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 #ifdef DEBUG
 130     av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
 131            bitrate, freq, s->frame_size, table, s->frame_frac_incr);
 132 #endif
 133
 134     for(i=0;i<s->nb_channels;i++)
 135         s->samples_offset[i] = 0;
 136
 137     for(i=0;i<257;i++) {
 138         int v;
 139         v = ff_mpa_enwindow[i];
 140 #if WFRAC_BITS != 16
 141         v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
 142 #endif
 143         filter_bank[i] = v;
 144         if ((i & 63) != 0)
 145             v = -v;
 146         if (i != 0)
 147             filter_bank[512 - i] = v;
 148     }
 149
 150     for(i=0;i<64;i++) {
 151         v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
 152         if (v <= 0)
 153             v = 1;
 154         scale_factor_table[i] = v;
 155 #ifdef USE_FLOATS
 156         scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
 157 #else
 158 #define P 15
 159         scale_factor_shift[i] = 21 - P - (i / 3);
 160         scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
 161 #endif
 162     }
 163     for(i=0;i<128;i++) {
 164         v = i - 64;
 165         if (v <= -3)
 166             v = 0;
 167         else if (v < 0)
 168             v = 1;
 169         else if (v == 0)
 170             v = 2;
 171         else if (v < 3)
 172             v = 3;
 173         else
 174             v = 4;
 175         scale_diff_table[i] = v;
 176     }
 177
 178     for(i=0;i<17;i++) {
 179         v = ff_mpa_quant_bits[i];
 180         if (v < 0)
 181             v = -v;
 182         else
 183             v = v * 3;
 184         total_quant_bits[i] = 12 * v;
 185     }
 186
 187     avctx->coded_frame= avcodec_alloc_frame();
 188     avctx->coded_frame->key_frame= 1;
 189
 190     return 0;
 191 }
 192
 193 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
 194 static void idct32(int *out, int *tab)
 195 {
 196     int i, j;
 197     int *t, *t1, xr;
 198     const int *xp = costab32;
 199
 200     for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
 201
 202     t = tab + 30;
 203     t1 = tab + 2;
 204     do {
 205         t[0] += t[-4];
 206         t[1] += t[1 - 4];
 207         t -= 4;
 208     } while (t != t1);
 209
 210     t = tab + 28;
 211     t1 = tab + 4;
 212     do {
 213         t[0] += t[-8];
 214         t[1] += t[1-8];
 215         t[2] += t[2-8];
 216         t[3] += t[3-8];
 217         t -= 8;
 218     } while (t != t1);
 219
 220     t = tab;
 221     t1 = tab + 32;
 222     do {
 223         t[ 3] = -t[ 3];
 224         t[ 6] = -t[ 6];
 225
 226         t[11] = -t[11];
 227         t[12] = -t[12];
 228         t[13] = -t[13];
 229         t[15] = -t[15];
 230         t += 16;
 231     } while (t != t1);
 232
 233
 234     t = tab;
 235     t1 = tab + 8;
 236     do {
 237         int x1, x2, x3, x4;
 238
 239         x3 = MUL(t[16], FIX(SQRT2*0.5));
 240         x4 = t[0] - x3;
 241         x3 = t[0] + x3;
 242
 243         x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
 244         x1 = MUL((t[8] - x2), xp[0]);
 245         x2 = MUL((t[8] + x2), xp[1]);
 246
 247         t[ 0] = x3 + x1;
 248         t[ 8] = x4 - x2;
 249         t[16] = x4 + x2;
 250         t[24] = x3 - x1;
 251         t++;
 252     } while (t != t1);
 253
 254     xp += 2;
 255     t = tab;
 256     t1 = tab + 4;
 257     do {
 258         xr = MUL(t[28],xp[0]);
 259         t[28] = (t[0] - xr);
 260         t[0] = (t[0] + xr);
 261
 262         xr = MUL(t[4],xp[1]);
 263         t[ 4] = (t[24] - xr);
 264         t[24] = (t[24] + xr);
 265
 266         xr = MUL(t[20],xp[2]);
 267         t[20] = (t[8] - xr);
 268         t[ 8] = (t[8] + xr);
 269
 270         xr = MUL(t[12],xp[3]);
 271         t[12] = (t[16] - xr);
 272         t[16] = (t[16] + xr);
 273         t++;
 274     } while (t != t1);
 275     xp += 4;
 276
 277     for (i = 0; i < 4; i++) {
 278         xr = MUL(tab[30-i*4],xp[0]);
 279         tab[30-i*4] = (tab[i*4] - xr);
 280         tab[   i*4] = (tab[i*4] + xr);
 281
 282         xr = MUL(tab[ 2+i*4],xp[1]);
 283         tab[ 2+i*4] = (tab[28-i*4] - xr);
 284         tab[28-i*4] = (tab[28-i*4] + xr);
 285
 286         xr = MUL(tab[31-i*4],xp[0]);
 287         tab[31-i*4] = (tab[1+i*4] - xr);
 288         tab[ 1+i*4] = (tab[1+i*4] + xr);
 289
 290         xr = MUL(tab[ 3+i*4],xp[1]);
 291         tab[ 3+i*4] = (tab[29-i*4] - xr);
 292         tab[29-i*4] = (tab[29-i*4] + xr);
 293
 294         xp += 2;
 295     }
 296
