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wanpipe-3.5.12.tgz

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Harald Welte 8 months ago
parent
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c6acb46c69
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      .openzap
  2. 2
      .router_version
  3. 38
      ChangeLog.3.5
  4. 23
      Makefile
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      OSLEC/echo/Kconfig
  6. 1
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  7. 10
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  8. 228
      OSLEC/echo/bit_operations.h
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  11. 295
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  12. 281
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      Setup
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  78. 40
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  101. Some files were not shown because too many files have changed in this diff Show More

2
.router_version

@ -1 +1 @@
wanpipe-3.5.11
wanpipe-3.5.12

38
ChangeLog.3.5

@ -3,11 +3,47 @@ WANPIPE TDM VOICE - IP/WAN Package
------------------------------------------------------------------------------
Author: Nenad Corbic <ncorbic@sangoma.com>
Copyright (c) 1995-2009 Sangoma Technologies Inc.
Copyright (c) 1995-2010 Sangoma Technologies Inc.
For more info visit: http://wiki.sangoma.com
------------------------------------------------------------------------------
* Mon Jun 28 2010 Nenad Corbic <ncorbic@sangoma.com> - 3.5.12
===================================================================
- Fixed Dahdi 2.3 Support
- Fixed FreeSwitch Openzap HardHDLC option for AFT cards
- Fixed wanpipemon support for non aft cards.
- Merged USB FXO code from 3.6 release
- USB FXO bug fix for 2.6.32 kernels
- Support for B800 Analog card
- Fixed alarm reporting in DAHDI/ZAPTEL
- Added Extra EC DSP Configuration Options
HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation enabled with nlp (default)
# OCT_SPEECH: improves software tone detection by disabling NLP (echo possible)
# OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions.
HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the line - could break fax
HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acustic echo cancelation
HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software tone detection (possible echo)
HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default)
HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default rx audio level to be maintained (-20 default)
HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal
HWEC_RX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal
- Added AIS BLUE Alarm Maintenance Startup option
Allows a port to be started in BLUE alarm.
TE_AIS_MAINTENANCE = NO #NO: defualt YES: Start port in AIS Blue Alarm and keep line down
#wanpipemon -i w1g1 -c Ttx_ais_off to disable AIS maintenance mode
#wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode
- Fixed Legacy XDLC compile
- Fixed core edge case scenarios where
potential race condition could occour.
* Thu Apr 08 2010 Nenad Corbic <ncorbic@sangoma.com> - 3.5.11
===================================================================

