Fix a crash caused by a null coded_picture pointer in ffserver.
[libav.git] / libavcodec / mpegaudio.c
CommitLineData
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1/*
2 * The simplest mpeg audio layer 2 encoder
ff4ec49e 3 * Copyright (c) 2000, 2001 Fabrice Bellard.
de6d9b64 4 *
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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.
de6d9b64 9 *
ff4ec49e 10 * This library is distributed in the hope that it will be useful,
de6d9b64 11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Lesser General Public License for more details.
de6d9b64 14 *
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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
de6d9b64 18 */
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19#include "avcodec.h"
20#include "mpegaudio.h"
21
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22/* currently, cannot change these constants (need to modify
23 quantization stage) */
24#define FRAC_BITS 15
25#define WFRAC_BITS 14
26#define MUL(a,b) (((INT64)(a) * (INT64)(b)) >> FRAC_BITS)
27#define FIX(a) ((int)((a) * (1 << FRAC_BITS)))
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28
29#define SAMPLES_BUF_SIZE 4096
30
31typedef struct MpegAudioContext {
32 PutBitContext pb;
33 int nb_channels;
34 int freq, bit_rate;
35 int lsf; /* 1 if mpeg2 low bitrate selected */
36 int bitrate_index; /* bit rate */
37 int freq_index;
38 int frame_size; /* frame size, in bits, without padding */
39 INT64 nb_samples; /* total number of samples encoded */
40 /* padding computation */
41 int frame_frac, frame_frac_incr, do_padding;
42 short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
43 int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
44 int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
45 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
46 /* code to group 3 scale factors */
47 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
48 int sblimit; /* number of used subbands */
49 const unsigned char *alloc_table;
50} MpegAudioContext;
51
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52/* define it to use floats in quantization (I don't like floats !) */
53//#define USE_FLOATS
54
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55#include "mpegaudiotab.h"
56
57int MPA_encode_init(AVCodecContext *avctx)
58{
59 MpegAudioContext *s = avctx->priv_data;
60 int freq = avctx->sample_rate;
61 int bitrate = avctx->bit_rate;
62 int channels = avctx->channels;
2456e28d 63 int i, v, table;
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64 float a;
65
66 if (channels > 2)
67 return -1;
68 bitrate = bitrate / 1000;
69 s->nb_channels = channels;
70 s->freq = freq;
71 s->bit_rate = bitrate * 1000;
72 avctx->frame_size = MPA_FRAME_SIZE;
73 avctx->key_frame = 1; /* always key frame */
74
75 /* encoding freq */
76 s->lsf = 0;
77 for(i=0;i<3;i++) {
2456e28d 78 if (mpa_freq_tab[i] == freq)
de6d9b64 79 break;
2456e28d 80 if ((mpa_freq_tab[i] / 2) == freq) {
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81 s->lsf = 1;
82 break;
83 }
84 }
85 if (i == 3)
86 return -1;
87 s->freq_index = i;
88
89 /* encoding bitrate & frequency */
90 for(i=0;i<15;i++) {
2456e28d 91 if (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
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92 break;
93 }
94 if (i == 15)
95 return -1;
96 s->bitrate_index = i;
97
98 /* compute total header size & pad bit */
99
100 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
101 s->frame_size = ((int)a) * 8;
102
103 /* frame fractional size to compute padding */
104 s->frame_frac = 0;
105 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
106
107 /* select the right allocation table */
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108 table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
109
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110 /* number of used subbands */
111 s->sblimit = sblimit_table[table];
112 s->alloc_table = alloc_tables[table];
113
114#ifdef DEBUG
