Lets get the g726 lib in here for later use.

git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@775 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Brian West committed 2006-03-07 05:41:10 +00:00
1 parent a723adb3e0
commit 8c79672f12
25 files changed
+42890

No files matched your search

+213
View File
@@ -0,0 +1,213 @@
/*
* 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);
}
}
+190
View File
@@ -0,0 +1,190 @@
/*
* 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_24.c
*
* Description:
*
* g723_24_encoder(), g723_24_decoder()
*
* These routines comprise an implementation of the CCITT G.723 24 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_24.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_24 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_24 code word to log of scale factor multiplier. */
static short _witab[8] = {-128, 960, 4384, 18624, 18624, 4384, 960, -128};
/*
* Maps G.723_24 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};
static int qtab_723_24[3] = {8, 218, 331};
/*
* g723_24_encoder()
*
* Encodes a linear PCM, A-law or u-law input sample and returns its 3-bit code.
* Returns -1 if invalid input coding value.
*/
int
g726_24_encoder(
int sl,
int in_coding,
g726_state *state_ptr)
{
int sezi;
int sei;
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; /* 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_24, 3); /* i = ADPCM code */
dq = reconstruct(i & 4, _dqlntab[i], y); /* quantized diff. */
sr = (dq < 0) ? se - (dq & 0x3FFF) : se + dq; /* reconstructed signal */
dqsez = sr + sez - se; /* pole prediction diff. */
update(3, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
return (i);
}
/*
* g723_24_decoder()
*
* Decodes a 3-bit CCITT G.723_24 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_24_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 &= 0x07; /* 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 & 0x04, _dqlntab[i], y); /* unquantize pred diff */
sr = (dq < 0) ? (se - (dq & 0x3FFF)) : (se + dq); /* reconst. signal */
dqsez = sr - se + sez; /* pole prediction diff. */
update(3, 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, 4, qtab_723_24));
case AUDIO_ENCODING_ULAW:
return (tandem_adjust_ulaw(sr, se, y, i, 4, qtab_723_24));
case AUDIO_ENCODING_LINEAR:
return (sr << 2); /* sr was of 14-bit dynamic range */
default:
return (-1);
}
}
+208
View File
@@ -0,0 +1,208 @@
/*
* 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 "g72x.h"
#include "private.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);
}
}
+211
View File
@@ -0,0 +1,211 @@
/*
* 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_40.c
*
* Description:
*
* g723_40_encoder(), g723_40_decoder()
*
* These routines comprise an implementation of the CCITT G.723 40Kbps
* 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 deviation from the bit level specification (lookup tables),
* 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.
*
* 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_40.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_40 code word to ructeconstructed scale factor normalized log
* magnitude values.
*/
static short _dqlntab[32] = {-2048, -66, 28, 104, 169, 224, 274, 318,
358, 395, 429, 459, 488, 514, 539, 566,
566, 539, 514, 488, 459, 429, 395, 358,
318, 274, 224, 169, 104, 28, -66, -2048};
/* Maps G.723_40 code word to log of scale factor multiplier. */
static short _witab[32] = {448, 448, 768, 1248, 1280, 1312, 1856, 3200,
4512, 5728, 7008, 8960, 11456, 14080, 16928, 22272,
22272, 16928, 14080, 11456, 8960, 7008, 5728, 4512,
3200, 1856, 1312, 1280, 1248, 768, 448, 448};
/*
* Maps G.723_40 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[32] = {0, 0, 0, 0, 0, 0x200, 0x200, 0x200,
0x200, 0x200, 0x400, 0x600, 0x800, 0xA00, 0xC00, 0xC00,
0xC00, 0xC00, 0xA00, 0x800, 0x600, 0x400, 0x200, 0x200,
0x200, 0x200, 0x200, 0, 0, 0, 0, 0};
static int qtab_723_40[15] = {-122, -16, 68, 139, 198, 250, 298, 339,
378, 413, 445, 475, 502, 528, 553};
/*
* g723_40_encoder()
*
* Encodes a 16-bit linear PCM, A-law or u-law input sample and returns
* the resulting 5-bit CCITT G.723 40Kbps code.
* Returns -1 if the input coding value is invalid.
*/
int
g726_40_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 difference */
/* quantize prediction difference */
y = step_size(state_ptr); /* adaptive quantizer step size */
i = quantize(d, y, qtab_723_40, 15); /* i = ADPCM code */
dq = reconstruct(i & 0x10, _dqlntab[i], y); /* quantized diff */
sr = (dq < 0) ? se - (dq & 0x7FFF) : se + dq; /* reconstructed signal */
dqsez = sr + sez - se; /* dqsez = pole prediction diff. */
update(5, y, _witab[i], _fitab[i], dq, sr, dqsez, state_ptr);
return (i);
}
/*
* g723_40_decoder()
*
* Decodes a 5-bit CCITT G.723 40Kbps 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_40_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 &= 0x1f; /* 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 & 0x10, _dqlntab[i], y); /* estimation diff. */
sr = (dq < 0) ? (se - (dq & 0x7FFF)) : (se + dq); /* reconst. signal */
dqsez = sr - se + sez; /* pole prediction diff. */
update(5, 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, 0x10, qtab_723_40));
