xref: /illumos-gate/usr/src/cmd/audio/utilities/g721.c (revision 2a8bcb4efb45d99ac41c94a75c396b362c414f7f)
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 /*
28  *
29  * Description:
30  *
31  * g721_encode(), g721_decode(), g721_set_law()
32  *
33  * These routines comprise an implementation of the CCITT G.721 ADPCM coding
34  * algorithm.  Essentially, this implementation is identical to
35  * the bit level description except for a few deviations which
36  * take advantage of work station attributes, such as hardware 2's
37  * complement arithmetic and large memory. Specifically, certain time
38  * consuming operations such as multiplications are replaced
39  * with look up tables and software 2's complement operations are
40  * replaced with hardware 2's complement.
41  *
42  * The deviation (look up tables) from the bit level
43  * specification, preserves the bit level performance specifications.
44  *
45  * As outlined in the G.721 Recommendation, the algorithm is broken
46  * down into modules.  Each section of code below is preceded by
47  * the name of the module which it is implementing.
48  *
49  */
50 #include <stdlib.h>
51 #include <libaudio.h>
52 
53 /*
54  * Maps G.721 code word to reconstructed scale factor normalized log
55  * magnitude values.
56  */
57 static short	_dqlntab[16] = {-2048, 4, 135, 213, 273, 323, 373, 425,
58 		    425, 373, 323, 273, 213, 135, 4, -2048};
59 
60 /* Maps G.721 code word to log of scale factor multiplier. */
61 static long	_witab[16] = {-384, 576, 1312, 2048, 3584, 6336, 11360, 35904,
62 		    35904, 11360, 6336, 3584, 2048, 1312, 576, -384};
63 
64 /*
65  * Maps G.721 code words to a set of values whose long and short
66  * term averages are computed and then compared to give an indication
67  * how stationary (steady state) the signal is.
68  */
69 static short	_fitab[16] = {0, 0, 0, 0x200, 0x200, 0x200, 0x600, 0xE00,
70 		    0xE00, 0x600, 0x200, 0x200, 0x200, 0, 0, 0};
71 
72 /*
73  * g721_init_state()
74  *
75  * Description:
76  *
77  * This routine initializes and/or resets the audio_g72x_state structure
78  * pointed to by 'state_ptr'.
79  * All the initial state values are specified in the G.721 standard specs.
80  */
81 void
g721_init_state(struct audio_g72x_state * state_ptr)82 g721_init_state(
83 	struct audio_g72x_state *state_ptr)
84 {
85 	int cnta;
86 
87 	state_ptr->yl = 34816;
88 	state_ptr->yu = 544;
89 	state_ptr->dms = 0;
90 	state_ptr->dml = 0;
91 	state_ptr->ap = 0;
92 	for (cnta = 0; cnta < 2; cnta++) {
93 		state_ptr->a[cnta] = 0;
94 		state_ptr->pk[cnta] = 0;
95 		state_ptr->sr[cnta] = 32;
96 	}
97 	for (cnta = 0; cnta < 6; cnta++) {
98 		state_ptr->b[cnta] = 0;
99 		state_ptr->dq[cnta] = 32;
100 	}
101 	state_ptr->td = 0;
102 	state_ptr->leftover_cnt = 0;		/* no left over codes */
103 }
104 
105 /*
106  * _g721_fmult()
107  *
108  * returns the integer product of the "floating point" an and srn
109  * by the lookup table _fmultwanmant[].
110  *
111  */
112 static int
_g721_fmult(int an,int srn)113 _g721_fmult(
114 	int	an,
115 	int	srn)
116 {
117 	short	anmag, anexp, anmant;
118 	short	wanexp;
119 
120 	if (an == 0) {
121 		return ((srn >= 0) ?
