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ffmpeg / libavcodec / mpegaudioenc.c @ 755bfeab

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1
/*
2
 * The simplest mpeg audio layer 2 encoder
3
 * Copyright (c) 2000, 2001 Fabrice Bellard.
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 *
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 * 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,
13
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
15
 * 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
/**
23
 * @file mpegaudio.c
24
 * The simplest mpeg audio layer 2 encoder.
25
 */
26

    
27
#include "avcodec.h"
28
#include "bitstream.h"
29
#include "mpegaudio.h"
30

    
31
/* currently, cannot change these constants (need to modify
32
   quantization stage) */
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#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
34
#define FIX(a)   ((int)((a) * (1 << FRAC_BITS)))
35

    
36
#define SAMPLES_BUF_SIZE 4096
37

    
38
typedef struct MpegAudioContext {
39
    PutBitContext pb;
40
    int nb_channels;
41
    int freq, bit_rate;
42
    int lsf;           /* 1 if mpeg2 low bitrate selected */
43
    int bitrate_index; /* bit rate */
44
    int freq_index;
45
    int frame_size; /* frame size, in bits, without padding */
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    int64_t nb_samples; /* total number of samples encoded */
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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 */
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 */
56
    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 int MPA_encode_init(AVCodecContext *avctx)
66
{
67
    MpegAudioContext *s = avctx->priv_data;
68
    int freq = avctx->sample_rate;
69
    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
    s->freq = freq;
81
    s->bit_rate = bitrate * 1000;
82
    avctx->frame_size = MPA_FRAME_SIZE;
83

    
84
    /* encoding freq */
85
    s->lsf = 0;
86
    for(i=0;i<3;i++) {
87
        if (ff_mpa_freq_tab[i] == freq)
88
            break;
89
        if ((ff_mpa_freq_tab[i] / 2) == freq) {
90
            s->lsf = 1;
91
            break;
92
        }
93
    }
94
    if (i == 3){
95
        av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
96
        return -1;
97
    }
98
    s->freq_index = i;
99

    
100
    /* encoding bitrate & frequency */
101
    for(i=0;i<15;i++) {
102
        if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
103
            break;
104
    }
105
    if (i == 15){
106
        av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
107
        return -1;
108
    }
109
    s->bitrate_index = i;
110

    
111
    /* compute total header size & pad bit */
112

    
113
    a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
114
    s->frame_size = ((int)a) * 8;
115

    
116
    /* frame fractional size to compute padding */
117
    s->frame_frac = 0;
118
    s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
119

    
120
    /* select the right allocation table */
121
    table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
122

    
123
    /* number of used subbands */
124
    s->sblimit = ff_mpa_sblimit_table[table];
125
    s->alloc_table = ff_mpa_alloc_tables[table];
126

    
127
#ifdef DEBUG
128
    av_log(avctx, AV_LOG_DEBUG, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
129
           bitrate, freq, s->frame_size, table, s->frame_frac_incr);
130
#endif
131

    
132
    for(i=0;i<s->nb_channels;i++)
133
        s->samples_offset[i] = 0;
134

    
135
    for(i=0;i<257;i++) {
136
        int v;
137
        v = ff_mpa_enwindow[i];
138
#if WFRAC_BITS != 16
139
        v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
140
#endif
141
        filter_bank[i] = v;
142
        if ((i & 63) != 0)
143
            v = -v;
144
        if (i != 0)
145
            filter_bank[512 - i] = v;
146
    }
147

    
148
    for(i=0;i<64;i++) {
149
        v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
150
        if (v <= 0)
151
            v = 1;
152
        scale_factor_table[i] = v;
153
#ifdef USE_FLOATS
154
        scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20);
155
#else
156
#define P 15
157
        scale_factor_shift[i] = 21 - P - (i / 3);
158
        scale_factor_mult[i] = (1 << P) * pow(2.0, (i % 3) / 3.0);
159
#endif
160
    }
161
    for(i=0;i<128;i++) {
162
        v = i - 64;
163
        if (v <= -3)
164
            v = 0;
165
        else if (v < 0)
166
            v = 1;
167
        else if (v == 0)
168
            v = 2;
169
        else if (v < 3)
170
            v = 3;
171
        else
172
            v = 4;
173
        scale_diff_table[i] = v;
174
    }
175

