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mpegaudioenc_template.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 #include "mpegaudiodata.h"
39 #include "mpegaudiotab.h"
40 
41 /* currently, cannot change these constants (need to modify
42  quantization stage) */
43 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
44 
45 #define SAMPLES_BUF_SIZE 4096
46 
47 typedef struct MpegAudioContext {
50  int lsf; /* 1 if mpeg2 low bitrate selected */
51  int bitrate_index; /* bit rate */
53  int frame_size; /* frame size, in bits, without padding */
54  /* padding computation */
56  short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
57  int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
59  unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
60  /* code to group 3 scale factors */
62  int sblimit; /* number of used subbands */
63  const unsigned char *alloc_table;
64  int16_t filter_bank[512];
66  unsigned char scale_diff_table[128];
67 #if USE_FLOATS
68  float scale_factor_inv_table[64];
69 #else
70  int8_t scale_factor_shift[64];
71  unsigned short scale_factor_mult[64];
72 #endif
73  unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
75 
77 {
78  MpegAudioContext *s = avctx->priv_data;
79  int freq = avctx->sample_rate;
80  int bitrate = avctx->bit_rate;
81  int channels = avctx->channels;
82  int i, v, table;
83  float a;
84 
85  if (channels <= 0 || channels > 2){
86  av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
87  return AVERROR(EINVAL);
88  }
89  bitrate = bitrate / 1000;
90  s->nb_channels = channels;
91  avctx->frame_size = MPA_FRAME_SIZE;
92  avctx->delay = 512 - 32 + 1;
93 
94  /* encoding freq */
95  s->lsf = 0;
96  for(i=0;i<3;i++) {
97  if (avpriv_mpa_freq_tab[i] == freq)
98  break;
99  if ((avpriv_mpa_freq_tab[i] / 2) == freq) {
100  s->lsf = 1;
101  break;
102  }
103  }
104  if (i == 3){
105  av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
106  return AVERROR(EINVAL);
107  }
108  s->freq_index = i;
109 
110  /* encoding bitrate & frequency */
111  for(i=0;i<15;i++) {
112  if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
113  break;
114  }
115  if (i == 15){
116  av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
117  return AVERROR(EINVAL);
118  }
119  s->bitrate_index = i;
120 
121  /* compute total header size & pad bit */
122 
123  a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
124  s->frame_size = ((int)a) * 8;
125 
126  /* frame fractional size to compute padding */
127  s->frame_frac = 0;
128  s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
129 
130  /* select the right allocation table */
131  table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
132 
133  /* number of used subbands */
136 
137  av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
138  bitrate, freq, s->frame_size, table, s->frame_frac_incr);
139 
140  for(i=0;i<s->nb_channels;i++)
141  s->samples_offset[i] = 0;
142 
143  for(i=0;i<257;i++) {
144  int v;
145  v = ff_mpa_enwindow[i];
146 #if WFRAC_BITS != 16
147  v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
148 #endif
149  s->filter_bank[i] = v;
150  if ((i & 63) != 0)
151  v = -v;
152  if (i != 0)
153  s->filter_bank[512 - i] = v;
154  }
155 
156  for(i=0;i<64;i++) {
157  v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
158  if (v <= 0)
159  v = 1;
160  s->scale_factor_table[i] = v;
161 #if USE_FLOATS
162  s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
163 #else
164 #define P 15
165  s->scale_factor_shift[i] = 21 - P - (i / 3);
166  s->scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);
167 #endif
168  }
169  for(i=0;i<128;i++) {
170  v = i - 64;
171  if (v <= -3)
172  v = 0;
173  else if (v < 0)
174  v = 1;
175  else if (v == 0)
