summaryrefslogtreecommitdiffstats
path: root/libavcodec/mpegaudioenc.c
blob: c061d7f5cf96c0ca782499fc8fe0e13a89dadadc (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
/*
 * The simplest mpeg audio layer 2 encoder
 * Copyright (c) 2000, 2001 Fabrice Bellard.
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file mpegaudio.c
 * The simplest mpeg audio layer 2 encoder.
 */

#include "avcodec.h"
#include "bitstream.h"
#include "mpegaudio.h"

/* currently, cannot change these constants (need to modify
   quantization stage) */
#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)

#define SAMPLES_BUF_SIZE 4096

typedef struct MpegAudioContext {
    PutBitContext pb;
    int nb_channels;
    int freq, bit_rate;
    int lsf;           /* 1 if mpeg2 low bitrate selected */
    int bitrate_index; /* bit rate */
    int freq_index;
    int frame_size; /* frame size, in bits, without padding */
    int64_t nb_samples; /* total number of samples encoded */
    /* padding computation */
    int frame_frac, frame_frac_incr, do_padding;
    short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
    int samples_offset[MPA_MAX_CHANNELS];       /* offset in samples_buf */
    int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
    unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
    /* code to group 3 scale factors */
    unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
    int sblimit; /* number of used subbands */
    const unsigned char *alloc_table;
} MpegAudioContext;

/* define it to use floats in quantization (I don't like floats !) */
//#define USE_FLOATS

#include "mpegaudiodata.h"
#include "mpegaudiotab.h"

static av_cold int MPA_encode_init(AVCodecContext *avctx)
{
    MpegAudioContext *s = avctx->priv_data;
    int freq = avctx->sample_rate;
    int bitrate = avctx->bit_rate;
    int channels = avctx->channels;
    int i, v, table;
    float a;

    if (channels <= 0 || channels > 2){
        av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
        return -1;
    }
    bitrate = bitrate / 1000;
    s->nb_channels = channels;
    s->freq = freq;
    s->bit_rate = bitrate * 1000;
    avctx->frame_size = MPA_FRAME_SIZE;

    /* encoding freq */
    s->lsf = 0;
    for(i=0;i<3;i++) {
        if (ff_mpa_freq_tab[i] == freq)
            break;
        if ((ff_mpa_freq_tab[i] / 2) == freq) {
            s->lsf = 1;
            break;
        }
    }
    if (i == 3){
        av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
        return -1;
    }
    s->freq_index = i;

    /* encoding bitrate & frequency */
    for(i=0;i<15;i++) {
        if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
            break;
    }
    if (i == 15){
        av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
        return -1;
    }
    s->bitrate_index = i;

    /* compute total header size & pad bit */

    a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
    s->frame_size = ((int)a) * 8;

    /* frame fractional size to compute padding */
    s->frame_frac = 0;
    s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);

    /* select the right allocation table */
    table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);

    /* number of used subbands */
    s->sblimit = ff_mpa_sblimit_table[table];
    s->alloc_table = ff_mpa_alloc_tables[table];

#ifdef DEBUG
    av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
           bitrate, freq, s->frame_size, table, s->frame_frac_incr);
#endif

    for(i=0;i<s->nb_channels;i++)
        s->samples_offset[i] = 0;

    for(i=0;i<257;i++) {
        int v;
        v = ff_mpa_enwindow[i];
#if WFRAC_BITS != 16
        v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
#endif
        filter_bank[i] = v;
        if ((i & 63) != 0)
            v = -v;
        if (i != 0)
            filter_bank[512 - i] = v;
    }

    for(i=0;i<64;i++) {
        v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
        if (v <= 0)
            v = 1;
        scale_factor_table[i] = v;
#ifdef USE_FLOATS
        scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
#else
#define P 15
        scale_factor_shift[i] = 21 - P - (i / 3);
        scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
#endif
    }
    for(i=0;i<128;i++) {
        v = i - 64;
        if (v <= -3)
            v = 0;
        else if (v < 0)
            v = 1;
        else if (v == 0)
            v = 2;
        else if (v < 3)
            v = 3;
        else
            v = 4;
        scale_diff_table[i] = v;
    }

    for(i=0;i<17;i++) {
        v = ff_mpa_quant_bits[i];
        if (v < 0)
            v = -v;
        else
            v = v * 3;
        total_quant_bits[i] = 12 * v;
    }

