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/*
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 * 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
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 * it under the terms of the GNU General Public License as published by
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 * the Free Software Foundation; either version 2 of the License, or
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 * (at your option) any later version.
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 *
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 * This program is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
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 * GNU General Public License for more details.
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 *
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 * You should have received a copy of the GNU General Public License
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 * along with this program; if not, write to the Free Software
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 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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 */
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#include <stdlib.h>
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#include <stdio.h>
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#include <string.h>
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#include <math.h>
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#include "avcodec.h"
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#include "mpegaudio.h"
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26
#define NDEBUG
27
#include <assert.h>
28

    
29
/* define it to use floats in quantization (I don't like floats !) */
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//#define USE_FLOATS
31

    
32
#define MPA_STEREO  0
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#define MPA_JSTEREO 1
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#define MPA_DUAL    2
35
#define MPA_MONO    3
36

    
37
#include "mpegaudiotab.h"
38

    
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int MPA_encode_init(AVCodecContext *avctx)
40
{
41
    MpegAudioContext *s = avctx->priv_data;
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    int freq = avctx->sample_rate;
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    int bitrate = avctx->bit_rate;
44
    int channels = avctx->channels;
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    int i, v, table, ch_bitrate;
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    float a;
47

    
48
    if (channels > 2)
49
        return -1;
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    bitrate = bitrate / 1000;
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    s->nb_channels = channels;
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    s->freq = freq;
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    s->bit_rate = bitrate * 1000;
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    avctx->frame_size = MPA_FRAME_SIZE;
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    avctx->key_frame = 1; /* always key frame */
56

    
57
    /* encoding freq */
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    s->lsf = 0;
59
    for(i=0;i<3;i++) {
60
        if (freq_tab[i] == freq) 
61
            break;
62
        if ((freq_tab[i] / 2) == freq) {
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            s->lsf = 1;
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            break;
65
        }
66
    }
67
    if (i == 3)
68
        return -1;
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    s->freq_index = i;
70

    
71
    /* encoding bitrate & frequency */
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    for(i=0;i<15;i++) {
73
        if (bitrate_tab[1-s->lsf][i] == bitrate) 
74
            break;
75
    }
76
    if (i == 15)
77
        return -1;
78
    s->bitrate_index = i;
79

    
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    /* compute total header size & pad bit */
81
    
82
    a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
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    s->frame_size = ((int)a) * 8;
84

    
85
    /* frame fractional size to compute padding */
86
    s->frame_frac = 0;
87
    s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
88
    
89
    /* select the right allocation table */
90
    ch_bitrate = bitrate / s->nb_channels;
91
    if (!s->lsf) {
92
        if ((freq == 48000 && ch_bitrate >= 56) ||
93
            (ch_bitrate >= 56 && ch_bitrate <= 80)) 
94
            table = 0;
95
        else if (freq != 48000 && ch_bitrate >= 96) 
96
            table = 1;
97
        else if (freq != 32000 && ch_bitrate <= 48) 
98
            table = 2;
99
        else 
100
            table = 3;
101
    } else {
102
        table = 4;
103
    }
104
    /* number of used subbands */
105
    s->sblimit = sblimit_table[table];
106
    s->alloc_table = alloc_tables[table];
107

    
108
#ifdef DEBUG
109
    printf("%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", 
110
           bitrate, freq, s->frame_size, table, s->frame_frac_incr);
111
#endif
112

    
113
    for(i=0;i<s->nb_channels;i++)
114
        s->samples_offset[i] = 0;
115

    
116
    for(i=0;i<512;i++) {
117
        float a = enwindow[i] * 32768.0 * 16.0;
118
        filter_bank[i] = (int)(a);
119
    }
120
    for(i=0;i<64;i++) {
121
        v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
122
        if (v <= 0)
123
            v = 1;
124
        scale_factor_table[i] = v;
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#ifdef USE_FLOATS
126
        scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
127
#else
128
#define P 15
129
        scale_factor_shift[i] = 21 - P - (i / 3);
130
        scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
131
#endif
132
    }
133
    for(i=0;i<128;i++) {
134
        v = i - 64;
135
        if (v <= -3)
136
            v = 0;
137
        else if (v < 0)
138
            v = 1;
139
        else if (v == 0)
140
            v = 2;
141
        else if (v < 3)
142
            v = 3;
143
        else 
144
            v = 4;
145
        scale_diff_table[i] = v;
146
    }
147

