1/* 2 * The simplest mpeg audio layer 2 encoder 3 * Copyright (c) 2000, 2001 Fabrice Bellard 4 * 5 * This file is part of Libav. 6 * 7 * Libav 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 * Libav 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 Libav; 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 "internal.h" 29#include "put_bits.h" 30 31#define FRAC_BITS 15 /* fractional bits for sb_samples and dct */ 32#define WFRAC_BITS 14 /* fractional bits for window */ 33 34#include "mpegaudio.h" 35 36/* currently, cannot change these constants (need to modify 37 quantization stage) */ 38#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS) 39 40#define SAMPLES_BUF_SIZE 4096 41 42typedef struct MpegAudioContext { 43 PutBitContext pb; 44 int nb_channels; 45 int lsf; /* 1 if mpeg2 low bitrate selected */ 46 int bitrate_index; /* bit rate */ 47 int freq_index; 48 int frame_size; /* frame size, in bits, without padding */ 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 avctx->frame_size = MPA_FRAME_SIZE; 83 84 /* encoding freq */ 85 s->lsf = 0; 86 for(i=0;i<3;i++) { 87 if (avpriv_mpa_freq_tab[i] == freq) 88 break; 89 if ((avpriv_mpa_freq_tab[i] / 2) == freq) { 90 s->lsf = 1; 91 break; 92 } 93 } 94 if (i == 3){ 95 av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq); 96 return -1; 97 } 98 s->freq_index = i; 99 100 /* encoding bitrate & frequency */ 101 for(i=0;i<15;i++) { 102 if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate) 103 break; 104 } 105 if (i == 15){ 106 av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate); 107 return -1; 108 } 109 s->bitrate_index = i; 110 111 /* compute total header size & pad bit */ 112 113 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0); 114 s->frame_size = ((int)a) * 8; 115 116 /* frame fractional size to compute padding */ 117 s->frame_frac = 0; 118 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0); 119 120 /* select the right allocation table */ 121 table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf); 122 123 /* number of used subbands */ 124 s->sblimit = ff_mpa_sblimit_table[table]; 125 s->alloc_table = ff_mpa_alloc_tables[table]; 126 127 av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", 128 bitrate, freq, s->frame_size, table, s->frame_frac_incr); 129 130 for(i=0;i<s->nb_channels;i++) 131 s->samples_offset[i] = 0; 132 133 for(i=0;i<257;i++) { 134 int v; 135 v = ff_mpa_enwindow[i]; 136#if WFRAC_BITS != 16 137 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS); 138#endif 139 filter_bank[i] = v; 140 if ((i & 63) != 0) 141 v = -v; 142 if (i != 0) 143 filter_bank[512 - i] = v; 144 } 145 146 for(i=0;i<64;i++) { 147 v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20)); 148 if (v <= 0) 149 v = 1; 150 scale_factor_table[i] = v; 151#ifdef USE_FLOATS 152 scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20); 153#else 154#define P 15 155 scale_factor_shift[i] = 21 - P - (i / 3); 156 scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0); 157#endif 158 } 159 for(i=0;i<128;i++) { 160 v = i - 64; 161 if (v <= -3) 162 v = 0; 163 else if (v < 0) 164 v = 1; 165 else if (v == 0) 166 v = 2; 167 else if (v < 3) 168 v = 3; 169 else 170 v = 4; 171 scale_diff_table[i] = v; 172 } 173 174 for(i=0;i<17;i++) { 175 v = ff_mpa_quant_bits[i]; 176 if (v < 0) 177 v = -v; 178 else 179 v = v * 3; 180 total_quant_bits[i] = 12 * v; 181 } 182 183 avctx->coded_frame= avcodec_alloc_frame(); 