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1 de6d9b64 Fabrice Bellard
/*
2
 * The simplest mpeg audio layer 2 encoder
3 ff4ec49e Fabrice Bellard
 * Copyright (c) 2000, 2001 Fabrice Bellard.
4 de6d9b64 Fabrice Bellard
 *
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 * This library is free software; you can redistribute it and/or
6
 * modify it under the terms of the GNU Lesser General Public
7
 * License as published by the Free Software Foundation; either
8
 * version 2 of the License, or (at your option) any later version.
9 de6d9b64 Fabrice Bellard
 *
10 ff4ec49e Fabrice Bellard
 * This library is distributed in the hope that it will be useful,
11 de6d9b64 Fabrice Bellard
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 ff4ec49e Fabrice Bellard
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
13
 * Lesser General Public License for more details.
14 de6d9b64 Fabrice Bellard
 *
15 ff4ec49e Fabrice Bellard
 * You should have received a copy of the GNU Lesser General Public
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 * License along with this library; if not, write to the Free Software
17
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
18 de6d9b64 Fabrice Bellard
 */
19
#include "avcodec.h"
20
#include "mpegaudio.h"
21
22 afa982fd Fabrice Bellard
/* currently, cannot change these constants (need to modify
23
   quantization stage) */
24
#define FRAC_BITS 15
25
#define WFRAC_BITS  14
26
#define MUL(a,b) (((INT64)(a) * (INT64)(b)) >> FRAC_BITS)
27
#define FIX(a)   ((int)((a) * (1 << FRAC_BITS)))
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29
#define SAMPLES_BUF_SIZE 4096
30
31
typedef struct MpegAudioContext {
32
    PutBitContext pb;
33
    int nb_channels;
34
    int freq, bit_rate;
35
    int lsf;           /* 1 if mpeg2 low bitrate selected */
36
    int bitrate_index; /* bit rate */
37
    int freq_index;
38
    int frame_size; /* frame size, in bits, without padding */
39
    INT64 nb_samples; /* total number of samples encoded */
40
    /* padding computation */
41
    int frame_frac, frame_frac_incr, do_padding;
42
    short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
43
    int samples_offset[MPA_MAX_CHANNELS];       /* offset in samples_buf */
44
    int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
45
    unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
46
    /* code to group 3 scale factors */
47
    unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];       
48
    int sblimit; /* number of used subbands */
49
    const unsigned char *alloc_table;
50
} MpegAudioContext;
51
52 de6d9b64 Fabrice Bellard
/* define it to use floats in quantization (I don't like floats !) */
53
//#define USE_FLOATS
54
55
#include "mpegaudiotab.h"
56
57
int MPA_encode_init(AVCodecContext *avctx)
58
{
59
    MpegAudioContext *s = avctx->priv_data;
60
    int freq = avctx->sample_rate;
61
    int bitrate = avctx->bit_rate;
62
    int channels = avctx->channels;
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    int i, v, table;
64 de6d9b64 Fabrice Bellard
    float a;
65
66
    if (channels > 2)
67
        return -1;
68
    bitrate = bitrate / 1000;
69
    s->nb_channels = channels;
70
    s->freq = freq;
71
    s->bit_rate = bitrate * 1000;
72
    avctx->frame_size = MPA_FRAME_SIZE;
73
    avctx->key_frame = 1; /* always key frame */
74
75
    /* encoding freq */
76
    s->lsf = 0;
77
    for(i=0;i<3;i++) {
78 2456e28d Fabrice Bellard
        if (mpa_freq_tab[i] == freq) 
79 de6d9b64 Fabrice Bellard
            break;
80 2456e28d Fabrice Bellard
        if ((mpa_freq_tab[i] / 2) == freq) {
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            s->lsf = 1;
82
            break;
83
        }
84
    }
85
    if (i == 3)
86
        return -1;
87
    s->freq_index = i;
88
89
    /* encoding bitrate & frequency */
90
    for(i=0;i<15;i++) {
91 2456e28d Fabrice Bellard
        if (mpa_bitrate_tab[s->lsf][1][i] == bitrate) 
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            break;
93
    }
94
    if (i == 15)
95
        return -1;
96
    s->bitrate_index = i;
97
98
    /* compute total header size & pad bit */
99
    