23
Makefile

@ -13,6 +13,7 @@
PWD=$(shell pwd)
KBUILD_VERBOSE=0
CC=gcc
EXTRA_CFLAGS=
EXTRA_FLAGS=
@ -99,7 +100,15 @@ EXTRA_UTIL_FLAGS += -I$(PWD)/patches/kdrivers/wanec -I$(PWD)/patches/kdrivers/wa
ENABLE_WANPIPEMON_ZAP=NO
ZAPHDLC_PRIV=/etc/wanpipe/.zaphdlc
EXTRA_CFLAGS += $(EXTRA_FLAGS)
PRODUCT_DEFINES= -DCONFIG_PRODUCT_WANPIPE_BASE -DCONFIG_PRODUCT_WANPIPE_AFT -DCONFIG_PRODUCT_WANPIPE_AFT_CORE
PRODUCT_DEFINES+= -DCONFIG_PRODUCT_WANPIPE_AFT_TE1 -DCONFIG_PRODUCT_WANPIPE_AFT_TE3 -DCONFIG_PRODUCT_WANPIPE_AFT_56K
PRODUCT_DEFINES+= -DCONFIG_WANPIPE_HWEC -DCONFIG_PRODUCT_WANPIPE_SOCK_DATASCOPE -DCONFIG_PRODUCT_WANPIPE_AFT_BRI -DCONFIG_PRODUCT_WANPIPE_AFT_SERIAL
PRODUCT_DEFINES+= -DCONFIG_PRODUCT_WANPIPE_TDM_VOICE_DCHAN -DCONFIG_PRODUCT_WANPIPE_CODEC_SLINEAR_LAW -DCONFIG_PRODUCT_WANPIPE_AFT_RM
PRODUCT_DEFINES+= -DCONFIG_PRODUCT_WANPIPE_USB -DCONFIG_PRODUCT_WANPIPE_A700 -DCONFIG_PRODUCT_A600 -DCONFIG_PRODUCT_WANPIPE_AFT_A600 -DCONFIG_PRODUCT_WANPIPE_AFT_A700
PRODUCT_DEFINES+= -DCONFIG_PRODUCT_WANPIPE_AFT_B601 -DCONFIG_PRODUCT_WANPIPE_AFT_B800
EXTRA_CFLAGS += $(EXTRA_FLAGS) $(PRODUCT_DEFINES)
EXTRA_UTIL_FLAGS += $(PRODUCT_DEFINES)
#Check if zaptel exists
@ -170,6 +179,9 @@ openzap: all_src all_lib
freetdm: all_src all_lib
@touch .all_lib .openzap
g3ti: all_src all_lib
@touch .all_lib .openzap
openzap_ss7: all_src_ss7 all_lib
@touch .all_lib
@ -370,6 +382,15 @@ install_pri:
@eval "cd ssmg; ./get_sangoma_prid.sh; cd .."
$(MAKE) -C ssmg/sangoma_pri/ install SYSINC=$(PWD)/$(WINCLUDE) CC=$(CC) DESTDIR=$(INSTALLPREFIX)
install_pri_freetdm:
@eval "cd ssmg; ./get_sangoma_prid.sh; cd .."
$(MAKE) -C ssmg/sangoma_pri/ install_libs install_freetdm_lib SYSINC=$(PWD)/$(WINCLUDE) CC=$(CC) DESTDIR=$(INSTALLPREFIX)
install_pri_freetdm_update:
@eval "cd ssmg; ./get_sangoma_prid.sh --update; cd .."
$(MAKE) -C ssmg/sangoma_pri/ install_libs install_freetdm_lib SYSINC=$(PWD)/$(WINCLUDE) CC=$(CC) DESTDIR=$(INSTALLPREFIX)
install_pri_update:
@eval "cd ssmg; ./get_sangoma_prid.sh --update; cd .."
$(MAKE) -C ssmg/sangoma_pri/ install SYSINC=$(PWD)/$(WINCLUDE) CC=$(CC) DESTDIR=$(INSTALLPREFIX)

9
OSLEC/echo/Kconfig

@ -0,0 +1,9 @@
config ECHO
tristate "Line Echo Canceller support"
default n
---help---
This driver provides line echo cancelling support for mISDN and
Zaptel drivers.
To compile this driver as a module, choose M here. The module
will be called echo.

1
OSLEC/echo/Makefile

@ -0,0 +1 @@
obj-$(CONFIG_ECHO) += echo.o

10
OSLEC/echo/TODO

@ -0,0 +1,10 @@
TODO:
- checkpatch.pl cleanups
- Lindent
- typedef removals
- handle bit_operations.h (merge in or make part of common code?)
- remove proc interface, only use echo.h interface (proc interface is
racy and not correct.)
Please send patches to Greg Kroah-Hartman <greg@kroah.com> and Cc: Steve
Underwood <steveu@coppice.org> and David Rowe <david@rowetel.com>

