINDOW_FUNC(eight_short)
224 {
225  const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
226  const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
227  const float *in = audio + 448;
228  float *out = sce->ret_buf;
229  int w;
230 
231  for (w = 0; w < 8; w++) {
232  fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
233  out += 128;
234  in += 128;
235  fdsp->vector_fmul_reverse(out, in, swindow, 128);
236  out += 128;
237  }
238 }
239 
240 static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
242  const float *audio) = {
243  [ONLY_LONG_SEQUENCE] = apply_only_long_window,
244  [LONG_START_SEQUENCE] = apply_long_start_window,
245  [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
246  [LONG_STOP_SEQUENCE] = apply_long_stop_window
247 };
248 
250  float *audio)
251 {
252  int i;
253  float *output = sce->ret_buf;
254 
255  apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
256 
258  s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
259  else
260  for (i = 0; i < 1024; i += 128)
261  s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
262  memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
263 }
264 
265 /**
266  * Encode ics_info element.
267  * @see Table 4.6 (syntax of ics_info)
268  */
270 {
271  int w;
272 
273  put_bits(&s->pb, 1, 0); // ics_reserved bit
274  put_bits(&s->pb, 2, info->window_sequence[0]);
275  put_bits(&s->pb, 1, info->use_kb_window[0]);
276  if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
277  put_bits(&s->pb, 6, info->max_sfb);
278  put_bits(&s->pb, 1, 0); // no prediction
279  } else {
280  put_bits(&s->pb, 4, info->max_sfb);
281  for (w = 1; w < 8; w++)
282  put_bits(&s->pb, 1, !info->group_len[w]);
283  }
284 }
285 
286 /**
287  * Encode MS data.
288  * @see 4.6.8.1 "Joint Coding - M/S Stereo"
289  */
291 {
292  int i, w;
293 
294  put_bits(pb, 2, cpe->ms_mode);
295  if (cpe->ms_mode == 1)
296  for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
297  for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
298  put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
299 }
300 
301 /**
302  * Produce integer coefficients from scalefactors provided by the model.
303  */
304 static void adjust_frame_information(ChannelElement *cpe, int chans)
305 {
306  int i, w, w2, g, ch;
307  int start, maxsfb, cmaxsfb;
308 
309  for (ch = 0; ch < chans; ch++) {
310  IndividualChannelStream *ics = &cpe->ch[ch].ics;
311  start = 0;
312  maxsfb = 0;
313  cpe->ch[ch].pulse.num_pulse = 0;
314  for (w = 0; w < ics->num_windows*16; w += 16) {
315  for (g = 0; g < ics->num_swb; g++) {
316  //apply M/S
317  if (cpe->common_window && !ch && cpe->ms_mask[w + g]) {
318  for (i = 0; i < ics->swb_sizes[g]; i++) {
319  cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
320  cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
321  }
322  }
323  start += ics->swb_sizes[g];
324  }
325  for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
326  ;
327  maxsfb = FFMAX(maxsfb, cmaxsfb);
328  }
329  ics->max_sfb = maxsfb;
330 
331  //adjust zero bands for window groups
332  for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
333  for (g = 0; g < ics->max_sfb; g++) {
334  i = 1;
335  for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
336  if (!cpe->ch[ch].zeroes[w2*16 + g]) {
337  i = 0;
338  break;
339  }
340  }
341  cpe->ch[ch].zeroes[w*16 + g] = i;
342  }
343  }
344  }
345 
346  if (chans > 1 && cpe->common_window) {
347  IndividualChannelStream *ics0 = &cpe->ch[0].ics;
348  IndividualChannelStream *ics1 = &cpe->ch[1].ics;
349  int msc = 0;
350  ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
351  ics1->max_sfb = ics0->max_sfb;
352  for (w = 0; w < ics0->num_windows*16; w += 16)
353  for (i = 0; i < ics0->max_sfb; i++)
354  if (cpe->ms_mask[w+i])
355  msc++;
356  if (msc == 0 || ics0->max_sfb == 0)
357  cpe->ms_mode = 0;
358  else
359  cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
360  }
361 }
362 
363 /**
364  * Encode scalefactor band coding type.
365  */
367 {
368  int w;
369 
370  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
371  s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
372 }
373 
374 /**
375  * Encode scalefactors.