100
    a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
101
    s->frame_size = ((int)a) * 8;
102
103
    /* frame fractional size to compute padding */
104
    s->frame_frac = 0;
105
    s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
106
    
107
    /* select the right allocation table */
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    table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
109
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    /* number of used subbands */
111
    s->sblimit = sblimit_table[table];
112
    s->alloc_table = alloc_tables[table];
113
114
#ifdef DEBUG
115
    printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", 
116
           bitrate, freq, s->frame_size, table, s->frame_frac_incr);
117
#endif
118
119
    for(i=0;i<s->nb_channels;i++)
120
        s->samples_offset[i] = 0;
121
122 2456e28d Fabrice Bellard
    for(i=0;i<257;i++) {
123
        int v;
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        v = mpa_enwindow[i];
125
#if WFRAC_BITS != 16
126
        v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
127
#endif
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        filter_bank[i] = v;
129
        if ((i & 63) != 0)
130
            v = -v;
131
        if (i != 0)
132
            filter_bank[512 - i] = v;
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    }
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    for(i=0;i<64;i++) {
136
        v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
137
        if (v <= 0)
138
            v = 1;
139
        scale_factor_table[i] = v;
140
#ifdef USE_FLOATS
141
        scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
142
#else
143
#define P 15
144
        scale_factor_shift[i] = 21 - P - (i / 3);
145
        scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
146
#endif
147
    }
148
    for(i=0;i<128;i++) {
149
        v = i - 64;
150
        if (v <= -3)
151
            v = 0;
152
        else if (v < 0)
153
            v = 1;
154
        else if (v == 0)
155
            v = 2;
156
        else if (v < 3)
157
            v = 3;
158
        else 
159
            v = 4;
160
        scale_diff_table[i] = v;
161
    }
162
163
    for(i=0;i<17;i++) {
164
        v = quant_bits[i];
165
        if (v < 0) 
166
            v = -v;
167
        else
168
            v = v * 3;
169
        total_quant_bits[i] = 12 * v;
170
    }
171
172
    return 0;
173
}
174
175 2456e28d Fabrice Bellard
/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
176 afa982fd Fabrice Bellard
static void idct32(int *out, int *tab)
177 de6d9b64 Fabrice Bellard
{
178
    int i, j;
179
    int *t, *t1, xr;
180
    const int *xp = costab32;
181
182
    for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
183
    
184
    t = tab + 30;
185
    t1 = tab + 2;
186
    do {
187
        t[0] += t[-4];
188
        t[1] += t[1 - 4];
189
        t -= 4;
190
    } while (t != t1);
191
192
    t = tab + 28;
193
    t1 = tab + 4;
194
    do {
195
        t[0] += t[-8];
196
        t[1] += t[1-8];
197
        t[2] += t[2-8];
198
        t[3] += t[3-8];
199
        t -= 8;
200
    } while (t != t1);
201
    
202
    t = tab;
203
    t1 = tab + 32;
204
    do {
205
        t[ 3] = -t[ 3];    
206
        t[ 6] = -t[ 6];    
207
        
208
        t[11] = -t[11];    
209
        t[12] = -t[12];    
210
        t[13] = -t[13];    
211
        t[15] = -t[15]; 
212
        t += 16;
213
    } while (t != t1);
214
215
    
216
    t = tab;
217
    t1 = tab + 8;
218
    do {
219
        int x1, x2, x3, x4;
220
        
221
        x3 = MUL(t[16], FIX(SQRT2*0.5));
222
        x4 = t[0] - x3;
223
        x3 = t[0] + x3;
224
        
