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 libavcodec/mpegaudio.c
  24  * The simplest mpeg audio layer 2 encoder.
  25  */
  26
  27 #include "avcodec.h"
  28 #include "put_bits.h"
  29
  30 #undef  CONFIG_MPEGAUDIO_HP
  31 #define CONFIG_MPEGAUDIO_HP 0
  32 #include "mpegaudio.h"
  33
  34 /* currently, cannot change these constants (need to modify
  35    quantization stage) */
  36 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
  37
  38 #define SAMPLES_BUF_SIZE 4096
  39
  40 typedef struct MpegAudioContext {
  41     PutBitContext pb;
  42     int nb_channels;
  43     int freq, bit_rate;
  44     int lsf;           /* 1 if mpeg2 low bitrate selected */
  45     int bitrate_index; /* bit rate */
  46     int freq_index;
  47     int frame_size; /* frame size, in bits, without padding */
  48     int64_t nb_samples; /* total number of samples encoded */
  49     /* padding computation */
  50     int frame_frac, frame_frac_incr, do_padding;
  51     short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
  52     int samples_offset[MPA_MAX_CHANNELS];       /* offset in samples_buf */
  53     int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
  54     unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
  55     /* code to group 3 scale factors */
  56     unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
  57     int sblimit; /* number of used subbands */
  58     const unsigned char *alloc_table;
  59 } MpegAudioContext;
  60
  61 /* define it to use floats in quantization (I don't like floats !) */
  62 //#define USE_FLOATS
  63
  64 #include "mpegaudiodata.h"
  65 #include "mpegaudiotab.h"
  66
  67 static av_cold int MPA_encode_init(AVCodecContext *avctx)
  68 {
  69     MpegAudioContext *s = avctx->priv_data;
  70     int freq = avctx->sample_rate;
  71     int bitrate = avctx->bit_rate;
  72     int channels = avctx->channels;
  73     int i, v, table;
  74     float a;
  75
  76     if (channels <= 0 || channels > 2){
  77         av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
  78         return -1;
  79     }
  80     bitrate = bitrate / 1000;
  81     s->nb_channels = channels;
  82     s->freq = freq;
  83     s->bit_rate = bitrate * 1000;
  84     avctx->frame_size = MPA_FRAME_SIZE;
  85
  86     /* encoding freq */
  87     s->lsf = 0;
  88     for(i=0;i<3;i++) {
  89         if (ff_mpa_freq_tab[i] == freq)
  90             break;
  91         if ((ff_mpa_freq_tab[i] / 2) == freq) {
  92             s->lsf = 1;
  93             break;
  94         }
  95     }
  96     if (i == 3){
  97         av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
  98         return -1;
  99     }
 100     s->freq_index = i;
 101
 102     /* encoding bitrate & frequency */
 103     for(i=0;i<15;i++) {
 104         if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
 105             break;
 106     }
 107     if (i == 15){
 108         av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
 109         return -1;
 110     }
 111     s->bitrate_index = i;
 112
 113     /* compute total header size & pad bit */
 114
 115     a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
 116     s->frame_size = ((int)a) * 8;
 117
 118     /* frame fractional size to compute padding */
 119     s->frame_frac = 0;
 120     s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
 121
 122     /* select the right allocation table */
 123     table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
 124
 125     /* number of used subbands */
 126     s->sblimit = ff_mpa_sblimit_table[table];
 127     s->alloc_table = ff_mpa_alloc_tables[table];
 128
 129     dprintf(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
 130             bitrate, freq, s->frame_size, table, s->frame_frac_incr);
 131
 132     for(i=0;i<s->nb_channels;i++)
 133         s->samples_offset[i] = 0;
 134
 135     for(i=0;i<257;i++) {
 136         int v;
 137         v = ff_mpa_enwindow[i];
 138 #if WFRAC_BITS != 16
 139         v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
 140 #endif
 141         filter_bank[i] = v;
 142         if ((i & 63) != 0)
 143             v = -v;
 144         if (i != 0)
 145             filter_bank[512 - i] = v;
 146     }
 147
 148     for(i=0;i<64;i++) {
 149         v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
 150         if (v <= 0)
