www.pudn.com > g.726.rar > g726_16.c, change:2002-11-20,size:6632b


/* 
 * This source code is a product of Sun Microsystems, Inc. and is provided 
 * for unrestricted use.  Users may copy or modify this source code without 
 * charge. 
 * 
 * SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING 
 * THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR 
 * PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE. 
 * 
 * Sun source code is provided with no support and without any obligation on 
 * the part of Sun Microsystems, Inc. to assist in its use, correction, 
 * modification or enhancement. 
 * 
 * SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE 
 * INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE 
 * OR ANY PART THEREOF. 
 * 
 * In no event will Sun Microsystems, Inc. be liable for any lost revenue 
 * or profits or other special, indirect and consequential damages, even if 
 * Sun has been advised of the possibility of such damages. 
 * 
 * Sun Microsystems, Inc. 
 * 2550 Garcia Avenue 
 * Mountain View, California  94043 
 */ 
/* 16kbps version created, used 24kbps code and changing as little as possible. 
 * G.726 specs are available from ITU's gopher or WWW site (http://www.itu.ch) 
 * If any errors are found, please contact me at mrand@tamu.edu 
 *      -Marc Randolph 
 */ 
 
/* 
 * g726_16.c 
 * 
 * Description: 
 * 
 * g723_16_encoder(), g723_16_decoder() 
 * 
 * These routines comprise an implementation of the CCITT G.726 16 Kbps 
 * ADPCM coding algorithm.  Essentially, this implementation is identical to 
 * the bit level description except for a few deviations which take advantage 
 * of workstation attributes, such as hardware 2's complement arithmetic. 
 * 
 * The ITU-T G.726 coder is an adaptive differential pulse code modulation 
 * (ADPCM) waveform coding algorithm, suitable for coding of digitized 
 * telephone bandwidth (0.3-3.4 kHz) speech or audio signals sampled at 8 kHz. 
 * This coder operates on a sample-by-sample basis. Input samples may be  
 * represented in linear PCM or companded 8-bit G.711 (m-law/A-law) formats 
 * (i.e., 64 kbps). For 32 kbps operation, each sample is converted into a 
 * 4-bit quantized difference signal resulting in a compression ratio of  
 * 2:1 over the G.711 format. For 24 kbps 40 kbps operation, the quantized 
 * difference signal is 3 bits and 5 bits, respectively. 
 * 
 * $Log: g726_16.c,v $ 
 * Revision 1.4  2002/11/20 04:29:13  robertj 
 * Included optimisations for G.711 and G.726 codecs, thanks Ted Szoczei 
 * 
 * Revision 1.1  2002/02/11 23:24:23  robertj 
 * Updated to openH323 v1.8.0 
 * 
 * Revision 1.2  2002/02/10 21:14:54  dereks 
 * Add cvs log history to head of the file. 
 * Ensure file is terminated by a newline. 
 * 
 * 
 * 
 * 
 */ 
#include "g72x.h" 
#include "private.h" 
 
/* 
 * Maps G.723_16 code word to reconstructed scale factor normalized log 
 * magnitude values.  Comes from Table 11/G.726 
 */ 
static short	_dqlntab[4] = { 116, 365, 365, 116};  
 
/* Maps G.723_16 code word to log of scale factor multiplier. 
 * 
 * _witab[4] is actually {-22 , 439, 439, -22}, but FILTD wants it 
 * as WI << 5  (multiplied by 32), so we'll do that here  
 */ 
static short	_witab[4] = {-704, 14048, 14048, -704}; 
 
/* 
 * Maps G.723_16 code words to a set of values whose long and short 
 * term averages are computed and then compared to give an indication 
 * how stationary (steady state) the signal is. 
 */ 
 
/* Comes from FUNCTF */ 
static short	_fitab[4] = {0, 0xE00, 0xE00, 0}; 
 
/* Comes from quantizer decision level tables (Table 7/G.726) 
 */ 
static int qtab_723_16[1] = {261}; 
 
