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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
 *
5 ff4ec49e Fabrice Bellard
 * 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
16
 * 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 983e3246 Michael Niedermayer
 
20
/**
21
 * @file mpegaudio.c
22
 * The simplest mpeg audio layer 2 encoder.
23
 */
24
 
25 de6d9b64 Fabrice Bellard
#include "avcodec.h"
26
#include "mpegaudio.h"
27
28 afa982fd Fabrice Bellard
/* currently, cannot change these constants (need to modify
29
   quantization stage) */
30
#define FRAC_BITS 15
31
#define WFRAC_BITS  14
32 0c1a9eda Zdenek Kabelac
#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
33 afa982fd Fabrice Bellard
#define FIX(a)   ((int)((a) * (1 << FRAC_BITS)))
34 2456e28d Fabrice Bellard
35
#define SAMPLES_BUF_SIZE 4096
36
37
typedef struct MpegAudioContext {
38
    PutBitContext pb;
39
    int nb_channels;
40
    int freq, bit_rate;
41
    int lsf;           /* 1 if mpeg2 low bitrate selected */
42
    int bitrate_index; /* bit rate */
43
    int freq_index;
44
    int frame_size; /* frame size, in bits, without padding */
45 0c1a9eda Zdenek Kabelac
    int64_t nb_samples; /* total number of samples encoded */
46 2456e28d Fabrice Bellard
    /* padding computation */
47
    int frame_frac, frame_frac_incr, do_padding;
48
    short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
49
    int samples_offset[MPA_MAX_CHANNELS];       /* offset in samples_buf */
50
    int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
51
    unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
52
    /* code to group 3 scale factors */
53
    unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];       
54
    int sblimit; /* number of used subbands */
55
    const unsigned char *alloc_table;
56
} MpegAudioContext;
57
58 de6d9b64 Fabrice Bellard
/* define it to use floats in quantization (I don't like floats !) */
59
//#define USE_FLOATS
60
61
#include "mpegaudiotab.h"
62
63 5c91a675 Zdenek Kabelac
static int MPA_encode_init(AVCodecContext *avctx)
64 de6d9b64 Fabrice Bellard
{
65
    MpegAudioContext *s = avctx->priv_data;
66
    int freq = avctx->sample_rate;
67
    int bitrate = avctx->bit_rate;
68
    int channels = avctx->channels;
69 2456e28d Fabrice Bellard
    int i, v, table;
70 de6d9b64 Fabrice Bellard
    float a;
71
72
    if (channels > 2)
73
        return -1;
74
    bitrate = bitrate / 1000;
75
    s->nb_channels = channels;
76
    s->freq = freq;
77
    s->bit_rate = bitrate * 1000;
78
    avctx->frame_size = MPA_FRAME_SIZE;
79
80
    /* encoding freq */
81
    s->lsf = 0;
82
    for(i=0;i<3;i++) {
83 2456e28d Fabrice Bellard
        if (mpa_freq_tab[i] == freq) 
84 de6d9b64 Fabrice Bellard
            break;
85 2456e28d Fabrice Bellard
        if ((mpa_freq_tab[i] / 2) == freq) {
86 de6d9b64 Fabrice Bellard
            s->lsf = 1;
87
            break;
88
        }
89
    }
90
    if (i == 3)
91
        return -1;
92
    s->freq_index = i;
93
94
    /* encoding bitrate & frequency */
95
    for(i=0;i<15;i++) {
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        if (mpa_bitrate_tab[s->lsf][1][i] == bitrate) 
97 de6d9b64 Fabrice Bellard
            break;
98
    }
99
    if (i == 15)
100
        return -1;
101
    s->bitrate_index = i;
102
103
    /* compute total header size & pad bit */
104
    
105
    a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
106
    s->frame_size = ((int)a) * 8;
107
108
    /* frame fractional size to compute padding */
109
    s->frame_frac = 0;
110
    s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
111
    
