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mpegaudioenc.c
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1 /*
2  * The simplest mpeg audio layer 2 encoder
3  * Copyright (c) 2000, 2001 Fabrice Bellard
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  * The simplest mpeg audio layer 2 encoder.
25  */
26 
28 
29 #include "avcodec.h"
30 #include "internal.h"
31 #include "put_bits.h"
32 
33 #define FRAC_BITS 15 /* fractional bits for sb_samples and dct */
34 #define WFRAC_BITS 14 /* fractional bits for window */
35 
36 #include "mpegaudio.h"
37 #include "mpegaudiodsp.h"
38 
39 /* currently, cannot change these constants (need to modify
40  quantization stage) */
41 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
42 
43 #define SAMPLES_BUF_SIZE 4096
44 
45 typedef struct MpegAudioContext {
48  int lsf; /* 1 if mpeg2 low bitrate selected */
49  int bitrate_index; /* bit rate */
51  int frame_size; /* frame size, in bits, without padding */
52  /* padding computation */
54  short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
55  int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
57  unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
58  /* code to group 3 scale factors */
60  int sblimit; /* number of used subbands */
61  const unsigned char *alloc_table;
63 
64 /* define it to use floats in quantization (I don't like floats !) */
65 #define USE_FLOATS
66 
67 #include "mpegaudiodata.h"
68 #include "mpegaudiotab.h"
69 
71 {
72  MpegAudioContext *s = avctx->priv_data;
73  int freq = avctx->sample_rate;
74  int bitrate = avctx->bit_rate;
75  int channels = avctx->channels;
76  int i, v, table;
77  float a;
78 
79  if (channels <= 0 || channels > 2){
80  av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
81  return AVERROR(EINVAL);
82  }
83  bitrate = bitrate / 1000;
84  s->nb_channels = channels;
85  avctx->frame_size = MPA_FRAME_SIZE;
86  avctx->delay = 512 - 32 + 1;
87 
88  /* encoding freq */
89  s->lsf = 0;
90  for(i=0;i<3;i++) {
91  if (avpriv_mpa_freq_tab[i] == freq)
92  break;
93  if ((avpriv_mpa_freq_tab[i] / 2) == freq) {
94  s->lsf = 1;
95  break;
96  }
97  }
98  if (i == 3){
99  av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
100  return AVERROR(EINVAL);
101  }
102  s->freq_index = i;
103 
104  /* encoding bitrate & frequency */
105  for(i=0;i<15;i++) {
106  if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
107  break;
108  }
109  if (i == 15){
110  av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
111  return AVERROR(EINVAL);
112  }
113  s->bitrate_index = i;
114 
115  /* compute total header size & pad bit */
116 
117  a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
118  s->frame_size = ((int)a) * 8;
119 
120  /* frame fractional size to compute padding */
121  s->frame_frac = 0;
122  s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
123 
124  /* select the right allocation table */
125  table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
126 
127  /* number of used subbands */
130 
131  av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
132  bitrate, freq, s->frame_size, table, s->frame_frac_incr);
133 
134  for(i=0;i<s->nb_channels;i++)
135  s->samples_offset[i] = 0;
136 
137  for(i=0;i<257;i++) {
