1/* 2 * The simplest mpeg audio layer 2 encoder 3 * Copyright (c) 2000, 2001 Fabrice Bellard 4 * 5 * This file is part of FFmpeg. 6 * 7 * FFmpeg is free software; you can redistribute it and/or 8 * modify it under the terms of the GNU Lesser General Public 9 * License as published by the Free Software Foundation; either 10 * version 2.1 of the License, or (at your option) any later version. 11 * 12 * FFmpeg is distributed in the hope that it will be useful, 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of 14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 15 * Lesser General Public License for more details. 16 * 17 * You should have received a copy of the GNU Lesser General Public 18 * License along with FFmpeg; if not, write to the Free Software 19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 20 */ 21 22/** 23 * @file libavcodec/mpegaudio.c 24 * The simplest mpeg audio layer 2 encoder. 25 */ 26 27#include "avcodec.h" 28#include "bitstream.h" 29 30#undef CONFIG_MPEGAUDIO_HP 31#define CONFIG_MPEGAUDIO_HP 0 32#include "mpegaudio.h" 33 34/* currently, cannot change these constants (need to modify 35 quantization stage) */ 36#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS) 37 38#define SAMPLES_BUF_SIZE 4096 39 40typedef struct MpegAudioContext { 41 PutBitContext pb; 42 int nb_channels; 43 int freq, bit_rate; 44 int lsf; /* 1 if mpeg2 low bitrate selected */ 45 int bitrate_index; /* bit rate */ 46 int freq_index; 47 int frame_size; /* frame size, in bits, without padding */ 48 int64_t nb_samples; /* total number of samples encoded */ 49 /* padding computation */ 50 int frame_frac, frame_frac_incr, do_padding; 51 short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */ 52 int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */ 53 int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT]; 54 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */ 55 /* code to group 3 scale factors */ 56 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; 57 int sblimit; /* number of used subbands */ 58 const unsigned char *alloc_table; 59} MpegAudioContext; 60 61/* define it to use floats in quantization (I don't like floats !) */ 62//#define USE_FLOATS 63 64#include "mpegaudiodata.h" 65#include "mpegaudiotab.h" 66 67static av_cold int MPA_encode_init(AVCodecContext *avctx) 68{ 69 MpegAudioContext *s = avctx->priv_data; 70 int freq = avctx->sample_rate; 71 int bitrate = avctx->bit_rate; 72 int channels = avctx->channels; 73 int i, v, table; 74 float a; 75 76 if (channels <= 0 || channels > 2){ 77 av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels); 78 return -1; 79 } 80 bitrate = bitrate / 1000; 81 s->nb_channels = channels; 82 s->freq = freq; 83 s->bit_rate = bitrate * 1000; 84 avctx->frame_size = MPA_FRAME_SIZE; 85 86 /* encoding freq */ 87 s->lsf = 0; 88 for(i=0;i<3;i++) { 89 if (ff_mpa_freq_tab[i] == freq) 90 break; 91 if ((ff_mpa_freq_tab[i] / 2) == freq) { 92 s->lsf = 1; 93 break; 94 } 95 } 96 if (i == 3){ 97 av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq); 98 return -1; 99 } 100 s->freq_index = i; 101 102 /* encoding bitrate & frequency */ 103 for(i=0;i<15;i++) { 104 if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate) 105 break; 106 } 107 if (i == 15){ 108 av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate); 109 return -1; 110 } 111 