176  v = 2;
177  else if (v < 3)
178  v = 3;
179  else
180  v = 4;
181  s->scale_diff_table[i] = v;
182  }
183 
184  for(i=0;i<17;i++) {
185  v = ff_mpa_quant_bits[i];
186  if (v < 0)
187  v = -v;
188  else
189  v = v * 3;
190  s->total_quant_bits[i] = 12 * v;
191  }
192 
193  return 0;
194 }
195 
196 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
197 static void idct32(int *out, int *tab)
198 {
199  int i, j;
200  int *t, *t1, xr;
201  const int *xp = costab32;
202 
203  for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
204 
205  t = tab + 30;
206  t1 = tab + 2;
207  do {
208  t[0] += t[-4];
209  t[1] += t[1 - 4];
210  t -= 4;
211  } while (t != t1);
212 
213  t = tab + 28;
214  t1 = tab + 4;
215  do {
216  t[0] += t[-8];
217  t[1] += t[1-8];
218  t[2] += t[2-8];
219  t[3] += t[3-8];
220  t -= 8;
221  } while (t != t1);
222 
223  t = tab;
224  t1 = tab + 32;
225  do {
226  t[ 3] = -t[ 3];
227  t[ 6] = -t[ 6];
228 
229  t[11] = -t[11];
230  t[12] = -t[12];
231  t[13] = -t[13];
232  t[15] = -t[15];
233  t += 16;
234  } while (t != t1);
235 
236 
237  t = tab;
238  t1 = tab + 8;
239  do {
240  int x1, x2, x3, x4;
241 
242  x3 = MUL(t[16], FIX(SQRT2*0.5));
243  x4 = t[0] - x3;
244  x3 = t[0] + x3;
245 
246  x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
247  x1 = MUL((t[8] - x2), xp[0]);
248  x2 = MUL((t[8] + x2), xp[1]);
249 
250  t[ 0] = x3 + x1;
251  t[ 8] = x4 - x2;
252  t[16] = x4 + x2;
253  t[24] = x3 - x1;
254  t++;
255  } while (t != t1);
256 
257  xp += 2;
258  t = tab;
259  t1 = tab + 4;
260  do {
261  xr = MUL(t[28],xp[0]);
262  t[28] = (t[0] - xr);
263  t[0] = (t[0] + xr);
264 
265  xr = MUL(t[4],xp[1]);
266  t[ 4] = (t[24] - xr);
267  t[24] = (t[24] + xr);
268 
269  xr = MUL(t[20],xp[2]);
270  t[20] = (t[8] - xr);
271  t[ 8] = (t[8] + xr);
272 
273  xr = MUL(t[12],xp[3]);
274  t[12] = (t[16] - xr);
275  t[16] = (t[16] + xr);
276  t++;
277  } while (t != t1);
278  xp += 4;
279 
280  for (i = 0; i < 4; i++) {
281  xr = MUL(tab[30-i*4],xp[0]);
282  tab[30-i*4] = (tab[i*4] - xr);
283  tab[ i*4] = (tab[i*4] + xr);
284 
285  xr = MUL(tab[ 2+i*4],xp[1]);
286  tab[ 2+i*4] = (tab[28-i*4] - xr);
287  tab[28-i*4] = (tab[28-i*4] + xr);
288 
289  xr = MUL(tab[31-i*4],xp[0]);
290  tab[31-i*4] = (tab[1+i*4] - xr);
291  tab[ 1+i*4] = (tab[1+i*4] + xr);
292 
293  xr = MUL(tab[ 3+i*4],xp[1]);
294  tab[ 3+i*4] = (tab[29-i*4] - xr);
295  tab[29-i*4] = (tab[29-i*4] + xr);
296 
297  xp += 2;
298  }
299 
300  t = tab + 30;
301  t1 = tab + 1;
302  do {
303  xr = MUL(t1[0], *xp);
304  t1[0] = (t[0] - xr);
305  t[0] = (t[0] + xr);
306  t -= 2;
307  t1 += 2;
308  xp++;
309  } while (t >= tab);
310 
311  for(i=0;i<32;i++) {
312  out[i] = tab[bitinv32[i]];
313  }
314 }
315 
316 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
317 
318 static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
319 {
320  short *p, *q;
321  int sum, offset, i, j;
322  int tmp[64];
323  int tmp1[32];
324  int *out;
325 
326  offset = s->samples_offset[ch];
327  out = &s->sb_samples[ch][0][0][0];
328  for(j=0;j<36;j++) {
329  /* 32 samples at once */
330  for(i=0;i<32;i++) {
331  s->samples_buf[ch][offset + (31 - i)] = samples[0];
332  samples += incr;
333  }
334 
335  /* filter */
336  p = s->samples_buf[ch] + offset;
337  q = s->filter_bank;
338  /* maxsum = 23169 */
339  for(i=0;i<64;i++) {
340  sum = p[0*64] * q[0*64];
341  sum += p[1*64] * q[1*64];
342  sum += p[2*64] * q[2*64];
343  sum += p[3*64] * q[3*64];
344  sum += p[4*64] * q[4*64];
345  sum += p[5*64] * q[5*64];
346  sum += p[6*64] * q[6*64];
347  sum += p[7*64] * q[7*64];
348  tmp[i] = sum;
349  p++;
350  q++;
351  }
352  tmp1[0] = tmp[16] >> WSHIFT;
353  for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
