libavcodec/mpegaudio.c

   1 /*
   2  * The simplest mpeg audio layer 2 encoder
   3  * Copyright (c) 2000, 2001 Fabrice Bellard.
   4  *
   5  * This library is free software; you can redistribute it and/or
   6  * modify it under the terms of the GNU Lesser General Public
   7  * License as published by the Free Software Foundation; either
   8  * version 2 of the License, or (at your option) any later version.
   9  *
  10  * This library 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 GNU
  13  * Lesser General Public License for more details.
  14  *
  15  * You should have received a copy of the GNU Lesser General Public
  16  * License along with this library; if not, write to the Free Software
  17  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  18  */
  19
  20 /**
  21  * @file mpegaudio.c
  22  * The simplest mpeg audio layer 2 encoder.
  23  */
  24
  25 #include "avcodec.h"
  26 #include "mpegaudio.h"
  27
  28 /* currently, cannot change these constants (need to modify
  29    quantization stage) */
  30 #define FRAC_BITS 15
  31 #define WFRAC_BITS  14
  32 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
  33 #define FIX(a)   ((int)((a) * (1 << FRAC_BITS)))
  34
  35 #define SAMPLES_BUF_SIZE 4096
  36
  37 typedef struct MpegAudioContext {
  38     PutBitContext pb;
  39     int nb_channels;
  40     int freq, bit_rate;
  41     int lsf;           /* 1 if mpeg2 low bitrate selected */
  42     int bitrate_index; /* bit rate */
  43     int freq_index;
  44     int frame_size; /* frame size, in bits, without padding */
  45     int64_t nb_samples; /* total number of samples encoded */
  46     /* padding computation */
  47     int frame_frac, frame_frac_incr, do_padding;
  48     short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
  49     int samples_offset[MPA_MAX_CHANNELS];       /* offset in samples_buf */
  50     int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
  51     unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
  52     /* code to group 3 scale factors */
  53     unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
  54     int sblimit; /* number of used subbands */
  55     const unsigned char *alloc_table;
  56 } MpegAudioContext;
  57
  58 /* define it to use floats in quantization (I don't like floats !) */
  59 //#define USE_FLOATS
  60
  61 #include "mpegaudiotab.h"
  62
  63 static int MPA_encode_init(AVCodecContext *avctx)
  64 {
  65     MpegAudioContext *s = avctx->priv_data;
  66     int freq = avctx->sample_rate;
  67     int bitrate = avctx->bit_rate;
  68     int channels = avctx->channels;
  69     int i, v, table;
  70     float a;
  71
  72     if (channels > 2)
  73         return -1;
  74     bitrate = bitrate / 1000;
  75     s->nb_channels = channels;
  76     s->freq = freq;
  77     s->bit_rate = bitrate * 1000;
  78     avctx->frame_size = MPA_FRAME_SIZE;
  79
  80     /* encoding freq */
  81     s->lsf = 0;
  82     for(i=0;i<3;i++) {
  83         if (mpa_freq_tab[i] == freq)
  84             break;
  85         if ((mpa_freq_tab[i] / 2) == freq) {
  86             s->lsf = 1;
  87             break;
  88         }
  89     }
  90     if (i == 3)
  91         return -1;
  92     s->freq_index = i;
  93
  94     /* encoding bitrate & frequency */
  95     for(i=0;i<15;i++) {
  96         if (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
  97             break;
  98     }
  99     if (i == 15)
 100         return -1;
 101     s->bitrate_index = i;
 102
 103     /* compute total header size & pad bit */
 104
 105     a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
 106     s->frame_size = ((int)a) * 8;
 107
 108     /* frame fractional size to compute padding */
 109     s->frame_frac = 0;
 110     s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
 111
 112     /* select the right allocation table */
 113     table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
 114
 115     /* number of used subbands */
 116     s->sblimit = sblimit_table[table];
 117     s->alloc_table = alloc_tables[table];
 118
 119 #ifdef DEBUG