s->bitrate_index = i; 112 113 /* compute total header size & pad bit */ 114 115 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0); 116 s->frame_size = ((int)a) * 8; 117 118 /* frame fractional size to compute padding */ 119 s->frame_frac = 0; 120 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0); 121 122 /* select the right allocation table */ 123 table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf); 124 125 /* number of used subbands */ 126 s->sblimit = ff_mpa_sblimit_table[table]; 127 s->alloc_table = ff_mpa_alloc_tables[table]; 128 129#ifdef DEBUG 130 av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", 131 bitrate, freq, s->frame_size, table, s->frame_frac_incr); 132#endif 133 134 for(i=0;i<s->nb_channels;i++) 135 s->samples_offset[i] = 0; 136 137 for(i=0;i<257;i++) { 138 int v; 139 v = ff_mpa_enwindow[i]; 140#if WFRAC_BITS != 16 141 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS); 142#endif 143 filter_bank[i] = v; 144 if ((i & 63) != 0) 145 v = -v; 146 if (i != 0) 147 filter_bank[512 - i] = v; 148 } 149 150 for(i=0;i<64;i++) { 151 v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20)); 152 if (v <= 0) 153 v = 1; 154 scale_factor_table[i] = v; 155#ifdef USE_FLOATS 156 scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20); 157#else 158#define P 15 159 scale_factor_shift[i] = 21 - P - (i / 3); 160 scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0); 161#endif 162 } 163 for(i=0;i<128;i++) { 164 v = i - 64; 165 if (v <= -3) 166 v = 0; 167 else if (v < 0) 168 v = 1; 169 else if (v == 0) 170 v = 2; 171 else if (v < 3) 172 v = 3; 173 else 174 v = 4; 175 scale_diff_table[i] = v; 176 } 177 178 for(i=0;i<17;i++) { 179 v = ff_mpa_quant_bits[i]; 180 if (v < 0) 181 v = -v; 182 else 183 v = v * 3; 184 total_quant_bits[i] = 12 * v; 185 } 186 187 avctx->coded_frame= avcodec_alloc_frame(); 188 avctx->coded_frame->key_frame= 1; 189 190 return 0; 191} 192 193/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */ 194static void idct32(int *out, int *tab) 195{ 196 int i, j; 197 int *t, *t1, xr; 198 const int *xp = costab32; 199 200 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2]; 201 202 t = tab + 30; 203 t1 = tab + 2; 204 do { 205 t[0] += t[-4]; 206 t[1] += t[1 - 4]; 207 t -= 4; 208 } while (t != t1); 209 210 t = tab + 28; 211 t1 = tab + 4; 212 do { 213 t[0] += t[-8]; 214 t[1] += t[1-8]; 215 t[2] += t[2-8]; 216 t[3] += t[3-8]; 217 t -= 8; 218 } while (t != t1); 219 220 t = tab; 221 t1 = tab + 32; 222 do { 223 t[ 3] = -t[ 3]; 224 t[ 6] = -t[ 6]; 225 226 t[11] = -t[11]; 227 t[12] = -t[12]; 228 t[13] = -t[13]; 229 t[15] = -t[15]; 230 t += 16; 231 } while (t != t1); 232 233 234 t = tab; 235 t1 = tab + 8; 236 do { 237 int x1, x2, x3, x4; 238 239 x3 = MUL(t[16], FIX(SQRT2*0.5)); 240 x4 = t[0] - x3; 241 x3 = t[0] + x3; 242 243 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5)); 244 x1 = MUL((t[8] - x2), xp[0]); 245 x2 = MUL((t[8] + x2), xp[1]); 246 247 t[ 0] = x3 + x1; 248 t[ 8] = x4 - x2; 249 t[16] = x4 + x2; 250 t[24] = x3 - x1; 251 t++; 252 } while (t != t1); 253 254 xp += 2; 255 t = tab; 256 t1 = tab + 4; 257 do { 258 xr = MUL(t[28],xp[0]); 259 t[28] = (t[0] - xr); 260 t[0] = (t[0] + xr); 261 262 xr = MUL(t[4],xp[1]); 263 t[ 4] = (t[24] - xr); 264 t[24] = (t[24] + xr); 265 266 xr = MUL(t[20],xp[2]); 267 t[20] = (t[8] - xr); 268 t[ 8] = (t[8] + xr); 269 270 