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1
/*
2
 * The simplest mpeg audio layer 2 encoder
3
 * Copyright (c) 2000, 2001 Fabrice Bellard
4
 *
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 * This file is part of Libav.
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 *
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 * Libav is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * Libav 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 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 Libav; 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"
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30
#define FRAC_BITS   15   /* fractional bits for sb_samples and dct */
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#define WFRAC_BITS  14   /* fractional bits for window */
32

    
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#include "mpegaudio.h"
34

    
35
/* currently, cannot change these constants (need to modify
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   quantization stage) */
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#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
38

    
39
#define SAMPLES_BUF_SIZE 4096
40

    
41
typedef struct MpegAudioContext {
42
    PutBitContext pb;
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    int nb_channels;
44
    int lsf;           /* 1 if mpeg2 low bitrate selected */
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    int bitrate_index; /* bit rate */
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    int freq_index;
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    int frame_size; /* frame size, in bits, without padding */
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    /* 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 */
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    int samples_offset[MPA_MAX_CHANNELS];       /* offset in samples_buf */
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    int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
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    unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
54
    /* code to group 3 scale factors */
55
    unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
56
    int sblimit; /* number of used subbands */
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    const unsigned char *alloc_table;
58
} MpegAudioContext;
59

    
60
/* define it to use floats in quantization (I don't like floats !) */
61
#define USE_FLOATS
62

    
63
#include "mpegaudiodata.h"
64
#include "mpegaudiotab.h"
65

    
66
static av_cold int MPA_encode_init(AVCodecContext *avctx)
67
{
68
    MpegAudioContext *s = avctx->priv_data;
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    int freq = avctx->sample_rate;
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    int bitrate = avctx->bit_rate;
71
    int channels = avctx->channels;
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    int i, v, table;
73
    float a;
74

    
75
    if (channels <= 0 || channels > 2){
76
        av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
77
        return -1;
78
    }
79
    bitrate = bitrate / 1000;
80
    s->nb_channels = channels;
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    avctx->frame_size = MPA_FRAME_SIZE;
82

    
83
    /* encoding freq */
84
    s->lsf = 0;
85
    for(i=0;i<3;i++) {
86
        if (ff_mpa_freq_tab[i] == freq)
87
            break;
88
        if ((ff_mpa_freq_tab[i] / 2) == freq) {
89
            s->lsf = 1;
90
            break;
91
        }
92
    }
93
    if (i == 3){
94
        av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
95
        return -1;
96
    }
97
    s->freq_index = i;
98

    
99
    /* encoding bitrate & frequency */
100
    for(i=0;i<15;i++) {
101
        if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
102
            break;
103
    }
104
    if (i == 15){
105
        av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
106
        return -1;
107
    }
108
    s->bitrate_index = i;
109

    
110
    /* compute total header size & pad bit */
111

    
112
    a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
113
    s->frame_size = ((int)a) * 8;
114

    
115
    /* frame fractional size to compute padding */
116
    s->frame_frac = 0;
117
    s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
118

    
119
    /* select the right allocation table */
120
    table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
121

    
122
    /* number of used subbands */
123
    s->sblimit = ff_mpa_sblimit_table[table];
124
    s->alloc_table = ff_mpa_alloc_tables[table];
125

    
126
    av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
127
            bitrate, freq, s->frame_size, table, s->frame_frac_incr);
128

    
129
    for(i=0;i<s->nb_channels;i++)
130
        s->samples_offset[i] = 0;
131

    
132
    for(i=0;i<257;i++) {
133
        int v;
134
        v = ff_mpa_enwindow[i];
135
#if WFRAC_BITS != 16
136
        v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
137
#endif
138
        filter_bank[i] = v;
139
        if ((i & 63) != 0)
140
            v = -v;
141
        if (i != 0)
142
            filter_bank[512 - i] = v;
143
    }
144