225
        x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
226
        x1 = MUL((t[8] - x2), xp[0]);
227
        x2 = MUL((t[8] + x2), xp[1]);
228
229
        t[ 0] = x3 + x1;
230
        t[ 8] = x4 - x2;
231
        t[16] = x4 + x2;
232
        t[24] = x3 - x1;
233
        t++;
234
    } while (t != t1);
235
236
    xp += 2;
237
    t = tab;
238
    t1 = tab + 4;
239
    do {
240
        xr = MUL(t[28],xp[0]);
241
        t[28] = (t[0] - xr);
242
        t[0] = (t[0] + xr);
243
244
        xr = MUL(t[4],xp[1]);
245
        t[ 4] = (t[24] - xr);
246
        t[24] = (t[24] + xr);
247
        
248
        xr = MUL(t[20],xp[2]);
249
        t[20] = (t[8] - xr);
250
        t[ 8] = (t[8] + xr);
251
            
252
        xr = MUL(t[12],xp[3]);
253
        t[12] = (t[16] - xr);
254
        t[16] = (t[16] + xr);
255
        t++;
256
    } while (t != t1);
257
    xp += 4;
258
259
    for (i = 0; i < 4; i++) {
260
        xr = MUL(tab[30-i*4],xp[0]);
261
        tab[30-i*4] = (tab[i*4] - xr);
262
        tab[   i*4] = (tab[i*4] + xr);
263
        
264
        xr = MUL(tab[ 2+i*4],xp[1]);
265
        tab[ 2+i*4] = (tab[28-i*4] - xr);
266
        tab[28-i*4] = (tab[28-i*4] + xr);
267
        
268
        xr = MUL(tab[31-i*4],xp[0]);
269
        tab[31-i*4] = (tab[1+i*4] - xr);
270
        tab[ 1+i*4] = (tab[1+i*4] + xr);
271
        
272
        xr = MUL(tab[ 3+i*4],xp[1]);
273
        tab[ 3+i*4] = (tab[29-i*4] - xr);
274
        tab[29-i*4] = (tab[29-i*4] + xr);
275
        
276
        xp += 2;
277
    }
278
279
    t = tab + 30;
280
    t1 = tab + 1;
281
    do {
282
        xr = MUL(t1[0], *xp);
283
        t1[0] = (t[0] - xr);
284
        t[0] = (t[0] + xr);
285
        t -= 2;
286
        t1 += 2;
287
        xp++;
288
    } while (t >= tab);
289
290
    for(i=0;i<32;i++) {
291 afa982fd Fabrice Bellard
        out[i] = tab[bitinv32[i]];
292 de6d9b64 Fabrice Bellard
    }
293
}
294
295 afa982fd Fabrice Bellard
#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
296
297 de6d9b64 Fabrice Bellard
static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
298
{
299
    short *p, *q;
300 afa982fd Fabrice Bellard
    int sum, offset, i, j;
301
    int tmp[64];
302 de6d9b64 Fabrice Bellard
    int tmp1[32];
303
    int *out;
304
305
    //    print_pow1(samples, 1152);
306
307
    offset = s->samples_offset[ch];
308
    out = &s->sb_samples[ch][0][0][0];
309
    for(j=0;j<36;j++) {
310
        /* 32 samples at once */
311
        for(i=0;i<32;i++) {
312
            s->samples_buf[ch][offset + (31 - i)] = samples[0];
313
            samples += incr;
314
        }
315
316
        /* filter */
317
        p = s->samples_buf[ch] + offset;
318
        q = filter_bank;
319
        /* maxsum = 23169 */
320
        for(i=0;i<64;i++) {
321
            sum = p[0*64] * q[0*64];
322
            sum += p[1*64] * q[1*64];
323
            sum += p[2*64] * q[2*64];
324
            sum += p[3*64] * q[3*64];
325
            sum += p[4*64] * q[4*64];
326
            sum += p[5*64] * q[5*64];
327
            sum += p[6*64] * q[6*64];
328
            sum += p[7*64] * q[7*64];
329 afa982fd Fabrice Bellard
            tmp[i] = sum;
330 de6d9b64 Fabrice Bellard
            p++;
331
            q++;
332
        }
333 afa982fd Fabrice Bellard
        tmp1[0] = tmp[16] >> WSHIFT;
334
        for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
335
        for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
336 de6d9b64 Fabrice Bellard
337 afa982fd Fabrice Bellard
        idct32(out, tmp1);
338 de6d9b64 Fabrice Bellard
339
        /* advance of 32 samples */
340
        offset -= 32;
341
        out += 32;
342
        /* handle the wrap around */
343
        if (offset < 0) {
344
            memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32), 
345
                    s->samples_buf[ch], (512 - 32) * 2);
346
            offset = SAMPLES_BUF_SIZE - 512;
347
        }
348
    }
349
    s->samples_offset[ch] = offset;
350
351
    //    print_pow(s->sb_samples, 1152);
352
}
353
354
static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
355
                                  unsigned char scale_factors[SBLIMIT][3], 
356
                                  int sb_samples[3][12][SBLIMIT],
357
                                  int sblimit)
358
{
359
    int *p, vmax, v, n, i, j, k, code;
360
    int index, d1, d2;
361
    unsigned char *sf = &scale_factors[0][0];
362
    
