Statistics
| Branch: | Revision:

ffmpeg / libavcodec / mpegaudio.c @ 935442b5

History | View | Annotate | Download (21.7 KB)

1
/*
2
 * The simplest mpeg audio layer 2 encoder
3
 * Copyright (c) 2000 Gerard Lantau.
4
 *
5
 * This program is free software; you can redistribute it and/or modify
6
 * it under the terms of the GNU General Public License as published by
7
 * the Free Software Foundation; either version 2 of the License, or
8
 * (at your option) any later version.
9
 *
10
 * This program is distributed in the hope that it will be useful,
11
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
13
 * GNU General Public License for more details.
14
 *
15
 * You should have received a copy of the GNU General Public License
16
 * along with this program; if not, write to the Free Software
17
 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
18
 */
19
#include "avcodec.h"
20
#include <math.h>
21
#include "mpegaudio.h"
22

    
23
/* define it to use floats in quantization (I don't like floats !) */
24
//#define USE_FLOATS
25

    
26
#define MPA_STEREO  0
27
#define MPA_JSTEREO 1
28
#define MPA_DUAL    2
29
#define MPA_MONO    3
30

    
31
#include "mpegaudiotab.h"
32

    
33
int MPA_encode_init(AVCodecContext *avctx)
34
{
35
    MpegAudioContext *s = avctx->priv_data;
36
    int freq = avctx->sample_rate;
37
    int bitrate = avctx->bit_rate;
38
    int channels = avctx->channels;
39
    int i, v, table, ch_bitrate;
40
    float a;
41

    
42
    if (channels > 2)
43
        return -1;
44
    bitrate = bitrate / 1000;
45
    s->nb_channels = channels;
46
    s->freq = freq;
47
    s->bit_rate = bitrate * 1000;
48
    avctx->frame_size = MPA_FRAME_SIZE;
49
    avctx->key_frame = 1; /* always key frame */
50

    
51
    /* encoding freq */
52
    s->lsf = 0;
53
    for(i=0;i<3;i++) {
54
        if (freq_tab[i] == freq) 
55
            break;
56
        if ((freq_tab[i] / 2) == freq) {
57
            s->lsf = 1;
58
            break;
59
        }
60
    }
61
    if (i == 3)
62
        return -1;
63
    s->freq_index = i;
64

    
65
    /* encoding bitrate & frequency */
66
    for(i=0;i<15;i++) {
67
        if (bitrate_tab[1-s->lsf][i] == bitrate) 
68
            break;
69
    }
70
    if (i == 15)
71
        return -1;
72
    s->bitrate_index = i;
73

    
74
    /* compute total header size & pad bit */
75
    
76
    a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
77
    s->frame_size = ((int)a) * 8;
78

    
79
    /* frame fractional size to compute padding */
80
    s->frame_frac = 0;
81
    s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
82
    
83
    /* select the right allocation table */
84
    ch_bitrate = bitrate / s->nb_channels;
85
    if (!s->lsf) {
86
        if ((freq == 48000 && ch_bitrate >= 56) ||
87
            (ch_bitrate >= 56 && ch_bitrate <= 80)) 
88
            table = 0;
89
        else if (freq != 48000 && ch_bitrate >= 96) 
90
            table = 1;
91
        else if (freq != 32000 && ch_bitrate <= 48) 
92
            table = 2;
93
        else 
94
            table = 3;
95
    } else {
96
        table = 4;
97
    }
98
    /* number of used subbands */
99
    s->sblimit = sblimit_table[table];
100
    s->alloc_table = alloc_tables[table];
101

    
102
#ifdef DEBUG
103
    printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", 
104
           bitrate, freq, s->frame_size, table, s->frame_frac_incr);
105
#endif
106

    
107
    for(i=0;i<s->nb_channels;i++)
108
        s->samples_offset[i] = 0;
109

    
110
    for(i=0;i<512;i++) {
111
        float a = enwindow[i] * 32768.0 * 16.0;
112
        filter_bank[i] = (int)(a);
113
    }
114
    for(i=0;i<64;i++) {
115
        v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
116
        if (v <= 0)
117
            v = 1;
118
        scale_factor_table[i] = v;
119
#ifdef USE_FLOATS
120
        scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
121
#else
122
#define P 15
123
        scale_factor_shift[i] = 21 - P - (i / 3);
124
        scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
125
#endif
126
    }
127
    for(i=0;i<128;i++) {
128
        v = i - 64;
129
        if (v <= -3)
130
            v = 0;
131
        else if (v < 0)
132
            v = 1;
133
        else if (v == 0)
134
            v = 2;
135
        else if (v < 3)
136
            v = 3;
137
        else 
138
            v = 4;
139
        scale_diff_table[i] = v;
140
    }
141