    
176
    for(i=0;i<17;i++) {
177
        v = ff_mpa_quant_bits[i];
178
        if (v < 0)
179
            v = -v;
180
        else
181
            v = v * 3;
182
        total_quant_bits[i] = 12 * v;
183
    }
184

    
185
    avctx->coded_frame= avcodec_alloc_frame();
186
    avctx->coded_frame->key_frame= 1;
187

    
188
    return 0;
189
}
190

    
191
/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
192
static void idct32(int *out, int *tab)
193
{
194
    int i, j;
195
    int *t, *t1, xr;
196
    const int *xp = costab32;
197

    
198
    for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
199

    
200
    t = tab + 30;
201
    t1 = tab + 2;
202
    do {
203
        t[0] += t[-4];
204
        t[1] += t[1 - 4];
205
        t -= 4;
206
    } while (t != t1);
207

    
208
    t = tab + 28;
209
    t1 = tab + 4;
210
    do {
211
        t[0] += t[-8];
212
        t[1] += t[1-8];
213
        t[2] += t[2-8];
214
        t[3] += t[3-8];
215
        t -= 8;
216
    } while (t != t1);
217

    
218
    t = tab;
219
    t1 = tab + 32;
220
    do {
221
        t[ 3] = -t[ 3];
222
        t[ 6] = -t[ 6];
223

    
224
        t[11] = -t[11];
225
        t[12] = -t[12];
226
        t[13] = -t[13];
227
        t[15] = -t[15];
228
        t += 16;
229
    } while (t != t1);
230

    
231

    
232
    t = tab;
233
    t1 = tab + 8;
234
    do {
235
        int x1, x2, x3, x4;
236

    
237
        x3 = MUL(t[16], FIX(SQRT2*0.5));
238
        x4 = t[0] - x3;
239
        x3 = t[0] + x3;
240

    
241
        x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5));
242
        x1 = MUL((t[8] - x2), xp[0]);
243
        x2 = MUL((t[8] + x2), xp[1]);
244

    
245
        t[ 0] = x3 + x1;
246
        t[ 8] = x4 - x2;
247
        t[16] = x4 + x2;
248
        t[24] = x3 - x1;
249
        t++;
250
    } while (t != t1);
251

    
252
    xp += 2;
253
    t = tab;
254
    t1 = tab + 4;
255
    do {
256
        xr = MUL(t[28],xp[0]);
257
        t[28] = (t[0] - xr);
258
        t[0] = (t[0] + xr);
259

    
260
        xr = MUL(t[4],xp[1]);
261
        t[ 4] = (t[24] - xr);
262
        t[24] = (t[24] + xr);
263

    
264
        xr = MUL(t[20],xp[2]);
265
        t[20] = (t[8] - xr);
266
        t[ 8] = (t[8] + xr);
267

    
268
        xr = MUL(t[12],xp[3]);
269
        t[12] = (t[16] - xr);
270
        t[16] = (t[16] + xr);
271
        t++;
272
    } while (t != t1);
273
    xp += 4;
274

    
275
    for (i = 0; i < 4; i++) {
276
        xr = MUL(tab[30-i*4],xp[0]);
277
        tab[30-i*4] = (tab[i*4] - xr);
278
        tab[   i*4] = (tab[i*4] + xr);
279

    
280
        xr = MUL(tab[ 2+i*4],xp[1]);
281
        tab[ 2+i*4] = (tab[28-i*4] - xr);
282
        tab[28-i*4] = (tab[28-i*4] + xr);
283

    
284
        xr = MUL(tab[31-i*4],xp[0]);
285
        tab[31-i*4] = (tab[1+i*4] - xr);
286
        tab[ 1+i*4] = (tab[1+i*4] + xr);
287

    
288
        xr = MUL(tab[ 3+i*4],xp[1]);
289
        tab[ 3+i*4] = (tab[29-i*4] - xr);
290
        tab[29-i*4] = (tab[29-i*4] + xr);
291

    
292
        xp += 2;
293
    }
294

    
295
    t = tab + 30;
296
    t1 = tab + 1;
297
    do {
298
        xr = MUL(t1[0], *xp);
299
        t1[0] = (t[0] - xr);
300
        t[0] = (t[0] + xr);
301
        t -= 2;
302
        t1 += 2;
303
        xp++;
304
    } while (t >= tab);
305

