/*
* CDDL HEADER START
*
* The contents of this file are subject to the terms of the
* Common Development and Distribution License, Version 1.0 only
* (the "License"). You may not use this file except in compliance
* with the License.
*
* You can obtain a copy of the license at usr/src/OPENSOLARIS.LICENSE
* or http://www.opensolaris.org/os/licensing.
* See the License for the specific language governing permissions
* and limitations under the License.
*
* When distributing Covered Code, include this CDDL HEADER in each
* file and include the License file at usr/src/OPENSOLARIS.LICENSE.
* If applicable, add the following below this CDDL HEADER, with the
* fields enclosed by brackets "[]" replaced with your own identifying
* information: Portions Copyright [yyyy] [name of copyright owner]
*
* CDDL HEADER END
*/
/*
* Copyright (c) 1992-2001 by Sun Microsystems, Inc.
* All rights reserved.
*/
#pragma ident "%Z%%M% %I% %E% SMI"
/*
* Description:
*
* g723_init_state(), g723_encode(), g723_decode()
*
* These routines comprise an implementation of the CCITT G.723 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 look up tables and software 2's complement operations are
* replaced with hardware 2's complement.
*
* The deviation (look up tables) from the bit level
* specification, preserves the bit level performance specifications.
*
* As outlined in the G.723 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.
*
*/
#include <stdlib.h>
#include <libaudio.h>
/*
* g723_tables.c
*
* Description:
*
* This file contains statically defined lookup tables for
* use with the G.723 coding routines.
*/
/*
* Maps G.723 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 code word to log of scale factor multiplier. */
static short _witab[8] = {-128, 960, 4384, 18624, 18624, 4384, 960, -128};
/*
* Maps G.723 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};
/*
* g723_init_state()
*
* Description:
*
* This routine initializes and/or resets the audio_encode_state structure
* pointed to by 'state_ptr'.
* All the state initial values are specified in the G.723 standard specs.
*/
void
g723_init_state(
struct audio_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;
state_ptr->leftover_cnt = 0; /* no left over codes */
}
/*
* _g723_fmult()
*
* returns the integer product of the "floating point" an and srn
* by the lookup table _fmultwanmant[].
*
*/
static int
_g723_fmult(
int an,
int srn)
{
short anmag, anexp, anmant;
short wanexp;
if (an == 0) {
return ((srn >= 0) ?
((srn & 077) + 1) >> (18 - (srn >> 6)) :
-(((srn & 077) + 1) >> (2 - (srn >> 6))));
} else if (an > 0) {
anexp = _fmultanexp[an] - 12;
anmant = ((anexp >= 0) ? an >> anexp : an << -anexp) & 07700;
if (srn >= 0) {
wanexp = anexp + (srn >> 6) - 7;
return ((wanexp >= 0) ?
(_fmultwanmant[(srn & 077) + anmant] << wanexp)
& 0x7FFF :
_fmultwanmant[(srn & 077) + anmant] >> -wanexp);
} else {
wanexp = anexp + (srn >> 6) - 0xFFF7;
return ((wanexp >= 0) ?
-((_fmultwanmant[(srn & 077) + anmant] << wanexp)
& 0x7FFF) :
-(_fmultwanmant[(srn & 077) + anmant] >> -wanexp));
}
} else {
anmag = (-an) & 0x1FFF;
anexp = _fmultanexp[anmag] - 12;
anmant = ((anexp >= 0) ? anmag >> anexp : anmag << -anexp)
& 07700;
if (srn >= 0) {
wanexp = anexp + (srn >> 6) - 7;
return ((wanexp >= 0) ?
-((_fmultwanmant[(srn & 077) + anmant] << wanexp)
& 0x7FFF) :
-(_fmultwanmant[(srn & 077) + anmant] >> -wanexp));
} else {
wanexp = anexp + (srn >> 6) - 0xFFF7;
return ((wanexp >= 0) ?