 151             v = 1;
 152         scale_factor_table[i] = v;
 153 #ifdef USE_FLOATS
 154         scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
 155 #else
 156 #define P 15
 157         scale_factor_shift[i] = 21 - P - (i / 3);
 158         scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
 159 #endif
 160     }
 161     for(i=0;i<128;i++) {
 162         v = i - 64;
 163         if (v <= -3)
 164             v = 0;
 165         else if (v < 0)
 166             v = 1;
 167         else if (v == 0)
 168             v = 2;
 169         else if (v < 3)
 170             v = 3;
 171         else
 172             v = 4;
 173         scale_diff_table[i] = v;
 174     }
 175
 176     for(i=0;i<17;i++) {
 177         v = ff_mpa_quant_bits[i];
 178         if (v < 0)
 179             v = -v;
 180         else
 181             v = v * 3;
 182         total_quant_bits[i] = 12 * v;
 183     }
 184
 185     avctx->coded_frame= avcodec_alloc_frame();
 186     avctx->coded_frame->key_frame= 1;
 187
 188     return 0;
 189 }
 190
 191 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
 192 static void idct32(int *out, int *tab)
 193 {
 194     int i, j;
 195     int *t, *t1, xr;
 196     const int *xp = costab32;
 197
 198     for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
 199
 200     t = tab + 30;
 201     t1 = tab + 2;
 202     do {
 203         t[0] += t[-4];
 204         t[1] += t[1 - 4];
 205         t -= 4;
 206     } while (t != t1);
 207
 208     t = tab + 28;
 209     t1 = tab + 4;
 210     do {
 211         t[0] += t[-8];
 212         t[1] += t[1-8];
 213         t[2] += t[2-8];
 214         t[3] += t[3-8];
 215         t -= 8;
 216     } while (t != t1);
 217
 218     t = tab;
 219     t1 = tab + 32;
 220     do {
 221         t[ 3] = -t[ 3];
 222         t[ 6] = -t[ 6];
 223
 224         t[11] = -t[11];
 225         t[12] = -t[12];
 226         t[13] = -t[13];
 227         t[15] = -t[15];
 228         t += 16;
 229     } while (t != t1);
 230
 231
 232     t = tab;
 233     t1 = tab + 8;
 234     do {
 235         int x1, x2, x3, x4;
 236
 237         x3 = MUL(t[16], FIX(SQRT2*0.5));
 238         x4 = t[0] - x3;
 239         x3 = t[0] + x3;
 240
 241         x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
 242         x1 = MUL((t[8] - x2), xp[0]);
 243         x2 = MUL((t[8] + x2), xp[1]);
 244
 245         t[ 0] = x3 + x1;
 246         t[ 8] = x4 - x2;
 247         t[16] = x4 + x2;
 248         t[24] = x3 - x1;
 249         t++;
 250     } while (t != t1);
 251
 252     xp += 2;
 253     t = tab;
 254     t1 = tab + 4;
 255     do {
 256         xr = MUL(t[28],xp[0]);
 257         t[28] = (t[0] - xr);
 258         t[0] = (t[0] + xr);
 259
 260         xr = MUL(t[4],xp[1]);
 261         t[ 4] = (t[24] - xr);
 262         t[24] = (t[24] + xr);
 263
 264         xr = MUL(t[20],xp[2]);
 265         t[20] = (t[8] - xr);
 266         t[ 8] = (t[8] + xr);
 267
 268         xr = MUL(t[12],xp[3]);
 269         t[12] = (t[16] - xr);
 270         t[16] = (t[16] + xr);
 271         t++;
 272     } while (t != t1);
 273     xp += 4;
 274
 275     for (i = 0; i < 4; i++) {
 276         xr = MUL(tab[30-i*4],xp[0]);
 277         tab[30-i*4] = (tab[i*4] - xr);
 278         tab[   i*4] = (tab[i*4] + xr);
 279
 280         xr = MUL(tab[ 2+i*4],xp[1]);
 281         tab[ 2+i*4] = (tab[28-i*4] - xr);
 282         tab[28-i*4] = (tab[28-i*4] + xr);
 283
 284         xr = MUL(tab[31-i*4],xp[0]);
 285         tab[31-i*4] = (tab[1+i*4] - xr);
 286         tab[ 1+i*4] = (tab[1+i*4] + xr);
 287
 288         xr = MUL(tab[ 3+i*4],xp[1]);
 289         tab[ 3+i*4] = (tab[29-i*4] - xr);
 290         tab[29-i*4] = (tab[29-i*4] + xr);
 291
 292         xp += 2;
 293     }
 294
 295     t = tab + 30;
 296     t1 = tab + 1;
 297     do {
 298         xr = MUL(t1[0], *xp);
 299         t1[0] = (t[0] - xr);
 300         t[0] = (t[0] + xr);
 301         t -= 2;
 302         t1 += 2;
 303         xp++;
 304     } while (t >= tab);
 305
 306     for(i=0;i<32;i++) {
 307         out[i] = tab[bitinv32[i]];
 308     }
 309 }
 310
 311 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
 312
 313 static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
 314 {
 315     short *p, *q;
 316     int sum, offset, i, j;
 317     int tmp[64];
 318     int tmp1[32];
 319     int *out;
 320
 321     //    print_pow1(samples, 1152);