    
145
    for(i=0;i<64;i++) {
146
        v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
147
        if (v <= 0)
148
            v = 1;
149
        scale_factor_table[i] = v;
150
#ifdef USE_FLOATS
151
        scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
152
#else
153
#define P 15
154
        scale_factor_shift[i] = 21 - P - (i / 3);
155
        scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
156
#endif
157
    }
158
    for(i=0;i<128;i++) {
159
        v = i - 64;
160
        if (v <= -3)
161
            v = 0;
162
        else if (v < 0)
163
            v = 1;
164
        else if (v == 0)
165
            v = 2;
166
        else if (v < 3)
167
            v = 3;
168
        else
169
            v = 4;
170
        scale_diff_table[i] = v;
171
    }
172

    
173
    for(i=0;i<17;i++) {
174
        v = ff_mpa_quant_bits[i];
175
        if (v < 0)
176
            v = -v;
177
        else
178
            v = v * 3;
179
        total_quant_bits[i] = 12 * v;
180
    }
181

    
182
    avctx->coded_frame= avcodec_alloc_frame();
183
    avctx->coded_frame->key_frame= 1;
184

    
185
    return 0;
186
}
187

    
188
/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
189
static void idct32(int *out, int *tab)
190
{
191
    int i, j;
192
    int *t, *t1, xr;
193
    const int *xp = costab32;
194

    
195
    for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
196

    
197
    t = tab + 30;
198
    t1 = tab + 2;
199
    do {
200
        t[0] += t[-4];
201
        t[1] += t[1 - 4];
202
        t -= 4;
203
    } while (t != t1);
204

    
205
    t = tab + 28;
206
    t1 = tab + 4;
207
    do {
208
        t[0] += t[-8];
209
        t[1] += t[1-8];
210
        t[2] += t[2-8];
211
        t[3] += t[3-8];
212
        t -= 8;
213
    } while (t != t1);
214

    
215
    t = tab;
216
    t1 = tab + 32;
217
    do {
218
        t[ 3] = -t[ 3];
219
        t[ 6] = -t[ 6];
220

    
221
        t[11] = -t[11];
222
        t[12] = -t[12];
223
        t[13] = -t[13];
224
        t[15] = -t[15];
225
        t += 16;
226
    } while (t != t1);
227

    
228

    
229
    t = tab;
230
    t1 = tab + 8;
231
    do {
232
        int x1, x2, x3, x4;
233

    
234
        x3 = MUL(t[16], FIX(SQRT2*0.5));
235
        x4 = t[0] - x3;
236
        x3 = t[0] + x3;
237

    
238
        x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
239
        x1 = MUL((t[8] - x2), xp[0]);
240
        x2 = MUL((t[8] + x2), xp[1]);
241

    
242
        t[ 0] = x3 + x1;
243
        t[ 8] = x4 - x2;
244
        t[16] = x4 + x2;
245
        t[24] = x3 - x1;
246
        t++;
247
    } while (t != t1);
248

    
249
    xp += 2;
250
    t = tab;
251
    t1 = tab + 4;
252
    do {
253
        xr = MUL(t[28],xp[0]);
254
        t[28] = (t[0] - xr);
255
        t[0] = (t[0] + xr);
256

    
257
        xr = MUL(t[4],xp[1]);
258
        t[ 4] = (t[24] - xr);
259
        t[24] = (t[24] + xr);
260

    
261
        xr = MUL(t[20],xp[2]);
262
        t[20] = (t[8] - xr);
263
        t[ 8] = (t[8] + xr);
264

    
265
        xr = MUL(t[12],xp[3]);
266
        t[12] = (t[16] - xr);
267
        t[16] = (t[16] + xr);
268
        t++;
269
    } while (t != t1);
270
    xp += 4;
271

    
272
    for (i = 0; i < 4; i++) {
273
        xr = MUL(tab[30-i*4],xp[0]);
274
        tab[30-i*4] = (tab[i*4] - xr);
275
        tab[   i*4] = (tab[i*4] + xr);
276