    
142
    for(i=0;i<17;i++) {
143
        v = quant_bits[i];
144
        if (v < 0) 
145
            v = -v;
146
        else
147
            v = v * 3;
148
        total_quant_bits[i] = 12 * v;
149
    }
150

    
151
    return 0;
152
}
153

    
154
/* 32 point floating point IDCT */
155
static void idct32(int *out, int *tab, int sblimit, int left_shift)
156
{
157
    int i, j;
158
    int *t, *t1, xr;
159
    const int *xp = costab32;
160

    
161
    for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
162
    
163
    t = tab + 30;
164
    t1 = tab + 2;
165
    do {
166
        t[0] += t[-4];
167
        t[1] += t[1 - 4];
168
        t -= 4;
169
    } while (t != t1);
170

    
171
    t = tab + 28;
172
    t1 = tab + 4;
173
    do {
174
        t[0] += t[-8];
175
        t[1] += t[1-8];
176
        t[2] += t[2-8];
177
        t[3] += t[3-8];
178
        t -= 8;
179
    } while (t != t1);
180
    
181
    t = tab;
182
    t1 = tab + 32;
183
    do {
184
        t[ 3] = -t[ 3];    
185
        t[ 6] = -t[ 6];    
186
        
187
        t[11] = -t[11];    
188
        t[12] = -t[12];    
189
        t[13] = -t[13];    
190
        t[15] = -t[15]; 
191
        t += 16;
192
    } while (t != t1);
193

    
194
    
195
    t = tab;
196
    t1 = tab + 8;
197
    do {
198
        int x1, x2, x3, x4;
199
        
200
        x3 = MUL(t[16], FIX(SQRT2*0.5));
201
        x4 = t[0] - x3;
202
        x3 = t[0] + x3;
203
        
204
        x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
205
        x1 = MUL((t[8] - x2), xp[0]);
206
        x2 = MUL((t[8] + x2), xp[1]);
207

    
208
        t[ 0] = x3 + x1;
209
        t[ 8] = x4 - x2;
210
        t[16] = x4 + x2;
211
        t[24] = x3 - x1;
212
        t++;
213
    } while (t != t1);
214

    
215
    xp += 2;
216
    t = tab;
217
    t1 = tab + 4;
218
    do {
219
        xr = MUL(t[28],xp[0]);
220
        t[28] = (t[0] - xr);
221
        t[0] = (t[0] + xr);
222

    
223
        xr = MUL(t[4],xp[1]);
224
        t[ 4] = (t[24] - xr);
225
        t[24] = (t[24] + xr);
226
        
227
        xr = MUL(t[20],xp[2]);
228
        t[20] = (t[8] - xr);
229
        t[ 8] = (t[8] + xr);
230
            
231
        xr = MUL(t[12],xp[3]);
232
        t[12] = (t[16] - xr);
233
        t[16] = (t[16] + xr);
234
        t++;
235
    } while (t != t1);
236
    xp += 4;
237

    
238
    for (i = 0; i < 4; i++) {
239
        xr = MUL(tab[30-i*4],xp[0]);
240
        tab[30-i*4] = (tab[i*4] - xr);
241
        tab[   i*4] = (tab[i*4] + xr);
242
        
243
        xr = MUL(tab[ 2+i*4],xp[1]);
244
        tab[ 2+i*4] = (tab[28-i*4] - xr);
245
        tab[28-i*4] = (tab[28-i*4] + xr);
246
        
247
        xr = MUL(tab[31-i*4],xp[0]);
248
        tab[31-i*4] = (tab[1+i*4] - xr);
249
        tab[ 1+i*4] = (tab[1+i*4] + xr);
250
        
251
        xr = MUL(tab[ 3+i*4],xp[1]);
252
        tab[ 3+i*4] = (tab[29-i*4] - xr);
253
        tab[29-i*4] = (tab[29-i*4] + xr);
254
        