 120     av_log(AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
 121            bitrate, freq, s->frame_size, table, s->frame_frac_incr);
 122 #endif
 123
 124     for(i=0;i<s->nb_channels;i++)
 125         s->samples_offset[i] = 0;
 126
 127     for(i=0;i<257;i++) {
 128         int v;
 129         v = mpa_enwindow[i];
 130 #if WFRAC_BITS != 16
 131         v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
 132 #endif
 133         filter_bank[i] = v;
 134         if ((i & 63) != 0)
 135             v = -v;
 136         if (i != 0)
 137             filter_bank[512 - i] = v;
 138     }
 139
 140     for(i=0;i<64;i++) {
 141         v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
 142         if (v <= 0)
 143             v = 1;
 144         scale_factor_table[i] = v;
 145 #ifdef USE_FLOATS
 146         scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
 147 #else
 148 #define P 15
 149         scale_factor_shift[i] = 21 - P - (i / 3);
 150         scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
 151 #endif
 152     }
 153     for(i=0;i<128;i++) {
 154         v = i - 64;
 155         if (v <= -3)
 156             v = 0;
 157         else if (v < 0)
 158             v = 1;
 159         else if (v == 0)
 160             v = 2;
 161         else if (v < 3)
 162             v = 3;
 163         else
 164             v = 4;
 165         scale_diff_table[i] = v;
 166     }
 167
 168     for(i=0;i<17;i++) {
 169         v = quant_bits[i];
 170         if (v < 0)
 171             v = -v;
 172         else
 173             v = v * 3;
 174         total_quant_bits[i] = 12 * v;
 175     }
 176
 177     avctx->coded_frame= avcodec_alloc_frame();
 178     avctx->coded_frame->key_frame= 1;
 179
 180     return 0;
 181 }
 182
 183 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
 184 static void idct32(int *out, int *tab)
 185 {
 186     int i, j;
 187     int *t, *t1, xr;
 188     const int *xp = costab32;
 189
 190     for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
 191
 192     t = tab + 30;
 193     t1 = tab + 2;
 194     do {
 195         t[0] += t[-4];
 196         t[1] += t[1 - 4];
 197         t -= 4;
 198     } while (t != t1);
 199
 200     t = tab + 28;
 201     t1 = tab + 4;
 202     do {
 203         t[0] += t[-8];
 204         t[1] += t[1-8];
 205         t[2] += t[2-8];
 206         t[3] += t[3-8];
 207         t -= 8;
 208     } while (t != t1);
 209
 210     t = tab;
 211     t1 = tab + 32;
 212     do {
 213         t[ 3] = -t[ 3];
 214         t[ 6] = -t[ 6];
 215
 216         t[11] = -t[11];
 217         t[12] = -t[12];
 218         t[13] = -t[13];
 219         t[15] = -t[15];
 220         t += 16;
 221     } while (t != t1);
 222
 223
 224     t = tab;
 225     t1 = tab + 8;
 226     do {
 227         int x1, x2, x3, x4;
 228
 229         x3 = MUL(t[16], FIX(SQRT2*0.5));
 230         x4 = t[0] - x3;
 231         x3 = t[0] + x3;
 232
 233         x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
 234         x1 = MUL((t[8] - x2), xp[0]);
 235         x2 = MUL((t[8] + x2), xp[1]);
 236
 237         t[ 0] = x3 + x1;
 238         t[ 8] = x4 - x2;
 239         t[16] = x4 + x2;
 240         t[24] = x3 - x1;
 241         t++;
 242     } while (t != t1);
 243
 244     xp += 2;
 245     t = tab;
 246     t1 = tab + 4;
 247     do {
 248         xr = MUL(t[28],xp[0]);
 249         t[28] = (t[0] - xr);
 250         t[0] = (t[0] + xr);
 251
 252         xr = MUL(t[4],xp[1]);
 253         t[ 4] = (t[24] - xr);
 254         t[24] = (t[24] + xr);
 255
 256         xr = MUL(t[20],xp[2]);
 257         t[20] = (t[8] - xr);
 258         t[ 8] = (t[8] + xr);
 259
 260         xr = MUL(t[12],xp[3]);
 261         t[12] = (t[16] - xr);
 262         t[16] = (t[16] + xr);
 263         t++;
 264     } while (t != t1);
 265     xp += 4;
 266
 267     for (i = 0; i < 4; i++) {
 268         xr = MUL(tab[30-i*4],xp[0]);
 269         tab[30-i*4] = (tab[i*4] - xr);
 270         tab[   i*4] = (tab[i*4] + xr);
 271
 272         xr = MUL(tab[ 2+i*4],xp[1]);
 273         tab[ 2+i*4] = (tab[28-i*4] - xr);
 274         tab[28-i*4] = (tab[28-i*4] + xr);