112
    /* select the right allocation table */
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    table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
114
115 de6d9b64 Fabrice Bellard
    /* number of used subbands */
116
    s->sblimit = sblimit_table[table];
117
    s->alloc_table = alloc_tables[table];
118
119
#ifdef DEBUG
120 3d0ef6dd Michael Niedermayer
    av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", 
121 de6d9b64 Fabrice Bellard
           bitrate, freq, s->frame_size, table, s->frame_frac_incr);
122
#endif
123
124
    for(i=0;i<s->nb_channels;i++)
125
        s->samples_offset[i] = 0;
126
127 2456e28d Fabrice Bellard
    for(i=0;i<257;i++) {
128
        int v;
129 afa982fd Fabrice Bellard
        v = mpa_enwindow[i];
130
#if WFRAC_BITS != 16
131
        v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
132
#endif
133 2456e28d Fabrice Bellard
        filter_bank[i] = v;
134
        if ((i & 63) != 0)
135
            v = -v;
136
        if (i != 0)
137
            filter_bank[512 - i] = v;
138 de6d9b64 Fabrice Bellard
    }
139 2456e28d Fabrice Bellard
140 de6d9b64 Fabrice Bellard
    for(i=0;i<64;i++) {
141
        v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
142
        if (v <= 0)
143
            v = 1;
144
        scale_factor_table[i] = v;
145
#ifdef USE_FLOATS
146
        scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
147
#else
148
#define P 15
149
        scale_factor_shift[i] = 21 - P - (i / 3);
150
        scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
151
#endif
152
    }
153
    for(i=0;i<128;i++) {
154
        v = i - 64;
155
        if (v <= -3)
156
            v = 0;
157
        else if (v < 0)
158
            v = 1;
159
        else if (v == 0)
160
            v = 2;
161
        else if (v < 3)
162
            v = 3;
163
        else 
164
            v = 4;
165
        scale_diff_table[i] = v;
166
    }
167
168
    for(i=0;i<17;i++) {
169
        v = quant_bits[i];
170
        if (v < 0) 
171
            v = -v;
172
        else
173
            v = v * 3;
174
        total_quant_bits[i] = 12 * v;
175
    }
176
177 492cd3a9 Michael Niedermayer
    avctx->coded_frame= avcodec_alloc_frame();
178
    avctx->coded_frame->key_frame= 1;
179
180 de6d9b64 Fabrice Bellard
    return 0;
181
}
182
183 2456e28d Fabrice Bellard
/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
184 afa982fd Fabrice Bellard
static void idct32(int *out, int *tab)
185 de6d9b64 Fabrice Bellard
{
186
    int i, j;
187
    int *t, *t1, xr;
188
    const int *xp = costab32;
189
190
    for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
191
    
192
    t = tab + 30;
193
    t1 = tab + 2;
194
    do {
195
        t[0] += t[-4];
196
        t[1] += t[1 - 4];
197
        t -= 4;
198
    } while (t != t1);
199
200
    t = tab + 28;
201
    t1 = tab + 4;
202
    do {
203
        t[0] += t[-8];
204
        t[1] += t[1-8];
205
        t[2] += t[2-8];
206
        t[3] += t[3-8];
207
        t -= 8;
208
    } while (t != t1);
209
    
210
    t = tab;
211
    t1 = tab + 32;
212
    do {
213
        t[ 3] = -t[ 3];    
214
        t[ 6] = -t[ 6];    
215
        
216
        t[11] = -t[11];    
217
        t[12] = -t[12];    
218
        t[13] = -t[13];    
219
        t[15] = -t[15]; 
220
        t += 16;
221
    } while (t != t1);
222
223
    
224
    t = tab;
225
    t1 = tab + 8;
226
    do {
227
        int x1, x2, x3, x4;
228
        
229
        x3 = MUL(t[16], FIX(SQRT2*0.5));
230
        x4 = t[0] - x3;
231
        x3 = t[0] + x3;
232
        
