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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 "internal.h" | |
31 | #include "put_bits.h" | |
32 | ||
33 | #define FRAC_BITS 15 /* fractional bits for sb_samples and dct */ | |
34 | #define WFRAC_BITS 14 /* fractional bits for window */ | |
35 | ||
36 | #include "mpegaudio.h" | |
37 | #include "mpegaudiodsp.h" | |
38 | #include "mpegaudiodata.h" | |
39 | #include "mpegaudiotab.h" | |
40 | ||
41 | /* currently, cannot change these constants (need to modify | |
42 | quantization stage) */ | |
43 | #define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS) | |
44 | ||
45 | #define SAMPLES_BUF_SIZE 4096 | |
46 | ||
47 | typedef struct MpegAudioContext { | |
48 | PutBitContext pb; | |
49 | int nb_channels; | |
50 | int lsf; /* 1 if mpeg2 low bitrate selected */ | |
51 | int bitrate_index; /* bit rate */ | |
52 | int freq_index; | |
53 | int frame_size; /* frame size, in bits, without padding */ | |
54 | /* padding computation */ | |
55 | int frame_frac, frame_frac_incr, do_padding; | |
56 | short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */ | |
57 | int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */ | |
58 | int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT]; | |
59 | unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */ | |
60 | /* code to group 3 scale factors */ | |
61 | unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; | |
62 | int sblimit; /* number of used subbands */ | |
63 | const unsigned char *alloc_table; | |
64 | int16_t filter_bank[512]; | |
65 | int scale_factor_table[64]; | |
66 | unsigned char scale_diff_table[128]; | |
67 | #if USE_FLOATS | |
68 | float scale_factor_inv_table[64]; | |
69 | #else | |
70 | int8_t scale_factor_shift[64]; | |
71 | unsigned short scale_factor_mult[64]; | |
72 | #endif | |
73 | unsigned short total_quant_bits[17]; /* total number of bits per allocation group */ | |
74 | } MpegAudioContext; | |
75 | ||
76 | static av_cold int MPA_encode_init(AVCodecContext *avctx) | |
77 | { | |
78 | MpegAudioContext *s = avctx->priv_data; | |
79 | int freq = avctx->sample_rate; | |
80 | int bitrate = avctx->bit_rate; | |
81 | int channels = avctx->channels; | |
82 | int i, v, table; | |
83 | float a; | |
84 | ||
85 | if (channels <= 0 || channels > 2){ | |
86 | av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels); | |
87 | return AVERROR(EINVAL); | |
88 | } | |
89 | bitrate = bitrate / 1000; | |
90 | s->nb_channels = channels; | |
91 | avctx->frame_size = MPA_FRAME_SIZE; | |
f6fa7814 | 92 | avctx->initial_padding = 512 - 32 + 1; |
2ba45a60 DM |
93 | |
94 | /* encoding freq */ | |
95 | s->lsf = 0; | |
96 | for(i=0;i<3;i++) { | |
97 | if (avpriv_mpa_freq_tab[i] == freq) | |
98 | break; | |
99 | if ((avpriv_mpa_freq_tab[i] / 2) == freq) { | |
100 | s->lsf = 1; | |
101 | break; | |
102 | } | |
103 | } | |
104 | if (i == 3){ | |
105 | av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq); | |
106 | return AVERROR(EINVAL); | |
107 | } | |
108 | s->freq_index = i; | |
109 | ||
110 | /* encoding bitrate & frequency */ | |
111 | for(i=1;i<15;i++) { | |
112 | if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate) | |
113 | break; | |
114 | } | |
115 | if (i == 15 && !avctx->bit_rate) { | |
116 | i = 14; | |
117 | bitrate = avpriv_mpa_bitrate_tab[s->lsf][1][i]; | |
118 | avctx->bit_rate = bitrate * 1000; | |