138  int v;
139  v = ff_mpa_enwindow[i];
140 #if WFRAC_BITS != 16
141  v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
142 #endif
143  filter_bank[i] = v;
144  if ((i & 63) != 0)
145  v = -v;
146  if (i != 0)
147  filter_bank[512 - i] = v;
148  }
149 
150  for(i=0;i<64;i++) {
151  v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
152  if (v <= 0)
153  v = 1;
154  scale_factor_table[i] = v;
155 #ifdef USE_FLOATS
156  scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
157 #else
158 #define P 15
159  scale_factor_shift[i] = 21 - P - (i / 3);
160  scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);
161 #endif
162  }
163  for(i=0;i<128;i++) {
164  v = i - 64;
165  if (v <= -3)
166  v = 0;
167  else if (v < 0)
168  v = 1;
169  else if (v == 0)
170  v = 2;
171  else if (v < 3)
172  v = 3;
173  else
174  v = 4;
175  scale_diff_table[i] = v;
176  }
177 
178  for(i=0;i<17;i++) {
179  v = ff_mpa_quant_bits[i];
180  if (v < 0)
181  v = -v;
182  else
183  v = v * 3;
184  total_quant_bits[i] = 12 * v;
185  }
186 
187  return 0;
188 }
189 
190 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
191 static void idct32(int *out, int *tab)
192 {
193  int i, j;
194  int *t, *t1, xr;
195  const int *xp = costab32;
196 
197  for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
198 
199  t = tab + 30;
200  t1 = tab + 2;
201  do {
202  t[0] += t[-4];
203  t[1] += t[1 - 4];
204  t -= 4;
205  } while (t != t1);
206 
207  t = tab + 28;
208  t1 = tab + 4;
209  do {
210  t[0] += t[-8];
211  t[1] += t[1-8];
212  t[2] += t[2-8];
213  t[3] += t[3-8];
214  t -= 8;
215  } while (t != t1);
216 
217  t = tab;
218  t1 = tab + 32;
219  do {
220  t[ 3] = -t[ 3];
221  t[ 6] = -t[ 6];
222 
223  t[11] = -t[11];
224  t[12] = -t[12];
225  t[13] = -t[13];
226  t[15] = -t[15];
227  t += 16;
228  } while (t != t1);
229 
230 
231  t = tab;
232  t1 = tab + 8;
233  do {
234  int x1, x2, x3, x4;
235 
236  x3 = MUL(t[16], FIX(SQRT2*0.5));
237  x4 = t[0] - x3;
238  x3 = t[0] + x3;
239 
240  x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
241  x1 = MUL((t[8] - x2), xp[0]);
242  x2 = MUL((t[8] + x2), xp[1]);
243 
244  t[ 0] = x3 + x1;
245  t[ 8] = x4 - x2;
246  t[16] = x4 + x2;
247  t[24] = x3 - x1;
248  t++;
249  } while (t != t1);
250 
251  xp += 2;
252  t = tab;
253  t1 = tab + 4;
254  do {
255  xr = MUL(t[28],xp[0]);
256  t[28] = (t[0] - xr);
257  t[0] = (t[0] + xr);
258 
259  xr = MUL(t[4],xp[1]);
260  t[ 4] = (t[24] - xr);
261  t[24] = (t[24] + xr);
262 
263  xr = MUL(t[20],xp[2]);
264  t[20] = (t[8] - xr);
265  t[ 8] = (t[8] + xr);
266 
267  xr = MUL(t[12],xp[3]);
268  t[12] = (t[16] - xr);
269  t[16] = (t[16] + xr);
270  t++;
271  } while (t != t1);
272  xp += 4;
273 
274  for (i = 0; i < 4; i++) {
275  xr = MUL(tab[30-i*4],xp[0]);
276  tab[30-i*4] = (tab[i*4] - xr);
277  tab[ i*4] = (tab[i*4] + xr);
278 
279  xr = MUL(tab[ 2+i*4],xp[1]);
280  tab[ 2+i*4] = (tab[28-i*4] - xr);
281  tab[28-i*4] = (tab[28-i*4] + xr);
282 
283  xr = MUL(tab[31-i*4],xp[0]);
284  tab[31-i*4] = (tab[1+i*4] - xr);
285  tab[ 1+i*4] = (tab[1+i*4] + xr);
286 
287  xr = MUL(tab[ 3+i*4],xp[1]);
288  tab[ 3+i*4] = (tab[29-i*4] - xr);
289  tab[29-i*4] = (tab[29-i*4] + xr);
290 
291  xp += 2;
292  }
293 
294  t = tab + 30;
