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 "libavutil/channel_layout.h" 28 29#include "avcodec.h" 30#include "internal.h" 31#include "put_bits.h" 32 33#define FRAC_BITS 15 /* fractional bits for sb_samples and dct */ 34#define WFRAC_BITS 14 /* fractional bits for window */ 35 36#include "mpegaudio.h" 37#include "mpegaudiodsp.h" 38#include "mpegaudiodata.h" 39#include "mpegaudiotab.h" 40 41/* currently, cannot change these constants (need to modify 42 quantization stage) */ 43#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS) 44 45#define SAMPLES_BUF_SIZE 4096 46 47typedef struct MpegAudioContext { 48 PutBitContext pb; 49 int nb_channels; 50 int lsf; /* 1 if mpeg2 low bitrate selected */ 51 int bitrate_index; /* bit rate */ 52 int freq_index; 53 int frame_size; /* frame size, in bits, without padding */ 54 /* padding computation */ 55 int frame_frac, frame_frac_incr, do_padding; 56 short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */ 57 int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */ 58 int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT]; 59 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */ 60 /* code to group 3 scale factors */ 61 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; 62 int sblimit; /* number of used subbands */ 63 const unsigned char *alloc_table; 64 int16_t filter_bank[512]; 65 int scale_factor_table[64]; 66 unsigned char scale_diff_table[128]; 67#if USE_FLOATS 68 float scale_factor_inv_table[64]; 69#else 70 int8_t scale_factor_shift[64]; 71 unsigned short scale_factor_mult[64]; 72#endif 73 unsigned short total_quant_bits[17]; /* total number of bits per allocation group */ 74} MpegAudioContext; 75 76static av_cold int MPA_encode_init(AVCodecContext *avctx) 77{ 78 MpegAudioContext *s = avctx->priv_data; 79 int freq = avctx->sample_rate; 80 int bitrate = avctx->bit_rate; 81 int channels = avctx->channels; 82 int i, v, table; 83 float a; 84 85 if (channels <= 0 || channels > 2){ 86 av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels); 87 return AVERROR(EINVAL); 88 } 89 bitrate = bitrate / 1000; 90 s->nb_channels = channels; 91 avctx->frame_size = MPA_FRAME_SIZE; 92 avctx->delay = 512 - 32 + 1; 93 94 /* encoding freq */ 95 s->lsf = 0; 96 for(i=0;i<3;i++) { 97 if (avpriv_mpa_freq_tab[i] == freq) 98 break; 99 if ((avpriv_mpa_freq_tab[i] / 2) == freq) { 100 s->lsf = 1; 101 break; 102 } 103 } 104 if (i == 3){ 105 av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq); 106 return AVERROR(EINVAL); 107 } 108 s->freq_index = i; 109 110 /* encoding bitrate & frequency */ 111 for(i=1;i<15;i++) { 112 if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate) 113 break; 114 } 115 if (i == 15 && !avctx->bit_rate) { 116 i = 14; 117 bitrate = avpriv_mpa_bitrate_tab[s->lsf][1][i]; 118 avctx->bit_rate = bitrate * 1000; 119 } 120 if (i == 15){ 121 av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate); 122 return AVERROR(EINVAL); 123 } 124 s->bitrate_index = i; 125 126 /* compute total header size & pad bit */ 127 128 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0); 129 s->frame_size = ((int)a) * 8; 130 131 /* frame fractional size to compute padding */ 132 s->frame_frac = 0; 