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ffmpeg / libavcodec / mpegaudioenc.c @ dfd2a005

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/*
2
 * The simplest mpeg audio layer 2 encoder
3
 * Copyright (c) 2000, 2001 Fabrice Bellard
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 *
5
 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
8
 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
10
 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
21

    
22
/**
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 * @file
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 * The simplest mpeg audio layer 2 encoder.
25
 */
26

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

    
30
#undef  CONFIG_MPEGAUDIO_HP
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#define CONFIG_MPEGAUDIO_HP 0
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#include "mpegaudio.h"
33

    
34
/* currently, cannot change these constants (need to modify
35
   quantization stage) */
36
#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
37

    
38
#define SAMPLES_BUF_SIZE 4096
39

    
40
typedef struct MpegAudioContext {
41
    PutBitContext pb;
42
    int nb_channels;
43
    int lsf;           /* 1 if mpeg2 low bitrate selected */
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    int bitrate_index; /* bit rate */
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    int freq_index;
46
    int frame_size; /* frame size, in bits, without padding */
47
    /* padding computation */
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    int frame_frac, frame_frac_incr, do_padding;
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    short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
50
    int samples_offset[MPA_MAX_CHANNELS];       /* offset in samples_buf */
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    int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
52
    unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
53
    /* code to group 3 scale factors */
54
    unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
55
    int sblimit; /* number of used subbands */
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    const unsigned char *alloc_table;
57
} MpegAudioContext;
58

    
59
/* define it to use floats in quantization (I don't like floats !) */
60
#define USE_FLOATS
61

    
62
#include "mpegaudiodata.h"
63
#include "mpegaudiotab.h"
64

    
65
static av_cold int MPA_encode_init(AVCodecContext *avctx)
66
{
67
    MpegAudioContext *s = avctx->priv_data;
68
    int freq = avctx->sample_rate;
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    int bitrate = avctx->bit_rate;
70
    int channels = avctx->channels;
71
    int i, v, table;
72
    float a;
73

    
74
    if (channels <= 0 || channels > 2){
75
        av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
76
        return -1;
77
    }
78
    bitrate = bitrate / 1000;
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    s->nb_channels = channels;
80
    avctx->frame_size = MPA_FRAME_SIZE;
81

    
82
    /* encoding freq */
83
    s->lsf = 0;
84
    for(i=0;i<3;i++) {
85
        if (ff_mpa_freq_tab[i] == freq)
86
            break;
87
        if ((ff_mpa_freq_tab[i] / 2) == freq) {
88
            s->lsf = 1;
89
            break;
90
        }
91
    }
92
    if (i == 3){
93
        av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
94
        return -1;
95
    }
96
    s->freq_index = i;
97

    
98
    /* encoding bitrate & frequency */
99
    for(i=0;i<15;i++) {
100
        if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
101
            break;
102
    }
103
    if (i == 15){
104
        av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
105
        return -1;
106
    }
107
    s->bitrate_index = i;
108

    
109
    /* compute total header size & pad bit */
110

    
111
    a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
112
    s->frame_size = ((int)a) * 8;
113

    
114
    /* frame fractional size to compute padding */
115
    s->frame_frac = 0;
116
    s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
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118
    /* select the right allocation table */
119
    table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
120

    
121
    /* number of used subbands */
122
    s->sblimit = ff_mpa_sblimit_table[table];
123
    s->alloc_table = ff_mpa_alloc_tables[table];
124

    
125
    av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
126
            bitrate, freq, s->frame_size, table, s->frame_frac_incr);
127

    
128
    for(i=0;i<s->nb_channels;i++)
129
        s->samples_offset[i] = 0;
130

    
131
    for(i=0;i<257;i++) {
132
        int v;
133
        v = ff_mpa_enwindow[i];
134
#if WFRAC_BITS != 16
135
        v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
136
#endif
137
        filter_bank[i] = v;
138
        if ((i & 63) != 0)
139
            v = -v;
140
        if (i != 0)
141
            filter_bank[512 - i] = v;
142
    }
143

