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 24 * The simplest mpeg audio layer 2 encoder. 25 */ 26 27#include "avcodec.h" 28#include "put_bits.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 dprintf(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", 130 bitrate, freq, s->frame_size, table, s->frame_frac_incr); 131 132 for(i=0;i<s->nb_channels;i++) 133 s->samples_offset[i] = 0; 134 135 for(i=0;i<257;i++) { 136 int v; 137 v = ff_mpa_enwindow[i]; 138#if WFRAC_BITS != 16 139 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS); 140#endif 141 filter_bank[i] = v; 142 if ((i & 63) != 0) 143 v = -v; 144 if (i != 0) 145 filter_bank[512 - i] = v; 146 } 147 148 for(i=0;i<64;i++) { 149 v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20)); 150 if (v <= 0) 151 v = 1; 152 scale_factor_table[i] = v; 153#ifdef USE_FLOATS 154 scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20); 155#else 156#define P 15 157 scale_factor_shift[i] = 21 - P - (i / 3); 158 scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0); 159#endif 160 } 161 for(i=0;i<128;i++) { 162 v = i - 64; 163 if (v <= -3) 164 v = 0; 165 else if (v < 0) 166 v = 1; 167 else if (v == 0) 168 v = 2; 169 else if (v < 3) 170 v = 3; 171 else 172 v = 4; 173 scale_diff_table[i] = v; 174 } 175 176 for(i=0;i<17;i++) { 177 v = ff_mpa_quant_bits[i]; 178 if (v < 0) 179 v = -v; 180 else 181 v = v * 3; 182 total_quant_bits[i] = 12 * v; 183 } 184 185 avctx->coded_frame= avcodec_alloc_frame(); 186 avctx->coded_frame->key_frame= 1; 187 188 return 0; 189} 190 191/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */ 192static void idct32(int *out, int *tab) 193{ 194 int i, j; 195 int *t, *t1, xr; 196 const int *xp = costab32; 197 198 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2]; 199 200 t = tab + 30; 201 t1 = tab + 2; 202 do { 203 t[0] += t[-4]; 204 t[1] += t[1 - 4]; 205 t -= 4; 206 } while (t != t1); 207 208 t = tab + 28; 209 t1 = tab + 4; 210 do { 211 t[0] += t[-8]; 212 t[1] += t[1-8]; 213 t[2] += t[2-8]; 214 t[3] += t[3-8]; 215 t -= 8; 216 } while (t != t1); 217 218 t = tab; 219 t1 = tab + 32; 220 do { 221 t[ 3] = -t[ 3]; 222 t[ 6] = -t[ 6]; 223 224 t[11] = -t[11]; 225 t[12] = -t[12]; 226 t[13] = -t[13]; 227 t[15] = -t[15]; 228 t += 16; 229 } while (t != t1); 230 231 232 t = tab; 233 t1 = tab + 8; 234 do { 235 int x1, x2, x3, x4; 236 237 x3 = MUL(t[16], FIX(SQRT2*0.5)); 238 x4 = t[0] - x3; 239 x3 = t[0] + x3; 240 241 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5)); 242 x1 = MUL((t[8] - x2), xp[0]); 243 x2 = MUL((t[8] + x2), xp[1]); 244 245 t[ 0] = x3 + x1; 246 t[ 8] = x4 - x2; 247 t[16] = x4 + x2; 248 t[24] = x3 - x1; 249 t++; 250 } while (t != t1); 251 252 xp += 2; 253 t = tab; 254 t1 = tab + 4; 255 do { 256 xr = MUL(t[28],xp[0]); 257 t[28] = (t[0] - xr); 258 t[0] = (t[0] + xr); 259 260 xr = MUL(t[4],xp[1]); 261 t[ 4] = (t[24] - xr); 262 t[24] = (t[24] + xr); 263 264 xr = MUL(t[20],xp[2]); 265 t[20] = (t[8] - xr); 266 t[ 8] = (t[8] + xr); 267 268 xr = MUL(t[12],xp[3]); 269 t[12] = (t[16] - xr); 270 t[16] = (t[16] + xr); 271 t++; 272 } while (t != t1); 273 xp += 4; 274 275 for (i = 0; i < 4; i++) { 276 xr = MUL(tab[30-i*4],xp[0]); 277 tab[30-i*4] = (tab[i*4] - xr); 278 tab[ i*4] = (tab[i*4] + xr); 279 280 xr = MUL(tab[ 2+i*4],xp[1]); 281 tab[ 2+i*4] = (tab[28-i*4] - xr); 