376  */
379 {
380  int off = sce->sf_idx[0], diff;
381  int i, w;
382 
383  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
384  for (i = 0; i < sce->ics.max_sfb; i++) {
385  if (!sce->zeroes[w*16 + i]) {
386  diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
387  av_assert0(diff >= 0 && diff <= 120);
388  off = sce->sf_idx[w*16 + i];
390  }
391  }
392  }
393 }
394 
395 /**
396  * Encode pulse data.
397  */
398 static void encode_pulses(AACEncContext *s, Pulse *pulse)
399 {
400  int i;
401 
402  put_bits(&s->pb, 1, !!pulse->num_pulse);
403  if (!pulse->num_pulse)
404  return;
405 
406  put_bits(&s->pb, 2, pulse->num_pulse - 1);
407  put_bits(&s->pb, 6, pulse->start);
408  for (i = 0; i < pulse->num_pulse; i++) {
409  put_bits(&s->pb, 5, pulse->pos[i]);
410  put_bits(&s->pb, 4, pulse->amp[i]);
411  }
412 }
413 
414 /**
415  * Encode spectral coefficients processed by psychoacoustic model.
416  */
418 {
419  int start, i, w, w2;
420 
421  for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
422  start = 0;
423  for (i = 0; i < sce->ics.max_sfb; i++) {
424  if (sce->zeroes[w*16 + i]) {
425  start += sce->ics.swb_sizes[i];
426  continue;
427  }
428  for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
429  s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
430  sce->ics.swb_sizes[i],
431  sce->sf_idx[w*16 + i],
432  sce->band_type[w*16 + i],
433  s->lambda);
434  start += sce->ics.swb_sizes[i];
435  }
436  }
437 }
438 
439 /**
440  * Encode one channel of audio data.
441  */
444  int common_window)
445 {
446  put_bits(&s->pb, 8, sce->sf_idx[0]);
447  if (!common_window)
448  put_ics_info(s, &sce->ics);
449  encode_band_info(s, sce);
450  encode_scale_factors(avctx, s, sce);
451  encode_pulses(s, &sce->pulse);
452  put_bits(&s->pb, 1, 0); //tns
453  put_bits(&s->pb, 1, 0); //ssr
454  encode_spectral_coeffs(s, sce);
455  return 0;
456 }
457 
458 /**
459  * Write some auxiliary information about the created AAC file.
460  */
461 static void put_bitstream_info(AACEncContext *s, const char *name)
462 {
463  int i, namelen, padbits;
464 
465  namelen = strlen(name) + 2;
466  put_bits(&s->pb, 3, TYPE_FIL);
467  put_bits(&s->pb, 4, FFMIN(namelen, 15));
468  if (namelen >= 15)
469  put_bits(&s->pb, 8, namelen - 14);
470  put_bits(&s->pb, 4, 0); //extension type - filler
471  padbits = -put_bits_count(&s->pb) & 7;
473  for (i = 0; i < namelen - 2; i++)
474  put_bits(&s->pb, 8, name[i]);
475  put_bits(&s->pb, 12 - padbits, 0);
476 }
477 
478 /*
479  * Copy input samples.
480  * Channels are reordered from libavcodec's default order to AAC order.
481  */
483 {
484  int ch;
485  int end = 2048 + (frame ? frame->nb_samples : 0);
486  const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
487 
488  /* copy and remap input samples */
489  for (ch = 0; ch < s->channels; ch++) {
490  /* copy last 1024 samples of previous frame to the start of the current frame */
491  memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
492 
493  /* copy new samples and zero any remaining samples */
494  if (frame) {
495  memcpy(&s->planar_samples[ch][2048],
496  frame->extended_data[channel_map[ch]],
497  frame->nb_samples * sizeof(s->planar_samples[0][0]));
498  }
499  memset(&s->planar_samples[ch][end], 0,
500  (3072 - end) * sizeof(s->planar_samples[0][0]));
501  }
502 }
503 
504 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
505  const AVFrame *frame, int *got_packet_ptr)
506 {
507  AACEncContext *s = avctx->priv_data;
508  float **samples = s->planar_samples, *samples2, *la, *overlap;
509  ChannelElement *cpe;
510  int i, ch, w, g, chans, tag, start_ch, ret;
511  int chan_el_counter[4];
513 
514  if (s->last_frame == 2)
515  return 0;
516 
517  /* add current frame to queue */
518  if (frame) {
519  if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
520  return ret;
521  }
522 
523  copy_input_samples(s, frame);
524  if (s->psypp)
526 
527  if (!avctx->frame_number)
528  return 0;
529 
530  start_ch = 0;
531  for (i = 0; i < s->chan_map[0]; i++) {
532  FFPsyWindowInfo* wi = windows + start_ch;
533  tag = s->chan_map[i+1];
534  chans = tag == TYPE_CPE ? 2 : 1;
535  cpe = &s->cpe[i];
536  for (ch = 0; ch < chans; ch++) {
537  IndividualChannelStream *ics = &cpe->ch[ch].ics;
538  int cur_channel = start_ch + ch;
539  overlap = &samples[cur_channel][0];
540  samples2 = overlap + 1024;
541  la = samples2 + (448+64);
542  if (!frame)
543  la = NULL;
544  if (tag == TYPE_LFE) {
545  wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
546  wi[ch].window_shape = 0;
547  wi[ch].num_windows = 1;
548  wi[ch].grouping[0] = 1;
549 
550  /* Only the lowest 12 coefficients are used in a LFE channel.