115 printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
116 bitrate, freq, s->frame_size, table, s->frame_frac_incr);
117#endif
118
119 for(i=0;i<s->nb_channels;i++)
120 s->samples_offset[i] = 0;
121
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122 for(i=0;i<257;i++) {
123 int v;
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124 v = mpa_enwindow[i];
125#if WFRAC_BITS != 16
126 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
127#endif
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128 filter_bank[i] = v;
129 if ((i & 63) != 0)
130 v = -v;
131 if (i != 0)
132 filter_bank[512 - i] = v;
de6d9b64 133 }
2456e28d 134
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135 for(i=0;i<64;i++) {
136 v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
137 if (v <= 0)
138 v = 1;
139 scale_factor_table[i] = v;
140#ifdef USE_FLOATS
141 scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
142#else
143#define P 15
144 scale_factor_shift[i] = 21 - P - (i / 3);
145 scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
146#endif
147 }
148 for(i=0;i<128;i++) {
149 v = i - 64;
150 if (v <= -3)
151 v = 0;
152 else if (v < 0)
153 v = 1;
154 else if (v == 0)
155 v = 2;
156 else if (v < 3)
157 v = 3;
158 else
159 v = 4;
160 scale_diff_table[i] = v;
161 }
162
163 for(i=0;i<17;i++) {
164 v = quant_bits[i];
165 if (v < 0)
166 v = -v;
167 else
168 v = v * 3;
169 total_quant_bits[i] = 12 * v;
170 }
171
172 return 0;
173}
174
2456e28d 175/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
afa982fd 176static void idct32(int *out, int *tab)
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177{
178 int i, j;
179 int *t, *t1, xr;
180 const int *xp = costab32;
181
182 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
183
184 t = tab + 30;
185 t1 = tab + 2;
186 do {
187 t[0] += t[-4];
188 t[1] += t[1 - 4];
189 t -= 4;
190 } while (t != t1);
191
192 t = tab + 28;
193 t1 = tab + 4;
194 do {
195 t[0] += t[-8];
196 t[1] += t[1-8];
197 t[2] += t[2-8];
198 t[3] += t[3-8];
199 t -= 8;
200 } while (t != t1);
201
202 t = tab;
203 t1 = tab + 32;
204 do {
205 t[ 3] = -t[ 3];
206 t[ 6] = -t[ 6];
207
208 t[11] = -t[11];
209 t[12] = -t[12];
210 t[13] = -t[13];
211 t[15] = -t[15];
212 t += 16;
213 } while (t != t1);
214
215
216 t = tab;
217 t1 = tab + 8;
218 do {
219 int x1, x2, x3, x4;
220
221 x3 = MUL(t[16], FIX(SQRT2*0.5));
222 x4 = t[0] - x3;
223 x3 = t[0] + x3;
224
225 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
226 x1 = MUL((t[8] - x2), xp[0]);
227 x2 = MUL((t[8] + x2), xp[1]);
228
229 t[ 0] = x3 + x1;
230 t[ 8] = x4 - x2;
231 t[16] = x4 + x2;
232 t[24] = x3 - x1;
233 t++;
234 } while (t != t1);
235
236 xp += 2;
237 t = tab;
238 t1 = tab + 4;
239 do {
240 xr = MUL(t[28],xp[0]);
241 t[28] = (t[0] - xr);
242 t[0] = (t[0] + xr);
243
244 xr = MUL(t[4],xp[1]);
245 t[ 4] = (t[24] - xr);
246 t[24] = (t[24] + xr);
247
248 xr = MUL(t[20],xp[2]);
249 t[20] = (t[8] - xr);
250 t[ 8] = (t[8] + xr);
251
252 xr = MUL(t[12],xp[3]);
253 t[12] = (t[16] - xr);
254 t[16] = (t[16] + xr);
255 t++;
256 } while (t != t1);
257 xp += 4;
258
259 for (i = 0; i < 4; i++) {
260 xr = MUL(tab[30-i*4],xp[0]);
261 tab[30-i*4] = (tab[i*4] - xr);
262 tab[ i*4] = (tab[i*4] + xr);
263
264 xr = MUL(tab[ 2+i*4],xp[1]);
265 tab[ 2+i*4] = (tab[28-i*4] - xr);
266 tab[28-i*4] = (tab[28-i*4] + xr);
267
268 xr = MUL(tab[31-i*4],xp[0]);
269 tab[31-i*4] = (tab[1+i*4] - xr);
270 tab[ 1+i*4] = (tab[1+i*4] + xr);
271
272 xr = MUL(tab[ 3+i*4],xp[1]);
273 tab[ 3+i*4] = (tab[29-i*4] - xr);
274 tab[29-i*4] = (tab[29-i*4] + xr);
275
276 xp += 2;
277 }
278
279 t = tab + 30;
280 t1 = tab + 1;
281 do {
282 xr = MUL(t1[0], *xp);
283 t1[0] = (t[0] - xr);
284 t[0] = (t[0] + xr);
285 t -= 2;
286 t1 += 2;
287 xp++;
288 } while (t >= tab);
289
290 for(i=0;i<32;i++) {
afa982fd 291 out[i] = tab[bitinv32[i]];
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292 }
293}
294
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295#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
296
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297static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