363
    for(j=0;j<sblimit;j++) {
364
        for(i=0;i<3;i++) {
365
            /* find the max absolute value */
366
            p = &sb_samples[i][0][j];
367
            vmax = abs(*p);
368
            for(k=1;k<12;k++) {
369
                p += SBLIMIT;
370
                v = abs(*p);
371
                if (v > vmax)
372
                    vmax = v;
373
            }
374
            /* compute the scale factor index using log 2 computations */
375
            if (vmax > 0) {
376 935442b5 Fabrice Bellard
                n = av_log2(vmax);
377 de6d9b64 Fabrice Bellard
                /* n is the position of the MSB of vmax. now 
378
                   use at most 2 compares to find the index */
379
                index = (21 - n) * 3 - 3;
380
                if (index >= 0) {
381
                    while (vmax <= scale_factor_table[index+1])
382
                        index++;
383
                } else {
384
                    index = 0; /* very unlikely case of overflow */
385
                }
386
            } else {
387 afa982fd Fabrice Bellard
                index = 62; /* value 63 is not allowed */
388 de6d9b64 Fabrice Bellard
            }
389 afa982fd Fabrice Bellard
390 de6d9b64 Fabrice Bellard
#if 0
391
            printf("%2d:%d in=%x %x %d\n", 
392
                   j, i, vmax, scale_factor_table[index], index);
393
#endif
394
            /* store the scale factor */
395
            assert(index >=0 && index <= 63);
396
            sf[i] = index;
397
        }
398
399
        /* compute the transmission factor : look if the scale factors
400
           are close enough to each other */
401
        d1 = scale_diff_table[sf[0] - sf[1] + 64];
402
        d2 = scale_diff_table[sf[1] - sf[2] + 64];
403
        
404
        /* handle the 25 cases */
405
        switch(d1 * 5 + d2) {
406
        case 0*5+0:
407
        case 0*5+4:
408
        case 3*5+4:
409
        case 4*5+0:
410
        case 4*5+4:
411
            code = 0;
412
            break;
413
        case 0*5+1:
414
        case 0*5+2:
415
        case 4*5+1:
416
        case 4*5+2:
417
            code = 3;
418
            sf[2] = sf[1];
419
            break;
420
        case 0*5+3:
421
        case 4*5+3:
422
            code = 3;
423
            sf[1] = sf[2];
424
            break;
425
        case 1*5+0:
426
        case 1*5+4:
427
        case 2*5+4:
428
            code = 1;
429
            sf[1] = sf[0];
430
            break;
431
        case 1*5+1:
432
        case 1*5+2:
433
        case 2*5+0:
434
        case 2*5+1:
435
        case 2*5+2:
436
            code = 2;
437
            sf[1] = sf[2] = sf[0];
438
            break;
439
        case 2*5+3:
440
        case 3*5+3:
441
            code = 2;
442
            sf[0] = sf[1] = sf[2];
443
            break;
444
        case 3*5+0:
445
        case 3*5+1:
446
        case 3*5+2:
447
            code = 2;
448
            sf[0] = sf[2] = sf[1];
449
            break;
450
        case 1*5+3:
451
            code = 2;
452
            if (sf[0] > sf[2])
453
              sf[0] = sf[2];
454
            sf[1] = sf[2] = sf[0];
455
            break;
456
        default:
457
            abort();
458
        }
459
        