122 		    ((srn & 077) + 1) >> (18 - (srn >> 6)) :
123 		    -(((srn & 077) + 1) >> (2 - (srn >> 6))));
124 	} else if (an > 0) {
125 		anexp = _fmultanexp[an] - 12;
126 		anmant = ((anexp >= 0) ? an >> anexp : an << -anexp) & 07700;
127 		if (srn >= 0) {
128 			wanexp = anexp + (srn >> 6) - 7;
129 			return ((wanexp >= 0) ?
130 			    (_fmultwanmant[(srn & 077) + anmant] << wanexp)
131 			    & 0x7FFF :
132 			    _fmultwanmant[(srn & 077) + anmant] >> -wanexp);
133 		} else {
134 			wanexp = anexp + (srn >> 6) - 0xFFF7;
135 			return ((wanexp >= 0) ?
136 			    -((_fmultwanmant[(srn & 077) + anmant] << wanexp)
137 			    & 0x7FFF) :
138 			    -(_fmultwanmant[(srn & 077) + anmant] >> -wanexp));
139 		}
140 	} else {
141 		anmag = (-an) & 0x1FFF;
142 		anexp = _fmultanexp[anmag] - 12;
143 		anmant = ((anexp >= 0) ? anmag >> anexp : anmag << -anexp)
144 		    & 07700;
145 		if (srn >= 0) {
146 			wanexp = anexp + (srn >> 6) - 7;
147 			return ((wanexp >= 0) ?
148 			    -((_fmultwanmant[(srn & 077) + anmant] << wanexp)
149 			    & 0x7FFF) :
150 			    -(_fmultwanmant[(srn & 077) + anmant] >> -wanexp));
151 		} else {
152 			wanexp = anexp + (srn >> 6) - 0xFFF7;
153 			return ((wanexp >= 0) ?
154 			    (_fmultwanmant[(srn & 077) + anmant] << wanexp)
155 			    & 0x7FFF :
156 			    _fmultwanmant[(srn & 077) + anmant] >> -wanexp);
157 		}
158 	}
159 }
160 
161 /*
162  * _g721_update()
163  *
164  * updates the state variables for each output code
165  *
166  */
167 static void
_g721_update(int y,int i,int dq,int sr,int pk0,struct audio_g72x_state * state_ptr,int sigpk)168 _g721_update(
169 	int	y,
170 	int	i,
171 	int	dq,
172 	int	sr,
173 	int	pk0,
174 	struct audio_g72x_state *state_ptr,
175 	int	sigpk)
176 {
177 	int	cnt;
178 	long	fi;				/* FUNCTF */
179 	short	mag, exp;			/* FLOAT A */
180 	short	a2p;				/* LIMC */
181 	short	a1ul;				/* UPA1 */
182 	short	pks1, fa1;			/* UPA2 */
183 	char	tr;				/* tone/transition detector */
184 	short	thr2;
185 
186 	mag = dq & 0x3FFF;
187 	/* TRANS */
188 	if (state_ptr->td == 0) {
189 		tr = 0;
190 	} else if (state_ptr->yl > 0x40000) {
191 		tr = (mag <= 0x2F80) ? 0 : 1;
192 	} else {
193 		thr2 = (0x20 + ((state_ptr->yl >> 10) & 0x1F)) <<
194 		    (state_ptr->yl >> 15);
195 		if (mag >= thr2) {
196 			tr = 1;
197 		} else {
198 			tr = (mag <= (thr2 - (thr2 >> 2))) ? 0 : 1;
199 		}
200 	}
201 
202 	/*
203 	 * Quantizer scale factor adaptation.
204 	 */
205 
206 	/* FUNCTW & FILTD & DELAY */
207 	state_ptr->yu = y + ((_witab[i] - y) >> 5);
208 
209 	/* LIMB */
210 	if (state_ptr->yu < 544) {
211 		state_ptr->yu = 544;
212 	} else if (state_ptr->yu > 5120) {
213 		state_ptr->yu = 5120;
214 	}
215 
216 	/* FILTE & DELAY */
217 	state_ptr->yl += state_ptr->yu + ((-state_ptr->yl) >> 6);
218 
219 	/*
220 	 * Adaptive predictor.