255
        xp += 2;
256
    }
257

    
258
    t = tab + 30;
259
    t1 = tab + 1;
260
    do {
261
        xr = MUL(t1[0], *xp);
262
        t1[0] = (t[0] - xr);
263
        t[0] = (t[0] + xr);
264
        t -= 2;
265
        t1 += 2;
266
        xp++;
267
    } while (t >= tab);
268

    
269
    for(i=0;i<32;i++) {
270
        out[i] = tab[bitinv32[i]] << left_shift;
271
    }
272
}
273

    
274
static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
275
{
276
    short *p, *q;
277
    int sum, offset, i, j, norm, n;
278
    short tmp[64];
279
    int tmp1[32];
280
    int *out;
281

    
282
    //    print_pow1(samples, 1152);
283

    
284
    offset = s->samples_offset[ch];
285
    out = &s->sb_samples[ch][0][0][0];
286
    for(j=0;j<36;j++) {
287
        /* 32 samples at once */
288
        for(i=0;i<32;i++) {
289
            s->samples_buf[ch][offset + (31 - i)] = samples[0];
290
            samples += incr;
291
        }
292

    
293
        /* filter */
294
        p = s->samples_buf[ch] + offset;
295
        q = filter_bank;
296
        /* maxsum = 23169 */
297
        for(i=0;i<64;i++) {
298
            sum = p[0*64] * q[0*64];
299
            sum += p[1*64] * q[1*64];
300
            sum += p[2*64] * q[2*64];
301
            sum += p[3*64] * q[3*64];
302
            sum += p[4*64] * q[4*64];
303
            sum += p[5*64] * q[5*64];
304
            sum += p[6*64] * q[6*64];
305
            sum += p[7*64] * q[7*64];
306
            tmp[i] = sum >> 14;
307
            p++;
308
            q++;
309
        }
310
        tmp1[0] = tmp[16];
311
        for( i=1; i<=16; i++ ) tmp1[i] = tmp[i+16]+tmp[16-i];
312
        for( i=17; i<=31; i++ ) tmp1[i] = tmp[i+16]-tmp[80-i];
313

    
314
        /* integer IDCT 32 with normalization. XXX: There may be some
315
           overflow left */
316
        norm = 0;
317
        for(i=0;i<32;i++) {
318
            norm |= abs(tmp1[i]);
319
        }
320
        n = av_log2(norm) - 12;
321
        if (n > 0) {
322
            for(i=0;i<32;i++) 
323
                tmp1[i] >>= n;
324
        } else {
325
            n = 0;
326
        }
327

    
328
        idct32(out, tmp1, s->sblimit, n);
329

    
330
        /* advance of 32 samples */
331
        offset -= 32;
332
        out += 32;
333
        /* handle the wrap around */
334
        if (offset < 0) {
335
            memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32), 
336
                    s->samples_buf[ch], (512 - 32) * 2);
337
            offset = SAMPLES_BUF_SIZE - 512;
338
        }
339
    }
340
    s->samples_offset[ch] = offset;
341

    
342
    //    print_pow(s->sb_samples, 1152);
343
}
344

    
345
static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
346
                                  unsigned char scale_factors[SBLIMIT][3], 
347
                                  int sb_samples[3][12][SBLIMIT],
348
                                  int sblimit)
349
{
350
    int *p, vmax, v, n, i, j, k, code;
351
    int index, d1, d2;
352
    unsigned char *sf = &scale_factors[0][0];
353
    
354
    for(j=0;j<sblimit;j++) {
355
        for(i=0;i<3;i++) {
356
            /* find the max absolute value */
357
            p = &sb_samples[i][0][j];
358
            vmax = abs(*p);
359
            for(k=1;k<12;k++) {
360
                p += SBLIMIT;
361
                v = abs(*p);
362
                if (v > vmax)
363
                    vmax = v;
364
            }
365
            /* compute the scale factor index using log 2 computations */
366
            if (vmax > 0) {
367
                n = av_log2(vmax);
368
                /* n is the position of the MSB of vmax. now 
369
                   use at most 2 compares to find the index */
370
                index = (21 - n) * 3 - 3;
371
                if (index >= 0) {
372
                    while (vmax <= scale_factor_table[index+1])
373
                        index++;
374
                } else {
375
                    index = 0; /* very unlikely case of overflow */
376
                }
377
            } else {
378
                index = 63;
379
            }
380
            
381
#if 0
382
            printf("%2d:%d in=%x %x %d\n", 
383
                   j, i, vmax, scale_factor_table[index], index);
384
#endif
385
            /* store the scale factor */
386
            assert(index >=0 && index <= 63);
387
            sf[i] = index;
388
        }
389