282 tab[28-i*4] = (tab[28-i*4] + xr); 283 284 xr = MUL(tab[31-i*4],xp[0]); 285 tab[31-i*4] = (tab[1+i*4] - xr); 286 tab[ 1+i*4] = (tab[1+i*4] + xr); 287 288 xr = MUL(tab[ 3+i*4],xp[1]); 289 tab[ 3+i*4] = (tab[29-i*4] - xr); 290 tab[29-i*4] = (tab[29-i*4] + xr); 291 292 xp += 2; 293 } 294 295 t = tab + 30; 296 t1 = tab + 1; 297 do { 298 xr = MUL(t1[0], *xp); 299 t1[0] = (t[0] - xr); 300 t[0] = (t[0] + xr); 301 t -= 2; 302 t1 += 2; 303 xp++; 304 } while (t >= tab); 305 306 for(i=0;i<32;i++) { 307 out[i] = tab[bitinv32[i]]; 308 } 309} 310 311#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS) 312 313static void filter(MpegAudioContext *s, int ch, short *samples, int incr) 314{ 315 short *p, *q; 316 int sum, offset, i, j; 317 int tmp[64]; 318 int tmp1[32]; 319 int *out; 320 321 // print_pow1(samples, 1152); 322 323 offset = s->samples_offset[ch]; 324 out = &s->sb_samples[ch][0][0][0]; 325 for(j=0;j<36;j++) { 326 /* 32 samples at once */ 327 for(i=0;i<32;i++) { 328 s->samples_buf[ch][offset + (31 - i)] = samples[0]; 329 samples += incr; 330 } 331 332 /* filter */ 333 p = s->samples_buf[ch] + offset; 334 q = filter_bank; 335 /* maxsum = 23169 */ 336 for(i=0;i<64;i++) { 337 sum = p[0*64] * q[0*64]; 338 sum += p[1*64] * q[1*64]; 339 sum += p[2*64] * q[2*64]; 340 sum += p[3*64] * q[3*64]; 341 sum += p[4*64] * q[4*64]; 342 sum += p[5*64] * q[5*64]; 343 sum += p[6*64] * q[6*64]; 344 sum += p[7*64] * q[7*64]; 345 tmp[i] = sum; 346 p++; 347 q++; 348 } 349 tmp1[0] = tmp[16] >> WSHIFT; 350 for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT; 351 for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT; 352 353 idct32(out, tmp1); 354 355 /* advance of 32 samples */ 356 offset -= 32; 357 out += 32; 358 /* handle the wrap around */ 359 if (offset < 0) { 360 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32), 361 s->samples_buf[ch], (512 - 32) * 2); 362 offset = SAMPLES_BUF_SIZE - 512; 363 } 364 } 365 s->samples_offset[ch] = offset; 366 367 // print_pow(s->sb_samples, 1152); 368} 369 370static void compute_scale_factors(unsigned char scale_code[SBLIMIT], 371 unsigned char scale_factors[SBLIMIT][3], 372 int sb_samples[3][12][SBLIMIT], 373 int sblimit) 374{ 375 int *p, vmax, v, n, i, j, k, code; 376 int index, d1, d2; 377 unsigned char *sf = &scale_factors[0][0]; 378 379 for(j=0;j<sblimit;j++) { 380 for(i=0;i<3;i++) { 381 /* find the max absolute value */ 382 p = &sb_samples[i][0][j]; 383 vmax = abs(*p); 384 for(k=1;k<12;k++) { 385 p += SBLIMIT; 386 v = abs(*p); 387 if (v > vmax) 388 vmax = v; 389 } 390 /* compute the scale factor index using log 2 computations */ 391 if (vmax > 1) { 392 n = av_log2(vmax); 393 /* n is the position of the MSB of vmax. now 394 use at most 2 compares to find the index */ 395 index = (21 - n) * 3 - 3; 396 if (index >= 0) { 397 while (vmax <= scale_factor_table[index+1]) 398 index++; 399 } else { 400 index = 0; /* very unlikely case of overflow */ 401 } 402 } else { 403 index = 62; /* value 63 is not allowed */ 404 } 405 406#if 0 407 printf("%2d:%d in=%x %x %d\n", 408 j, i, vmax, scale_factor_table[index], index); 409#endif 410 /* store the scale factor */ 411 assert(index >=0 && index <= 63); 412 sf[i] = index; 413 } 414 415 /* compute the transmission factor : look if the scale factors 416 are close