    
144
    for(i=0;i<64;i++) {
145
        v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
146
        if (v <= 0)
147
            v = 1;
148
        scale_factor_table[i] = v;
149
#ifdef USE_FLOATS
150
        scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
151
#else
152
#define P 15
153
        scale_factor_shift[i] = 21 - P - (i / 3);
154
        scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
155
#endif
156
    }
157
    for(i=0;i<128;i++) {
158
        v = i - 64;
159
        if (v <= -3)
160
            v = 0;
161
        else if (v < 0)
162
            v = 1;
163
        else if (v == 0)
164
            v = 2;
165
        else if (v < 3)
166
            v = 3;
167
        else
168
            v = 4;
169
        scale_diff_table[i] = v;
170
    }
171

    
172
    for(i=0;i<17;i++) {
173
        v = ff_mpa_quant_bits[i];
174
        if (v < 0)
175
            v = -v;
176
        else
177
            v = v * 3;
178
        total_quant_bits[i] = 12 * v;
179
    }
180

    
181
    avctx->coded_frame= avcodec_alloc_frame();
182
    avctx->coded_frame->key_frame= 1;
183

    
184
    return 0;
185
}
186

    
187
/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
188
static void idct32(int *out, int *tab)
189
{
190
    int i, j;
191
    int *t, *t1, xr;
192
    const int *xp = costab32;
193

    
194
    for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
195

    
196
    t = tab + 30;
197
    t1 = tab + 2;
198
    do {
199
        t[0] += t[-4];
200
        t[1] += t[1 - 4];
201
        t -= 4;
202
    } while (t != t1);
203

    
204
    t = tab + 28;
205
    t1 = tab + 4;
206
    do {
207
        t[0] += t[-8];
208
        t[1] += t[1-8];
209
        t[2] += t[2-8];
210
        t[3] += t[3-8];
211
        t -= 8;
212
    } while (t != t1);
213

    
214
    t = tab;
215
    t1 = tab + 32;
216
    do {
217
        t[ 3] = -t[ 3];
218
        t[ 6] = -t[ 6];
219

    
220
        t[11] = -t[11];
221
        t[12] = -t[12];
222
        t[13] = -t[13];
223
        t[15] = -t[15];
224
        t += 16;
225
    } while (t != t1);
226

    
227

    
228
    t = tab;
229
    t1 = tab + 8;
230
    do {
231
        int x1, x2, x3, x4;
232

    
233
        x3 = MUL(t[16], FIX(SQRT2*0.5));
234
        x4 = t[0] - x3;
235
        x3 = t[0] + x3;
236

    
237
        x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
238
        x1 = MUL((t[8] - x2), xp[0]);
239
        x2 = MUL((t[8] + x2), xp[1]);
240

    
241
        t[ 0] = x3 + x1;
242
        t[ 8] = x4 - x2;
243
        t[16] = x4 + x2;
244
        t[24] = x3 - x1;
245
        t++;
246
    } while (t != t1);
247

    
248
    xp += 2;
249
    t = tab;
250
    t1 = tab + 4;
251
    do {
252
        xr = MUL(t[28],xp[0]);
253
        t[28] = (t[0] - xr);
254
        t[0] = (t[0] + xr);
255

    
256
        xr = MUL(t[4],xp[1]);
257
        t[ 4] = (t[24] - xr);
258
        t[24] = (t[24] + xr);
259

    
260
        xr = MUL(t[20],xp[2]);
261
        t[20] = (t[8] - xr);
262
        t[ 8] = (t[8] + xr);
263

    
264
        xr = MUL(t[12],xp[3]);
265
        t[12] = (t[16] - xr);
266
        t[16] = (t[16] + xr);
267
        t++;
268
    } while (t != t1);
269
    xp += 4;
270

    
271
    for (i = 0; i < 4; i++) {
272
        xr = MUL(tab[30-i*4],xp[0]);
273
        tab[30-i*4] = (tab[i*4] - xr);
274
        tab[   i*4] = (tab[i*4] + xr);
275

    
276
        xr = MUL(tab[ 2+i*4],xp[1]);
277
        tab[ 2+i*4] = (tab[28-i*4] - xr);
278
        tab[28-i*4] = (tab[28-i*4] + xr);
279

    
280
        xr = MUL(tab[31-i*4],xp[0]);
281
        tab[31-i*4] = (tab[1+i*4] - xr);
282
        tab[ 1+i*4] = (tab[1+i*4] + xr);
283

    
284
        xr = MUL(tab[ 3+i*4],xp[1]);
285
        tab[ 3+i*4] = (tab[29-i*4] - xr);
286
        tab[29-i*4] = (tab[29-i*4] + xr);
287

