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 "libavutil/channel_layout.h"
28
29 #include "avcodec.h"
30 #include "encode.h"
31 #include "internal.h"
32 #include "put_bits.h"
33
34 #define FRAC_BITS 15 /* fractional bits for sb_samples and dct */
35 #define WFRAC_BITS 14 /* fractional bits for window */
36
37 #include "mpegaudio.h"
38 #include "mpegaudiodsp.h"
39 #include "mpegaudiodata.h"
40 #include "mpegaudiotab.h"
41
42 /* currently, cannot change these constants (need to modify
43 quantization stage) */
44 #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
45
46 #define SAMPLES_BUF_SIZE 4096
47
48 typedef struct MpegAudioContext {
49 PutBitContext pb;
50 int nb_channels;
51 int lsf; /* 1 if mpeg2 low bitrate selected */
52 int bitrate_index; /* bit rate */
53 int freq_index;
54 int frame_size; /* frame size, in bits, without padding */
55 /* padding computation */
56 int frame_frac, frame_frac_incr, do_padding;
57 short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
58 int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
59 int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
60 unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
61 /* code to group 3 scale factors */
62 unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
63 int sblimit; /* number of used subbands */
64 const unsigned char *alloc_table;
65 int16_t filter_bank[512];
66 int scale_factor_table[64];
67 unsigned char scale_diff_table[128];
68 #if USE_FLOATS
69 float scale_factor_inv_table[64];
70 #else
71 int8_t scale_factor_shift[64];
72 unsigned short scale_factor_mult[64];
73 #endif
74 unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
75 } MpegAudioContext;
76
MPA_encode_init(AVCodecContext * avctx)77 static av_cold int MPA_encode_init(AVCodecContext *avctx)
78 {
79 MpegAudioContext *s = avctx->priv_data;
80 int freq = avctx->sample_rate;
81 int bitrate = avctx->bit_rate;
82 int channels = avctx->ch_layout.nb_channels;
83 int i, v, table;
84 float a;
85
86 if (channels <= 0 || channels > 2){
87 av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
88 return AVERROR(EINVAL);
89 }
90 bitrate = bitrate / 1000;
91 s->nb_channels = channels;
92 avctx->frame_size = MPA_FRAME_SIZE;
93 avctx->initial_padding = 512 - 32 + 1;
94
95 /* encoding freq */
96 s->lsf = 0;
97 for(i=0;i<3;i++) {
98 if (ff_mpa_freq_tab[i] == freq)
99 break;
100 if ((ff_mpa_freq_tab[i] / 2) == freq) {
101 s->lsf = 1;
102 break;
103 }
104 }
105 if (i == 3){
106 av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
107 return AVERROR(EINVAL);
108 }
109 s->freq_index = i;
110
111 /* encoding bitrate & frequency */
112 for(i=1;i<15;i++) {
113 if (ff_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
114 break;
115 }
116 if (i == 15 && !avctx->bit_rate) {
117 i = 14;
118 bitrate = ff_mpa_bitrate_tab[s->lsf][1][i];
119 avctx->bit_rate = bitrate * 1000;
120 }
121 if (i == 15){
122 av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
123 return AVERROR(EINVAL);
124 }
125 s->bitrate_index = i;
126
127 /* compute total header size & pad bit */
128
129 a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
130 s->frame_size = ((int)a) * 8;
131
132 /* frame fractional size to compute padding */
133 s->frame_frac = 0;
134 s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
135
136 /* select the right allocation table */