enough to each other */ 417 d1 = scale_diff_table[sf[0] - sf[1] + 64]; 418 d2 = scale_diff_table[sf[1] - sf[2] + 64]; 419 420 /* handle the 25 cases */ 421 switch(d1 * 5 + d2) { 422 case 0*5+0: 423 case 0*5+4: 424 case 3*5+4: 425 case 4*5+0: 426 case 4*5+4: 427 code = 0; 428 break; 429 case 0*5+1: 430 case 0*5+2: 431 case 4*5+1: 432 case 4*5+2: 433 code = 3; 434 sf[2] = sf[1]; 435 break; 436 case 0*5+3: 437 case 4*5+3: 438 code = 3; 439 sf[1] = sf[2]; 440 break; 441 case 1*5+0: 442 case 1*5+4: 443 case 2*5+4: 444 code = 1; 445 sf[1] = sf[0]; 446 break; 447 case 1*5+1: 448 case 1*5+2: 449 case 2*5+0: 450 case 2*5+1: 451 case 2*5+2: 452 code = 2; 453 sf[1] = sf[2] = sf[0]; 454 break; 455 case 2*5+3: 456 case 3*5+3: 457 code = 2; 458 sf[0] = sf[1] = sf[2]; 459 break; 460 case 3*5+0: 461 case 3*5+1: 462 case 3*5+2: 463 code = 2; 464 sf[0] = sf[2] = sf[1]; 465 break; 466 case 1*5+3: 467 code = 2; 468 if (sf[0] > sf[2]) 469 sf[0] = sf[2]; 470 sf[1] = sf[2] = sf[0]; 471 break; 472 default: 473 assert(0); //cannot happen 474 code = 0; /* kill warning */ 475 } 476 477#if 0 478 printf("%d: %2d %2d %2d %d %d -> %d\n", j, 479 sf[0], sf[1], sf[2], d1, d2, code); 480#endif 481 scale_code[j] = code; 482 sf += 3; 483 } 484} 485 486/* The most important function : psycho acoustic module. In this 487 encoder there is basically none, so this is the worst you can do, 488 but also this is the simpler. */ 489static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) 490{ 491 int i; 492 493 for(i=0;i<s->sblimit;i++) { 494 smr[i] = (int)(fixed_smr[i] * 10); 495 } 496} 497 498 499#define SB_NOTALLOCATED 0 500#define SB_ALLOCATED 1 501#define SB_NOMORE 2 502 503/* Try to maximize the smr while using a number of bits inferior to 504 the frame size. I tried to make the code simpler, faster and 505 smaller than other encoders :-) */ 506static void compute_bit_allocation(MpegAudioContext *s, 507 short smr1[MPA_MAX_CHANNELS][SBLIMIT], 508 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 509 int *padding) 510{ 511 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size; 512 int incr; 513 short smr[MPA_MAX_CHANNELS][SBLIMIT]; 514 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT]; 515 const unsigned char *alloc; 516 517 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT); 518 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT); 519 memset(bit_alloc, 0, s->nb_channels * SBLIMIT); 520 521 /* compute frame size and padding */ 522 max_frame_size = s->frame_size; 523 s->frame_frac += s->frame_frac_incr; 524 if (s->frame_frac >= 65536) { 525 s->frame_frac -= 65536; 526 s->do_padding = 1; 527 max_frame_size += 8; 528 } else { 529 s->do_padding = 0; 530 } 531 532 /* compute the header + bit alloc size */ 533 current_frame_size = 32; 534 alloc = s->alloc_table; 535 for(i=0;i<s->sblimit;i++) { 536 incr = alloc[0]; 537 current_frame_size += incr * s->nb_channels; 538 alloc += 1 << incr; 539 } 540 for(;;) { 541 /* look for the subband with the largest signal to mask ratio */ 542 max_sb = -1; 543 max_ch = -1; 544 max_smr = INT_MIN; 545 for(ch=0;ch<s->nb_channels;ch++) { 546 for(i=0;i<s->sblimit;i++) { 547 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) { 548 max_smr = smr[ch][i]; 549 max_sb = i; 550 max_ch = ch; 551 } 552 } 553 } 554#if 0 555 printf("current=%d