221 	 */
222 	if (tr == 1) {
223 		state_ptr->a[0] = 0;
224 		state_ptr->a[1] = 0;
225 		state_ptr->b[0] = 0;
226 		state_ptr->b[1] = 0;
227 		state_ptr->b[2] = 0;
228 		state_ptr->b[3] = 0;
229 		state_ptr->b[4] = 0;
230 		state_ptr->b[5] = 0;
231 	} else {
232 
233 		/* UPA2 */
234 		pks1 = pk0 ^ state_ptr->pk[0];
235 
236 		a2p = state_ptr->a[1] - (state_ptr->a[1] >> 7);
237 		if (sigpk == 0) {
238 			fa1 = (pks1) ? state_ptr->a[0] : -state_ptr->a[0];
239 			if (fa1 < -8191) {
240 				a2p -= 0x100;
241 			} else if (fa1 > 8191) {
242 				a2p += 0xFF;
243 			} else {
244 				a2p += fa1 >> 5;
245 			}
246 
247 			if (pk0 ^ state_ptr->pk[1]) {
248 				/* LIMC */
249 				if (a2p <= -12160) {
250 					a2p = -12288;
251 				} else if (a2p >= 12416) {
252 					a2p = 12288;
253 				} else {
254 					a2p -= 0x80;
255 				}
256 			} else if (a2p <= -12416) {
257 				a2p = -12288;
258 			} else if (a2p >= 12160) {
259 				a2p = 12288;
260 			} else {
261 				a2p += 0x80;
262 			}
263 		}
264 
265 		/* TRIGB & DELAY */
266 		state_ptr->a[1] = a2p;
267 
268 		/* UPA1 */
269 		state_ptr->a[0] -= state_ptr->a[0] >> 8;
270 		if (sigpk == 0) {
271 			if (pks1 == 0) {
272 				state_ptr->a[0] += 192;
273 			} else {
274 				state_ptr->a[0] -= 192;
275 			}
276 		}
277 
278 		/* LIMD */
279 		a1ul = 15360 - a2p;
280 		if (state_ptr->a[0] < -a1ul)
281 			state_ptr->a[0] = -a1ul;
282 		else if (state_ptr->a[0] > a1ul)
283 			state_ptr->a[0] = a1ul;
284 
285 		/* UPB : update of b's */
286 		for (cnt = 0; cnt < 6; cnt++) {
287 			state_ptr->b[cnt] -= state_ptr->b[cnt] >> 8;
288 			if (dq & 0x3FFF) {
289 				/* XOR */
290 				if ((dq ^ state_ptr->dq[cnt]) >= 0)
291 					state_ptr->b[cnt] += 128;
292 				else
293 					state_ptr->b[cnt] -= 128;
294 			}
295 		}
296 	}
297 
298 	for (cnt = 5; cnt > 0; cnt--)
299 		state_ptr->dq[cnt] = state_ptr->dq[cnt-1];
300 	/* FLOAT A */
301 	if (mag == 0) {
302 		state_ptr->dq[0] = (dq >= 0) ? 0x20 : 0xFC20;
303 	} else {
304 		exp = _fmultanexp[mag];
305 		state_ptr->dq[0] = (dq >= 0) ?
306 		    (exp << 6) + ((mag << 6) >> exp) :
307 		    (exp << 6) + ((mag << 6) >> exp) - 0x400;
308 	}
309 
310 	state_ptr->sr[1] = state_ptr->sr[0];
311 	/* FLOAT B */
312 	if (sr == 0) {
313 		state_ptr->sr[0] = 0x20;
314 	} else if (sr > 0) {
315 		exp = _fmultanexp[sr];
316 		state_ptr->sr[0] = (exp << 6) + ((sr << 6) >> exp);
317 	} else {
318 		mag = -sr;
319 		exp = _fmultanexp[mag];
320 		state_ptr->sr[0] =  (exp << 6) + ((mag << 6) >> exp) - 0x400;
321 	}
322 
323 	/* DELAY A */
324 	state_ptr->pk[1] = state_ptr->pk[0];
325 	state_ptr->pk[0] = pk0;
326 
327 	/* TONE */
328 	if (tr == 1)
329 		state_ptr->td = 0;
330 	else if (a2p < -11776)
331 		state_ptr->td = 1;
332 	else
333 		state_ptr->td = 0;
334 
335 	/*
336 	 * Adaptation speed control.