228
OSLEC/echo/bit_operations.h

@ -0,0 +1,228 @@
/*
* SpanDSP - a series of DSP components for telephony
*
* bit_operations.h - Various bit level operations, such as bit reversal
*
* Written by Steve Underwood <steveu@coppice.org>
*
* Copyright (C) 2006 Steve Underwood
*
* All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2, as
* published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* $Id: bit_operations.h,v 1.11 2006/11/28 15:37:03 steveu Exp $
*/
/*! \file */
#if !defined(_BIT_OPERATIONS_H_)
#define _BIT_OPERATIONS_H_
#if defined(__i386__) || defined(__x86_64__)
/*! \brief Find the bit position of the highest set bit in a word
\param bits The word to be searched
\return The bit number of the highest set bit, or -1 if the word is zero. */
static __inline__ int top_bit(unsigned int bits)
{
int res;
__asm__(" xorl %[res],%[res];\n"
" decl %[res];\n"
" bsrl %[bits],%[res]\n"
:[res] "=&r" (res)
:[bits] "rm"(bits)
);
return res;
}
/*! \brief Find the bit position of the lowest set bit in a word
\param bits The word to be searched
\return The bit number of the lowest set bit, or -1 if the word is zero. */
static __inline__ int bottom_bit(unsigned int bits)
{
int res;
__asm__(" xorl %[res],%[res];\n"
" decl %[res];\n"
" bsfl %[bits],%[res]\n"
:[res] "=&r" (res)
:[bits] "rm"(bits)
);
return res;
}
#else
static __inline__ int top_bit(unsigned int bits)
{
int i;
if (bits == 0)
return -1;
i = 0;
if (bits & 0xFFFF0000) {
bits &= 0xFFFF0000;
i += 16;
}
if (bits & 0xFF00FF00) {
bits &= 0xFF00FF00;
i += 8;
}
if (bits & 0xF0F0F0F0) {
bits &= 0xF0F0F0F0;
i += 4;
}
if (bits & 0xCCCCCCCC) {
bits &= 0xCCCCCCCC;
i += 2;
}
if (bits & 0xAAAAAAAA) {
bits &= 0xAAAAAAAA;
i += 1;
}
return i;
}
static __inline__ int bottom_bit(unsigned int bits)
{
int i;
if (bits == 0)
return -1;
i = 32;
if (bits & 0x0000FFFF) {
bits &= 0x0000FFFF;
i -= 16;
}
if (bits & 0x00FF00FF) {
bits &= 0x00FF00FF;
i -= 8;
}
if (bits & 0x0F0F0F0F) {
bits &= 0x0F0F0F0F;
i -= 4;
}
if (bits & 0x33333333) {
bits &= 0x33333333;
i -= 2;
}
if (bits & 0x55555555) {
bits &= 0x55555555;
i -= 1;
}
return i;
}
#endif
/*! \brief Bit reverse a byte.
\param data The byte to be reversed.
\return The bit reversed version of data. */
static __inline__ uint8_t bit_reverse8(uint8_t x)
{
#if defined(__i386__) || defined(__x86_64__)
/* If multiply is fast */
return ((x * 0x0802U & 0x22110U) | (x * 0x8020U & 0x88440U)) *
0x10101U >> 16;
#else
/* If multiply is slow, but we have a barrel shifter */
x = (x >> 4) | (x << 4);
x = ((x & 0xCC) >> 2) | ((x & 0x33) << 2);
return ((x & 0xAA) >> 1) | ((x & 0x55) << 1);
#endif
}
/*! \brief Bit reverse a 16 bit word.
\param data The word to be reversed.
\return The bit reversed version of data. */
uint16_t bit_reverse16(uint16_t data);
/*! \brief Bit reverse a 32 bit word.
\param data The word to be reversed.
\return The bit reversed version of data. */
uint32_t bit_reverse32(uint32_t data);
/*! \brief Bit reverse each of the four bytes in a 32 bit word.
\param data The word to be reversed.
\return The bit reversed version of data. */
uint32_t bit_reverse_4bytes(uint32_t data);
/*! \brief Find the number of set bits in a 32 bit word.
\param x The word to be searched.
\return The number of set bits. */
int one_bits32(uint32_t x);
/*! \brief Create a mask as wide as the number in a 32 bit word.
\param x The word to be searched.
\return The mask. */
uint32_t make_mask32(uint32_t x);
/*! \brief Create a mask as wide as the number in a 16 bit word.
\param x The word to be searched.
\return The mask. */
uint16_t make_mask16(uint16_t x);
/*! \brief Find the least significant one in a word, and return a word
with just that bit set.
\param x The word to be searched.
\return The word with the single set bit. */
static __inline__ uint32_t least_significant_one32(uint32_t x)
{
return (x & (-(int32_t) x));
}
/*! \brief Find the most significant one in a word, and return a word
with just that bit set.
\param x The word to be searched.
\return The word with the single set bit. */
static __inline__ uint32_t most_significant_one32(uint32_t x)
{
#if defined(__i386__) || defined(__x86_64__)
return 1 << top_bit(x);
#else
x = make_mask32(x);
return (x ^ (x >> 1));
#endif
}
/*! \brief Find the parity of a byte.
\param x The byte to be checked.
\return 1 for odd, or 0 for even. */
static __inline__ int parity8(uint8_t x)
{
x = (x ^ (x >> 4)) & 0x0F;
return (0x6996 >> x) & 1;
}
/*! \brief Find the parity of a 16 bit word.
\param x The word to be checked.
\return 1 for odd, or 0 for even. */
static __inline__ int parity16(uint16_t x)
{
x ^= (x >> 8);
x = (x ^ (x >> 4)) & 0x0F;
return (0x6996 >> x) & 1;
}
/*! \brief Find the parity of a 32 bit word.
\param x The word to be checked.
\return 1 for odd, or 0 for even. */
static __inline__ int parity32(uint32_t x)
{
x ^= (x >> 16);
x ^= (x >> 8);
x = (x ^ (x >> 4)) & 0x0F;
return (0x6996 >> x) & 1;
}
#endif
/*- End of file ------------------------------------------------------------*/

