more info about why init failed
[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 "mpegaudio.h"
27
28 /* currently, cannot change these constants (need to modify
29 quantization stage) */
30 #define FRAC_BITS 15
31 #define WFRAC_BITS 14
32 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
33 #define FIX(a) ((int)((a) * (1 << FRAC_BITS)))
34
35 #define SAMPLES_BUF_SIZE 4096
36
37 typedef struct MpegAudioContext {
38 PutBitContext pb;
39 int nb_channels;
40 int freq, bit_rate;
41 int lsf; /* 1 if mpeg2 low bitrate selected */
42 int bitrate_index; /* bit rate */
43 int freq_index;
44 int frame_size; /* frame size, in bits, without padding */
45 int64_t nb_samples; /* total number of samples encoded */
46 /* padding computation */
47 int frame_frac, frame_frac_incr, do_padding;
48 short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
49 int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
50 int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
51 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
52 /* code to group 3 scale factors */
53 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
54 int sblimit; /* number of used subbands */
55 const unsigned char *alloc_table;
56 } MpegAudioContext;
57
58 /* define it to use floats in quantization (I don't like floats !) */
59 //#define USE_FLOATS
60
61 #include "mpegaudiotab.h"
62
63 static int MPA_encode_init(AVCodecContext *avctx)
64 {
65 MpegAudioContext *s = avctx->priv_data;
66 int freq = avctx->sample_rate;
67 int bitrate = avctx->bit_rate;
68 int channels = avctx->channels;
69 int i, v, table;
70 float a;
71
72 if (channels > 2)
73 return -1;
74 bitrate = bitrate / 1000;
75 s->nb_channels = channels;
76 s->freq = freq;
77 s->bit_rate = bitrate * 1000;
78 avctx->frame_size = MPA_FRAME_SIZE;
79
80 /* encoding freq */
81 s->lsf = 0;
82 for(i=0;i<3;i++) {
83 if (mpa_freq_tab[i] == freq)
84 break;
85 if ((mpa_freq_tab[i] / 2) == freq) {
86 s->lsf = 1;
87 break;
88 }
89 }
90 if (i == 3){
91 av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
92 return -1;
93 }
94 s->freq_index = i;
95
96 /* encoding bitrate & frequency */
97 for(i=0;i<15;i++) {
98 if (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
99 break;
100 }
101 if (i == 15){
102 av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
103 return -1;
104 }
105 s->bitrate_index = i;
106
107 /* compute total header size & pad bit */
108
109 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
110 s->frame_size = ((int)a) * 8;
111
112 /* frame fractional size to compute padding */
113 s->frame_frac = 0;
114 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
115
116 /* select the right allocation table */
117 table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
118
119 /* number of used subbands */
120 s->sblimit = sblimit_table[table];
121 s->alloc_table = alloc_tables[table];
122
123 #ifdef DEBUG
124 av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
125 bitrate, freq, s->frame_size, table, s->frame_frac_incr);
126 #endif
127
128 for(i=0;i<s->nb_channels;i++)
129 s->samples_offset[i] = 0;
130
131 for(i=0;i<257;i++) {
132 int v;
