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fteqw/engine/sndcodec/g72x.c
Spoike 7f00235804 Non-Functional voice communication.
git-svn-id: https://svn.code.sf.net/p/fteqw/code/trunk@16 fc73d0e0-1445-4013-8a0c-d673dee63da5
2004-08-23 01:36:14 +00:00

573 lines
14 KiB
C

#include "bothdefs.h"
#ifdef VOICECHAT
/*
* 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
*/
/*
* g72x.c
*
* Common routines for G.721 and G.723 conversions.
*/
#include "g72x.h"
#include <math.h>
#include <stdlib.h>
static short power2[15] = {1, 2, 4, 8, 0x10, 0x20, 0x40, 0x80,
0x100, 0x200, 0x400, 0x800, 0x1000, 0x2000, 0x4000};
/*
* quan()
*
* quantizes the input val against the table of size short integers.
* It returns i if table[i - 1] <= val < table[i].
*
* Using linear search for simple coding.
*/
static int
quan(
int val,
short *table,
int size)
{
int i;
for (i = 0; i < size; i++)
if (val < *table++)
break;
return (i);
}
/*
* fmult()
*
* returns the integer product of the 14-bit integer "an" and
* "floating point" representation (4-bit exponent, 6-bit mantessa) "srn".
*/
static int
fmult(
int an,
int srn)
{
short anmag, anexp, anmant;
short wanexp, wanmant;
short retval;
anmag = (an > 0) ? an : ((-an) & 0x1FFF);
anexp = quan(anmag, power2, 15) - 6;
anmant = (anmag == 0) ? 32 :
(anexp >= 0) ? anmag >> anexp : anmag << -anexp;
wanexp = anexp + ((srn >> 6) & 0xF) - 13;
wanmant = (anmant * (srn & 077) + 0x30) >> 4;
retval = (wanexp >= 0) ? ((wanmant << wanexp) & 0x7FFF) :
(wanmant >> -wanexp);
return (((an ^ srn) < 0) ? -retval : retval);
}
/*
* g72x_init_state()
*
* This routine initializes and/or resets the g72x_state structure
* pointed to by 'state_ptr'.
* All the initial state values are specified in the CCITT G.721 document.
*/
void
g72x_init_state(
struct g72x_state *state_ptr)
{
int cnta;
state_ptr->yl = 34816;
state_ptr->yu = 544;
state_ptr->dms = 0;
state_ptr->dml = 0;
state_ptr->ap = 0;
for (cnta = 0; cnta < 2; cnta++) {
state_ptr->a[cnta] = 0;
state_ptr->pk[cnta] = 0;
state_ptr->sr[cnta] = 32;
}
for (cnta = 0; cnta < 6; cnta++) {
state_ptr->b[cnta] = 0;
state_ptr->dq[cnta] = 32;
}
state_ptr->td = 0;
}
/*
* predictor_zero()
*
* computes the estimated signal from 6-zero predictor.
*
*/
int
predictor_zero(
struct g72x_state *state_ptr)
{
int i;
int sezi;
sezi = fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]);
for (i = 1; i < 6; i++) /* ACCUM */
sezi += fmult(state_ptr->b[i] >> 2, state_ptr->dq[i]);
return (sezi);
}
/*
* predictor_pole()
*
* computes the estimated signal from 2-pole predictor.
*
*/
int
predictor_pole(
struct g72x_state *state_ptr)
{
return (fmult(state_ptr->a[1] >> 2, state_ptr->sr[1]) +
fmult(state_ptr->a[0] >> 2, state_ptr->sr[0]));
}
/*
* step_size()
*
* computes the quantization step size of the adaptive quantizer.
*
*/
int
step_size(
struct g72x_state *state_ptr)
{
int y;
int dif;
int al;
if (state_ptr->ap >= 256)
return (state_ptr->yu);
else {
y = state_ptr->yl >> 6;
dif = state_ptr->yu - y;
al = state_ptr->ap >> 2;
if (dif > 0)
y += (dif * al) >> 6;
else if (dif < 0)
y += (dif * al + 0x3F) >> 6;
return (y);
}
}
/*
* quantize()
*
* Given a raw sample, 'd', of the difference signal and a
* quantization step size scale factor, 'y', this routine returns the
* ADPCM codeword to which that sample gets quantized. The step
* size scale factor division operation is done in the log base 2 domain
* as a subtraction.
