1 /* 2 * CDDL HEADER START 3 * 4 * The contents of this file are subject to the terms of the 5 * Common Development and Distribution License, Version 1.0 only 6 * (the "License"). You may not use this file except in compliance 7 * with the License. 8 * 9 * You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE 10 * or http://www.opensolaris.org/os/licensing. 11 * See the License for the specific language governing permissions 12 * and limitations under the License. 13 * 14 * When distributing Covered Code, include this CDDL HEADER in each 15 * file and include the License file at usr/src/OPENSOLARIS.LICENSE. 16 * If applicable, add the following below this CDDL HEADER, with the 17 * fields enclosed by brackets "[]" replaced with your own identifying 18 * information: Portions Copyright [yyyy] [name of copyright owner] 19 * 20 * CDDL HEADER END 21 */ 22 /* 23 * Copyright (c) 1992-2001 by Sun Microsystems, Inc. 24 * All rights reserved. 25 */ 26 27 #pragma ident "%Z%%M% %I% %E% SMI" 28 29 /* 30 * Description: 31 * 32 * g723_init_state(), g723_encode(), g723_decode() 33 * 34 * These routines comprise an implementation of the CCITT G.723 ADPCM coding 35 * algorithm. Essentially, this implementation is identical to 36 * the bit level description except for a few deviations which 37 * take advantage of work station attributes, such as hardware 2's 38 * complement arithmetic and large memory. Specifically, certain time 39 * consuming operations such as multiplications are replaced 40 * with look up tables and software 2's complement operations are 41 * replaced with hardware 2's complement. 42 * 43 * The deviation (look up tables) from the bit level 44 * specification, preserves the bit level performance specifications. 45 * 46 * As outlined in the G.723 Recommendation, the algorithm is broken 47 * down into modules. Each section of code below is preceded by 48 * the name of the module which it is implementing. 49 * 50 */ 51 #include <stdlib.h> 52 #include <libaudio.h> 53 54 /* 55 * g723_tables.c 56 * 57 * Description: 58 * 59 * This file contains statically defined lookup tables for 60 * use with the G.723 coding routines. 61 */ 62 63 /* 64 * Maps G.723 code word to reconstructed scale factor normalized log 65 * magnitude values. 66 */ 67 static short _dqlntab[8] = {-2048, 135, 273, 373, 373, 273, 135, -2048}; 68 69 /* Maps G.723 code word to log of scale factor multiplier. */ 70 static short _witab[8] = {-128, 960, 4384, 18624, 18624, 4384, 960, -128}; 71 72 /* 73 * Maps G.723 code words to a set of values whose long and short 74 * term averages are computed and then compared to give an indication 75 * how stationary (steady state) the signal is. 76 */ 77 static short _fitab[8] = {0, 0x200, 0x400, 0xE00, 0xE00, 0x400, 0x200, 0}; 78 79 /* 80 * g723_init_state() 81 * 82 * Description: 83 * 84 * This routine initializes and/or resets the audio_encode_state structure 85 * pointed to by 'state_ptr'. 