133 v = mpa_enwindow[i];
134 #if WFRAC_BITS != 16
135 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
136 #endif
137 filter_bank[i] = v;
138 if ((i & 63) != 0)
139 v = -v;
140 if (i != 0)
141 filter_bank[512 - i] = v;
142 }
143
144 for(i=0;i<64;i++) {
145 v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
146 if (v <= 0)
147 v = 1;
148 scale_factor_table[i] = v;
149 #ifdef USE_FLOATS
150 scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
151 #else
152 #define P 15
153 scale_factor_shift[i] = 21 - P - (i / 3);
154 scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
155 #endif
156 }
157 for(i=0;i<128;i++) {
158 v = i - 64;
159 if (v <= -3)
160 v = 0;
161 else if (v < 0)
162 v = 1;
163 else if (v == 0)
164 v = 2;
165 else if (v < 3)
166 v = 3;
167 else
168 v = 4;
169 scale_diff_table[i] = v;
170 }
171
172 for(i=0;i<17;i++) {
173 v = quant_bits[i];
174 if (v < 0)
175 v = -v;
176 else
177 v = v * 3;
178 total_quant_bits[i] = 12 * v;
179 }
180
181 avctx->coded_frame= avcodec_alloc_frame();
182 avctx->coded_frame->key_frame= 1;
183
184 return 0;
185 }
186
187 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
188 static void idct32(int *out, int *tab)
189 {
190 int i, j;
191 int *t, *t1, xr;
192 const int *xp = costab32;
193
194 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
195
196 t = tab + 30;
197 t1 = tab + 2;
198 do {
199 t[0] += t[-4];
200 t[1] += t[1 - 4];
201 t -= 4;
202 } while (t != t1);
203
204 t = tab + 28;
205 t1 = tab + 4;
206 do {
207 t[0] += t[-8];
208 t[1] += t[1-8];
209 t[2] += t[2-8];
210 t[3] += t[3-8];
211 t -= 8;
212 } while (t != t1);
213
214 t = tab;
215 t1 = tab + 32;
216 do {
217 t[ 3] = -t[ 3];
218 t[ 6] = -t[ 6];
219
220 t[11] = -t[11];
221 t[12] = -t[12];
222 t[13] = -t[13];
223 t[15] = -t[15];
224 t += 16;
225 } while (t != t1);
226
227
228 t = tab;
229 t1 = tab + 8;
230 do {
231 int x1, x2, x3, x4;
232
233 x3 = MUL(t[16], FIX(SQRT2*0.5));
234 x4 = t[0] - x3;
235 x3 = t[0] + x3;
236
237 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
238 x1 = MUL((t[8] - x2), xp[0]);
239 x2 = MUL((t[8] + x2), xp[1]);
240
241 t[ 0] = x3 + x1;
242 t[ 8] = x4 - x2;
243 t[16] = x4 + x2;
244 t[24] = x3 - x1;
245 t++;
246 } while (t != t1);
247
248 xp += 2;
249 t = tab;
250 t1 = tab + 4;
251 do {
252 xr = MUL(t[28],xp[0]);
253 t[28] = (t[0] - xr);
254 t[0] = (t[0] + xr);
255
256 xr = MUL(t[4],xp[1]);
257 t[ 4] = (t[24] - xr);
258 t[24] = (t[24] + xr);
259
260 xr = MUL(t[20],xp[2]);
261 t[20] = (t[8] - xr);
262 t[ 8] = (t[8] + xr);
263
264 xr = MUL(t[12],xp[3]);
265 t[12] = (t[16] - xr);
266 t[16] = (t[16] + xr);
267 t++;
268 } while (t != t1);
269 xp += 4;
270
271 for (i = 0; i < 4; i++) {
272 xr = MUL(tab[30-i*4],xp[0]);
273 tab[30-i*4] = (tab[i*4] - xr);
274 tab[ i*4] = (tab[i*4] + xr);
275
276 xr = MUL(tab[ 2+i*4],xp[1]);
277 tab[ 2+i*4] = (tab[28-i*4] - xr);
278 tab[28-i*4] = (tab[28-i*4] + xr);
279
280 xr = MUL(tab[31-i*4],xp[0]);
281 tab[31-i*4] = (tab[1+i*4] - xr);
282 tab[ 1+i*4] = (tab[1+i*4] + xr);
283
284 xr = MUL(tab[ 3+i*4],xp[1]);
285 tab[ 3+i*4] = (tab[29-i*4] - xr);
286 tab[29-i*4] = (tab[29-i*4] + xr);
287
288 xp += 2;
289 }
290
291 t = tab + 30;
