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added mpeg audio decoder tables
[libav.git] / libavcodec / mpegaudio.c
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de6d9b64
FB		 1/*
2 * The simplest mpeg audio layer 2 encoder
3 * Copyright (c) 2000 Gerard Lantau.
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; either version 2 of the License, or
8 * (at your option) any later version.
9 *
10 * This program is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 * GNU General Public License for more details.
14 *
15 * You should have received a copy of the GNU General Public License
16 * along with this program; if not, write to the Free Software
17 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
18 */
de6d9b64 19#include "avcodec.h"
1a565432 20#include <math.h>
de6d9b64
FB		 21#include "mpegaudio.h"
22
de6d9b64
FB		 23/* define it to use floats in quantization (I don't like floats !) */
24//#define USE_FLOATS
25
26#define MPA_STEREO 0
27#define MPA_JSTEREO 1
28#define MPA_DUAL 2
29#define MPA_MONO 3
30
31#include "mpegaudiotab.h"
32
33int MPA_encode_init(AVCodecContext *avctx)
34{
35 MpegAudioContext *s = avctx->priv_data;
36 int freq = avctx->sample_rate;
37 int bitrate = avctx->bit_rate;
38 int channels = avctx->channels;
39 int i, v, table, ch_bitrate;
40 float a;
41
42 if (channels > 2)
43 return -1;
44 bitrate = bitrate / 1000;
45 s->nb_channels = channels;
46 s->freq = freq;
47 s->bit_rate = bitrate * 1000;
48 avctx->frame_size = MPA_FRAME_SIZE;
49 avctx->key_frame = 1; /* always key frame */
50
51 /* encoding freq */
52 s->lsf = 0;
53 for(i=0;i<3;i++) {
54 if (freq_tab[i] == freq)
55 break;
56 if ((freq_tab[i] / 2) == freq) {
57 s->lsf = 1;
58 break;
59 }
60 }
61 if (i == 3)
62 return -1;
63 s->freq_index = i;
64
65 /* encoding bitrate & frequency */
66 for(i=0;i<15;i++) {
67 if (bitrate_tab[1-s->lsf][i] == bitrate)
68 break;
69 }
70 if (i == 15)
71 return -1;
72 s->bitrate_index = i;
73
74 /* compute total header size & pad bit */
75
76 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
77 s->frame_size = ((int)a) * 8;
78
79 /* frame fractional size to compute padding */
80 s->frame_frac = 0;
81 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
82
83 /* select the right allocation table */
84 ch_bitrate = bitrate / s->nb_channels;
85 if (!s->lsf) {
86 if ((freq == 48000 && ch_bitrate >= 56) ||
87 (ch_bitrate >= 56 && ch_bitrate <= 80))
88 table = 0;
89 else if (freq != 48000 && ch_bitrate >= 96)
90 table = 1;
91 else if (freq != 32000 && ch_bitrate <= 48)
92 table = 2;
93 else
94 table = 3;
95 } else {
96 table = 4;
97 }
98 /* number of used subbands */
99 s->sblimit = sblimit_table[table];
100 s->alloc_table = alloc_tables[table];
101
102#ifdef DEBUG
103 printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
104 bitrate, freq, s->frame_size, table, s->frame_frac_incr);
105#endif
106
107 for(i=0;i<s->nb_channels;i++)
108 s->samples_offset[i] = 0;
109
110 for(i=0;i<512;i++) {
111 float a = enwindow[i] * 32768.0 * 16.0;
112 filter_bank[i] = (int)(a);
113 }
114 for(i=0;i<64;i++) {
115 v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
116 if (v <= 0)
117 v = 1;
118 scale_factor_table[i] = v;
119#ifdef USE_FLOATS
120 scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
121#else
122#define P 15
123 scale_factor_shift[i] = 21 - P - (i / 3);
124 scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
125#endif
126 }
127 for(i=0;i<128;i++) {
128 v = i - 64;
129 if (v <= -3)
130 v = 0;
131 else if (v < 0)
132 v = 1;
133 else if (v == 0)
134 v = 2;
135 else if (v < 3)
136 v = 3;
137 else
138 v = 4;