    
148
    for(i=0;i<17;i++) {
149
        v = quant_bits[i];
150
        if (v < 0) 
151
            v = -v;
152
        else
153
            v = v * 3;
154
        total_quant_bits[i] = 12 * v;
155
    }
156

    
157
    return 0;
158
}
159

    
160
/* 32 point floating point IDCT */
161
static void idct32(int *out, int *tab, int sblimit, int left_shift)
162
{
163
    int i, j;
164
    int *t, *t1, xr;
165
    const int *xp = costab32;
166

    
167
    for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
168
    
169
    t = tab + 30;
170
    t1 = tab + 2;
171
    do {
172
        t[0] += t[-4];
173
        t[1] += t[1 - 4];
174
        t -= 4;
175
    } while (t != t1);
176

    
177
    t = tab + 28;
178
    t1 = tab + 4;
179
    do {
180
        t[0] += t[-8];
181
        t[1] += t[1-8];
182
        t[2] += t[2-8];
183
        t[3] += t[3-8];
184
        t -= 8;
185
    } while (t != t1);
186
    
187
    t = tab;
188
    t1 = tab + 32;
189
    do {
190
        t[ 3] = -t[ 3];    
191
        t[ 6] = -t[ 6];    
192
        
193
        t[11] = -t[11];    
194
        t[12] = -t[12];    
195
        t[13] = -t[13];    
196
        t[15] = -t[15]; 
197
        t += 16;
198
    } while (t != t1);
199

    
200
    
201
    t = tab;
202
    t1 = tab + 8;
203
    do {
204
        int x1, x2, x3, x4;
205
        
206
        x3 = MUL(t[16], FIX(SQRT2*0.5));
207
        x4 = t[0] - x3;
208
        x3 = t[0] + x3;
209
        
210
        x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
211
        x1 = MUL((t[8] - x2), xp[0]);
212
        x2 = MUL((t[8] + x2), xp[1]);
213

    
214
        t[ 0] = x3 + x1;
215
        t[ 8] = x4 - x2;
216
        t[16] = x4 + x2;
217
        t[24] = x3 - x1;
218
        t++;
219
    } while (t != t1);
220

    
221
    xp += 2;
222
    t = tab;
223
    t1 = tab + 4;
224
    do {
225
        xr = MUL(t[28],xp[0]);
226
        t[28] = (t[0] - xr);
227
        t[0] = (t[0] + xr);
228

    
229
        xr = MUL(t[4],xp[1]);
230
        t[ 4] = (t[24] - xr);
231
        t[24] = (t[24] + xr);
232
        
233
        xr = MUL(t[20],xp[2]);
234
        t[20] = (t[8] - xr);
235
        t[ 8] = (t[8] + xr);
236
            
237
        xr = MUL(t[12],xp[3]);
238
        t[12] = (t[16] - xr);
239
        t[16] = (t[16] + xr);
240
        t++;
241
    } while (t != t1);
242
    xp += 4;
243

    
244
    for (i = 0; i < 4; i++) {
245
        xr = MUL(tab[30-i*4],xp[0]);
246
        tab[30-i*4] = (tab[i*4] - xr);
247
        tab[   i*4] = (tab[i*4] + xr);
248
        
249
        xr = MUL(tab[ 2+i*4],xp[1]);
250
        tab[ 2+i*4] = (tab[28-i*4] - xr);
251
        tab[28-i*4] = (tab[28-i*4] + xr);
252
        
253
        xr = MUL(tab[31-i*4],xp[0]);
254
        tab[31-i*4] = (tab[1+i*4] - xr);
255
        tab[ 1+i*4] = (tab[1+i*4] + xr);
256
        
257
        xr = MUL(tab[ 3+i*4],xp[1]);
258
        tab[ 3+i*4] = (tab[29-i*4] - xr);
259
        tab[29-i*4] = (tab[29-i*4] + xr);
260
        
261
        xp += 2;
262
    }
263

    
264
    t = tab + 30;
265
    t1 = tab + 1;
266
    do {
267
        xr = MUL(t1[0], *xp);
268
        t1[0] = (t[0] - xr);
269
        t[0] = (t[0] + xr);
270
        t -= 2;
271
        t1 += 2;
272
        xp++;
273
    } while (t >= tab);
274