137 table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
138
139 /* number of used subbands */
140 s->sblimit = ff_mpa_sblimit_table[table];
141 s->alloc_table = ff_mpa_alloc_tables[table];
142
143 ff_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
144 bitrate, freq, s->frame_size, table, s->frame_frac_incr);
145
146 for(i=0;i<s->nb_channels;i++)
147 s->samples_offset[i] = 0;
148
149 for(i=0;i<257;i++) {
150 int v;
151 v = ff_mpa_enwindow[i];
152 #if WFRAC_BITS != 16
153 v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
154 #endif
155 s->filter_bank[i] = v;
156 if ((i & 63) != 0)
157 v = -v;
158 if (i != 0)
159 s->filter_bank[512 - i] = v;
160 }
161
162 for(i=0;i<64;i++) {
163 v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
164 if (v <= 0)
165 v = 1;
166 s->scale_factor_table[i] = v;
167 #if USE_FLOATS
168 s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
169 #else
170 #define P 15
171 s->scale_factor_shift[i] = 21 - P - (i / 3);
172 s->scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);
173 #endif
174 }
175 for(i=0;i<128;i++) {
176 v = i - 64;
177 if (v <= -3)
178 v = 0;
179 else if (v < 0)
180 v = 1;
181 else if (v == 0)
182 v = 2;
183 else if (v < 3)
184 v = 3;
185 else
186 v = 4;
187 s->scale_diff_table[i] = v;
188 }
189
190 for(i=0;i<17;i++) {
191 v = ff_mpa_quant_bits[i];
192 if (v < 0)
193 v = -v;
194 else
195 v = v * 3;
196 s->total_quant_bits[i] = 12 * v;
197 }
198
199 return 0;
200 }
201
202 /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
idct32(int * out,int * tab)203 static void idct32(int *out, int *tab)
204 {
205 int i, j;
206 int *t, *t1, xr;
207 const int *xp = costab32;
208
209 for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
210
211 t = tab + 30;
212 t1 = tab + 2;
213 do {
214 t[0] += t[-4];
215 t[1] += t[1 - 4];
216 t -= 4;
217 } while (t != t1);
218
219 t = tab + 28;
220 t1 = tab + 4;
221 do {
222 t[0] += t[-8];
223 t[1] += t[1-8];
224 t[2] += t[2-8];
225 t[3] += t[3-8];
226 t -= 8;
227 } while (t != t1);
228
229 t = tab;
230 t1 = tab + 32;
231 do {
232 t[ 3] = -t[ 3];
233 t[ 6] = -t[ 6];
234
235 t[11] = -t[11];
236 t[12] = -t[12];
237 t[13] = -t[13];
238 t[15] = -t[15];
239 t += 16;
240 } while (t != t1);
241
242
243 t = tab;
244 t1 = tab + 8;
245 do {
246 int x1, x2, x3, x4;
247
248 x3 = MUL(t[16], FIX(M_SQRT2*0.5));
249 x4 = t[0] - x3;
250 x3 = t[0] + x3;
251
252 x2 = MUL(-(t[24] + t[8]), FIX(M_SQRT2*0.5));
253 x1 = MUL((t[8] - x2), xp[0]);
254 x2 = MUL((t[8] + x2), xp[1]);
255
256 t[ 0] = x3 + x1;
257 t[ 8] = x4 - x2;
258 t[16] = x4 + x2;
259 t[24] = x3 - x1;
260 t++;
261 } while (t != t1);
262
263 xp += 2;
264 t = tab;
265 t1 = tab + 4;
266 do {
267 xr = MUL(t[28],xp[0]);
268 t[28] = (t[0] - xr);
269 t[0] = (t[0] + xr);
270
271 xr = MUL(t[4],xp[1]);
272 t[ 4] = (t[24] - xr);
273 t[24] = (t[24] + xr);
274
275 xr = MUL(t[20],xp[2]);
276 t[20] = (t[8] - xr);
277 t[ 8] = (t[8] + xr);
278
279 xr = MUL(t[12],xp[3]);
280 t[12] = (t[16] - xr);
281 t[16] = (t[16] + xr);
282 t++;
283 } while (t != t1);
284 xp += 4;
285
286 for (i = 0; i < 4; i++) {
287 xr = MUL(tab[30-i*4],xp[0]);
288 tab[30-i*4] = (tab[i*4] - xr);
289 tab[ i*4] = (tab[i*4] + xr);
290
291 xr = MUL(tab[ 2+i*4],xp[1]);
292 tab[ 2+i*4] = (tab[28-i*4] - xr);
293 tab[28-i*4] = (tab[28-i*4] + xr);
294
295 xr = MUL(tab[31-i*4],xp[0]);