    
390
        /* compute the transmission factor : look if the scale factors
391
           are close enough to each other */
392
        d1 = scale_diff_table[sf[0] - sf[1] + 64];
393
        d2 = scale_diff_table[sf[1] - sf[2] + 64];
394
        
395
        /* handle the 25 cases */
396
        switch(d1 * 5 + d2) {
397
        case 0*5+0:
398
        case 0*5+4:
399
        case 3*5+4:
400
        case 4*5+0:
401
        case 4*5+4:
402
            code = 0;
403
            break;
404
        case 0*5+1:
405
        case 0*5+2:
406
        case 4*5+1:
407
        case 4*5+2:
408
            code = 3;
409
            sf[2] = sf[1];
410
            break;
411
        case 0*5+3:
412
        case 4*5+3:
413
            code = 3;
414
            sf[1] = sf[2];
415
            break;
416
        case 1*5+0:
417
        case 1*5+4:
418
        case 2*5+4:
419
            code = 1;
420
            sf[1] = sf[0];
421
            break;
422
        case 1*5+1:
423
        case 1*5+2:
424
        case 2*5+0:
425
        case 2*5+1:
426
        case 2*5+2:
427
            code = 2;
428
            sf[1] = sf[2] = sf[0];
429
            break;
430
        case 2*5+3:
431
        case 3*5+3:
432
            code = 2;
433
            sf[0] = sf[1] = sf[2];
434
            break;
435
        case 3*5+0:
436
        case 3*5+1:
437
        case 3*5+2:
438
            code = 2;
439
            sf[0] = sf[2] = sf[1];
440
            break;
441
        case 1*5+3:
442
            code = 2;
443
            if (sf[0] > sf[2])
444
              sf[0] = sf[2];
445
            sf[1] = sf[2] = sf[0];
446
            break;
447
        default:
448
            abort();
449
        }
450
        
451
#if 0
452
        printf("%d: %2d %2d %2d %d %d -> %d\n", j, 
453
               sf[0], sf[1], sf[2], d1, d2, code);
454
#endif
455
        scale_code[j] = code;
456
        sf += 3;
457
    }
458
}
459

    
460
/* The most important function : psycho acoustic module. In this
461
   encoder there is basically none, so this is the worst you can do,
462
   but also this is the simpler. */
463
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
464
{
465
    int i;
466

    
467
    for(i=0;i<s->sblimit;i++) {
468
        smr[i] = (int)(fixed_smr[i] * 10);
469
    }
470
}
471

    
472

    
473
#define SB_NOTALLOCATED  0
474
#define SB_ALLOCATED     1
475
#define SB_NOMORE        2
476

    
477
/* Try to maximize the smr while using a number of bits inferior to
478
   the frame size. I tried to make the code simpler, faster and
479
   smaller than other encoders :-) */
480
static void compute_bit_allocation(MpegAudioContext *s, 
481
                                   short smr1[MPA_MAX_CHANNELS][SBLIMIT],
482
                                   unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
483
                                   int *padding)
484
{
485
    int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
486
    int incr;
487
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
488
    unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
489
    const unsigned char *alloc;
490

    
491
    memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
492
    memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
493
    memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
494
    
495
    /* compute frame size and padding */
496
    max_frame_size = s->frame_size;
497
    s->frame_frac += s->frame_frac_incr;
498
    if (s->frame_frac >= 65536) {
499
        s->frame_frac -= 65536;
500
        s->do_padding = 1;
501
        max_frame_size += 8;
502
    } else {
503
        s->do_padding = 0;
504
    }
505

    
506
    /* compute the header + bit alloc size */
507
    current_frame_size = 32;
508
    alloc = s->alloc_table;
509
    for(i=0;i<s->sblimit;i++) {
510
        incr = alloc[0];
511
        current_frame_size += incr * s->nb_channels;
512
        alloc += 1 << incr;
513
    }
514
    for(;;) {
515
        /* look for the subband with the largest signal to mask ratio */
516
        max_sb = -1;
517
        max_ch = -1;
518
        max_smr = 0x80000000;
519
        for(ch=0;ch<s->nb_channels;ch++) {
520
            for(i=0;i<s->sblimit;i++) {
521
                if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
522
                    max_smr = smr[ch][i];
523
                    max_sb = i;
524
                    max_ch = ch;
525
                }
526
            }
527
        }
528
#if 0
529
        printf("current=%d max=%d max_sb=%d alloc=%d\n", 
530
               current_frame_size, max_frame_size, max_sb,
531
               bit_alloc[max_sb]);
532
#endif        
533
        if (max_sb < 0)
534
            break;
535
        