    
275
    for(i=0;i<32;i++) {
276
        out[i] = tab[bitinv32[i]] << left_shift;
277
    }
278
}
279

    
280
static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
281
{
282
    short *p, *q;
283
    int sum, offset, i, j, norm, n;
284
    short tmp[64];
285
    int tmp1[32];
286
    int *out;
287

    
288
    //    print_pow1(samples, 1152);
289

    
290
    offset = s->samples_offset[ch];
291
    out = &s->sb_samples[ch][0][0][0];
292
    for(j=0;j<36;j++) {
293
        /* 32 samples at once */
294
        for(i=0;i<32;i++) {
295
            s->samples_buf[ch][offset + (31 - i)] = samples[0];
296
            samples += incr;
297
        }
298

    
299
        /* filter */
300
        p = s->samples_buf[ch] + offset;
301
        q = filter_bank;
302
        /* maxsum = 23169 */
303
        for(i=0;i<64;i++) {
304
            sum = p[0*64] * q[0*64];
305
            sum += p[1*64] * q[1*64];
306
            sum += p[2*64] * q[2*64];
307
            sum += p[3*64] * q[3*64];
308
            sum += p[4*64] * q[4*64];
309
            sum += p[5*64] * q[5*64];
310
            sum += p[6*64] * q[6*64];
311
            sum += p[7*64] * q[7*64];
312
            tmp[i] = sum >> 14;
313
            p++;
314
            q++;
315
        }
316
        tmp1[0] = tmp[16];
317
        for( i=1; i<=16; i++ ) tmp1[i] = tmp[i+16]+tmp[16-i];
318
        for( i=17; i<=31; i++ ) tmp1[i] = tmp[i+16]-tmp[80-i];
319

    
320
        /* integer IDCT 32 with normalization. XXX: There may be some
321
           overflow left */
322
        norm = 0;
323
        for(i=0;i<32;i++) {
324
            norm |= abs(tmp1[i]);
325
        }
326
        n = log2(norm) - 12;
327
        if (n > 0) {
328
            for(i=0;i<32;i++) 
329
                tmp1[i] >>= n;
330
        } else {
331
            n = 0;
332
        }
333

    
334
        idct32(out, tmp1, s->sblimit, n);
335

    
336
        /* advance of 32 samples */
337
        offset -= 32;
338
        out += 32;
339
        /* handle the wrap around */
340
        if (offset < 0) {
341
            memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32), 
342
                    s->samples_buf[ch], (512 - 32) * 2);
343
            offset = SAMPLES_BUF_SIZE - 512;
344
        }
345
    }
346
    s->samples_offset[ch] = offset;
347

    
348
    //    print_pow(s->sb_samples, 1152);
349
}
350

    
351
static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
352
                                  unsigned char scale_factors[SBLIMIT][3], 
353
                                  int sb_samples[3][12][SBLIMIT],
354
                                  int sblimit)
355
{
356
    int *p, vmax, v, n, i, j, k, code;
357
    int index, d1, d2;
358
    unsigned char *sf = &scale_factors[0][0];
359
    
360
    for(j=0;j<sblimit;j++) {
361
        for(i=0;i<3;i++) {
362
            /* find the max absolute value */
363
            p = &sb_samples[i][0][j];
364
            vmax = abs(*p);
365
            for(k=1;k<12;k++) {
366
                p += SBLIMIT;
367
                v = abs(*p);
368
                if (v > vmax)
369
                    vmax = v;
370
            }
371
            /* compute the scale factor index using log 2 computations */
372
            if (vmax > 0) {
373
                n = log2(vmax);
374
                /* n is the position of the MSB of vmax. now 
375
                   use at most 2 compares to find the index */
376
                index = (21 - n) * 3 - 3;
377
                if (index >= 0) {
378
                    while (vmax <= scale_factor_table[index+1])
379
                        index++;
380
                } else {
381
                    index = 0; /* very unlikely case of overflow */
382
                }
383
            } else {
384
                index = 63;
385
            }
386
            
387
#if 0
388
            printf("%2d:%d in=%x %x %d\n", 
389
                   j, i, vmax, scale_factor_table[index], index);
390
#endif
391
            /* store the scale factor */
392
            assert(index >=0 && index <= 63);
393
            sf[i] = index;
394
        }
395