296 tab[31-i*4] = (tab[1+i*4] - xr);
297 tab[ 1+i*4] = (tab[1+i*4] + xr);
298
299 xr = MUL(tab[ 3+i*4],xp[1]);
300 tab[ 3+i*4] = (tab[29-i*4] - xr);
301 tab[29-i*4] = (tab[29-i*4] + xr);
302
303 xp += 2;
304 }
305
306 t = tab + 30;
307 t1 = tab + 1;
308 do {
309 xr = MUL(t1[0], *xp);
310 t1[0] = (t[0] - xr);
311 t[0] = (t[0] + xr);
312 t -= 2;
313 t1 += 2;
314 xp++;
315 } while (t >= tab);
316
317 for(i=0;i<32;i++) {
318 out[i] = tab[bitinv32[i]];
319 }
320 }
321
322 #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
323
filter(MpegAudioContext * s,int ch,const short * samples,int incr)324 static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
325 {
326 short *p, *q;
327 int sum, offset, i, j;
328 int tmp[64];
329 int tmp1[32];
330 int *out;
331
332 offset = s->samples_offset[ch];
333 out = &s->sb_samples[ch][0][0][0];
334 for(j=0;j<36;j++) {
335 /* 32 samples at once */
336 for(i=0;i<32;i++) {
337 s->samples_buf[ch][offset + (31 - i)] = samples[0];
338 samples += incr;
339 }
340
341 /* filter */
342 p = s->samples_buf[ch] + offset;
343 q = s->filter_bank;
344 /* maxsum = 23169 */
345 for(i=0;i<64;i++) {
346 sum = p[0*64] * q[0*64];
347 sum += p[1*64] * q[1*64];
348 sum += p[2*64] * q[2*64];
349 sum += p[3*64] * q[3*64];
350 sum += p[4*64] * q[4*64];
351 sum += p[5*64] * q[5*64];
352 sum += p[6*64] * q[6*64];
353 sum += p[7*64] * q[7*64];
354 tmp[i] = sum;
355 p++;
356 q++;
357 }
358 tmp1[0] = tmp[16] >> WSHIFT;
359 for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
360 for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
361
362 idct32(out, tmp1);
363
364 /* advance of 32 samples */
365 offset -= 32;
366 out += 32;
367 /* handle the wrap around */
368 if (offset < 0) {
369 memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
370 s->samples_buf[ch], (512 - 32) * 2);
371 offset = SAMPLES_BUF_SIZE - 512;
372 }
373 }
374 s->samples_offset[ch] = offset;
375 }
376
compute_scale_factors(MpegAudioContext * s,unsigned char scale_code[SBLIMIT],unsigned char scale_factors[SBLIMIT][3],int sb_samples[3][12][SBLIMIT],int sblimit)377 static void compute_scale_factors(MpegAudioContext *s,
378 unsigned char scale_code[SBLIMIT],
379 unsigned char scale_factors[SBLIMIT][3],
380 int sb_samples[3][12][SBLIMIT],
381 int sblimit)
382 {
383 int *p, vmax, v, n, i, j, k, code;
384 int index, d1, d2;
385 unsigned char *sf = &scale_factors[0][0];
386
387 for(j=0;j<sblimit;j++) {
388 for(i=0;i<3;i++) {
389 /* find the max absolute value */
390 p = &sb_samples[i][0][j];
391 vmax = abs(*p);
392 for(k=1;k<12;k++) {
393 p += SBLIMIT;
394 v = abs(*p);
395 if (v > vmax)
396 vmax = v;
397 }
398 /* compute the scale factor index using log 2 computations */
399 if (vmax > 1) {
400 n = av_log2(vmax);
401 /* n is the position of the MSB of vmax. now
402 use at most 2 compares to find the index */
403 index = (21 - n) * 3 - 3;
404 if (index >= 0) {
405 while (vmax <= s->scale_factor_table[index+1])
406 index++;
407 } else {
408 index = 0; /* very unlikely case of overflow */
409 }
410 } else {
411 index = 62; /* value 63 is not allowed */
412 }
413
414 ff_dlog(NULL, "%2d:%d in=%x %x %d\n",
415 j, i, vmax, s->scale_factor_table[index], index);
416 /* store the scale factor */
417 av_assert2(index >=0 && index <= 63);
418 sf[i] = index;
419 }
420
421 /* compute the transmission factor : look if the scale factors
