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d521acc588c99df03a1bac48861e1183f86a06b3
[libav.git] / libavcodec / mpegaudio.c
1 /*
2 * The simplest mpeg audio layer 2 encoder
3 * Copyright (c) 2000, 2001 Fabrice Bellard.
4 *
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Lesser General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
9 *
10 * This library 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 GNU
13 * Lesser General Public License for more details.
14 *
15 * You should have received a copy of the GNU Lesser General Public
16 * License along with this library; if not, write to the Free Software
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
18 */
19
20 /**
21 * @file mpegaudio.c
22 * The simplest mpeg audio layer 2 encoder.
23 */
24
25 #include "avcodec.h"
26 #include "bitstream.h"
27 #include "mpegaudio.h"
28
29 /* currently, cannot change these constants (need to modify
30 quantization stage) */
31 #define FRAC_BITS 15
32 #define WFRAC_BITS 14
33 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
34 #define FIX(a) ((int)((a) * (1 << FRAC_BITS)))
35
36 #define SAMPLES_BUF_SIZE 4096
37
38 typedef struct MpegAudioContext {
39 PutBitContext pb;
40 int nb_channels;
41 int freq, bit_rate;
42 int lsf; /* 1 if mpeg2 low bitrate selected */
43 int bitrate_index; /* bit rate */
44 int freq_index;
45 int frame_size; /* frame size, in bits, without padding */
46 int64_t nb_samples; /* total number of samples encoded */
47 /* padding computation */
48 int frame_frac, frame_frac_incr, do_padding;
49 short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
50 int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
51 int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
52 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
53 /* code to group 3 scale factors */
54 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
55 int sblimit; /* number of used subbands */
56 const unsigned char *alloc_table;
57 } MpegAudioContext;
58
59 /* define it to use floats in quantization (I don't like floats !) */
60 //#define USE_FLOATS
61
62 #include "mpegaudiotab.h"
63
64 static int MPA_encode_init(AVCodecContext *avctx)
65 {
66 MpegAudioContext *s = avctx->priv_data;
67 int freq = avctx->sample_rate;
68 int bitrate = avctx->bit_rate;
69 int channels = avctx->channels;
70 int i, v, table;
71 float a;
72
73 if (channels > 2)
74 return -1;
75 bitrate = bitrate / 1000;
76 s->nb_channels = channels;
77 s->freq = freq;
78 s->bit_rate = bitrate * 1000;
79 avctx->frame_size = MPA_FRAME_SIZE;
80
81 /* encoding freq */
82 s->lsf = 0;
83 for(i=0;i<3;i++) {
84 if (mpa_freq_tab[i] == freq)
85 break;
86 if ((mpa_freq_tab[i] / 2) == freq) {
87 s->lsf = 1;
88 break;
89 }
90 }
91 if (i == 3){
92 av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
93 return -1;
94 }
95 s->freq_index = i;
96
97 /* encoding bitrate & frequency */
98 for(i=0;i<15;i++) {
99 if (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
100 break;
101 }
102 if (i == 15){
103 av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
104 return -1;
105 }
106 s->bitrate_index = i;
107
108 /* compute total header size & pad bit */
109
110 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
111 s->frame_size = ((int)a) * 8;
112
113 /* frame fractional size to compute padding */
114 s->frame_frac = 0;
115 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
116
117 /* select the right allocation table */
118 table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
119
120 /* number of used subbands */
121 s->sblimit = sblimit_table[table];
122 s->alloc_table = alloc_tables[table];
123
124 #ifdef DEBUG
125 av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