    
396
        /* compute the transmission factor : look if the scale factors
397
           are close enough to each other */
398
        d1 = scale_diff_table[sf[0] - sf[1] + 64];
399
        d2 = scale_diff_table[sf[1] - sf[2] + 64];
400
        
401
        /* handle the 25 cases */
402
        switch(d1 * 5 + d2) {
403
        case 0*5+0:
404
        case 0*5+4:
405
        case 3*5+4:
406
        case 4*5+0:
407
        case 4*5+4:
408
            code = 0;
409
            break;
410
        case 0*5+1:
411
        case 0*5+2:
412
        case 4*5+1:
413
        case 4*5+2:
414
            code = 3;
415
            sf[2] = sf[1];
416
            break;
417
        case 0*5+3:
418
        case 4*5+3:
419
            code = 3;
420
            sf[1] = sf[2];
421
            break;
422
        case 1*5+0:
423
        case 1*5+4:
424
        case 2*5+4:
425
            code = 1;
426
            sf[1] = sf[0];
427
            break;
428
        case 1*5+1:
429
        case 1*5+2:
430
        case 2*5+0:
431
        case 2*5+1:
432
        case 2*5+2:
433
            code = 2;
434
            sf[1] = sf[2] = sf[0];
435
            break;
436
        case 2*5+3:
437
        case 3*5+3:
438
            code = 2;
439
            sf[0] = sf[1] = sf[2];
440
            break;
441
        case 3*5+0:
442
        case 3*5+1:
443
        case 3*5+2:
444
            code = 2;
445
            sf[0] = sf[2] = sf[1];
446
            break;
447
        case 1*5+3:
448
            code = 2;
449
            if (sf[0] > sf[2])
450
              sf[0] = sf[2];
451
            sf[1] = sf[2] = sf[0];
452
            break;
453
        default:
454
            abort();
455
        }
456
        
457
#if 0
458
        printf("%d: %2d %2d %2d %d %d -> %d\n", j, 
459
               sf[0], sf[1], sf[2], d1, d2, code);
460
#endif
461
        scale_code[j] = code;
462
        sf += 3;
463
    }
464
}
465

    
466
/* The most important function : psycho acoustic module. In this
467
   encoder there is basically none, so this is the worst you can do,
468
   but also this is the simpler. */
469
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
470
{
471
    int i;
472

    
473
    for(i=0;i<s->sblimit;i++) {
474
        smr[i] = (int)(fixed_smr[i] * 10);
475
    }
476
}
477

    
478

    
479
#define SB_NOTALLOCATED  0
480
#define SB_ALLOCATED     1
481
#define SB_NOMORE        2
482

    
483
/* Try to maximize the smr while using a number of bits inferior to
484
   the frame size. I tried to make the code simpler, faster and
485
   smaller than other encoders :-) */
486
static void compute_bit_allocation(MpegAudioContext *s, 
487
                                   short smr1[MPA_MAX_CHANNELS][SBLIMIT],
488
                                   unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
489
                                   int *padding)
490
{
491
    int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
492
    int incr;
493
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
494
    unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
495
    const unsigned char *alloc;
496

    
497
    memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
498
    memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
499
    memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
500
    
501
    /* compute frame size and padding */
502
    max_frame_size = s->frame_size;
503
    s->frame_frac += s->frame_frac_incr;
504
    if (s->frame_frac >= 65536) {
505
        s->frame_frac -= 65536;
506
        s->do_padding = 1;
507
        max_frame_size += 8;
508
    } else {
509
        s->do_padding = 0;
510
    }
511

    
512
    /* compute the header + bit alloc size */
513
    current_frame_size = 32;
514
    alloc = s->alloc_table;
515
    for(i=0;i<s->sblimit;i++) {
516
        incr = alloc[0];
517
        current_frame_size += incr * s->nb_channels;
518
        alloc += 1 << incr;
519
    }
520
    for(;;) {
521
        /* look for the subband with the largest signal to mask ratio */
522
        max_sb = -1;
523
        max_ch = -1;
524
        max_smr = 0x80000000;
525
        for(ch=0;ch<s->nb_channels;ch++) {
526
            for(i=0;i<s->sblimit;i++) {
527
                if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
528
                    max_smr = smr[ch][i];
529
                    max_sb = i;
530
                    max_ch = ch;
531
                }
532
            }
533
        }
534
#if 0
535
        printf("current=%d max=%d max_sb=%d alloc=%d\n", 
536
               current_frame_size, max_frame_size, max_sb,
537
               bit_alloc[max_sb]);
538
#endif        
539
        if (max_sb < 0)
540
            break;
541
        