    
288
        xp += 2;
289
    }
290

    
291
    t = tab + 30;
292
    t1 = tab + 1;
293
    do {
294
        xr = MUL(t1[0], *xp);
295
        t1[0] = (t[0] - xr);
296
        t[0] = (t[0] + xr);
297
        t -= 2;
298
        t1 += 2;
299
        xp++;
300
    } while (t >= tab);
301

    
302
    for(i=0;i<32;i++) {
303
        out[i] = tab[bitinv32[i]];
304
    }
305
}
306

    
307
#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
308

    
309
static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
310
{
311
    short *p, *q;
312
    int sum, offset, i, j;
313
    int tmp[64];
314
    int tmp1[32];
315
    int *out;
316

    
317
    //    print_pow1(samples, 1152);
318

    
319
    offset = s->samples_offset[ch];
320
    out = &s->sb_samples[ch][0][0][0];
321
    for(j=0;j<36;j++) {
322
        /* 32 samples at once */
323
        for(i=0;i<32;i++) {
324
            s->samples_buf[ch][offset + (31 - i)] = samples[0];
325
            samples += incr;
326
        }
327

    
328
        /* filter */
329
        p = s->samples_buf[ch] + offset;
330
        q = filter_bank;
331
        /* maxsum = 23169 */
332
        for(i=0;i<64;i++) {
333
            sum = p[0*64] * q[0*64];
334
            sum += p[1*64] * q[1*64];
335
            sum += p[2*64] * q[2*64];
336
            sum += p[3*64] * q[3*64];
337
            sum += p[4*64] * q[4*64];
338
            sum += p[5*64] * q[5*64];
339
            sum += p[6*64] * q[6*64];
340
            sum += p[7*64] * q[7*64];
341
            tmp[i] = sum;
342
            p++;
343
            q++;
344
        }
345
        tmp1[0] = tmp[16] >> WSHIFT;
346
        for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
347
        for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
348

    
349
        idct32(out, tmp1);
350

    
351
        /* advance of 32 samples */
352
        offset -= 32;
353
        out += 32;
354
        /* handle the wrap around */
355
        if (offset < 0) {
356
            memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
357
                    s->samples_buf[ch], (512 - 32) * 2);
358
            offset = SAMPLES_BUF_SIZE - 512;
359
        }
360
    }
361
    s->samples_offset[ch] = offset;
362

    
363
    //    print_pow(s->sb_samples, 1152);
364
}
365

    
366
static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
367
                                  unsigned char scale_factors[SBLIMIT][3],
368
                                  int sb_samples[3][12][SBLIMIT],
369
                                  int sblimit)
370
{
371
    int *p, vmax, v, n, i, j, k, code;
372
    int index, d1, d2;
373
    unsigned char *sf = &scale_factors[0][0];
374

    
375
    for(j=0;j<sblimit;j++) {
376
        for(i=0;i<3;i++) {
377
            /* find the max absolute value */
378
            p = &sb_samples[i][0][j];
379
            vmax = abs(*p);
380
            for(k=1;k<12;k++) {
381
                p += SBLIMIT;
382
                v = abs(*p);
383
                if (v > vmax)
384
                    vmax = v;
385
            }
386
            /* compute the scale factor index using log 2 computations */
387
            if (vmax > 1) {
388
                n = av_log2(vmax);
389
                /* n is the position of the MSB of vmax. now
390
                   use at most 2 compares to find the index */
391
                index = (21 - n) * 3 - 3;
392
                if (index >= 0) {
393
                    while (vmax <= scale_factor_table[index+1])
394
                        index++;
395
                } else {
396
                    index = 0; /* very unlikely case of overflow */
397
                }
398
            } else {
399
                index = 62; /* value 63 is not allowed */
400
            }
401

    
402
#if 0
403
            printf("%2d:%d in=%x %x %d\n",
404
                   j, i, vmax, scale_factor_table[index], index);
405
#endif
406
            /* store the scale factor */
407
            assert(index >=0 && index <= 63);
408
            sf[i] = index;
409
        }
410

    
411
        /* compute the transmission factor : look if the scale factors
412
           are close enough to each other */
413
        d1 = scale_diff_table[sf[0] - sf[1] + 64];
414
        d2 = scale_diff_table[sf[1] - sf[2] + 64];
415