86 * All the state initial values are specified in the G.723 standard specs. 87 */ 88 void 89 g723_init_state( 90 struct audio_g72x_state *state_ptr) 91 { 92 int cnta; 93 94 state_ptr->yl = 34816; 95 state_ptr->yu = 544; 96 state_ptr->dms = 0; 97 state_ptr->dml = 0; 98 state_ptr->ap = 0; 99 for (cnta = 0; cnta < 2; cnta++) { 100 state_ptr->a[cnta] = 0; 101 state_ptr->pk[cnta] = 0; 102 state_ptr->sr[cnta] = 32; 103 } 104 for (cnta = 0; cnta < 6; cnta++) { 105 state_ptr->b[cnta] = 0; 106 state_ptr->dq[cnta] = 32; 107 } 108 state_ptr->td = 0; 109 state_ptr->leftover_cnt = 0; /* no left over codes */ 110 } 111 112 /* 113 * _g723_fmult() 114 * 115 * returns the integer product of the "floating point" an and srn 116 * by the lookup table _fmultwanmant[]. 117 * 118 */ 119 static int 120 _g723_fmult( 121 int an, 122 int srn) 123 { 124 short anmag, anexp, anmant; 125 short wanexp; 126 127 if (an == 0) { 128 return ((srn >= 0) ? 129 ((srn & 077) + 1) >> (18 - (srn >> 6)) : 130 -(((srn & 077) + 1) >> (2 - (srn >> 6)))); 131 } else if (an > 0) { 132 anexp = _fmultanexp[an] - 12; 133 anmant = ((anexp >= 0) ? an >> anexp : an << -anexp) & 07700; 134 if (srn >= 0) { 135 wanexp = anexp + (srn >> 6) - 7; 136 return ((wanexp >= 0) ? 137 (_fmultwanmant[(srn & 077) + anmant] << wanexp) 138 & 0x7FFF : 139 _fmultwanmant[(srn & 077) + anmant] >> -wanexp); 140 } else { 141 wanexp = anexp + (srn >> 6) - 0xFFF7; 142 return ((wanexp >= 0) ? 143 -((_fmultwanmant[(srn & 077) + anmant] << wanexp) 144 & 0x7FFF) : 145 -(_fmultwanmant[(srn & 077) + anmant] >> -wanexp)); 146 } 147 } else { 148 anmag = (-an) & 0x1FFF; 149 anexp = _fmultanexp[anmag] - 12; 150 anmant = ((anexp >= 0) ? anmag >> anexp : anmag << -anexp) 151 & 07700; 152 if (srn >= 0) { 153 wanexp = anexp + (srn >> 6) - 7; 154 return ((wanexp >= 0) ? 155 -((_fmultwanmant[(srn & 077) + anmant] << wanexp) 156 & 0x7FFF) : 157 -(_fmultwanmant[(srn & 077) + anmant] >> -wanexp)); 158 } else { 159 wanexp = anexp + (srn >> 6) - 0xFFF7; 160 return ((wanexp >= 0) ? 161 (_fmultwanmant[(srn & 077) + anmant] << wanexp) 162 & 0x7FFF : 163 _fmultwanmant[(srn & 077) + anmant] >> -wanexp); 164 } 165 } 166 167 } 168 169 /* 170 * _g723_update() 171 * 172 * updates the state variables for each output code 173 * 174 */ 175 static void 176 _g723_update( 177 int y, 178 int i, 179 int dq, 180 int sr, 181 int pk0, 182 struct audio_g72x_state *state_ptr, 183 int sigpk) 184 { 185 int cnt; 186 long fi; /* Adaptation speed control, FUNCTF */ 187 short mag, exp; /* Adaptive predictor, FLOAT A */ 188 short a2p; /* LIMC */ 189 short a1ul; /* UPA1 */ 190 short pks1, fa1; /* UPA2 */ 191 char tr; /* tone/transition detector */ 192 short thr2; 193 194 mag = dq & 0x3FFF; 195 /* TRANS */ 196 if (state_ptr->td == 0) 197 tr = 0; 198 else if (state_ptr->yl > 0x40000) 199 tr = (mag <= 0x2F80) ? 