292 t1 = tab + 1;
293 do {
294 xr = MUL(t1[0], *xp);
295 t1[0] = (t[0] - xr);
296 t[0] = (t[0] + xr);
297 t -= 2;
298 t1 += 2;
299 xp++;
300 } while (t >= tab);
301
302 for(i=0;i<32;i++) {
303 out[i] = tab[bitinv32[i]];
304 }
305 }
306
307 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
308
309 static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
310 {
311 short *p, *q;
312 int sum, offset, i, j;
313 int tmp[64];
314 int tmp1[32];
315 int *out;
316
317 // print_pow1(samples, 1152);
318
319 offset = s->samples_offset[ch];
320 out = &s->sb_samples[ch][0][0][0];
321 for(j=0;j<36;j++) {
322 /* 32 samples at once */
323 for(i=0;i<32;i++) {
324 s->samples_buf[ch][offset + (31 - i)] = samples[0];
325 samples += incr;
326 }
327
328 /* filter */
329 p = s->samples_buf[ch] + offset;
330 q = filter_bank;
331 /* maxsum = 23169 */
332 for(i=0;i<64;i++) {
333 sum = p[0*64] * q[0*64];
334 sum += p[1*64] * q[1*64];
335 sum += p[2*64] * q[2*64];
336 sum += p[3*64] * q[3*64];
337 sum += p[4*64] * q[4*64];
338 sum += p[5*64] * q[5*64];
339 sum += p[6*64] * q[6*64];
340 sum += p[7*64] * q[7*64];
341 tmp[i] = sum;
342 p++;
343 q++;
344 }
345 tmp1[0] = tmp[16] >> WSHIFT;
346 for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
347 for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
348
349 idct32(out, tmp1);
350
351 /* advance of 32 samples */
352 offset -= 32;
353 out += 32;
354 /* handle the wrap around */
355 if (offset < 0) {
356 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
357 s->samples_buf[ch], (512 - 32) * 2);
358 offset = SAMPLES_BUF_SIZE - 512;
359 }
360 }
361 s->samples_offset[ch] = offset;
362
363 // print_pow(s->sb_samples, 1152);
364 }
365
366 static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
367 unsigned char scale_factors[SBLIMIT][3],
368 int sb_samples[3][12][SBLIMIT],
369 int sblimit)
370 {
371 int *p, vmax, v, n, i, j, k, code;
372 int index, d1, d2;
373 unsigned char *sf = &scale_factors[0][0];
374
375 for(j=0;j<sblimit;j++) {
376 for(i=0;i<3;i++) {
377 /* find the max absolute value */
378 p = &sb_samples[i][0][j];
379 vmax = abs(*p);
380 for(k=1;k<12;k++) {
381 p += SBLIMIT;
382 v = abs(*p);
383 if (v > vmax)
384 vmax = v;
385 }
386 /* compute the scale factor index using log 2 computations */
387 if (vmax > 0) {
388 n = av_log2(vmax);
389 /* n is the position of the MSB of vmax. now
390 use at most 2 compares to find the index */
391 index = (21 - n) * 3 - 3;
392 if (index >= 0) {
393 while (vmax <= scale_factor_table[index+1])
394 index++;
395 } else {
396 index = 0; /* very unlikely case of overflow */
397 }
398 } else {
399 index = 62; /* value 63 is not allowed */
400 }
401
402 #if 0
403 printf("%2d:%d in=%x %x %d\n",
404 j, i, vmax, scale_factor_table[index], index);
405 #endif
406 /* store the scale factor */
407 assert(index >=0 && index <= 63);
408 sf[i] = index;
409 }
410
411 /* compute the transmission factor : look if the scale factors
412 are close enough to each other */
413 d1 = scale_diff_table[sf[0] - sf[1] + 64];
414 d2 = scale_diff_table[sf[1] - sf[2] + 64];
415
416 /* handle the 25 cases */
417 switch(d1 * 5 + d2) {