 322
 323     offset = s->samples_offset[ch];
 324     out = &s->sb_samples[ch][0][0][0];
 325     for(j=0;j<36;j++) {
 326         /* 32 samples at once */
 327         for(i=0;i<32;i++) {
 328             s->samples_buf[ch][offset + (31 - i)] = samples[0];
 329             samples += incr;
 330         }
 331
 332         /* filter */
 333         p = s->samples_buf[ch] + offset;
 334         q = filter_bank;
 335         /* maxsum = 23169 */
 336         for(i=0;i<64;i++) {
 337             sum = p[0*64] * q[0*64];
 338             sum += p[1*64] * q[1*64];
 339             sum += p[2*64] * q[2*64];
 340             sum += p[3*64] * q[3*64];
 341             sum += p[4*64] * q[4*64];
 342             sum += p[5*64] * q[5*64];
 343             sum += p[6*64] * q[6*64];
 344             sum += p[7*64] * q[7*64];
 345             tmp[i] = sum;
 346             p++;
 347             q++;
 348         }
 349         tmp1[0] = tmp[16] >> WSHIFT;
 350         for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
 351         for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
 352
 353         idct32(out, tmp1);
 354
 355         /* advance of 32 samples */
 356         offset -= 32;
 357         out += 32;
 358         /* handle the wrap around */
 359         if (offset < 0) {
 360             memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
 361                     s->samples_buf[ch], (512 - 32) * 2);
 362             offset = SAMPLES_BUF_SIZE - 512;
 363         }
 364     }
 365     s->samples_offset[ch] = offset;
 366
 367     //    print_pow(s->sb_samples, 1152);
 368 }
 369
 370 static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
 371                                   unsigned char scale_factors[SBLIMIT][3],
 372                                   int sb_samples[3][12][SBLIMIT],
 373                                   int sblimit)
 374 {
 375     int *p, vmax, v, n, i, j, k, code;
 376     int index, d1, d2;
 377     unsigned char *sf = &scale_factors[0][0];
 378
 379     for(j=0;j<sblimit;j++) {
 380         for(i=0;i<3;i++) {
 381             /* find the max absolute value */
 382             p = &sb_samples[i][0][j];
 383             vmax = abs(*p);
 384             for(k=1;k<12;k++) {
 385                 p += SBLIMIT;
 386                 v = abs(*p);
 387                 if (v > vmax)
 388                     vmax = v;
 389             }
 390             /* compute the scale factor index using log 2 computations */
 391             if (vmax > 1) {
 392                 n = av_log2(vmax);
 393                 /* n is the position of the MSB of vmax. now
 394                    use at most 2 compares to find the index */
 395                 index = (21 - n) * 3 - 3;
 396                 if (index >= 0) {
 397                     while (vmax <= scale_factor_table[index+1])
 398                         index++;
 399                 } else {
 400                     index = 0; /* very unlikely case of overflow */
 401                 }
 402             } else {
 403                 index = 62; /* value 63 is not allowed */
 404             }
 405
 406 #if 0
 407             printf("%2d:%d in=%x %x %d\n",
 408                    j, i, vmax, scale_factor_table[index], index);
 409 #endif
 410             /* store the scale factor */
 411             assert(index >=0 && index <= 63);
 412             sf[i] = index;
 413         }
 414
 415         /* compute the transmission factor : look if the scale factors
 416            are close enough to each other */
 417         d1 = scale_diff_table[sf[0] - sf[1] + 64];
 418         d2 = scale_diff_table[sf[1] - sf[2] + 64];
 419
 420         /* handle the 25 cases */
 421         switch(d1 * 5 + d2) {
 422         case 0*5+0:
 423         case 0*5+4:
 424         case 3*5+4:
 425         case 4*5+0:
 426         case 4*5+4:
 427             code = 0;
 428             break;
 429         case 0*5+1:
 430         case 0*5+2:
 431         case 4*5+1:
 432         case 4*5+2:
 433             code = 3;
 434             sf[2] = sf[1];
 435             break;
 436         case 0*5+3:
 437         case 4*5+3:
 438             code = 3;
 439             sf[1] = sf[2];
 440             break;
 441         case 1*5+0:
 442         case 1*5+4:
 443         case 2*5+4:
 444             code = 1;
 445             sf[1] = sf[0];