 
/* 
 * g723_16_encoder() 
 * 
 * Encodes a linear PCM, A-law or u-law input sample and returns its 2-bit code. 
 * Returns -1 if invalid input coding value. 
 */ 
int 
g726_16_encoder( 
	int		sl, 
	int		in_coding, 
	g726_state *state_ptr) 
{ 
	int		sezi; 
	int		sez;			/* ACCUM */ 
	int		sei; 
	int		se; 
	int		d;				/* SUBTA */ 
	int		y;				/* MIX */ 
	int		i; 
	int		dq; 
	int		sr;				/* ADDB */ 
	int		dqsez;			/* ADDC */ 
 
	switch (in_coding) {	/* linearize input sample to 14-bit PCM */ 
	case AUDIO_ENCODING_ALAW: 
		sl = alaw2linear(sl) >> 2; 
		break; 
	case AUDIO_ENCODING_ULAW: 
		sl = ulaw2linear(sl) >> 2; 
		break; 
	case AUDIO_ENCODING_LINEAR: 
		sl >>= 2;		/* sl of 14-bit dynamic range */ 
		break; 
	default: 
		return (-1); 
	} 
 
	sezi = predictor_zero(state_ptr); 
	sez = sezi >> 1; 
	sei = sezi + predictor_pole(state_ptr); 
	se = sei >> 1;			/* se = estimated signal */ 
 
	d = sl - se;			/* d = estimation diff. */ 
 
	/* quantize prediction difference d */ 
	y = step_size(state_ptr);	/* quantizer step size */ 
	i = quantize(d, y, qtab_723_16, 1);  /* i = ADPCM code */ 
 
	      /* Since quantize() only produces a three level output 
	       * (1, 2, or 3), we must create the fourth one on our own 
	       */ 
	if (i == 3)                          /* i code for the zero region */ 
	  if ((d & 0x8000) == 0)             /* If d > 0, i=3 isn't right... */ 
	    i = 0; 
	     
	dq = reconstruct(i & 2, _dqlntab[i], y); /* quantized diff. */ 
 
	sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; /* reconstructed signal */ 
 
	dqsez = sr + sez - se;		/* pole prediction diff. */ 
 
	update(2, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr); 
 
	return (i); 
} 
 
/* 
 * g723_16_decoder() 
 * 
 * Decodes a 2-bit CCITT G.723_16 ADPCM code and returns 
 * the resulting 16-bit linear PCM, A-law or u-law sample value. 
 * -1 is returned if the output coding is unknown. 
 */ 
int 
g726_16_decoder( 
	int		i, 
	int		out_coding, 
	g726_state *state_ptr) 
{ 
	int		sezi; 
	int		sez;			/* ACCUM */ 
	int		sei; 
	int		se; 
	int		y;				/* MIX */ 
	int		dq; 
	int		sr;				/* ADDB */ 
	int		dqsez; 
 
	i &= 0x03;			/* mask to get proper bits */ 
	sezi = predictor_zero(state_ptr); 
	sez = sezi >> 1; 
	sei = sezi + predictor_pole(state_ptr); 
	se = sei >> 1;			/* se = estimated signal */ 
 
	y = step_size(state_ptr);	/* adaptive quantizer step size */ 
	dq = reconstruct(i & 0x02, _dqlntab[i], y); /* unquantize pred diff */ 
 
	sr = (dq < 0) ? (se - (dq & 0x3FFF)) : (se + dq); /* reconst. signal */ 
 
	dqsez = sr - se + sez;			/* pole prediction diff. */ 
 
	update(2, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr); 
 
	switch (out_coding) { 
	case AUDIO_ENCODING_ALAW: 
		return (tandem_adjust_alaw(sr, se, y, i, 2, qtab_723_16)); 
	case AUDIO_ENCODING_ULAW: 
		return (tandem_adjust_ulaw(sr, se, y, i, 2, qtab_723_16)); 
	case AUDIO_ENCODING_LINEAR: 
		return (sr << 2);	/* sr was of 14-bit dynamic range */ 
	default: 
		return (-1); 
	} 
}