354  for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
355 
356  idct32(out, tmp1);
357 
358  /* advance of 32 samples */
359  offset -= 32;
360  out += 32;
361  /* handle the wrap around */
362  if (offset < 0) {
363  memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
364  s->samples_buf[ch], (512 - 32) * 2);
365  offset = SAMPLES_BUF_SIZE - 512;
366  }
367  }
368  s->samples_offset[ch] = offset;
369 }
370 
372  unsigned char scale_code[SBLIMIT],
373  unsigned char scale_factors[SBLIMIT][3],
374  int sb_samples[3][12][SBLIMIT],
375  int sblimit)
376 {
377  int *p, vmax, v, n, i, j, k, code;
378  int index, d1, d2;
379  unsigned char *sf = &scale_factors[0][0];
380 
381  for(j=0;j<sblimit;j++) {
382  for(i=0;i<3;i++) {
383  /* find the max absolute value */
384  p = &sb_samples[i][0][j];
385  vmax = abs(*p);
386  for(k=1;k<12;k++) {
387  p += SBLIMIT;
388  v = abs(*p);
389  if (v > vmax)
390  vmax = v;
391  }
392  /* compute the scale factor index using log 2 computations */
393  if (vmax > 1) {
394  n = av_log2(vmax);
395  /* n is the position of the MSB of vmax. now
396  use at most 2 compares to find the index */
397  index = (21 - n) * 3 - 3;
398  if (index >= 0) {
399  while (vmax <= s->scale_factor_table[index+1])
400  index++;
401  } else {
402  index = 0; /* very unlikely case of overflow */
403  }
404  } else {
405  index = 62; /* value 63 is not allowed */
406  }
407 
408  av_dlog(NULL, "%2d:%d in=%x %x %d\n",
409  j, i, vmax, s->scale_factor_table[index], index);
410  /* store the scale factor */
411  av_assert2(index >=0 && index <= 63);
412  sf[i] = index;
413  }
414 
415  /* compute the transmission factor : look if the scale factors
416  are close enough to each other */
417  d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
418  d2 = s->scale_diff_table[sf[1] - sf[2] + 64];
419 
420  /* handle the 25 cases */
421  switch(d1 * 5 + d2) {
422  case 0*5+0:
423  case 0*5+4:
424  case 3*5+4:
425  case 4*5+0:
426  case 4*5+4:
427  code = 0;
428  break;
429  case 0*5+1:
430  case 0*5+2:
431  case 4*5+1:
432  case 4*5+2:
433  code = 3;
434  sf[2] = sf[1];
435  break;
436  case 0*5+3:
437  case 4*5+3:
438  code = 3;
439  sf[1] = sf[2];
440  break;
441  case 1*5+0:
442  case 1*5+4:
443  case 2*5+4:
444  code = 1;
445  sf[1] = sf[0];
446  break;
447  case 1*5+1:
448  case 1*5+2:
449  case 2*5+0:
450  case 2*5+1:
451  case 2*5+2:
452  code = 2;
453  sf[1] = sf[2] = sf[0];
454  break;
455  case 2*5+3:
456  case 3*5+3:
457  code = 2;
458  sf[0] = sf[1] = sf[2];
459  break;
460  case 3*5+0:
461  case 3*5+1:
462  case 3*5+2:
463  code = 2;
464  sf[0] = sf[2] = sf[1];
465  break;
466  case 1*5+3:
467  code = 2;
468  if (sf[0] > sf[2])
469  sf[0] = sf[2];
470  sf[1] = sf[2] = sf[0];
471  break;
472  default:
473  av_assert2(0); //cannot happen
474  code = 0; /* kill warning */
475  }
476 
477  av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,
478  sf[0], sf[1], sf[2], d1, d2, code);
479  scale_code[j] = code;
480  sf += 3;
481  }
482 }
483 
484 /* The most important function : psycho acoustic module. In this
485  encoder there is basically none, so this is the worst you can do,
486  but also this is the simpler. */
488 {
489  int i;
490 
491  for(i=0;i<s->sblimit;i++) {
492  smr[i] = (int)(fixed_smr[i] * 10);
493  }
494 }
495 
496 
497 #define SB_NOTALLOCATED 0
498 #define SB_ALLOCATED 1
499 #define SB_NOMORE 2
500 
501 /* Try to maximize the smr while using a number of bits inferior to
502  the frame size. I tried to make the code simpler, faster and
503  smaller than other encoders :-) */
505  short smr1[MPA_MAX_CHANNELS][SBLIMIT],
506  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
507  int *padding)
508 {
509  int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
510  int incr;