233
        x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
234
        x1 = MUL((t[8] - x2), xp[0]);
235
        x2 = MUL((t[8] + x2), xp[1]);
236
237
        t[ 0] = x3 + x1;
238
        t[ 8] = x4 - x2;
239
        t[16] = x4 + x2;
240
        t[24] = x3 - x1;
241
        t++;
242
    } while (t != t1);
243
244
    xp += 2;
245
    t = tab;
246
    t1 = tab + 4;
247
    do {
248
        xr = MUL(t[28],xp[0]);
249
        t[28] = (t[0] - xr);
250
        t[0] = (t[0] + xr);
251
252
        xr = MUL(t[4],xp[1]);
253
        t[ 4] = (t[24] - xr);
254
        t[24] = (t[24] + xr);
255
        
256
        xr = MUL(t[20],xp[2]);
257
        t[20] = (t[8] - xr);
258
        t[ 8] = (t[8] + xr);
259
            
260
        xr = MUL(t[12],xp[3]);
261
        t[12] = (t[16] - xr);
262
        t[16] = (t[16] + xr);
263
        t++;
264
    } while (t != t1);
265
    xp += 4;
266
267
    for (i = 0; i < 4; i++) {
268
        xr = MUL(tab[30-i*4],xp[0]);
269
        tab[30-i*4] = (tab[i*4] - xr);
270
        tab[   i*4] = (tab[i*4] + xr);
271
        
272
        xr = MUL(tab[ 2+i*4],xp[1]);
273
        tab[ 2+i*4] = (tab[28-i*4] - xr);
274
        tab[28-i*4] = (tab[28-i*4] + xr);
275
        
276
        xr = MUL(tab[31-i*4],xp[0]);
277
        tab[31-i*4] = (tab[1+i*4] - xr);
278
        tab[ 1+i*4] = (tab[1+i*4] + xr);
279
        
280
        xr = MUL(tab[ 3+i*4],xp[1]);
281
        tab[ 3+i*4] = (tab[29-i*4] - xr);
282
        tab[29-i*4] = (tab[29-i*4] + xr);
283
        
284
        xp += 2;
285
    }
286
287
    t = tab + 30;
288
    t1 = tab + 1;
289
    do {
290
        xr = MUL(t1[0], *xp);
291
        t1[0] = (t[0] - xr);
292
        t[0] = (t[0] + xr);
293
        t -= 2;
294
        t1 += 2;
295
        xp++;
296
    } while (t >= tab);
297
298
    for(i=0;i<32;i++) {
299 afa982fd Fabrice Bellard
        out[i] = tab[bitinv32[i]];
300 de6d9b64 Fabrice Bellard
    }
301
}
302
303 afa982fd Fabrice Bellard
#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
304
305 de6d9b64 Fabrice Bellard
static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
306
{
307
    short *p, *q;
308 afa982fd Fabrice Bellard
    int sum, offset, i, j;
309
    int tmp[64];
310 de6d9b64 Fabrice Bellard
    int tmp1[32];
311
    int *out;
312
313
    //    print_pow1(samples, 1152);
314
315
    offset = s->samples_offset[ch];
316
    out = &s->sb_samples[ch][0][0][0];
317
    for(j=0;j<36;j++) {
318
        /* 32 samples at once */
319
        for(i=0;i<32;i++) {
320
            s->samples_buf[ch][offset + (31 - i)] = samples[0];
321
            samples += incr;
322
        }
323
324
        /* filter */
325
        p = s->samples_buf[ch] + offset;
326
        q = filter_bank;
327
        /* maxsum = 23169 */
328
        for(i=0;i<64;i++) {
329
            sum = p[0*64] * q[0*64];
330
            sum += p[1*64] * q[1*64];
331
            sum += p[2*64] * q[2*64];
332
            sum += p[3*64] * q[3*64];
333
            sum += p[4*64] * q[4*64];
334
            sum += p[5*64] * q[5*64];
335
            sum += p[6*64] * q[6*64];
336
            sum += p[7*64] * q[7*64];
337 afa982fd Fabrice Bellard
            tmp[i] = sum;
338 de6d9b64 Fabrice Bellard
            p++;
339
            q++;
340
        }
341 afa982fd Fabrice Bellard
        tmp1[0] = tmp[16] >> WSHIFT;
342
        for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
343
        for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
344 de6d9b64 Fabrice Bellard
345 afa982fd Fabrice Bellard
        idct32(out, tmp1);
346 de6d9b64 Fabrice Bellard
347
        /* advance of 32 samples */
348
        offset -= 32;
349
        out += 32;
350
        /* handle the wrap around */
351
        if (offset < 0) {
352
            memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32), 
353
                    s->samples_buf[ch], (512 - 32) * 2);
354
            offset = SAMPLES_BUF_SIZE - 512;
355
        }
356
    }
357
    s->samples_offset[ch] = offset;
358
359
    //    print_pow(s->sb_samples, 1152);
360
}
361
362
static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
363
                                  unsigned char scale_factors[SBLIMIT][3], 
364
                                  int sb_samples[3][12][SBLIMIT],
365
                                  int sblimit)
366
{
367
    int *p, vmax, v, n, i, j, k, code;
368
    int index, d1, d2;
369
    unsigned char *sf = &scale_factors[0][0];
370
    