 446             break;
 447         case 1*5+1:
 448         case 1*5+2:
 449         case 2*5+0:
 450         case 2*5+1:
 451         case 2*5+2:
 452             code = 2;
 453             sf[1] = sf[2] = sf[0];
 454             break;
 455         case 2*5+3:
 456         case 3*5+3:
 457             code = 2;
 458             sf[0] = sf[1] = sf[2];
 459             break;
 460         case 3*5+0:
 461         case 3*5+1:
 462         case 3*5+2:
 463             code = 2;
 464             sf[0] = sf[2] = sf[1];
 465             break;
 466         case 1*5+3:
 467             code = 2;
 468             if (sf[0] > sf[2])
 469               sf[0] = sf[2];
 470             sf[1] = sf[2] = sf[0];
 471             break;
 472         default:
 473             assert(0); //cannot happen
 474             code = 0;           /* kill warning */
 475         }
 476
 477 #if 0
 478         printf("%d: %2d %2d %2d %d %d -> %d\n", j,
 479                sf[0], sf[1], sf[2], d1, d2, code);
 480 #endif
 481         scale_code[j] = code;
 482         sf += 3;
 483     }
 484 }
 485
 486 /* The most important function : psycho acoustic module. In this
 487    encoder there is basically none, so this is the worst you can do,
 488    but also this is the simpler. */
 489 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
 490 {
 491     int i;
 492
 493     for(i=0;i<s->sblimit;i++) {
 494         smr[i] = (int)(fixed_smr[i] * 10);
 495     }
 496 }
 497
 498
 499 #define SB_NOTALLOCATED  0
 500 #define SB_ALLOCATED     1
 501 #define SB_NOMORE        2
 502
 503 /* Try to maximize the smr while using a number of bits inferior to
 504    the frame size. I tried to make the code simpler, faster and
 505    smaller than other encoders :-) */
 506 static void compute_bit_allocation(MpegAudioContext *s,
 507                                    short smr1[MPA_MAX_CHANNELS][SBLIMIT],
 508                                    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
 509                                    int *padding)
 510 {
 511     int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
 512     int incr;
 513     short smr[MPA_MAX_CHANNELS][SBLIMIT];
 514     unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
 515     const unsigned char *alloc;
 516
 517     memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
 518     memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
 519     memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
 520
 521     /* compute frame size and padding */
 522     max_frame_size = s->frame_size;
 523     s->frame_frac += s->frame_frac_incr;
 524     if (s->frame_frac >= 65536) {
 525         s->frame_frac -= 65536;
 526         s->do_padding = 1;
 527         max_frame_size += 8;
 528     } else {
 529         s->do_padding = 0;
 530     }
 531
 532     /* compute the header + bit alloc size */
 533     current_frame_size = 32;
 534     alloc = s->alloc_table;
 535     for(i=0;i<s->sblimit;i++) {
 536         incr = alloc[0];
 537         current_frame_size += incr * s->nb_channels;
 538         alloc += 1 << incr;
 539     }
 540     for(;;) {
 541         /* look for the subband with the largest signal to mask ratio */
 542         max_sb = -1;
 543         max_ch = -1;
 544         max_smr = INT_MIN;
 545         for(ch=0;ch<s->nb_channels;ch++) {
 546             for(i=0;i<s->sblimit;i++) {
 547                 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
 548                     max_smr = smr[ch][i];
 549                     max_sb = i;
 550                     max_ch = ch;
 551                 }
 552             }
 553         }
 554 #if 0
 555         printf("current=%d max=%d max_sb=%d alloc=%d\n",
 556                current_frame_size, max_frame_size, max_sb,
 557                bit_alloc[max_sb]);
 558 #endif
 559         if (max_sb < 0)
 560             break;
 561
 562         /* find alloc table entry (XXX: not optimal, should use
 563            pointer table) */
 564         alloc = s->alloc_table;
 565         for(i=0;i<max_sb;i++) {
 566             alloc += 1 << alloc[0];
 567         }
 568
 569         if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
 570             /* nothing was coded for this band: add the necessary bits */
 571             incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
 572             incr += total_quant_bits[alloc[1]];