298{
299 short *p, *q;
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300 int sum, offset, i, j;
301 int tmp[64];
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302 int tmp1[32];
303 int *out;
304
305 // print_pow1(samples, 1152);
306
307 offset = s->samples_offset[ch];
308 out = &s->sb_samples[ch][0][0][0];
309 for(j=0;j<36;j++) {
310 /* 32 samples at once */
311 for(i=0;i<32;i++) {
312 s->samples_buf[ch][offset + (31 - i)] = samples[0];
313 samples += incr;
314 }
315
316 /* filter */
317 p = s->samples_buf[ch] + offset;
318 q = filter_bank;
319 /* maxsum = 23169 */
320 for(i=0;i<64;i++) {
321 sum = p[0*64] * q[0*64];
322 sum += p[1*64] * q[1*64];
323 sum += p[2*64] * q[2*64];
324 sum += p[3*64] * q[3*64];
325 sum += p[4*64] * q[4*64];
326 sum += p[5*64] * q[5*64];
327 sum += p[6*64] * q[6*64];
328 sum += p[7*64] * q[7*64];
afa982fd 329 tmp[i] = sum;
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330 p++;
331 q++;
332 }
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333 tmp1[0] = tmp[16] >> WSHIFT;
334 for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
335 for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
de6d9b64 336
afa982fd 337 idct32(out, tmp1);
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338
339 /* advance of 32 samples */
340 offset -= 32;
341 out += 32;
342 /* handle the wrap around */
343 if (offset < 0) {
344 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
345 s->samples_buf[ch], (512 - 32) * 2);
346 offset = SAMPLES_BUF_SIZE - 512;
347 }
348 }
349 s->samples_offset[ch] = offset;
350
351 // print_pow(s->sb_samples, 1152);
352}
353
354static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
355 unsigned char scale_factors[SBLIMIT][3],
356 int sb_samples[3][12][SBLIMIT],
357 int sblimit)
358{
359 int *p, vmax, v, n, i, j, k, code;
360 int index, d1, d2;
361 unsigned char *sf = &scale_factors[0][0];
362
363 for(j=0;j<sblimit;j++) {
364 for(i=0;i<3;i++) {
365 /* find the max absolute value */
366 p = &sb_samples[i][0][j];
367 vmax = abs(*p);
368 for(k=1;k<12;k++) {
369 p += SBLIMIT;
370 v = abs(*p);
371 if (v > vmax)
372 vmax = v;
373 }
374 /* compute the scale factor index using log 2 computations */
375 if (vmax > 0) {
935442b5 376 n = av_log2(vmax);
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377 /* n is the position of the MSB of vmax. now
378 use at most 2 compares to find the index */
379 index = (21 - n) * 3 - 3;
380 if (index >= 0) {
381 while (vmax <= scale_factor_table[index+1])
382 index++;
383 } else {
384 index = 0; /* very unlikely case of overflow */
385 }
386 } else {
afa982fd 387 index = 62; /* value 63 is not allowed */
de6d9b64 388 }
afa982fd 389
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390#if 0
391 printf("%2d:%d in=%x %x %d\n",
392 j, i, vmax, scale_factor_table[index], index);
393#endif
394 /* store the scale factor */
395 assert(index >=0 && index <= 63);
396 sf[i] = index;
397 }
398
399 /* compute the transmission factor : look if the scale factors
400 are close enough to each other */
401 d1 = scale_diff_table[sf[0] - sf[1] + 64];
402 d2 = scale_diff_table[sf[1] - sf[2] + 64];
403
404 /* handle the 25 cases */
405 switch(d1 * 5 + d2) {
406 case 0*5+0:
407 case 0*5+4:
408 case 3*5+4:
409 case 4*5+0:
410 case 4*5+4:
411 code = 0;
412 break;
413 case 0*5+1:
414 case 0*5+2:
415 case 4*5+1:
416 case 4*5+2:
417 code = 3;
418 sf[2] = sf[1];
419 break;
420 case 0*5+3:
421 case 4*5+3:
422 code = 3;
423 sf[1] = sf[2];
424 break;
425 case 1*5+0:
426 case 1*5+4:
427 case 2*5+4:
428 code = 1;
429 sf[1] = sf[0];
430 break;
431 case 1*5+1:
432 case 1*5+2:
433 case 2*5+0:
434 case 2*5+1:
435 case 2*5+2:
436 code = 2;
437 sf[1] = sf[2] = sf[0];
438 break;
439 case 2*5+3:
440 case 3*5+3:
441 code = 2;
442 sf[0] = sf[1] = sf[2];
443 break;
444 case 3*5+0:
445 case 3*5+1:
446 case 3*5+2:
447 code = 2;
448 sf[0] = sf[2] = sf[1];
449 break;
450 case 1*5+3:
451 code = 2;
452 if (sf[0] > sf[2])
453 sf[0] = sf[2];
454 sf[1] = sf[2] = sf[0];
455 break;