 275
 276         xr = MUL(tab[31-i*4],xp[0]);
 277         tab[31-i*4] = (tab[1+i*4] - xr);
 278         tab[ 1+i*4] = (tab[1+i*4] + xr);
 279
 280         xr = MUL(tab[ 3+i*4],xp[1]);
 281         tab[ 3+i*4] = (tab[29-i*4] - xr);
 282         tab[29-i*4] = (tab[29-i*4] + xr);
 283
 284         xp += 2;
 285     }
 286
 287     t = tab + 30;
 288     t1 = tab + 1;
 289     do {
 290         xr = MUL(t1[0], *xp);
 291         t1[0] = (t[0] - xr);
 292         t[0] = (t[0] + xr);
 293         t -= 2;
 294         t1 += 2;
 295         xp++;
 296     } while (t >= tab);
 297
 298     for(i=0;i<32;i++) {
 299         out[i] = tab[bitinv32[i]];
 300     }
 301 }
 302
 303 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
 304
 305 static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
 306 {
 307     short *p, *q;
 308     int sum, offset, i, j;
 309     int tmp[64];
 310     int tmp1[32];
 311     int *out;
 312
 313     //    print_pow1(samples, 1152);
 314
 315     offset = s->samples_offset[ch];
 316     out = &s->sb_samples[ch][0][0][0];
 317     for(j=0;j<36;j++) {
 318         /* 32 samples at once */
 319         for(i=0;i<32;i++) {
 320             s->samples_buf[ch][offset + (31 - i)] = samples[0];
 321             samples += incr;
 322         }
 323
 324         /* filter */
 325         p = s->samples_buf[ch] + offset;
 326         q = filter_bank;
 327         /* maxsum = 23169 */
 328         for(i=0;i<64;i++) {
 329             sum = p[0*64] * q[0*64];
 330             sum += p[1*64] * q[1*64];
 331             sum += p[2*64] * q[2*64];
 332             sum += p[3*64] * q[3*64];
 333             sum += p[4*64] * q[4*64];
 334             sum += p[5*64] * q[5*64];
 335             sum += p[6*64] * q[6*64];
 336             sum += p[7*64] * q[7*64];
 337             tmp[i] = sum;
 338             p++;
 339             q++;
 340         }
 341         tmp1[0] = tmp[16] >> WSHIFT;
 342         for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
 343         for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
 344
 345         idct32(out, tmp1);
 346
 347         /* advance of 32 samples */
 348         offset -= 32;
 349         out += 32;
 350         /* handle the wrap around */
 351         if (offset < 0) {
 352             memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
 353                     s->samples_buf[ch], (512 - 32) * 2);
 354             offset = SAMPLES_BUF_SIZE - 512;
 355         }
 356     }
 357     s->samples_offset[ch] = offset;
 358
 359     //    print_pow(s->sb_samples, 1152);
 360 }
 361
 362 static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
 363                                   unsigned char scale_factors[SBLIMIT][3],
 364                                   int sb_samples[3][12][SBLIMIT],
 365                                   int sblimit)
 366 {
 367     int *p, vmax, v, n, i, j, k, code;
 368     int index, d1, d2;
 369     unsigned char *sf = &scale_factors[0][0];
 370
 371     for(j=0;j<sblimit;j++) {
 372         for(i=0;i<3;i++) {
 373             /* find the max absolute value */
 374             p = &sb_samples[i][0][j];
 375             vmax = abs(*p);
 376             for(k=1;k<12;k++) {
 377                 p += SBLIMIT;
 378                 v = abs(*p);
 379                 if (v > vmax)
 380                     vmax = v;
 381             }
 382             /* compute the scale factor index using log 2 computations */
 383             if (vmax > 0) {
 384                 n = av_log2(vmax);
 385                 /* n is the position of the MSB of vmax. now
 386                    use at most 2 compares to find the index */
 387                 index = (21 - n) * 3 - 3;
 388                 if (index >= 0) {
 389                     while (vmax <= scale_factor_table[index+1])
 390                         index++;
 391                 } else {
 392                     index = 0; /* very unlikely case of overflow */
 393                 }
 394             } else {
 395                 index = 62; /* value 63 is not allowed */
 396             }
 397
 398 #if 0