(_fmultwanmant[(srn & 077) + anmant] << wanexp)
& 0x7FFF :
_fmultwanmant[(srn & 077) + anmant] >> -wanexp);
}
}
}
/*
* _g723_update()
*
* updates the state variables for each output code
*
*/
static void
_g723_update(
int y,
int i,
int dq,
int sr,
int pk0,
struct audio_g72x_state *state_ptr,
int sigpk)
{
int cnt;
long fi; /* Adaptation speed control, FUNCTF */
short mag, exp; /* Adaptive predictor, FLOAT A */
short a2p; /* LIMC */
short a1ul; /* UPA1 */
short pks1, fa1; /* UPA2 */
char tr; /* tone/transition detector */
short thr2;
mag = dq & 0x3FFF;
/* TRANS */
if (state_ptr->td == 0)
tr = 0;
else if (state_ptr->yl > 0x40000)
tr = (mag <= 0x2F80) ? 0 : 1;
else {
thr2 = (0x20 + ((state_ptr->yl >> 10) & 0x1F)) <<
(state_ptr->yl >> 15);
if (mag >= thr2)
tr = 1;
else
tr = (mag <= (thr2 - (thr2 >> 2))) ? 0 : 1;
}
/*
* Quantizer scale factor adaptation.
*/
/* FUNCTW & FILTD & DELAY */
state_ptr->yu = y + ((_witab[i] - y) >> 5);
/* LIMB */
if (state_ptr->yu < 544)
state_ptr->yu = 544;
else if (state_ptr->yu > 5120)
state_ptr->yu = 5120;
/* FILTE & DELAY */
state_ptr->yl += state_ptr->yu + ((-state_ptr->yl) >> 6);
/*
* Adaptive predictor coefficients.
*/
if (tr == 1) {
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 {
/* UPA2 */
pks1 = pk0 ^ state_ptr->pk[0];
a2p = state_ptr->a[1] - (state_ptr->a[1] >> 7);
if (sigpk == 0) {
fa1 = (pks1) ? state_ptr->a[0] : -state_ptr->a[0];
if (fa1 < -8191)
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 */
state_ptr->a[0] -= state_ptr->a[0] >> 8;
if (sigpk == 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 of b's */
for (cnt = 0; cnt < 6; cnt++) {
state_ptr->b[cnt] -= state_ptr->b[cnt] >> 8;
if (dq & 0x3FFF) {
/* 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 */
if (mag == 0) {
state_ptr->dq[0] = (dq >= 0) ? 0x20 : 0xFC20;
} else {
exp = _fmultanexp[mag];
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 */
if (sr == 0) {
state_ptr->sr[0] = 0x20;
} else if (sr > 0) {
exp = _fmultanexp[sr];
state_ptr->sr[0] = (exp << 6) + ((sr << 6) >> exp);
} else {
mag = -sr;
exp = _fmultanexp[mag];
state_ptr->sr[0] = (exp << 6) + ((mag << 6) >> exp) - 0x400;
}
/* DELAY A */
state_ptr->pk[1] = state_ptr->pk[0];
state_ptr->pk[0] = pk0;
/* TONE */
if (tr == 1)
state_ptr->td = 0;
else if (a2p < -11776)
state_ptr->td = 1;
else
state_ptr->td = 0;
/*
* Adaptation speed control.
*/
fi = _fitab[i]; /* FUNCTF */
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;
}
/*
* _g723_quantize()
*
* Description:
*
* Given a raw sample, 'd', of the difference signal and a
* quantization step size scale factor, 'y', this routine returns the
* G.723 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.
*/
static unsigned int
_g723_quantize(
int d, /* Raw difference signal sample. */
int y) /* Step size multiplier. */
{
/* LOG */
short dqm; /* Magnitude of 'd'. */
short exp; /* Integer part of base 2 log of magnitude of 'd'. */
short mant; /* Fractional part of base 2 log. */
short dl; /* Log of magnitude of 'd'. */
/* SUBTB */
short dln; /* Step size scale factor normalized log. */
/* QUAN */
unsigned char i; /* G.723 codeword. */
/*
* LOG
*
* Compute base 2 log of 'd', and store in 'dln'.
*
*/
dqm = abs(d);
exp = _fmultanexp[dqm >> 1];
mant = ((dqm << 7) >> exp) & 0x7F; /* Fractional portion. */
dl = (exp << 7) + mant;
/*
* SUBTB
*
* "Divide" by step size multiplier.