133 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0); 134 135 /* select the right allocation table */ 136 table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf); 137 138 /* number of used subbands */ 139 s->sblimit = ff_mpa_sblimit_table[table]; 140 s->alloc_table = ff_mpa_alloc_tables[table]; 141 142 av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", 143 bitrate, freq, s->frame_size, table, s->frame_frac_incr); 144 145 for(i=0;i<s->nb_channels;i++) 146 s->samples_offset[i] = 0; 147 148 for(i=0;i<257;i++) { 149 int v; 150 v = ff_mpa_enwindow[i]; 151#if WFRAC_BITS != 16 152 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS); 153#endif 154 s->filter_bank[i] = v; 155 if ((i & 63) != 0) 156 v = -v; 157 if (i != 0) 158 s->filter_bank[512 - i] = v; 159 } 160 161 for(i=0;i<64;i++) { 162 v = (int)(exp2((3 - i) / 3.0) * (1 << 20)); 163 if (v <= 0) 164 v = 1; 165 s->scale_factor_table[i] = v; 166#if USE_FLOATS 167 s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20); 168#else 169#define P 15 170 s->scale_factor_shift[i] = 21 - P - (i / 3); 171 s->scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0); 172#endif 173 } 174 for(i=0;i<128;i++) { 175 v = i - 64; 176 if (v <= -3) 177 v = 0; 178 else if (v < 0) 179 v = 1; 180 else if (v == 0) 181 v = 2; 182 else if (v < 3) 183 v = 3; 184 else 185 v = 4; 186 s->scale_diff_table[i] = v; 187 } 188 189 for(i=0;i<17;i++) { 190 v = ff_mpa_quant_bits[i]; 191 if (v < 0) 192 v = -v; 193 else 194 v = v * 3; 195 s->total_quant_bits[i] = 12 * v; 196 } 197 198 return 0; 199} 200 201/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */ 202static void idct32(int *out, int *tab) 203{ 204 int i, j; 205 int *t, *t1, xr; 206 const int *xp = costab32; 207 208 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2]; 209 210 t = tab + 30; 211 t1 = tab + 2; 212 do { 213 t[0] += t[-4]; 214 t[1] += t[1 - 4]; 215 t -= 4; 216 } while (t != t1); 217 218 t = tab + 28; 219 t1 = tab + 4; 220 do { 221 t[0] += t[-8]; 222 t[1] += t[1-8]; 223 t[2] += t[2-8]; 224 t[3] += t[3-8]; 225 t -= 8; 226 } while (t != t1); 227 228 t = tab; 229 t1 = tab + 32; 230 do { 231 t[ 3] = -t[ 3]; 232 t[ 6] = -t[ 6]; 233 234 t[11] = -t[11]; 235 t[12] = -t[12]; 236 t[13] = -t[13]; 237 t[15] = -t[15]; 238 t += 16; 239 } while (t != t1); 240 241 242 t = tab; 243 t1 = tab + 8; 244 do { 245 int x1, x2, x3, x4; 246 247 x3 = MUL(t[16], FIX(SQRT2*0.5)); 248 x4 = t[0] - x3; 249 x3 = t[0] + x3; 250 251 x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5)); 252 x1 = MUL((t[8] - x2), xp[0]); 253 x2 = MUL((t[8] + x2), xp[1]); 254 255 t[ 0] = x3 + x1; 256 t[ 8] = x4 - x2; 257 t[16] = x4 + x2; 258 t[24] = x3 - x1; 259 t++; 260 } while (t != t1); 261 262 xp += 2; 263 t = tab; 264 t1 = tab + 4; 265 do { 266 xr = MUL(t[28],xp[0]); 267 t[28] = (t[0] - xr); 268 t[0] = (t[0] + xr); 269 270 xr = MUL(t[4],xp[1]); 271 t[ 4] = (t[24] - xr); 272 t[24] = (t[24] + xr); 273 274 xr = MUL(t[20],xp[2]); 275 t[20] = (t[8] - xr); 276 t[ 8] = (t[8] + xr); 277 278 xr = MUL(t[12],xp[3]); 279 t[12] = (t[16] - xr); 280 t[16] = (t[16] + xr); 281 t++; 282 } while (t != t1); 283 xp += 4; 284 285 for (i = 0; i < 4; i++) { 286 xr = MUL(tab[30-i*4],xp[0]); 287 tab[30-i*4] = (tab[i*4] - xr); 