139 scale_diff_table[i] = v;
140 }
141
142 for(i=0;i<17;i++) {
143 v = quant_bits[i];
144 if (v < 0)
145 v = -v;
146 else
147 v = v * 3;
148 total_quant_bits[i] = 12 * v;
149 }
150
151 return 0;
152}
153
154/* 32 point floating point IDCT */
155static void idct32(int *out, int *tab, int sblimit, int left_shift)
156{
157 int i, j;
158 int *t, *t1, xr;
159 const int *xp = costab32;
160
161 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
162
163 t = tab + 30;
164 t1 = tab + 2;
165 do {
166 t[0] += t[-4];
167 t[1] += t[1 - 4];
168 t -= 4;
169 } while (t != t1);
170
171 t = tab + 28;
172 t1 = tab + 4;
173 do {
174 t[0] += t[-8];
175 t[1] += t[1-8];
176 t[2] += t[2-8];
177 t[3] += t[3-8];
178 t -= 8;
179 } while (t != t1);
180
181 t = tab;
182 t1 = tab + 32;
183 do {
184 t[ 3] = -t[ 3];
185 t[ 6] = -t[ 6];
186
187 t[11] = -t[11];
188 t[12] = -t[12];
189 t[13] = -t[13];
190 t[15] = -t[15];
191 t += 16;
192 } while (t != t1);
193
194
195 t = tab;
196 t1 = tab + 8;
197 do {
198 int x1, x2, x3, x4;
199
200 x3 = MUL(t[16], FIX(SQRT2*0.5));
201 x4 = t[0] - x3;
202 x3 = t[0] + x3;
203
204 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
205 x1 = MUL((t[8] - x2), xp[0]);
206 x2 = MUL((t[8] + x2), xp[1]);
207
208 t[ 0] = x3 + x1;
209 t[ 8] = x4 - x2;
210 t[16] = x4 + x2;
211 t[24] = x3 - x1;
212 t++;
213 } while (t != t1);
214
215 xp += 2;
216 t = tab;
217 t1 = tab + 4;
218 do {
219 xr = MUL(t[28],xp[0]);
220 t[28] = (t[0] - xr);
221 t[0] = (t[0] + xr);
222
223 xr = MUL(t[4],xp[1]);
224 t[ 4] = (t[24] - xr);
225 t[24] = (t[24] + xr);
226
227 xr = MUL(t[20],xp[2]);
228 t[20] = (t[8] - xr);
229 t[ 8] = (t[8] + xr);
230
231 xr = MUL(t[12],xp[3]);
232 t[12] = (t[16] - xr);
233 t[16] = (t[16] + xr);
234 t++;
235 } while (t != t1);
236 xp += 4;
237
238 for (i = 0; i < 4; i++) {
239 xr = MUL(tab[30-i*4],xp[0]);
240 tab[30-i*4] = (tab[i*4] - xr);
241 tab[ i*4] = (tab[i*4] + xr);
242
243 xr = MUL(tab[ 2+i*4],xp[1]);
244 tab[ 2+i*4] = (tab[28-i*4] - xr);
245 tab[28-i*4] = (tab[28-i*4] + xr);
246
247 xr = MUL(tab[31-i*4],xp[0]);
248 tab[31-i*4] = (tab[1+i*4] - xr);
249 tab[ 1+i*4] = (tab[1+i*4] + xr);
250
251 xr = MUL(tab[ 3+i*4],xp[1]);
252 tab[ 3+i*4] = (tab[29-i*4] - xr);
253 tab[29-i*4] = (tab[29-i*4] + xr);
254
255 xp += 2;
256 }
257
258 t = tab + 30;
259 t1 = tab + 1;
260 do {
261 xr = MUL(t1[0], *xp);
262 t1[0] = (t[0] - xr);
263 t[0] = (t[0] + xr);
264 t -= 2;
265 t1 += 2;
266 xp++;
267 } while (t >= tab);
268
269 for(i=0;i<32;i++) {
270 out[i] = tab[bitinv32[i]] << left_shift;
271 }
272}
273
274static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
275{
276 short *p, *q;
277 int sum, offset, i, j, norm, n;
278 short tmp[64];
279 int tmp1[32];
280 int *out;
281
282 // print_pow1(samples, 1152);
283
284 offset = s->samples_offset[ch];
285 out = &s->sb_samples[ch][0][0][0];
286 for(j=0;j<36;j++) {
287 /* 32 samples at once */
288 for(i=0;i<32;i++) {
289 s->samples_buf[ch][offset + (31 - i)] = samples[0];
290 samples += incr;
291 }
292
293 /* filter */
294 p = s->samples_buf[ch] + offset;
295 q = filter_bank;
296 /* maxsum = 23169 */
297 for(i=0;i<64;i++) {
298 sum = p[0*64] * q[0*64];
299 sum += p[1*64] * q[1*64];
300 sum += p[2*64] * q[2*64];
301 sum += p[3*64] * q[3*64];
302 sum += p[4*64] * q[4*64];
303 sum += p[5*64] * q[5*64];
304 sum += p[6*64] * q[6*64];
305 sum += p[7*64] * q[7*64];
306 tmp[i] = sum >> 14;
307 p++;
308 q++;
309 }
310 tmp1[0] = tmp[16];
311 for( i=1; i<=16; i++ ) tmp1[i] = tmp[i+16]+tmp[16-i];
312 for( i=17; i<=31; i++ ) tmp1[i] = tmp[i+16]-tmp[80-i];
313
314 /* integer IDCT 32 with normalization. XXX: There may be some