    avctx->coded_frame= avcodec_alloc_frame();
    avctx->coded_frame->key_frame= 1;

    return 0;
}

/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
static void idct32(int *out, int *tab)
{
    int i, j;
    int *t, *t1, xr;
    const int *xp = costab32;

    for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];

    t = tab + 30;
    t1 = tab + 2;
    do {
        t[0] += t[-4];
        t[1] += t[1 - 4];
        t -= 4;
    } while (t != t1);

    t = tab + 28;
    t1 = tab + 4;
    do {
        t[0] += t[-8];
        t[1] += t[1-8];
        t[2] += t[2-8];
        t[3] += t[3-8];
        t -= 8;
    } while (t != t1);

    t = tab;
    t1 = tab + 32;
    do {
        t[ 3] = -t[ 3];
        t[ 6] = -t[ 6];

        t[11] = -t[11];
        t[12] = -t[12];
        t[13] = -t[13];
        t[15] = -t[15];
        t += 16;
    } while (t != t1);


    t = tab;
    t1 = tab + 8;
    do {
        int x1, x2, x3, x4;

        x3 = MUL(t[16], FIX(SQRT2*0.5));
        x4 = t[0] - x3;
        x3 = t[0] + x3;

        x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
        x1 = MUL((t[8] - x2), xp[0]);
        x2 = MUL((t[8] + x2), xp[1]);

        t[ 0] = x3 + x1;
        t[ 8] = x4 - x2;
        t[16] = x4 + x2;
        t[24] = x3 - x1;
        t++;
    } while (t != t1);

    xp += 2;
    t = tab;
    t1 = tab + 4;
    do {
        xr = MUL(t[28],xp[0]);
        t[28] = (t[0] - xr);
        t[0] = (t[0] + xr);

        xr = MUL(t[4],xp[1]);
        t[ 4] = (t[24] - xr);
        t[24] = (t[24] + xr);

        xr = MUL(t[20],xp[2]);
        t[20] = (t[8] - xr);
        t[ 8] = (t[8] + xr);

        xr = MUL(t[12],xp[3]);
        t[12] = (t[16] - xr);
        t[16] = (t[16] + xr);
        t++;
    } while (t != t1);
    xp += 4;

    for (i = 0; i < 4; i++) {
        xr = MUL(tab[30-i*4],xp[0]);
        tab[30-i*4] = (tab[i*4] - xr);
        tab[   i*4] = (tab[i*4] + xr);

        xr = MUL(tab[ 2+i*4],xp[1]);
        tab[ 2+i*4] = (tab[28-i*4] - xr);
        tab[28-i*4] = (tab[28-i*4] + xr);

        xr = MUL(tab[31-i*4],xp[0]);
        tab[31-i*4] = (tab[1+i*4] - xr);
        tab[ 1+i*4] = (tab[1+i*4] + xr);

        xr = MUL(tab[ 3+i*4],xp[1]);
        tab[ 3+i*4] = (tab[29-i*4] - xr);
        tab[29-i*4] = (tab[29-i*4] + xr);

        xp += 2;
    }

    t = tab + 30;
    t1 = tab + 1;
    do {
        xr = MUL(t1[0], *xp);
        t1[0] = (t[0] - xr);
        t[0] = (t[0] + xr);
        t -= 2;
        t1 += 2;
        xp++;
    } while (t >= tab);

    for(i=0;i<32;i++) {
        out[i] = tab[bitinv32[i]];
    }
}

#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)

static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
{
    short *p, *q;
    int sum, offset, i, j;
    int tmp[64];
    int tmp1[32];
    int *out;

    //    print_pow1(samples, 1152);

    offset = s->samples_offset[ch];
    out = &s->sb_samples[ch][0][0][0];
    for(j=0;j<36;j++) {
        /* 32 samples at once */
        for(i=0;i<32;i++) {
            s->samples_buf[ch][offset + (31 - i)] = samples[0];
            samples += incr;
        }

        /* filter */
        p = s->samples_buf[ch] + offset;
        q = filter_bank;
        /* maxsum = 23169 */
        for(i=0;i<64;i++) {
            sum = p[0*64] * q[0*64];
            sum += p[1*64] * q[1*64];
            sum += p[2*64] * q[2*64];
            sum += p[3*64] * q[3*64];
            sum += p[4*64] * q[4*64];
            sum += p[5*64] * q[5*64];
            sum += p[6*64] * q[6*64];
            sum += p[7*64] * q[7*64];
            tmp[i] = sum;
            p++;
            q++;
        }
        tmp1[0] = tmp[16] >> WSHIFT;
        for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
        for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;

        idct32(out, tmp1);

