xref: /illumos-gate/usr/src/cmd/audio/utilities/g723.c (revision d8048045f75b60d97deaf376b39d437b027becd1)
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  * Description:
29  *
30  * g723_init_state(), g723_encode(), g723_decode()
31  *
32  * These routines comprise an implementation of the CCITT G.723 ADPCM coding
33  * algorithm.  Essentially, this implementation is identical to
34  * the bit level description except for a few deviations which
35  * take advantage of work station attributes, such as hardware 2's
36  * complement arithmetic and large memory. Specifically, certain time
37  * consuming operations such as multiplications are replaced
38  * with look up tables and software 2's complement operations are
39  * replaced with hardware 2's complement.
40  *
41  * The deviation (look up tables) from the bit level
42  * specification, preserves the bit level performance specifications.
43  *
44  * As outlined in the G.723 Recommendation, the algorithm is broken
45  * down into modules.  Each section of code below is preceded by
46  * the name of the module which it is implementing.
47  *
48  */
49 #include <stdlib.h>
50 #include <libaudio.h>
51 
52 /*
53  * g723_tables.c
54  *
55  * Description:
56  *
57  * This file contains statically defined lookup tables for
58  * use with the G.723 coding routines.
59  */
60 
61 /*
62  * Maps G.723 code word to reconstructed scale factor normalized log
63  * magnitude values.
64  */
65 static short	_dqlntab[8] = {-2048, 135, 273, 373, 373, 273, 135, -2048};
66 
67 /* Maps G.723 code word to log of scale factor multiplier. */
68 static short	_witab[8] = {-128, 960, 4384, 18624, 18624, 4384, 960, -128};
69 
70 /*
71  * Maps G.723 code words to a set of values whose long and short
72  * term averages are computed and then compared to give an indication
73  * how stationary (steady state) the signal is.
74  */
75 static short	_fitab[8] = {0, 0x200, 0x400, 0xE00, 0xE00, 0x400, 0x200, 0};
76 
77 /*
78  * g723_init_state()
79  *
80  * Description:
81  *
82  * This routine initializes and/or resets the audio_encode_state structure
83  * pointed to by 'state_ptr'.
84  * All the state initial values are specified in the G.723 standard specs.
85  */
86 void
g723_init_state(struct audio_g72x_state * state_ptr)87 g723_init_state(
88 	struct audio_g72x_state *state_ptr)
89 {
90 	int cnta;
91 
92 	state_ptr->yl = 34816;
93 	state_ptr->yu = 544;
94 	state_ptr->dms = 0;
95 	state_ptr->dml = 0;
96 	state_ptr->ap = 0;
97 	for (cnta = 0; cnta < 2; cnta++) {
98 		state_ptr->a[cnta] = 0;
99 		state_ptr->pk[cnta] = 0;
100 		state_ptr->sr[cnta] = 32;
101 	}
102 	for (cnta = 0; cnta < 6; cnta++) {
103 		state_ptr->b[cnta] = 0;
104 		state_ptr->dq[cnta] = 32;
105 	}
106 	state_ptr->td = 0;
107 	state_ptr->leftover_cnt = 0;		/* no left over codes */
108 }
109 
110 /*
111  * _g723_fmult()
112  *
113  * returns the integer product of the "floating point" an and srn
114  * by the lookup table _fmultwanmant[].
115  *
116  */
117 static int
_g723_fmult(int an,int srn)118 _g723_fmult(
119 		int an,
120 		int srn)
121 {
122 	short	anmag, anexp, anmant;
123 	short	wanexp;
124 
125 	if (an == 0) {
126 		return ((srn >= 0) ?
127 		    ((srn & 077) + 1) >> (18 - (srn >> 6)) :
128 		    -(((srn & 077) + 1) >> (2 - (srn >> 6))));
129 	} else if (an > 0) {
130 		anexp = _fmultanexp[an] - 12;
131 		anmant = ((anexp >= 0) ? an >> anexp : an << -anexp) & 07700;
132 		if (srn >= 0) {
133 			wanexp = anexp + (srn >> 6) - 7;
134 			return ((wanexp >= 0) ?
135 			    (_fmultwanmant[(srn & 077) + anmant] << wanexp)
136 			    & 0x7FFF :
137 			    _fmultwanmant[(srn & 077) + anmant] >> -wanexp);
138 		} else {
139 			wanexp = anexp + (srn >> 6) - 0xFFF7;
140 			return ((wanexp >= 0) ?
141 			    -((_fmultwanmant[(srn & 077) + anmant] << wanexp)
142 			    & 0x7FFF) :
143 			    -(_fmultwanmant[(srn & 077) + anmant] >> -wanexp));
144 		}
145 	} else {
146 		anmag = (-an) & 0x1FFF;
147 		anexp = _fmultanexp[anmag] - 12;
148 		anmant = ((anexp >= 0) ? anmag >> anexp : anmag << -anexp)
149 		    & 07700;
150 		if (srn >= 0) {
151 			wanexp = anexp + (srn >> 6) - 7;
152 			return ((wanexp >= 0) ?