315 overflow left */
316 norm = 0;
317 for(i=0;i<32;i++) {
318 norm |= abs(tmp1[i]);
319 }
935442b5 320 n = av_log2(norm) - 12;
de6d9b64
FB		 321 if (n > 0) {
322 for(i=0;i<32;i++)
323 tmp1[i] >>= n;
324 } else {
325 n = 0;
326 }
327
328 idct32(out, tmp1, s->sblimit, n);
329
330 /* advance of 32 samples */
331 offset -= 32;
332 out += 32;
333 /* handle the wrap around */
334 if (offset < 0) {
335 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
336 s->samples_buf[ch], (512 - 32) * 2);
337 offset = SAMPLES_BUF_SIZE - 512;
338 }
339 }
340 s->samples_offset[ch] = offset;
341
342 // print_pow(s->sb_samples, 1152);
343}
344
345static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
346 unsigned char scale_factors[SBLIMIT][3],
347 int sb_samples[3][12][SBLIMIT],
348 int sblimit)
349{
350 int *p, vmax, v, n, i, j, k, code;
351 int index, d1, d2;
352 unsigned char *sf = &scale_factors[0][0];
353
354 for(j=0;j<sblimit;j++) {
355 for(i=0;i<3;i++) {
356 /* find the max absolute value */
357 p = &sb_samples[i][0][j];
358 vmax = abs(*p);
359 for(k=1;k<12;k++) {
360 p += SBLIMIT;
361 v = abs(*p);
362 if (v > vmax)
363 vmax = v;
364 }
365 /* compute the scale factor index using log 2 computations */
366 if (vmax > 0) {
935442b5 367 n = av_log2(vmax);
de6d9b64
FB		 368 /* n is the position of the MSB of vmax. now
369 use at most 2 compares to find the index */
370 index = (21 - n) * 3 - 3;
371 if (index >= 0) {
372 while (vmax <= scale_factor_table[index+1])
373 index++;
374 } else {
375 index = 0; /* very unlikely case of overflow */
376 }
377 } else {
378 index = 63;
379 }
380
381#if 0
382 printf("%2d:%d in=%x %x %d\n",
383 j, i, vmax, scale_factor_table[index], index);
384#endif
385 /* store the scale factor */
386 assert(index >=0 && index <= 63);
387 sf[i] = index;
388 }
389
390 /* compute the transmission factor : look if the scale factors
391 are close enough to each other */
392 d1 = scale_diff_table[sf[0] - sf[1] + 64];
393 d2 = scale_diff_table[sf[1] - sf[2] + 64];
394
395 /* handle the 25 cases */
396 switch(d1 * 5 + d2) {
397 case 0*5+0:
398 case 0*5+4:
399 case 3*5+4:
400 case 4*5+0:
401 case 4*5+4:
402 code = 0;
403 break;
404 case 0*5+1:
405 case 0*5+2:
406 case 4*5+1:
407 case 4*5+2:
408 code = 3;
409 sf[2] = sf[1];
410 break;
411 case 0*5+3:
412 case 4*5+3:
413 code = 3;
414 sf[1] = sf[2];
415 break;
416 case 1*5+0:
417 case 1*5+4:
418 case 2*5+4:
419 code = 1;
420 sf[1] = sf[0];
421 break;
422 case 1*5+1:
423 case 1*5+2:
424 case 2*5+0:
425 case 2*5+1:
426 case 2*5+2:
427 code = 2;
428 sf[1] = sf[2] = sf[0];
429 break;
430 case 2*5+3:
431 case 3*5+3:
432 code = 2;
433 sf[0] = sf[1] = sf[2];
434 break;
435 case 3*5+0:
436 case 3*5+1:
437 case 3*5+2:
438 code = 2;
439 sf[0] = sf[2] = sf[1];
440 break;
441 case 1*5+3:
442 code = 2;
443 if (sf[0] > sf[2])
444 sf[0] = sf[2];
445 sf[1] = sf[2] = sf[0];
446 break;
447 default:
448 abort();
449 }
450
451#if 0
452 printf("%d: %2d %2d %2d %d %d -> %d\n", j,
453 sf[0], sf[1], sf[2], d1, d2, code);
454#endif
455 scale_code[j] = code;
456 sf += 3;
457 }
458}
459
460/* The most important function : psycho acoustic module. In this
461 encoder there is basically none, so this is the worst you can do,
462 but also this is the simpler. */
463static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
464{
465 int i;
466
467 for(i=0;i<s->sblimit;i++) {
468 smr[i] = (int)(fixed_smr[i] * 10);
469 }
470}
471
472
473#define SB_NOTALLOCATED 0
474#define SB_ALLOCATED 1
475#define SB_NOMORE 2
476
477/* Try to maximize the smr while using a number of bits inferior to
478 the frame size. I tried to make the code simpler, faster and