0 : 1; 200 else { 201 thr2 = (0x20 + ((state_ptr->yl >> 10) & 0x1F)) << 202 (state_ptr->yl >> 15); 203 if (mag >= thr2) 204 tr = 1; 205 else 206 tr = (mag <= (thr2 - (thr2 >> 2))) ? 0 : 1; 207 } 208 209 /* 210 * Quantizer scale factor adaptation. 211 */ 212 213 /* FUNCTW & FILTD & DELAY */ 214 state_ptr->yu = y + ((_witab[i] - y) >> 5); 215 216 /* LIMB */ 217 if (state_ptr->yu < 544) 218 state_ptr->yu = 544; 219 else if (state_ptr->yu > 5120) 220 state_ptr->yu = 5120; 221 222 /* FILTE & DELAY */ 223 state_ptr->yl += state_ptr->yu + ((-state_ptr->yl) >> 6); 224 225 /* 226 * Adaptive predictor coefficients. 227 */ 228 if (tr == 1) { 229 state_ptr->a[0] = 0; 230 state_ptr->a[1] = 0; 231 state_ptr->b[0] = 0; 232 state_ptr->b[1] = 0; 233 state_ptr->b[2] = 0; 234 state_ptr->b[3] = 0; 235 state_ptr->b[4] = 0; 236 state_ptr->b[5] = 0; 237 } else { 238 239 /* UPA2 */ 240 pks1 = pk0 ^ state_ptr->pk[0]; 241 242 a2p = state_ptr->a[1] - (state_ptr->a[1] >> 7); 243 if (sigpk == 0) { 244 fa1 = (pks1) ? state_ptr->a[0] : -state_ptr->a[0]; 245 if (fa1 < -8191) 246 a2p -= 0x100; 247 else if (fa1 > 8191) 248 a2p += 0xFF; 249 else 250 a2p += fa1 >> 5; 251 252 if (pk0 ^ state_ptr->pk[1]) 253 /* LIMC */ 254 if (a2p <= -12160) 255 a2p = -12288; 256 else if (a2p >= 12416) 257 a2p = 12288; 258 else 259 a2p -= 0x80; 260 else if (a2p <= -12416) 261 a2p = -12288; 262 else if (a2p >= 12160) 263 a2p = 12288; 264 else 265 a2p += 0x80; 266 } 267 268 /* TRIGB & DELAY */ 269 state_ptr->a[1] = a2p; 270 271 /* UPA1 */ 272 state_ptr->a[0] -= state_ptr->a[0] >> 8; 273 if (sigpk == 0) 274 if (pks1 == 0) 275 state_ptr->a[0] += 192; 276 else 277 state_ptr->a[0] -= 192; 278 279 /* LIMD */ 280 a1ul = 15360 - a2p; 281 if (state_ptr->a[0] < -a1ul) 282 state_ptr->a[0] = -a1ul; 283 else if (state_ptr->a[0] > a1ul) 284 state_ptr->a[0] = a1ul; 285 286 /* UPB : update of b's */ 287 for (cnt = 0; cnt < 6; cnt++) { 288 state_ptr->b[cnt] -= state_ptr->b[cnt] >> 8; 289 if (dq & 0x3FFF) { 290 /* XOR */ 291 if ((dq ^ state_ptr->dq[cnt]) >= 0) 292 state_ptr->b[cnt] += 128; 293 else 294 state_ptr->b[cnt] -= 128; 295 } 296 } 297 } 298 299 for (cnt = 5; cnt > 0; cnt--) 300 state_ptr->dq[cnt] = state_ptr->dq[cnt-1]; 301 /* FLOAT A */ 302 if (mag == 0) { 303 state_ptr->dq[0] = (dq >= 0) ? 0x20 : 0xFC20; 304 } else { 305 exp = _fmultanexp[mag]; 306 state_ptr->dq[0] = (dq >= 0) ? 307 (exp << 6) + ((mag << 6) >> exp) : 308 (exp << 6) + ((mag << 6) >> exp) - 0x400; 309 } 310 311 state_ptr->sr[1] = state_ptr->sr[0]; 312 /* FLOAT B */ 313 if (sr == 0) { 314 state_ptr->sr[0] = 0x20; 315 } else if (sr > 0) { 316 exp = _fmultanexp[sr]; 317 state_ptr->sr[0] = (exp << 6) + ((sr << 6) >> exp); 318 } else { 319 mag = -sr; 320 exp = _fmultanexp[mag]; 321 state_ptr->sr[0] = (exp << 6) + ((mag << 6) >> exp) - 0x400; 322 } 323 324 /* DELAY A */ 325 state_ptr->pk[1] = state_ptr->pk[0]; 326 state_ptr->pk[0] = pk0; 327 328 /* TONE */ 329 if (tr == 1) 330 state_ptr->td = 0; 331 else if (a2p < -11776) 332 state_ptr->td = 1; 333 else 334 state_ptr->td = 0; 335 336 /* 337 * Adaptation speed control. 