126 bitrate, freq, s->frame_size, table, s->frame_frac_incr);
127 #endif
128
129 for(i=0;i<s->nb_channels;i++)
130 s->samples_offset[i] = 0;
131
132 for(i=0;i<257;i++) {
133 int v;
134 v = mpa_enwindow[i];
135 #if WFRAC_BITS != 16
136 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
137 #endif
138 filter_bank[i] = v;
139 if ((i & 63) != 0)
140 v = -v;
141 if (i != 0)
142 filter_bank[512 - i] = v;
143 }
144
145 for(i=0;i<64;i++) {
146 v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
147 if (v <= 0)
148 v = 1;
149 scale_factor_table[i] = v;
150 #ifdef USE_FLOATS
151 scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
152 #else
153 #define P 15
154 scale_factor_shift[i] = 21 - P - (i / 3);
155 scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
156 #endif
157 }
158 for(i=0;i<128;i++) {
159 v = i - 64;
160 if (v <= -3)
161 v = 0;
162 else if (v < 0)
163 v = 1;
164 else if (v == 0)
165 v = 2;
166 else if (v < 3)
167 v = 3;
168 else
169 v = 4;
170 scale_diff_table[i] = v;
171 }
172
173 for(i=0;i<17;i++) {
174 v = quant_bits[i];
175 if (v < 0)
176 v = -v;
177 else
178 v = v * 3;
179 total_quant_bits[i] = 12 * v;
180 }
181
182 avctx->coded_frame= avcodec_alloc_frame();
183 avctx->coded_frame->key_frame= 1;
184
185 return 0;
186 }
187
188 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
189 static void idct32(int *out, int *tab)
190 {
191 int i, j;
192 int *t, *t1, xr;
193 const int *xp = costab32;
194
195 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
196
197 t = tab + 30;
198 t1 = tab + 2;
199 do {
200 t[0] += t[-4];
201 t[1] += t[1 - 4];
202 t -= 4;
203 } while (t != t1);
204
205 t = tab + 28;
206 t1 = tab + 4;
207 do {
208 t[0] += t[-8];
209 t[1] += t[1-8];
210 t[2] += t[2-8];
211 t[3] += t[3-8];
212 t -= 8;
213 } while (t != t1);
214
215 t = tab;
216 t1 = tab + 32;
217 do {
218 t[ 3] = -t[ 3];
219 t[ 6] = -t[ 6];
220
221 t[11] = -t[11];
222 t[12] = -t[12];
223 t[13] = -t[13];
224 t[15] = -t[15];
225 t += 16;
226 } while (t != t1);
227
228
229 t = tab;
230 t1 = tab + 8;
231 do {
232 int x1, x2, x3, x4;
233
234 x3 = MUL(t[16], FIX(SQRT2*0.5));
235 x4 = t[0] - x3;
236 x3 = t[0] + x3;
237
238 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
239 x1 = MUL((t[8] - x2), xp[0]);
240 x2 = MUL((t[8] + x2), xp[1]);
241
242 t[ 0] = x3 + x1;
243 t[ 8] = x4 - x2;
244 t[16] = x4 + x2;
245 t[24] = x3 - x1;
246 t++;
247 } while (t != t1);
248
249 xp += 2;
250 t = tab;
251 t1 = tab + 4;
252 do {
253 xr = MUL(t[28],xp[0]);
254 t[28] = (t[0] - xr);
255 t[0] = (t[0] + xr);
256
257 xr = MUL(t[4],xp[1]);
258 t[ 4] = (t[24] - xr);
259 t[24] = (t[24] + xr);
260
261 xr = MUL(t[20],xp[2]);
262 t[20] = (t[8] - xr);
263 t[ 8] = (t[8] + xr);
264
265 xr = MUL(t[12],xp[3]);
266 t[12] = (t[16] - xr);
267 t[16] = (t[16] + xr);
268 t++;
269 } while (t != t1);
270 xp += 4;
271
272 for (i = 0; i < 4; i++) {
273 xr = MUL(tab[30-i*4],xp[0]);
274 tab[30-i*4] = (tab[i*4] - xr);
275 tab[ i*4] = (tab[i*4] + xr);
276
277 xr = MUL(tab[ 2+i*4],xp[1]);
278 tab[ 2+i*4] = (tab[28-i*4] - xr);
279 tab[28-i*4] = (tab[28-i*4] + xr);
280
281 xr = MUL(tab[31-i*4],xp[0]);
282 tab[31-i*4] = (tab[1+i*4] - xr);
283 tab[ 1+i*4] = (tab[1+i*4] + xr);
284
285 xr = MUL(tab[ 3+i*4],xp[1]);
286 tab[ 3+i*4] = (tab[29-i*4] - xr);
287 tab[29-i*4] = (tab[29-i*4] + xr);
288
289 xp += 2;
290 }
291
292 t = tab + 30;
293 t1 = tab + 1;
294 do {
295 xr = MUL(t1[0], *xp);
296 t1[0] = (t[0] - xr);
297 t[0] = (t[0] + xr);
298 t -= 2;
299 t1 += 2;
300 xp++;
301 } while (t >= tab);
302
303 for(i=0;i<32;i++) {
304 out[i] = tab[bitinv32[i]];
305 }
306 }
307
308 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
309
310 static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
311 {
312 short *p, *q;