 297     t = tab + 30;
 298     t1 = tab + 1;
 299     do {
 300         xr = MUL(t1[0], *xp);
 301         t1[0] = (t[0] - xr);
 302         t[0] = (t[0] + xr);
 303         t -= 2;
 304         t1 += 2;
 305         xp++;
 306     } while (t >= tab);
 307
 308     for(i=0;i<32;i++) {
 309         out[i] = tab[bitinv32[i]];
 310     }
 311 }
 312
 313 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
 314
 315 static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
 316 {
 317     short *p, *q;
 318     int sum, offset, i, j;
 319     int tmp[64];
 320     int tmp1[32];
 321     int *out;
 322
 323     //    print_pow1(samples, 1152);
 324
 325     offset = s->samples_offset[ch];
 326     out = &s->sb_samples[ch][0][0][0];
 327     for(j=0;j<36;j++) {
 328         /* 32 samples at once */
 329         for(i=0;i<32;i++) {
 330             s->samples_buf[ch][offset + (31 - i)] = samples[0];
 331             samples += incr;
 332         }
 333
 334         /* filter */
 335         p = s->samples_buf[ch] + offset;
 336         q = filter_bank;
 337         /* maxsum = 23169 */
 338         for(i=0;i<64;i++) {
 339             sum = p[0*64] * q[0*64];
 340             sum += p[1*64] * q[1*64];
 341             sum += p[2*64] * q[2*64];
 342             sum += p[3*64] * q[3*64];
 343             sum += p[4*64] * q[4*64];
 344             sum += p[5*64] * q[5*64];
 345             sum += p[6*64] * q[6*64];
 346             sum += p[7*64] * q[7*64];
 347             tmp[i] = sum;
 348             p++;
 349             q++;
 350         }
 351         tmp1[0] = tmp[16] >> WSHIFT;
 352         for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
 353         for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
 354
 355         idct32(out, tmp1);
 356
 357         /* advance of 32 samples */
 358         offset -= 32;
 359         out += 32;
 360         /* handle the wrap around */
 361         if (offset < 0) {
 362             memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
 363                     s->samples_buf[ch], (512 - 32) * 2);
 364             offset = SAMPLES_BUF_SIZE - 512;
 365         }
 366     }
 367     s->samples_offset[ch] = offset;
 368
 369     //    print_pow(s->sb_samples, 1152);
 370 }
 371
 372 static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
 373                                   unsigned char scale_factors[SBLIMIT][3],
 374                                   int sb_samples[3][12][SBLIMIT],
 375                                   int sblimit)
 376 {
 377     int *p, vmax, v, n, i, j, k, code;
 378     int index, d1, d2;
 379     unsigned char *sf = &scale_factors[0][0];
 380
 381     for(j=0;j<sblimit;j++) {
 382         for(i=0;i<3;i++) {
 383             /* find the max absolute value */
 384             p = &sb_samples[i][0][j];
 385             vmax = abs(*p);
 386             for(k=1;k<12;k++) {
 387                 p += SBLIMIT;
 388                 v = abs(*p);
 389                 if (v > vmax)
 390                     vmax = v;
 391             }
 392             /* compute the scale factor index using log 2 computations */
 393             if (vmax > 1) {
 394                 n = av_log2(vmax);
 395                 /* n is the position of the MSB of vmax. now
 396                    use at most 2 compares to find the index */
 397                 index = (21 - n) * 3 - 3;
 398                 if (index >= 0) {
 399                     while (vmax <= scale_factor_table[index+1])
 400                         index++;
 401                 } else {
 402                     index = 0; /* very unlikely case of overflow */
 403                 }
 404             } else {
 405                 index = 62; /* value 63 is not allowed */
 406             }
 407
 408 #if 0
 409             printf("%2d:%d in=%x %x %d\n",
 410                    j, i, vmax, scale_factor_table[index], index);
 411 #endif
 412             /* store the scale factor */
 413             assert(index >=0 && index <= 63);
 414             sf[i] = index;
 415         }
 416
 417         /* compute the transmission factor : look if the scale factors
 418            are close enough to each other */
 419         d1 = scale_diff_table[sf[0] - sf[1] + 64];
 420         d2 = scale_diff_table[sf[1] - sf[2] + 64];
 421
 422         /* handle the 25 cases */
 423         switch(d1 * 5 + d2) {
 424         case 0*5+0:
 425         case 0*5+4:
 426         case 3*5+4:
 427         case 4*5+0:
 428         case 4*5+4:
 429             code = 0;