338 */ 339 fi = _fitab[i]; /* FUNCTF */ 340 state_ptr->dms += (fi - state_ptr->dms) >> 5; /* FILTA */ 341 state_ptr->dml += (((fi << 2) - state_ptr->dml) >> 7); /* FILTB */ 342 343 if (tr == 1) 344 state_ptr->ap = 256; 345 else if (y < 1536) /* SUBTC */ 346 state_ptr->ap += (0x200 - state_ptr->ap) >> 4; 347 else if (state_ptr->td == 1) 348 state_ptr->ap += (0x200 - state_ptr->ap) >> 4; 349 else if (abs((state_ptr->dms << 2) - state_ptr->dml) >= 350 (state_ptr->dml >> 3)) 351 state_ptr->ap += (0x200 - state_ptr->ap) >> 4; 352 else 353 state_ptr->ap += (-state_ptr->ap) >> 4; 354 } 355 356 /* 357 * _g723_quantize() 358 * 359 * Description: 360 * 361 * Given a raw sample, 'd', of the difference signal and a 362 * quantization step size scale factor, 'y', this routine returns the 363 * G.723 codeword to which that sample gets quantized. The step 364 * size scale factor division operation is done in the log base 2 domain 365 * as a subtraction. 366 */ 367 static unsigned int 368 _g723_quantize( 369 int d, /* Raw difference signal sample. */ 370 int y) /* Step size multiplier. */ 371 { 372 /* LOG */ 373 short dqm; /* Magnitude of 'd'. */ 374 short exp; /* Integer part of base 2 log of magnitude of 'd'. */ 375 short mant; /* Fractional part of base 2 log. */ 376 short dl; /* Log of magnitude of 'd'. */ 377 378 /* SUBTB */ 379 short dln; /* Step size scale factor normalized log. */ 380 381 /* QUAN */ 382 unsigned char i; /* G.723 codeword. */ 383 384 /* 385 * LOG 386 * 387 * Compute base 2 log of 'd', and store in 'dln'. 388 * 389 */ 390 dqm = abs(d); 391 exp = _fmultanexp[dqm >> 1]; 392 mant = ((dqm << 7) >> exp) & 0x7F; /* Fractional portion. */ 393 dl = (exp << 7) + mant; 394 395 /* 396 * SUBTB 397 * 398 * "Divide" by step size multiplier. 399 */ 400 dln = dl - (y >> 2); 401 402 /* 403 * QUAN 404 * 405 * Obtain codword for 'd'. 406 */ 407 i = _g723quani[dln & 0xFFF]; 408 if (d < 0) 409 i ^= 7; /* Stuff in sign of 'd'. */ 410 else if (i == 0) 411 i = 7; /* New in 1988 revision */ 412 413 return (i); 414 } 415 416 /* 417 * _g723_reconstr() 418 * 419 * Description: 420 * 421 * Returns reconstructed difference signal 'dq' obtained from 422 * G.723 codeword 'i' and quantization step size scale factor 'y'. 423 * Multiplication is performed in log base 2 domain as addition. 424 */ 425 static int 426 _g723_reconstr( 427 int i, /* G.723 codeword. */ 428 unsigned long y) /* Step size multiplier. */ 429 { 430 /* ADD A */ 431 short dql; /* Log of 'dq' magnitude. */ 432 433 /* ANTILOG */ 434 short dex; /* Integer part of log. */ 435 short dqt; 436 short dq; /* Reconstructed difference signal sample. */ 437 438 439 dql = _dqlntab[i] + (y >> 2); /* ADDA */ 440 441 if (dql < 0) 442 dq = 0; 443 else { /* ANTILOG */ 444 dex = (dql >> 7) & 15; 445 dqt = 128 + (dql & 127); 446 dq = (dqt << 7) >> (14 - dex); 447 } 448 if (i & 4) 449 dq -= 0x8000; 450 451 return (dq); 452 } 453 454 /* 455 * _tandem_adjust(sr, se, y, i) 456 * 457 * Description: 458 * 459 * At the end of ADPCM decoding, it simulates an encoder which may be receiving 460 * the output of this decoder as a tandem process. If the output of the 461 * simulated encoder differs from the input to this decoder, the decoder output 462 * is adjusted by one level of A-law or Mu-law codes. 463 * 464 * Input: 465 * sr decoder output linear PCM sample, 466 * se predictor estimate sample, 467 * y quantizer step size, 468 * i decoder input code 469 * 470 * Return: 471 * adjusted A-law or Mu-law compressed sample. 472 */ 473 static int 474 _tandem_adjust_alaw( 475 int sr, /* decoder output linear PCM sample */ 476 int se, /* predictor estimate sample */ 477 int y, /* quantizer step size */ 478 int i) /* decoder input code */ 479 { 480 unsigned char sp; /* A-law compressed 8-bit code */ 481 short dx; /* prediction error */ 482 char id; /* quantized prediction error */ 483 int sd; /* adjusted A-law decoded sample value */ 484 int im; /* biased magnitude of i */ 485 int imx; /* biased magnitude of id */ 486 487 sp = audio_s2a((sr <= -0x2000)? -0x8000 : 488 (sr < 0x1FFF)? sr << 2 : 0x7FFF); /* short to A-law compression */ 489 dx = (audio_a2s(sp) >> 2) - se; /* 16-bit prediction error */ 490 id = _g723_quantize(dx, y); 491 492 if (id == i) /* no adjustment on sp */ 493 return (sp); 494 else { /* sp adjustment needed */ 495 im = i ^ 4; /* 2's complement to biased unsigned */ 496 imx = id ^ 4; 497 498 if (imx > im) { /* sp adjusted to next lower value */ 499 if (sp & 0x80) 500 sd = (sp == 0xD5)? 0x55 : 501 ((sp ^ 0x55) - 1) ^ 0x55; 502 else 503 sd = (sp == 0x2A)? 0x2A : 504 ((sp ^ 0x55) + 1) ^ 0x55; 505 } else { /* sp adjusted to next higher value */ 506 if (sp & 0x80) 507 sd = (sp == 0xAA)? 0xAA : 508 ((sp ^ 0x55) + 1) ^ 0x55; 509 else 510 sd = (sp == 0x55)? 0xD5 : 511 ((sp ^ 0x55) - 1) ^ 0x55; 512 } 513 return (sd); 514 } 515 } 516 517 static int 518 _tandem_adjust_ulaw( 519 int sr, /* decoder output linear PCM sample */ 520 int se, /* predictor estimate sample */ 521 int y, /* quantizer step size */ 522 int i) /* decoder input code */ 523 { 524 unsigned char sp; /* A-law compressed 8-bit code */ 525 short dx; /* prediction error */ 526 char id; /* quantized prediction error */ 527 int sd; /* adjusted A-law decoded sample value */ 528 int im; /* biased magnitude of i */ 529 int imx; /* biased magnitude of id */ 530 531 sp = audio_s2u((sr <= -0x2000)? -0x8000 : 532 (sr >= 0x1FFF)? 0x7FFF : sr << 2); /* short to u-law compression */ 533 dx = (audio_u2s(sp) >> 2) - se; /* 16-bit prediction error */ 534 id = _g723_quantize(dx, y); 535 if (id == i) 536 return (sp); 537 else { 538 /* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */ 539 im = i ^ 4; /* 2's complement to biased unsigned */ 540 imx = id ^ 4; 541 542 /* u-law codes : 0, 1, ... 