    
306
    for(i=0;i<32;i++) {
307
        out[i] = tab[bitinv32[i]];
308
    }
309
}
310

    
311
#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
312

    
313
static void filter(MpegAudioContext *s, int ch, short *samples, int incr)
314
{
315
    short *p, *q;
316
    int sum, offset, i, j;
317
    int tmp[64];
318
    int tmp1[32];
319
    int *out;
320

    
321
    //    print_pow1(samples, 1152);
322

    
323
    offset = s->samples_offset[ch];
324
    out = &s->sb_samples[ch][0][0][0];
325
    for(j=0;j<36;j++) {
326
        /* 32 samples at once */
327
        for(i=0;i<32;i++) {
328
            s->samples_buf[ch][offset + (31 - i)] = samples[0];
329
            samples += incr;
330
        }
331

    
332
        /* filter */
333
        p = s->samples_buf[ch] + offset;
334
        q = filter_bank;
335
        /* maxsum = 23169 */
336
        for(i=0;i<64;i++) {
337
            sum = p[0*64] * q[0*64];
338
            sum += p[1*64] * q[1*64];
339
            sum += p[2*64] * q[2*64];
340
            sum += p[3*64] * q[3*64];
341
            sum += p[4*64] * q[4*64];
342
            sum += p[5*64] * q[5*64];
343
            sum += p[6*64] * q[6*64];
344
            sum += p[7*64] * q[7*64];
345
            tmp[i] = sum;
346
            p++;
347
            q++;
348
        }
349
        tmp1[0] = tmp[16] >> WSHIFT;
350
        for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
351
        for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
352

    
353
        idct32(out, tmp1);
354

    
355
        /* advance of 32 samples */
356
        offset -= 32;
357
        out += 32;
358
        /* handle the wrap around */
359
        if (offset < 0) {
360
            memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
361
                    s->samples_buf[ch], (512 - 32) * 2);
362
            offset = SAMPLES_BUF_SIZE - 512;
363
        }
364
    }
365
    s->samples_offset[ch] = offset;
366

    
367
    //    print_pow(s->sb_samples, 1152);
368
}
369

    
370
static void compute_scale_factors(unsigned char scale_code[SBLIMIT],
371
                                  unsigned char scale_factors[SBLIMIT][3],
372
                                  int sb_samples[3][12][SBLIMIT],
373
                                  int sblimit)
374
{
375
    int *p, vmax, v, n, i, j, k, code;
376
    int index, d1, d2;
377
    unsigned char *sf = &scale_factors[0][0];
378

    
379
    for(j=0;j<sblimit;j++) {
380
        for(i=0;i<3;i++) {
381
            /* find the max absolute value */
382
            p = &sb_samples[i][0][j];
383
            vmax = abs(*p);
384
            for(k=1;k<12;k++) {
385
                p += SBLIMIT;
386
                v = abs(*p);
387
                if (v > vmax)
388
                    vmax = v;
389
            }
390
            /* compute the scale factor index using log 2 computations */
391
            if (vmax > 0) {
392
                n = av_log2(vmax);
393
                /* n is the position of the MSB of vmax. now
394
                   use at most 2 compares to find the index */
395
                index = (21 - n) * 3 - 3;
396
                if (index >= 0) {
397
                    while (vmax <= scale_factor_table[index+1])
398
                        index++;
399
                } else {
400
                    index = 0; /* very unlikely case of overflow */
401
                }
402
            } else {
403
                index = 62; /* value 63 is not allowed */
404
            }
405

    
406
#if 0
407
            printf("%2d:%d in=%x %x %d\n",
408
                   j, i, vmax, scale_factor_table[index], index);
409
#endif
410
            /* store the scale factor */
411
            assert(index >=0 && index <= 63);
412
            sf[i] = index;
413
        }
414

    
415
        /* compute the transmission factor : look if the scale factors
416
           are close enough to each other */
417
        d1 = scale_diff_table[sf[0] - sf[1] + 64];
418
        d2 = scale_diff_table[sf[1] - sf[2] + 64];
419