288 tab[ i*4] = (tab[i*4] + xr); 289 290 xr = MUL(tab[ 2+i*4],xp[1]); 291 tab[ 2+i*4] = (tab[28-i*4] - xr); 292 tab[28-i*4] = (tab[28-i*4] + xr); 293 294 xr = MUL(tab[31-i*4],xp[0]); 295 tab[31-i*4] = (tab[1+i*4] - xr); 296 tab[ 1+i*4] = (tab[1+i*4] + xr); 297 298 xr = MUL(tab[ 3+i*4],xp[1]); 299 tab[ 3+i*4] = (tab[29-i*4] - xr); 300 tab[29-i*4] = (tab[29-i*4] + xr); 301 302 xp += 2; 303 } 304 305 t = tab + 30; 306 t1 = tab + 1; 307 do { 308 xr = MUL(t1[0], *xp); 309 t1[0] = (t[0] - xr); 310 t[0] = (t[0] + xr); 311 t -= 2; 312 t1 += 2; 313 xp++; 314 } while (t >= tab); 315 316 for(i=0;i<32;i++) { 317 out[i] = tab[bitinv32[i]]; 318 } 319} 320 321#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS) 322 323static void filter(MpegAudioContext *s, int ch, const short *samples, int incr) 324{ 325 short *p, *q; 326 int sum, offset, i, j; 327 int tmp[64]; 328 int tmp1[32]; 329 int *out; 330 331 offset = s->samples_offset[ch]; 332 out = &s->sb_samples[ch][0][0][0]; 333 for(j=0;j<36;j++) { 334 /* 32 samples at once */ 335 for(i=0;i<32;i++) { 336 s->samples_buf[ch][offset + (31 - i)] = samples[0]; 337 samples += incr; 338 } 339 340 /* filter */ 341 p = s->samples_buf[ch] + offset; 342 q = s->filter_bank; 343 /* maxsum = 23169 */ 344 for(i=0;i<64;i++) { 345 sum = p[0*64] * q[0*64]; 346 sum += p[1*64] * q[1*64]; 347 sum += p[2*64] * q[2*64]; 348 sum += p[3*64] * q[3*64]; 349 sum += p[4*64] * q[4*64]; 350 sum += p[5*64] * q[5*64]; 351 sum += p[6*64] * q[6*64]; 352 sum += p[7*64] * q[7*64]; 353 tmp[i] = sum; 354 p++; 355 q++; 356 } 357 tmp1[0] = tmp[16] >> WSHIFT; 358 for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT; 359 for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT; 360 361 idct32(out, tmp1); 362 363 /* advance of 32 samples */ 364 offset -= 32; 365 out += 32; 366 /* handle the wrap around */ 367 if (offset < 0) { 368 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32), 369 s->samples_buf[ch], (512 - 32) * 2); 370 offset = SAMPLES_BUF_SIZE - 512; 371 } 372 } 373 s->samples_offset[ch] = offset; 374} 375 376static void compute_scale_factors(MpegAudioContext *s, 377 unsigned char scale_code[SBLIMIT], 378 unsigned char scale_factors[SBLIMIT][3], 379 int sb_samples[3][12][SBLIMIT], 380 int sblimit) 381{ 382 int *p, vmax, v, n, i, j, k, code; 383 int index, d1, d2; 384 unsigned char *sf = &scale_factors[0][0]; 385 386 for(j=0;j<sblimit;j++) { 387 for(i=0;i<3;i++) { 388 /* find the max absolute value */ 389 p = &sb_samples[i][0][j]; 390 vmax = abs(*p); 391 for(k=1;k<12;k++) { 392 p += SBLIMIT; 393 v = abs(*p); 394 if (v > vmax) 395 vmax = v; 396 } 397 /* compute the scale factor index using log 2 computations */ 398 if (vmax > 1) { 399 n = av_log2(vmax); 400 /* n is the position of the MSB of vmax. now 401 use at most 2 compares to find the index */ 402 index = (21 - n) * 3 - 3; 403 if (index >= 0) { 404 while (vmax <= s->scale_factor_table[index+1]) 405 index++; 406 } else { 407 index = 0; /* very unlikely case of overflow */ 408 } 409 } else { 410 index = 62; /* value 63 is not allowed */ 411 } 412 413 av_dlog(NULL, "%2d:%d in=%x %x %d\n", 414 j, i, vmax, s->scale_factor_table[index], index); 415 /* store the scale factor */ 416 av_assert2(index >=0 && index <= 63); 417 sf[i] = index; 418 } 419 420 /* compute the transmission factor : look if the scale factors 421 are close enough to each other */ 422 d1 = s->scale_diff_table[sf[0] - sf[1] + 64]; 423 d2 = s->scale_diff_table[sf[1] - sf[2] + 64]; 424 425 /* handle the 25 cases */ 426 switch(d1 * 5 + d2) { 427 case 0*5+0: 428 case 0*5+4: 429 case 3*5+4: 430 case 4*5+0: 431 case 4*5+4: 432 code = 0; 433 break; 434 case 0*5+1: 435 case 0*5+2: 436 case 4*5+1: 437 case 4*5+2: 438 code = 3; 439 sf[2] = sf[1]; 440 break; 441 case 0*5+3: 442 case 4*5+3: 443 code = 3; 444 sf[1] = sf[2]; 445 break; 446 case 1*5+0: 447 case 1*5+4: 448 case 2*5+4: 449 code = 1; 450 sf[1] = sf[0]; 451 break; 452 case 1*5+1: 453 case 1*5+2: 454 case 2*5+0: 455 case 2*5+1: 456 case 2*5+2: 457 code = 2; 458 sf[1] = sf[2] = sf[0]; 459 break; 460 case 2*5+3: 461 case 3*5+3: 462 code = 2; 463 sf[0] = sf[1] = sf[2]; 464 break; 465 case 3*5+0: 466 case 3*5+1: 467 case 3*5+2: 468 code = 2; 469 sf[0] = sf[2] = sf[1]; 470 break; 471 case 1*5+3: 472 code = 2; 473 if (sf[0] > sf[2]) 474 sf[0] = sf[2]; 475 sf[1] = sf[2] = sf[0]; 476 break; 477 default: 478 av_assert2(0); //cannot happen 479 code = 0; /* kill warning */ 480 } 481 482 av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j, 483 sf[0], sf[1], sf[2], d1, d2, code); 484 scale_code[j] = code; 485 sf += 3; 486 } 487} 488 489/* The most important function : psycho acoustic module. In this 490 encoder there is basically none, so this is the worst you can do, 491 but also this is the simpler. */ 492static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) 493{ 494 int i; 495 496 for(i=0;i<s->sblimit;i++) { 497 smr[i] = (int)(fixed_smr[i] * 10); 498 } 499} 500 501 502#define SB_NOTALLOCATED 0 503#define SB_ALLOCATED 1 504#define SB_NOMORE 2 505 506/* Try to maximize the smr while using a number of bits inferior to 507 the frame size. I tried to make the code simpler, faster and 508 smaller than other encoders :-) */ 509static void compute_bit_allocation(MpegAudioContext *s, 510 short smr1[MPA_MAX_CHANNELS][SBLIMIT], 511 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 512 int *padding) 513{ 514 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size; 515 int incr; 516 short smr[MPA_MAX_CHANNELS][SBLIMIT]; 517 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT]; 518 const unsigned char *alloc; 519 520 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT); 521 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT); 522 memset(bit_alloc, 0, s->nb_channels * SBLIMIT); 523 524 /* compute frame size and padding */ 525 max_frame_size = s->frame_size; 526 s->frame_frac += s->frame_frac_incr; 527 if (s->frame_frac >= 65536) { 528 s->frame_frac -= 65536; 529 s->do_padding = 1; 530 max_frame_size += 8; 531 } else { 532 s->do_padding = 0; 533 } 534 535 /* compute the header + bit alloc size */ 536 current_frame_size = 32; 537 alloc = s->alloc_table; 538 for(i=0;i<s->sblimit;i++) { 539 incr = alloc[0]; 540 current_frame_size += incr * s->nb_channels; 541 alloc += 1 << incr; 542 } 543 for(;;) { 544 /* look for the subband with the largest signal to mask ratio */ 545 max_sb = -1; 546 max_ch = -1; 547 