    
416
        /* handle the 25 cases */
417
        switch(d1 * 5 + d2) {
418
        case 0*5+0:
419
        case 0*5+4:
420
        case 3*5+4:
421
        case 4*5+0:
422
        case 4*5+4:
423
            code = 0;
424
            break;
425
        case 0*5+1:
426
        case 0*5+2:
427
        case 4*5+1:
428
        case 4*5+2:
429
            code = 3;
430
            sf[2] = sf[1];
431
            break;
432
        case 0*5+3:
433
        case 4*5+3:
434
            code = 3;
435
            sf[1] = sf[2];
436
            break;
437
        case 1*5+0:
438
        case 1*5+4:
439
        case 2*5+4:
440
            code = 1;
441
            sf[1] = sf[0];
442
            break;
443
        case 1*5+1:
444
        case 1*5+2:
445
        case 2*5+0:
446
        case 2*5+1:
447
        case 2*5+2:
448
            code = 2;
449
            sf[1] = sf[2] = sf[0];
450
            break;
451
        case 2*5+3:
452
        case 3*5+3:
453
            code = 2;
454
            sf[0] = sf[1] = sf[2];
455
            break;
456
        case 3*5+0:
457
        case 3*5+1:
458
        case 3*5+2:
459
            code = 2;
460
            sf[0] = sf[2] = sf[1];
461
            break;
462
        case 1*5+3:
463
            code = 2;
464
            if (sf[0] > sf[2])
465
              sf[0] = sf[2];
466
            sf[1] = sf[2] = sf[0];
467
            break;
468
        default:
469
            assert(0); //cannot happen
470
            code = 0;           /* kill warning */
471
        }
472

    
473
#if 0
474
        printf("%d: %2d %2d %2d %d %d -> %d\n", j,
475
               sf[0], sf[1], sf[2], d1, d2, code);
476
#endif
477
        scale_code[j] = code;
478
        sf += 3;
479
    }
480
}
481

    
482
/* The most important function : psycho acoustic module. In this
483
   encoder there is basically none, so this is the worst you can do,
484
   but also this is the simpler. */
485
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
486
{
487
    int i;
488

    
489
    for(i=0;i<s->sblimit;i++) {
490
        smr[i] = (int)(fixed_smr[i] * 10);
491
    }
492
}
493

    
494

    
495
#define SB_NOTALLOCATED  0
496
#define SB_ALLOCATED     1
497
#define SB_NOMORE        2
498

    
499
/* Try to maximize the smr while using a number of bits inferior to
500
   the frame size. I tried to make the code simpler, faster and
501
   smaller than other encoders :-) */
502
static void compute_bit_allocation(MpegAudioContext *s,
503
                                   short smr1[MPA_MAX_CHANNELS][SBLIMIT],
504
                                   unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
505
                                   int *padding)
506
{
507
    int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
508
    int incr;
509
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
510
    unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
511
    const unsigned char *alloc;
512

    
513
    memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
514
    memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
515
    memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
516

    
517
    /* compute frame size and padding */
518
    max_frame_size = s->frame_size;
519
    s->frame_frac += s->frame_frac_incr;
520
    if (s->frame_frac >= 65536) {
521
        s->frame_frac -= 65536;
522
        s->do_padding = 1;
523
        max_frame_size += 8;
524
    } else {
525
        s->do_padding = 0;
526
    }
527

    
528
    /* compute the header + bit alloc size */
529
    current_frame_size = 32;
530
    alloc = s->alloc_table;
531
    for(i=0;i<s->sblimit;i++) {
532
        incr = alloc[0];
533
        current_frame_size += incr * s->nb_channels;
534
        alloc += 1 << incr;
535
    }
536
    for(;;) {
537
        /* look for the subband with the largest signal to mask ratio */
538
        max_sb = -1;
539
        max_ch = -1;
540
        max_smr = INT_MIN;
541
        for(ch=0;ch<s->nb_channels;ch++) {
542
            for(i=0;i<s->sblimit;i++) {
543
                if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
544
                    max_smr = smr[ch][i];
545
                    max_sb = i;
546
                    max_ch = ch;
547
                }
548
            }
549
        }
550
#if 0
551
        printf("current=%d max=%d max_sb=%d alloc=%d\n",
552
               current_frame_size, max_frame_size, max_sb,
553
               bit_alloc[max_sb]);
554
#endif
555
        if (max_sb < 0)
556
            break;
557