 430             break;
 431         case 0*5+1:
 432         case 0*5+2:
 433         case 4*5+1:
 434         case 4*5+2:
 435             code = 3;
 436             sf[2] = sf[1];
 437             break;
 438         case 0*5+3:
 439         case 4*5+3:
 440             code = 3;
 441             sf[1] = sf[2];
 442             break;
 443         case 1*5+0:
 444         case 1*5+4:
 445         case 2*5+4:
 446             code = 1;
 447             sf[1] = sf[0];
 448             break;
 449         case 1*5+1:
 450         case 1*5+2:
 451         case 2*5+0:
 452         case 2*5+1:
 453         case 2*5+2:
 454             code = 2;
 455             sf[1] = sf[2] = sf[0];
 456             break;
 457         case 2*5+3:
 458         case 3*5+3:
 459             code = 2;
 460             sf[0] = sf[1] = sf[2];
 461             break;
 462         case 3*5+0:
 463         case 3*5+1:
 464         case 3*5+2:
 465             code = 2;
 466             sf[0] = sf[2] = sf[1];
 467             break;
 468         case 1*5+3:
 469             code = 2;
 470             if (sf[0] > sf[2])
 471               sf[0] = sf[2];
 472             sf[1] = sf[2] = sf[0];
 473             break;
 474         default:
 475             assert(0); //cannot happen
 476             code = 0;           /* kill warning */
 477         }
 478
 479 #if 0
 480         printf("%d: %2d %2d %2d %d %d -> %d\n", j,
 481                sf[0], sf[1], sf[2], d1, d2, code);
 482 #endif
 483         scale_code[j] = code;
 484         sf += 3;
 485     }
 486 }
 487
 488 /* The most important function : psycho acoustic module. In this
 489    encoder there is basically none, so this is the worst you can do,
 490    but also this is the simpler. */
 491 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
 492 {
 493     int i;
 494
 495     for(i=0;i<s->sblimit;i++) {
 496         smr[i] = (int)(fixed_smr[i] * 10);
 497     }
 498 }
 499
 500
 501 #define SB_NOTALLOCATED  0
 502 #define SB_ALLOCATED     1
 503 #define SB_NOMORE        2
 504
 505 /* Try to maximize the smr while using a number of bits inferior to
 506    the frame size. I tried to make the code simpler, faster and
 507    smaller than other encoders :-) */
 508 static void compute_bit_allocation(MpegAudioContext *s,
 509                                    short smr1[MPA_MAX_CHANNELS][SBLIMIT],
 510                                    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
 511                                    int *padding)
 512 {
 513     int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
 514     int incr;
 515     short smr[MPA_MAX_CHANNELS][SBLIMIT];
 516     unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
 517     const unsigned char *alloc;
 518
 519     memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
 520     memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
 521     memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
 522
 523     /* compute frame size and padding */
 524     max_frame_size = s->frame_size;
 525     s->frame_frac += s->frame_frac_incr;
 526     if (s->frame_frac >= 65536) {
 527         s->frame_frac -= 65536;
 528         s->do_padding = 1;
 529         max_frame_size += 8;
 530     } else {
 531         s->do_padding = 0;
 532     }
 533
 534     /* compute the header + bit alloc size */
 535     current_frame_size = 32;
 536     alloc = s->alloc_table;
 537     for(i=0;i<s->sblimit;i++) {
 538         incr = alloc[0];
 539         current_frame_size += incr * s->nb_channels;
 540         alloc += 1 << incr;
 541     }
 542     for(;;) {
 543         /* look for the subband with the largest signal to mask ratio */
 544         max_sb = -1;
 545         max_ch = -1;
 546         max_smr = INT_MIN;
 547         for(ch=0;ch<s->nb_channels;ch++) {
 548             for(i=0;i<s->sblimit;i++) {
 549                 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
 550                     max_smr = smr[ch][i];
 551                     max_sb = i;
 552                     max_ch = ch;
 553                 }
 554             }
 555         }
 556 #if 0
 557         printf("current=%d max=%d max_sb=%d alloc=%d\n",
 558                current_frame_size, max_frame_size, max_sb,
 559                bit_alloc[max_sb]);
 560 #endif
 561         if (max_sb < 0)
 562             break;
 563
 564         /* find alloc table entry (XXX: not optimal, should use
 565            pointer table) */
 566         alloc = s->alloc_table;
 567         for(i=0;i<max_sb;i++) {
