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
g723_init_state(struct audio_g72x_state * state_ptr)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
_g723_fmult(int an,int srn)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
_g723_update(int y,int i,int dq,int sr,int pk0,struct audio_g72x_state * state_ptr,int sigpk)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
_g723_quantize(int d,int y)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
_g723_reconstr(int i,unsigned long y)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
_tandem_adjust_alaw(int sr,int se,int y,int i)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
_tandem_adjust_ulaw(int sr,int se,int y,int i)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
_encoder(int sl,struct audio_g72x_state * state_ptr)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
g723_encode(void * in_buf,int data_size,Audio_hdr * in_header,unsigned char * out_buf,int * out_size,struct audio_g72x_state * state_ptr)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
g723_decode(unsigned char * in_buf,int data_size,Audio_hdr * out_header,void * out_buf,int * out_size,struct audio_g72x_state * state_ptr)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