xr = MUL(t[12],xp[3]); 271 t[12] = (t[16] - xr); 272 t[16] = (t[16] + xr); 273 t++; 274 } while (t != t1); 275 xp += 4; 276 277 for (i = 0; i < 4; i++) { 278 xr = MUL(tab[30-i*4],xp[0]); 279 tab[30-i*4] = (tab[i*4] - xr); 280 tab[ i*4] = (tab[i*4] + xr); 281 282 xr = MUL(tab[ 2+i*4],xp[1]); 283 tab[ 2+i*4] = (tab[28-i*4] - xr); 284 tab[28-i*4] = (tab[28-i*4] + xr); 285 286 xr = MUL(tab[31-i*4],xp[0]); 287 tab[31-i*4] = (tab[1+i*4] - xr); 288 tab[ 1+i*4] = (tab[1+i*4] + xr); 289 290 xr = MUL(tab[ 3+i*4],xp[1]); 291 tab[ 3+i*4] = (tab[29-i*4] - xr); 292 tab[29-i*4] = (tab[29-i*4] + xr); 293 294 xp += 2; 295 } 296 297 t = tab + 30; 298 t1 = tab + 1; 299 do { 300 xr = MUL(t1[0], *xp); 301 t1[0] = (t[0] - xr); 302 t[0] = (t[0] + xr); 303 t -= 2; 304 t1 += 2; 305 xp++; 306 } while (t >= tab); 307 308 for(i=0;i<32;i++) { 309 out[i] = tab[bitinv32[i]]; 310 } 311} 312 313#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS) 314 315static void filter(MpegAudioContext *s, int ch, short *samples, int incr) 316{ 317 short *p, *q; 318 int sum, offset, i, j; 319 int tmp[64]; 320 int tmp1[32]; 321 int *out; 322 323 // print_pow1(samples, 1152); 324 325 offset = s->samples_offset[ch]; 326 out = &s->sb_samples[ch][0][0][0]; 327 for(j=0;j<36;j++) { 328 /* 32 samples at once */ 329 for(i=0;i<32;i++) { 330 s->samples_buf[ch][offset + (31 - i)] = samples[0]; 331 samples += incr; 332 } 333 334 /* filter */ 335 p = s->samples_buf[ch] + offset; 336 q = filter_bank; 337 /* maxsum = 23169 */ 338 for(i=0;i<64;i++) { 339 sum = p[0*64] * q[0*64]; 340 sum += p[1*64] * q[1*64]; 341 sum += p[2*64] * q[2*64]; 342 sum += p[3*64] * q[3*64]; 343 sum += p[4*64] * q[4*64]; 344 sum += p[5*64] * q[5*64]; 345 sum += p[6*64] * q[6*64]; 346 sum += p[7*64] * q[7*64]; 347 tmp[i] = sum; 348 p++; 349 q++; 350 } 351 tmp1[0] = tmp[16] >> WSHIFT; 352 for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT; 353 for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT; 354 355 idct32(out, tmp1); 356 357 /* advance of 32 samples */ 358 offset -= 32; 359 out += 32; 360 /* handle the wrap around */ 361 if (offset < 0) { 362 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32), 363 s->samples_buf[ch], (512 - 32) * 2); 364 offset = SAMPLES_BUF_SIZE - 512; 365 } 366 } 367 s->samples_offset[ch] = offset; 368 369 // print_pow(s->sb_samples, 1152); 370} 371 372static void compute_scale_factors(unsigned char scale_code[SBLIMIT], 373 unsigned char scale_factors[SBLIMIT][3], 374 int sb_samples[3][12][SBLIMIT], 375 int sblimit) 376{ 377 int *p, vmax, v, n, i, j, k, code; 378 int index, d1, d2; 379 unsigned char *sf = &scale_factors[0][0]; 380 381 for(j=0;j<sblimit;j++) { 382 for(i=0;i<3;i++) { 383 /* find the max absolute value */ 384 p = &sb_samples[i][0][j]; 385 vmax = abs(*p); 386 for(k=1;k<12;k++) { 387 p += SBLIMIT; 388 v = abs(*p); 389 if (v > vmax) 390 vmax = v; 391 } 392 /* compute the scale factor index using log 2 computations */ 393 if (vmax > 1) { 394 n = av_log2(vmax); 395 /* n is the position of the MSB of vmax. now 396 use at most 2 compares to find the index */ 397 index = (21 - n) * 3 - 3; 398 if (index >= 0) { 399 while (vmax <= scale_factor_table[index+1]) 400 index++; 401 } else { 402 index = 0; /* very unlikely case of overflow */ 403 } 404 } else { 405 index = 62; /* value 63 is not allowed */ 406 } 407 408#if 0 409 printf("%2d:%d