371
    for(j=0;j<sblimit;j++) {
372
        for(i=0;i<3;i++) {
373
            /* find the max absolute value */
374
            p = &sb_samples[i][0][j];
375
            vmax = abs(*p);
376
            for(k=1;k<12;k++) {
377
                p += SBLIMIT;
378
                v = abs(*p);
379
                if (v > vmax)
380
                    vmax = v;
381
            }
382
            /* compute the scale factor index using log 2 computations */
383
            if (vmax > 0) {
384 935442b5 Fabrice Bellard
                n = av_log2(vmax);
385 de6d9b64 Fabrice Bellard
                /* n is the position of the MSB of vmax. now 
386
                   use at most 2 compares to find the index */
387
                index = (21 - n) * 3 - 3;
388
                if (index >= 0) {
389
                    while (vmax <= scale_factor_table[index+1])
390
                        index++;
391
                } else {
392
                    index = 0; /* very unlikely case of overflow */
393
                }
394
            } else {
395 afa982fd Fabrice Bellard
                index = 62; /* value 63 is not allowed */
396 de6d9b64 Fabrice Bellard
            }
397 afa982fd Fabrice Bellard
398 de6d9b64 Fabrice Bellard
#if 0
399
            printf("%2d:%d in=%x %x %d\n", 
400
                   j, i, vmax, scale_factor_table[index], index);
401
#endif
402
            /* store the scale factor */
403
            assert(index >=0 && index <= 63);
404
            sf[i] = index;
405
        }
406
407
        /* compute the transmission factor : look if the scale factors
408
           are close enough to each other */
409
        d1 = scale_diff_table[sf[0] - sf[1] + 64];
410
        d2 = scale_diff_table[sf[1] - sf[2] + 64];
411
        
412
        /* handle the 25 cases */
413
        switch(d1 * 5 + d2) {
414
        case 0*5+0:
415
        case 0*5+4:
416
        case 3*5+4:
417
        case 4*5+0:
418
        case 4*5+4:
419
            code = 0;
420
            break;
421
        case 0*5+1:
422
        case 0*5+2:
423
        case 4*5+1:
424
        case 4*5+2:
425
            code = 3;
426
            sf[2] = sf[1];
427
            break;
428
        case 0*5+3:
429
        case 4*5+3:
430
            code = 3;
431
            sf[1] = sf[2];
432
            break;
433
        case 1*5+0:
434
        case 1*5+4:
435
        case 2*5+4:
436
            code = 1;
437
            sf[1] = sf[0];
438
            break;
439
        case 1*5+1:
440
        case 1*5+2:
441
        case 2*5+0:
442
        case 2*5+1:
443
        case 2*5+2:
444
            code = 2;
445
            sf[1] = sf[2] = sf[0];
446
            break;
447
        case 2*5+3:
448
        case 3*5+3:
449
            code = 2;
450
            sf[0] = sf[1] = sf[2];
451
            break;
452
        case 3*5+0:
453
        case 3*5+1:
454
        case 3*5+2:
455
            code = 2;
456
            sf[0] = sf[2] = sf[1];
457
            break;
458
        case 1*5+3:
459
            code = 2;
460
            if (sf[0] > sf[2])
461
              sf[0] = sf[2];
462
            sf[1] = sf[2] = sf[0];
463
            break;
464
        default:
465 02ac3136 Philip Gladstone
            av_abort();
466 de6d9b64 Fabrice Bellard
        }
467
        
