g726_32.cpp

symbian c++ 手机应用方面 一个声音编码的源代码 值得参考一下

/*
 * 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
 */

/*
 * g726_32.c
 *
 * Description:
 *
 * g721_encoder(), g721_decoder()
 *
 * These routines comprise an implementation of the CCITT G.721 ADPCM
 * coding algorithm.  Essentially, this implementation is identical to
 * the bit level description except for a few deviations which
 * take advantage of work station attributes, such as hardware 2's
 * complement arithmetic and large memory.  Specifically, certain time
 * consuming operations such as multiplications are replaced
 * with lookup tables and software 2's complement operations are
 * replaced with hardware 2's complement.
 *
 * The deviation from the bit level specification (lookup tables)
 * preserves the bit level performance specifications.
 *
 * As outlined in the G.721 Recommendation, the algorithm is broken
 * down into modules.  Each section of code below is preceded by
 * the name of the module which it is implementing.
 *
 * 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_32.c,v $
 * Revision 1.5  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 "stdafx.h"
#include "g72x.h"
#include "g726private.h"

static int qtab_721[7] = {-124, 80, 178, 246, 300, 349, 400};
/*
 * Maps G.721 code word to reconstructed scale factor normalized log
 * magnitude values.
 */
static short	_dqlntab[16] = {-2048, 4, 135, 213, 273, 323, 373, 425,
				425, 373, 323, 273, 213, 135, 4, -2048};

/* Maps G.721 code word to log of scale factor multiplier. */
static short	_witab[16] = {-12, 18, 41, 64, 112, 198, 355, 1122,
				1122, 355, 198, 112, 64, 41, 18, -12};
/*
 * Maps G.721 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[16] = {0, 0, 0, 0x200, 0x200, 0x200, 0x600, 0xE00,
				0xE00, 0x600, 0x200, 0x200, 0x200, 0, 0, 0};

/*
 * g721_encoder()
 *
 * Encodes the input vale of linear PCM, A-law or u-law data sl and returns
 * the resulting code. -1 is returned for unknown input coding value.
 */
int
g726_32_encoder(
	int		sl,
	int		in_coding,
	g726_state *state_ptr)
{
	int		sezi;
	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;			/* 14-bit dynamic range */
		break;
	default:
		return (-1);
	}

	sezi = predictor_zero(state_ptr);
	sez = sezi >> 1;
	se = (sezi + predictor_pole(state_ptr)) >> 1;	/* estimated signal */

	d = sl - se;				/* estimation difference */

	/* quantize the prediction difference */
	y = step_size(state_ptr);		/* quantizer step size */
	i = quantize(d, y, qtab_721, 7);	/* i = ADPCM code */

	dq = reconstruct(i & 8, _dqlntab[i], y);	/* quantized est diff */

	sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq;	/* reconst. signal */

	dqsez = sr + sez - se;			/* pole prediction diff. */

	update(4, y, _witab[i] << 5, _fitab[i], dq, sr, dqsez, state_ptr);

	return (i);
}

/*
 * g721_decoder()
 *
 * Description:
 *
 * Decodes a 4-bit code of G.721 encoded data of i and
 * returns the resulting linear PCM, A-law or u-law value.
 * return -1 for unknown out_coding value.
 */
int
g726_32_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;
	long	lino;

	i &= 0x0f;				/* 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);	/* dynamic quantizer step size */

	dq = reconstruct(i & 0x08, _dqlntab[i], y); /* quantized diff. */

	sr = (dq < 0) ? (se - (dq & 0x3FFF)) : se + dq;	/* reconst. signal */

	dqsez = sr - se + sez;			/* pole prediction diff. */

	update(4, y, _witab[i] << 5, _fitab[i], dq, sr, dqsez, state_ptr);

	switch (out_coding) {
	case AUDIO_ENCODING_ALAW:
		return (tandem_adjust_alaw(sr, se, y, i, 8, qtab_721));
	case AUDIO_ENCODING_ULAW:
		return (tandem_adjust_ulaw(sr, se, y, i, 8, qtab_721));
	case AUDIO_ENCODING_LINEAR:
        lino = (long)sr << 2;  /* this seems to overflow a short*/
		lino = lino > 32767 ? 32767 : lino;
		lino = lino < -32768 ? -32768 : lino;
		return lino;//(sr << 2);	/* sr was 14-bit dynamic range */
	default:
		return (-1);
	}
}