in=%x %x %d\n", 410 j, i, vmax, scale_factor_table[index], index); 411#endif 412 /* store the scale factor */ 413 assert(index >=0 && index <= 63); 414 sf[i] = index; 415 } 416 417 /* compute the transmission factor : look if the scale factors 418 are close enough to each other */ 419 d1 = scale_diff_table[sf[0] - sf[1] + 64]; 420 d2 = scale_diff_table[sf[1] - sf[2] + 64]; 421 422 /* handle the 25 cases */ 423 switch(d1 * 5 + d2) { 424 case 0*5+0: 425 case 0*5+4: 426 case 3*5+4: 427 case 4*5+0: 428 case 4*5+4: 429 code = 0; 430 break; 431 case 0*5+1: 432 case 0*5+2: 433 case 4*5+1: 434 case 4*5+2: 435 code = 3; 436 sf[2] = sf[1]; 437 break; 438 case 0*5+3: 439 case 4*5+3: 440 code = 3; 441 sf[1] = sf[2]; 442 break; 443 case 1*5+0: 444 case 1*5+4: 445 case 2*5+4: 446 code = 1; 447 sf[1] = sf[0]; 448 break; 449 case 1*5+1: 450 case 1*5+2: 451 case 2*5+0: 452 case 2*5+1: 453 case 2*5+2: 454 code = 2; 455 sf[1] = sf[2] = sf[0]; 456 break; 457 case 2*5+3: 458 case 3*5+3: 459 code = 2; 460 sf[0] = sf[1] = sf[2]; 461 break; 462 case 3*5+0: 463 case 3*5+1: 464 case 3*5+2: 465 code = 2; 466 sf[0] = sf[2] = sf[1]; 467 break; 468 case 1*5+3: 469 code = 2; 470 if (sf[0] > sf[2]) 471 sf[0] = sf[2]; 472 sf[1] = sf[2] = sf[0]; 473 break; 474 default: 475 assert(0); //cannot happen 476 code = 0; /* kill warning */ 477 } 478 479#if 0 480 printf("%d: %2d %2d %2d %d %d -> %d\n", j, 481 sf[0], sf[1], sf[2], d1, d2, code); 482#endif 483 scale_code[j] = code; 484 sf += 3; 485 } 486} 487 488/* The most important function : psycho acoustic module. In this 489 encoder there is basically none, so this is the worst you can do, 490 but also this is the simpler. */ 491static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) 492{ 493 int i; 494 495 for(i=0;i<s->sblimit;i++) { 496 smr[i] = (int)(fixed_smr[i] * 10); 497 } 498} 499 500 501#define SB_NOTALLOCATED 0 502#define SB_ALLOCATED 1 503#define SB_NOMORE 2 504 505/* Try to maximize the smr while using a number of bits inferior to 506 the frame size. I tried to make the code simpler, faster and 507 smaller than other encoders :-) */ 508static void compute_bit_allocation(MpegAudioContext *s, 509 short smr1[MPA_MAX_CHANNELS][SBLIMIT], 510 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 511 int *padding) 512{ 513 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size; 514 int incr; 515 short smr[MPA_MAX_CHANNELS][SBLIMIT]; 516 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT]; 517 const unsigned char *alloc; 518 519 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT); 520 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT); 521 memset(bit_alloc, 0, s->nb_channels * SBLIMIT); 522 523 /* compute frame size and padding */ 524 max_frame_size = s->frame_size; 525 s->frame_frac += s->frame_frac_incr; 526 if (s->frame_frac >= 65536) { 527 s->frame_frac -= 65536; 528 s->do_padding = 1; 529 max_frame_size += 8; 530 } else { 531 s->do_padding = 0; 532 } 533 534 /* compute the header + bit alloc size */ 535 current_frame_size = 32; 536 alloc = s->alloc_table; 537 for(i=0;i<s->sblimit;i++) { 538 incr = alloc[0]; 539 current_frame_size += incr * s->nb_channels; 540 alloc += 1 << incr; 541 } 542 for(;;) { 543 /* look for the subband with the largest signal to mask ratio */ 544 max_sb = -1; 545 max_ch = -1; 546 max_smr = INT_MIN; 547 for(ch=0;ch<s->nb_channels;ch++) { 