| 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; |
| 92 | avctx->delay = 512 - 32 + 1; |
| 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) |
| 774 | avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->delay); |
| 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 | |