*/
int
quantize(
int d, /* Raw difference signal sample */
int y, /* Step size multiplier */
short *table, /* quantization table */
int size) /* table size of short integers */
{
short dqm; /* Magnitude of 'd' */
short exp; /* Integer part of base 2 log of 'd' */
short mant; /* Fractional part of base 2 log */
short dl; /* Log of magnitude of 'd' */
short dln; /* Step size scale factor normalized log */
int i;
/*
* LOG
*
* Compute base 2 log of 'd', and store in 'dl'.
*/
dqm = abs(d);
exp = quan(dqm >> 1, power2, 15);
mant = ((dqm << 7) >> exp) & 0x7F; /* Fractional portion. */
dl = (exp << 7) + mant;
/*
* SUBTB
*
* "Divide" by step size multiplier.
*/
dln = dl - (y >> 2);
/*
* QUAN
*
* Obtain codword i for 'd'.
*/
i = quan(dln, table, size);
if (d < 0) /* take 1's complement of i */
return ((size << 1) + 1 - i);
else if (i == 0) /* take 1's complement of 0 */
return ((size << 1) + 1); /* new in 1988 */
else
return (i);
}
/*
* reconstruct()
*
* Returns reconstructed difference signal 'dq' obtained from
* codeword 'i' and quantization step size scale factor 'y'.
* Multiplication is performed in log base 2 domain as addition.
*/
int
reconstruct(
int sign, /* 0 for non-negative value */
int dqln, /* G.72x codeword */
int y) /* Step size multiplier */
{
short dql; /* Log of 'dq' magnitude */
short dex; /* Integer part of log */
short dqt;
short dq; /* Reconstructed difference signal sample */
dql = dqln + (y >> 2); /* ADDA */
if (dql < 0) {
return ((sign) ? -0x8000 : 0);
} else { /* ANTILOG */
dex = (dql >> 7) & 15;
dqt = 128 + (dql & 127);
dq = (dqt << 7) >> (14 - dex);
return ((sign) ? (dq - 0x8000) : dq);
}
}
/*
* update()
*
* updates the state variables for each output code
*/
void
update(
int code_size, /* distinguish 723_40 with others */
int y, /* quantizer step size */
int wi, /* scale factor multiplier */
int fi, /* for long/short term energies */
int dq, /* quantized prediction difference */
int sr, /* reconstructed signal */
int dqsez, /* difference from 2-pole predictor */
struct g72x_state *state_ptr) /* coder state pointer */
{
int cnt;
short mag, exp; /* Adaptive predictor, FLOAT A */
short a2p; /* LIMC */
short a1ul; /* UPA1 */
short pks1; /* UPA2 */
short fa1;
char tr; /* tone/transition detector */
short ylint, thr2, dqthr;
short ylfrac, thr1;
short pk0;
pk0 = (dqsez < 0) ? 1 : 0; /* needed in updating predictor poles */
mag = dq & 0x7FFF; /* prediction difference magnitude */
/* TRANS */
ylint = (short)(state_ptr->yl >> 15); /* exponent part of yl */
ylfrac = (state_ptr->yl >> 10) & 0x1F; /* fractional part of yl */
thr1 = (32 + ylfrac) << ylint; /* threshold */
thr2 = (ylint > 9) ? 31 << 10 : thr1; /* limit thr2 to 31 << 10 */
dqthr = (thr2 + (thr2 >> 1)) >> 1; /* dqthr = 0.75 * thr2 */
if (state_ptr->td == 0) /* signal supposed voice */
tr = 0;
else if (mag <= dqthr) /* supposed data, but small mag */
tr = 0; /* treated as voice */
else /* signal is data (modem) */
tr = 1;
/*
* Quantizer scale factor adaptation.
*/
/* FUNCTW & FILTD & DELAY */
/* update non-steady state step size multiplier */
state_ptr->yu = y + ((wi - y) >> 5);
/* LIMB */
if (state_ptr->yu < 544) /* 544 <= yu <= 5120 */
state_ptr->yu = 544;
else if (state_ptr->yu > 5120)
state_ptr->yu = 5120;
/* FILTE & DELAY */
/* update steady state step size multiplier */
state_ptr->yl += state_ptr->yu + ((-state_ptr->yl) >> 6);
/*
* Adaptive predictor coefficients.
