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[libav.git] / libavcodec / mpegaudio.c
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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 */
19#include <stdlib.h>
20#include <stdio.h>
21#include <string.h>
22#include <math.h>
23#include "avcodec.h"
24#include "mpegaudio.h"
25
26#define NDEBUG
27#include <assert.h>
28
29/* define it to use floats in quantization (I don't like floats !) */
30//#define USE_FLOATS
31
32#define MPA_STEREO 0
33#define MPA_JSTEREO 1
34#define MPA_DUAL 2
35#define MPA_MONO 3
36
37#include "mpegaudiotab.h"
38
39int MPA_encode_init(AVCodecContext *avctx)
40{
41 MpegAudioContext *s = avctx->priv_data;
42 int freq = avctx->sample_rate;
43 int bitrate = avctx->bit_rate;
44 int channels = avctx->channels;
45 int i, v, table, ch_bitrate;
46 float a;
47
48 if (channels > 2)
49 return -1;
50 bitrate = bitrate / 1000;
51 s->nb_channels = channels;
52 s->freq = freq;
53 s->bit_rate = bitrate * 1000;
54 avctx->frame_size = MPA_FRAME_SIZE;
55 avctx->key_frame = 1; /* always key frame */
56
57 /* encoding freq */
58 s->lsf = 0;
59 for(i=0;i<3;i++) {
60 if (freq_tab[i] == freq)
61 break;
62 if ((freq_tab[i] / 2) == freq) {
63 s->lsf = 1;
64 break;
65 }
66 }
67 if (i == 3)
68 return -1;
69 s->freq_index = i;
70
71 /* encoding bitrate & frequency */
72 for(i=0;i<15;i++) {
73 if (bitrate_tab[1-s->lsf][i] == bitrate)
74 break;
75 }
76 if (i == 15)
77 return -1;
78 s->bitrate_index = i;
79
80 /* compute total header size & pad bit */
81
82 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
83 s->frame_size = ((int)a) * 8;
84
85 /* frame fractional size to compute padding */
86 s->frame_frac = 0;
87 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
88
89 /* select the right allocation table */
90 ch_bitrate = bitrate / s->nb_channels;
91 if (!s->lsf) {
92 if ((freq == 48000 && ch_bitrate >= 56) ||
93 (ch_bitrate >= 56 && ch_bitrate <= 80))
94 table = 0;
95 else if (freq != 48000 && ch_bitrate >= 96)
96 table = 1;
97 else if (freq != 32000 && ch_bitrate <= 48)
98 table = 2;
99 else
100 table = 3;
101 } else {
102 table = 4;
103 }
104 /* number of used subbands */
105 s->sblimit = sblimit_table[table];
106 s->alloc_table = alloc_tables[table];
107
108#ifdef DEBUG
109 printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
110 bitrate, freq, s->frame_size, table, s->frame_frac_incr);
111#endif
112
113 for(i=0;i<s->nb_channels;i++)
114 s->samples_offset[i] = 0;
115
116 for(i=0;i<512;i++) {
117 float a = enwindow[i] * 32768.0 * 16.0;
118 filter_bank[i] = (int)(a);
119 }
120 for(i=0;i<64;i++) {
121 v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
122 if (v <= 0)
123 v = 1;
124 scale_factor_table[i] = v;
125#ifdef USE_FLOATS
126 scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
127#else
128#define P 15
129 scale_factor_shift[i] = 21 - P - (i / 3);
130 scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
131#endif
132 }
133 for(i=0;i<128;i++) {
134 v = i - 64;