460
#if 0
461
        printf("%d: %2d %2d %2d %d %d -> %d\n", j, 
462
               sf[0], sf[1], sf[2], d1, d2, code);
463
#endif
464
        scale_code[j] = code;
465
        sf += 3;
466
    }
467
}
468
469
/* The most important function : psycho acoustic module. In this
470
   encoder there is basically none, so this is the worst you can do,
471
   but also this is the simpler. */
472
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
473
{
474
    int i;
475
476
    for(i=0;i<s->sblimit;i++) {
477
        smr[i] = (int)(fixed_smr[i] * 10);
478
    }
479
}
480
481
482
#define SB_NOTALLOCATED  0
483
#define SB_ALLOCATED     1
484
#define SB_NOMORE        2
485
486
/* Try to maximize the smr while using a number of bits inferior to
487
   the frame size. I tried to make the code simpler, faster and
488
   smaller than other encoders :-) */
489
static void compute_bit_allocation(MpegAudioContext *s, 
490
                                   short smr1[MPA_MAX_CHANNELS][SBLIMIT],
491
                                   unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
492
                                   int *padding)
493
{
494
    int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
495
    int incr;
496
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
497
    unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
498
    const unsigned char *alloc;
499
500
    memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
501
    memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
502
    memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
503
    
504
    /* compute frame size and padding */
505
    max_frame_size = s->frame_size;
506
    s->frame_frac += s->frame_frac_incr;
507
    if (s->frame_frac >= 65536) {
508
        s->frame_frac -= 65536;
509
        s->do_padding = 1;
510
        max_frame_size += 8;
511
    } else {
512
        s->do_padding = 0;
513
    }
514
515
    /* compute the header + bit alloc size */
516
    current_frame_size = 32;
517
    alloc = s->alloc_table;
518
    for(i=0;i<s->sblimit;i++) {
519
        incr = alloc[0];
520
        current_frame_size += incr * s->nb_channels;
521
        alloc += 1 << incr;
522
    }
523
    for(;;) {
524
        /* look for the subband with the largest signal to mask ratio */
525
        max_sb = -1;
526
        max_ch = -1;
527
        max_smr = 0x80000000;
528
        for(ch=0;ch<s->nb_channels;ch++) {
529
            for(i=0;i<s->sblimit;i++) {
530
                if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
531
                    max_smr = smr[ch][i];
532
                    max_sb = i;
533
                    max_ch = ch;
534
                }
535
            }
536
        }
537
#if 0
538
        printf("current=%d max=%d max_sb=%d alloc=%d\n", 
539
               current_frame_size, max_frame_size, max_sb,
540
               bit_alloc[max_sb]);
541
#endif        
542
        if (max_sb < 0)
543
            break;
544
        