*/
dln = dl - (y >> 2);
/*
* QUAN
*
* Obtain codword for 'd'.
*/
i = _g723quani[dln & 0xFFF];
if (d < 0)
i ^= 7; /* Stuff in sign of 'd'. */
else if (i == 0)
i = 7; /* New in 1988 revision */
return (i);
}
/*
* _g723_reconstr()
*
* Description:
*
* Returns reconstructed difference signal 'dq' obtained from
* G.723 codeword 'i' and quantization step size scale factor 'y'.
* Multiplication is performed in log base 2 domain as addition.
*/
static int
_g723_reconstr(
int i, /* G.723 codeword. */
unsigned long y) /* Step size multiplier. */
{
/* ADD A */
short dql; /* Log of 'dq' magnitude. */
/* ANTILOG */
short dex; /* Integer part of log. */
short dqt;
short dq; /* Reconstructed difference signal sample. */
dql = _dqlntab[i] + (y >> 2); /* ADDA */
if (dql < 0)
dq = 0;
else { /* ANTILOG */
dex = (dql >> 7) & 15;
dqt = 128 + (dql & 127);
dq = (dqt << 7) >> (14 - dex);
}
if (i & 4)
dq -= 0x8000;
return (dq);
}
/*
* _tandem_adjust(sr, se, y, i)
*
* Description:
*
* 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 Mu-law codes.
*
* Input:
* sr decoder output linear PCM sample,
* se predictor estimate sample,
* y quantizer step size,
* i decoder input code
*
* Return:
* adjusted A-law or Mu-law compressed sample.
*/
static 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 */
{
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 */
sp = audio_s2a((sr <= -0x2000)? -0x8000 :
(sr < 0x1FFF)? sr << 2 : 0x7FFF); /* short to A-law compression */
dx = (audio_a2s(sp) >> 2) - se; /* 16-bit prediction error */
id = _g723_quantize(dx, y);
if (id == i) /* no adjustment on sp */
return (sp);
else { /* sp adjustment needed */
im = i ^ 4; /* 2's complement to biased unsigned */
imx = id ^ 4;
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);
}
}
static 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 */
{
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 */
sp = audio_s2u((sr <= -0x2000)? -0x8000 :
(sr >= 0x1FFF)? 0x7FFF : sr << 2); /* short to u-law compression */
dx = (audio_u2s(sp) >> 2) - se; /* 16-bit prediction error */
id = _g723_quantize(dx, y);
if (id == i)
return (sp);
else {
/* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */
im = i ^ 4; /* 2's complement to biased unsigned */
imx = id ^ 4;
/* u-law codes : 0, 1, ... 7E, 7F, FF, FE, ... 81, 80 */
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);
}
}
static unsigned char
_encoder(
int sl,
struct audio_g72x_state *state_ptr)
{
short sei, sezi, se, sez; /* ACCUM */
short d; /* SUBTA */
float al; /* use floating point for faster multiply */
short y, dif; /* MIX */
short sr; /* ADDB */
short pk0, sigpk, dqsez; /* ADDC */
short dq, i;
int cnt;
/* ACCUM */
sezi = _g723_fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]);
for (cnt = 1; cnt < 6; cnt++)
sezi = sezi + _g723_fmult(state_ptr->b[cnt] >> 2,
state_ptr->dq[cnt]);
sei = sezi;
for (cnt = 1; cnt > -1; cnt--)
sei = sei + _g723_fmult(state_ptr->a[cnt] >> 2,
state_ptr->sr[cnt]);
sez = sezi >> 1;
se = sei >> 1;
d = sl - se; /* SUBTA */
if (state_ptr->ap >= 256)
y = state_ptr->yu;
else {
y = state_ptr->yl >> 6;
dif = state_ptr->yu - y;
al = state_ptr->ap >> 2;
if (dif > 0)
y += ((int)(dif * al)) >> 6;
else if (dif < 0)
y += ((int)(dif * al) + 0x3F) >> 6;
}
i = _g723_quantize(d, y);
dq = _g723_reconstr(i, y);
sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; /* ADDB */
dqsez = sr + sez - se; /* ADDC */
if (dqsez == 0) {
pk0 = 0;
sigpk = 1;
} else {
pk0 = (dqsez < 0) ? 1 : 0;
sigpk = 0;
}
_g723_update(y, i, dq, sr, pk0, state_ptr, sigpk);
return (i);
}
/*
* g723_encode()
*
* Description:
*
* Encodes a buffer of linear PCM, A-law or Mu-law data pointed to by 'in_buf'
* according the G.723 encoding algorithm and packs the resulting code words
* into bytes. The bytes of codewords are written to a buffer
* pointed to by 'out_buf'.