468
#if 0
469
        printf("%d: %2d %2d %2d %d %d -> %d\n", j, 
470
               sf[0], sf[1], sf[2], d1, d2, code);
471
#endif
472
        scale_code[j] = code;
473
        sf += 3;
474
    }
475
}
476
477
/* The most important function : psycho acoustic module. In this
478
   encoder there is basically none, so this is the worst you can do,
479
   but also this is the simpler. */
480
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
481
{
482
    int i;
483
484
    for(i=0;i<s->sblimit;i++) {
485
        smr[i] = (int)(fixed_smr[i] * 10);
486
    }
487
}
488
489
490
#define SB_NOTALLOCATED  0
491
#define SB_ALLOCATED     1
492
#define SB_NOMORE        2
493
494
/* Try to maximize the smr while using a number of bits inferior to
495
   the frame size. I tried to make the code simpler, faster and
496
   smaller than other encoders :-) */
497
static void compute_bit_allocation(MpegAudioContext *s, 
498
                                   short smr1[MPA_MAX_CHANNELS][SBLIMIT],
499
                                   unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
500
                                   int *padding)
501
{
502
    int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
503
    int incr;
504
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
505
    unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
506
    const unsigned char *alloc;
507
508
    memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
509
    memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
510
    memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
511
    
512
    /* compute frame size and padding */
513
    max_frame_size = s->frame_size;
514
    s->frame_frac += s->frame_frac_incr;
515
    if (s->frame_frac >= 65536) {
516
        s->frame_frac -= 65536;
517
        s->do_padding = 1;
518
        max_frame_size += 8;
519
    } else {
520
        s->do_padding = 0;
521
    }
522
523
    /* compute the header + bit alloc size */
524
    current_frame_size = 32;
525
    alloc = s->alloc_table;
526
    for(i=0;i<s->sblimit;i++) {
527
        incr = alloc[0];
528
        current_frame_size += incr * s->nb_channels;
529
        alloc += 1 << incr;
530
    }
531
    for(;;) {
532
        /* look for the subband with the largest signal to mask ratio */
533
        max_sb = -1;
534
        max_ch = -1;
535
        max_smr = 0x80000000;
536
        for(ch=0;ch<s->nb_channels;ch++) {
537
            for(i=0;i<s->sblimit;i++) {
538
                if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
539
                    max_smr = smr[ch][i];
540
                    max_sb = i;
541
                    max_ch = ch;
542
                }
543
            }
544
        }
545
#if 0
546
        printf("current=%d max=%d max_sb=%d alloc=%d\n", 
547
               current_frame_size, max_frame_size, max_sb,
548
               bit_alloc[max_sb]);
549
#endif        
550
        if (max_sb < 0)
551
            break;
552
        