*/
if (tr == 1) { /* reset a's and b's for modem signal */
state_ptr->a[0] = 0;
state_ptr->a[1] = 0;
state_ptr->b[0] = 0;
state_ptr->b[1] = 0;
state_ptr->b[2] = 0;
state_ptr->b[3] = 0;
state_ptr->b[4] = 0;
state_ptr->b[5] = 0;
} else { /* update a's and b's */
pks1 = pk0 ^ state_ptr->pk[0]; /* UPA2 */
/* update predictor pole a[1] */
a2p = state_ptr->a[1] - (state_ptr->a[1] >> 7);
if (dqsez != 0) {
fa1 = (pks1) ? state_ptr->a[0] : -state_ptr->a[0];
if (fa1 < -8191) /* a2p = function of fa1 */
a2p -= 0x100;
else if (fa1 > 8191)
a2p += 0xFF;
else
a2p += fa1 >> 5;
if (pk0 ^ state_ptr->pk[1])
/* LIMC */
if (a2p <= -12160)
a2p = -12288;
else if (a2p >= 12416)
a2p = 12288;
else
a2p -= 0x80;
else if (a2p <= -12416)
a2p = -12288;
else if (a2p >= 12160)
a2p = 12288;
else
a2p += 0x80;
}
/* TRIGB & DELAY */
state_ptr->a[1] = a2p;
/* UPA1 */
/* update predictor pole a[0] */
state_ptr->a[0] -= state_ptr->a[0] >> 8;
if (dqsez != 0)
{
if (pks1 == 0)
state_ptr->a[0] += 192;
else
state_ptr->a[0] -= 192;
}
/* LIMD */
a1ul = 15360 - a2p;
if (state_ptr->a[0] < -a1ul)
state_ptr->a[0] = -a1ul;
else if (state_ptr->a[0] > a1ul)
state_ptr->a[0] = a1ul;
/* UPB : update predictor zeros b[6] */
for (cnt = 0; cnt < 6; cnt++) {
if (code_size == 5) /* for 40Kbps G.723 */
state_ptr->b[cnt] -= state_ptr->b[cnt] >> 9;
else /* for G.721 and 24Kbps G.723 */
state_ptr->b[cnt] -= state_ptr->b[cnt] >> 8;
if (dq & 0x7FFF) { /* XOR */
if ((dq ^ state_ptr->dq[cnt]) >= 0)
state_ptr->b[cnt] += 128;
else
state_ptr->b[cnt] -= 128;
}
}
}
for (cnt = 5; cnt > 0; cnt--)
state_ptr->dq[cnt] = state_ptr->dq[cnt-1];
/* FLOAT A : convert dq[0] to 4-bit exp, 6-bit mantissa f.p. */
if (mag == 0) {
state_ptr->dq[0] = (dq >= 0) ? 0x20 : 0xFC20;
} else {
exp = quan(mag, power2, 15);
state_ptr->dq[0] = (dq >= 0) ?
(exp << 6) + ((mag << 6) >> exp) :
(exp << 6) + ((mag << 6) >> exp) - 0x400;
}
state_ptr->sr[1] = state_ptr->sr[0];
/* FLOAT B : convert sr to 4-bit exp., 6-bit mantissa f.p. */
if (sr == 0) {
state_ptr->sr[0] = 0x20;
} else if (sr > 0) {
exp = quan(sr, power2, 15);
state_ptr->sr[0] = (exp << 6) + ((sr << 6) >> exp);
} else if (sr > -32768) {
mag = -sr;
exp = quan(mag, power2, 15);
state_ptr->sr[0] = (exp << 6) + ((mag << 6) >> exp) - 0x400;
} else
state_ptr->sr[0] = (short)0xFC20;
/* DELAY A */
state_ptr->pk[1] = state_ptr->pk[0];
state_ptr->pk[0] = pk0;
/* TONE */
if (tr == 1) /* this sample has been treated as data */
state_ptr->td = 0; /* next one will be treated as voice */
else if (a2p < -11776) /* small sample-to-sample correlation */
state_ptr->td = 1; /* signal may be data */
else /* signal is voice */
state_ptr->td = 0;
/*
* Adaptation speed control.
