www.pudn.com > g.726.rar > g726_24.c, change:2002-11-20,size:5887b


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/* 
 * g726_24.c 
 * 
 * Description: 
 * 
 * g723_24_encoder(), g723_24_decoder() 
 * 
 * These routines comprise an implementation of the CCITT G.723 24 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_24.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_24 code word to reconstructed scale factor normalized log 
 * magnitude values. 
 */ 
static short	_dqlntab[8] = {-2048, 135, 273, 373, 373, 273, 135, -2048}; 
 
/* Maps G.723_24 code word to log of scale factor multiplier. */ 
static short	_witab[8] = {-128, 960, 4384, 18624, 18624, 4384, 960, -128}; 
 
/* 
 * Maps G.723_24 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. 
 */ 
static short	_fitab[8] = {0, 0x200, 0x400, 0xE00, 0xE00, 0x400, 0x200, 0}; 
 
static int qtab_723_24[3] = {8, 218, 331}; 
 
/* 
 * g723_24_encoder() 
 * 
 * Encodes a linear PCM, A-law or u-law input sample and returns its 3-bit code. 
 * Returns -1 if invalid input coding value. 
 */ 
int 
g726_24_encoder( 
	int		sl, 
	int		in_coding, 
	g726_state *state_ptr) 
{ 
	int		sezi; 
	int		sei; 
	int		sez;			/* ACCUM */ 
	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_24, 3);	/* i = ADPCM code */ 
	dq = reconstruct(i & 4, _dqlntab[i], y); /* quantized diff. */ 
 
	sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; /* reconstructed signal */ 
 
	dqsez = sr + sez - se;		/* pole prediction diff. */ 
 
	update(3, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr); 
 
	return (i); 
} 
 
/* 
 * g723_24_decoder() 
 * 
 * Decodes a 3-bit CCITT G.723_24 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_24_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 &= 0x07;				/* 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 & 0x04, _dqlntab[i], y); /* unquantize pred diff */ 
 
	sr = (dq < 0) ? (se - (dq & 0x3FFF)) : (se + dq); /* reconst. signal */ 
 
	dqsez = sr - se + sez;			/* pole prediction diff. */ 
 
	update(3, 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, 4, qtab_723_24)); 
	case AUDIO_ENCODING_ULAW: 
		return (tandem_adjust_ulaw(sr, se, y, i, 4, qtab_723_24)); 
	case AUDIO_ENCODING_LINEAR: 
		return (sr << 2);	/* sr was of 14-bit dynamic range */ 
	default: 
		return (-1); 
	} 
}