119 | } | |
120 | if (i == 15){ | |
121 | av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate); | |
122 | return AVERROR(EINVAL); | |
123 | } | |
124 | s->bitrate_index = i; | |
125 | ||
126 | /* compute total header size & pad bit */ | |
127 | ||
128 | a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0); | |
129 | s->frame_size = ((int)a) * 8; | |
130 | ||
131 | /* frame fractional size to compute padding */ | |
132 | s->frame_frac = 0; | |
133 | s->frame_frac_incr = (int)((a - floor(a)) * 65536.0); | |
134 | ||
135 | /* select the right allocation table */ | |
136 | table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf); | |
137 | ||
138 | /* number of used subbands */ | |
139 | s->sblimit = ff_mpa_sblimit_table[table]; | |
140 | s->alloc_table = ff_mpa_alloc_tables[table]; | |
141 | ||
142 | av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", | |
143 | bitrate, freq, s->frame_size, table, s->frame_frac_incr); | |
144 | ||
145 | for(i=0;i<s->nb_channels;i++) | |
146 | s->samples_offset[i] = 0; | |
147 | ||
148 | for(i=0;i<257;i++) { | |
149 | int v; | |
150 | v = ff_mpa_enwindow[i]; | |
151 | #if WFRAC_BITS != 16 | |
152 | v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS); | |
153 | #endif | |
154 | s->filter_bank[i] = v; | |
155 | if ((i & 63) != 0) | |
156 | v = -v; | |
157 | if (i != 0) | |
158 | s->filter_bank[512 - i] = v; | |
159 | } | |
160 | ||
161 | for(i=0;i<64;i++) { | |
162 | v = (int)(exp2((3 - i) / 3.0) * (1 << 20)); | |
163 | if (v <= 0) | |
164 | v = 1; | |
165 | s->scale_factor_table[i] = v; | |
166 | #if USE_FLOATS | |
167 | s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20); | |
168 | #else | |
169 | #define P 15 | |
170 | s->scale_factor_shift[i] = 21 - P - (i / 3); | |
171 | s->scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0); | |
172 | #endif | |
173 | } | |
174 | for(i=0;i<128;i++) { | |
175 | v = i - 64; | |
176 | if (v <= -3) | |
177 | v = 0; | |
178 | else if (v < 0) | |
179 | v = 1; | |
180 | else if (v == 0) | |
181 | v = 2; | |
182 | else if (v < 3) | |
183 | v = 3; | |
184 | else | |
185 | v = 4; | |
186 | s->scale_diff_table[i] = v; | |
187 | } | |
188 | ||
189 | for(i=0;i<17;i++) { | |
190 | v = ff_mpa_quant_bits[i]; | |
191 | if (v < 0) | |
192 | v = -v; | |
193 | else | |
194 | v = v * 3; | |
195 | s->total_quant_bits[i] = 12 * v; | |
196 | } | |
197 | ||
198 | return 0; | |
199 | } | |
200 | ||
201 | /* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */ | |
202 | static void idct32(int *out, int *tab) | |
203 | { | |
204 | int i, j; | |
205 | int *t, *t1, xr; | |
206 | const int *xp = costab32; | |
207 | ||
208 | for(j=31;j>=3;j-=2) tab[j] += tab[j - 2]; | |
209 | ||
210 | t = tab + 30; | |
211 | t1 = tab + 2; | |
212 | do { | |
213 | t[0] += t[-4]; | |
214 | t[1] += t[1 - 4]; | |
215 | t -= 4; | |
216 | } while (t != t1); | |
217 | ||
218 | t = tab + 28; | |
219 | t1 = tab + 4; | |
220 | do { | |
221 | t[0] += t[-8]; | |
222 | t[1] += t[1-8]; | |
223 | t[2] += t[2-8]; | |
224 | t[3] += t[3-8]; | |
225 | t -= 8; | |
226 | } while (t != t1); | |
227 | ||
228 | t = tab; | |
229 | t1 = tab + 32; | |
230 | do { | |
231 | t[ 3] = -t[ 3]; | |
232 | t[ 6] = -t[ 6]; | |
233 | ||
234 | t[11] = -t[11]; | |
235 | t[12] = -t[12]; | |
236 | t[13] = -t[13]; | |
237 | t[15] = -t[15]; | |
238 | t += 16; | |
239 | } while (t != t1); | |
240 | ||
241 | ||
242 | t = tab; | |