    
277
        xr = MUL(tab[ 2+i*4],xp[1]);
278
        tab[ 2+i*4] = (tab[28-i*4] - xr);
279
        tab[28-i*4] = (tab[28-i*4] + xr);
280

    
281
        xr = MUL(tab[31-i*4],xp[0]);
282
        tab[31-i*4] = (tab[1+i*4] - xr);
283
        tab[ 1+i*4] = (tab[1+i*4] + xr);
284

    
285
        xr = MUL(tab[ 3+i*4],xp[1]);
286
        tab[ 3+i*4] = (tab[29-i*4] - xr);
287
        tab[29-i*4] = (tab[29-i*4] + xr);
288

    
289
        xp += 2;
290
    }
291

    
292
    t = tab + 30;
293
    t1 = tab + 1;
294
    do {
295
        xr = MUL(t1[0], *xp);
296
        t1[0] = (t[0] - xr);
297
        t[0] = (t[0] + xr);
298
        t -= 2;
299
        t1 += 2;
300
        xp++;
301
    } while (t >= tab);
302

    
303
    for(i=0;i<32;i++) {
304
        out[i] = tab[bitinv32[i]];
305
    }
306
}
307

    
308
#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
309

    
310
static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
311
{
312
    short *p, *q;
313
    int sum, offset, i, j;
314
    int tmp[64];
315
    int tmp1[32];
316
    int *out;
317

    
318
    //    print_pow1(samples, 1152);
319

    
320
    offset = s->samples_offset[ch];
321
    out = &s->sb_samples[ch][0][0][0];
322
    for(j=0;j<36;j++) {
323
        /* 32 samples at once */
324
        for(i=0;i<32;i++) {
325
            s->samples_buf[ch][offset + (31 - i)] = samples[0];
326
            samples += incr;
327
        }
328

    
329
        /* filter */
330
        p = s->samples_buf[ch] + offset;
331
        q = filter_bank;
332
        /* maxsum = 23169 */
333
        for(i=0;i<64;i++) {
334
            sum = p[0*64] * q[0*64];
335
            sum += p[1*64] * q[1*64];
336
            sum += p[2*64] * q[2*64];
337
            sum += p[3*64] * q[3*64];
338
            sum += p[4*64] * q[4*64];
339
            sum += p[5*64] * q[5*64];
340
            sum += p[6*64] * q[6*64];
341
            sum += p[7*64] * q[7*64];
342
            tmp[i] = sum;
343
            p++;
344
            q++;
345
        }
346
        tmp1[0] = tmp[16] >> WSHIFT;
347
        for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
348
        for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
349

    
350
        idct32(out, tmp1);
351

    
352
        /* advance of 32 samples */
353
        offset -= 32;
354
        out += 32;
355
        /* handle the wrap around */
356
        if (offset < 0) {
357
            memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
358
                    s->samples_buf[ch], (512 - 32) * 2);
359
            offset = SAMPLES_BUF_SIZE - 512;
360
        }
361
    }
362
    s->samples_offset[ch] = offset;
363

    
364
    //    print_pow(s->sb_samples, 1152);
365
}
366

    
367
static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
368
                                  unsigned char scale_factors[SBLIMIT][3],
369
                                  int sb_samples[3][12][SBLIMIT],
370
                                  int sblimit)
371
{
372
    int *p, vmax, v, n, i, j, k, code;
373
    int index, d1, d2;
374
    unsigned char *sf = &scale_factors[0][0];
375

    
376
    for(j=0;j<sblimit;j++) {
377
        for(i=0;i<3;i++) {
378
            /* find the max absolute value */
379
            p = &sb_samples[i][0][j];
380
            vmax = abs(*p);
381
            for(k=1;k<12;k++) {
382
                p += SBLIMIT;
383
                v = abs(*p);
384
                if (v > vmax)
385
                    vmax = v;
386
            }
387
            /* compute the scale factor index using log 2 computations */
388
            if (vmax > 1) {
389
                n = av_log2(vmax);
390
                /* n is the position of the MSB of vmax. now
391
                   use at most 2 compares to find the index */
392
                index = (21 - n) * 3 - 3;
393
                if (index >= 0) {
394
                    while (vmax <= scale_factor_table[index+1])
395
                        index++;
396
                } else {
397
                    index = 0; /* very unlikely case of overflow */
398
                }
399
            } else {
400
                index = 62; /* value 63 is not allowed */
401
            }
402