638
OSLEC/echo/echo.c

@ -0,0 +1,638 @@
/*
* SpanDSP - a series of DSP components for telephony
*
* echo.c - A line echo canceller. This code is being developed
* against and partially complies with G168.
*
* Written by Steve Underwood <steveu@coppice.org>
* and David Rowe <david_at_rowetel_dot_com>
*
* Copyright (C) 2001, 2003 Steve Underwood, 2007 David Rowe
*
* Based on a bit from here, a bit from there, eye of toad, ear of
* bat, 15 years of failed attempts by David and a few fried brain
* cells.
*
* All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2, as
* published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* $Id: echo.c,v 1.20 2006/12/01 18:00:48 steveu Exp $
*/
/*! \file */
/* Implementation Notes
David Rowe
April 2007
This code started life as Steve's NLMS algorithm with a tap
rotation algorithm to handle divergence during double talk. I
added a Geigel Double Talk Detector (DTD) [2] and performed some
G168 tests. However I had trouble meeting the G168 requirements,
especially for double talk - there were always cases where my DTD
failed, for example where near end speech was under the 6dB
threshold required for declaring double talk.
So I tried a two path algorithm [1], which has so far given better
results. The original tap rotation/Geigel algorithm is available
in SVN http://svn.rowetel.com/software/oslec/tags/before_16bit.
It's probably possible to make it work if some one wants to put some
serious work into it.
At present no special treatment is provided for tones, which
generally cause NLMS algorithms to diverge. Initial runs of a
subset of the G168 tests for tones (e.g ./echo_test 6) show the
current algorithm is passing OK, which is kind of surprising. The
full set of tests needs to be performed to confirm this result.
One other interesting change is that I have managed to get the NLMS
code to work with 16 bit coefficients, rather than the original 32
bit coefficents. This reduces the MIPs and storage required.
I evaulated the 16 bit port using g168_tests.sh and listening tests
on 4 real-world samples.
I also attempted the implementation of a block based NLMS update
[2] but although this passes g168_tests.sh it didn't converge well
on the real-world samples. I have no idea why, perhaps a scaling
problem. The block based code is also available in SVN
http://svn.rowetel.com/software/oslec/tags/before_16bit. If this
code can be debugged, it will lead to further reduction in MIPS, as
the block update code maps nicely onto DSP instruction sets (it's a
dot product) compared to the current sample-by-sample update.
Steve also has some nice notes on echo cancellers in echo.h
References:
[1] Ochiai, Areseki, and Ogihara, "Echo Canceller with Two Echo
Path Models", IEEE Transactions on communications, COM-25,
No. 6, June
1977.
http://www.rowetel.com/images/echo/dual_path_paper.pdf
[2] The classic, very useful paper that tells you how to
actually build a real world echo canceller:
Messerschmitt, Hedberg, Cole, Haoui, Winship, "Digital Voice
Echo Canceller with a TMS320020,
http://www.rowetel.com/images/echo/spra129.pdf
[3] I have written a series of blog posts on this work, here is
Part 1: http://www.rowetel.com/blog/?p=18
[4] The source code http://svn.rowetel.com/software/oslec/
[5] A nice reference on LMS filters:
http://en.wikipedia.org/wiki/Least_mean_squares_filter
Credits:
Thanks to Steve Underwood, Jean-Marc Valin, and Ramakrishnan
Muthukrishnan for their suggestions and email discussions. Thanks
also to those people who collected echo samples for me such as
Mark, Pawel, and Pavel.
*/
#include <linux/kernel.h> /* We're doing kernel work */
#include <linux/module.h>
#include <linux/slab.h>
#include "bit_operations.h"
#include "echo.h"
#define MIN_TX_POWER_FOR_ADAPTION 64
#define MIN_RX_POWER_FOR_ADAPTION 64
#define DTD_HANGOVER 600 /* 600 samples, or 75ms */
#define DC_LOG2BETA 3 /* log2() of DC filter Beta */
/*-----------------------------------------------------------------------*\
FUNCTIONS
\*-----------------------------------------------------------------------*/
/* adapting coeffs using the traditional stochastic descent (N)LMS algorithm */
#ifdef __bfin__
static void __inline__ lms_adapt_bg(struct oslec_state *ec, int clean,
int shift)
{
int i, j;
int offset1;
int offset2;
int factor;
int exp;
int16_t *phist;
int n;
if (shift > 0)
factor = clean << shift;
else
factor = clean >> -shift;
/* Update the FIR taps */
offset2 = ec->curr_pos;
offset1 = ec->taps - offset2;
phist = &ec->fir_state_bg.history[offset2];
/* st: and en: help us locate the assembler in echo.s */
//asm("st:");
n = ec->taps;
for (i = 0, j = offset2; i < n; i++, j++) {
exp = *phist++ * factor;
ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
}
//asm("en:");
/* Note the asm for the inner loop above generated by Blackfin gcc
4.1.1 is pretty good (note even parallel instructions used):
R0 = W [P0++] (X);
R0 *= R2;
R0 = R0 + R3 (NS) ||
R1 = W [P1] (X) ||
nop;
R0 >>>= 15;
R0 = R0 + R1;
W [P1++] = R0;
A block based update algorithm would be much faster but the
above can't be improved on much. Every instruction saved in
the loop above is 2 MIPs/ch! The for loop above is where the
Blackfin spends most of it's time - about 17 MIPs/ch measured
with speedtest.c with 256 taps (32ms). Write-back and
Write-through cache gave about the same performance.
*/
}
/*
IDEAS for further optimisation of lms_adapt_bg():
1/ The rounding is quite costly. Could we keep as 32 bit coeffs
then make filter pluck the MS 16-bits of the coeffs when filtering?
However this would lower potential optimisation of filter, as I
think the dual-MAC architecture requires packed 16 bit coeffs.
2/ Block based update would be more efficient, as per comments above,
could use dual MAC architecture.
3/ Look for same sample Blackfin LMS code, see if we can get dual-MAC
packing.
4/ Execute the whole e/c in a block of say 20ms rather than sample
by sample. Processing a few samples every ms is inefficient.
*/
#else
static __inline__ void lms_adapt_bg(struct oslec_state *ec, int clean,
int shift)
{
int i;
int offset1;
int offset2;
int factor;
int exp;
if (shift > 0)
factor = clean << shift;
else
factor = clean >> -shift;
/* Update the FIR taps */
offset2 = ec->curr_pos;
offset1 = ec->taps - offset2;
for (i = ec->taps - 1; i >= offset1; i--) {
exp = (ec->fir_state_bg.history[i - offset1] * factor);
ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
}
for (; i >= 0; i--) {
exp = (ec->fir_state_bg.history[i + offset2] * factor);
ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
}
}
#endif
struct oslec_state *oslec_create(int len, int adaption_mode)
{
struct oslec_state *ec;
int i;
ec = kzalloc(sizeof(*ec), GFP_KERNEL);
if (!ec)
return NULL;
ec->taps = len;