243 | t1 = tab + 8; | |
244 | do { | |
245 | int x1, x2, x3, x4; | |
246 | ||
247 | x3 = MUL(t[16], FIX(SQRT2*0.5)); | |
248 | x4 = t[0] - x3; | |
249 | x3 = t[0] + x3; | |
250 | ||
251 | x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5)); | |
252 | x1 = MUL((t[8] - x2), xp[0]); | |
253 | x2 = MUL((t[8] + x2), xp[1]); | |
254 | ||
255 | t[ 0] = x3 + x1; | |
256 | t[ 8] = x4 - x2; | |
257 | t[16] = x4 + x2; | |
258 | t[24] = x3 - x1; | |
259 | t++; | |
260 | } while (t != t1); | |
261 | ||
262 | xp += 2; | |
263 | t = tab; | |
264 | t1 = tab + 4; | |
265 | do { | |
266 | xr = MUL(t[28],xp[0]); | |
267 | t[28] = (t[0] - xr); | |
268 | t[0] = (t[0] + xr); | |
269 | ||
270 | xr = MUL(t[4],xp[1]); | |
271 | t[ 4] = (t[24] - xr); | |
272 | t[24] = (t[24] + xr); | |
273 | ||
274 | xr = MUL(t[20],xp[2]); | |
275 | t[20] = (t[8] - xr); | |
276 | t[ 8] = (t[8] + xr); | |
277 | ||
278 | xr = MUL(t[12],xp[3]); | |
279 | t[12] = (t[16] - xr); | |
280 | t[16] = (t[16] + xr); | |
281 | t++; | |
282 | } while (t != t1); | |
283 | xp += 4; | |
284 | ||
285 | for (i = 0; i < 4; i++) { | |
286 | xr = MUL(tab[30-i*4],xp[0]); | |
287 | tab[30-i*4] = (tab[i*4] - xr); | |
288 | tab[ i*4] = (tab[i*4] + xr); | |
289 | ||
290 | xr = MUL(tab[ 2+i*4],xp[1]); | |
291 | tab[ 2+i*4] = (tab[28-i*4] - xr); | |
292 | tab[28-i*4] = (tab[28-i*4] + xr); | |
293 | ||
294 | xr = MUL(tab[31-i*4],xp[0]); | |
295 | tab[31-i*4] = (tab[1+i*4] - xr); | |
296 | tab[ 1+i*4] = (tab[1+i*4] + xr); | |
297 | ||
298 | xr = MUL(tab[ 3+i*4],xp[1]); | |
299 | tab[ 3+i*4] = (tab[29-i*4] - xr); | |
300 | tab[29-i*4] = (tab[29-i*4] + xr); | |
301 | ||
302 | xp += 2; | |
303 | } | |
304 | ||
305 | t = tab + 30; | |
306 | t1 = tab + 1; | |
307 | do { | |
308 | xr = MUL(t1[0], *xp); | |
309 | t1[0] = (t[0] - xr); | |
310 | t[0] = (t[0] + xr); | |
311 | t -= 2; | |
312 | t1 += 2; | |
313 | xp++; | |
314 | } while (t >= tab); | |
315 | ||
316 | for(i=0;i<32;i++) { | |
317 | out[i] = tab[bitinv32[i]]; | |
318 | } | |
319 | } | |
320 | ||
321 | #define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS) | |
322 | ||
323 | static void filter(MpegAudioContext *s, int ch, const short *samples, int incr) | |
324 | { | |
325 | short *p, *q; | |
326 | int sum, offset, i, j; | |
327 | int tmp[64]; | |
328 | int tmp1[32]; | |
329 | int *out; | |
330 | ||
331 | offset = s->samples_offset[ch]; | |
332 | out = &s->sb_samples[ch][0][0][0]; | |
333 | for(j=0;j<36;j++) { | |
334 | /* 32 samples at once */ | |
335 | for(i=0;i<32;i++) { | |
336 | s->samples_buf[ch][offset + (31 - i)] = samples[0]; | |
337 | samples += incr; | |
338 | } | |
339 | ||
340 | /* filter */ | |
341 | p = s->samples_buf[ch] + offset; | |
342 | q = s->filter_bank; | |
343 | /* maxsum = 23169 */ | |
344 | for(i=0;i<64;i++) { | |
345 | sum = p[0*64] * q[0*64]; | |
346 | sum += p[1*64] * q[1*64]; | |
347 | sum += p[2*64] * q[2*64]; | |
348 | sum += p[3*64] * q[3*64]; | |
349 | sum += p[4*64] * q[4*64]; | |
350 | sum += p[5*64] * q[5*64]; | |
351 | sum += p[6*64] * q[6*64]; | |
352 | sum += p[7*64] * q[7*64]; | |
353 | tmp[i] = sum; | |
354 | p++; | |
355 | q++; | |
356 | } | |
357 | tmp1[0] = tmp[16] >> WSHIFT; | |
358 | for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT; | |
359 | for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT; | |
360 | ||
361 | idct32(out, tmp1); | |
362 | ||
363 | /* advance of 32 samples */ | |