545
        /* find alloc table entry (XXX: not optimal, should use
546
           pointer table) */
547
        alloc = s->alloc_table;
548
        for(i=0;i<max_sb;i++) {
549
            alloc += 1 << alloc[0];
550
        }
551
552
        if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
553
            /* nothing was coded for this band: add the necessary bits */
554
            incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
555
            incr += total_quant_bits[alloc[1]];
556
        } else {
557
            /* increments bit allocation */
558
            b = bit_alloc[max_ch][max_sb];
559
            incr = total_quant_bits[alloc[b + 1]] - 
560
                total_quant_bits[alloc[b]];
561
        }
562
563
        if (current_frame_size + incr <= max_frame_size) {
564
            /* can increase size */
565
            b = ++bit_alloc[max_ch][max_sb];
566
            current_frame_size += incr;
567
            /* decrease smr by the resolution we added */
568
            smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
569
            /* max allocation size reached ? */
570
            if (b == ((1 << alloc[0]) - 1))
571
                subband_status[max_ch][max_sb] = SB_NOMORE;
572
            else
573
                subband_status[max_ch][max_sb] = SB_ALLOCATED;
574
        } else {
575
            /* cannot increase the size of this subband */
576
            subband_status[max_ch][max_sb] = SB_NOMORE;
577
        }
578
    }
579
    *padding = max_frame_size - current_frame_size;
580
    assert(*padding >= 0);
581
582
#if 0
583
    for(i=0;i<s->sblimit;i++) {
584
        printf("%d ", bit_alloc[i]);
585
    }
586
    printf("\n");
587
#endif
588
}
589
590
/*
591
 * Output the mpeg audio layer 2 frame. Note how the code is small
592
 * compared to other encoders :-)
593
 */
594
static void encode_frame(MpegAudioContext *s,
595
                         unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
596
                         int padding)
597
{
598
    int i, j, k, l, bit_alloc_bits, b, ch;
599
    unsigned char *sf;
600
    int q[3];
601
    PutBitContext *p = &s->pb;
602
603
    /* header */
604
605
    put_bits(p, 12, 0xfff);
606
    put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
607
    put_bits(p, 2, 4-2);  /* layer 2 */
608
    put_bits(p, 1, 1); /* no error protection */
609
    put_bits(p, 4, s->bitrate_index);
610
    put_bits(p, 2, s->freq_index);
611
    put_bits(p, 1, s->do_padding); /* use padding */
612
    put_bits(p, 1, 0);             /* private_bit */
613
    put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
614
    put_bits(p, 2, 0); /* mode_ext */
615
    put_bits(p, 1, 0); /* no copyright */
616
    put_bits(p, 1, 1); /* original */
617
    put_bits(p, 2, 0); /* no emphasis */
618
619
    /* bit allocation */
620
    j = 0;
621
    for(i=0;i<s->sblimit;i++) {
622
        bit_alloc_bits = s->alloc_table[j];
623
        for(ch=0;ch<s->nb_channels;ch++) {
624
            put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
625
        }
626
        j += 1 << bit_alloc_bits;
627
    }
628
    
629
    /* scale codes */
630
    for(i=0;i<s->sblimit;i++) {
631
        for(ch=0;ch<s->nb_channels;ch++) {
632
            if (bit_alloc[ch][i]) 
633
                put_bits(p, 2, s->scale_code[ch][i]);
634
        }
635
    }
636
637
    /* scale factors */
638
    for(i=0;i<s->sblimit;i++) {
639
        for(ch=0;ch<s->nb_channels;ch++) {
640
            if (bit_alloc[ch][i]) {
641
                sf = &s->scale_factors[ch][i][0];
642
                switch(s->scale_code[ch][i]) {
643
                case 0:
644
                    put_bits(p, 6, sf[0]);
645
                    put_bits(p, 6, sf[1]);
646
                    put_bits(p, 6, sf[2]);
647
                    break;
648
                case 3:
649
                case 1:
650
                    put_bits(p, 6, sf[0]);
651
                    put_bits(p, 6, sf[2]);
652
                    break;
653
                case 2:
654
                    put_bits(p, 6, sf[0]);
655
                    break;
656
                }
657
            }
658
        }
659
    }
660
    