337 	 */
338 	fi = _fitab[i];						/* FUNCTF */
339 	state_ptr->dms += (fi - state_ptr->dms) >> 5;		/* FILTA */
340 	state_ptr->dml += (((fi << 2) - state_ptr->dml) >> 7);	/* FILTB */
341 
342 	if (tr == 1)
343 		state_ptr->ap = 256;
344 	else if (y < 1536)					/* SUBTC */
345 		state_ptr->ap += (0x200 - state_ptr->ap) >> 4;
346 	else if (state_ptr->td == 1)
347 		state_ptr->ap += (0x200 - state_ptr->ap) >> 4;
348 	else if (abs((state_ptr->dms << 2) - state_ptr->dml) >=
349 	    (state_ptr->dml >> 3))
350 		state_ptr->ap += (0x200 - state_ptr->ap) >> 4;
351 	else
352 		state_ptr->ap += (-state_ptr->ap) >> 4;
353 }
354 
355 /*
356  * _g721_quantize()
357  *
358  * Description:
359  *
360  * Given a raw sample, 'd', of the difference signal and a
361  * quantization step size scale factor, 'y', this routine returns the
362  * G.721 codeword to which that sample gets quantized.  The step
363  * size scale factor division operation is done in the log base 2 domain
364  * as a subtraction.
365  */
366 static unsigned int
_g721_quantize(int d,int y)367 _g721_quantize(
368 	int	d,	/* Raw difference signal sample. */
369 	int	y)	/* Step size multiplier. */
370 {
371 	/* LOG */
372 	short	dqm;	/* Magnitude of 'd'. */
373 	short	exp;	/* Integer part of base 2 log of magnitude of 'd'. */
374 	short	mant;	/* Fractional part of base 2 log. */
375 	short	dl;	/* Log of magnitude of 'd'. */
376 
377 	/* SUBTB */
378 	short	dln;	/* Step size scale factor normalized log. */
379 
380 	/* QUAN */
381 	char	i;	/* G.721 codeword. */
382 
383 	/*
384 	 * LOG
385 	 *
386 	 * Compute base 2 log of 'd', and store in 'dln'.
387 	 *
388 	 */
389 	dqm = abs(d);
390 	exp = _fmultanexp[dqm >> 1];
391 	mant = ((dqm << 7) >> exp) & 0x7F;	/* Fractional portion. */
392 	dl = (exp << 7) + mant;
393 
394 	/*
395 	 * SUBTB
396 	 *
397 	 * "Divide" by step size multiplier.
398 	 */
399 	dln = dl - (y >> 2);
400 
401 	/*
402 	 * QUAN
403 	 *
404 	 * Obtain codword for 'd'.
405 	 */
406 	i = _quani[dln & 0xFFF];
407 	if (d < 0)
408 		i ^= 0xF;	/* Stuff in sign of 'd'. */
409 	else if (i == 0)
410 		i = 0xF;	/* New in 1988 revision */
411 
412 	return (i);
413 }
414 
415 /*
416  * _g721_reconstr()
417  *
418  * Description:
419  *
420  * Returns reconstructed difference signal 'dq' obtained from
421  * G.721 codeword 'i' and quantization step size scale factor 'y'.
422  * Multiplication is performed in log base 2 domain as addition.