422 are close enough to each other */
423 d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
424 d2 = s->scale_diff_table[sf[1] - sf[2] + 64];
425
426 /* handle the 25 cases */
427 switch(d1 * 5 + d2) {
428 case 0*5+0:
429 case 0*5+4:
430 case 3*5+4:
431 case 4*5+0:
432 case 4*5+4:
433 code = 0;
434 break;
435 case 0*5+1:
436 case 0*5+2:
437 case 4*5+1:
438 case 4*5+2:
439 code = 3;
440 sf[2] = sf[1];
441 break;
442 case 0*5+3:
443 case 4*5+3:
444 code = 3;
445 sf[1] = sf[2];
446 break;
447 case 1*5+0:
448 case 1*5+4:
449 case 2*5+4:
450 code = 1;
451 sf[1] = sf[0];
452 break;
453 case 1*5+1:
454 case 1*5+2:
455 case 2*5+0:
456 case 2*5+1:
457 case 2*5+2:
458 code = 2;
459 sf[1] = sf[2] = sf[0];
460 break;
461 case 2*5+3:
462 case 3*5+3:
463 code = 2;
464 sf[0] = sf[1] = sf[2];
465 break;
466 case 3*5+0:
467 case 3*5+1:
468 case 3*5+2:
469 code = 2;
470 sf[0] = sf[2] = sf[1];
471 break;
472 case 1*5+3:
473 code = 2;
474 if (sf[0] > sf[2])
475 sf[0] = sf[2];
476 sf[1] = sf[2] = sf[0];
477 break;
478 default:
479 av_assert2(0); //cannot happen
480 code = 0; /* kill warning */
481 }
482
483 ff_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,
484 sf[0], sf[1], sf[2], d1, d2, code);
485 scale_code[j] = code;
486 sf += 3;
487 }
488 }
489
490 /* The most important function : psycho acoustic module. In this
491 encoder there is basically none, so this is the worst you can do,
492 but also this is the simpler. */
psycho_acoustic_model(MpegAudioContext * s,short smr[SBLIMIT])493 static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
494 {
495 int i;
496
497 for(i=0;i<s->sblimit;i++) {
498 smr[i] = (int)(fixed_smr[i] * 10);
499 }
500 }
501
502
503 #define SB_NOTALLOCATED 0
504 #define SB_ALLOCATED 1
505 #define SB_NOMORE 2
506
507 /* Try to maximize the smr while using a number of bits inferior to
508 the frame size. I tried to make the code simpler, faster and
509 smaller than other encoders :-) */
compute_bit_allocation(MpegAudioContext * s,short smr1[MPA_MAX_CHANNELS][SBLIMIT],unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],int * padding)510 static void compute_bit_allocation(MpegAudioContext *s,
511 short smr1[MPA_MAX_CHANNELS][SBLIMIT],
512 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
513 int *padding)
514 {
515 int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
516 int incr;
517 short smr[MPA_MAX_CHANNELS][SBLIMIT];
518 unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
519 const unsigned char *alloc;
520
521 memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
522 memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
523 memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
524
525 /* compute frame size and padding */
526 max_frame_size = s->frame_size;
527 s->frame_frac += s->frame_frac_incr;
528 if (s->frame_frac >= 65536) {
529 s->frame_frac -= 65536;
530 s->do_padding = 1;
531 max_frame_size += 8;
532 } else {
533 s->do_padding = 0;
534 }
535
536 /* compute the header + bit alloc size */
537 current_frame_size = 32;
538 alloc = s->alloc_table;
539 for(i=0;i<s->sblimit;i++) {
540 incr = alloc[0];
541 current_frame_size += incr * s->nb_channels;
542 alloc += 1 << incr;
543 }
544 for(;;) {
545 /* look for the subband with the largest signal to mask ratio */
546 max_sb = -1;
547 max_ch = -1;