295  t1 = tab + 1;
296  do {
297  xr = MUL(t1[0], *xp);
298  t1[0] = (t[0] - xr);
299  t[0] = (t[0] + xr);
300  t -= 2;
301  t1 += 2;
302  xp++;
303  } while (t >= tab);
304 
305  for(i=0;i<32;i++) {
306  out[i] = tab[bitinv32[i]];
307  }
308 }
309 
310 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
311 
312 static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
313 {
314  short *p, *q;
315  int sum, offset, i, j;
316  int tmp[64];
317  int tmp1[32];
318  int *out;
319 
320  offset = s->samples_offset[ch];
321  out = &s->sb_samples[ch][0][0][0];
322  for(j=0;j<36;j++) {
323  /* 32 samples at once */
324  for(i=0;i<32;i++) {
325  s->samples_buf[ch][offset + (31 - i)] = samples[0];
326  samples += incr;
327  }
328 
329  /* filter */
330  p = s->samples_buf[ch] + offset;
331  q = filter_bank;
332  /* maxsum = 23169 */
333  for(i=0;i<64;i++) {
334  sum = p[0*64] * q[0*64];
335  sum += p[1*64] * q[1*64];
336  sum += p[2*64] * q[2*64];
337  sum += p[3*64] * q[3*64];
338  sum += p[4*64] * q[4*64];
339  sum += p[5*64] * q[5*64];
340  sum += p[6*64] * q[6*64];
341  sum += p[7*64] * q[7*64];
342  tmp[i] = sum;
343  p++;
344  q++;
345  }
346  tmp1[0] = tmp[16] >> WSHIFT;
347  for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
348  for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
349 
350  idct32(out, tmp1);
351 
352  /* advance of 32 samples */
353  offset -= 32;
354  out += 32;
355  /* handle the wrap around */
356  if (offset < 0) {
357  memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
358  s->samples_buf[ch], (512 - 32) * 2);
359  offset = SAMPLES_BUF_SIZE - 512;
360  }
361  }
362  s->samples_offset[ch] = offset;
363 }
364 
365 static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
366  unsigned char scale_factors[SBLIMIT][3],
367  int sb_samples[3][12][SBLIMIT],
368  int sblimit)
369 {
370  int *p, vmax, v, n, i, j, k, code;
371  int index, d1, d2;
372  unsigned char *sf = &scale_factors[0][0];
373 
374  for(j=0;j<sblimit;j++) {
375  for(i=0;i<3;i++) {
376  /* find the max absolute value */
377  p = &sb_samples[i][0][j];
378  vmax = abs(*p);
379  for(k=1;k<12;k++) {
380  p += SBLIMIT;
381  v = abs(*p);
382  if (v > vmax)
383  vmax = v;
384  }
385  /* compute the scale factor index using log 2 computations */
386  if (vmax > 1) {
387  n = av_log2(vmax);
388  /* n is the position of the MSB of vmax. now
389  use at most 2 compares to find the index */
390  index = (21 - n) * 3 - 3;
391  if (index >= 0) {
392  while (vmax <= scale_factor_table[index+1])
393  index++;
394  } else {
395  index = 0; /* very unlikely case of overflow */
396  }
397  } else {
398  index = 62; /* value 63 is not allowed */
399  }
400 
401  av_dlog(NULL, "%2d:%d in=%x %x %d\n",
402  j, i, vmax, scale_factor_table[index], index);
403  /* store the scale factor */
404  av_assert2(index >=0 && index <= 63);
405  sf[i] = index;
406  }
407 
408  /* compute the transmission factor : look if the scale factors
409  are close enough to each other */
410  d1 = scale_diff_table[sf[0] - sf[1] + 64];
411  d2 = scale_diff_table[sf[1] - sf[2] + 64];
412 
413  /* handle the 25 cases */
414  switch(d1 * 5 + d2) {
415  case 0*5+0:
416  case 0*5+4:
417  case 3*5+4:
418  case 4*5+0:
419  case 4*5+4:
420  code = 0;
421  break;
422  case 0*5+1:
423  case 0*5+2:
424  case 4*5+1:
425  case 4*5+2:
426  code = 3;
427  sf[2] = sf[1];
428  break;
429  case 0*5+3:
430  case 4*5+3:
431  code = 3;
432  sf[1] = sf[2];
433  break;
434  case 1*5+0:
435  case 1*5+4:
436  case 2*5+4:
437  code = 1;
438  sf[1] = sf[0];
439  break;
440  case 1*5+1:
441  case 1*5+2:
442  case 2*5+0:
443  case 2*5+1:
444  case 2*5+2:
445  code = 2;
446  sf[1] = sf[2] = sf[0];
447  break;
448  case 2*5+3:
449  case 3*5+3:
450  code = 2;
451  sf[0] = sf[1] = sf[2];
452  break;
453  case 3*5+0:
454  case 3*5+1:
455  case 3*5+2:
456  code = 2;
457  sf[0] = sf[2] = sf[1];
458  break;
459  case 1*5+3:
460  code = 2;
461  if (sf[0] > sf[2])
462  sf[0] = sf[2];
463  sf[1] = sf[2] = sf[0];
464  break;
465  default:
466  av_assert2(0); //cannot happen
467  code = 0; /* kill warning */
468  }
469 
470  av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,
471  sf[0], sf[1], sf[2], d1, d2, code);
472  scale_code[j] = code;
473  sf += 3;
474  }
475 }
476 
477 /* The most important function : psycho acoustic module. In this
478  encoder there is basically none, so this is the worst you can do,
479  but also this is the simpler. */
481 {
482  int i;
483 
484  for(i=0;i<s->sblimit;i++) {
485  smr[i] = (int)(fixed_smr[i] * 10);
486  }
487 }
488 
489 
490 #define SB_NOTALLOCATED 0
491 #define SB_ALLOCATED 1
492 #define SB_NOMORE 2
493 
494 /* Try to maximize the smr while using a number of bits inferior to
495  the frame size. I tried to make the code simpler, faster and
496  smaller than other encoders :-) */
498  short smr1[MPA_MAX_CHANNELS][SBLIMIT],
499  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
500  int *padding)
501 {
502  int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
503  int incr;
504  short smr[MPA_MAX_CHANNELS][SBLIMIT];
505  unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
506  const unsigned char *alloc;
507 
508  memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
509  memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
510  memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
511 
512  /* compute frame size and padding */
513  max_frame_size = s->frame_size;
514  s->frame_frac += s->frame_frac_incr;
515  if (s->frame_frac >= 65536) {
516  s->frame_frac -= 65536;
517  s->do_padding = 1;
518  max_frame_size += 8;
519  } else {
520  s->do_padding = 0;
521  }
522 
523  /* compute the header + bit alloc size */
524  current_frame_size = 32;
525  alloc = s->alloc_table;
526  for(i=0;i<s->sblimit;i++) {
527  incr = alloc[0];
528  current_frame_size += incr * s->nb_channels;
529  alloc += 1 << incr;
530  }
531  for(;;) {
532  /* look for the subband with the largest signal to mask ratio */
533  max_sb = -1;
534  max_ch = -1;
535  max_smr = INT_MIN;
536  for(ch=0;ch<s->nb_channels;ch++) {
537  for(i=0;i<s->sblimit;i++) {
538  if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
539  max_smr = smr[ch][i];
540  max_sb = i;
541  max_ch = ch;
542  }
543  }
544  }
545  if (max_sb < 0)
546  break;
547  av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n",
548  current_frame_size, max_frame_size, max_sb, max_ch,
549  bit_alloc[max_ch][max_sb]);
550 
551  /* find alloc table entry (XXX: not optimal, should use