418 case 0*5+0:
419 case 0*5+4:
420 case 3*5+4:
421 case 4*5+0:
422 case 4*5+4:
423 code = 0;
424 break;
425 case 0*5+1:
426 case 0*5+2:
427 case 4*5+1:
428 case 4*5+2:
429 code = 3;
430 sf[2] = sf[1];
431 break;
432 case 0*5+3:
433 case 4*5+3:
434 code = 3;
435 sf[1] = sf[2];
436 break;
437 case 1*5+0:
438 case 1*5+4:
439 case 2*5+4:
440 code = 1;
441 sf[1] = sf[0];
442 break;
443 case 1*5+1:
444 case 1*5+2:
445 case 2*5+0:
446 case 2*5+1:
447 case 2*5+2:
448 code = 2;
449 sf[1] = sf[2] = sf[0];
450 break;
451 case 2*5+3:
452 case 3*5+3:
453 code = 2;
454 sf[0] = sf[1] = sf[2];
455 break;
456 case 3*5+0:
457 case 3*5+1:
458 case 3*5+2:
459 code = 2;
460 sf[0] = sf[2] = sf[1];
461 break;
462 case 1*5+3:
463 code = 2;
464 if (sf[0] > sf[2])
465 sf[0] = sf[2];
466 sf[1] = sf[2] = sf[0];
467 break;
468 default:
469 av_abort();
470 }
471
472 #if 0
473 printf("%d: %2d %2d %2d %d %d -> %d\n", j,
474 sf[0], sf[1], sf[2], d1, d2, code);
475 #endif
476 scale_code[j] = code;
477 sf += 3;
478 }
479 }
480
481 /* The most important function : psycho acoustic module. In this
482 encoder there is basically none, so this is the worst you can do,
483 but also this is the simpler. */
484 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
485 {
486 int i;
487
488 for(i=0;i<s->sblimit;i++) {
489 smr[i] = (int)(fixed_smr[i] * 10);
490 }
491 }
492
493
494 #define SB_NOTALLOCATED 0
495 #define SB_ALLOCATED 1
496 #define SB_NOMORE 2
497
498 /* Try to maximize the smr while using a number of bits inferior to
499 the frame size. I tried to make the code simpler, faster and
500 smaller than other encoders :-) */
501 static void compute_bit_allocation(MpegAudioContext *s,
502 short smr1[MPA_MAX_CHANNELS][SBLIMIT],
503 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
504 int *padding)
505 {
506 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
507 int incr;
508 short smr[MPA_MAX_CHANNELS][SBLIMIT];
509 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
510 const unsigned char *alloc;
511
512 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
513 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
514 memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
515
516 /* compute frame size and padding */
517 max_frame_size = s->frame_size;
518 s->frame_frac += s->frame_frac_incr;
519 if (s->frame_frac >= 65536) {
520 s->frame_frac -= 65536;
521 s->do_padding = 1;
522 max_frame_size += 8;
523 } else {
524 s->do_padding = 0;
525 }
526
527 /* compute the header + bit alloc size */
528 current_frame_size = 32;
529 alloc = s->alloc_table;
530 for(i=0;i<s->sblimit;i++) {
531 incr = alloc[0];
532 current_frame_size += incr * s->nb_channels;
533 alloc += 1 << incr;
534 }
535 for(;;) {
536 /* look for the subband with the largest signal to mask ratio */
537 max_sb = -1;
538 max_ch = -1;
539 max_smr = 0x80000000;
540 for(ch=0;ch<s->nb_channels;ch++) {
541 for(i=0;i<s->sblimit;i++) {
542 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
543 max_smr = smr[ch][i];
544 max_sb = i;
545 max_ch = ch;
546 }
547 }
548 }
549 #if 0
550 printf("current=%d max=%d max_sb=%d alloc=%d\n",
551 current_frame_size, max_frame_size, max_sb,