ec->log2taps = top_bit(len);
ec->curr_pos = ec->taps - 1;
for (i = 0; i < 2; i++) {
ec->fir_taps16[i] =
kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
if (!ec->fir_taps16[i])
goto error_oom;
}
fir16_create(&ec->fir_state, ec->fir_taps16[0], ec->taps);
fir16_create(&ec->fir_state_bg, ec->fir_taps16[1], ec->taps);
for (i = 0; i < 5; i++) {
ec->xvtx[i] = ec->yvtx[i] = ec->xvrx[i] = ec->yvrx[i] = 0;
}
ec->cng_level = 1000;
oslec_adaption_mode(ec, adaption_mode);
ec->snapshot = kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
if (!ec->snapshot)
goto error_oom;
ec->cond_met = 0;
ec->Pstates = 0;
ec->Ltxacc = ec->Lrxacc = ec->Lcleanacc = ec->Lclean_bgacc = 0;
ec->Ltx = ec->Lrx = ec->Lclean = ec->Lclean_bg = 0;
ec->tx_1 = ec->tx_2 = ec->rx_1 = ec->rx_2 = 0;
ec->Lbgn = ec->Lbgn_acc = 0;
ec->Lbgn_upper = 200;
ec->Lbgn_upper_acc = ec->Lbgn_upper << 13;
return ec;
error_oom:
for (i = 0; i < 2; i++)
kfree(ec->fir_taps16[i]);
kfree(ec);
return NULL;
}
EXPORT_SYMBOL_GPL(oslec_create);
void oslec_free(struct oslec_state *ec)
{
int i;
fir16_free(&ec->fir_state);
fir16_free(&ec->fir_state_bg);
for (i = 0; i < 2; i++)
kfree(ec->fir_taps16[i]);
kfree(ec->snapshot);
kfree(ec);
}
EXPORT_SYMBOL_GPL(oslec_free);
void oslec_adaption_mode(struct oslec_state *ec, int adaption_mode)
{
ec->adaption_mode = adaption_mode;
}
EXPORT_SYMBOL_GPL(oslec_adaption_mode);
void oslec_flush(struct oslec_state *ec)
{
int i;
ec->Ltxacc = ec->Lrxacc = ec->Lcleanacc = ec->Lclean_bgacc = 0;
ec->Ltx = ec->Lrx = ec->Lclean = ec->Lclean_bg = 0;
ec->tx_1 = ec->tx_2 = ec->rx_1 = ec->rx_2 = 0;
ec->Lbgn = ec->Lbgn_acc = 0;
ec->Lbgn_upper = 200;
ec->Lbgn_upper_acc = ec->Lbgn_upper << 13;
ec->nonupdate_dwell = 0;
fir16_flush(&ec->fir_state);
fir16_flush(&ec->fir_state_bg);
ec->fir_state.curr_pos = ec->taps - 1;
ec->fir_state_bg.curr_pos = ec->taps - 1;
for (i = 0; i < 2; i++)
memset(ec->fir_taps16[i], 0, ec->taps * sizeof(int16_t));
ec->curr_pos = ec->taps - 1;
ec->Pstates = 0;
}
EXPORT_SYMBOL_GPL(oslec_flush);
void oslec_snapshot(struct oslec_state *ec)
{
memcpy(ec->snapshot, ec->fir_taps16[0], ec->taps * sizeof(int16_t));
}
EXPORT_SYMBOL_GPL(oslec_snapshot);
/* Dual Path Echo Canceller ------------------------------------------------*/
int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
{
int32_t echo_value;
int clean_bg;
int tmp, tmp1;
/* Input scaling was found be required to prevent problems when tx
starts clipping. Another possible way to handle this would be the
filter coefficent scaling. */
ec->tx = tx;
ec->rx = rx;
tx >>= 1;
rx >>= 1;
/*
Filter DC, 3dB point is 160Hz (I think), note 32 bit precision required
otherwise values do not track down to 0. Zero at DC, Pole at (1-Beta)
only real axis. Some chip sets (like Si labs) don't need
this, but something like a $10 X100P card does. Any DC really slows
down convergence.
Note: removes some low frequency from the signal, this reduces
the speech quality when listening to samples through headphones
but may not be obvious through a telephone handset.
Note that the 3dB frequency in radians is approx Beta, e.g. for
Beta = 2^(-3) = 0.125, 3dB freq is 0.125 rads = 159Hz.
*/
if (ec->adaption_mode & ECHO_CAN_USE_RX_HPF) {
tmp = rx << 15;
#if 1
/* Make sure the gain of the HPF is 1.0. This can still saturate a little under
impulse conditions, and it might roll to 32768 and need clipping on sustained peak