135 if (v <= -3)
136 v = 0;
137 else if (v < 0)
138 v = 1;
139 else if (v == 0)
140 v = 2;
141 else if (v < 3)
142 v = 3;
143 else
144 v = 4;
145 scale_diff_table[i] = v;
146 }
147
148 for(i=0;i<17;i++) {
149 v = quant_bits[i];
150 if (v < 0)
151 v = -v;
152 else
153 v = v * 3;
154 total_quant_bits[i] = 12 * v;
155 }
156
157 return 0;
158}
159
160/* 32 point floating point IDCT */
161static void idct32(int *out, int *tab, int sblimit, int left_shift)
162{
163 int i, j;
164 int *t, *t1, xr;
165 const int *xp = costab32;
166
167 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
168
169 t = tab + 30;
170 t1 = tab + 2;
171 do {
172 t[0] += t[-4];
173 t[1] += t[1 - 4];
174 t -= 4;
175 } while (t != t1);
176
177 t = tab + 28;
178 t1 = tab + 4;
179 do {
180 t[0] += t[-8];
181 t[1] += t[1-8];
182 t[2] += t[2-8];
183 t[3] += t[3-8];
184 t -= 8;
185 } while (t != t1);
186
187 t = tab;
188 t1 = tab + 32;
189 do {
190 t[ 3] = -t[ 3];
191 t[ 6] = -t[ 6];
192
193 t[11] = -t[11];
194 t[12] = -t[12];
195 t[13] = -t[13];
196 t[15] = -t[15];
197 t += 16;
198 } while (t != t1);
199
200
201 t = tab;
202 t1 = tab + 8;
203 do {
204 int x1, x2, x3, x4;
205
206 x3 = MUL(t[16], FIX(SQRT2*0.5));
207 x4 = t[0] - x3;
208 x3 = t[0] + x3;
209
210 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
211 x1 = MUL((t[8] - x2), xp[0]);
212 x2 = MUL((t[8] + x2), xp[1]);
213
214 t[ 0] = x3 + x1;
215 t[ 8] = x4 - x2;
216 t[16] = x4 + x2;
217 t[24] = x3 - x1;
218 t++;
219 } while (t != t1);
220
221 xp += 2;
222 t = tab;
223 t1 = tab + 4;
224 do {
225 xr = MUL(t[28],xp[0]);
226 t[28] = (t[0] - xr);
227 t[0] = (t[0] + xr);
228
229 xr = MUL(t[4],xp[1]);
230 t[ 4] = (t[24] - xr);
231 t[24] = (t[24] + xr);
232
233 xr = MUL(t[20],xp[2]);
234 t[20] = (t[8] - xr);
235 t[ 8] = (t[8] + xr);
236
237 xr = MUL(t[12],xp[3]);
238 t[12] = (t[16] - xr);
239 t[16] = (t[16] + xr);
240 t++;
241 } while (t != t1);
242 xp += 4;
243
244 for (i = 0; i < 4; i++) {
245 xr = MUL(tab[30-i*4],xp[0]);
246 tab[30-i*4] = (tab[i*4] - xr);
247 tab[ i*4] = (tab[i*4] + xr);
248
249 xr = MUL(tab[ 2+i*4],xp[1]);
250 tab[ 2+i*4] = (tab[28-i*4] - xr);
251 tab[28-i*4] = (tab[28-i*4] + xr);
252
253 xr = MUL(tab[31-i*4],xp[0]);
254 tab[31-i*4] = (tab[1+i*4] - xr);
255 tab[ 1+i*4] = (tab[1+i*4] + xr);
256
257 xr = MUL(tab[ 3+i*4],xp[1]);
258 tab[ 3+i*4] = (tab[29-i*4] - xr);
259 tab[29-i*4] = (tab[29-i*4] + xr);
260
261 xp += 2;
262 }
263
264 t = tab + 30;
265 t1 = tab + 1;
266 do {
267 xr = MUL(t1[0], *xp);
268 t1[0] = (t[0] - xr);
269 t[0] = (t[0] + xr);
270 t -= 2;
271 t1 += 2;
272 xp++;
273 } while (t >= tab);
274
275 for(i=0;i<32;i++) {
276 out[i] = tab[bitinv32[i]] << left_shift;
277 }
278}
279
280static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
281{
282 short *p, *q;
283 int sum, offset, i, j, norm, n;
284 short tmp[64];
285 int tmp1[32];
286 int *out;
287
288 // print_pow1(samples, 1152);
289
290 offset = s->samples_offset[ch];
291 out = &s->sb_samples[ch][0][0][0];
292 for(j=0;j<36;j++) {