7E, 7F, FF, FE, ... 81, 80 */ 543 if (imx > im) { /* sp adjusted to next lower value */ 544 if (sp & 0x80) 545 sd = (sp == 0xFF)? 0x7E : sp + 1; 546 else 547 sd = (sp == 0)? 0 : sp - 1; 548 549 } else { /* sp adjusted to next higher value */ 550 if (sp & 0x80) 551 sd = (sp == 0x80)? 0x80 : sp - 1; 552 else 553 sd = (sp == 0x7F)? 0xFE : sp + 1; 554 } 555 return (sd); 556 } 557 } 558 559 static unsigned char 560 _encoder( 561 int sl, 562 struct audio_g72x_state *state_ptr) 563 { 564 short sei, sezi, se, sez; /* ACCUM */ 565 short d; /* SUBTA */ 566 float al; /* use floating point for faster multiply */ 567 short y, dif; /* MIX */ 568 short sr; /* ADDB */ 569 short pk0, sigpk, dqsez; /* ADDC */ 570 short dq, i; 571 int cnt; 572 573 /* ACCUM */ 574 sezi = _g723_fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]); 575 for (cnt = 1; cnt < 6; cnt++) 576 sezi = sezi + _g723_fmult(state_ptr->b[cnt] >> 2, 577 state_ptr->dq[cnt]); 578 sei = sezi; 579 for (cnt = 1; cnt > -1; cnt--) 580 sei = sei + _g723_fmult(state_ptr->a[cnt] >> 2, 581 state_ptr->sr[cnt]); 582 sez = sezi >> 1; 583 se = sei >> 1; 584 585 d = sl - se; /* SUBTA */ 586 587 if (state_ptr->ap >= 256) 588 y = state_ptr->yu; 589 else { 590 y = state_ptr->yl >> 6; 591 dif = state_ptr->yu - y; 592 al = state_ptr->ap >> 2; 593 if (dif > 0) 594 y += ((int)(dif * al)) >> 6; 595 else if (dif < 0) 596 y += ((int)(dif * al) + 0x3F) >> 6; 597 } 598 599 i = _g723_quantize(d, y); 600 dq = _g723_reconstr(i, y); 601 602 sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; /* ADDB */ 603 604 dqsez = sr + sez - se; /* ADDC */ 605 if (dqsez == 0) { 606 pk0 = 0; 607 sigpk = 1; 608 } else { 609 pk0 = (dqsez < 0) ? 1 : 0; 610 sigpk = 0; 611 } 612 613 _g723_update(y, i, dq, sr, pk0, state_ptr, sigpk); 614 615 return (i); 616 } 617 618 /* 619 * g723_encode() 620 * 621 * Description: 622 * 623 * Encodes a buffer of linear PCM, A-law or Mu-law data pointed to by 'in_buf' 624 * according the G.723 encoding algorithm and packs the resulting code words 625 * into bytes. The bytes of codewords are written to a buffer 626 * pointed to by 'out_buf'. 627 * 628 * Notes: 629 * 630 * In the event that the number packed codes is shorter than a sample unit, 631 * the remainder is saved in the state stucture till next call. It is then 632 * packed into the new buffer on the next call. 633 * The number of valid bytes in 'out_buf' is returned in *out_size. Note that 634 * this will not always be equal to 3/8 of 'data_size' on input. On the 635 * final call to 'g723_encode()' the calling program might want to 636 * check if any code bits was left over. This can be 637 * done by calling 'g723_encode()' with data_size = 0, which returns in 638 * *out_size a* 0 if nothing was leftover and the number of bits left over in 639 * the state structure which now is in out_buf[0]. 