479 smaller than other encoders :-) */
480static void compute_bit_allocation(MpegAudioContext *s,
481 short smr1[MPA_MAX_CHANNELS][SBLIMIT],
482 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
483 int *padding)
484{
485 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
486 int incr;
487 short smr[MPA_MAX_CHANNELS][SBLIMIT];
488 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
489 const unsigned char *alloc;
490
491 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
492 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
493 memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
494
495 /* compute frame size and padding */
496 max_frame_size = s->frame_size;
497 s->frame_frac += s->frame_frac_incr;
498 if (s->frame_frac >= 65536) {
499 s->frame_frac -= 65536;
500 s->do_padding = 1;
501 max_frame_size += 8;
502 } else {
503 s->do_padding = 0;
504 }
505
506 /* compute the header + bit alloc size */
507 current_frame_size = 32;
508 alloc = s->alloc_table;
509 for(i=0;i<s->sblimit;i++) {
510 incr = alloc[0];
511 current_frame_size += incr * s->nb_channels;
512 alloc += 1 << incr;
513 }
514 for(;;) {
515 /* look for the subband with the largest signal to mask ratio */
516 max_sb = -1;
517 max_ch = -1;
518 max_smr = 0x80000000;
519 for(ch=0;ch<s->nb_channels;ch++) {
520 for(i=0;i<s->sblimit;i++) {
521 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
522 max_smr = smr[ch][i];
523 max_sb = i;
524 max_ch = ch;
525 }
526 }
527 }
528#if 0
529 printf("current=%d max=%d max_sb=%d alloc=%d\n",
530 current_frame_size, max_frame_size, max_sb,
531 bit_alloc[max_sb]);
532#endif
533 if (max_sb < 0)
534 break;
535
536 /* find alloc table entry (XXX: not optimal, should use
537 pointer table) */
538 alloc = s->alloc_table;
539 for(i=0;i<max_sb;i++) {
540 alloc += 1 << alloc[0];
541 }
542
543 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
544 /* nothing was coded for this band: add the necessary bits */
545 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
546 incr += total_quant_bits[alloc[1]];
547 } else {
548 /* increments bit allocation */
549 b = bit_alloc[max_ch][max_sb];
550 incr = total_quant_bits[alloc[b + 1]] -
551 total_quant_bits[alloc[b]];
552 }
553
554 if (current_frame_size + incr <= max_frame_size) {
555 /* can increase size */
556 b = ++bit_alloc[max_ch][max_sb];
557 current_frame_size += incr;
558 /* decrease smr by the resolution we added */
559 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
560 /* max allocation size reached ? */
561 if (b == ((1 << alloc[0]) - 1))
562 subband_status[max_ch][max_sb] = SB_NOMORE;
563 else
564 subband_status[max_ch][max_sb] = SB_ALLOCATED;
565 } else {
566 /* cannot increase the size of this subband */
567 subband_status[max_ch][max_sb] = SB_NOMORE;
568 }
569 }
570 *padding = max_frame_size - current_frame_size;
571 assert(*padding >= 0);
572
573#if 0
574 for(i=0;i<s->sblimit;i++) {
575 printf("%d ", bit_alloc[i]);
576 }
577 printf("\n");
578#endif
579}
580
581/*
582 * Output the mpeg audio layer 2 frame. Note how the code is small
583 * compared to other encoders :-)
584 */
585static void encode_frame(MpegAudioContext *s,
586 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
587 int padding)
588{
589 int i, j, k, l, bit_alloc_bits, b, ch;
590 unsigned char *sf;
591 int q[3];
592 PutBitContext *p = &s->pb;
593
594 /* header */
595
596 put_bits(p, 12, 0xfff);
597 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
598 put_bits(p, 2, 4-2); /* layer 2 */
599 put_bits(p, 1, 1); /* no error protection */
600 put_bits(p, 4, s->bitrate_index);