 573         } else {
 574             /* increments bit allocation */
 575             b = bit_alloc[max_ch][max_sb];
 576             incr = total_quant_bits[alloc[b + 1]] -
 577                 total_quant_bits[alloc[b]];
 578         }
 579
 580         if (current_frame_size + incr <= max_frame_size) {
 581             /* can increase size */
 582             b = ++bit_alloc[max_ch][max_sb];
 583             current_frame_size += incr;
 584             /* decrease smr by the resolution we added */
 585             smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
 586             /* max allocation size reached ? */
 587             if (b == ((1 << alloc[0]) - 1))
 588                 subband_status[max_ch][max_sb] = SB_NOMORE;
 589             else
 590                 subband_status[max_ch][max_sb] = SB_ALLOCATED;
 591         } else {
 592             /* cannot increase the size of this subband */
 593             subband_status[max_ch][max_sb] = SB_NOMORE;
 594         }
 595     }
 596     *padding = max_frame_size - current_frame_size;
 597     assert(*padding >= 0);
 598
 599 #if 0
 600     for(i=0;i<s->sblimit;i++) {
 601         printf("%d ", bit_alloc[i]);
 602     }
 603     printf("\n");
 604 #endif
 605 }
 606
 607 /*
 608  * Output the mpeg audio layer 2 frame. Note how the code is small
 609  * compared to other encoders :-)
 610  */
 611 static void encode_frame(MpegAudioContext *s,
 612                          unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
 613                          int padding)
 614 {
 615     int i, j, k, l, bit_alloc_bits, b, ch;
 616     unsigned char *sf;
 617     int q[3];
 618     PutBitContext *p = &s->pb;
 619
 620     /* header */
 621
 622     put_bits(p, 12, 0xfff);
 623     put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
 624     put_bits(p, 2, 4-2);  /* layer 2 */
 625     put_bits(p, 1, 1); /* no error protection */
 626     put_bits(p, 4, s->bitrate_index);
 627     put_bits(p, 2, s->freq_index);
 628     put_bits(p, 1, s->do_padding); /* use padding */
 629     put_bits(p, 1, 0);             /* private_bit */
 630     put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
 631     put_bits(p, 2, 0); /* mode_ext */
 632     put_bits(p, 1, 0); /* no copyright */
 633     put_bits(p, 1, 1); /* original */
 634     put_bits(p, 2, 0); /* no emphasis */
 635
 636     /* bit allocation */
 637     j = 0;
 638     for(i=0;i<s->sblimit;i++) {
 639         bit_alloc_bits = s->alloc_table[j];
 640         for(ch=0;ch<s->nb_channels;ch++) {
 641             put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
 642         }
 643         j += 1 << bit_alloc_bits;
 644     }
 645
 646     /* scale codes */
 647     for(i=0;i<s->sblimit;i++) {
 648         for(ch=0;ch<s->nb_channels;ch++) {
 649             if (bit_alloc[ch][i])
 650                 put_bits(p, 2, s->scale_code[ch][i]);
 651         }
 652     }
 653
 654     /* scale factors */
 655     for(i=0;i<s->sblimit;i++) {
 656         for(ch=0;ch<s->nb_channels;ch++) {
 657             if (bit_alloc[ch][i]) {
 658                 sf = &s->scale_factors[ch][i][0];
 659                 switch(s->scale_code[ch][i]) {
 660                 case 0:
 661                     put_bits(p, 6, sf[0]);
 662                     put_bits(p, 6, sf[1]);
 663                     put_bits(p, 6, sf[2]);
 664                     break;
 665                 case 3:
 666                 case 1:
 667                     put_bits(p, 6, sf[0]);
 668                     put_bits(p, 6, sf[2]);
 669                     break;
 670                 case 2:
 671                     put_bits(p, 6, sf[0]);
 672                     break;
 673                 }
 674             }
 675         }
 676     }
 677
 678     /* quantization & write sub band samples */
 679
 680     for(k=0;k<3;k++) {
 681         for(l=0;l<12;l+=3) {
 682             j = 0;
 683             for(i=0;i<s->sblimit;i++) {
 684                 bit_alloc_bits = s->alloc_table[j];
 685                 for(ch=0;ch<s->nb_channels;ch++) {
 686                     b = bit_alloc[ch][i];
 687                     if (b) {
 688                         int qindex, steps, m, sample, bits;
 689                         /* we encode 3 sub band samples of the same sub band at a time */
 690                         qindex = s->alloc_table[j+b];