max_smr = INT_MIN; 548 for(ch=0;ch<s->nb_channels;ch++) { 549 for(i=0;i<s->sblimit;i++) { 550 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) { 551 max_smr = smr[ch][i]; 552 max_sb = i; 553 max_ch = ch; 554 } 555 } 556 } 557 if (max_sb < 0) 558 break; 559 av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n", 560 current_frame_size, max_frame_size, max_sb, max_ch, 561 bit_alloc[max_ch][max_sb]); 562 563 /* find alloc table entry (XXX: not optimal, should use 564 pointer table) */ 565 alloc = s->alloc_table; 566 for(i=0;i<max_sb;i++) { 567 alloc += 1 << alloc[0]; 568 } 569 570 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) { 571 /* nothing was coded for this band: add the necessary bits */ 572 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6; 573 incr += s->total_quant_bits[alloc[1]]; 574 } else { 575 /* increments bit allocation */ 576 b = bit_alloc[max_ch][max_sb]; 577 incr = s->total_quant_bits[alloc[b + 1]] - 578 s->total_quant_bits[alloc[b]]; 579 } 580 581 if (current_frame_size + incr <= max_frame_size) { 582 /* can increase size */ 583 b = ++bit_alloc[max_ch][max_sb]; 584 current_frame_size += incr; 585 /* decrease smr by the resolution we added */ 586 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]]; 587 /* max allocation size reached ? */ 588 if (b == ((1 << alloc[0]) - 1)) 589 subband_status[max_ch][max_sb] = SB_NOMORE; 590 else 591 subband_status[max_ch][max_sb] = SB_ALLOCATED; 592 } else { 593 /* cannot increase the size of this subband */ 594 subband_status[max_ch][max_sb] = SB_NOMORE; 595 } 596 } 597 *padding = max_frame_size - current_frame_size; 598 av_assert0(*padding >= 0); 599} 600 601/* 602 * Output the mpeg audio layer 2 frame. Note how the code is small 603 * compared to other encoders :-) 604 */ 605static void encode_frame(MpegAudioContext *s, 606 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], 607 int padding) 608{ 609 int i, j, k, l, bit_alloc_bits, b, ch; 610 unsigned char *sf; 611 int q[3]; 612 PutBitContext *p = &s->pb; 613 614 /* header */ 615 616 put_bits(p, 12, 0xfff); 617 put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */ 618 put_bits(p, 2, 4-2); /* layer 2 */ 619 put_bits(p, 1, 1); /* no error protection */ 620 put_bits(p, 4, s->bitrate_index); 621 put_bits(p, 2, s->freq_index); 622 put_bits(p, 1, s->do_padding); /* use padding */ 623 put_bits(p, 1, 0); /* private_bit */ 624 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO); 625 put_bits(p, 2, 0); /* mode_ext */ 626 put_bits(p, 1, 0); /* no copyright */ 627 put_bits(p, 1, 1); /* original */ 628 put_bits(p, 2, 0); /* no emphasis */ 629 630 /* bit allocation */ 631 j = 0; 632 for(i=0;i<s->sblimit;i++) { 633 bit_alloc_bits = s->alloc_table[j]; 634 for(ch=0;ch<s->nb_channels;ch++) { 635 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]); 636 } 637 j += 1 << bit_alloc_bits; 638 } 639 640 /* scale codes */ 641 for(i=0;i<s->sblimit;i++) { 642 for(ch=0;ch<s->nb_channels;ch++) { 643 if (bit_alloc[ch][i]) 644 put_bits(p, 2, s->scale_code[ch][i]); 645 } 646 } 647 648 /* scale factors */ 649 for(i=0;i<s->sblimit;i++) { 650 for(ch=0;ch<s->nb_channels;ch++) { 651 if (bit_alloc[ch][i]) { 652 sf = &s->scale_factors[ch][i][0]; 