    
420
        /* handle the 25 cases */
421
        switch(d1 * 5 + d2) {
422
        case 0*5+0:
423
        case 0*5+4:
424
        case 3*5+4:
425
        case 4*5+0:
426
        case 4*5+4:
427
            code = 0;
428
            break;
429
        case 0*5+1:
430
        case 0*5+2:
431
        case 4*5+1:
432
        case 4*5+2:
433
            code = 3;
434
            sf[2] = sf[1];
435
            break;
436
        case 0*5+3:
437
        case 4*5+3:
438
            code = 3;
439
            sf[1] = sf[2];
440
            break;
441
        case 1*5+0:
442
        case 1*5+4:
443
        case 2*5+4:
444
            code = 1;
445
            sf[1] = sf[0];
446
            break;
447
        case 1*5+1:
448
        case 1*5+2:
449
        case 2*5+0:
450
        case 2*5+1:
451
        case 2*5+2:
452
            code = 2;
453
            sf[1] = sf[2] = sf[0];
454
            break;
455
        case 2*5+3:
456
        case 3*5+3:
457
            code = 2;
458
            sf[0] = sf[1] = sf[2];
459
            break;
460
        case 3*5+0:
461
        case 3*5+1:
462
        case 3*5+2:
463
            code = 2;
464
            sf[0] = sf[2] = sf[1];
465
            break;
466
        case 1*5+3:
467
            code = 2;
468
            if (sf[0] > sf[2])
469
              sf[0] = sf[2];
470
            sf[1] = sf[2] = sf[0];
471
            break;
472
        default:
473
            assert(0); //cannot happen
474
            code = 0;           /* kill warning */
475
        }
476

    
477
#if 0
478
        printf("%d: %2d %2d %2d %d %d -> %d\n", j,
479
               sf[0], sf[1], sf[2], d1, d2, code);
480
#endif
481
        scale_code[j] = code;
482
        sf += 3;
483
    }
484
}
485

    
486
/* The most important function : psycho acoustic module. In this
487
   encoder there is basically none, so this is the worst you can do,
488
   but also this is the simpler. */
489
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
490
{
491
    int i;
492

    
493
    for(i=0;i<s->sblimit;i++) {
494
        smr[i] = (int)(fixed_smr[i] * 10);
495
    }
496
}
497

    
498

    
499
#define SB_NOTALLOCATED  0
500
#define SB_ALLOCATED     1
501
#define SB_NOMORE        2
502

    
503
/* Try to maximize the smr while using a number of bits inferior to
504
   the frame size. I tried to make the code simpler, faster and
505
   smaller than other encoders :-) */
506
static void compute_bit_allocation(MpegAudioContext *s,
507
                                   short smr1[MPA_MAX_CHANNELS][SBLIMIT],
508
                                   unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
509
                                   int *padding)
510
{
511
    int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
512
    int incr;
513
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
514
    unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
515
    const unsigned char *alloc;
516

    
517
    memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
518
    memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
519
    memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
520

    
521
    /* compute frame size and padding */
522
    max_frame_size = s->frame_size;
523
    s->frame_frac += s->frame_frac_incr;
524
    if (s->frame_frac >= 65536) {
525
        s->frame_frac -= 65536;
526
        s->do_padding = 1;
527
        max_frame_size += 8;
528
    } else {
529
        s->do_padding = 0;
530
    }
531

    
532
    /* compute the header + bit alloc size */
533
    current_frame_size = 32;
534
    alloc = s->alloc_table;
535
    for(i=0;i<s->sblimit;i++) {
536
        incr = alloc[0];
537
        current_frame_size += incr * s->nb_channels;
538
        alloc += 1 << incr;
539
    }
540
    for(;;) {
541
        /* look for the subband with the largest signal to mask ratio */
542
        max_sb = -1;
543
        max_ch = -1;
544
        max_smr = 0x80000000;
545
        for(ch=0;ch<s->nb_channels;ch++) {
546
            for(i=0;i<s->sblimit;i++) {
547
                if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
548
                    max_smr = smr[ch][i];
549
                    max_sb = i;
550
                    max_ch = ch;
551
                }
552
            }
553
        }
554
#if 0
555
        printf("current=%d max=%d max_sb=%d alloc=%d\n",
556
               current_frame_size, max_frame_size, max_sb,
557
               bit_alloc[max_sb]);
558
#endif
559
        if (max_sb < 0)
560
            break;
561

    
562
        /* find alloc table entry (XXX: not optimal, should use
563
           pointer table) */
564
        alloc = s->alloc_table;
565
        for(i=0;i<max_sb;i++) {
566
            alloc += 1 << alloc[0];
567
        }
568