 568             alloc += 1 << alloc[0];
 569         }
 570
 571         if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
 572             /* nothing was coded for this band: add the necessary bits */
 573             incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
 574             incr += total_quant_bits[alloc[1]];
 575         } else {
 576             /* increments bit allocation */
 577             b = bit_alloc[max_ch][max_sb];
 578             incr = total_quant_bits[alloc[b + 1]] -
 579                 total_quant_bits[alloc[b]];
 580         }
 581
 582         if (current_frame_size + incr <= max_frame_size) {
 583             /* can increase size */
 584             b = ++bit_alloc[max_ch][max_sb];
 585             current_frame_size += incr;
 586             /* decrease smr by the resolution we added */
 587             smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
 588             /* max allocation size reached ? */
 589             if (b == ((1 << alloc[0]) - 1))
 590                 subband_status[max_ch][max_sb] = SB_NOMORE;
 591             else
 592                 subband_status[max_ch][max_sb] = SB_ALLOCATED;
 593         } else {
 594             /* cannot increase the size of this subband */
 595             subband_status[max_ch][max_sb] = SB_NOMORE;
 596         }
 597     }
 598     *padding = max_frame_size - current_frame_size;
 599     assert(*padding >= 0);
 600
 601 #if 0
 602     for(i=0;i<s->sblimit;i++) {
 603         printf("%d ", bit_alloc[i]);
 604     }
 605     printf("\n");
 606 #endif
 607 }
 608
 609 /*
 610  * Output the mpeg audio layer 2 frame. Note how the code is small
 611  * compared to other encoders :-)
 612  */
 613 static void encode_frame(MpegAudioContext *s,
 614                          unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
 615                          int padding)
 616 {
 617     int i, j, k, l, bit_alloc_bits, b, ch;
 618     unsigned char *sf;
 619     int q[3];
 620     PutBitContext *p = &s->pb;
 621
 622     /* header */
 623
 624     put_bits(p, 12, 0xfff);
 625     put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
 626     put_bits(p, 2, 4-2);  /* layer 2 */
 627     put_bits(p, 1, 1); /* no error protection */
 628     put_bits(p, 4, s->bitrate_index);
 629     put_bits(p, 2, s->freq_index);
 630     put_bits(p, 1, s->do_padding); /* use padding */
 631     put_bits(p, 1, 0);             /* private_bit */
 632     put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
 633     put_bits(p, 2, 0); /* mode_ext */
 634     put_bits(p, 1, 0); /* no copyright */
 635     put_bits(p, 1, 1); /* original */
 636     put_bits(p, 2, 0); /* no emphasis */
 637
 638     /* bit allocation */
 639     j = 0;
 640     for(i=0;i<s->sblimit;i++) {
 641         bit_alloc_bits = s->alloc_table[j];
 642         for(ch=0;ch<s->nb_channels;ch++) {
 643             put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
 644         }
 645         j += 1 << bit_alloc_bits;
 646     }
 647
 648     /* scale codes */
 649     for(i=0;i<s->sblimit;i++) {
 650         for(ch=0;ch<s->nb_channels;ch++) {
 651             if (bit_alloc[ch][i])
 652                 put_bits(p, 2, s->scale_code[ch][i]);
 653         }
 654     }
 655
 656     /* scale factors */
 657     for(i=0;i<s->sblimit;i++) {
 658         for(ch=0;ch<s->nb_channels;ch++) {
 659             if (bit_alloc[ch][i]) {
 660                 sf = &s->scale_factors[ch][i][0];
 661                 switch(s->scale_code[ch][i]) {
 662                 case 0:
 663                     put_bits(p, 6, sf[0]);
 664                     put_bits(p, 6, sf[1]);
 665                     put_bits(p, 6, sf[2]);
 666                     break;
 667                 case 3:
 668                 case 1:
 669                     put_bits(p, 6, sf[0]);
 670                     put_bits(p, 6, sf[2]);
 671                     break;
 672                 case 2:
 673                     put_bits(p, 6, sf[0]);
 674                     break;
 675                 }
 676             }
 677         }
 678     }
 679
 680     /* quantization & write sub band samples */
 681
 682     for(k=0;k<3;k++) {
 683         for(l=0;l<12;l+=3) {
 684             j = 0;
 685             for(i=0;i<s->sblimit;i++) {
 686                 bit_alloc_bits = s->alloc_table[j];
 687                 for(ch=0;ch<s->nb_channels;ch++) {
 688                     b = bit_alloc[ch][i];
 689                     if (b) {
 690                         int qindex, steps, m, sample, bits;