536
        /* find alloc table entry (XXX: not optimal, should use
537
           pointer table) */
538
        alloc = s->alloc_table;
539
        for(i=0;i<max_sb;i++) {
540
            alloc += 1 << alloc[0];
541
        }
542

    
543
        if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
544
            /* nothing was coded for this band: add the necessary bits */
545
            incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
546
            incr += total_quant_bits[alloc[1]];
547
        } else {
548
            /* increments bit allocation */
549
            b = bit_alloc[max_ch][max_sb];
550
            incr = total_quant_bits[alloc[b + 1]] - 
551
                total_quant_bits[alloc[b]];
552
        }
553

    
554
        if (current_frame_size + incr <= max_frame_size) {
555
            /* can increase size */
556
            b = ++bit_alloc[max_ch][max_sb];
557
            current_frame_size += incr;
558
            /* decrease smr by the resolution we added */
559
            smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
560
            /* max allocation size reached ? */
561
            if (b == ((1 << alloc[0]) - 1))
562
                subband_status[max_ch][max_sb] = SB_NOMORE;
563
            else
564
                subband_status[max_ch][max_sb] = SB_ALLOCATED;
565
        } else {
566
            /* cannot increase the size of this subband */
567
            subband_status[max_ch][max_sb] = SB_NOMORE;
568
        }
569
    }
570
    *padding = max_frame_size - current_frame_size;
571
    assert(*padding >= 0);
572

    
573
#if 0
574
    for(i=0;i<s->sblimit;i++) {
575
        printf("%d ", bit_alloc[i]);
576
    }
577
    printf("\n");
578
#endif
579
}
580

    
581
/*
582
 * Output the mpeg audio layer 2 frame. Note how the code is small
583
 * compared to other encoders :-)
584
 */
585
static void encode_frame(MpegAudioContext *s,
586
                         unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
587
                         int padding)
588
{
589
    int i, j, k, l, bit_alloc_bits, b, ch;
590
    unsigned char *sf;
591
    int q[3];
592
    PutBitContext *p = &s->pb;
593

    
594
    /* header */
595

    
596
    put_bits(p, 12, 0xfff);
597
    put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
598
    put_bits(p, 2, 4-2);  /* layer 2 */
599
    put_bits(p, 1, 1); /* no error protection */
600
    put_bits(p, 4, s->bitrate_index);
601
    put_bits(p, 2, s->freq_index);
602
    put_bits(p, 1, s->do_padding); /* use padding */
603
    put_bits(p, 1, 0);             /* private_bit */
604
    put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
605
    put_bits(p, 2, 0); /* mode_ext */
606
    put_bits(p, 1, 0); /* no copyright */
607
    put_bits(p, 1, 1); /* original */
608
    put_bits(p, 2, 0); /* no emphasis */
609

    
610
    /* bit allocation */
611
    j = 0;
612
    for(i=0;i<s->sblimit;i++) {
613
        bit_alloc_bits = s->alloc_table[j];
614
        for(ch=0;ch<s->nb_channels;ch++) {
615
            put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
616
        }
617
        j += 1 << bit_alloc_bits;
618
    }
619
    
620
    /* scale codes */
621
    for(i=0;i<s->sblimit;i++) {
622
        for(ch=0;ch<s->nb_channels;ch++) {
623
            if (bit_alloc[ch][i]) 
624
                put_bits(p, 2, s->scale_code[ch][i]);
625
        }
626
    }
627

    
628
    /* scale factors */
629
    for(i=0;i<s->sblimit;i++) {
630
        for(ch=0;ch<s->nb_channels;ch++) {
631
            if (bit_alloc[ch][i]) {
632
                sf = &s->scale_factors[ch][i][0];
633
                switch(s->scale_code[ch][i]) {
634
                case 0:
635
                    put_bits(p, 6, sf[0]);
636
                    put_bits(p, 6, sf[1]);
637
                    put_bits(p, 6, sf[2]);
638
                    break;
639
                case 3:
640
                case 1:
641
                    put_bits(p, 6, sf[0]);
642
                    put_bits(p, 6, sf[2]);
643
                    break;
644
                case 2:
645
                    put_bits(p, 6, sf[0]);
646
                    break;
647
                }
648
            }
649
        }
650
    }
651
    