313 int sum, offset, i, j;
314 int tmp[64];
315 int tmp1[32];
316 int *out;
317
318 // print_pow1(samples, 1152);
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 // print_pow(s->sb_samples, 1152);
365 }
366
367 static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
368 unsigned char scale_factors[SBLIMIT][3],
369 int sb_samples[3][12][SBLIMIT],
370 int sblimit)
371 {
372 int *p, vmax, v, n, i, j, k, code;
373 int index, d1, d2;
374 unsigned char *sf = &scale_factors[0][0];
375
376 for(j=0;j<sblimit;j++) {
377 for(i=0;i<3;i++) {
378 /* find the max absolute value */
379 p = &sb_samples[i][0][j];
380 vmax = abs(*p);
381 for(k=1;k<12;k++) {
382 p += SBLIMIT;
383 v = abs(*p);
384 if (v > vmax)
385 vmax = v;
386 }
387 /* compute the scale factor index using log 2 computations */
388 if (vmax > 0) {
389 n = av_log2(vmax);
390 /* n is the position of the MSB of vmax. now
391 use at most 2 compares to find the index */
392 index = (21 - n) * 3 - 3;
393 if (index >= 0) {
394 while (vmax <= scale_factor_table[index+1])
395 index++;
396 } else {
397 index = 0; /* very unlikely case of overflow */
398 }
399 } else {
400 index = 62; /* value 63 is not allowed */
401 }
402
403 #if 0
404 printf("%2d:%d in=%x %x %d\n",
405 j, i, vmax, scale_factor_table[index], index);
406 #endif
407 /* store the scale factor */
408 assert(index >=0 && index <= 63);
409 sf[i] = index;
410 }
411
412 /* compute the transmission factor : look if the scale factors
413 are close enough to each other */
414 d1 = scale_diff_table[sf[0] - sf[1] + 64];
415 d2 = scale_diff_table[sf[1] - sf[2] + 64];
416
417 /* handle the 25 cases */
418 switch(d1 * 5 + d2) {
419 case 0*5+0:
420 case 0*5+4:
421 case 3*5+4:
422 case 4*5+0:
423 case 4*5+4:
424 code = 0;
425 break;
426 case 0*5+1:
427 case 0*5+2:
428 case 4*5+1:
429 case 4*5+2:
430 code = 3;
431 sf[2] = sf[1];
432 break;
433 case 0*5+3:
434 case 4*5+3:
435 code = 3;
436 sf[1] = sf[2];
437 break;
438 case 1*5+0:
439 case 1*5+4:
440 case 2*5+4:
441 code = 1;
442 sf[1] = sf[0];
443 break;
444 case 1*5+1:
445 case 1*5+2:
446 case 2*5+0:
447 case 2*5+1:
448 case 2*5+2:
449 code = 2;
450 sf[1] = sf[2] = sf[0];
451 break;
452 case 2*5+3:
453 case 3*5+3:
454 code = 2;
455 sf[0] = sf[1] = sf[2];
456 break;
457 case 3*5+0:
458 case 3*5+1:
459 case 3*5+2:
460 code = 2;
461 sf[0] = sf[2] = sf[1];
462 break;
463 case 1*5+3:
464 code = 2;
465 if (sf[0] > sf[2])
466 sf[0] = sf[2];
467 sf[1] = sf[2] = sf[0];
468 break;
469 default:
470 assert(0); //cant happen
471 code = 0; /* kill warning */
472 }
473
474 #if 0
475 printf("%d: %2d %2d %2d %d %d -> %d\n", j,
476 sf[0], sf[1], sf[2], d1, d2, code);
477 #endif
478 scale_code[j] = code;
479 sf += 3;
480 }
481 }
482
483 /* The most important function : psycho acoustic module. In this
484 encoder there is basically none, so this is the worst you can do,
485 but also this is the simpler. */
486 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
487 {
488 int i;
489
490 for(i=0;i<s->sblimit;i++) {
491 smr[i] = (int)(fixed_smr[i] * 10);
492 }
493 }
494
495
496 #define SB_NOTALLOCATED 0
497 #define SB_ALLOCATED 1
498 #define SB_NOMORE 2
499
500 /* Try to maximize the smr while using a number of bits inferior to
501 the frame size. I tried to make the code simpler, faster and
502 smaller than other encoders :-) */
503 static void compute_bit_allocation(MpegAudioContext *s,
504 short smr1[MPA_MAX_CHANNELS][SBLIMIT],
505 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
506 int *padding)
507 {
508 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
509 int incr;
510 short smr[MPA_MAX_CHANNELS][SBLIMIT];
511 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
512 const unsigned char *alloc;
513