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

    
410
        /* compute the transmission factor : look if the scale factors
411
           are close enough to each other */
412
        d1 = scale_diff_table[sf[0] - sf[1] + 64];
413
        d2 = scale_diff_table[sf[1] - sf[2] + 64];
414

    
415
        /* handle the 25 cases */
416
        switch(d1 * 5 + d2) {
417
        case 0*5+0:
418
        case 0*5+4:
419
        case 3*5+4:
420
        case 4*5+0:
421
        case 4*5+4:
422
            code = 0;
423
            break;
424
        case 0*5+1:
425
        case 0*5+2:
426
        case 4*5+1:
427
        case 4*5+2:
428
            code = 3;
429
            sf[2] = sf[1];
430
            break;
431
        case 0*5+3:
432
        case 4*5+3:
433
            code = 3;
434
            sf[1] = sf[2];
435
            break;
436
        case 1*5+0:
437
        case 1*5+4:
438
        case 2*5+4:
439
            code = 1;
440
            sf[1] = sf[0];
441
            break;
442
        case 1*5+1:
443
        case 1*5+2:
444
        case 2*5+0:
445
        case 2*5+1:
446
        case 2*5+2:
447
            code = 2;
448
            sf[1] = sf[2] = sf[0];
449
            break;
450
        case 2*5+3:
451
        case 3*5+3:
452
            code = 2;
453
            sf[0] = sf[1] = sf[2];
454
            break;
455
        case 3*5+0:
456
        case 3*5+1:
457
        case 3*5+2:
458
            code = 2;
459
            sf[0] = sf[2] = sf[1];
460
            break;
461
        case 1*5+3:
462
            code = 2;
463
            if (sf[0] > sf[2])
464
              sf[0] = sf[2];
465
            sf[1] = sf[2] = sf[0];
466
            break;
467
        default:
468
            assert(0); //cannot happen
469
            code = 0;           /* kill warning */
470
        }
471

    
472
        av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,
473
                sf[0], sf[1], sf[2], d1, d2, code);
474
        scale_code[j] = code;
475
        sf += 3;
476
    }
477
}
478

    
479
/* The most important function : psycho acoustic module. In this
480
   encoder there is basically none, so this is the worst you can do,
481
   but also this is the simpler. */
482
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
483
{
484
    int i;
485

    
486
    for(i=0;i<s->sblimit;i++) {
487
        smr[i] = (int)(fixed_smr[i] * 10);
488
    }
489
}
490

    
491

    
492
#define SB_NOTALLOCATED  0
493
#define SB_ALLOCATED     1
494
#define SB_NOMORE        2
495

    
496
/* Try to maximize the smr while using a number of bits inferior to
497
   the frame size. I tried to make the code simpler, faster and
498
   smaller than other encoders :-) */
499
static void compute_bit_allocation(MpegAudioContext *s,
500
                                   short smr1[MPA_MAX_CHANNELS][SBLIMIT],
501
                                   unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
502
                                   int *padding)
503
{
504
    int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
505
    int incr;
506
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
507
    unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
508
    const unsigned char *alloc;
509

    
510
    memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
511
    memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
512
    memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
513

    
514
    /* compute frame size and padding */
515
    max_frame_size = s->frame_size;
516
    s->frame_frac += s->frame_frac_incr;
517
    if (s->frame_frac >= 65536) {
518
        s->frame_frac -= 65536;
519
        s->do_padding = 1;
520
        max_frame_size += 8;
521
    } else {
522
        s->do_padding = 0;
523
    }
524