640 * 641 * The 3 lower significant bits of an individual byte in the output byte 642 * stream is packed with a G.723 code first. Then the 3 higher order 643 * bits are packed with the next code. 644 */ 645 int 646 g723_encode( 647 void *in_buf, 648 int data_size, 649 Audio_hdr *in_header, 650 unsigned char *out_buf, 651 int *out_size, 652 struct audio_g72x_state *state_ptr) 653 { 654 int i; 655 unsigned char *out_ptr; 656 unsigned char *leftover; 657 unsigned int bits; 658 unsigned int codes; 659 int offset; 660 short *short_ptr; 661 unsigned char *char_ptr; 662 663 /* Dereference the array pointer for faster access */ 664 leftover = &state_ptr->leftover[0]; 665 666 /* Return all cached leftovers */ 667 if (data_size == 0) { 668 for (i = 0; state_ptr->leftover_cnt > 0; i++) { 669 *out_buf++ = leftover[i]; 670 state_ptr->leftover_cnt -= 8; 671 } 672 if (i > 0) { 673 /* Round up to a complete sample unit */ 674 for (; i < 3; i++) 675 *out_buf++ = 0; 676 } 677 *out_size = i; 678 state_ptr->leftover_cnt = 0; 679 return (AUDIO_SUCCESS); 680 } 681 682 /* XXX - if linear, it had better be 16-bit! */ 683 if (in_header->encoding == AUDIO_ENCODING_LINEAR) { 684 if (data_size & 1) { 685 return (AUDIO_ERR_BADFRAME); 686 } else { 687 data_size >>= 1; 688 short_ptr = (short *)in_buf; 689 } 690 } else { 691 char_ptr = (unsigned char *)in_buf; 692 } 693 out_ptr = (unsigned char *)out_buf; 694 695 offset = state_ptr->leftover_cnt / 8; 696 bits = state_ptr->leftover_cnt % 8; 697 codes = (bits > 0) ? leftover[offset] : 0; 698 699 while (data_size--) { 700 switch (in_header->encoding) { 701 case AUDIO_ENCODING_LINEAR: 702 i = _encoder(*short_ptr++ >> 2, state_ptr); 703 break; 704 case AUDIO_ENCODING_ALAW: 705 i = _encoder(audio_a2s(*char_ptr++) >> 2, state_ptr); 706 break; 707 case AUDIO_ENCODING_ULAW: 708 i = _encoder(audio_u2s(*char_ptr++) >> 2, state_ptr); 709 break; 710 default: 711 return (AUDIO_ERR_ENCODING); 712 } 713 /* pack the resulting code into leftover buffer */ 714 codes += i << bits; 715 bits += 3; 716 if (bits >= 8) { 717 leftover[offset] = codes & 0xff; 718 bits -= 8; 719 codes >>= 8; 720 offset++; 721 } 722 state_ptr->leftover_cnt += 3; 723 724 /* got a whole sample unit so copy it out and reset */ 725 if (bits == 0) { 726 *out_ptr++ = leftover[0]; 727 *out_ptr++ = leftover[1]; 728 *out_ptr++ = leftover[2]; 729 codes = 0; 730 state_ptr->leftover_cnt = 0; 731 offset = 0; 732 } 733 } 734 /* If any residual bits, save them for the next call */ 735 if (bits > 0) { 736 leftover[offset] = codes & 0xff; 737 state_ptr->leftover_cnt += bits; 738 } 739 *out_size = (out_ptr - (unsigned char *)out_buf); 740 return (AUDIO_SUCCESS); 741 } 742 743 /* 744 * g723_decode() 745 * 746 * Description: 747 * 748 * Decodes a buffer of G.723 encoded data pointed to by 'in_buf' and 749 * writes the resulting linear PCM, A-law or Mu-law words into a buffer 750 * pointed to by 'out_buf'. 