g726_16.c

sloedgy open sip stack source code

/*
 * 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 [e-mail]
 *      -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 1.1  2006/06/26 03:02:36  joegenbaclor
 * I have decided to include the latest development realease  of OPAL tagged Deimos Devel 1 (June 8 2006) as inegrated classes to opensipstack to avoid future version conflicts due to the fast pace in OPAL development.   This move is also aimed to reduce the size of projects using OPAL componets such as the soon to be relased OpenSIPPhone.
 *
 * Revision 1.2  2002/11/20 04:53:16  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);
	}
}