    
525
    /* compute the header + bit alloc size */
526
    current_frame_size = 32;
527
    alloc = s->alloc_table;
528
    for(i=0;i<s->sblimit;i++) {
529
        incr = alloc[0];
530
        current_frame_size += incr * s->nb_channels;
531
        alloc += 1 << incr;
532
    }
533
    for(;;) {
534
        /* look for the subband with the largest signal to mask ratio */
535
        max_sb = -1;
536
        max_ch = -1;
537
        max_smr = INT_MIN;
538
        for(ch=0;ch<s->nb_channels;ch++) {
539
            for(i=0;i<s->sblimit;i++) {
540
                if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
541
                    max_smr = smr[ch][i];
542
                    max_sb = i;
543
                    max_ch = ch;
544
                }
545
            }
546
        }
547
        av_dlog(NULL, "current=%d max=%d max_sb=%d alloc=%d\n",
548
                current_frame_size, max_frame_size, max_sb,
549
                bit_alloc[max_sb]);
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

    
591
/*
592
 * Output the mpeg audio layer 2 frame. Note how the code is small
593
 * compared to other encoders :-)
594
 */
595
static void encode_frame(MpegAudioContext *s,
596
                         unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
597
                         int padding)
598
{
599
    int i, j, k, l, bit_alloc_bits, b, ch;
600
    unsigned char *sf;
601
    int q[3];
602
    PutBitContext *p = &s->pb;
603

    
604
    /* header */
605

    
606
    put_bits(p, 12, 0xfff);
607
    put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
608
    put_bits(p, 2, 4-2);  /* layer 2 */
609
    put_bits(p, 1, 1); /* no error protection */
610
    put_bits(p, 4, s->bitrate_index);
611
    put_bits(p, 2, s->freq_index);
612
    put_bits(p, 1, s->do_padding); /* use padding */
613
    put_bits(p, 1, 0);             /* private_bit */
614
    put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
615
    put_bits(p, 2, 0); /* mode_ext */
616
    put_bits(p, 1, 0); /* no copyright */
617
    put_bits(p, 1, 1); /* original */
618
    put_bits(p, 2, 0); /* no emphasis */
619

    
620
    /* bit allocation */
621
    j = 0;
622
    for(i=0;i<s->sblimit;i++) {
623
        bit_alloc_bits = s->alloc_table[j];
624
        for(ch=0;ch<s->nb_channels;ch++) {
625
            put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
626
        }
627
        j += 1 << bit_alloc_bits;
628
    }
629

    
630
    /* scale codes */
631
    for(i=0;i<s->sblimit;i++) {
632
        for(ch=0;ch<s->nb_channels;ch++) {
633
            if (bit_alloc[ch][i])
634
                put_bits(p, 2, s->scale_code[ch][i]);
635
        }
636
    }
637

    
638
    /* scale factors */
639
    for(i=0;i<s->sblimit;i++) {
640
        for(ch=0;ch<s->nb_channels;ch++) {
641
            if (bit_alloc[ch][i]) {
642
                sf = &s->scale_factors[ch][i][0];
643
                switch(s->scale_code[ch][i]) {
644
                case 0:
645
                    put_bits(p, 6, sf[0]);
646
                    put_bits(p, 6, sf[1]);
647
                    put_bits(p, 6, sf[2]);
648
                    break;
649
                case 3:
650
                case 1:
651
                    put_bits(p, 6, sf[0]);
652
                    put_bits(p, 6, sf[2]);
653
                    break;
654
                case 2:
655
                    put_bits(p, 6, sf[0]);
656
                    break;
657
                }
658
            }
659
        }
660
    }
661