max=%d max_sb=%d alloc=%d\n", 556 current_frame_size, max_frame_size, max_sb, 557 bit_alloc[max_sb]); 558#endif 559 if (max_sb < 0) 560 break; 561 562 /* find alloc table entry (XXX: not optimal, should use 563 pointer table) */ 564 alloc = s->alloc_table; 565 for(i=0;i<max_sb;i++) { 566 alloc += 1 << alloc[0]; 567 } 568 569 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) { 570 /* nothing was coded for this band: add the necessary bits */ 571 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6; 572 incr += total_quant_bits[alloc[1]]; 573 } else { 574 /* increments bit allocation */ 575 b = bit_alloc[max_ch][max_sb]; 576 incr = total_quant_bits[alloc[b + 1]] - 577 total_quant_bits[alloc[b]]; 578 } 579 580 if (current_frame_size + incr <= max_frame_size) { 581 /* can increase size */ 582 b = ++bit_alloc[max_ch][max_sb]; 583 current_frame_size += incr; 584 /* decrease smr by the resolution we added */ 585 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]]; 586 /* max allocation size reached ? */ 587 if (b == ((1 << alloc[0]) - 1)) 588 subband_status[max_ch][max_sb] = SB_NOMORE; 589 else 590 subband_status[max_ch][max_sb] = SB_ALLOCATED; 591 } else { 592 /* cannot increase the size of this subband */ 593 subband_status[max_ch][max_sb] = SB_NOMORE; 594 } 595 } 596 *padding = max_frame_size - current_frame_size; 597 assert(*padding >= 0); 598 599#if 0 600 for(i=0;i<s->sblimit;i++) { 601 printf("%d ", bit_alloc[i]); 602 } 603 printf("\n"); 604#endif 605} 606 607/* 608 * Output the mpeg audio layer 2 frame. Note how the code is small 609 * compared to other encoders :-) 610 */ 611static void encode_frame(MpegAudioContext *s, 612 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 613 int padding) 614{ 615 int i, j, k, l, bit_alloc_bits, b, ch; 616 unsigned char *sf; 617 int q[3]; 618 PutBitContext *p = &s->pb; 619 620 /* header */ 621 622 put_bits(p, 12, 0xfff); 623 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */ 624 put_bits(p, 2, 4-2); /* layer 2 */ 625 put_bits(p, 1, 1); /* no error protection */ 626 put_bits(p, 4, s->bitrate_index); 627 put_bits(p, 2, s->freq_index); 628 put_bits(p, 1, s->do_padding); /* use padding */ 629 put_bits(p, 1, 0); /* private_bit */ 630 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO); 631 put_bits(p, 2, 0); /* mode_ext */ 632 put_bits(p, 1, 0); /* no copyright */ 633 put_bits(p, 1, 1); /* original */ 634 put_bits(p, 2, 0); /* no emphasis */ 635 636 /* bit allocation */ 637 j = 0; 638 for(i=0;i<s->sblimit;i++) { 639 bit_alloc_bits = s->alloc_table[j]; 640 for(ch=0;ch<s->nb_channels;ch++) { 641 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]); 642 } 643 j += 1 << bit_alloc_bits; 644 } 645 646 /* scale codes */ 647 for(i=0;i<s->sblimit;i++) { 648 for(ch=0;ch<s->nb_channels;ch++) { 649 if (bit_alloc[ch][i]) 650 put_bits(p, 2, s->scale_code[ch][i]); 651 } 652 } 653 654 /* scale factors */ 655 for(i=0;i<s->sblimit;i++) { 656 for(ch=0;ch<s->nb_channels;ch++) { 657 if (bit_alloc[ch][i]) { 658 sf = &s->scale_factors[ch][i][0]; 659 switch(s->scale_code[ch][i]) { 660 case 0: 661 put_bits(p, 6, sf[0]); 662 put_bits(p, 6, sf[1]); 663 put_bits(p, 6, sf[2]); 664 break; 665 case 3: 666 case 1: 667 put_bits(p, 6, sf[0]); 668 put_bits(p, 6, sf[2]); 669 break; 670 case 2: 671 put_bits(p, 