Mercurial > libavcodec.hg
comparison mpegaudio.c @ 0:986e461dc072 libavcodec
Initial revision
author | glantau |
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date | Sun, 22 Jul 2001 14:18:56 +0000 |
parents | |
children | 5aa6292a1660 |
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-1:000000000000 | 0:986e461dc072 |
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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 | |
39 int 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 */ | |
161 static 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 | |
280 static 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 | |
351 static 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. */ | |
469 static 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 :-) */ | |
486 static 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 */ | |
591 static 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 | |
735 int 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 | |
766 AVCodec 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 }; |