542
        /* find alloc table entry (XXX: not optimal, should use
543
           pointer table) */
544
        alloc = s->alloc_table;
545
        for(i=0;i<max_sb;i++) {
546
            alloc += 1 << alloc[0];
547
        }
548

    
549
        if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
550
            /* nothing was coded for this band: add the necessary bits */
551
            incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
552
            incr += total_quant_bits[alloc[1]];
553
        } else {
554
            /* increments bit allocation */
555
            b = bit_alloc[max_ch][max_sb];
556
            incr = total_quant_bits[alloc[b + 1]] - 
557
                total_quant_bits[alloc[b]];
558
        }
559

    
560
        if (current_frame_size + incr <= max_frame_size) {
561
            /* can increase size */
562
            b = ++bit_alloc[max_ch][max_sb];
563
            current_frame_size += incr;
564
            /* decrease smr by the resolution we added */
565
            smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
566
            /* max allocation size reached ? */
567
            if (b == ((1 << alloc[0]) - 1))
568
                subband_status[max_ch][max_sb] = SB_NOMORE;
569
            else
570
                subband_status[max_ch][max_sb] = SB_ALLOCATED;
571
        } else {
572
            /* cannot increase the size of this subband */
573
            subband_status[max_ch][max_sb] = SB_NOMORE;
574
        }
575
    }
576
    *padding = max_frame_size - current_frame_size;
577
    assert(*padding >= 0);
578

    
579
#if 0
580
    for(i=0;i<s->sblimit;i++) {
581
        printf("%d ", bit_alloc[i]);
582
    }
583
    printf("\n");
584
#endif
585
}
586

    
587
/*
588
 * Output the mpeg audio layer 2 frame. Note how the code is small
589
 * compared to other encoders :-)
590
 */
591
static void encode_frame(MpegAudioContext *s,
592
                         unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
593
                         int padding)
594
{
595
    int i, j, k, l, bit_alloc_bits, b, ch;
596
    unsigned char *sf;
597
    int q[3];
598
    PutBitContext *p = &s->pb;
599

    
600
    /* header */
601

    
602
    put_bits(p, 12, 0xfff);
603
    put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
604
    put_bits(p, 2, 4-2);  /* layer 2 */
605
    put_bits(p, 1, 1); /* no error protection */
606
    put_bits(p, 4, s->bitrate_index);
607
    put_bits(p, 2, s->freq_index);
608
    put_bits(p, 1, s->do_padding); /* use padding */
609
    put_bits(p, 1, 0);             /* private_bit */
610
    put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
611
    put_bits(p, 2, 0); /* mode_ext */
612
    put_bits(p, 1, 0); /* no copyright */
613
    put_bits(p, 1, 1); /* original */
614
    put_bits(p, 2, 0); /* no emphasis */
615

    
616
    /* bit allocation */
617
    j = 0;
618
    for(i=0;i<s->sblimit;i++) {
619
        bit_alloc_bits = s->alloc_table[j];
620
        for(ch=0;ch<s->nb_channels;ch++) {
621
            put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
622
        }
623
        j += 1 << bit_alloc_bits;
624
    }
625
    
626
    /* scale codes */
627
    for(i=0;i<s->sblimit;i++) {
628
        for(ch=0;ch<s->nb_channels;ch++) {
629
            if (bit_alloc[ch][i]) 
630
                put_bits(p, 2, s->scale_code[ch][i]);
631
        }
632
    }
633

    
634
    /* scale factors */
635
    for(i=0;i<s->sblimit;i++) {
636
        for(ch=0;ch<s->nb_channels;ch++) {
637
            if (bit_alloc[ch][i]) {
638
                sf = &s->scale_factors[ch][i][0];
639
                switch(s->scale_code[ch][i]) {
640
                case 0:
641
                    put_bits(p, 6, sf[0]);
642
                    put_bits(p, 6, sf[1]);
643
                    put_bits(p, 6, sf[2]);
644
                    break;
645
                case 3:
646
                case 1:
647
                    put_bits(p, 6, sf[0]);
648
                    put_bits(p, 6, sf[2]);
649
                    break;
650
                case 2:
651
                    put_bits(p, 6, sf[0]);
652
                    break;
653
                }
654
            }
655
        }
656
    }
657
    