6, sf[0]); 672 break; 673 } 674 } 675 } 676 } 677 678 /* quantization & write sub band samples */ 679 680 for(k=0;k<3;k++) { 681 for(l=0;l<12;l+=3) { 682 j = 0; 683 for(i=0;i<s->sblimit;i++) { 684 bit_alloc_bits = s->alloc_table[j]; 685 for(ch=0;ch<s->nb_channels;ch++) { 686 b = bit_alloc[ch][i]; 687 if (b) { 688 int qindex, steps, m, sample, bits; 689 /* we encode 3 sub band samples of the same sub band at a time */ 690 qindex = s->alloc_table[j+b]; 691 steps = ff_mpa_quant_steps[qindex]; 692 for(m=0;m<3;m++) { 693 sample = s->sb_samples[ch][k][l + m][i]; 694 /* divide by scale factor */ 695#ifdef USE_FLOATS 696 { 697 float a; 698 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]]; 699 q[m] = (int)((a + 1.0) * steps * 0.5); 700 } 701#else 702 { 703 int q1, e, shift, mult; 704 e = s->scale_factors[ch][i][k]; 705 shift = scale_factor_shift[e]; 706 mult = scale_factor_mult[e]; 707 708 /* normalize to P bits */ 709 if (shift < 0) 710 q1 = sample << (-shift); 711 else 712 q1 = sample >> shift; 713 q1 = (q1 * mult) >> P; 714 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1); 715 } 716#endif 717 if (q[m] >= steps) 718 q[m] = steps - 1; 719 assert(q[m] >= 0 && q[m] < steps); 720 } 721 bits = ff_mpa_quant_bits[qindex]; 722 if (bits < 0) { 723 /* group the 3 values to save bits */ 724 put_bits(p, -bits, 725 q[0] + steps * (q[1] + steps * q[2])); 726#if 0 727 printf("%d: gr1 %d\n", 728 i, q[0] + steps * (q[1] + steps * q[2])); 729#endif 730 } else { 731#if 0 732 printf("%d: gr3 %d %d %d\n", 733 i, q[0], q[1], q[2]); 734#endif 735 put_bits(p, bits, q[0]); 736 put_bits(p, bits, q[1]); 737 put_bits(p, bits, q[2]); 738 } 739 } 740 } 741 /* next subband in alloc table */ 742 j += 1 << bit_alloc_bits; 743 } 744 } 745 } 746 747 /* padding */ 748 for(i=0;i<padding;i++) 749 put_bits(p, 1, 0); 750 751 /* flush */ 752 flush_put_bits(p); 753} 754 755static int MPA_encode_frame(AVCodecContext *avctx, 756 unsigned char *frame, int buf_size, void *data) 757{ 758 MpegAudioContext *s = avctx->priv_data; 759 short *samples = data; 760 short smr[MPA_MAX_CHANNELS][SBLIMIT]; 761 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT]; 762 int padding, i; 763 764 for(i=0;i<s->nb_channels;i++) { 765 filter(s, i, samples + i, s->nb_channels); 766 } 767 768 for(i=0;i<s->nb_channels;i++) { 769 compute_scale_factors(s->scale_code[i], s->scale_factors[i], 770 s->sb_samples[i], s->sblimit); 771 } 772 for(i=0;i<s->nb_channels;i++) { 773 psycho_acoustic_model(s, smr[i]); 774 } 775 compute_bit_allocation(s, smr, bit_alloc, &padding); 776 777 init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE); 778 779 encode_frame(s, bit_alloc, padding); 780 781 s->nb_samples += MPA_FRAME_SIZE; 782 return put_bits_ptr(&s->pb) - s->pb.buf; 783} 784 785static av_cold int MPA_encode_close(AVCodecContext *avctx) 786{ 787 av_freep(&avctx->coded_frame); 788 return 0; 789} 790 791AVCodec mp2_encoder = { 792 "mp2", 793 AVMEDIA_TYPE_AUDIO, 794 CODEC_ID_MP2, 795 sizeof(MpegAudioContext), 796 MPA_encode_init, 797 MPA_encode_frame, 798 MPA_encode_close, 799 NULL, 800 .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, 801 .supported_samplerates= (const int[]){44100, 48000, 32000, 22050, 24000, 16000, 0}, 802 .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), 803}; 804 805#undef FIX 806