551  * The expression below results in only the bottom 8 coefficients
552  * being used for 11.025kHz to 16kHz sample rates.
553  */
554  ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
555  } else {
556  wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
557  ics->window_sequence[0]);
558  }
559  ics->window_sequence[1] = ics->window_sequence[0];
560  ics->window_sequence[0] = wi[ch].window_type[0];
561  ics->use_kb_window[1] = ics->use_kb_window[0];
562  ics->use_kb_window[0] = wi[ch].window_shape;
563  ics->num_windows = wi[ch].num_windows;
564  ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
565  ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
566  for (w = 0; w < ics->num_windows; w++)
567  ics->group_len[w] = wi[ch].grouping[w];
568 
569  apply_window_and_mdct(s, &cpe->ch[ch], overlap);
570  if (isnan(cpe->ch->coeffs[0])) {
571  av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n");
572  return AVERROR(EINVAL);
573  }
574  }
575  start_ch += chans;
576  }
577  if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0)
578  return ret;
579  do {
580  int frame_bits;
581 
582  init_put_bits(&s->pb, avpkt->data, avpkt->size);
583 
584  if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
586  start_ch = 0;
587  memset(chan_el_counter, 0, sizeof(chan_el_counter));
588  for (i = 0; i < s->chan_map[0]; i++) {
589  FFPsyWindowInfo* wi = windows + start_ch;
590  const float *coeffs[2];
591  tag = s->chan_map[i+1];
592  chans = tag == TYPE_CPE ? 2 : 1;
593  cpe = &s->cpe[i];
594  put_bits(&s->pb, 3, tag);
595  put_bits(&s->pb, 4, chan_el_counter[tag]++);
596  for (ch = 0; ch < chans; ch++)
597  coeffs[ch] = cpe->ch[ch].coeffs;
598  s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
599  for (ch = 0; ch < chans; ch++) {
600  s->cur_channel = start_ch + ch;
601  s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
602  }
603  cpe->common_window = 0;
604  if (chans > 1
605  && wi[0].window_type[0] == wi[1].window_type[0]
606  && wi[0].window_shape == wi[1].window_shape) {
607 
608  cpe->common_window = 1;
609  for (w = 0; w < wi[0].num_windows; w++) {
610  if (wi[0].grouping[w] != wi[1].grouping[w]) {
611  cpe->common_window = 0;
612  break;
613  }
614  }
615  }
616  s->cur_channel = start_ch;
617  if (s->options.stereo_mode && cpe->common_window) {
618  if (s->options.stereo_mode > 0) {
619  IndividualChannelStream *ics = &cpe->ch[0].ics;
620  for (w = 0; w < ics->num_windows; w += ics->group_len[w])
621  for (g = 0; g < ics->num_swb; g++)
622  cpe->ms_mask[w*16+g] = 1;
623  } else if (s->coder->search_for_ms) {
624  s->coder->search_for_ms(s, cpe, s->lambda);
625  }
626  }
627  adjust_frame_information(cpe, chans);
628  if (chans == 2) {
629  put_bits(&s->pb, 1, cpe->common_window);
630  if (cpe->common_window) {
631  put_ics_info(s, &cpe->ch[0].ics);
632  encode_ms_info(&s->pb, cpe);
633  }
634  }
635  for (ch = 0; ch < chans; ch++) {
636  s->cur_channel = start_ch + ch;
637  encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
638  }
639  start_ch += chans;
640  }
641 
642  frame_bits = put_bits_count(&s->pb);
643  if (frame_bits <= 6144 * s->channels - 3) {
644  s->psy.bitres.bits = frame_bits / s->channels;
645  break;
646  }
647 
648  s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
649 
650  } while (1);
651 
652  put_bits(&s->pb, 3, TYPE_END);
653  flush_put_bits(&s->pb);
654  avctx->frame_bits = put_bits_count(&s->pb);
655 
656  // rate control stuff
657  if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
658  float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
659  s->lambda *= ratio;
660  s->lambda = FFMIN(s->lambda, 65536.f);
661  }
662 
663  if (!frame)
664  s->last_frame++;
665 
666  ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
667  &avpkt->duration);
668 
669  avpkt->size = put_bits_count(&s->pb) >> 3;
670  *got_packet_ptr = 1;
671  return 0;
672 }
673 
675 {