552  pointer table) */
553  alloc = s->alloc_table;
554  for(i=0;i<max_sb;i++) {
555  alloc += 1 << alloc[0];
556  }
557 
558  if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
559  /* nothing was coded for this band: add the necessary bits */
560  incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
561  incr += total_quant_bits[alloc[1]];
562  } else {
563  /* increments bit allocation */
564  b = bit_alloc[max_ch][max_sb];
565  incr = total_quant_bits[alloc[b + 1]] -
566  total_quant_bits[alloc[b]];
567  }
568 
569  if (current_frame_size + incr <= max_frame_size) {
570  /* can increase size */
571  b = ++bit_alloc[max_ch][max_sb];
572  current_frame_size += incr;
573  /* decrease smr by the resolution we added */
574  smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
575  /* max allocation size reached ? */
576  if (b == ((1 << alloc[0]) - 1))
577  subband_status[max_ch][max_sb] = SB_NOMORE;
578  else
579  subband_status[max_ch][max_sb] = SB_ALLOCATED;
580  } else {
581  /* cannot increase the size of this subband */
582  subband_status[max_ch][max_sb] = SB_NOMORE;
583  }
584  }
585  *padding = max_frame_size - current_frame_size;
586  av_assert0(*padding >= 0);
587 }
588 
589 /*
590  * Output the mpeg audio layer 2 frame. Note how the code is small
591  * compared to other encoders :-)
592  */
594  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
595  int padding)
596 {
597  int i, j, k, l, bit_alloc_bits, b, ch;
598  unsigned char *sf;
599  int q[3];
600  PutBitContext *p = &s->pb;
601 
602  /* header */
603 
604  put_bits(p, 12, 0xfff);
605  put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
606  put_bits(p, 2, 4-2); /* layer 2 */
607  put_bits(p, 1, 1); /* no error protection */
608  put_bits(p, 4, s->bitrate_index);
609  put_bits(p, 2, s->freq_index);
610  put_bits(p, 1, s->do_padding); /* use padding */
611  put_bits(p, 1, 0); /* private_bit */
612  put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
613  put_bits(p, 2, 0); /* mode_ext */
614  put_bits(p, 1, 0); /* no copyright */
615  put_bits(p, 1, 1); /* original */
616  put_bits(p, 2, 0); /* no emphasis */
617 
618  /* bit allocation */
619  j = 0;
620  for(i=0;i<s->sblimit;i++) {
621  bit_alloc_bits = s->alloc_table[j];
622  for(ch=0;ch<s->nb_channels;ch++) {
623  put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
624  }
625  j += 1 << bit_alloc_bits;
626  }
627 
628  /* scale codes */
629  for(i=0;i<s->sblimit;i++) {
630  for(ch=0;ch<s->nb_channels;ch++) {
631  if (bit_alloc[ch][i])
632  put_bits(p, 2, s->scale_code[ch][i]);
633  }
634  }
635 
636  /* scale factors */
637  for(i=0;i<s->sblimit;i++) {
638  for(ch=0;ch<s->nb_channels;ch++) {
639  if (bit_alloc[ch][i]) {
640  sf = &s->scale_factors[ch][i][0];
641  switch(s->scale_code[ch][i]) {
642  case 0:
643  put_bits(p, 6, sf[0]);
644  put_bits(p, 6, sf[1]);
645  put_bits(p, 6, sf[2]);
646  break;
647  case 3:
648  case 1:
649  put_bits(p, 6, sf[0]);
650  put_bits(p, 6, sf[2]);
651  break;
652  case 2:
653  put_bits(p, 6, sf[0]);
654  break;
655  }
656  }
657  }
658  }
659 
660  /* quantization & write sub band samples */
661 
662  for(k=0;k<3;k++) {
663  for(l=0;l<12;l+=3) {
664  j = 0;
665  for(i=0;i<s->sblimit;i++) {
