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phoned/lib/libpvf/usr.c

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Import of libpvf
2005-06-13 22:11:33 -04:00
/*
* usr.c
*
* Conversion pvf <--> USR GSM and ADPCM formats.
*
* $Id: usr.c,v 1.5 2001/12/22 19:44:09 marcs Exp $
*
*/
#include "../include/voice.h"
/* Forward defs of the format-specific routines
*/
static int pvftousrgsm (FILE *fd_in, FILE *fd_out, pvf_header *header_in);
static int pvftousradpcm (FILE *fd_in, FILE *fd_out, pvf_header *header_in);
static int usrgsmtopvf (FILE *fd_in, FILE *fd_out, pvf_header *header_out);
static int usradpcmtopvf (FILE *fd_in, FILE *fd_out, pvf_header *header_out);
int pvftousr(FILE *fd_in, FILE *fd_out, int compression,
pvf_header *header_in) {
switch (compression) {
case 1:
return (pvftousrgsm(fd_in, fd_out, header_in));
case 4:
return (pvftousradpcm(fd_in, fd_out, header_in));
default:
return -1;
}
}
int usrtopvf (FILE *fd_in, FILE *fd_out, int compression,
pvf_header *header_out) {
switch (compression) {
case 1:
return (usrgsmtopvf(fd_in, fd_out, header_out));
case 4:
return (usradpcmtopvf(fd_in, fd_out, header_out));
default:
return -1;
}
}
/*****************
** GSM SECTION **
*****************/
#include "../libmgsm/gsm.h"
/* USR's GSM data format consists of 38-byte frames of data where the
* first two bytes of the frame (usually "0xFE 0xFE" for valid data and
* "0xB6 0xB6" for silence, and 3 bytes of trailer ("0x0 0xA5 0xA5") can
* be discarded, giving 33 bytes of useful data.
* Newer models can also generate frames with raw data (without the
* trailing and leading bytes).
* The decoding function tries to detect the frame type and pass the
* 33 bytes of data to a garden variety GSM decode process.
* In my case, I used GSM 06.10 from Technische
* Universitaet Berlin ftp://ftp.cs.tu-berlin.de/pub/local/kbs/tubmik/gsm/
*
* The pvftousrgsm function just generates the old type of frame
* since it can be played on both new and old models.
*/
unsigned char gsm_head[2] = { 0xfe, 0xfe };
unsigned char gsm_tail[3] = { 0x0, 0xa5, 0xa5 };
static int pvftousrgsm (FILE *fd_in, FILE *fd_out, pvf_header *header_in)
{
gsm r;
gsm_signal s[ 160 ];
gsm_frame d;
int opt_fast = 0;
int opt_verbose = 0;
int opt_ltp_cut = 0;
int i;
int sample = 0;
if (!(r = gsm_create())) {
perror("gsm_create");
return -1;
}
(void)gsm_option(r, GSM_OPT_FAST, &opt_fast);
(void)gsm_option(r, GSM_OPT_VERBOSE, &opt_verbose);
(void)gsm_option(r, GSM_OPT_LTP_CUT, &opt_ltp_cut);
/* GSM operates on frames of 160 samples each, so we may run up against
* the end of the file and be forced to backfill with zeroes.
