libavcodec/mpegaudioenc.c

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