423  */
424 static unsigned long
_g721_reconstr(int i,unsigned long y)425 _g721_reconstr(
426 	int		i,	/* G.721 codeword. */
427 	unsigned long	y)	/* Step size multiplier. */
428 {
429 	/* ADD A */
430 	short	dql;	/* Log of 'dq' magnitude. */
431 
432 	/* ANTILOG */
433 	short	dex;	/* Integer part of log. */
434 	short	dqt;
435 	short	dq;	/* Reconstructed difference signal sample. */
436 
437 	dql = _dqlntab[i] + (y >> 2);	/* ADDA */
438 
439 	if (dql < 0)
440 		dq = 0;
441 	else {				/* ANTILOG */
442 		dex = (dql >> 7) & 15;
443 		dqt = 128 + (dql & 127);
444 		dq = (dqt << 7) >> (14 - dex);
445 	}
446 	if (i & 8)
447 		dq -= 0x4000;
448 
449 	return (dq);
450 }
451 
452 /*
453  * _tandem_adjust(sr, se, y, i)
454  *
455  * Description:
456  *
457  * At the end of ADPCM decoding, it simulates an encoder which may be receiving
458  * the output of this decoder as a tandem process. If the output of the
459  * simulated encoder differs from the input to this decoder, the decoder output
460  * is adjusted by one level of A-law or u-law codes.
461  *
462  * Input:
463  *	sr	decoder output linear PCM sample,
464  *	se	predictor estimate sample,
465  *	y	quantizer step size,
466  *	i	decoder input code
467  *
468  * Return:
469  *	adjusted A-law or u-law compressed sample.
470  */
471 static int
_tandem_adjust_alaw(int sr,int se,int y,int i)472 _tandem_adjust_alaw(
473 	int	sr,	/* decoder output linear PCM sample */
474 	int	se,	/* predictor estimate sample */
475 	int	y,	/* quantizer step size */
476 	int	i)	/* decoder input code */
477 {
478 	unsigned char	sp;	/* A-law compressed 8-bit code */
479 	short	dx;		/* prediction error */
480 	char	id;		/* quantized prediction error */
481 	int	sd;		/* adjusted A-law decoded sample value */
482 	int	im;		/* biased magnitude of i */
483 	int	imx;		/* biased magnitude of id */
484 
485 	sp = audio_s2a((sr <= -0x2000)? -0x8000 :
486 	    (sr >= 0x1FFF)? 0x7FFF : sr << 2);	/* short to A-law compression */
487 	dx = (audio_a2s(sp) >> 2) - se; 	/* 16-bit prediction error */
488 	id = _g721_quantize(dx, y);
489 
490 	if (id == i)			/* no adjustment on sp */
491 		return (sp);
492 	else {				/* sp adjustment needed */
493 		/* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */
494 		im = i ^ 8;		/* 2's complement to biased unsigned */
495 		imx = id ^ 8;
496 
497 		if (imx > im) {		/* sp adjusted to next lower value */
498 			if (sp & 0x80)
499 				sd = (sp == 0xD5)? 0x55 :
500 				    ((sp ^ 0x55) - 1) ^ 0x55;
501 			else
502 				sd = (sp == 0x2A)? 0x2A :
503 				    ((sp ^ 0x55) + 1) ^ 0x55;
504 		} else {		/* sp adjusted to next higher value */
505 			if (sp & 0x80)
506 				sd = (sp == 0xAA)? 0xAA :
507 				    ((sp ^ 0x55) + 1) ^ 0x55;
508 			else
509 				sd = (sp == 0x55)? 0xD5 :
510 				    ((sp ^ 0x55) - 1) ^ 0x55;
511 		}
512 		return (sd);
513 	}
514 }
515 
516 static int
_tandem_adjust_ulaw(int sr,int se,int y,int i)517 _tandem_adjust_ulaw(
518 	int	sr,	/* decoder output linear PCM sample */
519 	int	se,	/* predictor estimate sample */
520 	int	y,	/* quantizer step size */
521 	int	i)	/* decoder input code */
522 {
523 	unsigned char   sp;	/* A-law compressed 8-bit code */