661
    /* quantization & write sub band samples */
662
663
    for(k=0;k<3;k++) {
664
        for(l=0;l<12;l+=3) {
665
            j = 0;
666
            for(i=0;i<s->sblimit;i++) {
667
                bit_alloc_bits = s->alloc_table[j];
668
                for(ch=0;ch<s->nb_channels;ch++) {
669
                    b = bit_alloc[ch][i];
670
                    if (b) {
671
                        int qindex, steps, m, sample, bits;
672
                        /* we encode 3 sub band samples of the same sub band at a time */
673
                        qindex = s->alloc_table[j+b];
674
                        steps = quant_steps[qindex];
675
                        for(m=0;m<3;m++) {
676
                            sample = s->sb_samples[ch][k][l + m][i];
677
                            /* divide by scale factor */
678
#ifdef USE_FLOATS
679
                            {
680
                                float a;
681
                                a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
682
                                q[m] = (int)((a + 1.0) * steps * 0.5);
683
                            }
684
#else
685
                            {
686
                                int q1, e, shift, mult;
687
                                e = s->scale_factors[ch][i][k];
688
                                shift = scale_factor_shift[e];
689
                                mult = scale_factor_mult[e];
690
                                
691
                                /* normalize to P bits */
692
                                if (shift < 0)
693
                                    q1 = sample << (-shift);
694
                                else
695
                                    q1 = sample >> shift;
696
                                q1 = (q1 * mult) >> P;
697
                                q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
698
                            }
699
#endif
700
                            if (q[m] >= steps)
701
                                q[m] = steps - 1;
702
                            assert(q[m] >= 0 && q[m] < steps);
703
                        }
704
                        bits = quant_bits[qindex];
705
                        if (bits < 0) {
706
                            /* group the 3 values to save bits */
707
                            put_bits(p, -bits, 
708
                                     q[0] + steps * (q[1] + steps * q[2]));
709
#if 0
710
                            printf("%d: gr1 %d\n", 
711
                                   i, q[0] + steps * (q[1] + steps * q[2]));
712
#endif
713
                        } else {
714
#if 0
715
                            printf("%d: gr3 %d %d %d\n", 
716
                                   i, q[0], q[1], q[2]);
717
#endif                               
718
                            put_bits(p, bits, q[0]);
719
                            put_bits(p, bits, q[1]);
720
                            put_bits(p, bits, q[2]);
721
                        }
722
                    }
723
                }
724
                /* next subband in alloc table */
725
                j += 1 << bit_alloc_bits; 
726
            }
727
        }
728
    }
729
730
    /* padding */
731
    for(i=0;i<padding;i++)
732
        put_bits(p, 1, 0);
733
734
    /* flush */
735
    flush_put_bits(p);
736
}
737
738
int MPA_encode_frame(AVCodecContext *avctx,
739
                     unsigned char *frame, int buf_size, void *data)
740
{
741
    MpegAudioContext *s = avctx->priv_data;
742
    short *samples = data;
743
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
744
    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
745
    int padding, i;
746
747
    for(i=0;i<s->nb_channels;i++) {
748
        filter(s, i, samples + i, s->nb_channels);
749
    }
750
751
    for(i=0;i<s->nb_channels;i++) {
752
        compute_scale_factors(s->scale_code[i], s->scale_factors[i], 
753
                              s->sb_samples[i], s->sblimit);
754
    }
755
    for(i=0;i<s->nb_channels;i++) {
756
        psycho_acoustic_model(s, smr[i]);
757
    }
758
    compute_bit_allocation(s, smr, bit_alloc, &padding);
759
760
    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
761
762
    encode_frame(s, bit_alloc, padding);
763
    
764
    s->nb_samples += MPA_FRAME_SIZE;
765 17592475 Michael Niedermayer
    return pbBufPtr(&s->pb) - s->pb.buf;
766 de6d9b64 Fabrice Bellard
}
767
768
769
AVCodec mp2_encoder = {
770
    "mp2",
771
    CODEC_TYPE_AUDIO,
772
    CODEC_ID_MP2,
773
    sizeof(MpegAudioContext),
774
    MPA_encode_init,
775
    MPA_encode_frame,
776
    NULL,
777
};
778 cd4af68a Zdenek Kabelac
779
#undef FIX