*
* Notes:
*
* In the event that the number packed codes is shorter than a sample unit,
* the remainder is saved in the state stucture till next call. It is then
* packed into the new buffer on the next call.
* The number of valid bytes in 'out_buf' is returned in *out_size. Note that
* this will not always be equal to 3/8 of 'data_size' on input. On the
* final call to 'g723_encode()' the calling program might want to
* check if any code bits was left over. This can be
* done by calling 'g723_encode()' with data_size = 0, which returns in
* *out_size a* 0 if nothing was leftover and the number of bits left over in
* the state structure which now is in out_buf[0].
*
* The 3 lower significant bits of an individual byte in the output byte
* stream is packed with a G.723 code first. Then the 3 higher order
* bits are packed with the next code.
*/
int
g723_encode(
void *in_buf,
int data_size,
Audio_hdr *in_header,
unsigned char *out_buf,
int *out_size,
struct audio_g72x_state *state_ptr)
{
int i;
unsigned char *out_ptr;
unsigned char *leftover;
unsigned int bits;
unsigned int codes;
int offset;
short *short_ptr;
unsigned char *char_ptr;
/* Dereference the array pointer for faster access */
leftover = &state_ptr->leftover[0];
/* Return all cached leftovers */
if (data_size == 0) {
for (i = 0; state_ptr->leftover_cnt > 0; i++) {
*out_buf++ = leftover[i];
state_ptr->leftover_cnt -= 8;
}
if (i > 0) {
/* Round up to a complete sample unit */
for (; i < 3; i++)
*out_buf++ = 0;
}
*out_size = i;
state_ptr->leftover_cnt = 0;
return (AUDIO_SUCCESS);
}
/* XXX - if linear, it had better be 16-bit! */
if (in_header->encoding == AUDIO_ENCODING_LINEAR) {
if (data_size & 1) {
return (AUDIO_ERR_BADFRAME);
} else {
data_size >>= 1;
short_ptr = (short *)in_buf;
}
} else {
char_ptr = (unsigned char *)in_buf;
}
out_ptr = (unsigned char *)out_buf;
offset = state_ptr->leftover_cnt / 8;
bits = state_ptr->leftover_cnt % 8;
codes = (bits > 0) ? leftover[offset] : 0;
while (data_size--) {
switch (in_header->encoding) {
case AUDIO_ENCODING_LINEAR:
i = _encoder(*short_ptr++ >> 2, state_ptr);
break;
case AUDIO_ENCODING_ALAW:
i = _encoder(audio_a2s(*char_ptr++) >> 2, state_ptr);
break;
case AUDIO_ENCODING_ULAW:
i = _encoder(audio_u2s(*char_ptr++) >> 2, state_ptr);
break;
default:
return (AUDIO_ERR_ENCODING);
}
/* pack the resulting code into leftover buffer */
codes += i << bits;
bits += 3;
if (bits >= 8) {
leftover[offset] = codes & 0xff;
bits -= 8;
codes >>= 8;
offset++;
}
state_ptr->leftover_cnt += 3;
/* got a whole sample unit so copy it out and reset */
if (bits == 0) {
*out_ptr++ = leftover[0];
*out_ptr++ = leftover[1];
*out_ptr++ = leftover[2];
codes = 0;
state_ptr->leftover_cnt = 0;
offset = 0;
}
}
/* If any residual bits, save them for the next call */
if (bits > 0) {
leftover[offset] = codes & 0xff;
state_ptr->leftover_cnt += bits;
}
*out_size = (out_ptr - (unsigned char *)out_buf);
return (AUDIO_SUCCESS);
}
/*
* g723_decode()
*
* Description:
*
* Decodes a buffer of G.723 encoded data pointed to by 'in_buf' and
* writes the resulting linear PCM, A-law or Mu-law words into a buffer
* pointed to by 'out_buf'.