548 for(i=0;i<s->sblimit;i++) { 549 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) { 550 max_smr = smr[ch][i]; 551 max_sb = i; 552 max_ch = ch; 553 } 554 } 555 } 556#if 0 557 printf("current=%d max=%d max_sb=%d alloc=%d\n", 558 current_frame_size, max_frame_size, max_sb, 559 bit_alloc[max_sb]); 560#endif 561 if (max_sb < 0) 562 break; 563 564 /* find alloc table entry (XXX: not optimal, should use 565 pointer table) */ 566 alloc = s->alloc_table; 567 for(i=0;i<max_sb;i++) { 568 alloc += 1 << alloc[0]; 569 } 570 571 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) { 572 /* nothing was coded for this band: add the necessary bits */ 573 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6; 574 incr += total_quant_bits[alloc[1]]; 575 } else { 576 /* increments bit allocation */ 577 b = bit_alloc[max_ch][max_sb]; 578 incr = total_quant_bits[alloc[b + 1]] - 579 total_quant_bits[alloc[b]]; 580 } 581 582 if (current_frame_size + incr <= max_frame_size) { 583 /* can increase size */ 584 b = ++bit_alloc[max_ch][max_sb]; 585 current_frame_size += incr; 586 /* decrease smr by the resolution we added */ 587 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]]; 588 /* max allocation size reached ? */ 589 if (b == ((1 << alloc[0]) - 1)) 590 subband_status[max_ch][max_sb] = SB_NOMORE; 591 else 592 subband_status[max_ch][max_sb] = SB_ALLOCATED; 593 } else { 594 /* cannot increase the size of this subband */ 595 subband_status[max_ch][max_sb] = SB_NOMORE; 596 } 597 } 598 *padding = max_frame_size - current_frame_size; 599 assert(*padding >= 0); 600 601#if 0 602 for(i=0;i<s->sblimit;i++) { 603 printf("%d ", bit_alloc[i]); 604 } 605 printf("\n"); 606#endif 607} 608 609/* 610 * Output the mpeg audio layer 2 frame. Note how the code is small 611 * compared to other encoders :-) 612 */ 613static void encode_frame(MpegAudioContext *s, 614 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 615 int padding) 616{ 617 int i, j, k, l, bit_alloc_bits, b, ch; 618 unsigned char *sf; 619 int q[3]; 620 PutBitContext *p = &s->pb; 621 622 /* header */ 623 624 put_bits(p, 12, 0xfff); 625 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */ 626 put_bits(p, 2, 4-2); /* layer 2 */ 627 put_bits(p, 1, 1); /* no error protection */ 628 put_bits(p, 4, s->bitrate_index); 629 put_bits(p, 2, s->freq_index); 630 put_bits(p, 1, s->do_padding); /* use padding */ 631 put_bits(p, 1, 0); /* private_bit */ 632 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO); 633 put_bits(p, 2, 0); /* mode_ext */ 634 put_bits(p, 1, 0); /* no copyright */ 635 put_bits(p, 1, 1); /* original */ 636 put_bits(p, 2, 0); /* no emphasis */ 637 638 /* bit allocation */ 639 j = 0; 640 for(i=0;i<s->sblimit;i++) { 641 bit_alloc_bits = s->alloc_table[j]; 642 for(ch=0;ch<s->nb_channels;ch++) { 643 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]); 644 } 645 j += 1 << bit_alloc_bits; 646 } 647 648 /* scale codes */ 649 for(i=0;i<s->sblimit;i++) { 650 for(ch=0;ch<s->nb_channels;ch++) { 651 if (bit_alloc[ch][i]) 652 put_bits(p, 2, s->scale_code[ch][i]); 653 } 654 } 655 656 /* scale factors */ 657 for(i=0;i<s->sblimit;i++) { 658 for(ch=0;ch<s->nb_channels;ch++) { 659 if (bit_alloc[ch][i]) { 660 sf = &s->scale_factors[ch][i][0]; 661 switch(s->scale_code[ch][i]) { 662 case 0: 663 put_bits(p, 6, sf[0]); 664 put_bits(p, 6, sf[1]); 665 