 399             printf("%2d:%d in=%x %x %d\n",
 400                    j, i, vmax, scale_factor_table[index], index);
 401 #endif
 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             av_abort();
 466         }
 467
 468 #if 0
 469         printf("%d: %2d %2d %2d %d %d -> %d\n", j,
 470                sf[0], sf[1], sf[2], d1, d2, code);
 471 #endif
 472         scale_code[j] = code;
 473         sf += 3;
 474     }
 475 }
 476
 477 /* The most important function : psycho acoustic module. In this
 478    encoder there is basically none, so this is the worst you can do,
 479    but also this is the simpler. */
 480 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
 481 {
 482     int i;
 483
 484     for(i=0;i<s->sblimit;i++) {
 485         smr[i] = (int)(fixed_smr[i] * 10);
 486     }
 487 }
 488
 489
 490 #define SB_NOTALLOCATED  0
 491 #define SB_ALLOCATED     1
 492 #define SB_NOMORE        2
 493
 494 /* Try to maximize the smr while using a number of bits inferior to
 495    the frame size. I tried to make the code simpler, faster and
 496    smaller than other encoders :-) */
 497 static void compute_bit_allocation(MpegAudioContext *s,
 498                                    short smr1[MPA_MAX_CHANNELS][SBLIMIT],
 499                                    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
 500                                    int *padding)
 501 {
 502     int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
 503     int incr;
 504     short smr[MPA_MAX_CHANNELS][SBLIMIT];
 505     unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
 506     const unsigned char *alloc;
 507
 508     memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
 509     memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
 510     memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
 511
 512     /* compute frame size and padding */
 513     max_frame_size = s->frame_size;
 514     s->frame_frac += s->frame_frac_incr;
 515     if (s->frame_frac >= 65536) {
 516         s->frame_frac -= 65536;
 517         s->do_padding = 1;
 518         max_frame_size += 8;
 519     } else {
 520         s->do_padding = 0;
 521     }
 522
 523     /* compute the header + bit alloc size */
 524     current_frame_size = 32;
 525     alloc = s->alloc_table;
 526     for(i=0;i<s->sblimit;i++) {
 527         incr = alloc[0];
 528         current_frame_size += incr * s->nb_channels;
 529         alloc += 1 << incr;
 530     }
 531     for(;;) {
 532         /* look for the subband with the largest signal to mask ratio */
 533         max_sb = -1;
 534         max_ch = -1;
 535         max_smr = 0x80000000;
 536         for(ch=0;ch<s->nb_channels;ch++) {
 537             for(i=0;i<s->sblimit;i++) {
 538                 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
 539                     max_smr = smr[ch][i];
 540                     max_sb = i;
 541                     max_ch = ch;
 542                 }
 543             }
 544         }
 545 #if 0
 546         printf("current=%d max=%d max_sb=%d alloc=%d\n",
 547                current_frame_size, max_frame_size, max_sb,
 548                bit_alloc[max_sb]);
 549 #endif
 550         if (max_sb < 0)
 551             break;
 552
 553         /* find alloc table entry (XXX: not optimal, should use
 554            pointer table) */
 555         alloc = s->alloc_table;
 556         for(i=0;i<max_sb;i++) {
 557             alloc += 1 << alloc[0];
 558         }
 559
 560         if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
 561             /* nothing was coded for this band: add the necessary bits */
 562             incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
 563             incr += total_quant_bits[alloc[1]];
 564         } else {
 565             /* increments bit allocation */
 566             b = bit_alloc[max_ch][max_sb];
 567             incr = total_quant_bits[alloc[b + 1]] -
 568                 total_quant_bits[alloc[b]];
 569         }
 570
 571         if (current_frame_size + incr <= max_frame_size) {
 572             /* can increase size */
