mirror of
https://github.com/signalwire/freeswitch.git
synced 2025-02-25 10:54:45 +00:00
8c79672f12
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@775 d0543943-73ff-0310-b7d9-9358b9ac24b2
214 lines
6.3 KiB
C
214 lines
6.3 KiB
C
/*
|
|
* 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);
|
|
}
|
|
}
|
|
|