511  short smr[MPA_MAX_CHANNELS][SBLIMIT];
512  unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
513  const unsigned char *alloc;
514 
515  memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
516  memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
517  memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
518 
519  /* compute frame size and padding */
520  max_frame_size = s->frame_size;
521  s->frame_frac += s->frame_frac_incr;
522  if (s->frame_frac >= 65536) {
523  s->frame_frac -= 65536;
524  s->do_padding = 1;
525  max_frame_size += 8;
526  } else {
527  s->do_padding = 0;
528  }
529 
530  /* compute the header + bit alloc size */
531  current_frame_size = 32;
532  alloc = s->alloc_table;
533  for(i=0;i<s->sblimit;i++) {
534  incr = alloc[0];
535  current_frame_size += incr * s->nb_channels;
536  alloc += 1 << incr;
537  }
538  for(;;) {
539  /* look for the subband with the largest signal to mask ratio */
540  max_sb = -1;
541  max_ch = -1;
542  max_smr = INT_MIN;
543  for(ch=0;ch<s->nb_channels;ch++) {
544  for(i=0;i<s->sblimit;i++) {
545  if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
546  max_smr = smr[ch][i];
547  max_sb = i;
548  max_ch = ch;
549  }
550  }
551  }
552  if (max_sb < 0)
553  break;
554  av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n",
555  current_frame_size, max_frame_size, max_sb, max_ch,
556  bit_alloc[max_ch][max_sb]);
557 
558  /* find alloc table entry (XXX: not optimal, should use
559  pointer table) */
560  alloc = s->alloc_table;
561  for(i=0;i<max_sb;i++) {
562  alloc += 1 << alloc[0];
563  }
564 
565  if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
566  /* nothing was coded for this band: add the necessary bits */
567  incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
568  incr += s->total_quant_bits[alloc[1]];
569  } else {
570  /* increments bit allocation */
571  b = bit_alloc[max_ch][max_sb];
572  incr = s->total_quant_bits[alloc[b + 1]] -
573  s->total_quant_bits[alloc[b]];
574  }
575 
576  if (current_frame_size + incr <= max_frame_size) {
577  /* can increase size */
578  b = ++bit_alloc[max_ch][max_sb];
579  current_frame_size += incr;
580  /* decrease smr by the resolution we added */
581  smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
582  /* max allocation size reached ? */
583  if (b == ((1 << alloc[0]) - 1))
584  subband_status[max_ch][max_sb] = SB_NOMORE;
585  else
586  subband_status[max_ch][max_sb] = SB_ALLOCATED;
587  } else {
588  /* cannot increase the size of this subband */
589  subband_status[max_ch][max_sb] = SB_NOMORE;
590  }
591  }
592  *padding = max_frame_size - current_frame_size;
593  av_assert0(*padding >= 0);
594 }
595 
596 /*
597  * Output the mpeg audio layer 2 frame. Note how the code is small
598  * compared to other encoders :-)
599  */
601  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
602  int padding)
603 {
604  int i, j, k, l, bit_alloc_bits, b, ch;
605  unsigned char *sf;
606  int q[3];
607  PutBitContext *p = &s->pb;
608 
609  /* header */
610 
611  put_bits(p, 12, 0xfff);
612  put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
613  put_bits(p, 2, 4-2); /* layer 2 */
614  put_bits(p, 1, 1); /* no error protection */
615  put_bits(p, 4, s->bitrate_index);
616  put_bits(p, 2, s->freq_index);
617  put_bits(p, 1, s->do_padding); /* use padding */
618  put_bits(p, 1, 0); /* private_bit */
619  put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
620  put_bits(p, 2, 0); /* mode_ext */
621  put_bits(p, 1, 0); /* no copyright */
622  put_bits(p, 1, 1); /* original */
623  put_bits(p, 2, 0); /* no emphasis */
624 
625  /* bit allocation */
626  j = 0;
627  for(i=0;i<s->sblimit;i++) {
628  bit_alloc_bits = s->alloc_table[j];