        /* advance of 32 samples */
        offset -= 32;
        out += 32;
        /* handle the wrap around */
        if (offset < 0) {
            memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
                    s->samples_buf[ch], (512 - 32) * 2);
            offset = SAMPLES_BUF_SIZE - 512;
        }
    }
    s->samples_offset[ch] = offset;

    //    print_pow(s->sb_samples, 1152);
}

static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
                                  unsigned char scale_factors[SBLIMIT][3],
                                  int sb_samples[3][12][SBLIMIT],
                                  int sblimit)
{
    int *p, vmax, v, n, i, j, k, code;
    int index, d1, d2;
    unsigned char *sf = &scale_factors[0][0];

    for(j=0;j<sblimit;j++) {
        for(i=0;i<3;i++) {
            /* find the max absolute value */
            p = &sb_samples[i][0][j];
            vmax = abs(*p);
            for(k=1;k<12;k++) {
                p += SBLIMIT;
                v = abs(*p);
                if (v > vmax)
                    vmax = v;
            }
            /* compute the scale factor index using log 2 computations */
            if (vmax > 1) {
                n = av_log2(vmax);
                /* n is the position of the MSB of vmax. now
                   use at most 2 compares to find the index */
                index = (21 - n) * 3 - 3;
                if (index >= 0) {
                    while (vmax <= scale_factor_table[index+1])
                        index++;
                } else {
                    index = 0; /* very unlikely case of overflow */
                }
            } else {
                index = 62; /* value 63 is not allowed */
            }

#if 0
            printf("%2d:%d in=%x %x %d\n",
                   j, i, vmax, scale_factor_table[index], index);
#endif
            /* store the scale factor */
            assert(index >=0 && index <= 63);
            sf[i] = index;
        }

        /* compute the transmission factor : look if the scale factors
           are close enough to each other */
        d1 = scale_diff_table[sf[0] - sf[1] + 64];
        d2 = scale_diff_table[sf[1] - sf[2] + 64];

        /* handle the 25 cases */
        switch(d1 * 5 + d2) {
        case 0*5+0:
        case 0*5+4:
        case 3*5+4:
        case 4*5+0:
        case 4*5+4:
            code = 0;
            break;
        case 0*5+1:
        case 0*5+2:
        case 4*5+1:
        case 4*5+2:
            code = 3;
            sf[2] = sf[1];
            break;
        case 0*5+3:
        case 4*5+3:
            code = 3;
            sf[1] = sf[2];
            break;
        case 1*5+0:
        case 1*5+4:
        case 2*5+4:
            code = 1;
            sf[1] = sf[0];
            break;
        case 1*5+1:
        case 1*5+2:
        case 2*5+0:
        case 2*5+1:
        case 2*5+2:
            code = 2;
            sf[1] = sf[2] = sf[0];
            break;
        case 2*5+3:
        case 3*5+3:
            code = 2;
            sf[0] = sf[1] = sf[2];
            break;
        case 3*5+0:
        case 3*5+1:
        case 3*5+2:
            code = 2;
            sf[0] = sf[2] = sf[1];
            break;
        case 1*5+3:
            code = 2;
            if (sf[0] > sf[2])
              sf[0] = sf[2];
            sf[1] = sf[2] = sf[0];
            break;
        default:
            assert(0); //cannot happen
            code = 0;           /* kill warning */
        }

#if 0
        printf("%d: %2d %2d %2d %d %d -> %d\n", j,
               sf[0], sf[1], sf[2], d1, d2, code);
#endif
        scale_code[j] = code;
        sf += 3;
    }
}

/* The most important function : psycho acoustic module. In this
   encoder there is basically none, so this is the worst you can do,
   but also this is the simpler. */
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
{
    int i;

    for(i=0;i<s->sblimit;i++) {
        smr[i] = (int)(fixed_smr[i] * 10);
    }
}


#define SB_NOTALLOCATED  0
#define SB_ALLOCATED     1
#define SB_NOMORE        2

/* Try to maximize the smr while using a number of bits inferior to
   the frame size. I tried to make the code simpler, faster and
   smaller than other encoders :-) */
static void compute_bit_allocation(MpegAudioContext *s,
                                   short smr1[MPA_MAX_CHANNELS][SBLIMIT],
                                   unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
                                   int *padding)
{
    int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
    int incr;
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
    unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
    const unsigned char *alloc;

    memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
    memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
    memset(bit_alloc, 0, s->nb_channels * SBLIMIT);