652
    /* quantization & write sub band samples */
653

    
654
    for(k=0;k<3;k++) {
655
        for(l=0;l<12;l+=3) {
656
            j = 0;
657
            for(i=0;i<s->sblimit;i++) {
658
                bit_alloc_bits = s->alloc_table[j];
659
                for(ch=0;ch<s->nb_channels;ch++) {
660
                    b = bit_alloc[ch][i];
661
                    if (b) {
662
                        int qindex, steps, m, sample, bits;
663
                        /* we encode 3 sub band samples of the same sub band at a time */
664
                        qindex = s->alloc_table[j+b];
665
                        steps = quant_steps[qindex];
666
                        for(m=0;m<3;m++) {
667
                            sample = s->sb_samples[ch][k][l + m][i];
668
                            /* divide by scale factor */
669
#ifdef USE_FLOATS
670
                            {
671
                                float a;
672
                                a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
673
                                q[m] = (int)((a + 1.0) * steps * 0.5);
674
                            }
675
#else
676
                            {
677
                                int q1, e, shift, mult;
678
                                e = s->scale_factors[ch][i][k];
679
                                shift = scale_factor_shift[e];
680
                                mult = scale_factor_mult[e];
681
                                
682
                                /* normalize to P bits */
683
                                if (shift < 0)
684
                                    q1 = sample << (-shift);
685
                                else
686
                                    q1 = sample >> shift;
687
                                q1 = (q1 * mult) >> P;
688
                                q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
689
                            }
690
#endif
691
                            if (q[m] >= steps)
692
                                q[m] = steps - 1;
693
                            assert(q[m] >= 0 && q[m] < steps);
694
                        }
695
                        bits = quant_bits[qindex];
696
                        if (bits < 0) {
697
                            /* group the 3 values to save bits */
698
                            put_bits(p, -bits, 
699
                                     q[0] + steps * (q[1] + steps * q[2]));
700
#if 0
701
                            printf("%d: gr1 %d\n", 
702
                                   i, q[0] + steps * (q[1] + steps * q[2]));
703
#endif
704
                        } else {
705
#if 0
706
                            printf("%d: gr3 %d %d %d\n", 
707
                                   i, q[0], q[1], q[2]);
708
#endif                               
709
                            put_bits(p, bits, q[0]);
710
                            put_bits(p, bits, q[1]);
711
                            put_bits(p, bits, q[2]);
712
                        }
713
                    }
714
                }
715
                /* next subband in alloc table */
716
                j += 1 << bit_alloc_bits; 
717
            }
718
        }
719
    }
720

    
721
    /* padding */
722
    for(i=0;i<padding;i++)
723
        put_bits(p, 1, 0);
724

    
725
    /* flush */
726
    flush_put_bits(p);
727
}
728

    
729
int MPA_encode_frame(AVCodecContext *avctx,
730
                     unsigned char *frame, int buf_size, void *data)
731
{
732
    MpegAudioContext *s = avctx->priv_data;
733
    short *samples = data;
734
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
735
    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
736
    int padding, i;
737

    
738
    for(i=0;i<s->nb_channels;i++) {
739
        filter(s, i, samples + i, s->nb_channels);
740
    }
741

    
742
    for(i=0;i<s->nb_channels;i++) {
743
        compute_scale_factors(s->scale_code[i], s->scale_factors[i], 
744
                              s->sb_samples[i], s->sblimit);
745
    }
746
    for(i=0;i<s->nb_channels;i++) {
747
        psycho_acoustic_model(s, smr[i]);
748
    }
749
    compute_bit_allocation(s, smr, bit_alloc, &padding);
750

    
751
    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
752

    
753
    encode_frame(s, bit_alloc, padding);
754
    
755
    s->nb_samples += MPA_FRAME_SIZE;
756
    return s->pb.buf_ptr - s->pb.buf;
757
}
758

    
759

    
760
AVCodec mp2_encoder = {
761
    "mp2",
762
    CODEC_TYPE_AUDIO,
763
    CODEC_ID_MP2,
764
    sizeof(MpegAudioContext),
765
    MPA_encode_init,
766
    MPA_encode_frame,
767
    NULL,
768
};