364 | offset -= 32; | |
365 | out += 32; | |
366 | /* handle the wrap around */ | |
367 | if (offset < 0) { | |
368 | memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32), | |
369 | s->samples_buf[ch], (512 - 32) * 2); | |
370 | offset = SAMPLES_BUF_SIZE - 512; | |
371 | } | |
372 | } | |
373 | s->samples_offset[ch] = offset; | |
374 | } | |
375 | ||
376 | static void compute_scale_factors(MpegAudioContext *s, | |
377 | unsigned char scale_code[SBLIMIT], | |
378 | unsigned char scale_factors[SBLIMIT][3], | |
379 | int sb_samples[3][12][SBLIMIT], | |
380 | int sblimit) | |
381 | { | |
382 | int *p, vmax, v, n, i, j, k, code; | |
383 | int index, d1, d2; | |
384 | unsigned char *sf = &scale_factors[0][0]; | |
385 | ||
386 | for(j=0;j<sblimit;j++) { | |
387 | for(i=0;i<3;i++) { | |
388 | /* find the max absolute value */ | |
389 | p = &sb_samples[i][0][j]; | |
390 | vmax = abs(*p); | |
391 | for(k=1;k<12;k++) { | |
392 | p += SBLIMIT; | |
393 | v = abs(*p); | |
394 | if (v > vmax) | |
395 | vmax = v; | |
396 | } | |
397 | /* compute the scale factor index using log 2 computations */ | |
398 | if (vmax > 1) { | |
399 | n = av_log2(vmax); | |
400 | /* n is the position of the MSB of vmax. now | |
401 | use at most 2 compares to find the index */ | |
402 | index = (21 - n) * 3 - 3; | |
403 | if (index >= 0) { | |
404 | while (vmax <= s->scale_factor_table[index+1]) | |
405 | index++; | |
406 | } else { | |
407 | index = 0; /* very unlikely case of overflow */ | |
408 | } | |
409 | } else { | |
410 | index = 62; /* value 63 is not allowed */ | |
411 | } | |
412 | ||
413 | av_dlog(NULL, "%2d:%d in=%x %x %d\n", | |
414 | j, i, vmax, s->scale_factor_table[index], index); | |
415 | /* store the scale factor */ | |
416 | av_assert2(index >=0 && index <= 63); | |
417 | sf[i] = index; | |
418 | } | |
419 | ||
420 | /* compute the transmission factor : look if the scale factors | |
421 | are close enough to each other */ | |
422 | d1 = s->scale_diff_table[sf[0] - sf[1] + 64]; | |
423 | d2 = s->scale_diff_table[sf[1] - sf[2] + 64]; | |
424 | ||
425 | /* handle the 25 cases */ | |
426 | switch(d1 * 5 + d2) { | |
427 | case 0*5+0: | |
428 | case 0*5+4: | |
429 | case 3*5+4: | |
430 | case 4*5+0: | |
431 | case 4*5+4: | |
432 | code = 0; | |
433 | break; | |
434 | case 0*5+1: | |
435 | case 0*5+2: | |
436 | case 4*5+1: | |
437 | case 4*5+2: | |
438 | code = 3; | |
439 | sf[2] = sf[1]; | |
440 | break; | |
441 | case 0*5+3: | |
442 | case 4*5+3: | |
443 | code = 3; | |
444 | sf[1] = sf[2]; | |
445 | break; | |
446 | case 1*5+0: | |
447 | case 1*5+4: | |
448 | case 2*5+4: | |
449 | code = 1; | |
450 | sf[1] = sf[0]; | |
451 | break; | |
452 | case 1*5+1: | |
453 | case 1*5+2: | |
454 | case 2*5+0: | |
455 | case 2*5+1: | |
456 | case 2*5+2: | |
457 | code = 2; | |
458 | sf[1] = sf[2] = sf[0]; | |
459 | break; | |
460 | case 2*5+3: | |
461 | case 3*5+3: | |
462 | code = 2; | |
463 | sf[0] = sf[1] = sf[2]; | |
464 | break; | |
465 | case 3*5+0: | |
466 | case 3*5+1: | |
467 | case 3*5+2: | |
468 | code = 2; | |
469 | sf[0] = sf[2] = sf[1]; | |
470 | break; | |
471 | case 1*5+3: | |
472 | code = 2; | |
473 | if (sf[0] > sf[2]) | |
474 | sf[0] = sf[2]; | |
475 | sf[1] = sf[2] = sf[0]; | |
476 | break; | |
477 | default: | |
478 | av_assert2(0); //cannot happen | |
479 | code = 0; /* kill warning */ | |
480 | } | |
481 | ||
482 | av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j, | |