    
558
        /* find alloc table entry (XXX: not optimal, should use
559
           pointer table) */
560
        alloc = s->alloc_table;
561
        for(i=0;i<max_sb;i++) {
562
            alloc += 1 << alloc[0];
563
        }
564

    
565
        if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
566
            /* nothing was coded for this band: add the necessary bits */
567
            incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
568
            incr += total_quant_bits[alloc[1]];
569
        } else {
570
            /* increments bit allocation */
571
            b = bit_alloc[max_ch][max_sb];
572
            incr = total_quant_bits[alloc[b + 1]] -
573
                total_quant_bits[alloc[b]];
574
        }
575

    
576
        if (current_frame_size + incr <= max_frame_size) {
577
            /* can increase size */
578
            b = ++bit_alloc[max_ch][max_sb];
579
            current_frame_size += incr;
580
            /* decrease smr by the resolution we added */
581
            smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
582
            /* max allocation size reached ? */
583
            if (b == ((1 << alloc[0]) - 1))
584
                subband_status[max_ch][max_sb] = SB_NOMORE;
585
            else
586
                subband_status[max_ch][max_sb] = SB_ALLOCATED;
587
        } else {
588
            /* cannot increase the size of this subband */
589
            subband_status[max_ch][max_sb] = SB_NOMORE;
590
        }
591
    }
592
    *padding = max_frame_size - current_frame_size;
593
    assert(*padding >= 0);
594

    
595
#if 0
596
    for(i=0;i<s->sblimit;i++) {
597
        printf("%d ", bit_alloc[i]);
598
    }
599
    printf("\n");
600
#endif
601
}
602

    
603
/*
604
 * Output the mpeg audio layer 2 frame. Note how the code is small
605
 * compared to other encoders :-)
606
 */
607
static void encode_frame(MpegAudioContext *s,
608
                         unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
609
                         int padding)
610
{
611
    int i, j, k, l, bit_alloc_bits, b, ch;
612
    unsigned char *sf;
613
    int q[3];
614
    PutBitContext *p = &s->pb;
615

    
616
    /* header */
617

    
618
    put_bits(p, 12, 0xfff);
619
    put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
620
    put_bits(p, 2, 4-2);  /* layer 2 */
621
    put_bits(p, 1, 1); /* no error protection */
622
    put_bits(p, 4, s->bitrate_index);
623
    put_bits(p, 2, s->freq_index);
624
    put_bits(p, 1, s->do_padding); /* use padding */
625
    put_bits(p, 1, 0);             /* private_bit */
626
    put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
627
    put_bits(p, 2, 0); /* mode_ext */
628
    put_bits(p, 1, 0); /* no copyright */
629
    put_bits(p, 1, 1); /* original */
630
    put_bits(p, 2, 0); /* no emphasis */
631

    
632
    /* bit allocation */
633
    j = 0;
634
    for(i=0;i<s->sblimit;i++) {
635
        bit_alloc_bits = s->alloc_table[j];
636
        for(ch=0;ch<s->nb_channels;ch++) {
637
            put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
638
        }
639
        j += 1 << bit_alloc_bits;
640
    }
641

    
642
    /* scale codes */
643
    for(i=0;i<s->sblimit;i++) {
644
        for(ch=0;ch<s->nb_channels;ch++) {
645
            if (bit_alloc[ch][i])
646
                put_bits(p, 2, s->scale_code[ch][i]);
647
        }
648
    }
649

    
650
    /* scale factors */
651
    for(i=0;i<s->sblimit;i++) {
652
        for(ch=0;ch<s->nb_channels;ch++) {
653
            if (bit_alloc[ch][i]) {
654
                sf = &s->scale_factors[ch][i][0];
655
                switch(s->scale_code[ch][i]) {
656
                case 0:
657
                    put_bits(p, 6, sf[0]);
658
                    put_bits(p, 6, sf[1]);
659
                    put_bits(p, 6, sf[2]);
660
                    break;
661
                case 3:
662
                case 1:
663
                    put_bits(p, 6, sf[0]);
664
                    put_bits(p, 6, sf[2]);
665
                    break;
666
                case 2:
667
                    put_bits(p, 6, sf[0]);
668
                    break;
669
                }
670
            }
671
        }
672
    }
673