524 	short	dx;		/* prediction error */
525 	char	id;		/* quantized prediction error */
526 	int	sd;		/* adjusted A-law decoded sample value */
527 	int	im;		/* biased magnitude of i */
528 	int	imx;		/* biased magnitude of id */
529 
530 	sp = audio_s2u((sr <= -0x2000)? -0x8000 :
531 	    (sr >= 0x1FFF)? 0x7FFF : sr << 2); /* short to u-law compression */
532 	dx = (audio_u2s(sp) >> 2) - se;  /* 16-bit prediction error */
533 	id = _g721_quantize(dx, y);
534 	if (id == i)
535 		return (sp);
536 	else {
537 		/* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */
538 		im = i ^ 8;		/* 2's complement to biased unsigned */
539 		imx = id ^ 8;
540 		if (imx > im) {		/* sp adjusted to next lower value */
541 			if (sp & 0x80)
542 				sd = (sp == 0xFF)? 0x7F : sp + 1;
543 			else
544 				sd = (sp == 0)? 0 : sp - 1;
545 
546 		} else {		/* sp adjusted to next higher value */
547 			if (sp & 0x80)
548 				sd = (sp == 0x80)? 0x80 : sp - 1;
549 			else
550 				sd = (sp == 0x7F)? 0xFF : sp + 1;
551 		}
552 		return (sd);
553 	}
554 }
555 
556 /*
557  * g721_encode()
558  *
559  * Description:
560  *
561  * Encodes a buffer of linear PCM, A-law or u-law data pointed to by
562  * 'in_buf' according * the G.721 encoding algorithm and packs the
563  * resulting code words into bytes. The bytes of codeword pairs are
564  * written to a buffer pointed to by 'out_buf'.
565  *
566  * Notes:
567  *
568  * In the event that the total number of codewords which have to be
569  * written is odd, the last unpairable codeword is saved in the
570  * state structure till the next call. It is then paired off and
571  * packed with the first codeword of the new buffer. The number of
572  * valid bytes in 'out_buf' is returned in *out_size. Note that
573  * *out_size will not always be equal to half * of 'data_size' on input.
574  * On the final call to 'g721_encode()' the calling program might want to
575  * check if a codeword was left over. This can be
576  * done by calling 'g721_encode()' with data_size = 0, which returns in
577  * *out_size a 0 if nothing was leftover and 1 if a codeword was leftover
578  * which now is in out_buf[0].
579  *
580  * The 4 lower significant bits of an individual byte in the output byte
581  * stream is packed with a G.721 codeword first.  Then the 4 higher order
582  * bits are packed with the next codeword.
583  */
584 int
g721_encode(void * in_buf,int data_size,Audio_hdr * in_header,unsigned char * out_buf,int * out_size,struct audio_g72x_state * state_ptr)585 g721_encode(
586 	void		*in_buf,
587 	int		data_size,
588 	Audio_hdr	*in_header,
589 	unsigned char	*out_buf,
590 	int		*out_size,
591 	struct audio_g72x_state *state_ptr)
592 {
593 	short	sl;				/* EXPAND */
594 	short	sei, sezi, se, sez;		/* ACCUM */
595 	short	d;				/* SUBTA */
596 	float	al;		/* use floating point for faster multiply */
597 	short	y, dif;				/* MIX */
598 	short	sr;				/* ADDB */
599 	short	pk0, sigpk, dqsez;		/* ADDC */
600 	short	dq, i;
601 	int	cnt, cnta;
602 	int	out_leng;
603 	unsigned char *char_in;
604 	unsigned char *char_out;
605 	short	*short_ptr;
606 
607 	if (data_size == 0) {