653 switch(s->scale_code[ch][i]) { 654 case 0: 655 put_bits(p, 6, sf[0]); 656 put_bits(p, 6, sf[1]); 657 put_bits(p, 6, sf[2]); 658 break; 659 case 3: 660 case 1: 661 put_bits(p, 6, sf[0]); 662 put_bits(p, 6, sf[2]); 663 break; 664 case 2: 665 put_bits(p, 6, sf[0]); 666 break; 667 } 668 } 669 } 670 } 671 672 /* quantization & write sub band samples */ 673 674 for(k=0;k<3;k++) { 675 for(l=0;l<12;l+=3) { 676 j = 0; 677 for(i=0;i<s->sblimit;i++) { 678 bit_alloc_bits = s->alloc_table[j]; 679 for(ch=0;ch<s->nb_channels;ch++) { 680 b = bit_alloc[ch][i]; 681 if (b) { 682 int qindex, steps, m, sample, bits; 683 /* we encode 3 sub band samples of the same sub band at a time */ 684 qindex = s->alloc_table[j+b]; 685 steps = ff_mpa_quant_steps[qindex]; 686 for(m=0;m<3;m++) { 687 sample = s->sb_samples[ch][k][l + m][i]; 688 /* divide by scale factor */ 689#if USE_FLOATS 690 { 691 float a; 692 a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]]; 693 q[m] = (int)((a + 1.0) * steps * 0.5); 694 } 695#else 696 { 697 int q1, e, shift, mult; 698 e = s->scale_factors[ch][i][k]; 699 shift = s->scale_factor_shift[e]; 700 mult = s->scale_factor_mult[e]; 701 702 /* normalize to P bits */ 703 if (shift < 0) 704 q1 = sample << (-shift); 705 else 706 q1 = sample >> shift; 707 q1 = (q1 * mult) >> P; 708 q1 += 1 << P; 709 if (q1 < 0) 710 q1 = 0; 711 q[m] = (q1 * (unsigned)steps) >> (P + 1); 712 } 713#endif 714 if (q[m] >= steps) 715 q[m] = steps - 1; 716 av_assert2(q[m] >= 0 && q[m] < steps); 717 } 718 bits = ff_mpa_quant_bits[qindex]; 719 if (bits < 0) { 720 /* group the 3 values to save bits */ 721 put_bits(p, -bits, 722 q[0] + steps * (q[1] + steps * q[2])); 723 } else { 724 put_bits(p, bits, q[0]); 725 put_bits(p, bits, q[1]); 726 put_bits(p, bits, q[2]); 727 } 728 } 729 } 730 /* next subband in alloc table */ 731 j += 1 << bit_alloc_bits; 732 } 733 } 734 } 735 736 /* padding */ 737 for(i=0;i<padding;i++) 738 put_bits(p, 1, 0); 739 740 /* flush */ 741 flush_put_bits(p); 742} 743 744static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, 745 const AVFrame *frame, int *got_packet_ptr) 746{ 747 MpegAudioContext *s = avctx->priv_data; 748 const int16_t *samples = (const int16_t *)frame->data[0]; 749 short smr[MPA_MAX_CHANNELS][SBLIMIT]; 750 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT]; 751 int padding, i, ret; 752 753 for(i=0;i<s->nb_channels;i++) { 754 filter(s, i, samples + i, s->nb_channels); 755 } 756 757 for(i=0;i<s->nb_channels;i++) { 758 compute_scale_factors(s, s->scale_code[i], s->scale_factors[i], 759 s->sb_samples[i], s->sblimit); 760 } 761 for(i=0;i<s->nb_channels;i++) { 762 psycho_acoustic_model(s, smr[i]); 763 } 764 compute_bit_allocation(s, smr, bit_alloc, &padding); 765 766 if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0) 767 return ret; 768 769 init_put_bits(&s->pb, avpkt->data, avpkt->size); 770 771 encode_frame(s, bit_alloc, padding); 772 773 if (frame->pts != AV_NOPTS_VALUE) 774 avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay); 775 776 avpkt->size = put_bits_count(&s->pb) / 8; 777 *got_packet_ptr = 1; 778 return 0; 779} 780 781static const AVCodecDefault mp2_defaults[] = { 782 { "b", "0" }, 783 { NULL }, 784}; 785 786