548 max_smr = INT_MIN;
549 for(ch=0;ch<s->nb_channels;ch++) {
550 for(i=0;i<s->sblimit;i++) {
551 if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
552 max_smr = smr[ch][i];
553 max_sb = i;
554 max_ch = ch;
555 }
556 }
557 }
558 if (max_sb < 0)
559 break;
560 ff_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n",
561 current_frame_size, max_frame_size, max_sb, max_ch,
562 bit_alloc[max_ch][max_sb]);
563
564 /* find alloc table entry (XXX: not optimal, should use
565 pointer table) */
566 alloc = s->alloc_table;
567 for(i=0;i<max_sb;i++) {
568 alloc += 1 << alloc[0];
569 }
570
571 if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
572 /* nothing was coded for this band: add the necessary bits */
573 incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
574 incr += s->total_quant_bits[alloc[1]];
575 } else {
576 /* increments bit allocation */
577 b = bit_alloc[max_ch][max_sb];
578 incr = s->total_quant_bits[alloc[b + 1]] -
579 s->total_quant_bits[alloc[b]];
580 }
581
582 if (current_frame_size + incr <= max_frame_size) {
583 /* can increase size */
584 b = ++bit_alloc[max_ch][max_sb];
585 current_frame_size += incr;
586 /* decrease smr by the resolution we added */
587 smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
588 /* max allocation size reached ? */
589 if (b == ((1 << alloc[0]) - 1))
590 subband_status[max_ch][max_sb] = SB_NOMORE;
591 else
592 subband_status[max_ch][max_sb] = SB_ALLOCATED;
593 } else {
594 /* cannot increase the size of this subband */
595 subband_status[max_ch][max_sb] = SB_NOMORE;
596 }
597 }
598 *padding = max_frame_size - current_frame_size;
599 av_assert0(*padding >= 0);
600 }
601
602 /*
603 * Output the MPEG audio layer 2 frame. Note how the code is small
604 * compared to other encoders :-)
605 */
encode_frame(MpegAudioContext * s,unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],int padding)606 static void encode_frame(MpegAudioContext *s,
607 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
608 int padding)
609 {
610 int i, j, k, l, bit_alloc_bits, b, ch;
611 unsigned char *sf;
612 int q[3];
613 PutBitContext *p = &s->pb;
614
615 /* header */
616
617 put_bits(p, 12, 0xfff);
618 put_bits(p, 1, 1 - s->lsf); /* 1 = MPEG-1 ID, 0 = MPEG-2 lsf ID */
619 put_bits(p, 2, 4-2); /* layer 2 */
620 put_bits(p, 1, 1); /* no error protection */
621 put_bits(p, 4, s->bitrate_index);
622 put_bits(p, 2, s->freq_index);
623 put_bits(p, 1, s->do_padding); /* use padding */
624 put_bits(p, 1, 0); /* private_bit */
625 put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
626 put_bits(p, 2, 0); /* mode_ext */
627 put_bits(p, 1, 0); /* no copyright */
628 put_bits(p, 1, 1); /* original */
629 put_bits(p, 2, 0); /* no emphasis */
630
631 /* bit allocation */
632 j = 0;
633 for(i=0;i<s->sblimit;i++) {
634 bit_alloc_bits = s->alloc_table[j];
635 for(ch=0;ch<s->nb_channels;ch++) {
636 put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
637 }
638 j += 1 << bit_alloc_bits;
639 }
640
641 /* scale codes */
642 for(i=0;i<s->sblimit;i++) {
643 for(ch=0;ch<s->nb_channels;ch++) {
644 if (bit_alloc[ch][i])
645 put_bits(p, 2, s->scale_code[ch][i]);
646 }
647 }
648
649 /* scale factors */
650 for(i=0;i<s->sblimit;i++) {
651 for(ch=0;ch<s->nb_channels;ch++) {
652 if (bit_alloc[ch][i]) {
653 sf = &s->scale_factors[ch][i][0];