*/
state_ptr->dms += (fi - state_ptr->dms) >> 5; /* FILTA */
state_ptr->dml += (((fi << 2) - state_ptr->dml) >> 7); /* FILTB */
if (tr == 1)
state_ptr->ap = 256;
else if (y < 1536) /* SUBTC */
state_ptr->ap += (0x200 - state_ptr->ap) >> 4;
else if (state_ptr->td == 1)
state_ptr->ap += (0x200 - state_ptr->ap) >> 4;
else if (abs((state_ptr->dms << 2) - state_ptr->dml) >=
(state_ptr->dml >> 3))
state_ptr->ap += (0x200 - state_ptr->ap) >> 4;
else
state_ptr->ap += (-state_ptr->ap) >> 4;
}
/*
* tandem_adjust(sr, se, y, i, sign)
*
* At the end of ADPCM decoding, it simulates an encoder which may be receiving
* the output of this decoder as a tandem process. If the output of the
* simulated encoder differs from the input to this decoder, the decoder output
* is adjusted by one level of A-law or u-law codes.
*
* Input:
* sr decoder output linear PCM sample,
* se predictor estimate sample,
* y quantizer step size,
* i decoder input code,
* sign sign bit of code i
*
* Return:
* adjusted A-law or u-law compressed sample.
*/
int
tandem_adjust_alaw(
int sr, /* decoder output linear PCM sample */
int se, /* predictor estimate sample */
int y, /* quantizer step size */
int i, /* decoder input code */
int sign,
short *qtab)
{
unsigned char sp; /* A-law compressed 8-bit code */
short dx; /* prediction error */
char id; /* quantized prediction error */
int sd; /* adjusted A-law decoded sample value */
int im; /* biased magnitude of i */
int imx; /* biased magnitude of id */
if (sr <= -32768)
sr = -1;
sp = linear2alaw((sr >> 1) << 3); /* short to A-law compression */
dx = (alaw2linear(sp) >> 2) - se; /* 16-bit prediction error */
id = quantize(dx, y, qtab, sign - 1);
if (id == i) { /* no adjustment on sp */
return (sp);
} else { /* sp adjustment needed */
/* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */
im = i ^ sign; /* 2's complement to biased unsigned */
imx = id ^ sign;
if (imx > im) { /* sp adjusted to next lower value */
if (sp & 0x80) {
sd = (sp == 0xD5) ? 0x55 :
((sp ^ 0x55) - 1) ^ 0x55;
} else {
sd = (sp == 0x2A) ? 0x2A :
((sp ^ 0x55) + 1) ^ 0x55;
}
} else { /* sp adjusted to next higher value */
if (sp & 0x80)
sd = (sp == 0xAA) ? 0xAA :
((sp ^ 0x55) + 1) ^ 0x55;
else
sd = (sp == 0x55) ? 0xD5 :
((sp ^ 0x55) - 1) ^ 0x55;
}
return (sd);
}
}
int
tandem_adjust_ulaw(
int sr, /* decoder output linear PCM sample */
int se, /* predictor estimate sample */
int y, /* quantizer step size */
int i, /* decoder input code */
int sign,
short *qtab)
{
unsigned char sp; /* u-law compressed 8-bit code */
short dx; /* prediction error */
char id; /* quantized prediction error */
int sd; /* adjusted u-law decoded sample value */
int im; /* biased magnitude of i */
int imx; /* biased magnitude of id */
if (sr <= -32768)
sr = 0;
sp = linear2ulaw(sr << 2); /* short to u-law compression */
dx = (ulaw2linear(sp) >> 2) - se; /* 16-bit prediction error */
id = quantize(dx, y, qtab, sign - 1);
if (id == i) {
return (sp);
} else {
/* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */
im = i ^ sign; /* 2's complement to biased unsigned */
imx = id ^ sign;
if (imx > im) { /* sp adjusted to next lower value */
if (sp & 0x80)
sd = (sp == 0xFF) ? 0x7E : sp + 1;
else
sd = (sp == 0) ? 0 : sp - 1;
} else { /* sp adjusted to next higher value */
if (sp & 0x80)
sd = (sp == 0x80) ? 0x80 : sp - 1;
else
sd = (sp == 0x7F) ? 0xFE : sp + 1;
}
return (sd);
}
}
#endif