    
674
    /* quantization & write sub band samples */
675

    
676
    for(k=0;k<3;k++) {
677
        for(l=0;l<12;l+=3) {
678
            j = 0;
679
            for(i=0;i<s->sblimit;i++) {
680
                bit_alloc_bits = s->alloc_table[j];
681
                for(ch=0;ch<s->nb_channels;ch++) {
682
                    b = bit_alloc[ch][i];
683
                    if (b) {
684
                        int qindex, steps, m, sample, bits;
685
                        /* we encode 3 sub band samples of the same sub band at a time */
686
                        qindex = s->alloc_table[j+b];
687
                        steps = ff_mpa_quant_steps[qindex];
688
                        for(m=0;m<3;m++) {
689
                            sample = s->sb_samples[ch][k][l + m][i];
690
                            /* divide by scale factor */
691
#ifdef USE_FLOATS
692
                            {
693
                                float a;
694
                                a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
695
                                q[m] = (int)((a + 1.0) * steps * 0.5);
696
                            }
697
#else
698
                            {
699
                                int q1, e, shift, mult;
700
                                e = s->scale_factors[ch][i][k];
701
                                shift = scale_factor_shift[e];
702
                                mult = scale_factor_mult[e];
703

    
704
                                /* normalize to P bits */
705
                                if (shift < 0)
706
                                    q1 = sample << (-shift);
707
                                else
708
                                    q1 = sample >> shift;
709
                                q1 = (q1 * mult) >> P;
710
                                q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
711
                            }
712
#endif
713
                            if (q[m] >= steps)
714
                                q[m] = steps - 1;
715
                            assert(q[m] >= 0 && q[m] < steps);
716
                        }
717
                        bits = ff_mpa_quant_bits[qindex];
718
                        if (bits < 0) {
719
                            /* group the 3 values to save bits */
720
                            put_bits(p, -bits,
721
                                     q[0] + steps * (q[1] + steps * q[2]));
722
#if 0
723
                            printf("%d: gr1 %d\n",
724
                                   i, q[0] + steps * (q[1] + steps * q[2]));
725
#endif
726
                        } else {
727
#if 0
728
                            printf("%d: gr3 %d %d %d\n",
729
                                   i, q[0], q[1], q[2]);
730
#endif
731
                            put_bits(p, bits, q[0]);
732
                            put_bits(p, bits, q[1]);
733
                            put_bits(p, bits, q[2]);
734
                        }
735
                    }
736
                }
737
                /* next subband in alloc table */
738
                j += 1 << bit_alloc_bits;
739
            }
740
        }
741
    }
742

    
743
    /* padding */
744
    for(i=0;i<padding;i++)
745
        put_bits(p, 1, 0);
746

    
747
    /* flush */
748
    flush_put_bits(p);
749
}
750

    
751
static int MPA_encode_frame(AVCodecContext *avctx,
752
                            unsigned char *frame, int buf_size, void *data)
753
{
754
    MpegAudioContext *s = avctx->priv_data;
755
    const short *samples = data;
756
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
757
    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
758
    int padding, i;
759

    
760
    for(i=0;i<s->nb_channels;i++) {
761
        filter(s, i, samples + i, s->nb_channels);
762
    }
763

    
764
    for(i=0;i<s->nb_channels;i++) {
765
        compute_scale_factors(s->scale_code[i], s->scale_factors[i],
766
                              s->sb_samples[i], s->sblimit);
767
    }
768
    for(i=0;i<s->nb_channels;i++) {
769
        psycho_acoustic_model(s, smr[i]);
770
    }
771
    compute_bit_allocation(s, smr, bit_alloc, &padding);
772

    
773
    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
774

    
775
    encode_frame(s, bit_alloc, padding);
776

    
777
    return put_bits_ptr(&s->pb) - s->pb.buf;
778
}
779

    
780
static av_cold int MPA_encode_close(AVCodecContext *avctx)
781
{
782
    av_freep(&avctx->coded_frame);
783
    return 0;
784
}
785

    
786
AVCodec ff_mp2_encoder = {
787
    "mp2",
788
    AVMEDIA_TYPE_AUDIO,
789
    CODEC_ID_MP2,
790
    sizeof(MpegAudioContext),
791
    MPA_encode_init,
792
    MPA_encode_frame,
793
    MPA_encode_close,
794
    NULL,
795
    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
796
    .supported_samplerates= (const int[]){44100, 48000,  32000, 22050, 24000, 16000, 0},
797
    .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
798
};
799

    
800
#undef FIX