184 avctx->coded_frame->key_frame= 1; 185 186 return 0; 187} 188 189/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */ 190static void idct32(int *out, int *tab) 191{ 192 int i, j; 193 int *t, *t1, xr; 194 const int *xp = costab32; 195 196 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2]; 197 198 t = tab + 30; 199 t1 = tab + 2; 200 do { 201 t[0] += t[-4]; 202 t[1] += t[1 - 4]; 203 t -= 4; 204 } while (t != t1); 205 206 t = tab + 28; 207 t1 = tab + 4; 208 do { 209 t[0] += t[-8]; 210 t[1] += t[1-8]; 211 t[2] += t[2-8]; 212 t[3] += t[3-8]; 213 t -= 8; 214 } while (t != t1); 215 216 t = tab; 217 t1 = tab + 32; 218 do { 219 t[ 3] = -t[ 3]; 220 t[ 6] = -t[ 6]; 221 222 t[11] = -t[11]; 223 t[12] = -t[12]; 224 t[13] = -t[13]; 225 t[15] = -t[15]; 226 t += 16; 227 } while (t != t1); 228 229 230 t = tab; 231 t1 = tab + 8; 232 do { 233 int x1, x2, x3, x4; 234 235 x3 = MUL(t[16], FIX(SQRT2*0.5)); 236 x4 = t[0] - x3; 237 x3 = t[0] + x3; 238 239 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5)); 240 x1 = MUL((t[8] - x2), xp[0]); 241 x2 = MUL((t[8] + x2), xp[1]); 242 243 t[ 0] = x3 + x1; 244 t[ 8] = x4 - x2; 245 t[16] = x4 + x2; 246 t[24] = x3 - x1; 247 t++; 248 } while (t != t1); 249 250 xp += 2; 251 t = tab; 252 t1 = tab + 4; 253 do { 254 xr = MUL(t[28],xp[0]); 255 t[28] = (t[0] - xr); 256 t[0] = (t[0] + xr); 257 258 xr = MUL(t[4],xp[1]); 259 t[ 4] = (t[24] - xr); 260 t[24] = (t[24] + xr); 261 262 xr = MUL(t[20],xp[2]); 263 t[20] = (t[8] - xr); 264 t[ 8] = (t[8] + xr); 265 266 xr = MUL(t[12],xp[3]); 267 t[12] = (t[16] - xr); 268 t[16] = (t[16] + xr); 269 t++; 270 } while (t != t1); 271 xp += 4; 272 273 for (i = 0; i < 4; i++) { 274 xr = MUL(tab[30-i*4],xp[0]); 275 tab[30-i*4] = (tab[i*4] - xr); 276 tab[ i*4] = (tab[i*4] + xr); 277 278 xr = MUL(tab[ 2+i*4],xp[1]); 279 tab[ 2+i*4] = (tab[28-i*4] - xr); 280 tab[28-i*4] = (tab[28-i*4] + xr); 281 282 xr = MUL(tab[31-i*4],xp[0]); 283 tab[31-i*4] = (tab[1+i*4] - xr); 284 tab[ 1+i*4] = (tab[1+i*4] + xr); 285 286 xr = MUL(tab[ 3+i*4],xp[1]); 287 tab[ 3+i*4] = (tab[29-i*4] - xr); 288 tab[29-i*4] = (tab[29-i*4] + xr); 289 290 xp += 2; 291 } 292 293 t = tab + 30; 294 t1 = tab + 1; 295 do { 296 xr = MUL(t1[0], *xp); 297 t1[0] = (t[0] - xr); 298 t[0] = (t[0] + xr); 299 t -= 2; 300 t1 += 2; 301 xp++; 302 } while (t >= tab); 303 304 for(i=0;i<32;i++) { 305 out[i] = tab[bitinv32[i]]; 306 } 307} 308 309#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS) 310 311static void filter(MpegAudioContext *s, int ch, const short *samples, int incr) 312{ 313 short *p, *q; 314 int sum, offset, i, j; 315 int tmp[64]; 316 int tmp1[32]; 317 int *out; 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 364static void compute_scale_factors(unsigned char scale_code[SBLIMIT], 365 unsigned char scale_factors[SBLIMIT][3], 366 int sb_samples[3][12][SBLIMIT], 367 int sblimit) 368{ 369 int *p, vmax, v, n, i, j, k, code; 370 int index, d1, d2; 371 unsigned char *sf = &scale_factors[0][0]; 372 373 for(j=0;j<sblimit;j++) { 374 for(i=0;i<3;i++) { 375 /* find the max absolute value */ 376 p = &sb_samples[i][0][j]; 377 vmax = abs(*p); 378 for(k=1;k<12;k++) { 379 p += SBLIMIT; 380 v = abs(*p); 381 if (v > vmax) 382 vmax = v; 383 } 384 /* compute the scale factor index using log 2 computations */ 385 if (vmax > 1) { 386 n = av_log2(vmax); 387 /* n is the position of the MSB of vmax. now 388 use at most 2 compares to find the index */ 389 index = (21 - n) * 3 - 3; 390 if (index >= 0) { 391 while (vmax <= scale_factor_table[index+1]) 392 index++; 393 } else { 394 index = 0; /* very unlikely case of overflow */ 395 } 396 } else { 397 index = 62; /* value 63 is not allowed */ 398 } 399 400 av_dlog(NULL, "%2d:%d in=%x %x %d\n", 401 j, i, vmax, scale_factor_table[index], index); 402 /* store the scale factor */ 403 assert(index >=0 && index <= 63); 404 sf[i] = index; 405 } 406 407 /* compute the transmission factor : look if the scale factors 408 are close enough to each other */ 409 d1 = scale_diff_table[sf[0] - sf[1] + 64]; 410 d2 = scale_diff_table[sf[1] - sf[2] + 64]; 411 412 /* handle the 25 cases */ 413 switch(d1 * 5 + d2) { 414 case 0*5+0: 415 case 0*5+4: 416 case 3*5+4: 417 case 4*5+0: 418 case 4*5+4: 419 code = 0; 420 break; 421 case 0*5+1: 422 case 0*5+2: 423 case 4*5+1: 424 case 4*5+2: 425 code = 3; 426 sf[2] = sf[1]; 427 break; 428 case 0*5+3: 429 case 4*5+3: 430 code = 3; 431 sf[1] = sf[2]; 432 break; 433 case 1*5+0: 434 case 1*5+4: 435 case 2*5+4: 436 code = 1; 437 sf[1] = sf[0]; 438 break; 439 case 1*5+1: 440 case 1*5+2: 441 case 2*5+0: 442 case 2*5+1: 443 case 2*5+2: 444 code = 2; 445 sf[1] = sf[2] = sf[0]; 446 break; 447 case 2*5+3: 448 case 3*5+3: 449 code = 2; 450 sf[0] = sf[1] = sf[2]; 451 break; 452 case 3*5+0: 453 case 3*5+1: 454 case 3*5+2: 455 code = 2; 456 sf[0] = sf[2] = sf[1]; 457 break; 458 case 1*5+3: 459 code = 2; 460 if (sf[0] > sf[2]) 461 sf[0] = sf[2]; 462 sf[1] = sf[2] = sf[0]; 463 break; 464 default: 465 assert(0); //cannot happen 466 code = 0; /* kill warning */ 467 } 468 469 av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j, 470 sf[0], sf[1], sf[2], d1, d2, code); 471 scale_code[j] = code; 472 sf += 3; 473 } 474} 475 476/* The most important function : psycho acoustic module. In this 477 encoder there is basically none, so this is the worst you can do, 478 but also this is the simpler. */ 479static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) 480{ 481 int i; 482 483 for(i=0;i<s->sblimit;i++) { 484 smr[i] = (int)(fixed_smr[i] * 10); 485 } 486} 487 488 489#define SB_NOTALLOCATED 0 490#define SB_ALLOCATED 1 491#define SB_NOMORE 2 492 493/* Try to maximize the smr while using a number of bits inferior to 494 the frame size. I tried to make the code simpler, faster and 495 smaller than other encoders :-) */ 496static void compute_bit_allocation(MpegAudioContext *s, 497 short smr1[MPA_MAX_CHANNELS][SBLIMIT], 498 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 499 int *padding) 500{ 501 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size; 502 int incr; 503 short smr[MPA_MAX_CHANNELS][SBLIMIT]; 504 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT]; 505 const unsigned char *alloc; 506 507 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT); 508 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT); 509 memset(bit_alloc, 0, s->nb_channels * SBLIMIT); 510 511 /* compute frame size and padding */ 512 max_frame_size = s->frame_size; 513 s->frame_frac += s->frame_frac_incr; 514 if (s->frame_frac >= 65536) { 515 s->frame_frac -= 65536; 516 s->do_padding = 1; 517 max_frame_size += 8; 518 } else { 519 s->do_padding = 0; 520 } 521 522 /* compute the header + bit alloc size */ 523 current_frame_size = 32; 524 alloc = s->alloc_table; 525 for(i=0;i<s->sblimit;i++) { 526 incr = alloc[0]; 527 current_frame_size += incr * s->nb_channels; 528 alloc += 1 << incr; 529 } 530 for(;;) { 531 /* look for the subband with the largest signal to mask ratio */ 532 max_sb = -1; 533 max_ch = -1; 534 max_smr = INT_MIN; 535 for(ch=0;ch<s->nb_channels;ch++) { 536 for(i=0;i<s->sblimit;i++) { 537 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) { 538 max_smr = smr[ch][i]; 539 max_sb = i; 540 max_ch = ch; 541 } 542 } 543 } 544 if (max_sb < 0) 545 break; 546 av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n", 547 current_frame_size, max_frame_size, max_sb, max_ch, 548 bit_alloc[max_ch][max_sb]); 549 550 /* find alloc table entry (XXX: not optimal, should use 551 pointer table) */ 552 alloc = s->alloc_table; 553 for(i=0;i<max_sb;i++) { 554 alloc += 1 << alloc[0]; 555 } 556 557 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) { 558 /* nothing was coded for this band: add the necessary bits */ 559 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6; 560 incr += total_quant_bits[alloc[1]]; 561 } else { 562 /* increments bit allocation */ 563 b = bit_alloc[max_ch][max_sb]; 564 incr = total_quant_bits[alloc[b + 1]] - 565 total_quant_bits[alloc[b]]; 566 } 567 568 if (current_frame_size + incr <= max_frame_size) { 569 /* can increase size */ 570 b = ++bit_alloc[max_ch][max_sb]; 571 current_frame_size += incr; 572 /* decrease smr by the resolution we added */ 573 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]]; 574 /* max allocation size reached ? */ 575 if (b == ((1 << alloc[0]) - 1)) 576 subband_status[max_ch][max_sb] = SB_NOMORE; 577 else 578 subband_status[max_ch][max_sb] = SB_ALLOCATED; 579 } else { 580 /* cannot increase the size of this subband */ 581 subband_status[max_ch][max_sb] = SB_NOMORE; 582 } 583 } 584 *padding = max_frame_size - current_frame_size; 585 assert(*padding >= 0); 586} 587 588/* 589 * Output the mpeg audio layer 2 frame. Note how the code is small 590 * compared to other encoders :-) 591 */ 592static void encode_frame(MpegAudioContext *s, 593 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 594 int padding) 595{ 596 int i, j, k, l, bit_alloc_bits, b, ch; 597 unsigned char *sf; 598 int q[3]; 599 PutBitContext *p = &s->pb; 600 601 /* header */ 602 603 put_bits(p, 12, 0xfff); 604 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */ 605 put_bits(p, 2, 4-2); /* layer 2 */ 606 put_bits(p, 1, 1); /* no error protection */ 607 put_bits(p, 4, s->bitrate_index); 608 put_bits(p, 2, s->freq_index); 609 put_bits(p, 1, s->do_padding); /* use padding */ 610 put_bits(p, 1, 0); /* private_bit */ 611 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO); 612 put_bits(p, 2, 0); /* mode_ext */ 613 put_bits(p, 1, 0); /* no copyright */ 614 put_bits(p, 1, 1); /* original */ 615 put_bits(p, 2, 0); /* no emphasis */ 616 617 /* bit allocation */ 618 j = 0; 619 for(i=0;i<s->sblimit;i++) { 620 bit_alloc_bits = s->alloc_table[j]; 621 for(ch=0;ch<s->nb_channels;ch++) { 622 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]); 623 } 624 j += 1 << bit_alloc_bits; 625 } 626 627 /* scale codes */ 628 for(i=0;i<s->sblimit;i++) { 629 for(ch=0;ch<s->nb_channels;ch++) { 630 if (bit_alloc[ch][i]) 631 put_bits(p, 2, s->scale_code[ch][i]); 632 } 633 } 634 635 /* scale factors */ 636 for(i=0;i<s->sblimit;i++) { 637 for(ch=0;ch<s->nb_channels;ch++) { 638 if (bit_alloc[ch][i]) { 639 sf = &s->scale_factors[ch][i][0]; 640 switch(s->scale_code[ch][i]) { 641 case 0: 642 put_bits(p, 6, sf[0]); 643 put_bits(p, 6, sf[1]); 