483 | sf[0], sf[1], sf[2], d1, d2, code); | |
484 | scale_code[j] = code; | |
485 | sf += 3; | |
486 | } | |
487 | } | |
488 | ||
489 | /* The most important function : psycho acoustic module. In this | |
490 | encoder there is basically none, so this is the worst you can do, | |
491 | but also this is the simpler. */ | |
492 | static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) | |
493 | { | |
494 | int i; | |
495 | ||
496 | for(i=0;i<s->sblimit;i++) { | |
497 | smr[i] = (int)(fixed_smr[i] * 10); | |
498 | } | |
499 | } | |
500 | ||
501 | ||
502 | #define SB_NOTALLOCATED 0 | |
503 | #define SB_ALLOCATED 1 | |
504 | #define SB_NOMORE 2 | |
505 | ||
506 | /* Try to maximize the smr while using a number of bits inferior to | |
507 | the frame size. I tried to make the code simpler, faster and | |
508 | smaller than other encoders :-) */ | |
509 | static void compute_bit_allocation(MpegAudioContext *s, | |
510 | short smr1[MPA_MAX_CHANNELS][SBLIMIT], | |
511 | unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], | |
512 | int *padding) | |
513 | { | |
514 | int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size; | |
515 | int incr; | |
516 | short smr[MPA_MAX_CHANNELS][SBLIMIT]; | |
517 | unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT]; | |
518 | const unsigned char *alloc; | |
519 | ||
520 | memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT); | |
521 | memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT); | |
522 | memset(bit_alloc, 0, s->nb_channels * SBLIMIT); | |
523 | ||
524 | /* compute frame size and padding */ | |
525 | max_frame_size = s->frame_size; | |
526 | s->frame_frac += s->frame_frac_incr; | |
527 | if (s->frame_frac >= 65536) { | |
528 | s->frame_frac -= 65536; | |
529 | s->do_padding = 1; | |
530 | max_frame_size += 8; | |
531 | } else { | |
532 | s->do_padding = 0; | |
533 | } | |
534 | ||
535 | /* compute the header + bit alloc size */ | |
536 | current_frame_size = 32; | |
537 | alloc = s->alloc_table; | |
538 | for(i=0;i<s->sblimit;i++) { | |
539 | incr = alloc[0]; | |
540 | current_frame_size += incr * s->nb_channels; | |
541 | alloc += 1 << incr; | |
542 | } | |
543 | for(;;) { | |
544 | /* look for the subband with the largest signal to mask ratio */ | |
545 | max_sb = -1; | |
546 | max_ch = -1; | |
547 | max_smr = INT_MIN; | |
548 | for(ch=0;ch<s->nb_channels;ch++) { | |
549 | for(i=0;i<s->sblimit;i++) { | |
550 | if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) { | |
551 | max_smr = smr[ch][i]; | |
552 | max_sb = i; | |
553 | max_ch = ch; | |
554 | } | |
555 | } | |
556 | } | |
557 | if (max_sb < 0) | |
558 | break; | |
559 | av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n", | |
560 | current_frame_size, max_frame_size, max_sb, max_ch, | |
561 | bit_alloc[max_ch][max_sb]); | |
562 | ||
563 | /* find alloc table entry (XXX: not optimal, should use | |
564 | pointer table) */ | |
565 | alloc = s->alloc_table; | |
566 | for(i=0;i<max_sb;i++) { | |
567 | alloc += 1 << alloc[0]; | |
568 | } | |
569 | ||
570 | if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) { | |
571 | /* nothing was coded for this band: add the necessary bits */ | |
572 | incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6; | |
573 | incr += s->total_quant_bits[alloc[1]]; | |
574 | } else { | |
575 | /* increments bit allocation */ | |