*
*/
int
g723_decode(
unsigned char *in_buf, /* Buffer of g723 encoded data. */
int data_size, /* Size in bytes of in_buf. */
Audio_hdr *out_header,
void *out_buf, /* Decoded data buffer. */
int *out_size,
struct audio_g72x_state *state_ptr) /* the decoder's state structure. */
{
unsigned char *inbuf_end;
unsigned char *in_ptr, *out_ptr;
short *linear_ptr;
unsigned int codes;
unsigned int bits;
int cnt;
short sezi, sei, sez, se; /* ACCUM */
float al; /* use floating point for faster multiply */
short y, dif; /* MIX */
short sr; /* ADDB */
char pk0; /* ADDC */
short dq;
char sigpk;
short dqsez;
unsigned char i;
in_ptr = in_buf;
inbuf_end = in_buf + data_size;
out_ptr = (unsigned char *)out_buf;
linear_ptr = (short *)out_buf;
/* Leftovers in decoding are only up to 8 bits */
bits = state_ptr->leftover_cnt;
codes = (bits > 0) ? state_ptr->leftover[0] : 0;
while ((bits >= 3) || (in_ptr < (unsigned char *)inbuf_end)) {
if (bits < 3) {
codes += *in_ptr++ << bits;
bits += 8;
}
/* ACCUM */
sezi = _g723_fmult(state_ptr->b[0] >> 2, state_ptr->dq[0]);
for (cnt = 1; cnt < 6; cnt++)
sezi = sezi + _g723_fmult(state_ptr->b[cnt] >> 2,
state_ptr->dq[cnt]);
sei = sezi;
for (cnt = 1; cnt >= 0; cnt--)
sei = sei + _g723_fmult(state_ptr->a[cnt] >> 2,
state_ptr->sr[cnt]);
sez = sezi >> 1;
se = sei >> 1;
if (state_ptr->ap >= 256)
y = state_ptr->yu;
else {
y = state_ptr->yl >> 6;
dif = state_ptr->yu - y;
al = state_ptr->ap >> 2;
if (dif > 0)
y += ((int)(dif * al)) >> 6;
else if (dif < 0)
y += ((int)(dif * al) + 0x3F) >> 6;
}
i = codes & 7;
dq = _g723_reconstr(i, y);
/* ADDB */
if (dq < 0)
sr = se - (dq & 0x3FFF);
else
sr = se + dq;
dqsez = sr - se + sez; /* ADDC */
pk0 = (dqsez < 0) ? 1 : 0;
sigpk = (dqsez) ? 0 : 1;
_g723_update(y, i, dq, sr, pk0, state_ptr, sigpk);
switch (out_header->encoding) {
case AUDIO_ENCODING_LINEAR:
*linear_ptr++ = ((sr <= -0x2000) ? -0x8000 :
(sr >= 0x1FFF) ? 0x7FFF : sr << 2);
break;
case AUDIO_ENCODING_ALAW:
*out_ptr++ = _tandem_adjust_alaw(sr, se, y, i);
break;
case AUDIO_ENCODING_ULAW:
*out_ptr++ = _tandem_adjust_ulaw(sr, se, y, i);
break;
default:
return (AUDIO_ERR_ENCODING);
}
codes >>= 3;
bits -= 3;
}
state_ptr->leftover_cnt = bits;
if (bits > 0)
state_ptr->leftover[0] = codes;
/* Calculate number of samples returned */
if (out_header->encoding == AUDIO_ENCODING_LINEAR)
*out_size = linear_ptr - (short *)out_buf;
else
*out_size = out_ptr - (unsigned char *)out_buf;
return (AUDIO_SUCCESS);
}