    /* compute frame size and padding */
    max_frame_size = s->frame_size;
    s->frame_frac += s->frame_frac_incr;
    if (s->frame_frac >= 65536) {
        s->frame_frac -= 65536;
        s->do_padding = 1;
        max_frame_size += 8;
    } else {
        s->do_padding = 0;
    }

    /* compute the header + bit alloc size */
    current_frame_size = 32;
    alloc = s->alloc_table;
    for(i=0;i<s->sblimit;i++) {
        incr = alloc[0];
        current_frame_size += incr * s->nb_channels;
        alloc += 1 << incr;
    }
    for(;;) {
        /* look for the subband with the largest signal to mask ratio */
        max_sb = -1;
        max_ch = -1;
        max_smr = INT_MIN;
        for(ch=0;ch<s->nb_channels;ch++) {
            for(i=0;i<s->sblimit;i++) {
                if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
                    max_smr = smr[ch][i];
                    max_sb = i;
                    max_ch = ch;
                }
            }
        }
#if 0
        printf("current=%d max=%d max_sb=%d alloc=%d\n",
               current_frame_size, max_frame_size, max_sb,
               bit_alloc[max_sb]);
#endif
        if (max_sb < 0)
            break;

        /* find alloc table entry (XXX: not optimal, should use
           pointer table) */
        alloc = s->alloc_table;
        for(i=0;i<max_sb;i++) {
            alloc += 1 << alloc[0];
        }

        if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
            /* nothing was coded for this band: add the necessary bits */
            incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
            incr += total_quant_bits[alloc[1]];
        } else {
            /* increments bit allocation */
            b = bit_alloc[max_ch][max_sb];
            incr = total_quant_bits[alloc[b + 1]] -
                total_quant_bits[alloc[b]];
        }

        if (current_frame_size + incr <= max_frame_size) {
            /* can increase size */
            b = ++bit_alloc[max_ch][max_sb];
            current_frame_size += incr;
            /* decrease smr by the resolution we added */
            smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
            /* max allocation size reached ? */
            if (b == ((1 << alloc[0]) - 1))
                subband_status[max_ch][max_sb] = SB_NOMORE;
            else
                subband_status[max_ch][max_sb] = SB_ALLOCATED;
        } else {
            /* cannot increase the size of this subband */
            subband_status[max_ch][max_sb] = SB_NOMORE;
        }
    }
    *padding = max_frame_size - current_frame_size;
    assert(*padding >= 0);

#if 0
    for(i=0;i<s->sblimit;i++) {
        printf("%d ", bit_alloc[i]);
    }
    printf("\n");
#endif
}

/*
 * Output the mpeg audio layer 2 frame. Note how the code is small
 * compared to other encoders :-)
 */
static void encode_frame(MpegAudioContext *s,
                         unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
                         int padding)
{
    int i, j, k, l, bit_alloc_bits, b, ch;
    unsigned char *sf;
    int q[3];
    PutBitContext *p = &s->pb;

    /* header */

    put_bits(p, 12, 0xfff);
    put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
    put_bits(p, 2, 4-2);  /* layer 2 */
    put_bits(p, 1, 1); /* no error protection */
    put_bits(p, 4, s->bitrate_index);
    put_bits(p, 2, s->freq_index);
    put_bits(p, 1, s->do_padding); /* use padding */
    put_bits(p, 1, 0);             /* private_bit */
    put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
    put_bits(p, 2, 0); /* mode_ext */
    put_bits(p, 1, 0); /* no copyright */
    put_bits(p, 1, 1); /* original */
    put_bits(p, 2, 0); /* no emphasis */

    /* bit allocation */
    j = 0;
    for(i=0;i<s->sblimit;i++) {
        bit_alloc_bits = s->alloc_table[j];
        for(ch=0;ch<s->nb_channels;ch++) {
            put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
        }
        j += 1 << bit_alloc_bits;
    }

    /* scale codes */
    for(i=0;i<s->sblimit;i++) {
        for(ch=0;ch<s->nb_channels;ch++) {
            if (bit_alloc[ch][i])
                put_bits(p, 2, s->scale_code[ch][i]);
        }
    }