576 | b = bit_alloc[max_ch][max_sb]; | |
577 | incr = s->total_quant_bits[alloc[b + 1]] - | |
578 | s->total_quant_bits[alloc[b]]; | |
579 | } | |
580 | ||
581 | if (current_frame_size + incr <= max_frame_size) { | |
582 | /* can increase size */ | |
583 | b = ++bit_alloc[max_ch][max_sb]; | |
584 | current_frame_size += incr; | |
585 | /* decrease smr by the resolution we added */ | |
586 | smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]]; | |
587 | /* max allocation size reached ? */ | |
588 | if (b == ((1 << alloc[0]) - 1)) | |
589 | subband_status[max_ch][max_sb] = SB_NOMORE; | |
590 | else | |
591 | subband_status[max_ch][max_sb] = SB_ALLOCATED; | |
592 | } else { | |
593 | /* cannot increase the size of this subband */ | |
594 | subband_status[max_ch][max_sb] = SB_NOMORE; | |
595 | } | |
596 | } | |
597 | *padding = max_frame_size - current_frame_size; | |
598 | av_assert0(*padding >= 0); | |
599 | } | |
600 | ||
601 | /* | |
602 | * Output the mpeg audio layer 2 frame. Note how the code is small | |
603 | * compared to other encoders :-) | |
604 | */ | |
605 | static void encode_frame(MpegAudioContext *s, | |
606 | unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], | |
607 | int padding) | |
608 | { | |
609 | int i, j, k, l, bit_alloc_bits, b, ch; | |
610 | unsigned char *sf; | |
611 | int q[3]; | |
612 | PutBitContext *p = &s->pb; | |
613 | ||
614 | /* header */ | |
615 | ||
616 | put_bits(p, 12, 0xfff); | |
617 | put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */ | |
618 | put_bits(p, 2, 4-2); /* layer 2 */ | |
619 | put_bits(p, 1, 1); /* no error protection */ | |
620 | put_bits(p, 4, s->bitrate_index); | |
621 | put_bits(p, 2, s->freq_index); | |
622 | put_bits(p, 1, s->do_padding); /* use padding */ | |
623 | put_bits(p, 1, 0); /* private_bit */ | |
624 | put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO); | |
625 | put_bits(p, 2, 0); /* mode_ext */ | |
626 | put_bits(p, 1, 0); /* no copyright */ | |
627 | put_bits(p, 1, 1); /* original */ | |
628 | put_bits(p, 2, 0); /* no emphasis */ | |
629 | ||
630 | /* bit allocation */ | |
631 | j = 0; | |
632 | for(i=0;i<s->sblimit;i++) { | |
633 | bit_alloc_bits = s->alloc_table[j]; | |
634 | for(ch=0;ch<s->nb_channels;ch++) { | |
635 | put_bits(p, bit_alloc_bits, bit_alloc[ch][i]); | |
636 | } | |
637 | j += 1 << bit_alloc_bits; | |
638 | } | |
639 | ||
640 | /* scale codes */ | |
641 | for(i=0;i<s->sblimit;i++) { | |
642 | for(ch=0;ch<s->nb_channels;ch++) { | |
643 | if (bit_alloc[ch][i]) | |
644 | put_bits(p, 2, s->scale_code[ch][i]); | |
645 | } | |
646 | } | |
647 | ||
648 | /* scale factors */ | |
649 | for(i=0;i<s->sblimit;i++) { | |
650 | for(ch=0;ch<s->nb_channels;ch++) { | |
651 | if (bit_alloc[ch][i]) { | |
652 | sf = &s->scale_factors[ch][i][0]; | |
653 | switch(s->scale_code[ch][i]) { | |
654 | case 0: | |
655 | put_bits(p, 6, sf[0]); | |
656 | put_bits(p, 6, sf[1]); | |
657 | put_bits(p, 6, sf[2]); | |
658 | break; | |
659 | case 3: | |
660 | case 1: | |
661 | put_bits(p, 6, sf[0]); | |
662 | put_bits(p, 6, sf[2]); | |
663 | break; | |
664 | case 2: | |
665 | put_bits(p, 6, sf[0]); | |
666 | break; | |
667 | } | |
668 | } | |
669 | } | |
670 | } | |
671 | ||
672 | /* quantization & write sub band samples */ | |
673 | ||
674 | for(k=0;k<3;k++) { | |
675 | for(l=0;l<12;l+=3) { | |
676 | j = 0; | |
677 | for(i=0;i<s->sblimit;i++) { | |