552 bit_alloc[max_sb]);
553 #endif
554 if (max_sb < 0)
555 break;
556
557 /* find alloc table entry (XXX: not optimal, should use
558 pointer table) */
559 alloc = s->alloc_table;
560 for(i=0;i<max_sb;i++) {
561 alloc += 1 << alloc[0];
562 }
563
564 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
565 /* nothing was coded for this band: add the necessary bits */
566 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
567 incr += total_quant_bits[alloc[1]];
568 } else {
569 /* increments bit allocation */
570 b = bit_alloc[max_ch][max_sb];
571 incr = total_quant_bits[alloc[b + 1]] -
572 total_quant_bits[alloc[b]];
573 }
574
575 if (current_frame_size + incr <= max_frame_size) {
576 /* can increase size */
577 b = ++bit_alloc[max_ch][max_sb];
578 current_frame_size += incr;
579 /* decrease smr by the resolution we added */
580 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
581 /* max allocation size reached ? */
582 if (b == ((1 << alloc[0]) - 1))
583 subband_status[max_ch][max_sb] = SB_NOMORE;
584 else
585 subband_status[max_ch][max_sb] = SB_ALLOCATED;
586 } else {
587 /* cannot increase the size of this subband */
588 subband_status[max_ch][max_sb] = SB_NOMORE;
589 }
590 }
591 *padding = max_frame_size - current_frame_size;
592 assert(*padding >= 0);
593
594 #if 0
595 for(i=0;i<s->sblimit;i++) {
596 printf("%d ", bit_alloc[i]);
597 }
598 printf("\n");
599 #endif
600 }
601
602 /*
603 * Output the mpeg audio layer 2 frame. Note how the code is small
604 * compared to other encoders :-)
605 */
606 static void encode_frame(MpegAudioContext *s,
607 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
608 int padding)
609 {
610 int i, j, k, l, bit_alloc_bits, b, ch;
611 unsigned char *sf;
612 int q[3];
613 PutBitContext *p = &s->pb;
614
615 /* header */
616
617 put_bits(p, 12, 0xfff);
618 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
619 put_bits(p, 2, 4-2); /* layer 2 */
620 put_bits(p, 1, 1); /* no error protection */
621 put_bits(p, 4, s->bitrate_index);
622 put_bits(p, 2, s->freq_index);
623 put_bits(p, 1, s->do_padding); /* use padding */
624 put_bits(p, 1, 0); /* private_bit */
625 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
626 put_bits(p, 2, 0); /* mode_ext */
627 put_bits(p, 1, 0); /* no copyright */
628 put_bits(p, 1, 1); /* original */
629 put_bits(p, 2, 0); /* no emphasis */
630
631 /* bit allocation */
632 j = 0;
633 for(i=0;i<s->sblimit;i++) {
634 bit_alloc_bits = s->alloc_table[j];
635 for(ch=0;ch<s->nb_channels;ch++) {
636 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
637 }
638 j += 1 << bit_alloc_bits;
639 }
640
641 /* scale codes */
642 for(i=0;i<s->sblimit;i++) {
643 for(ch=0;ch<s->nb_channels;ch++) {
644 if (bit_alloc[ch][i])
645 put_bits(p, 2, s->scale_code[ch][i]);
646 }
647 }
648
649 /* scale factors */
650 for(i=0;i<s->sblimit;i++) {
651 for(ch=0;ch<s->nb_channels;ch++) {
652 if (bit_alloc[ch][i]) {
653 sf = &s->scale_factors[ch][i][0];
654 switch(s->scale_code[ch][i]) {
655 case 0:
656 put_bits(p, 6, sf[0]);
657 put_bits(p, 6, sf[1]);
658 put_bits(p, 6, sf[2]);
659 break;
660 case 3:
661 case 1:
662 put_bits(p, 6, sf[0]);