 573             b = ++bit_alloc[max_ch][max_sb];
 574             current_frame_size += incr;
 575             /* decrease smr by the resolution we added */
 576             smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
 577             /* max allocation size reached ? */
 578             if (b == ((1 << alloc[0]) - 1))
 579                 subband_status[max_ch][max_sb] = SB_NOMORE;
 580             else
 581                 subband_status[max_ch][max_sb] = SB_ALLOCATED;
 582         } else {
 583             /* cannot increase the size of this subband */
 584             subband_status[max_ch][max_sb] = SB_NOMORE;
 585         }
 586     }
 587     *padding = max_frame_size - current_frame_size;
 588     assert(*padding >= 0);
 589
 590 #if 0
 591     for(i=0;i<s->sblimit;i++) {
 592         printf("%d ", bit_alloc[i]);
 593     }
 594     printf("\n");
 595 #endif
 596 }
 597
 598 /*
 599  * Output the mpeg audio layer 2 frame. Note how the code is small
 600  * compared to other encoders :-)
 601  */
 602 static void encode_frame(MpegAudioContext *s,
 603                          unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
 604                          int padding)
 605 {
 606     int i, j, k, l, bit_alloc_bits, b, ch;
 607     unsigned char *sf;
 608     int q[3];
 609     PutBitContext *p = &s->pb;
 610
 611     /* header */
 612
 613     put_bits(p, 12, 0xfff);
 614     put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
 615     put_bits(p, 2, 4-2);  /* layer 2 */
 616     put_bits(p, 1, 1); /* no error protection */
 617     put_bits(p, 4, s->bitrate_index);
 618     put_bits(p, 2, s->freq_index);
 619     put_bits(p, 1, s->do_padding); /* use padding */
 620     put_bits(p, 1, 0);             /* private_bit */
 621     put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
 622     put_bits(p, 2, 0); /* mode_ext */
 623     put_bits(p, 1, 0); /* no copyright */
 624     put_bits(p, 1, 1); /* original */
 625     put_bits(p, 2, 0); /* no emphasis */
 626
 627     /* bit allocation */
 628     j = 0;
 629     for(i=0;i<s->sblimit;i++) {
 630         bit_alloc_bits = s->alloc_table[j];
 631         for(ch=0;ch<s->nb_channels;ch++) {
 632             put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
 633         }
 634         j += 1 << bit_alloc_bits;
 635     }
 636
 637     /* scale codes */
 638     for(i=0;i<s->sblimit;i++) {
 639         for(ch=0;ch<s->nb_channels;ch++) {
 640             if (bit_alloc[ch][i])
 641                 put_bits(p, 2, s->scale_code[ch][i]);
 642         }
 643     }
 644
 645     /* scale factors */
 646     for(i=0;i<s->sblimit;i++) {
 647         for(ch=0;ch<s->nb_channels;ch++) {
 648             if (bit_alloc[ch][i]) {
 649                 sf = &s->scale_factors[ch][i][0];
 650                 switch(s->scale_code[ch][i]) {
 651                 case 0:
 652                     put_bits(p, 6, sf[0]);
 653                     put_bits(p, 6, sf[1]);
 654                     put_bits(p, 6, sf[2]);
 655                     break;
 656                 case 3:
 657                 case 1:
 658                     put_bits(p, 6, sf[0]);
 659                     put_bits(p, 6, sf[2]);
 660                     break;
 661                 case 2:
 662                     put_bits(p, 6, sf[0]);
 663                     break;
 664                 }
 665             }
 666         }
 667     }
 668
 669     /* quantization & write sub band samples */
 670
 671     for(k=0;k<3;k++) {
 672         for(l=0;l<12;l+=3) {
 673             j = 0;
 674             for(i=0;i<s->sblimit;i++) {
 675                 bit_alloc_bits = s->alloc_table[j];
 676                 for(ch=0;ch<s->nb_channels;ch++) {
 677                     b = bit_alloc[ch][i];
 678                     if (b) {
 679                         int qindex, steps, m, sample, bits;
 680                         /* we encode 3 sub band samples of the same sub band at a time */
 681                         qindex = s->alloc_table[j+b];
 682                         steps = quant_steps[qindex];
 683                         for(m=0;m<3;m++) {