666  bit_alloc_bits = s->alloc_table[j];
667  for(ch=0;ch<s->nb_channels;ch++) {
668  b = bit_alloc[ch][i];
669  if (b) {
670  int qindex, steps, m, sample, bits;
671  /* we encode 3 sub band samples of the same sub band at a time */
672  qindex = s->alloc_table[j+b];
673  steps = ff_mpa_quant_steps[qindex];
674  for(m=0;m<3;m++) {
675  sample = s->sb_samples[ch][k][l + m][i];
676  /* divide by scale factor */
677 #ifdef USE_FLOATS
678  {
679  float a;
680  a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
681  q[m] = (int)((a + 1.0) * steps * 0.5);
682  }
683 #else
684  {
685  int q1, e, shift, mult;
686  e = s->scale_factors[ch][i][k];
687  shift = scale_factor_shift[e];
688  mult = scale_factor_mult[e];
689 
690  /* normalize to P bits */
691  if (shift < 0)
692  q1 = sample << (-shift);
693  else
694  q1 = sample >> shift;
695  q1 = (q1 * mult) >> P;
696  q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
697  }
698 #endif
699  if (q[m] >= steps)
700  q[m] = steps - 1;
701  av_assert2(q[m] >= 0 && q[m] < steps);
702  }
703  bits = ff_mpa_quant_bits[qindex];
704  if (bits < 0) {
705  /* group the 3 values to save bits */
706  put_bits(p, -bits,
707  q[0] + steps * (q[1] + steps * q[2]));
708  } else {
709  put_bits(p, bits, q[0]);
710  put_bits(p, bits, q[1]);
711  put_bits(p, bits, q[2]);
712  }
713  }
714  }
715  /* next subband in alloc table */
716  j += 1 << bit_alloc_bits;
717  }
718  }
719  }
720 
721  /* padding */
722  for(i=0;i<padding;i++)
723  put_bits(p, 1, 0);
724 
725  /* flush */
726  flush_put_bits(p);
727 }
728 
729 static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
730  const AVFrame *frame, int *got_packet_ptr)
731 {
732  MpegAudioContext *s = avctx->priv_data;
733  const int16_t *samples = (const int16_t *)frame->data[0];
734  short smr[MPA_MAX_CHANNELS][SBLIMIT];
735  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
736  int padding, i, ret;
737 
738  for(i=0;i<s->nb_channels;i++) {
739  filter(s, i, samples + i, s->nb_channels);
740  }
741 
742  for(i=0;i<s->nb_channels;i++) {
744  s->sb_samples[i], s->sblimit);
745  }
746  for(i=0;i<s->nb_channels;i++) {
747  psycho_acoustic_model(s, smr[i]);
748  }
749  compute_bit_allocation(s, smr, bit_alloc, &padding);
750 
751  if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0)
752  return ret;
753 
754  init_put_bits(&s->pb, avpkt->data, avpkt->size);
755 
756  encode_frame(s, bit_alloc, padding);
757 
758  if (frame->pts != AV_NOPTS_VALUE)
759  avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
760 
761  avpkt->size = put_bits_count(&s->pb) / 8;
762  *got_packet_ptr = 1;
763  return 0;
764 }
765 
766 static const AVCodecDefault mp2_defaults[] = {
767  { "b", "128k" },
768  { NULL },
769 };
770 
772  .name = "mp2",
773  .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
774  .type = AVMEDIA_TYPE_AUDIO,
775  .id = AV_CODEC_ID_MP2,
776  .priv_data_size = sizeof(MpegAudioContext),
778  .encode2 = MPA_encode_frame,
779  .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
781  .supported_samplerates = (const int[]){
782  44100, 48000, 32000, 22050, 24000, 16000, 0
783  },
784  .channel_layouts = (const uint64_t[]){ AV_CH_LAYOUT_MONO,
786  0 },
787  .defaults = mp2_defaults,
788 };