663 put_bits(p, 6, sf[2]);
664 break;
665 case 2:
666 put_bits(p, 6, sf[0]);
667 break;
668 }
669 }
670 }
671 }
672
673 /* quantization & write sub band samples */
674
675 for(k=0;k<3;k++) {
676 for(l=0;l<12;l+=3) {
677 j = 0;
678 for(i=0;i<s->sblimit;i++) {
679 bit_alloc_bits = s->alloc_table[j];
680 for(ch=0;ch<s->nb_channels;ch++) {
681 b = bit_alloc[ch][i];
682 if (b) {
683 int qindex, steps, m, sample, bits;
684 /* we encode 3 sub band samples of the same sub band at a time */
685 qindex = s->alloc_table[j+b];
686 steps = quant_steps[qindex];
687 for(m=0;m<3;m++) {
688 sample = s->sb_samples[ch][k][l + m][i];
689 /* divide by scale factor */
690 #ifdef USE_FLOATS
691 {
692 float a;
693 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
694 q[m] = (int)((a + 1.0) * steps * 0.5);
695 }
696 #else
697 {
698 int q1, e, shift, mult;
699 e = s->scale_factors[ch][i][k];
700 shift = scale_factor_shift[e];
701 mult = scale_factor_mult[e];
702
703 /* normalize to P bits */
704 if (shift < 0)
705 q1 = sample << (-shift);
706 else
707 q1 = sample >> shift;
708 q1 = (q1 * mult) >> P;
709 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
710 }
711 #endif
712 if (q[m] >= steps)
713 q[m] = steps - 1;
714 assert(q[m] >= 0 && q[m] < steps);
715 }
716 bits = quant_bits[qindex];
717 if (bits < 0) {
718 /* group the 3 values to save bits */
719 put_bits(p, -bits,
720 q[0] + steps * (q[1] + steps * q[2]));
721 #if 0
722 printf("%d: gr1 %d\n",
723 i, q[0] + steps * (q[1] + steps * q[2]));
724 #endif
725 } else {
726 #if 0
727 printf("%d: gr3 %d %d %d\n",
728 i, q[0], q[1], q[2]);
729 #endif
730 put_bits(p, bits, q[0]);
731 put_bits(p, bits, q[1]);
732 put_bits(p, bits, q[2]);
733 }
734 }
735 }
736 /* next subband in alloc table */
737 j += 1 << bit_alloc_bits;
738 }
739 }
740 }
741
742 /* padding */
743 for(i=0;i<padding;i++)
744 put_bits(p, 1, 0);
745
746 /* flush */
747 flush_put_bits(p);
748 }
749
750 static int MPA_encode_frame(AVCodecContext *avctx,
751 unsigned char *frame, int buf_size, void *data)
752 {
753 MpegAudioContext *s = avctx->priv_data;
754 short *samples = data;
755 short smr[MPA_MAX_CHANNELS][SBLIMIT];
756 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
757 int padding, i;
758
759 for(i=0;i<s->nb_channels;i++) {
760 filter(s, i, samples + i, s->nb_channels);
761 }
762
763 for(i=0;i<s->nb_channels;i++) {
764 compute_scale_factors(s->scale_code[i], s->scale_factors[i],
765 s->sb_samples[i], s->sblimit);
766 }
767 for(i=0;i<s->nb_channels;i++) {
768 psycho_acoustic_model(s, smr[i]);
769 }
770 compute_bit_allocation(s, smr, bit_alloc, &padding);
771
772 init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
773
774 encode_frame(s, bit_alloc, padding);
775
776 s->nb_samples += MPA_FRAME_SIZE;
777 return pbBufPtr(&s->pb) - s->pb.buf;
778 }
779
780 static int MPA_encode_close(AVCodecContext *avctx)
781 {
782 av_freep(&avctx->coded_frame);
783 return 0;
784 }
785
786 AVCodec mp2_encoder = {
787 "mp2",
788 CODEC_TYPE_AUDIO,
789 CODEC_ID_MP2,
790 sizeof(MpegAudioContext),
791 MPA_encode_init,
792 MPA_encode_frame,
793 MPA_encode_close,
794 NULL,
795 };
796
797 #undef FIX