153 			    -((_fmultwanmant[(srn & 077) + anmant] << wanexp)
154 			    & 0x7FFF) :
155 			    -(_fmultwanmant[(srn & 077) + anmant] >> -wanexp));
156 		} else {
157 			wanexp = anexp + (srn >> 6) - 0xFFF7;
158 			return ((wanexp >= 0) ?
159 			    (_fmultwanmant[(srn & 077) + anmant] << wanexp)
160 			    & 0x7FFF :
161 			    _fmultwanmant[(srn & 077) + anmant] >> -wanexp);
162 		}
163 	}
164 
165 }
166 
167 /*
168  * _g723_update()
169  *
170  * updates the state variables for each output code
171  *
172  */
173 static void
_g723_update(int y,int i,int dq,int sr,int pk0,struct audio_g72x_state * state_ptr,int sigpk)174 _g723_update(
175 	int	y,
176 	int	i,
177 	int	dq,
178 	int	sr,
179 	int	pk0,
180 	struct audio_g72x_state *state_ptr,
181 	int	sigpk)
182 {
183 	int	cnt;
184 	long	fi;			/* Adaptation speed control, FUNCTF */
185 	short	mag, exp;		/* Adaptive predictor, FLOAT A */
186 	short	a2p;			/* LIMC */
187 	short	a1ul;			/* UPA1 */
188 	short	pks1, fa1;		/* UPA2 */
189 	char	tr;			/* tone/transition detector */
190 	short	thr2;
191 
192 	mag = dq & 0x3FFF;
193 	/* TRANS */
194 	if (state_ptr->td == 0)
195 		tr = 0;
196 	else if (state_ptr->yl > 0x40000)
197 		tr = (mag <= 0x2F80) ? 0 : 1;
198 	else {
199 		thr2 = (0x20 + ((state_ptr->yl >> 10) & 0x1F)) <<
200 		    (state_ptr->yl >> 15);
201 		if (mag >= thr2)
202 			tr = 1;
203 		else
204 			tr = (mag <= (thr2 - (thr2 >> 2))) ? 0 : 1;
205 	}
206 
207 	/*
208 	 * Quantizer scale factor adaptation.
209 	 */
210 
211 	/* FUNCTW & FILTD & DELAY */
212 	state_ptr->yu = y + ((_witab[i] - y) >> 5);
213 
214 	/* LIMB */
215 	if (state_ptr->yu < 544)
216 		state_ptr->yu = 544;
217 	else if (state_ptr->yu > 5120)
218 		state_ptr->yu = 5120;
219 
220 	/* FILTE & DELAY */
221 	state_ptr->yl += state_ptr->yu + ((-state_ptr->yl) >> 6);
222 
223 	/*
224 	 * Adaptive predictor coefficients.
225 	 */
226 	if (tr == 1) {
227 		state_ptr->a[0] = 0;
228 		state_ptr->a[1] = 0;
229 		state_ptr->b[0] = 0;
230 		state_ptr->b[1] = 0;
231 		state_ptr->b[2] = 0;
232 		state_ptr->b[3] = 0;
233 		state_ptr->b[4] = 0;
234 		state_ptr->b[5] = 0;
235 	} else {
236 
237 		/* UPA2 */
238 		pks1 = pk0 ^ state_ptr->pk[0];
239 
240 		a2p = state_ptr->a[1] - (state_ptr->a[1] >> 7);
241 		if (sigpk == 0) {
242 			fa1 = (pks1) ? state_ptr->a[0] : -state_ptr->a[0];
243 			if (fa1 < -8191)
244 				a2p -= 0x100;
245 			else if (fa1 > 8191)
246 				a2p += 0xFF;
247 			else
248 				a2p += fa1 >> 5;
249 
250 			if (pk0 ^ state_ptr->pk[1])
251 				/* LIMC */
252 				if (a2p <= -12160)
253 					a2p = -12288;
254 				else if (a2p >= 12416)
255 					a2p = 12288;
256 				else
257 					a2p -= 0x80;
258 			else if (a2p <= -12416)
259 				a2p = -12288;
260 			else if (a2p >= 12160)
261 				a2p = 12288;
262 			else
263 				a2p += 0x80;
264 		}
265 
266 		/* TRIGB & DELAY */
267 		state_ptr->a[1] = a2p;
268 
269 		/* UPA1 */
270 		state_ptr->a[0] -= state_ptr->a[0] >> 8;
271 		if (sigpk == 0) {
272 			if (pks1 == 0) {
273 				state_ptr->a[0] += 192;
274 			} else {
275 				state_ptr->a[0] -= 192;
276 			}
277 		}
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