553
        /* find alloc table entry (XXX: not optimal, should use
554
           pointer table) */
555
        alloc = s->alloc_table;
556
        for(i=0;i<max_sb;i++) {
557
            alloc += 1 << alloc[0];
558
        }
559
560
        if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
561
            /* nothing was coded for this band: add the necessary bits */
562
            incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
563
            incr += total_quant_bits[alloc[1]];
564
        } else {
565
            /* increments bit allocation */
566
            b = bit_alloc[max_ch][max_sb];
567
            incr = total_quant_bits[alloc[b + 1]] - 
568
                total_quant_bits[alloc[b]];
569
        }
570
571
        if (current_frame_size + incr <= max_frame_size) {
572
            /* can increase size */
573
            b = ++bit_alloc[max_ch][max_sb];
574
            current_frame_size += incr;
575
            /* decrease smr by the resolution we added */
576
            smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
577
            /* max allocation size reached ? */
578
            if (b == ((1 << alloc[0]) - 1))
579
                subband_status[max_ch][max_sb] = SB_NOMORE;
580
            else
581
                subband_status[max_ch][max_sb] = SB_ALLOCATED;
582
        } else {
583
            /* cannot increase the size of this subband */
584
            subband_status[max_ch][max_sb] = SB_NOMORE;
585
        }
586
    }
587
    *padding = max_frame_size - current_frame_size;
588
    assert(*padding >= 0);
589
590
#if 0
591
    for(i=0;i<s->sblimit;i++) {
592
        printf("%d ", bit_alloc[i]);
593
    }
594
    printf("\n");
595
#endif
596
}
597
598
/*
599
 * Output the mpeg audio layer 2 frame. Note how the code is small
600
 * compared to other encoders :-)
601
 */
602
static void encode_frame(MpegAudioContext *s,
603
                         unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
604
                         int padding)
605
{
606
    int i, j, k, l, bit_alloc_bits, b, ch;
607
    unsigned char *sf;
608
    int q[3];
609
    PutBitContext *p = &s->pb;
610
611
    /* header */
612
613
    put_bits(p, 12, 0xfff);
614
    put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
615
    put_bits(p, 2, 4-2);  /* layer 2 */
616
    put_bits(p, 1, 1); /* no error protection */
617
    put_bits(p, 4, s->bitrate_index);
618
    put_bits(p, 2, s->freq_index);
619
    put_bits(p, 1, s->do_padding); /* use padding */
620
    put_bits(p, 1, 0);             /* private_bit */
621
    put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
622
    put_bits(p, 2, 0); /* mode_ext */
623
    put_bits(p, 1, 0); /* no copyright */
624
    put_bits(p, 1, 1); /* original */
625
    put_bits(p, 2, 0); /* no emphasis */
626
627
    /* bit allocation */
628
    j = 0;
629
    for(i=0;i<s->sblimit;i++) {
630
        bit_alloc_bits = s->alloc_table[j];
631
        for(ch=0;ch<s->nb_channels;ch++) {
632
            put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
633
        }
634
        j += 1 << bit_alloc_bits;
635
    }
636
    
637
    /* scale codes */
638
    for(i=0;i<s->sblimit;i++) {
639
        for(ch=0;ch<s->nb_channels;ch++) {
640
            if (bit_alloc[ch][i]) 
641
                put_bits(p, 2, s->scale_code[ch][i]);
642
        }
643
    }
644
645
    /* scale factors */
646
    for(i=0;i<s->sblimit;i++) {
647
        for(ch=0;ch<s->nb_channels;ch++) {
648
            if (bit_alloc[ch][i]) {
649
                sf = &s->scale_factors[ch][i][0];
650
                switch(s->scale_code[ch][i]) {
651
                case 0:
652
                    put_bits(p, 6, sf[0]);
653
                    put_bits(p, 6, sf[1]);
654
                    put_bits(p, 6, sf[2]);
655
                    break;
656
                case 3:
657
                case 1:
658
                    put_bits(p, 6, sf[0]);
659
                    put_bits(p, 6, sf[2]);
660
                    break;
661
                case 2:
662
                    put_bits(p, 6, sf[0]);
663
                    break;
664
                }
665
            }
666
        }
667
    }
668
    
669
    /* quantization & write sub band samples */
670
671
    for(k=0;k<3;k++) {
672
        for(l=0;l<12;l+=3) {
673
            j = 0;
674
            for(i=0;i<s->sblimit;i++) {
675
                bit_alloc_bits = s->alloc_table[j];
676
                for(ch=0;ch<s->nb_channels;ch++) {
677
                    b = bit_alloc[ch][i];
678
                    if (b) {
679
                        int qindex, steps, m, sample, bits;
680
                        /* we encode 3 sub band samples of the same sub band at a time */
681
                        qindex = s->alloc_table[j+b];
682
                        steps = quant_steps[qindex];
683
                        for(m=0;m<3;m++) {
684
                            sample = s->sb_samples[ch][k][l + m][i];
685
                            /* divide by scale factor */
686
#ifdef USE_FLOATS
687
                            {
688
                                float a;
689
                                a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
690
                                q[m] = (int)((a + 1.0) * steps * 0.5);
691
                            }
692
#else
693
                            {
694
                                int q1, e, shift, mult;
695
                                e = s->scale_factors[ch][i][k];
696
                                shift = scale_factor_shift[e];
697
                                mult = scale_factor_mult[e];
698
                                