 691                         /* we encode 3 sub band samples of the same sub band at a time */
 692                         qindex = s->alloc_table[j+b];
 693                         steps = ff_mpa_quant_steps[qindex];
 694                         for(m=0;m<3;m++) {
 695                             sample = s->sb_samples[ch][k][l + m][i];
 696                             /* divide by scale factor */
 697 #ifdef USE_FLOATS
 698                             {
 699                                 float a;
 700                                 a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
 701                                 q[m] = (int)((a + 1.0) * steps * 0.5);
 702                             }
 703 #else
 704                             {
 705                                 int q1, e, shift, mult;
 706                                 e = s->scale_factors[ch][i][k];
 707                                 shift = scale_factor_shift[e];
 708                                 mult = scale_factor_mult[e];
 709
 710                                 /* normalize to P bits */
 711                                 if (shift < 0)
 712                                     q1 = sample << (-shift);
 713                                 else
 714                                     q1 = sample >> shift;
 715                                 q1 = (q1 * mult) >> P;
 716                                 q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
 717                             }
 718 #endif
 719                             if (q[m] >= steps)
 720                                 q[m] = steps - 1;
 721                             assert(q[m] >= 0 && q[m] < steps);
 722                         }
 723                         bits = ff_mpa_quant_bits[qindex];
 724                         if (bits < 0) {
 725                             /* group the 3 values to save bits */
 726                             put_bits(p, -bits,
 727                                      q[0] + steps * (q[1] + steps * q[2]));
 728 #if 0
 729                             printf("%d: gr1 %d\n",
 730                                    i, q[0] + steps * (q[1] + steps * q[2]));
 731 #endif
 732                         } else {
 733 #if 0
 734                             printf("%d: gr3 %d %d %d\n",
 735                                    i, q[0], q[1], q[2]);
 736 #endif
 737                             put_bits(p, bits, q[0]);
 738                             put_bits(p, bits, q[1]);
 739                             put_bits(p, bits, q[2]);
 740                         }
 741                     }
 742                 }
 743                 /* next subband in alloc table */
 744                 j += 1 << bit_alloc_bits;
 745             }
 746         }
 747     }
 748
 749     /* padding */
 750     for(i=0;i<padding;i++)
 751         put_bits(p, 1, 0);
 752
 753     /* flush */
 754     flush_put_bits(p);
 755 }
 756
 757 static int MPA_encode_frame(AVCodecContext *avctx,
 758                             unsigned char *frame, int buf_size, void *data)
 759 {
 760     MpegAudioContext *s = avctx->priv_data;
 761     short *samples = data;
 762     short smr[MPA_MAX_CHANNELS][SBLIMIT];
 763     unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
 764     int padding, i;
 765
 766     for(i=0;i<s->nb_channels;i++) {
 767         filter(s, i, samples + i, s->nb_channels);
 768     }
 769
 770     for(i=0;i<s->nb_channels;i++) {
 771         compute_scale_factors(s->scale_code[i], s->scale_factors[i],
 772                               s->sb_samples[i], s->sblimit);
 773     }
 774     for(i=0;i<s->nb_channels;i++) {
 775         psycho_acoustic_model(s, smr[i]);
 776     }
 777     compute_bit_allocation(s, smr, bit_alloc, &padding);
 778
 779     init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
 780
 781     encode_frame(s, bit_alloc, padding);
 782
 783     s->nb_samples += MPA_FRAME_SIZE;
 784     return put_bits_ptr(&s->pb) - s->pb.buf;
 785 }
 786
 787 static av_cold int MPA_encode_close(AVCodecContext *avctx)
 788 {
 789     av_freep(&avctx->coded_frame);
 790     return 0;
 791 }
 792
 793 AVCodec mp2_encoder = {
 794     "mp2",
 795     CODEC_TYPE_AUDIO,
 796     CODEC_ID_MP2,
 797     sizeof(MpegAudioContext),
 798     MPA_encode_init,
 799     MPA_encode_frame,
 800     MPA_encode_close,
 801     NULL,
 802     .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
 803     .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
 804 };
 805
 806 #undef FIX