514 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
515 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
516 memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
517
518 /* compute frame size and padding */
519 max_frame_size = s->frame_size;
520 s->frame_frac += s->frame_frac_incr;
521 if (s->frame_frac >= 65536) {
522 s->frame_frac -= 65536;
523 s->do_padding = 1;
524 max_frame_size += 8;
525 } else {
526 s->do_padding = 0;
527 }
528
529 /* compute the header + bit alloc size */
530 current_frame_size = 32;
531 alloc = s->alloc_table;
532 for(i=0;i<s->sblimit;i++) {
533 incr = alloc[0];
534 current_frame_size += incr * s->nb_channels;
535 alloc += 1 << incr;
536 }
537 for(;;) {
538 /* look for the subband with the largest signal to mask ratio */
539 max_sb = -1;
540 max_ch = -1;
541 max_smr = 0x80000000;
542 for(ch=0;ch<s->nb_channels;ch++) {
543 for(i=0;i<s->sblimit;i++) {
544 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
545 max_smr = smr[ch][i];
546 max_sb = i;
547 max_ch = ch;
548 }
549 }
550 }
551 #if 0
552 printf("current=%d max=%d max_sb=%d alloc=%d\n",
553 current_frame_size, max_frame_size, max_sb,
554 bit_alloc[max_sb]);
555 #endif
556 if (max_sb < 0)
557 break;
558
559 /* find alloc table entry (XXX: not optimal, should use
560 pointer table) */
561 alloc = s->alloc_table;
562 for(i=0;i<max_sb;i++) {
563 alloc += 1 << alloc[0];
564 }
565
566 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
567 /* nothing was coded for this band: add the necessary bits */
568 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
569 incr += total_quant_bits[alloc[1]];
570 } else {
571 /* increments bit allocation */
572 b = bit_alloc[max_ch][max_sb];
573 incr = total_quant_bits[alloc[b + 1]] -
574 total_quant_bits[alloc[b]];
575 }
576
577 if (current_frame_size + incr <= max_frame_size) {
578 /* can increase size */
579 b = ++bit_alloc[max_ch][max_sb];
580 current_frame_size += incr;
581 /* decrease smr by the resolution we added */
582 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
583 /* max allocation size reached ? */
584 if (b == ((1 << alloc[0]) - 1))
585 subband_status[max_ch][max_sb] = SB_NOMORE;
586 else
587 subband_status[max_ch][max_sb] = SB_ALLOCATED;
588 } else {
589 /* cannot increase the size of this subband */
590 subband_status[max_ch][max_sb] = SB_NOMORE;
591 }
592 }
593 *padding = max_frame_size - current_frame_size;
594 assert(*padding >= 0);
595
596 #if 0
597 for(i=0;i<s->sblimit;i++) {
598 printf("%d ", bit_alloc[i]);
599 }
600 printf("\n");
601 #endif
602 }
603
604 /*
605 * Output the mpeg audio layer 2 frame. Note how the code is small
606 * compared to other encoders :-)
607 */
608 static void encode_frame(MpegAudioContext *s,
609 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
610 int padding)
611 {
612 int i, j, k, l, bit_alloc_bits, b, ch;
613 unsigned char *sf;
614 int q[3];
615 PutBitContext *p = &s->pb;
616
617 /* header */
618
619 put_bits(p, 12, 0xfff);
620 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
621 put_bits(p, 2, 4-2); /* layer 2 */
622 put_bits(p, 1, 1); /* no error protection */
623 put_bits(p, 4, s->bitrate_index);
624 put_bits(p, 2, s->freq_index);
625 put_bits(p, 1, s->do_padding); /* use padding */
626 put_bits(p, 1, 0); /* private_bit */
627 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
628 put_bits(p, 2, 0); /* mode_ext */
629 put_bits(p, 1, 0); /* no copyright */
630 put_bits(p, 1, 1); /* original */
631 put_bits(p, 2, 0); /* no emphasis */
632
633 /* bit allocation */
634 j = 0;
635 for(i=0;i<s->sblimit;i++) {
636 bit_alloc_bits = s->alloc_table[j];
637 for(ch=0;ch<s->nb_channels;ch++) {
638 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
639 }