293 /* 32 samples at once */
294 for(i=0;i<32;i++) {
295 s->samples_buf[ch][offset + (31 - i)] = samples[0];
296 samples += incr;
297 }
298
299 /* filter */
300 p = s->samples_buf[ch] + offset;
301 q = filter_bank;
302 /* maxsum = 23169 */
303 for(i=0;i<64;i++) {
304 sum = p[0*64] * q[0*64];
305 sum += p[1*64] * q[1*64];
306 sum += p[2*64] * q[2*64];
307 sum += p[3*64] * q[3*64];
308 sum += p[4*64] * q[4*64];
309 sum += p[5*64] * q[5*64];
310 sum += p[6*64] * q[6*64];
311 sum += p[7*64] * q[7*64];
312 tmp[i] = sum >> 14;
313 p++;
314 q++;
315 }
316 tmp1[0] = tmp[16];
317 for( i=1; i<=16; i++ ) tmp1[i] = tmp[i+16]+tmp[16-i];
318 for( i=17; i<=31; i++ ) tmp1[i] = tmp[i+16]-tmp[80-i];
319
320 /* integer IDCT 32 with normalization. XXX: There may be some
321 overflow left */
322 norm = 0;
323 for(i=0;i<32;i++) {
324 norm |= abs(tmp1[i]);
325 }
326 n = log2(norm) - 12;
327 if (n > 0) {
328 for(i=0;i<32;i++)
329 tmp1[i] >>= n;
330 } else {
331 n = 0;
332 }
333
334 idct32(out, tmp1, s->sblimit, n);
335
336 /* advance of 32 samples */
337 offset -= 32;
338 out += 32;
339 /* handle the wrap around */
340 if (offset < 0) {
341 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
342 s->samples_buf[ch], (512 - 32) * 2);
343 offset = SAMPLES_BUF_SIZE - 512;
344 }
345 }
346 s->samples_offset[ch] = offset;
347
348 // print_pow(s->sb_samples, 1152);
349}
350
351static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
352 unsigned char scale_factors[SBLIMIT][3],
353 int sb_samples[3][12][SBLIMIT],
354 int sblimit)
355{
356 int *p, vmax, v, n, i, j, k, code;
357 int index, d1, d2;
358 unsigned char *sf = &scale_factors[0][0];
359
360 for(j=0;j<sblimit;j++) {
361 for(i=0;i<3;i++) {
362 /* find the max absolute value */
363 p = &sb_samples[i][0][j];
364 vmax = abs(*p);
365 for(k=1;k<12;k++) {
366 p += SBLIMIT;
367 v = abs(*p);
368 if (v > vmax)
369 vmax = v;
370 }
371 /* compute the scale factor index using log 2 computations */
372 if (vmax > 0) {
373 n = log2(vmax);
374 /* n is the position of the MSB of vmax. now
375 use at most 2 compares to find the index */
376 index = (21 - n) * 3 - 3;
377 if (index >= 0) {
378 while (vmax <= scale_factor_table[index+1])
379 index++;
380 } else {
381 index = 0; /* very unlikely case of overflow */
382 }
383 } else {
384 index = 63;
385 }
386
387#if 0
388 printf("%2d:%d in=%x %x %d\n",
389 j, i, vmax, scale_factor_table[index], index);
390#endif
391 /* store the scale factor */
392 assert(index >=0 && index <= 63);
393 sf[i] = index;
394 }
395
396 /* compute the transmission factor : look if the scale factors
397 are close enough to each other */
398 d1 = scale_diff_table[sf[0] - sf[1] + 64];
399 d2 = scale_diff_table[sf[1] - sf[2] + 64];
400
401 /* handle the 25 cases */
402 switch(d1 * 5 + d2) {
403 case 0*5+0:
404 case 0*5+4:
405 case 3*5+4:
406 case 4*5+0:
407 case 4*5+4:
408 code = 0;
409 break;
410 case 0*5+1:
411 case 0*5+2:
412 case 4*5+1:
413 case 4*5+2:
414 code = 3;
415 sf[2] = sf[1];
416 break;
417 case 0*5+3:
418 case 4*5+3:
419 code = 3;
420 sf[1] = sf[2];
421 break;
422 case 1*5+0:
423 case 1*5+4:
424 case 2*5+4:
425 code = 1;
426 sf[1] = sf[0];
427 break;
428 case 1*5+1:
429 case 1*5+2:
430 case 2*5+0:
431 case 2*5+1:
432 case 2*5+2:
433 code = 2;
434 sf[1] = sf[2] = sf[0];
435 break;
436 case 2*5+3:
437 case 3*5+3:
438 code = 2;
439 sf[0] = sf[1] = sf[2];
440 break;
441 case 3*5+0:
442 case 3*5+1:
443 case 3*5+2:
444 code = 2;
445 sf[0] = sf[2] = sf[1];
446 break;
447 case 1*5+3:
448 code = 2;
449 if (sf[0] > sf[2])
450 sf[0] = sf[2];
451 sf[1] = sf[2] = sf[0];
452 break;
453 default:
454 abort();
455 }
456
457#if 0
458 printf("%d: %2d %2d %2d %d %d -> %d\n", j,
459 sf[0], sf[1], sf[2], d1, d2, code);
460#endif
461 scale_code[j] = code;
462 sf += 3;
463 }
464}
465
466/* The most important function : psycho acoustic module. In this
467 encoder there is basically none, so this is the worst you can do,
468 but also this is the simpler. */
469static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
470{
471 int i;
472
473 for(i=0;i<s->sblimit;i++) {
474 smr[i] = (int)(fixed_smr[i] * 10);
475 }
476}
477
478
479#define SB_NOTALLOCATED 0
480#define SB_ALLOCATED 1
481#define SB_NOMORE 2
482
483/* Try to maximize the smr while using a number of bits inferior to
484 the frame size. I tried to make the code simpler, faster and
485 smaller than other encoders :-) */
486static void compute_bit_allocation(MpegAudioContext *s,
487 short smr1[MPA_MAX_CHANNELS][SBLIMIT],
488 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
489 int *padding)
490{
491 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
492 int incr;
493 short smr[MPA_MAX_CHANNELS][SBLIMIT];
494 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
495 const unsigned char *alloc;
496
497 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
498 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
499 memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
500
501 /* compute frame size and padding */
502 max_frame_size = s->frame_size;
503 s->frame_frac += s->frame_frac_incr;
504 if (s->frame_frac >= 65536) {
505 s->frame_frac -= 65536;
506 s->do_padding = 1;
507 max_frame_size += 8;
508 } else {
509 s->do_padding = 0;
510 }
511
512 /* compute the header + bit alloc size */
513 current_frame_size = 32;
514 alloc = s->alloc_table;
515 for(i=0;i<s->sblimit;i++) {
516 incr = alloc[0];
517 current_frame_size += incr * s->nb_channels;
518 alloc += 1 << incr;
519 }
520 for(;;) {
521 /* look for the subband with the largest signal to mask ratio */
522 max_sb = -1;
523 max_ch = -1;
524 max_smr = 0x80000000;
525 for(ch=0;ch<s->nb_channels;ch++) {
526 for(i=0;i<s->sblimit;i++) {
527 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
528 max_smr = smr[ch][i];
529 max_sb = i;
530 max_ch = ch;
531 }
532 }
533 }
534#if 0
535 printf("current=%d max=%d max_sb=%d alloc=%d\n",
536 current_frame_size, max_frame_size, max_sb,
537 bit_alloc[max_sb]);
538#endif
539 if (max_sb < 0)
540 break;
541