654 switch(s->scale_code[ch][i]) {
655 case 0:
656 put_bits(p, 6, sf[0]);
657 put_bits(p, 6, sf[1]);
658 put_bits(p, 6, sf[2]);
659 break;
660 case 3:
661 case 1:
662 put_bits(p, 6, sf[0]);
663 put_bits(p, 6, sf[2]);
664 break;
665 case 2:
666 put_bits(p, 6, sf[0]);
667 break;
668 }
669 }
670 }
671 }
672
673 /* quantization & write sub band samples */
674
675 for(k=0;k<3;k++) {
676 for(l=0;l<12;l+=3) {
677 j = 0;
678 for(i=0;i<s->sblimit;i++) {
679 bit_alloc_bits = s->alloc_table[j];
680 for(ch=0;ch<s->nb_channels;ch++) {
681 b = bit_alloc[ch][i];
682 if (b) {
683 int qindex, steps, m, sample, bits;
684 /* we encode 3 sub band samples of the same sub band at a time */
685 qindex = s->alloc_table[j+b];
686 steps = ff_mpa_quant_steps[qindex];
687 for(m=0;m<3;m++) {
688 sample = s->sb_samples[ch][k][l + m][i];
689 /* divide by scale factor */
690 #if USE_FLOATS
691 {
692 float a;
693 a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]];
694 q[m] = (int)((a + 1.0) * steps * 0.5);
695 }
696 #else
697 {
698 int q1, e, shift, mult;
699 e = s->scale_factors[ch][i][k];
700 shift = s->scale_factor_shift[e];
701 mult = s->scale_factor_mult[e];
702
703 /* normalize to P bits */
704 if (shift < 0)
705 q1 = sample * (1 << -shift);
706 else
707 q1 = sample >> shift;
708 q1 = (q1 * mult) >> P;
709 q1 += 1 << P;
710 if (q1 < 0)
711 q1 = 0;
712 q[m] = (q1 * (unsigned)steps) >> (P + 1);
713 }
714 #endif
715 if (q[m] >= steps)
716 q[m] = steps - 1;
717 av_assert2(q[m] >= 0 && q[m] < steps);
718 }
719 bits = ff_mpa_quant_bits[qindex];
720 if (bits < 0) {
721 /* group the 3 values to save bits */
722 put_bits(p, -bits,
723 q[0] + steps * (q[1] + steps * q[2]));
724 } else {
725 put_bits(p, bits, q[0]);
726 put_bits(p, bits, q[1]);
727 put_bits(p, bits, q[2]);
728 }
729 }
730 }
731 /* next subband in alloc table */
732 j += 1 << bit_alloc_bits;
733 }
734 }
735 }
736
737 /* padding */
738 for(i=0;i<padding;i++)
739 put_bits(p, 1, 0);
740 }
741
MPA_encode_frame(AVCodecContext * avctx,AVPacket * avpkt,const AVFrame * frame,int * got_packet_ptr)742 static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
743 const AVFrame *frame, int *got_packet_ptr)
744 {
745 MpegAudioContext *s = avctx->priv_data;
746 const int16_t *samples = (const int16_t *)frame->data[0];
747 short smr[MPA_MAX_CHANNELS][SBLIMIT];
748 unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
749 int padding, i, ret;
750
751 for(i=0;i<s->nb_channels;i++) {
752 filter(s, i, samples + i, s->nb_channels);
753 }
754
755 for(i=0;i<s->nb_channels;i++) {
756 compute_scale_factors(s, s->scale_code[i], s->scale_factors[i],
757 s->sb_samples[i], s->sblimit);
758 }
759 for(i=0;i<s->nb_channels;i++) {
760 psycho_acoustic_model(s, smr[i]);
761 }
762 compute_bit_allocation(s, smr, bit_alloc, &padding);
763
764 if ((ret = ff_alloc_packet(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0)
765 return ret;
766
767 init_put_bits(&s->pb, avpkt->data, avpkt->size);
768
769 encode_frame(s, bit_alloc, padding);
770
771 /* flush */
772 flush_put_bits(&s->pb);
773 avpkt->size = put_bytes_output(&s->pb);
774
775 if (frame->pts != AV_NOPTS_VALUE)
776 avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
777
778 *got_packet_ptr = 1;
779 return 0;
780 }
781
782 static const FFCodecDefault mp2_defaults[] = {
783 { "b", "0" },
784 { NULL },
785 };
786
787