    
569
        if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
570
            /* nothing was coded for this band: add the necessary bits */
571
            incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
572
            incr += total_quant_bits[alloc[1]];
573
        } else {
574
            /* increments bit allocation */
575
            b = bit_alloc[max_ch][max_sb];
576
            incr = total_quant_bits[alloc[b + 1]] -
577
                total_quant_bits[alloc[b]];
578
        }
579

    
580
        if (current_frame_size + incr <= max_frame_size) {
581
            /* can increase size */
582
            b = ++bit_alloc[max_ch][max_sb];
583
            current_frame_size += incr;
584
            /* decrease smr by the resolution we added */
585
            smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
586
            /* max allocation size reached ? */
587
            if (b == ((1 << alloc[0]) - 1))
588
                subband_status[max_ch][max_sb] = SB_NOMORE;
589
            else
590
                subband_status[max_ch][max_sb] = SB_ALLOCATED;
591
        } else {
592
            /* cannot increase the size of this subband */
593
            subband_status[max_ch][max_sb] = SB_NOMORE;
594
        }
595
    }
596
    *padding = max_frame_size - current_frame_size;
597
    assert(*padding >= 0);
598

    
599
#if 0
600
    for(i=0;i<s->sblimit;i++) {
601
        printf("%d ", bit_alloc[i]);
602
    }
603
    printf("\n");
604
#endif
605
}
606

    
607
/*
608
 * Output the mpeg audio layer 2 frame. Note how the code is small
609
 * compared to other encoders :-)
610
 */
611
static void encode_frame(MpegAudioContext *s,
612
                         unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
613
                         int padding)
614
{
615
    int i, j, k, l, bit_alloc_bits, b, ch;
616
    unsigned char *sf;
617
    int q[3];
618
    PutBitContext *p = &s->pb;
619

    
620
    /* header */
621

    
622
    put_bits(p, 12, 0xfff);
623
    put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */
624
    put_bits(p, 2, 4-2);  /* layer 2 */
625
    put_bits(p, 1, 1); /* no error protection */
626
    put_bits(p, 4, s->bitrate_index);
627
    put_bits(p, 2, s->freq_index);
628
    put_bits(p, 1, s->do_padding); /* use padding */
629
    put_bits(p, 1, 0);             /* private_bit */
630
    put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
631
    put_bits(p, 2, 0); /* mode_ext */
632
    put_bits(p, 1, 0); /* no copyright */
633
    put_bits(p, 1, 1); /* original */
634
    put_bits(p, 2, 0); /* no emphasis */
635

    
636
    /* bit allocation */
637
    j = 0;
638
    for(i=0;i<s->sblimit;i++) {
639
        bit_alloc_bits = s->alloc_table[j];
640
        for(ch=0;ch<s->nb_channels;ch++) {
641
            put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
642
        }
643
        j += 1 << bit_alloc_bits;
644
    }
645

    
646
    /* scale codes */
647
    for(i=0;i<s->sblimit;i++) {
648
        for(ch=0;ch<s->nb_channels;ch++) {
649
            if (bit_alloc[ch][i])
650
                put_bits(p, 2, s->scale_code[ch][i]);
651
        }
652
    }
653

    
654
    /* scale factors */
655
    for(i=0;i<s->sblimit;i++) {
656
        for(ch=0;ch<s->nb_channels;ch++) {
657
            if (bit_alloc[ch][i]) {
658
                sf = &s->scale_factors[ch][i][0];
659
                switch(s->scale_code[ch][i]) {
660
                case 0:
661
                    put_bits(p, 6, sf[0]);
662
                    put_bits(p, 6, sf[1]);
663
                    put_bits(p, 6, sf[2]);
664
                    break;
665
                case 3:
666
                case 1:
667
                    put_bits(p, 6, sf[0]);
668
                    put_bits(p, 6, sf[2]);
669
                    break;
670
                case 2:
671
                    put_bits(p, 6, sf[0]);
672
                    break;
673
                }
674
            }
675
        }
676
    }
677