542 /* find alloc table entry (XXX: not optimal, should use
543 pointer table) */
544 alloc = s->alloc_table;
545 for(i=0;i<max_sb;i++) {
546 alloc += 1 << alloc[0];
547 }
548
549 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
550 /* nothing was coded for this band: add the necessary bits */
551 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
552 incr += total_quant_bits[alloc[1]];
553 } else {
554 /* increments bit allocation */
555 b = bit_alloc[max_ch][max_sb];
556 incr = total_quant_bits[alloc[b + 1]] -
557 total_quant_bits[alloc[b]];
558 }
559
560 if (current_frame_size + incr <= max_frame_size) {
561 /* can increase size */
562 b = ++bit_alloc[max_ch][max_sb];
563 current_frame_size += incr;
564 /* decrease smr by the resolution we added */
565 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
566 /* max allocation size reached ? */
567 if (b == ((1 << alloc[0]) - 1))
568 subband_status[max_ch][max_sb] = SB_NOMORE;
569 else
570 subband_status[max_ch][max_sb] = SB_ALLOCATED;
571 } else {
572 /* cannot increase the size of this subband */
573 subband_status[max_ch][max_sb] = SB_NOMORE;
574 }
575 }
576 *padding = max_frame_size - current_frame_size;
577 assert(*padding >= 0);
578
579#if 0
580 for(i=0;i<s->sblimit;i++) {
581 printf("%d ", bit_alloc[i]);
582 }
583 printf("\n");
584#endif
585}
586
587/*
588 * Output the mpeg audio layer 2 frame. Note how the code is small
589 * compared to other encoders :-)
590 */
591static void encode_frame(MpegAudioContext *s,
592 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
593 int padding)
594{
595 int i, j, k, l, bit_alloc_bits, b, ch;
596 unsigned char *sf;
597 int q[3];
598 PutBitContext *p = &s->pb;
599
600 /* header */
601
602 put_bits(p, 12, 0xfff);
603 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
604 put_bits(p, 2, 4-2); /* layer 2 */
605 put_bits(p, 1, 1); /* no error protection */
606 put_bits(p, 4, s->bitrate_index);
607 put_bits(p, 2, s->freq_index);
608 put_bits(p, 1, s->do_padding); /* use padding */
609 put_bits(p, 1, 0); /* private_bit */
610 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
611 put_bits(p, 2, 0); /* mode_ext */
612 put_bits(p, 1, 0); /* no copyright */
613 put_bits(p, 1, 1); /* original */
614 put_bits(p, 2, 0); /* no emphasis */
615
616 /* bit allocation */
617 j = 0;
618 for(i=0;i<s->sblimit;i++) {
619 bit_alloc_bits = s->alloc_table[j];
620 for(ch=0;ch<s->nb_channels;ch++) {
621 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
622 }
623 j += 1 << bit_alloc_bits;
624 }
625
626 /* scale codes */
627 for(i=0;i<s->sblimit;i++) {
628 for(ch=0;ch<s->nb_channels;ch++) {
629 if (bit_alloc[ch][i])
630 put_bits(p, 2, s->scale_code[ch][i]);
631 }
632 }
633
634 /* scale factors */
635 for(i=0;i<s->sblimit;i++) {
636 for(ch=0;ch<s->nb_channels;ch++) {
637 if (bit_alloc[ch][i]) {
638 sf = &s->scale_factors[ch][i][0];
639 switch(s->scale_code[ch][i]) {
640 case 0:
641 put_bits(p, 6, sf[0]);
642 put_bits(p, 6, sf[1]);
643 put_bits(p, 6, sf[2]);
644 break;
645 case 3:
646 case 1:
647 put_bits(p, 6, sf[0]);
648 put_bits(p, 6, sf[2]);
649 break;
650 case 2:
651 put_bits(p, 6, sf[0]);
652 break;
653 }
654 }
655 }
656 }
657
658 /* quantization & write sub band samples */
659
660 for(k=0;k<3;k++) {
661 for(l=0;l<12;l+=3) {
662 j = 0;
663 for(i=0;i<s->sblimit;i++) {
664 bit_alloc_bits = s->alloc_table[j];
665 for(ch=0;ch<s->nb_channels;ch++) {
666 b = bit_alloc[ch][i];
667 if (b) {
668 int qindex, steps, m, sample, bits;
669 /* we encode 3 sub band samples of the same sub band at a time */
670 qindex = s->alloc_table[j+b];
671 steps = quant_steps[qindex];
672 for(m=0;m<3;m++) {
673 sample = s->sb_samples[ch][k][l + m][i];
674 /* divide by scale factor */
675#ifdef USE_FLOATS
676 {
677 float a;
678 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
679 q[m] = (int)((a + 1.0) * steps * 0.5);
680 }
681#else
682 {
683 int q1, e, shift, mult;
684 e = s->scale_factors[ch][i][k];
685 shift = scale_factor_shift[e];
686 mult = scale_factor_mult[e];
687
688 /* normalize to P bits */
689 if (shift < 0)
690 q1 = sample << (-shift);
691 else
692 q1 = sample >> shift;
693 q1 = (q1 * mult) >> P;
694 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
695 }
696#endif
697 if (q[m] >= steps)
698 q[m] = steps - 1;
699 assert(q[m] >= 0 && q[m] < steps);
700 }
701 bits = quant_bits[qindex];
702 if (bits < 0) {
703 /* group the 3 values to save bits */
704 put_bits(p, -bits,
705 q[0] + steps * (q[1] + steps * q[2]));
706#if 0
707 printf("%d: gr1 %d\n",
708 i, q[0] + steps * (q[1] + steps * q[2]));
709#endif
710 } else {
711#if 0
712 printf("%d: gr3 %d %d %d\n",
713 i, q[0], q[1], q[2]);
714#endif
715 put_bits(p, bits, q[0]);
716 put_bits(p, bits, q[1]);
717 put_bits(p, bits, q[2]);
718 }
719 }
720 }
721 /* next subband in alloc table */
722 j += 1 << bit_alloc_bits;
723 }
724 }
725 }
726
727 /* padding */
728 for(i=0;i<padding;i++)
729 put_bits(p, 1, 0);
730
731 /* flush */
732 flush_put_bits(p);
733}
734
735int MPA_encode_frame(AVCodecContext *avctx,
736 unsigned char *frame, int buf_size, void *data)
737{
738 MpegAudioContext *s = avctx->priv_data;
739 short *samples = data;
740 short smr[MPA_MAX_CHANNELS][SBLIMIT];
741 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
742 int padding, i;
743
744 for(i=0;i<s->nb_channels;i++) {
745 filter(s, i, samples + i, s->nb_channels);
746 }
747
748 for(i=0;i<s->nb_channels;i++) {
749 compute_scale_factors(s->scale_code[i], s->scale_factors[i],
750 s->sb_samples[i], s->sblimit);
751 }
752 for(i=0;i<s->nb_channels;i++) {
753 psycho_acoustic_model(s, smr[i]);
754 }
755 compute_bit_allocation(s, smr, bit_alloc, &padding);
756
757 init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
758
759 encode_frame(s, bit_alloc, padding);
760
761 s->nb_samples += MPA_FRAME_SIZE;
762 return s->pb.buf_ptr - s->pb.buf;
763}
764
765
766AVCodec mp2_encoder = {
767 "mp2",
768 CODEC_TYPE_AUDIO,
769 CODEC_ID_MP2,
770 sizeof(MpegAudioContext),
771 MPA_encode_init,
772 MPA_encode_frame,
773 NULL,
774};