658
    /* quantization & write sub band samples */
659

    
660
    for(k=0;k<3;k++) {
661
        for(l=0;l<12;l+=3) {
662
            j = 0;
663
            for(i=0;i<s->sblimit;i++) {
664
                bit_alloc_bits = s->alloc_table[j];
665
                for(ch=0;ch<s->nb_channels;ch++) {
666
                    b = bit_alloc[ch][i];
667
                    if (b) {
668
                        int qindex, steps, m, sample, bits;
669
                        /* we encode 3 sub band samples of the same sub band at a time */
670
                        qindex = s->alloc_table[j+b];
671
                        steps = quant_steps[qindex];
672
                        for(m=0;m<3;m++) {
673
                            sample = s->sb_samples[ch][k][l + m][i];
674
                            /* divide by scale factor */
675
#ifdef USE_FLOATS
676
                            {
677
                                float a;
678
                                a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
679
                                q[m] = (int)((a + 1.0) * steps * 0.5);
680
                            }
681
#else
682
                            {
683
                                int q1, e, shift, mult;
684
                                e = s->scale_factors[ch][i][k];
685
                                shift = scale_factor_shift[e];
686
                                mult = scale_factor_mult[e];
687
                                
688
                                /* normalize to P bits */
689
                                if (shift < 0)
690
                                    q1 = sample << (-shift);
691
                                else
692
                                    q1 = sample >> shift;
693
                                q1 = (q1 * mult) >> P;
694
                                q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
695
                            }
696
#endif
697
                            if (q[m] >= steps)
698
                                q[m] = steps - 1;
699
                            assert(q[m] >= 0 && q[m] < steps);
700
                        }
701
                        bits = quant_bits[qindex];
702
                        if (bits < 0) {
703
                            /* group the 3 values to save bits */
704
                            put_bits(p, -bits, 
705
                                     q[0] + steps * (q[1] + steps * q[2]));
706
#if 0
707
                            printf("%d: gr1 %d\n", 
708
                                   i, q[0] + steps * (q[1] + steps * q[2]));
709
#endif
710
                        } else {
711
#if 0
712
                            printf("%d: gr3 %d %d %d\n", 
713
                                   i, q[0], q[1], q[2]);
714
#endif                               
715
                            put_bits(p, bits, q[0]);
716
                            put_bits(p, bits, q[1]);
717
                            put_bits(p, bits, q[2]);
718
                        }
719
                    }
720
                }
721
                /* next subband in alloc table */
722
                j += 1 << bit_alloc_bits; 
723
            }
724
        }
725
    }
726

    
727
    /* padding */
728
    for(i=0;i<padding;i++)
729
        put_bits(p, 1, 0);
730

    
731
    /* flush */
732
    flush_put_bits(p);
733
}
734

    
735
int MPA_encode_frame(AVCodecContext *avctx,
736
                     unsigned char *frame, int buf_size, void *data)
737
{
738
    MpegAudioContext *s = avctx->priv_data;
739
    short *samples = data;
740
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
741
    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
742
    int padding, i;
743

    
744
    for(i=0;i<s->nb_channels;i++) {
745
        filter(s, i, samples + i, s->nb_channels);
746
    }
747

    
748
    for(i=0;i<s->nb_channels;i++) {
749
        compute_scale_factors(s->scale_code[i], s->scale_factors[i], 
750
                              s->sb_samples[i], s->sblimit);
751
    }
752
    for(i=0;i<s->nb_channels;i++) {
753
        psycho_acoustic_model(s, smr[i]);
754
    }
755
    compute_bit_allocation(s, smr, bit_alloc, &padding);
756

    
757
    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE, NULL, NULL);
758

    
759
    encode_frame(s, bit_alloc, padding);
760
    
761
    s->nb_samples += MPA_FRAME_SIZE;
762
    return s->pb.buf_ptr - s->pb.buf;
763
}
764

    
765

    
766
AVCodec mp2_encoder = {
767
    "mp2",
768
    CODEC_TYPE_AUDIO,
769
    CODEC_ID_MP2,
770
    sizeof(MpegAudioContext),
771
    MPA_encode_init,
772
    MPA_encode_frame,
773
    NULL,
774
};