1/*
2 * The simplest mpeg audio layer 2 encoder
3 * Copyright (c) 2000, 2001 Fabrice Bellard
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22/**
23 * @file
24 * The simplest mpeg audio layer 2 encoder.
25 */
26
27#include "avcodec.h"
28#include "put_bits.h"
29
30#undef  CONFIG_MPEGAUDIO_HP
31#define CONFIG_MPEGAUDIO_HP 0
32#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
40typedef struct MpegAudioContext {
41    PutBitContext pb;
42    int nb_channels;
43    int freq, bit_rate;
44    int lsf;           /* 1 if mpeg2 low bitrate selected */
45    int bitrate_index; /* bit rate */
46    int freq_index;
47    int frame_size; /* frame size, in bits, without padding */
48    int64_t nb_samples; /* total number of samples encoded */
49    /* padding computation */
50    int frame_frac, frame_frac_incr, do_padding;
51    short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
52    int samples_offset[MPA_MAX_CHANNELS];       /* offset in samples_buf */
53    int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
54    unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
55    /* code to group 3 scale factors */
56    unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
57    int sblimit; /* number of used subbands */
58    const unsigned char *alloc_table;
59} MpegAudioContext;
60
61/* define it to use floats in quantization (I don't like floats !) */
62#define USE_FLOATS
63
64#include "mpegaudiodata.h"
65#include "mpegaudiotab.h"
66
67static av_cold int MPA_encode_init(AVCodecContext *avctx)
68{
69    MpegAudioContext *s = avctx->priv_data;
70    int freq = avctx->sample_rate;
71    int bitrate = avctx->bit_rate;
72    int channels = avctx->channels;
73    int i, v, table;
74    float a;
75
76    if (channels <= 0 || channels > 2){
77        av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
78        return -1;
79    }
80    bitrate = bitrate / 1000;
81    s->nb_channels = channels;
82    s->freq = freq;
83    s->bit_rate = bitrate * 1000;
84    avctx->frame_size = MPA_FRAME_SIZE;
85
86    /* encoding freq */
87    s->lsf = 0;
88    for(i=0;i<3;i++) {
89        if (ff_mpa_freq_tab[i] == freq)
90            break;
91        if ((ff_mpa_freq_tab[i] / 2) == freq) {
92            s->lsf = 1;
93            break;
94        }
95    }
96    if (i == 3){
97        av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
98        return -1;
99    }
100    s->freq_index = i;
101
102    /* encoding bitrate & frequency */
103    for(i=0;i<15;i++) {
104        if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
105            break;
106    }
107    if (i == 15){
108        av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
109        return -1;
110    }
111    s->bitrate_index = i;
112
113    /* compute total header size & pad bit */
114
115    a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
116    s->frame_size = ((int)a) * 8;
117
118    /* frame fractional size to compute padding */
119    s->frame_frac = 0;
120    s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
121
122    /* select the right allocation table */
123    table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
124
125    /* number of used subbands */
126    s->sblimit = ff_mpa_sblimit_table[table];
127    s->alloc_table = ff_mpa_alloc_tables[table];
128
129    dprintf(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
130            bitrate, freq, s->frame_size, table, s->frame_frac_incr);
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 */
192static 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
313static 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
370static 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 > 1) {
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. */
489static 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 :-) */
506static 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 = INT_MIN;
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 */
611static 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
755static 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 put_bits_ptr(&s->pb) - s->pb.buf;
783}
784
785static av_cold int MPA_encode_close(AVCodecContext *avctx)
786{
787    av_freep(&avctx->coded_frame);
788    return 0;
789}
790
791AVCodec mp2_encoder = {
792    "mp2",
793    AVMEDIA_TYPE_AUDIO,
794    CODEC_ID_MP2,
795    sizeof(MpegAudioContext),
796    MPA_encode_init,
797    MPA_encode_frame,
798    MPA_encode_close,
799    NULL,
800    .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
801    .supported_samplerates= (const int[]){44100, 48000,  32000, 22050, 24000, 16000, 0},
802    .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
803};
804
805#undef FIX
806