    /* scale factors */
    for(i=0;i<s->sblimit;i++) {
        for(ch=0;ch<s->nb_channels;ch++) {
            if (bit_alloc[ch][i]) {
                sf = &s->scale_factors[ch][i][0];
                switch(s->scale_code[ch][i]) {
                case 0:
                    put_bits(p, 6, sf[0]);
                    put_bits(p, 6, sf[1]);
                    put_bits(p, 6, sf[2]);
                    break;
                case 3:
                case 1:
                    put_bits(p, 6, sf[0]);
                    put_bits(p, 6, sf[2]);
                    break;
                case 2:
                    put_bits(p, 6, sf[0]);
                    break;
                }
            }
        }
    }

    /* quantization & write sub band samples */

    for(k=0;k<3;k++) {
        for(l=0;l<12;l+=3) {
            j = 0;
            for(i=0;i<s->sblimit;i++) {
                bit_alloc_bits = s->alloc_table[j];
                for(ch=0;ch<s->nb_channels;ch++) {
                    b = bit_alloc[ch][i];
                    if (b) {
                        int qindex, steps, m, sample, bits;
                        /* we encode 3 sub band samples of the same sub band at a time */
                        qindex = s->alloc_table[j+b];
                        steps = ff_mpa_quant_steps[qindex];
                        for(m=0;m<3;m++) {
                            sample = s->sb_samples[ch][k][l + m][i];
                            /* divide by scale factor */
#ifdef USE_FLOATS
                            {
                                float a;
                                a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
                                q[m] = (int)((a + 1.0) * steps * 0.5);
                            }
#else
                            {
                                int q1, e, shift, mult;
                                e = s->scale_factors[ch][i][k];
                                shift = scale_factor_shift[e];
                                mult = scale_factor_mult[e];

                                /* normalize to P bits */
                                if (shift < 0)
                                    q1 = sample << (-shift);
                                else
                                    q1 = sample >> shift;
                                q1 = (q1 * mult) >> P;
                                q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
                            }
#endif
                            if (q[m] >= steps)
                                q[m] = steps - 1;
                            assert(q[m] >= 0 && q[m] < steps);
                        }
                        bits = ff_mpa_quant_bits[qindex];
                        if (bits < 0) {
                            /* group the 3 values to save bits */
                            put_bits(p, -bits,
                                     q[0] + steps * (q[1] + steps * q[2]));
#if 0
                            printf("%d: gr1 %d\n",
                                   i, q[0] + steps * (q[1] + steps * q[2]));
#endif
                        } else {
#if 0
                            printf("%d: gr3 %d %d %d\n",
                                   i, q[0], q[1], q[2]);
#endif
                            put_bits(p, bits, q[0]);
                            put_bits(p, bits, q[1]);
                            put_bits(p, bits, q[2]);
                        }
                    }
                }
                /* next subband in alloc table */
                j += 1 << bit_alloc_bits;
            }
        }
    }

    /* padding */
    for(i=0;i<padding;i++)
        put_bits(p, 1, 0);

    /* flush */
    flush_put_bits(p);
}

static int MPA_encode_frame(AVCodecContext *avctx,
                            unsigned char *frame, int buf_size, void *data)
{
    MpegAudioContext *s = avctx->priv_data;
    short *samples = data;
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
    int padding, i;

    for(i=0;i<s->nb_channels;i++) {
        filter(s, i, samples + i, s->nb_channels);
    }

    for(i=0;i<s->nb_channels;i++) {
        compute_scale_factors(s->scale_code[i], s->scale_factors[i],
                              s->sb_samples[i], s->sblimit);
    }
    for(i=0;i<s->nb_channels;i++) {
        psycho_acoustic_model(s, smr[i]);
    }
    compute_bit_allocation(s, smr, bit_alloc, &padding);

    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);

    encode_frame(s, bit_alloc, padding);

    s->nb_samples += MPA_FRAME_SIZE;
    return pbBufPtr(&s->pb) - s->pb.buf;
}

static av_cold int MPA_encode_close(AVCodecContext *avctx)
{
    av_freep(&avctx->coded_frame);
    return 0;
}

AVCodec mp2_encoder = {
    "mp2",
    CODEC_TYPE_AUDIO,
    CODEC_ID_MP2,
    sizeof(MpegAudioContext),
    MPA_encode_init,
    MPA_encode_frame,
    MPA_encode_close,
    NULL,
    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
    .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
};

#undef FIX
OpenPOWER on IntegriCloud