640 j += 1 << bit_alloc_bits;
641 }
642
643 /* scale codes */
644 for(i=0;i<s->sblimit;i++) {
645 for(ch=0;ch<s->nb_channels;ch++) {
646 if (bit_alloc[ch][i])
647 put_bits(p, 2, s->scale_code[ch][i]);
648 }
649 }
650
651 /* scale factors */
652 for(i=0;i<s->sblimit;i++) {
653 for(ch=0;ch<s->nb_channels;ch++) {
654 if (bit_alloc[ch][i]) {
655 sf = &s->scale_factors[ch][i][0];
656 switch(s->scale_code[ch][i]) {
657 case 0:
658 put_bits(p, 6, sf[0]);
659 put_bits(p, 6, sf[1]);
660 put_bits(p, 6, sf[2]);
661 break;
662 case 3:
663 case 1:
664 put_bits(p, 6, sf[0]);
665 put_bits(p, 6, sf[2]);
666 break;
667 case 2:
668 put_bits(p, 6, sf[0]);
669 break;
670 }
671 }
672 }
673 }
674
675 /* quantization & write sub band samples */
676
677 for(k=0;k<3;k++) {
678 for(l=0;l<12;l+=3) {
679 j = 0;
680 for(i=0;i<s->sblimit;i++) {
681 bit_alloc_bits = s->alloc_table[j];
682 for(ch=0;ch<s->nb_channels;ch++) {
683 b = bit_alloc[ch][i];
684 if (b) {
685 int qindex, steps, m, sample, bits;
686 /* we encode 3 sub band samples of the same sub band at a time */
687 qindex = s->alloc_table[j+b];
688 steps = quant_steps[qindex];
689 for(m=0;m<3;m++) {
690 sample = s->sb_samples[ch][k][l + m][i];
691 /* divide by scale factor */
692 #ifdef USE_FLOATS
693 {
694 float a;
695 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
696 q[m] = (int)((a + 1.0) * steps * 0.5);
697 }
698 #else
699 {
700 int q1, e, shift, mult;
701 e = s->scale_factors[ch][i][k];
702 shift = scale_factor_shift[e];
703 mult = scale_factor_mult[e];
704
705 /* normalize to P bits */
706 if (shift < 0)
707 q1 = sample << (-shift);
708 else
709 q1 = sample >> shift;
710 q1 = (q1 * mult) >> P;
711 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
712 }
713 #endif
714 if (q[m] >= steps)
715 q[m] = steps - 1;
716 assert(q[m] >= 0 && q[m] < steps);
717 }
718 bits = quant_bits[qindex];
719 if (bits < 0) {
720 /* group the 3 values to save bits */
721 put_bits(p, -bits,
722 q[0] + steps * (q[1] + steps * q[2]));
723 #if 0
724 printf("%d: gr1 %d\n",
725 i, q[0] + steps * (q[1] + steps * q[2]));
726 #endif
727 } else {
728 #if 0
729 printf("%d: gr3 %d %d %d\n",
730 i, q[0], q[1], q[2]);
731 #endif
732 put_bits(p, bits, q[0]);
733 put_bits(p, bits, q[1]);
734 put_bits(p, bits, q[2]);
735 }
736 }
737 }
738 /* next subband in alloc table */
739 j += 1 << bit_alloc_bits;
740 }
741 }
742 }
743
744 /* padding */
745 for(i=0;i<padding;i++)
746 put_bits(p, 1, 0);
747
748 /* flush */
749 flush_put_bits(p);
750 }
751
752 static int MPA_encode_frame(AVCodecContext *avctx,
753 unsigned char *frame, int buf_size, void *data)
754 {
755 MpegAudioContext *s = avctx->priv_data;
756 short *samples = data;
757 short smr[MPA_MAX_CHANNELS][SBLIMIT];
758 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
759 int padding, i;
760
761 for(i=0;i<s->nb_channels;i++) {
762 filter(s, i, samples + i, s->nb_channels);
763 }
764
765 for(i=0;i<s->nb_channels;i++) {
766 compute_scale_factors(s->scale_code[i], s->scale_factors[i],
767 s->sb_samples[i], s->sblimit);
768 }
769 for(i=0;i<s->nb_channels;i++) {
770 psycho_acoustic_model(s, smr[i]);
771 }
772 compute_bit_allocation(s, smr, bit_alloc, &padding);
773
774 init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
775
776 encode_frame(s, bit_alloc, padding);
777
778 s->nb_samples += MPA_FRAME_SIZE;
779 return pbBufPtr(&s->pb) - s->pb.buf;
780 }
781
782 static int MPA_encode_close(AVCodecContext *avctx)
783 {
784 av_freep(&avctx->coded_frame);
785 return 0;
786 }
787
788 AVCodec mp2_encoder = {
789 "mp2",
790 CODEC_TYPE_AUDIO,
791 CODEC_ID_MP2,
792 sizeof(MpegAudioContext),
793 MPA_encode_init,
794 MPA_encode_frame,
795 MPA_encode_close,
796 NULL,
797 };
798
799 #undef FIX
Libav master