*/
while (!feof(fd_in)) {
for (i=0;i<160;i++) {
sample = header_in->read_pvf_data(fd_in);
if (feof(fd_in)) {
memset((char *)(&s[i]), 0, sizeof(gsm_signal)*(160-i));
} else {
sample >>= 8;
if (sample > 0x7fff)
sample = 0x7fff;
if (sample < -0x8000)
sample = -0x8000;
s[i] = sample;
}
}
gsm_encode(r, s, d);
fwrite((char *)gsm_head, 2, 1, fd_out);
fwrite((char *)d, sizeof(d), 1, fd_out);
fwrite((char *)gsm_tail, 3, 1, fd_out);
}
gsm_destroy(r);
return(OK);
}
static int usrgsmtopvf (FILE *fd_in, FILE *fd_out, pvf_header *header_out)
{
unsigned char inbuf[38];
gsm r;
gsm_byte *s;
gsm_signal d[ 160 ];
int opt_fast = 0;
int opt_verbose = 0;
int i, sample, bytes2read, chunksread;
if (!(r = gsm_create())) {
perror("gsm_create");
return -1;
}
(void)gsm_option(r, GSM_OPT_FAST, &opt_fast);
(void)gsm_option(r, GSM_OPT_VERBOSE, &opt_verbose);
/*
* read the first frame to see if it has an
* header or is raw data
*/
if ((chunksread=fread(inbuf, 33, 1, fd_in)) > 0) {
if ((inbuf[0] == inbuf[1]) &&
((inbuf[0] == 0xfe) || (inbuf[0] == 0xb6))) {
/*
* has an header
*/
fread(&inbuf[33], 5, 1, fd_in);
s=&inbuf[2];
bytes2read=38;
} else
{
/*
* raw data
*/
s=&inbuf[0];
bytes2read=33;
}
}
while (chunksread > 0) {
/* --- MNI_p/JoSch --->
* I don'n know how this (now redundant to leave libmgsm untouched
* -> see ../libmgsm/decode.c line 20) control for GSM_MAGIC
* works with other USR- modems. Do they realy produce wrong
* bytes so that it is necessary to control it?
* For my US Robotics Vi 28.8 Faxmodem (gr) with Personal Voice
* Mail (speed 33600) this seems to work.
* May be there is a modem-setting I don't know (I don't know
* any voice-settig-switches for this modem) that switches the
* modem to a workable compression. At this time this patch works
* for me until I get informations from the USR-Support.
*/
if ((((*s >> 4) & 0x0F) != GSM_MAGIC) || (((*s >> 4) & 0x0F) != 0))
*s |= (GSM_MAGIC << 4);
/* <--- MNI_p/JoSch --- */
if (gsm_decode(r, s, d)) {
fprintf(stderr, "%s: bad frame in input\n", program_name);
gsm_destroy(r);
return(ERROR);
}
for (i=0;i<160;i++) {
sample = d[i];
if (sample > 0x7fff) {
sample -= 0x10000;
}
header_out->write_pvf_data(fd_out, sample << 8);
}
chunksread=fread(inbuf, bytes2read, 1, fd_in);
}
return(OK);
}
/*******************
** ADPCM SECTION **
*******************/
/* This ADPCM code was released by Sun Microsystems, Inc. to the public domain
* as a part of the CCITT (International Telegraph and Telephone Consultative
* Committee) reference implementation of the G.711, G.721 and G.723 voice
* compression standards.
*
* The following files from the original distribution have been concatenated
* in this section:
*
* g72x.h header file for g721.c, g723_24.c and g723_40.c
* g72x.c common denominator of G.721 and G.723 ADPCM codes
* g721.c CCITT G.721 32Kbps ADPCM coder (with g72x.c)
*
* Below this, the pvf conversion routines follow.
*/
/*
* 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.h
*
* Header file for CCITT conversion routines.
*
*/
#ifndef _G72X_H
#define _G72X_H
#define AUDIO_ENCODING_LINEAR (3) /* PCM 2's-complement (0-center) */
/*
* The following is the definition of the state structure
* used by the G.721/G.723 encoder and decoder to preserve their internal
* state between successive calls. The meanings of the majority
* of the state structure fields are explained in detail in the
* CCITT Recommendation G.721. The field names are essentially indentical
* to variable names in the bit level description of the coding algorithm
* included in this Recommendation.
*/
struct g72x_state {
long yl; /* Locked or steady state step size multiplier. */
short yu; /* Unlocked or non-steady state step size multiplier. */
short dms; /* Short term energy estimate. */
short dml; /* Long term energy estimate. */
short ap; /* Linear weighting coefficient of 'yl' and 'yu'. */
short a[2]; /* Coefficients of pole portion of prediction filter. */
short b[6]; /* Coefficients of zero portion of prediction filter. */
short pk[2]; /*
* Signs of previous two samples of a partially
* reconstructed signal.