456 default:
02ac3136 457 av_abort();
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458 }
459
460#if 0
461 printf("%d: %2d %2d %2d %d %d -> %d\n", j,
462 sf[0], sf[1], sf[2], d1, d2, code);
463#endif
464 scale_code[j] = code;
465 sf += 3;
466 }
467}
468
469/* The most important function : psycho acoustic module. In this
470 encoder there is basically none, so this is the worst you can do,
471 but also this is the simpler. */
472static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
473{
474 int i;
475
476 for(i=0;i<s->sblimit;i++) {
477 smr[i] = (int)(fixed_smr[i] * 10);
478 }
479}
480
481
482#define SB_NOTALLOCATED 0
483#define SB_ALLOCATED 1
484#define SB_NOMORE 2
485
486/* Try to maximize the smr while using a number of bits inferior to
487 the frame size. I tried to make the code simpler, faster and
488 smaller than other encoders :-) */
489static void compute_bit_allocation(MpegAudioContext *s,
490 short smr1[MPA_MAX_CHANNELS][SBLIMIT],
491 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
492 int *padding)
493{
494 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
495 int incr;
496 short smr[MPA_MAX_CHANNELS][SBLIMIT];
497 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
498 const unsigned char *alloc;
499
500 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
501 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
502 memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
503
504 /* compute frame size and padding */
505 max_frame_size = s->frame_size;
506 s->frame_frac += s->frame_frac_incr;
507 if (s->frame_frac >= 65536) {
508 s->frame_frac -= 65536;
509 s->do_padding = 1;
510 max_frame_size += 8;
511 } else {
512 s->do_padding = 0;
513 }
514
515 /* compute the header + bit alloc size */
516 current_frame_size = 32;
517 alloc = s->alloc_table;
518 for(i=0;i<s->sblimit;i++) {
519 incr = alloc[0];
520 current_frame_size += incr * s->nb_channels;
521 alloc += 1 << incr;
522 }
523 for(;;) {
524 /* look for the subband with the largest signal to mask ratio */
525 max_sb = -1;
526 max_ch = -1;
527 max_smr = 0x80000000;
528 for(ch=0;ch<s->nb_channels;ch++) {
529 for(i=0;i<s->sblimit;i++) {
530 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
531 max_smr = smr[ch][i];
532 max_sb = i;
533 max_ch = ch;
534 }
535 }
536 }
537#if 0
538 printf("current=%d max=%d max_sb=%d alloc=%d\n",
539 current_frame_size, max_frame_size, max_sb,
540 bit_alloc[max_sb]);
541#endif
542 if (max_sb < 0)
543 break;
544
545 /* find alloc table entry (XXX: not optimal, should use
546 pointer table) */
547 alloc = s->alloc_table;
548 for(i=0;i<max_sb;i++) {
549 alloc += 1 << alloc[0];
550 }
551
552 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
553 /* nothing was coded for this band: add the necessary bits */
554 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
555 incr += total_quant_bits[alloc[1]];
556 } else {
557 /* increments bit allocation */
558 b = bit_alloc[max_ch][max_sb];
559 incr = total_quant_bits[alloc[b + 1]] -
560 total_quant_bits[alloc[b]];
561 }
562
563 if (current_frame_size + incr <= max_frame_size) {
564 /* can increase size */
565 b = ++bit_alloc[max_ch][max_sb];
566 current_frame_size += incr;
567 /* decrease smr by the resolution we added */
568 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
569 /* max allocation size reached ? */
570 if (b == ((1 << alloc[0]) - 1))
571 subband_status[max_ch][max_sb] = SB_NOMORE;
572 else
573 subband_status[max_ch][max_sb] = SB_ALLOCATED;
574 } else {
575 /* cannot increase the size of this subband */
576 subband_status[max_ch][max_sb] = SB_NOMORE;
577 }
578 }
579 *padding = max_frame_size - current_frame_size;
580 assert(*padding >= 0);
581
582#if 0
583 for(i=0;i<s->sblimit;i++) {
584 printf("%d ", bit_alloc[i]);
585 }
586 printf("\n");
587#endif
588}
589
590/*
591 * Output the mpeg audio layer 2 frame. Note how the code is small
592 * compared to other encoders :-)
593 */
594static void encode_frame(MpegAudioContext *s,
595 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
596 int padding)
597{