678 | bit_alloc_bits = s->alloc_table[j]; | |
679 | for(ch=0;ch<s->nb_channels;ch++) { | |
680 | b = bit_alloc[ch][i]; | |
681 | if (b) { | |
682 | int qindex, steps, m, sample, bits; | |
683 | /* we encode 3 sub band samples of the same sub band at a time */ | |
684 | qindex = s->alloc_table[j+b]; | |
685 | steps = ff_mpa_quant_steps[qindex]; | |
686 | for(m=0;m<3;m++) { | |
687 | sample = s->sb_samples[ch][k][l + m][i]; | |
688 | /* divide by scale factor */ | |
689 | #if USE_FLOATS | |
690 | { | |
691 | float a; | |
692 | a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]]; | |
693 | q[m] = (int)((a + 1.0) * steps * 0.5); | |
694 | } | |
695 | #else | |
696 | { | |
697 | int q1, e, shift, mult; | |
698 | e = s->scale_factors[ch][i][k]; | |
699 | shift = s->scale_factor_shift[e]; | |
700 | mult = s->scale_factor_mult[e]; | |
701 | ||
702 | /* normalize to P bits */ | |
703 | if (shift < 0) | |
704 | q1 = sample << (-shift); | |
705 | else | |
706 | q1 = sample >> shift; | |
707 | q1 = (q1 * mult) >> P; | |
708 | q1 += 1 << P; | |
709 | if (q1 < 0) | |
710 | q1 = 0; | |
711 | q[m] = (q1 * (unsigned)steps) >> (P + 1); | |
712 | } | |
713 | #endif | |
714 | if (q[m] >= steps) | |
715 | q[m] = steps - 1; | |
716 | av_assert2(q[m] >= 0 && q[m] < steps); | |
717 | } | |
718 | bits = ff_mpa_quant_bits[qindex]; | |
719 | if (bits < 0) { | |
720 | /* group the 3 values to save bits */ | |
721 | put_bits(p, -bits, | |
722 | q[0] + steps * (q[1] + steps * q[2])); | |
723 | } else { | |
724 | put_bits(p, bits, q[0]); | |
725 | put_bits(p, bits, q[1]); | |
726 | put_bits(p, bits, q[2]); | |
727 | } | |
728 | } | |
729 | } | |
730 | /* next subband in alloc table */ | |
731 | j += 1 << bit_alloc_bits; | |
732 | } | |
733 | } | |
734 | } | |
735 | ||
736 | /* padding */ | |
737 | for(i=0;i<padding;i++) | |
738 | put_bits(p, 1, 0); | |
739 | ||
740 | /* flush */ | |
741 | flush_put_bits(p); | |
742 | } | |
743 | ||
744 | static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, | |
745 | const AVFrame *frame, int *got_packet_ptr) | |
746 | { | |
747 | MpegAudioContext *s = avctx->priv_data; | |
748 | const int16_t *samples = (const int16_t *)frame->data[0]; | |
749 | short smr[MPA_MAX_CHANNELS][SBLIMIT]; | |
750 | unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT]; | |
751 | int padding, i, ret; | |
752 | ||
753 | for(i=0;i<s->nb_channels;i++) { | |
754 | filter(s, i, samples + i, s->nb_channels); | |
755 | } | |
756 | ||
757 | for(i=0;i<s->nb_channels;i++) { | |
758 | compute_scale_factors(s, s->scale_code[i], s->scale_factors[i], | |
759 | s->sb_samples[i], s->sblimit); | |
760 | } | |
761 | for(i=0;i<s->nb_channels;i++) { | |
762 | psycho_acoustic_model(s, smr[i]); | |
763 | } | |
764 | compute_bit_allocation(s, smr, bit_alloc, &padding); | |
765 | ||
766 | if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0) | |
767 | return ret; | |
768 | ||
769 | init_put_bits(&s->pb, avpkt->data, avpkt->size); | |
770 | ||
771 | encode_frame(s, bit_alloc, padding); | |
772 | ||
773 | if (frame->pts != AV_NOPTS_VALUE) | |
f6fa7814 | 774 | avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding); |
2ba45a60 DM |
775 | |
776 | avpkt->size = put_bits_count(&s->pb) / 8; | |
777 | *got_packet_ptr = 1; | |
778 | return 0; | |
779 | } | |
780 | ||
781 | static const AVCodecDefault mp2_defaults[] = { | |
782 | { "b", "0" }, | |
783 | { NULL }, | |
784 | }; | |
785 |