629  for(ch=0;ch<s->nb_channels;ch++) {
630  put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
631  }
632  j += 1 << bit_alloc_bits;
633  }
634 
635  /* scale codes */
636  for(i=0;i<s->sblimit;i++) {
637  for(ch=0;ch<s->nb_channels;ch++) {
638  if (bit_alloc[ch][i])
639  put_bits(p, 2, s->scale_code[ch][i]);
640  }
641  }
642 
643  /* scale factors */
644  for(i=0;i<s->sblimit;i++) {
645  for(ch=0;ch<s->nb_channels;ch++) {
646  if (bit_alloc[ch][i]) {
647  sf = &s->scale_factors[ch][i][0];
648  switch(s->scale_code[ch][i]) {
649  case 0:
650  put_bits(p, 6, sf[0]);
651  put_bits(p, 6, sf[1]);
652  put_bits(p, 6, sf[2]);
653  break;
654  case 3:
655  case 1:
656  put_bits(p, 6, sf[0]);
657  put_bits(p, 6, sf[2]);
658  break;
659  case 2:
660  put_bits(p, 6, sf[0]);
661  break;
662  }
663  }
664  }
665  }
666 
667  /* quantization & write sub band samples */
668 
669  for(k=0;k<3;k++) {
670  for(l=0;l<12;l+=3) {
671  j = 0;
672  for(i=0;i<s->sblimit;i++) {
673  bit_alloc_bits = s->alloc_table[j];
674  for(ch=0;ch<s->nb_channels;ch++) {
675  b = bit_alloc[ch][i];
676  if (b) {
677  int qindex, steps, m, sample, bits;
678  /* we encode 3 sub band samples of the same sub band at a time */
679  qindex = s->alloc_table[j+b];
680  steps = ff_mpa_quant_steps[qindex];
681  for(m=0;m<3;m++) {
682  sample = s->sb_samples[ch][k][l + m][i];
683  /* divide by scale factor */
684 #if USE_FLOATS
685  {
686  float a;
687  a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]];
688  q[m] = (int)((a + 1.0) * steps * 0.5);
689  }
690 #else
691  {
692  int q1, e, shift, mult;
693  e = s->scale_factors[ch][i][k];
694  shift = s->scale_factor_shift[e];
695  mult = s->scale_factor_mult[e];
696 
697  /* normalize to P bits */
698  if (shift < 0)
699  q1 = sample << (-shift);
700  else
701  q1 = sample >> shift;
702  q1 = (q1 * mult) >> P;
703  q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
704  if (q[m] < 0)
705  q[m] = 0;
706  }
707 #endif
708  if (q[m] >= steps)
709  q[m] = steps - 1;
710  av_assert2(q[m] >= 0 && q[m] < steps);
711  }
712  bits = ff_mpa_quant_bits[qindex];
713  if (bits < 0) {
714  /* group the 3 values to save bits */
715  put_bits(p, -bits,
716  q[0] + steps * (q[1] + steps * q[2]));
717  } else {
718  put_bits(p, bits, q[0]);
719  put_bits(p, bits, q[1]);
720  put_bits(p, bits, q[2]);
721  }
722  }
723  }
724  /* next subband in alloc table */
725  j += 1 << bit_alloc_bits;
726  }
727  }
728  }
729 
730  /* padding */
731  for(i=0;i<padding;i++)
732  put_bits(p, 1, 0);
733 
734  /* flush */
735  flush_put_bits(p);
736 }
737 
738 static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
739  const AVFrame *frame, int *got_packet_ptr)
740 {
741  MpegAudioContext *s = avctx->priv_data;
742  const int16_t *samples = (const int16_t *)frame->data[0];
743  short smr[MPA_MAX_CHANNELS][SBLIMIT];
744  unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
745  int padding, i, ret;
746 
747  for(i=0;i<s->nb_channels;i++) {
748  filter(s, i, samples + i, s->nb_channels);
749  }
750 
751  for(i=0;i<s->nb_channels;i++) {
753  s->sb_samples[i], s->sblimit);
754  }
755  for(i=0;i<s->nb_channels;i++) {
756  psycho_acoustic_model(s, smr[i]);
757  }
758  compute_bit_allocation(s, smr, bit_alloc, &padding);
759 
760  if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0)
761  return ret;
762 
763  init_put_bits(&s->pb, avpkt->data, avpkt->size);
764 
765  encode_frame(s, bit_alloc, padding);
766 
767  if (frame->pts != AV_NOPTS_VALUE)
768  avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay);
769 
770  avpkt->size = put_bits_count(&s->pb) / 8;
771  *got_packet_ptr = 1;
772  return 0;
773 }
774 
775 static const AVCodecDefault mp2_defaults[] = {
776  { "b", "128k" },
777  { NULL },
778 };
779