699
                                /* normalize to P bits */
700
                                if (shift < 0)
701
                                    q1 = sample << (-shift);
702
                                else
703
                                    q1 = sample >> shift;
704
                                q1 = (q1 * mult) >> P;
705
                                q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
706
                            }
707
#endif
708
                            if (q[m] >= steps)
709
                                q[m] = steps - 1;
710
                            assert(q[m] >= 0 && q[m] < steps);
711
                        }
712
                        bits = quant_bits[qindex];
713
                        if (bits < 0) {
714
                            /* group the 3 values to save bits */
715
                            put_bits(p, -bits, 
716
                                     q[0] + steps * (q[1] + steps * q[2]));
717
#if 0
718
                            printf("%d: gr1 %d\n", 
719
                                   i, q[0] + steps * (q[1] + steps * q[2]));
720
#endif
721
                        } else {
722
#if 0
723
                            printf("%d: gr3 %d %d %d\n", 
724
                                   i, q[0], q[1], q[2]);
725
#endif                               
726
                            put_bits(p, bits, q[0]);
727
                            put_bits(p, bits, q[1]);
728
                            put_bits(p, bits, q[2]);
729
                        }
730
                    }
731
                }
732
                /* next subband in alloc table */
733
                j += 1 << bit_alloc_bits; 
734
            }
735
        }
736
    }
737
738
    /* padding */
739
    for(i=0;i<padding;i++)
740
        put_bits(p, 1, 0);
741
742
    /* flush */
743
    flush_put_bits(p);
744
}
745
746 5c91a675 Zdenek Kabelac
static int MPA_encode_frame(AVCodecContext *avctx,
747
                            unsigned char *frame, int buf_size, void *data)
748 de6d9b64 Fabrice Bellard
{
749
    MpegAudioContext *s = avctx->priv_data;
750
    short *samples = data;
751
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
752
    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
753
    int padding, i;
754
755
    for(i=0;i<s->nb_channels;i++) {
756
        filter(s, i, samples + i, s->nb_channels);
757
    }
758
759
    for(i=0;i<s->nb_channels;i++) {
760
        compute_scale_factors(s->scale_code[i], s->scale_factors[i], 
761
                              s->sb_samples[i], s->sblimit);
762
    }
763
    for(i=0;i<s->nb_channels;i++) {
764
        psycho_acoustic_model(s, smr[i]);
765
    }
766
    compute_bit_allocation(s, smr, bit_alloc, &padding);
767
768 ed7debda Alex Beregszaszi
    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
769 de6d9b64 Fabrice Bellard
770
    encode_frame(s, bit_alloc, padding);
771
    
772
    s->nb_samples += MPA_FRAME_SIZE;
773 17592475 Michael Niedermayer
    return pbBufPtr(&s->pb) - s->pb.buf;
774 de6d9b64 Fabrice Bellard
}
775
776 492cd3a9 Michael Niedermayer
static int MPA_encode_close(AVCodecContext *avctx)
777
{
778
    av_freep(&avctx->coded_frame);
779 8e1e6f31 Fabrice Bellard
    return 0;
780 492cd3a9 Michael Niedermayer
}
781 de6d9b64 Fabrice Bellard
782
AVCodec mp2_encoder = {
783
    "mp2",
784
    CODEC_TYPE_AUDIO,
785
    CODEC_ID_MP2,
786
    sizeof(MpegAudioContext),
787
    MPA_encode_init,
788
    MPA_encode_frame,
789 492cd3a9 Michael Niedermayer
    MPA_encode_close,
790 de6d9b64 Fabrice Bellard
    NULL,
791
};
792 cd4af68a Zdenek Kabelac
793
#undef FIX