751 * 752 */ 753 int 754 g723_decode( 755 unsigned char *in_buf, /* Buffer of g723 encoded data. */ 756 int data_size, /* Size in bytes of in_buf. */ 757 Audio_hdr *out_header, 758 void *out_buf, /* Decoded data buffer. */ 759 int *out_size, 760 struct audio_g72x_state *state_ptr) /* the decoder's state structure. */ 761 { 762 unsigned char *inbuf_end; 763 unsigned char *in_ptr, *out_ptr; 764 short *linear_ptr; 765 unsigned int codes; 766 unsigned int bits; 767 int cnt; 768 769 short sezi, sei, sez, se; /* ACCUM */ 770 float al; /* use floating point for faster multiply */ 771 short y, dif; /* MIX */ 772 short sr; /* ADDB */ 773 char pk0; /* ADDC */ 774 short dq; 775 char sigpk; 776 short dqsez; 777 unsigned char i; 778 779 in_ptr = in_buf; 780 inbuf_end = in_buf + data_size; 781 out_ptr = (unsigned char *)out_buf; 782 linear_ptr = (short *)out_buf; 783 784 /* Leftovers in decoding are only up to 8 bits */ 785 bits = state_ptr->leftover_cnt; 786 codes = (bits > 0) ? state_ptr->leftover[0] : 0; 787 788 while ((bits >= 3) || (in_ptr < (unsigned char *)inbuf_end)) { 789 if (bits < 3) { 790 codes += *in_ptr++ << bits; 791 bits += 8; 792 } 793 794 /* ACCUM */ 795 sezi = _g723_fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]); 796 for (cnt = 1; cnt < 6; cnt++) 797 sezi = sezi + _g723_fmult(state_ptr->b[cnt] >> 2, 798 state_ptr->dq[cnt]); 799 sei = sezi; 800 for (cnt = 1; cnt >= 0; cnt--) 801 sei = sei + _g723_fmult(state_ptr->a[cnt] >> 2, 802 state_ptr->sr[cnt]); 803 804 sez = sezi >> 1; 805 se = sei >> 1; 806 if (state_ptr->ap >= 256) 807 y = state_ptr->yu; 808 else { 809 y = state_ptr->yl >> 6; 810 dif = state_ptr->yu - y; 811 al = state_ptr->ap >> 2; 812 if (dif > 0) 813 y += ((int)(dif * al)) >> 6; 814 else if (dif < 0) 815 y += ((int)(dif * al) + 0x3F) >> 6; 816 } 817 818 i = codes & 7; 819 dq = _g723_reconstr(i, y); 820 /* ADDB */ 821 if (dq < 0) 822 sr = se - (dq & 0x3FFF); 823 else 824 sr = se + dq; 825 826 827 dqsez = sr - se + sez; /* ADDC */ 828 pk0 = (dqsez < 0) ? 1 : 0; 829 sigpk = (dqsez) ? 0 : 1; 830 831 _g723_update(y, i, dq, sr, pk0, state_ptr, sigpk); 832 833 switch (out_header->encoding) { 834 case AUDIO_ENCODING_LINEAR: 835 *linear_ptr++ = ((sr <= -0x2000) ? -0x8000 : 836 (sr >= 0x1FFF) ? 0x7FFF : sr << 2); 837 break; 838 case AUDIO_ENCODING_ALAW: 839 *out_ptr++ = _tandem_adjust_alaw(sr, se, y, i); 840 break; 841 case AUDIO_ENCODING_ULAW: 842 *out_ptr++ = _tandem_adjust_ulaw(sr, se, y, i); 843 break; 844 default: 845 return (AUDIO_ERR_ENCODING); 846 } 847 codes >>= 3; 848 bits -= 3; 849 } 850 state_ptr->leftover_cnt = bits; 851 if (bits > 0) 852 state_ptr->leftover[0] = codes; 853 854 /* Calculate number of samples returned */ 855 if (out_header->encoding == AUDIO_ENCODING_LINEAR) 856 *out_size = linear_ptr - (short *)out_buf; 857 else 858 *out_size = out_ptr - (unsigned char *)out_buf; 859 860 return (AUDIO_SUCCESS); 861 } 862