608 		/* Actually, the leftover count will never be more than 4 */
609 		for (i = 0; state_ptr->leftover_cnt > 0; i++) {
610 			*out_buf++ = state_ptr->leftover[i];
611 			state_ptr->leftover_cnt -= 8;
612 		}
613 		*out_size = i;
614 		state_ptr->leftover_cnt = 0;
615 		return (AUDIO_SUCCESS);
616 	}
617 
618 	/* XXX - if linear, it had better be 16-bit! */
619 	if (in_header->encoding == AUDIO_ENCODING_LINEAR) {
620 		if (data_size & 1) {
621 			return (AUDIO_ERR_BADFRAME);
622 		} else {
623 			data_size >>= 1;	/* divide to get sample cnt */
624 			short_ptr = (short *)in_buf;
625 		}
626 	} else {
627 		char_in = (unsigned char *)in_buf;
628 	}
629 	char_out = (unsigned char *)out_buf;
630 	if (state_ptr->leftover_cnt > 0) {
631 		*char_out = state_ptr->leftover[0];
632 		state_ptr->leftover_cnt = 0;
633 		data_size += 1;
634 		cnta = 1;
635 	} else {
636 		cnta = 0;
637 	}
638 	out_leng = (data_size & ~0x01);		/* clear low order bit */
639 	for (; cnta < data_size; cnta++) {
640 		/*  EXPAND  */
641 		switch (in_header->encoding) {
642 		case AUDIO_ENCODING_LINEAR:
643 			sl = *short_ptr++ >> 2;
644 			break;
645 		case AUDIO_ENCODING_ALAW:
646 			sl = audio_a2s(*char_in++) >> 2;
647 			break;
648 		case AUDIO_ENCODING_ULAW:
649 			sl = audio_u2s(*char_in++) >> 2; /* u-law to short */
650 			break;
651 		default:
652 			return (AUDIO_ERR_ENCODING);
653 		}
654 
655 		/* ACCUM */
656 		sezi = _g721_fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]);
657 		for (cnt = 1; cnt < 6; cnt++)
658 			sezi = sezi + _g721_fmult(state_ptr->b[cnt] >> 2,
659 			    state_ptr->dq[cnt]);
660 		sei = sezi;
661 		for (cnt = 1; cnt > -1; cnt--)
662 			sei = sei + _g721_fmult(state_ptr->a[cnt] >> 2,
663 			    state_ptr->sr[cnt]);
664 		sez = sezi >> 1;
665 		se = sei >> 1;
666 		d = sl - se;				/* SUBTA */
667 
668 		if (state_ptr->ap >= 256)
669 			y = state_ptr->yu;
670 		else {
671 			y = state_ptr->yl >> 6;
672 			dif = state_ptr->yu - y;
673 			al = state_ptr->ap >> 2;
674 			if (dif > 0)
675 				y += ((int)(dif * al)) >> 6;
676 			else if (dif < 0)
677 				y += ((int)(dif * al) + 0x3F) >> 6;
678 		}
679 
680 		i = _g721_quantize(d, y);
681 		dq = _g721_reconstr(i, y);
682 		/* ADDB */
683 		sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq;
684 
685 		if (cnta & 1) {
686 			*char_out++ += i << 4;
687 		} else if (cnta < out_leng) {
688 			*char_out = i;
689 		} else {
690 			/*
691 			 * save the last codeword which can not be paired into
692 			 * a byte in the state stucture and set leftover_flag.
693 			 */
694 			state_ptr->leftover[0] = i;
695 			state_ptr->leftover_cnt = 4;
696 		}
697 
698 		dqsez = sr + sez - se;		/* ADDC */
699 		if (dqsez == 0) {
700 			pk0 = 0;
701 			sigpk = 1;
702 		} else {
703 			pk0 = (dqsez < 0) ? 1 : 0;
704 			sigpk = 0;
705 		}
706 
707 		_g721_update(y, i, dq, sr, pk0, state_ptr, sigpk);
708 	}
709 	*out_size = cnta >> 1;
710 
711 	return (AUDIO_SUCCESS);
712 }
713 
714 /*
715  * g721_decode()
716  *
717  * Description:
718  *
719  * Decodes a buffer of G.721 encoded data pointed to by 'in_buf' and
720  * writes the resulting linear PCM, A-law or Mu-law bytes into a buffer
721  * pointed to by 'out_buf'.