598 int i, j, k, l, bit_alloc_bits, b, ch;
599 unsigned char *sf;
600 int q[3];
601 PutBitContext *p = &s->pb;
602
603 /* header */
604
605 put_bits(p, 12, 0xfff);
606 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
607 put_bits(p, 2, 4-2); /* layer 2 */
608 put_bits(p, 1, 1); /* no error protection */
609 put_bits(p, 4, s->bitrate_index);
610 put_bits(p, 2, s->freq_index);
611 put_bits(p, 1, s->do_padding); /* use padding */
612 put_bits(p, 1, 0); /* private_bit */
613 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
614 put_bits(p, 2, 0); /* mode_ext */
615 put_bits(p, 1, 0); /* no copyright */
616 put_bits(p, 1, 1); /* original */
617 put_bits(p, 2, 0); /* no emphasis */
618
619 /* bit allocation */
620 j = 0;
621 for(i=0;i<s->sblimit;i++) {
622 bit_alloc_bits = s->alloc_table[j];
623 for(ch=0;ch<s->nb_channels;ch++) {
624 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
625 }
626 j += 1 << bit_alloc_bits;
627 }
628
629 /* scale codes */
630 for(i=0;i<s->sblimit;i++) {
631 for(ch=0;ch<s->nb_channels;ch++) {
632 if (bit_alloc[ch][i])
633 put_bits(p, 2, s->scale_code[ch][i]);
634 }
635 }
636
637 /* scale factors */
638 for(i=0;i<s->sblimit;i++) {
639 for(ch=0;ch<s->nb_channels;ch++) {
640 if (bit_alloc[ch][i]) {
641 sf = &s->scale_factors[ch][i][0];
642 switch(s->scale_code[ch][i]) {
643 case 0:
644 put_bits(p, 6, sf[0]);
645 put_bits(p, 6, sf[1]);
646 put_bits(p, 6, sf[2]);
647 break;
648 case 3:
649 case 1:
650 put_bits(p, 6, sf[0]);
651 put_bits(p, 6, sf[2]);
652 break;
653 case 2:
654 put_bits(p, 6, sf[0]);
655 break;
656 }
657 }
658 }
659 }
660
661 /* quantization & write sub band samples */
662
663 for(k=0;k<3;k++) {
664 for(l=0;l<12;l+=3) {
665 j = 0;
666 for(i=0;i<s->sblimit;i++) {
667 bit_alloc_bits = s->alloc_table[j];
668 for(ch=0;ch<s->nb_channels;ch++) {
669 b = bit_alloc[ch][i];
670 if (b) {
671 int qindex, steps, m, sample, bits;
672 /* we encode 3 sub band samples of the same sub band at a time */
673 qindex = s->alloc_table[j+b];
674 steps = quant_steps[qindex];
675 for(m=0;m<3;m++) {
676 sample = s->sb_samples[ch][k][l + m][i];
677 /* divide by scale factor */
678#ifdef USE_FLOATS
679 {
680 float a;
681 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
682 q[m] = (int)((a + 1.0) * steps * 0.5);
683 }
684#else
685 {
686 int q1, e, shift, mult;
687 e = s->scale_factors[ch][i][k];
688 shift = scale_factor_shift[e];
689 mult = scale_factor_mult[e];
690
691 /* normalize to P bits */
692 if (shift < 0)
693 q1 = sample << (-shift);
694 else
695 q1 = sample >> shift;
696 q1 = (q1 * mult) >> P;
697 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
698 }
699#endif
700 if (q[m] >= steps)
701 q[m] = steps - 1;
702 assert(q[m] >= 0 && q[m] < steps);
703 }
704 bits = quant_bits[qindex];
705 if (bits < 0) {
706 /* group the 3 values to save bits */
707 put_bits(p, -bits,
708 q[0] + steps * (q[1] + steps * q[2]));
709#if 0
710 printf("%d: gr1 %d\n",
711 i, q[0] + steps * (q[1] + steps * q[2]));
712#endif
713 } else {
714#if 0
715 printf("%d: gr3 %d %d %d\n",
716 i, q[0], q[1], q[2]);
717#endif
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
738int MPA_encode_frame(AVCodecContext *avctx,
739 unsigned char *frame, int buf_size, void *data)
740{
741 MpegAudioContext *s = avctx->priv_data;
742 short *samples = data;
743 short smr[MPA_MAX_CHANNELS][SBLIMIT];
744 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
745 int padding, i;
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++) {
752 compute_scale_factors(s->scale_code[i], s->scale_factors[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 init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
761
762 encode_frame(s, bit_alloc, padding);
763
764 s->nb_samples += MPA_FRAME_SIZE;
17592475 765 return pbBufPtr(&s->pb) - s->pb.buf;
de6d9b64
FB
766}
767
768
769AVCodec mp2_encoder = {
770 "mp2",
771 CODEC_TYPE_AUDIO,
772 CODEC_ID_MP2,
773 sizeof(MpegAudioContext),
774 MPA_encode_init,
775 MPA_encode_frame,
776 NULL,
777};
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778
779#undef FIX