case AUDIO_ENCODING_ULAW:
return (tandem_adjust_ulaw(sr, se, y, i, 0x10, qtab_723_40));
case AUDIO_ENCODING_LINEAR:
return (sr << 2); /* sr was of 14-bit dynamic range */
default:
return (-1);
}
}
+582
View File
@@ -0,0 +1,582 @@
/*
* 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
*/
/*
* $Log: g72x.c,v $
* Revision 1.3 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.
*
*
*
* Common routines for G.721 and G.723 conversions.
*/
#include "g72x.h"
#include "private.h"
static int 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,
int * 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)
{
int anmag;
int anexp;
int anmant;
int wanexp;
int wanmant;
int 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
g726_init_state(
g726_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(
g726_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(
g726_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(
g726_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 */
int * table, /* quantization table */
int size) /* table size of integers */
{
int dqm; /* Magnitude of 'd' */
int exp; /* Integer part of base 2 log of 'd' */
int mant; /* Fractional part of base 2 log */
int dl; /* Log of magnitude of 'd' */
int 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 */
{
int dql; /* Log of 'dq' magnitude */
int dex; /* Integer part of log */
int dqt;
int 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 = (short)((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 */
g726_state *state_ptr) /* coder state pointer */
{
int cnt;
int mag, exp; /* Adaptive predictor, FLOAT A */
int a2p; /* LIMC */
int a1ul; /* UPA1 */
int pks1; /* UPA2 */
int fa1;
int tr; /* tone/transition detector */
int ylint, thr2, dqthr;
int ylfrac, thr1;
int 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;
a2p = 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] = (short)((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] = 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,
int * qtab)
{
int sp; /* A-law compressed 8-bit code */
int dx; /* prediction error */
int 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,
int * qtab)
{
int sp; /* u-law compressed 8-bit code */
int dx; /* prediction error */
int 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);
}
}
/* Allocate and Init. codec struct */
+109
View File
@@ -0,0 +1,109 @@
/*
* 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_ULAW (1) /* ISDN u-law */
#define AUDIO_ENCODING_ALAW (2) /* ISDN A-law */
#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.
*/
typedef struct g726_state_s {
long yl; /* Locked or steady state step size multiplier. */
int yu; /* Unlocked or non-steady state step size multiplier. */
int dms; /* Short term energy estimate. */
int dml; /* Long term energy estimate. */
int ap; /* Linear weighting coefficient of 'yl' and 'yu'. */
int a[2]; /* Coefficients of pole portion of prediction filter. */
int b[6]; /* Coefficients of zero portion of prediction filter. */
int pk[2]; /* Signs of previous two samples of a partially
* reconstructed signal. */
short dq[6];/* int here fails in newupdate on encode!
* Previous 6 samples of the quantized difference
* signal represented in an internal floating point
* format.
*/
int sr[2]; /* Previous 2 samples of the quantized difference
* signal represented in an internal floating point
* format. */
int td; /* delayed tone detect, new in 1988 version */
} g726_state;
/* External function definitions. */
void g726_init_state( g726_state *);
int g726_32_encoder(
int sample,
int in_coding,
g726_state *state_ptr);
int g726_32_decoder(
int code,
int out_coding,
g726_state *state_ptr);
int g726_16_encoder(
int sample,
int in_coding,
g726_state *state_ptr);
int g726_16_decoder(
int code,
int out_coding,
g726_state *state_ptr);
int g726_24_encoder(
int sample,
int in_coding,
g726_state *state_ptr);
int g726_24_decoder(
int code,
int out_coding,
g726_state *state_ptr);
int g726_40_encoder(
int sample,
int in_coding,
g726_state *state_ptr);
int g726_40_decoder(
int code,
int out_coding,
g726_state *state_ptr);
#endif /* !_G72X_H */
+60
View File
@@ -0,0 +1,60 @@
/**
*
* $Log: private.h,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.
*
*
*
*/
#if !defined G726_PRIVATE
#define G726_PRIVATE
int linear2ulaw(int pcm_val);
int ulaw2linear(int u_val);
int linear2alaw(int pcm_val);
int alaw2linear(int u_val);
static int fmult(int an, int srn);
int predictor_zero( g726_state *state_ptr);
int predictor_pole( g726_state *state_ptr);
int step_size( g726_state *state_ptr);
int quantize( int d, /* Raw difference signal sample */
int y, /* Step size multiplier */
int * table, /* quantization table */
int size); /* table size of short integers */
int reconstruct( int sign, /* 0 for non-negative value */
int dqln, /* G.72x codeword */
int y); /* Step size multiplier */
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 */
g726_state *state_ptr); /* coder state pointer */
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,
int * qtab);
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,
int * qtab);
#endif // G726_PRIVATE