722  */
723 int
g721_decode(unsigned char * in_buf,int data_size,Audio_hdr * out_header,void * out_buf,int * out_size,struct audio_g72x_state * state_ptr)724 g721_decode(
725 	unsigned char	*in_buf,	/* Buffer of g721 encoded data. */
726 	int		data_size,	/* Size in bytes of in_buf. */
727 	Audio_hdr	*out_header,
728 	void		*out_buf,	/* Decoded data buffer. */
729 	int		*out_size,
730 	struct audio_g72x_state *state_ptr) /* the decoder's state structure. */
731 {
732 	short	sezi, sei, sez, se;		/* ACCUM */
733 	float	al;		/* use floating point for faster multiply */
734 	short	y, dif;				/* MIX */
735 	short sr;				/* ADDB */
736 	char	pk0, i;				/* ADDC */
737 	short	dq;
738 	char	sigpk;
739 	short	dqsez;
740 	unsigned char *char_in;
741 	unsigned char *char_out;
742 	int	cnt, cnta;
743 	short	*linear_out;
744 
745 	*out_size = data_size << 1;
746 	char_in = (unsigned char *)in_buf;
747 	char_out = (unsigned char *)out_buf;
748 	linear_out = (short *)out_buf;
749 	for (cnta = 0; cnta < *out_size; cnta++) {
750 		if (cnta & 1)
751 			i = *char_in++ >> 4;
752 		else
753 			i = *char_in & 0xF;
754 		/* ACCUM */
755 		sezi = _g721_fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]);
756 		for (cnt = 1; cnt < 6; cnt++)
757 			sezi = sezi + _g721_fmult(state_ptr->b[cnt] >> 2,
758 			    state_ptr->dq[cnt]);
759 		sei = sezi;
760 		for (cnt = 1; cnt >= 0; cnt--)
761 			sei = sei + _g721_fmult(state_ptr->a[cnt] >> 2,
762 			    state_ptr->sr[cnt]);
763 
764 		sez = sezi >> 1;
765 		se = sei >> 1;
766 		if (state_ptr->ap >= 256)
767 			y = state_ptr->yu;
768 		else {
769 			y = state_ptr->yl >> 6;
770 			dif = state_ptr->yu - y;
771 			al = state_ptr->ap >> 2;
772 			if (dif > 0)
773 				y += ((int)(dif * al)) >> 6;
774 			else if (dif < 0)
775 				y += ((int)(dif * al) + 0x3F) >> 6;
776 		}
777 
778 		dq = _g721_reconstr(i, y);
779 		/* ADDB */
780 		if (dq < 0)
781 			sr = se - (dq & 0x3FFF);
782 		else
783 			sr = se + dq;
784 
785 		switch (out_header->encoding) {
786 		case AUDIO_ENCODING_LINEAR:
787 			*linear_out++ = ((sr <= -0x2000) ? -0x8000 :
788 			    (sr >= 0x1FFF) ? 0x7FFF : sr << 2);
789 			break;
790 		case AUDIO_ENCODING_ALAW:
791 			*char_out++ = _tandem_adjust_alaw(sr, se, y, i);
792 			break;
793 		case AUDIO_ENCODING_ULAW:
794 			*char_out++ = _tandem_adjust_ulaw(sr, se, y, i);
795 			break;
796 		default:
797 			return (AUDIO_ERR_ENCODING);
798 		}
799 		/* ADDC */
800 		dqsez = sr - se + sez;
801 		pk0 = (dqsez < 0) ? 1 : 0;
802 		sigpk = (dqsez) ? 0 : 1;
803 
804 		_g721_update(y, i, dq, sr, pk0, state_ptr, sigpk);
805 	}
806 	*out_size = cnta;
807 
808 	return (AUDIO_SUCCESS);
809 }
810