676  AACEncContext *s = avctx->priv_data;
677 
678  ff_mdct_end(&s->mdct1024);
679  ff_mdct_end(&s->mdct128);
680  ff_psy_end(&s->psy);
681  if (s->psypp)
683  av_freep(&s->buffer.samples);
684  av_freep(&s->cpe);
685  av_freep(&s->fdsp);
686  ff_af_queue_close(&s->afq);
687  return 0;
688 }
689 
691 {
692  int ret = 0;
693 
695  if (!s->fdsp)
696  return AVERROR(ENOMEM);
697 
698  // window init
703 
704  if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
705  return ret;
706  if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0))
707  return ret;
708 
709  return 0;
710 }
711 
713 {
714  int ch;
715  FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);
716  FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);
717  FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
718 
719  for(ch = 0; ch < s->channels; ch++)
720  s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
721 
722  return 0;
723 alloc_fail:
724  return AVERROR(ENOMEM);
725 }
726 
728 {
729  AACEncContext *s = avctx->priv_data;
730  int i, ret = 0;
731  const uint8_t *sizes[2];
732  uint8_t grouping[AAC_MAX_CHANNELS];
733  int lengths[2];
734 
735  avctx->frame_size = 1024;
736 
737  for (i = 0; i < 16; i++)
739  break;
740 
741  s->channels = avctx->channels;
742 
743  ERROR_IF(i == 16,
744  "Unsupported sample rate %d\n", avctx->sample_rate);
746  "Unsupported number of channels: %d\n", s->channels);
748  "Unsupported profile %d\n", avctx->profile);
749  ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
750  "Too many bits per frame requested\n");
751 
752  s->samplerate_index = i;
753 
755 
756  if ((ret = dsp_init(avctx, s)) < 0)
757  goto fail;
758 
759  if ((ret = alloc_buffers(avctx, s)) < 0)
760  goto fail;
761 
762  avctx->extradata_size = 5;
764 
765  sizes[0] = swb_size_1024[i];
766  sizes[1] = swb_size_128[i];
767  lengths[0] = ff_aac_num_swb_1024[i];
768  lengths[1] = ff_aac_num_swb_128[i];
769  for (i = 0; i < s->chan_map[0]; i++)
770  grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
771  if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
772  s->chan_map[0], grouping)) < 0)
773  goto fail;
774  s->psypp = ff_psy_preprocess_init(avctx);
776 
777  if (HAVE_MIPSDSPR1)
779 
780  s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
781 
783 
784  for (i = 0; i < 428; i++)
785  ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
786 
787  avctx->initial_padding = 1024;
788  ff_af_queue_init(avctx, &s->afq);
789 
790  return 0;
791 fail:
792  aac_encode_end(avctx);
793  return ret;
794 }
795 
796 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
797 static const AVOption aacenc_options[] = {
798  {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
799  {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
800  {"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
801  {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
802  {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},
803  {"faac", "FAAC-inspired method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
804  {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
805  {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
806  {"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
807  {NULL}
808 };
809 
810 static const AVClass aacenc_class = {
811  "AAC encoder",
815 };
816 
817 /* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
818  * failures */
819 static const int mpeg4audio_sample_rates[16] = {
820  96000, 88200, 64000, 48000, 44100, 32000,
821  24000, 22050, 16000, 12000, 11025, 8000, 7350
822 };
823 
825  .name = "aac",
826  .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
827  .type = AVMEDIA_TYPE_AUDIO,
828  .id = AV_CODEC_ID_AAC,
829  .priv_data_size = sizeof(AACEncContext),
831  .encode2 = aac_encode_frame,
834  .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
836  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
838  .priv_class = &aacenc_class,
839 };