    
662
    /* quantization & write sub band samples */
663

    
664
    for(k=0;k<3;k++) {
665
        for(l=0;l<12;l+=3) {
666
            j = 0;
667
            for(i=0;i<s->sblimit;i++) {
668
                bit_alloc_bits = s->alloc_table[j];
669
                for(ch=0;ch<s->nb_channels;ch++) {
670
                    b = bit_alloc[ch][i];
671
                    if (b) {
672
                        int qindex, steps, m, sample, bits;
673
                        /* we encode 3 sub band samples of the same sub band at a time */
674
                        qindex = s->alloc_table[j+b];
675
                        steps = ff_mpa_quant_steps[qindex];
676
                        for(m=0;m<3;m++) {
677
                            sample = s->sb_samples[ch][k][l + m][i];
678
                            /* divide by scale factor */
679
#ifdef USE_FLOATS
680
                            {
681
                                float a;
682
                                a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
683
                                q[m] = (int)((a + 1.0) * steps * 0.5);
684
                            }
685
#else
686
                            {
687
                                int q1, e, shift, mult;
688
                                e = s->scale_factors[ch][i][k];
689
                                shift = scale_factor_shift[e];
690
                                mult = scale_factor_mult[e];
691

    
692
                                /* normalize to P bits */
693
                                if (shift < 0)
694
                                    q1 = sample << (-shift);
695
                                else
696
                                    q1 = sample >> shift;
697
                                q1 = (q1 * mult) >> P;
698
                                q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
699
                            }
700
#endif
701
                            if (q[m] >= steps)
702
                                q[m] = steps - 1;
703
                            assert(q[m] >= 0 && q[m] < steps);
704
                        }
705
                        bits = ff_mpa_quant_bits[qindex];
706
                        if (bits < 0) {
707
                            /* group the 3 values to save bits */
708
                            put_bits(p, -bits,
709
                                     q[0] + steps * (q[1] + steps * q[2]));
710
                        } else {
711
                            put_bits(p, bits, q[0]);
712
                            put_bits(p, bits, q[1]);
713
                            put_bits(p, bits, q[2]);
714
                        }
715
                    }
716
                }
717
                /* next subband in alloc table */
718
                j += 1 << bit_alloc_bits;
719
            }
720
        }
721
    }
722

    
723
    /* padding */
724
    for(i=0;i<padding;i++)
725
        put_bits(p, 1, 0);
726

    
727
    /* flush */
728
    flush_put_bits(p);
729
}
730

    
731
static int MPA_encode_frame(AVCodecContext *avctx,
732
                            unsigned char *frame, int buf_size, void *data)
733
{
734
    MpegAudioContext *s = avctx->priv_data;
735
    const short *samples = data;
736
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
737
    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
738
    int padding, i;
739

    
740
    for(i=0;i<s->nb_channels;i++) {
741
        filter(s, i, samples + i, s->nb_channels);
742
    }
743

    
744
    for(i=0;i<s->nb_channels;i++) {
745
        compute_scale_factors(s->scale_code[i], s->scale_factors[i],
746
                              s->sb_samples[i], s->sblimit);
747
    }
748
    for(i=0;i<s->nb_channels;i++) {
749
        psycho_acoustic_model(s, smr[i]);
750
    }
751
    compute_bit_allocation(s, smr, bit_alloc, &padding);
752

    
753
    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
754

    
755
    encode_frame(s, bit_alloc, padding);
756

    
757
    return put_bits_ptr(&s->pb) - s->pb.buf;
758
}
759

    
760
static av_cold int MPA_encode_close(AVCodecContext *avctx)
761
{
762
    av_freep(&avctx->coded_frame);
763
    return 0;
764
}
765

    
766
AVCodec ff_mp2_encoder = {
767
    "mp2",
768
    AVMEDIA_TYPE_AUDIO,
769
    CODEC_ID_MP2,
770
    sizeof(MpegAudioContext),
771
    MPA_encode_init,
772
    MPA_encode_frame,
773
    MPA_encode_close,
774
    NULL,
775
    .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
776
    .supported_samplerates= (const int[]){44100, 48000,  32000, 22050, 24000, 16000, 0},
777
    .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
778
};
779

    
780
#undef FIX