*/
short dq[6]; /*
* Previous 6 samples of the quantized difference
* signal represented in an internal floating point
* format.
*/
short sr[2]; /*
* Previous 2 samples of the quantized difference
* signal represented in an internal floating point
* format.
*/
char td; /* delayed tone detect, new in 1988 version */
};
#endif /* !_G72X_H */
/*
* 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.
*/
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, /* wanmag, */ 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.
*/
static 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.
*
*/
static 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.
*
*/
static 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.
*
*/
static 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.
*/
static 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.
*/
static 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
*/
static 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 /* , mant */ ; /* Adaptive predictor, FLOAT A */
short a2p = 0; /* LIMC */
short a1ul; /* UPA1 */
short /* ua2, */ pks1; /* UPA2 */
short /* uga2a, */ fa1;
/* short uga2b; */
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 = 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] = (unsigned 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;
}
/*
* 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
*/
/*
* g721.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.
*
*/
static short 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.
*/
static int
g721_encoder(
int sl,
int in_coding,
struct g72x_state *state_ptr)
{
short sezi, se, sez; /* ACCUM */
short d; /* SUBTA */
short sr; /* ADDB */
short y; /* MIX */
short dqsez; /* ADDC */
short dq, i;
switch (in_coding) { /* linearize input sample to 14-bit PCM */
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.
*/
static int
g721_decoder(
int i,
int out_coding,
struct g72x_state *state_ptr)
{
short sezi, sei, sez, se; /* ACCUM */
short y; /* MIX */
short sr; /* ADDB */
short dq;
short dqsez;
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_LINEAR:
return (sr << 2); /* sr was 14-bit dynamic range */
default:
return (-1);
}
}
static int
pack_output(
unsigned code,
int bits,
FILE* fd_out)
{
static unsigned int out_buffer = 0;
static int out_bits = 0;
unsigned char out_byte;
out_buffer |= (code << out_bits);
out_bits += bits;
if (out_bits >= 8) {
out_byte = out_buffer & 0xff;
out_bits -= 8;
out_buffer >>= 8;
fwrite(&out_byte, sizeof (char), 1, fd_out);
}
return (out_bits > 0);
}
static int
unpack_input(
unsigned char *code,
int bits,
FILE* fd_in)
{
static unsigned int in_buffer = 0;
static int in_bits = 0;
unsigned char in_byte;
if (in_bits < bits) {
if (fread(&in_byte, sizeof (char), 1, fd_in) != 1) {
*code = 0;
return (-1);
}
in_buffer |= (in_byte << in_bits);
in_bits += 8;
}
*code = in_buffer & ((1 << bits) - 1);
in_buffer >>= bits;
in_bits -= bits;
return (in_bits > 0);
}
int pvftousradpcm (FILE *fd_in, FILE *fd_out, pvf_header *header_in)
{
struct g72x_state state;
int r = 0;
int sample = 0;
unsigned char code = 0;
g72x_init_state(&state);
/* GSM operates on frames of 160 samples each, so we may run up against
* the end of the file and be forced to backfill with zeroes.
*/
while (!feof(fd_in)) {
sample = header_in->read_pvf_data(fd_in);
sample >>= 8;
if (sample > 0x7fff)
sample = 0x7fff;
if (sample < -0x8000)
sample = -0x8000;
code = g721_encoder(sample, AUDIO_ENCODING_LINEAR, &state);
r = pack_output(code, 4, fd_out);
}
while (r) {
r = pack_output(0, 4, fd_out);
}
return(OK);
}
static int usradpcmtopvf (FILE *fd_in, FILE *fd_out, pvf_header *header_out)
{
int sample;
struct g72x_state state;
unsigned char code = 0;
while (unpack_input(&code, 4, fd_in) >= 0) {
sample = g721_decoder(code, AUDIO_ENCODING_LINEAR, &state);
header_out->write_pvf_data(fd_out, sample << 8);
}
return(OK);
}
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