 684                             sample = s->sb_samples[ch][k][l + m][i];
 685                             /* divide by scale factor */
 686 #ifdef USE_FLOATS
 687                             {
 688                                 float a;
 689                                 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
 690                                 q[m] = (int)((a + 1.0) * steps * 0.5);
 691                             }
 692 #else
 693                             {
 694                                 int q1, e, shift, mult;
 695                                 e = s->scale_factors[ch][i][k];
 696                                 shift = scale_factor_shift[e];
 697                                 mult = scale_factor_mult[e];
 698
 699                                 /* normalize to P bits */
 700                                 if (shift < 0)
 701                                     q1 = sample << (-shift);
 702                                 else
 703                                     q1 = sample >> shift;
 704                                 q1 = (q1 * mult) >> P;
 705                                 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
 706                             }
 707 #endif
 708                             if (q[m] >= steps)
 709                                 q[m] = steps - 1;
 710                             assert(q[m] >= 0 && q[m] < steps);
 711                         }
 712                         bits = quant_bits[qindex];
 713                         if (bits < 0) {
 714                             /* group the 3 values to save bits */
 715                             put_bits(p, -bits,
 716                                      q[0] + steps * (q[1] + steps * q[2]));
 717 #if 0
 718                             printf("%d: gr1 %d\n",
 719                                    i, q[0] + steps * (q[1] + steps * q[2]));
 720 #endif
 721                         } else {
 722 #if 0
 723                             printf("%d: gr3 %d %d %d\n",
 724                                    i, q[0], q[1], q[2]);
 725 #endif
 726                             put_bits(p, bits, q[0]);
 727                             put_bits(p, bits, q[1]);
 728                             put_bits(p, bits, q[2]);
 729                         }
 730                     }
 731                 }
 732                 /* next subband in alloc table */
 733                 j += 1 << bit_alloc_bits;
 734             }
 735         }
 736     }
 737
 738     /* padding */
 739     for(i=0;i<padding;i++)
 740         put_bits(p, 1, 0);
 741
 742     /* flush */
 743     flush_put_bits(p);
 744 }
 745
 746 static int MPA_encode_frame(AVCodecContext *avctx,
 747                             unsigned char *frame, int buf_size, void *data)
 748 {
 749     MpegAudioContext *s = avctx->priv_data;
 750     short *samples = data;
 751     short smr[MPA_MAX_CHANNELS][SBLIMIT];
 752     unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
 753     int padding, i;
 754
 755     for(i=0;i<s->nb_channels;i++) {
 756         filter(s, i, samples + i, s->nb_channels);
 757     }
 758
 759     for(i=0;i<s->nb_channels;i++) {
 760         compute_scale_factors(s->scale_code[i], s->scale_factors[i],
 761                               s->sb_samples[i], s->sblimit);
 762     }
 763     for(i=0;i<s->nb_channels;i++) {
 764         psycho_acoustic_model(s, smr[i]);
 765     }
 766     compute_bit_allocation(s, smr, bit_alloc, &padding);
 767
 768     init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
 769
 770     encode_frame(s, bit_alloc, padding);
 771
 772     s->nb_samples += MPA_FRAME_SIZE;
 773     return pbBufPtr(&s->pb) - s->pb.buf;
 774 }
 775
 776 static int MPA_encode_close(AVCodecContext *avctx)
 777 {
 778     av_freep(&avctx->coded_frame);
 779     return 0;
 780 }
 781
 782 AVCodec mp2_encoder = {
 783     "mp2",
 784     CODEC_TYPE_AUDIO,
 785     CODEC_ID_MP2,
 786     sizeof(MpegAudioContext),
 787     MPA_encode_init,
 788     MPA_encode_frame,
 789     MPA_encode_close,
 790     NULL,
 791 };
 792
 793 #undef FIX