    
678
    /* quantization & write sub band samples */
679

    
680
    for(k=0;k<3;k++) {
681
        for(l=0;l<12;l+=3) {
682
            j = 0;
683
            for(i=0;i<s->sblimit;i++) {
684
                bit_alloc_bits = s->alloc_table[j];
685
                for(ch=0;ch<s->nb_channels;ch++) {
686
                    b = bit_alloc[ch][i];
687
                    if (b) {
688
                        int qindex, steps, m, sample, bits;
689
                        /* we encode 3 sub band samples of the same sub band at a time */
690
                        qindex = s->alloc_table[j+b];
691
                        steps = ff_mpa_quant_steps[qindex];
692
                        for(m=0;m<3;m++) {
693
                            sample = s->sb_samples[ch][k][l + m][i];
694
                            /* divide by scale factor */
695
#ifdef USE_FLOATS
696
                            {
697
                                float a;
698
                                a = (float)sample * scale_factor_inv_table[s->scale_factors[ch][i][k]];
699
                                q[m] = (int)((a + 1.0) * steps * 0.5);
700
                            }
701
#else
702
                            {
703
                                int q1, e, shift, mult;
704
                                e = s->scale_factors[ch][i][k];
705
                                shift = scale_factor_shift[e];
706
                                mult = scale_factor_mult[e];
707

    
708
                                /* normalize to P bits */
709
                                if (shift < 0)
710
                                    q1 = sample << (-shift);
711
                                else
712
                                    q1 = sample >> shift;
713
                                q1 = (q1 * mult) >> P;
714
                                q[m] = ((q1 + (1 << P)) * steps) >> (P + 1);
715
                            }
716
#endif
717
                            if (q[m] >= steps)
718
                                q[m] = steps - 1;
719
                            assert(q[m] >= 0 && q[m] < steps);
720
                        }
721
                        bits = ff_mpa_quant_bits[qindex];
722
                        if (bits < 0) {
723
                            /* group the 3 values to save bits */
724
                            put_bits(p, -bits,
725
                                     q[0] + steps * (q[1] + steps * q[2]));
726
#if 0
727
                            printf("%d: gr1 %d\n",
728
                                   i, q[0] + steps * (q[1] + steps * q[2]));
729
#endif
730
                        } else {
731
#if 0
732
                            printf("%d: gr3 %d %d %d\n",
733
                                   i, q[0], q[1], q[2]);
734
#endif
735
                            put_bits(p, bits, q[0]);
736
                            put_bits(p, bits, q[1]);
737
                            put_bits(p, bits, q[2]);
738
                        }
739
                    }
740
                }
741
                /* next subband in alloc table */
742
                j += 1 << bit_alloc_bits;
743
            }
744
        }
745
    }
746

    
747
    /* padding */
748
    for(i=0;i<padding;i++)
749
        put_bits(p, 1, 0);
750

    
751
    /* flush */
752
    flush_put_bits(p);
753
}
754

    
755
static int MPA_encode_frame(AVCodecContext *avctx,
756
                            unsigned char *frame, int buf_size, void *data)
757
{
758
    MpegAudioContext *s = avctx->priv_data;
759
    short *samples = data;
760
    short smr[MPA_MAX_CHANNELS][SBLIMIT];
761
    unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
762
    int padding, i;
763

    
764
    for(i=0;i<s->nb_channels;i++) {
765
        filter(s, i, samples + i, s->nb_channels);
766
    }
767

    
768
    for(i=0;i<s->nb_channels;i++) {
769
        compute_scale_factors(s->scale_code[i], s->scale_factors[i],
770
                              s->sb_samples[i], s->sblimit);
771
    }
772
    for(i=0;i<s->nb_channels;i++) {
773
        psycho_acoustic_model(s, smr[i]);
774
    }
775
    compute_bit_allocation(s, smr, bit_alloc, &padding);
776

    
777
    init_put_bits(&s->pb, frame, MPA_MAX_CODED_FRAME_SIZE);
778

    
779
    encode_frame(s, bit_alloc, padding);
780

    
781
    s->nb_samples += MPA_FRAME_SIZE;
782
    return pbBufPtr(&s->pb) - s->pb.buf;
783
}
784

    
785
static int MPA_encode_close(AVCodecContext *avctx)
786
{
787
    av_freep(&avctx->coded_frame);
788
    return 0;
789
}
790

    
791
AVCodec mp2_encoder = {
792
    "mp2",
793
    CODEC_TYPE_AUDIO,
794
    CODEC_ID_MP2,
795
    sizeof(MpegAudioContext),
796
    MPA_encode_init,
797
    MPA_encode_frame,
798
    MPA_encode_close,
799
    NULL,
800
};
801

    
802
#undef FIX