 691                         steps = ff_mpa_quant_steps[qindex];
 692                         for(m=0;m<3;m++) {
 693                             sample = s->sb_samples[ch][k][l + m][i];
 694                             /* divide by scale factor */
 695 #ifdef USE_FLOATS
 696                             {
 697                                 float a;
 698                                 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
 699                                 q[m] = (int)((a + 1.0) * steps * 0.5);
 700                             }
 701 #else
 702                             {
 703                                 int q1, e, shift, mult;
 704                                 e = s->scale_factors[ch][i][k];
 705                                 shift = scale_factor_shift[e];
 706                                 mult = scale_factor_mult[e];
 707
 708                                 /* normalize to P bits */
 709                                 if (shift < 0)
 710                                     q1 = sample << (-shift);
 711                                 else
 712                                     q1 = sample >> shift;
 713                                 q1 = (q1 * mult) >> P;
 714                                 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
 715                             }
 716 #endif
 717                             if (q[m] >= steps)
 718                                 q[m] = steps - 1;
 719                             assert(q[m] >= 0 && q[m] < steps);
 720                         }
 721                         bits = ff_mpa_quant_bits[qindex];
 722                         if (bits < 0) {
 723                             /* group the 3 values to save bits */
 724                             put_bits(p, -bits,
 725                                      q[0] + steps * (q[1] + steps * q[2]));
 726 #if 0
 727                             printf("%d: gr1 %d\n",
 728                                    i, q[0] + steps * (q[1] + steps * q[2]));
 729 #endif
 730                         } else {
 731 #if 0
 732                             printf("%d: gr3 %d %d %d\n",
 733                                    i, q[0], q[1], q[2]);
 734 #endif
 735                             put_bits(p, bits, q[0]);
 736                             put_bits(p, bits, q[1]);
 737                             put_bits(p, bits, q[2]);
 738                         }
 739                     }
 740                 }
 741                 /* next subband in alloc table */
 742                 j += 1 << bit_alloc_bits;
 743             }
 744         }
 745     }
 746
 747     /* padding */
 748     for(i=0;i<padding;i++)
 749         put_bits(p, 1, 0);
 750
 751     /* flush */
 752     flush_put_bits(p);
 753 }
 754
 755 static int MPA_encode_frame(AVCodecContext *avctx,
 756                             unsigned char *frame, int buf_size, void *data)
 757 {
 758     MpegAudioContext *s = avctx->priv_data;
 759     short *samples = data;
 760     short smr[MPA_MAX_CHANNELS][SBLIMIT];
 761     unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
 762     int padding, i;
 763
 764     for(i=0;i<s->nb_channels;i++) {
 765         filter(s, i, samples + i, s->nb_channels);
 766     }
 767
 768     for(i=0;i<s->nb_channels;i++) {
 769         compute_scale_factors(s->scale_code[i], s->scale_factors[i],
 770                               s->sb_samples[i], s->sblimit);
 771     }
 772     for(i=0;i<s->nb_channels;i++) {
 773         psycho_acoustic_model(s, smr[i]);
 774     }
 775     compute_bit_allocation(s, smr, bit_alloc, &padding);
 776
 777     init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
 778
 779     encode_frame(s, bit_alloc, padding);
 780
 781     s->nb_samples += MPA_FRAME_SIZE;
 782     return put_bits_ptr(&s->pb) - s->pb.buf;
 783 }
 784
 785 static av_cold int MPA_encode_close(AVCodecContext *avctx)
 786 {
 787     av_freep(&avctx->coded_frame);
 788     return 0;
 789 }
 790
 791 AVCodec mp2_encoder = {
 792     "mp2",
 793     CODEC_TYPE_AUDIO,
 794     CODEC_ID_MP2,
 795     sizeof(MpegAudioContext),
 796     MPA_encode_init,
 797     MPA_encode_frame,
 798     MPA_encode_close,
 799     NULL,
 800     .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
 801     .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
 802 };
 803
 804 #undef FIX