644 put_bits(p, 6, sf[2]); 645 break; 646 case 3: 647 case 1: 648 put_bits(p, 6, sf[0]); 649 put_bits(p, 6, sf[2]); 650 break; 651 case 2: 652 put_bits(p, 6, sf[0]); 653 break; 654 } 655 } 656 } 657 } 658 659 /* quantization & write sub band samples */ 660 661 for(k=0;k<3;k++) { 662 for(l=0;l<12;l+=3) { 663 j = 0; 664 for(i=0;i<s->sblimit;i++) { 665 bit_alloc_bits = s->alloc_table[j]; 666 for(ch=0;ch<s->nb_channels;ch++) { 667 b = bit_alloc[ch][i]; 668 if (b) { 669 int qindex, steps, m, sample, bits; 670 /* we encode 3 sub band samples of the same sub band at a time */ 671 qindex = s->alloc_table[j+b]; 672 steps = ff_mpa_quant_steps[qindex]; 673 for(m=0;m<3;m++) { 674 sample = s->sb_samples[ch][k][l + m][i]; 675 /* divide by scale factor */ 676#ifdef USE_FLOATS 677 { 678 float a; 679 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]]; 680 q[m] = (int)((a + 1.0) * steps * 0.5); 681 } 682#else 683 { 684 int q1, e, shift, mult; 685 e = s->scale_factors[ch][i][k]; 686 shift = scale_factor_shift[e]; 687 mult = scale_factor_mult[e]; 688 689 /* normalize to P bits */ 690 if (shift < 0) 691 q1 = sample << (-shift); 692 else 693 q1 = sample >> shift; 694 q1 = (q1 * mult) >> P; 695 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1); 696 } 697#endif 698 if (q[m] >= steps) 699 q[m] = steps - 1; 700 assert(q[m] >= 0 && q[m] < steps); 701 } 702 bits = ff_mpa_quant_bits[qindex]; 703 if (bits < 0) { 704 /* group the 3 values to save bits */ 705 put_bits(p, -bits, 706 q[0] + steps * (q[1] + steps * q[2])); 707 } else { 708 put_bits(p, bits, q[0]); 709 put_bits(p, bits, q[1]); 710 put_bits(p, bits, q[2]); 711 } 712 } 713 } 714 /* next subband in alloc table */ 715 j += 1 << bit_alloc_bits; 716 } 717 } 718 } 719 720 /* padding */ 721 for(i=0;i<padding;i++) 722 put_bits(p, 1, 0); 723 724 /* flush */ 725 flush_put_bits(p); 726} 727 728static int MPA_encode_frame(AVCodecContext *avctx, 729 unsigned char *frame, int buf_size, void *data) 730{ 731 MpegAudioContext *s = avctx->priv_data; 732 const short *samples = data; 733 short smr[MPA_MAX_CHANNELS][SBLIMIT]; 734 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT]; 735 int padding, i; 736 737 for(i=0;i<s->nb_channels;i++) { 738 filter(s, i, samples + i, s->nb_channels); 739 } 740 741 for(i=0;i<s->nb_channels;i++) { 742 compute_scale_factors(s->scale_code[i], s->scale_factors[i], 743 s->sb_samples[i], s->sblimit); 744 } 745 for(i=0;i<s->nb_channels;i++) { 746 psycho_acoustic_model(s, smr[i]); 747 } 748 compute_bit_allocation(s, smr, bit_alloc, &padding); 749 750 init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE); 751 752 encode_frame(s, bit_alloc, padding); 753 754 return put_bits_ptr(&s->pb) - s->pb.buf; 755} 756 757static av_cold int MPA_encode_close(AVCodecContext *avctx) 758{ 759 av_freep(&avctx->coded_frame); 760 return 0; 761} 762 763static const AVCodecDefault mp2_defaults[] = { 764 { "b", "128k" }, 765 { NULL }, 766}; 767 768AVCodec ff_mp2_encoder = { 769 .name = "mp2", 770 .type = AVMEDIA_TYPE_AUDIO, 771 .id = CODEC_ID_MP2, 772 .priv_data_size = sizeof(MpegAudioContext), 773 .init = MPA_encode_init, 774 .encode = MPA_encode_frame, 775 .close = MPA_encode_close, 776 .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, 777 .supported_samplerates= (const int[]){44100, 48000, 32000, 22050, 24000, 16000, 0}, 778 .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), 779 .defaults = mp2_defaults, 780}; 781