put_bits(p, 6, sf[2]); 666 break; 667 case 3: 668 case 1: 669 put_bits(p, 6, sf[0]); 670 put_bits(p, 6, sf[2]); 671 break; 672 case 2: 673 put_bits(p, 6, sf[0]); 674 break; 675 } 676 } 677 } 678 } 679 680 /* quantization & write sub band samples */ 681 682 for(k=0;k<3;k++) { 683 for(l=0;l<12;l+=3) { 684 j = 0; 685 for(i=0;i<s->sblimit;i++) { 686 bit_alloc_bits = s->alloc_table[j]; 687 for(ch=0;ch<s->nb_channels;ch++) { 688 b = bit_alloc[ch][i]; 689 if (b) { 690 int qindex, steps, m, sample, bits; 691 /* we encode 3 sub band samples of the same sub band at a time */ 692 qindex = s->alloc_table[j+b]; 693 steps = ff_mpa_quant_steps[qindex]; 694 for(m=0;m<3;m++) { 695 sample = s->sb_samples[ch][k][l + m][i]; 696 /* divide by scale factor */ 697#ifdef USE_FLOATS 698 { 699 float a; 700 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]]; 701 q[m] = (int)((a + 1.0) * steps * 0.5); 702 } 703#else 704 { 705 int q1, e, shift, mult; 706 e = s->scale_factors[ch][i][k]; 707 shift = scale_factor_shift[e]; 708 mult = scale_factor_mult[e]; 709 710 /* normalize to P bits */ 711 if (shift < 0) 712 q1 = sample << (-shift); 713 else 714 q1 = sample >> shift; 715 q1 = (q1 * mult) >> P; 716 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1); 717 } 718#endif 719 if (q[m] >= steps) 720 q[m] = steps - 1; 721 assert(q[m] >= 0 && q[m] < steps); 722 } 723 bits = ff_mpa_quant_bits[qindex]; 724 if (bits < 0) { 725 /* group the 3 values to save bits */ 726 put_bits(p, -bits, 727 q[0] + steps * (q[1] + steps * q[2])); 728#if 0 729 printf("%d: gr1 %d\n", 730 i, q[0] + steps * (q[1] + steps * q[2])); 731#endif 732 } else { 733#if 0 734 printf("%d: gr3 %d %d %d\n", 735 i, q[0], q[1], q[2]); 736#endif 737 put_bits(p, bits, q[0]); 738 put_bits(p, bits, q[1]); 739 put_bits(p, bits, q[2]); 740 } 741 } 742 } 743 /* next subband in alloc table */ 744 j += 1 << bit_alloc_bits; 745 } 746 } 747 } 748 749 /* padding */ 750 for(i=0;i<padding;i++) 751 put_bits(p, 1, 0); 752 753 /* flush */ 754 flush_put_bits(p); 755} 756 757static int MPA_encode_frame(AVCodecContext *avctx, 758 unsigned char *frame, int buf_size, void *data) 759{ 760 MpegAudioContext *s = avctx->priv_data; 761 short *samples = data; 762 short smr[MPA_MAX_CHANNELS][SBLIMIT]; 763 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT]; 764 int padding, i; 765 766 for(i=0;i<s->nb_channels;i++) { 767 filter(s, i, samples + i, s->nb_channels); 768 } 769 770 for(i=0;i<s->nb_channels;i++) { 771 compute_scale_factors(s->scale_code[i], s->scale_factors[i], 772 s->sb_samples[i], s->sblimit); 773 } 774 for(i=0;i<s->nb_channels;i++) { 775 psycho_acoustic_model(s, smr[i]); 776 } 777 compute_bit_allocation(s, smr, bit_alloc, &padding); 778 779 init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE); 780 781 encode_frame(s, bit_alloc, padding); 782 783 s->nb_samples += MPA_FRAME_SIZE; 784 return pbBufPtr(&s->pb) - s->pb.buf; 785} 786 787static av_cold int MPA_encode_close(AVCodecContext *avctx) 788{ 789 av_freep(&avctx->coded_frame); 790 return 0; 791} 792 793AVCodec mp2_encoder = { 794 "mp2", 795 CODEC_TYPE_AUDIO, 796 CODEC_ID_MP2, 797 sizeof(MpegAudioContext), 798 MPA_encode_init, 799 MPA_encode_frame, 800 MPA_encode_close, 801 NULL, 802 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, 803 .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), 804}; 805 806#undef FIX 807