level signals. However, the scale of such clipping is small, and the error due to
any saturation should not markedly affect the downstream processing. */
tmp -= (tmp >> 4);
#endif
ec->rx_1 += -(ec->rx_1 >> DC_LOG2BETA) + tmp - ec->rx_2;
/* hard limit filter to prevent clipping. Note that at this stage
rx should be limited to +/- 16383 due to right shift above */
tmp1 = ec->rx_1 >> 15;
if (tmp1 > 16383)
tmp1 = 16383;
if (tmp1 < -16383)
tmp1 = -16383;
rx = tmp1;
ec->rx_2 = tmp;
}
/* Block average of power in the filter states. Used for
adaption power calculation. */
{
int new, old;
/* efficient "out with the old and in with the new" algorithm so
we don't have to recalculate over the whole block of
samples. */
new = (int)tx *(int)tx;
old = (int)ec->fir_state.history[ec->fir_state.curr_pos] *
(int)ec->fir_state.history[ec->fir_state.curr_pos];
ec->Pstates +=
((new - old) + (1 << ec->log2taps)) >> ec->log2taps;
if (ec->Pstates < 0)
ec->Pstates = 0;
}
/* Calculate short term average levels using simple single pole IIRs */
ec->Ltxacc += abs(tx) - ec->Ltx;
ec->Ltx = (ec->Ltxacc + (1 << 4)) >> 5;
ec->Lrxacc += abs(rx) - ec->Lrx;
ec->Lrx = (ec->Lrxacc + (1 << 4)) >> 5;
/* Foreground filter --------------------------------------------------- */
ec->fir_state.coeffs = ec->fir_taps16[0];
echo_value = fir16(&ec->fir_state, tx);
ec->clean = rx - echo_value;
ec->Lcleanacc += abs(ec->clean) - ec->Lclean;
ec->Lclean = (ec->Lcleanacc + (1 << 4)) >> 5;
/* Background filter --------------------------------------------------- */
echo_value = fir16(&ec->fir_state_bg, tx);
clean_bg = rx - echo_value;
ec->Lclean_bgacc += abs(clean_bg) - ec->Lclean_bg;
ec->Lclean_bg = (ec->Lclean_bgacc + (1 << 4)) >> 5;
/* Background Filter adaption ----------------------------------------- */
/* Almost always adap bg filter, just simple DT and energy
detection to minimise adaption in cases of strong double talk.
However this is not critical for the dual path algorithm.
*/
ec->factor = 0;
ec->shift = 0;
if ((ec->nonupdate_dwell == 0)) {
int P, logP, shift;
/* Determine:
f = Beta * clean_bg_rx/P ------ (1)
where P is the total power in the filter states.
The Boffins have shown that if we obey (1) we converge
quickly and avoid instability.
The correct factor f must be in Q30, as this is the fixed
point format required by the lms_adapt_bg() function,
therefore the scaled version of (1) is:
(2^30) * f = (2^30) * Beta * clean_bg_rx/P
factor = (2^30) * Beta * clean_bg_rx/P ----- (2)
We have chosen Beta = 0.25 by experiment, so:
factor = (2^30) * (2^-2) * clean_bg_rx/P
(30 - 2 - log2(P))
factor = clean_bg_rx 2 ----- (3)
To avoid a divide we approximate log2(P) as top_bit(P),
which returns the position of the highest non-zero bit in
P. This approximation introduces an error as large as a
factor of 2, but the algorithm seems to handle it OK.
Come to think of it a divide may not be a big deal on a
modern DSP, so its probably worth checking out the cycles
for a divide versus a top_bit() implementation.
*/
P = MIN_TX_POWER_FOR_ADAPTION + ec->Pstates;
logP = top_bit(P) + ec->log2taps;
shift = 30 - 2 - logP;
ec->shift = shift;
lms_adapt_bg(ec, clean_bg, shift);
}
/* very simple DTD to make sure we dont try and adapt with strong
near end speech */
ec->adapt = 0;
if ((ec->Lrx > MIN_RX_POWER_FOR_ADAPTION) && (ec->Lrx > ec->Ltx))
ec->nonupdate_dwell = DTD_HANGOVER;
if (ec->nonupdate_dwell)
