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1
/*
2
 * The simplest mpeg audio layer 2 encoder
3
 * Copyright (c) 2000, 2001 Fabrice Bellard
4
 *
5
 * This file is part of FFmpeg.
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 *
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 * 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.
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 *
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 * 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.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
21

    
22
/**
23
 * @file libavcodec/mpegaudio.c
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 * The simplest mpeg audio layer 2 encoder.
25
 */
26

    
27
#include "avcodec.h"
28
#include "put_bits.h"
29

    
30
#undef  CONFIG_MPEGAUDIO_HP
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#define CONFIG_MPEGAUDIO_HP 0
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#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 */
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    int64_t nb_samples; /* total number of samples encoded */
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    /* 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 */
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    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 pbBufPtr(&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