601 put_bits(p, 2, s->freq_index);
602 put_bits(p, 1, s->do_padding); /* use padding */
603 put_bits(p, 1, 0); /* private_bit */
604 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
605 put_bits(p, 2, 0); /* mode_ext */
606 put_bits(p, 1, 0); /* no copyright */
607 put_bits(p, 1, 1); /* original */
608 put_bits(p, 2, 0); /* no emphasis */
609
610 /* bit allocation */
611 j = 0;
612 for(i=0;i<s->sblimit;i++) {
613 bit_alloc_bits = s->alloc_table[j];
614 for(ch=0;ch<s->nb_channels;ch++) {
615 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
616 }
617 j += 1 << bit_alloc_bits;
618 }
619
620 /* scale codes */
621 for(i=0;i<s->sblimit;i++) {
622 for(ch=0;ch<s->nb_channels;ch++) {
623 if (bit_alloc[ch][i])
624 put_bits(p, 2, s->scale_code[ch][i]);
625 }
626 }
627
628 /* scale factors */
629 for(i=0;i<s->sblimit;i++) {
630 for(ch=0;ch<s->nb_channels;ch++) {
631 if (bit_alloc[ch][i]) {
632 sf = &s->scale_factors[ch][i][0];
633 switch(s->scale_code[ch][i]) {
634 case 0:
635 put_bits(p, 6, sf[0]);
636 put_bits(p, 6, sf[1]);
637 put_bits(p, 6, sf[2]);
638 break;
639 case 3:
640 case 1:
641 put_bits(p, 6, sf[0]);
642 put_bits(p, 6, sf[2]);
643 break;
644 case 2:
645 put_bits(p, 6, sf[0]);
646 break;
647 }
648 }
649 }
650 }
651
652 /* quantization & write sub band samples */
653
654 for(k=0;k<3;k++) {
655 for(l=0;l<12;l+=3) {
656 j = 0;
657 for(i=0;i<s->sblimit;i++) {
658 bit_alloc_bits = s->alloc_table[j];
659 for(ch=0;ch<s->nb_channels;ch++) {
660 b = bit_alloc[ch][i];
661 if (b) {
662 int qindex, steps, m, sample, bits;
663 /* we encode 3 sub band samples of the same sub band at a time */
664 qindex = s->alloc_table[j+b];
665 steps = quant_steps[qindex];
666 for(m=0;m<3;m++) {
667 sample = s->sb_samples[ch][k][l + m][i];
668 /* divide by scale factor */
669#ifdef USE_FLOATS
670 {
671 float a;
672 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
673 q[m] = (int)((a + 1.0) * steps * 0.5);
674 }
675#else
676 {
677 int q1, e, shift, mult;
678 e = s->scale_factors[ch][i][k];
679 shift = scale_factor_shift[e];
680 mult = scale_factor_mult[e];
681
682 /* normalize to P bits */
683 if (shift < 0)
684 q1 = sample << (-shift);
685 else
686 q1 = sample >> shift;
687 q1 = (q1 * mult) >> P;
688 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
689 }
690#endif
691 if (q[m] >= steps)
692 q[m] = steps - 1;
693 assert(q[m] >= 0 && q[m] < steps);
694 }
695 bits = quant_bits[qindex];
696 if (bits < 0) {
697 /* group the 3 values to save bits */
698 put_bits(p, -bits,
699 q[0] + steps * (q[1] + steps * q[2]));
700#if 0
701 printf("%d: gr1 %d\n",
702 i, q[0] + steps * (q[1] + steps * q[2]));
703#endif
704 } else {
705#if 0
706 printf("%d: gr3 %d %d %d\n",
707 i, q[0], q[1], q[2]);
708#endif
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
729int MPA_encode_frame(AVCodecContext *avctx,
730 unsigned char *frame, int buf_size, void *data)
731{
732 MpegAudioContext *s = avctx->priv_data;
733 short *samples = data;
734 short smr[MPA_MAX_CHANNELS][SBLIMIT];
735 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
736 int padding, i;
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++) {
743 compute_scale_factors(s->scale_code[i], s->scale_factors[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 init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
752
753 encode_frame(s, bit_alloc, padding);
754
755 s->nb_samples += MPA_FRAME_SIZE;
756 return s->pb.buf_ptr - s->pb.buf;
757}
758
759
760AVCodec mp2_encoder = {
761 "mp2",
762 CODEC_TYPE_AUDIO,
763 CODEC_ID_MP2,
764 sizeof(MpegAudioContext),
765 MPA_encode_init,
766 MPA_encode_frame,
767 NULL,
768};
Libav master