ec->nonupdate_dwell--;
/* Transfer logic ------------------------------------------------------ */
/* These conditions are from the dual path paper [1], I messed with
them a bit to improve performance. */
if ((ec->adaption_mode & ECHO_CAN_USE_ADAPTION) &&
(ec->nonupdate_dwell == 0) &&
(8 * ec->Lclean_bg <
7 * ec->Lclean) /* (ec->Lclean_bg < 0.875*ec->Lclean) */ &&
(8 * ec->Lclean_bg <
ec->Ltx) /* (ec->Lclean_bg < 0.125*ec->Ltx) */ ) {
if (ec->cond_met == 6) {
/* BG filter has had better results for 6 consecutive samples */
ec->adapt = 1;
memcpy(ec->fir_taps16[0], ec->fir_taps16[1],
ec->taps * sizeof(int16_t));
} else
ec->cond_met++;
} else
ec->cond_met = 0;
/* Non-Linear Processing --------------------------------------------------- */
ec->clean_nlp = ec->clean;
if (ec->adaption_mode & ECHO_CAN_USE_NLP) {
/* Non-linear processor - a fancy way to say "zap small signals, to avoid
residual echo due to (uLaw/ALaw) non-linearity in the channel.". */
if ((16 * ec->Lclean < ec->Ltx)) {
/* Our e/c has improved echo by at least 24 dB (each factor of 2 is 6dB,
so 2*2*2*2=16 is the same as 6+6+6+6=24dB) */
if (ec->adaption_mode & ECHO_CAN_USE_CNG) {
ec->cng_level = ec->Lbgn;
/* Very elementary comfort noise generation. Just random
numbers rolled off very vaguely Hoth-like. DR: This
noise doesn't sound quite right to me - I suspect there
are some overlfow issues in the filtering as it's too
"crackly". TODO: debug this, maybe just play noise at
high level or look at spectrum.
*/
ec->cng_rndnum =
1664525U * ec->cng_rndnum + 1013904223U;
ec->cng_filter =
((ec->cng_rndnum & 0xFFFF) - 32768 +
5 * ec->cng_filter) >> 3;
ec->clean_nlp =
(ec->cng_filter * ec->cng_level * 8) >> 14;
} else if (ec->adaption_mode & ECHO_CAN_USE_CLIP) {
/* This sounds much better than CNG */
if (ec->clean_nlp > ec->Lbgn)
ec->clean_nlp = ec->Lbgn;
if (ec->clean_nlp < -ec->Lbgn)
ec->clean_nlp = -ec->Lbgn;
} else {
/* just mute the residual, doesn't sound very good, used mainly
in G168 tests */
ec->clean_nlp = 0;
}
} else {
/* Background noise estimator. I tried a few algorithms
here without much luck. This very simple one seems to
work best, we just average the level using a slow (1 sec
time const) filter if the current level is less than a
(experimentally derived) constant. This means we dont
include high level signals like near end speech. When
combined with CNG or especially CLIP seems to work OK.
*/
if (ec->Lclean < 40) {
ec->Lbgn_acc += abs(ec->clean) - ec->Lbgn;
ec->Lbgn = (ec->Lbgn_acc + (1 << 11)) >> 12;
}
}
}
/* Roll around the taps buffer */
if (ec->curr_pos <= 0)
ec->curr_pos = ec->taps;
ec->curr_pos--;
if (ec->adaption_mode & ECHO_CAN_DISABLE)
ec->clean_nlp = rx;
/* Output scaled back up again to match input scaling */
return (int16_t) ec->clean_nlp << 1;
}
EXPORT_SYMBOL_GPL(oslec_update);
/* This function is seperated from the echo canceller is it is usually called
as part of the tx process. See rx HP (DC blocking) filter above, it's
the same design.
Some soft phones send speech signals with a lot of low frequency
energy, e.g. down to 20Hz. This can make the hybrid non-linear
which causes the echo canceller to fall over. This filter can help
by removing any low frequency before it gets to the tx port of the
hybrid.
It can also help by removing and DC in the tx signal. DC is bad
for LMS algorithms.
This is one of the classic DC removal filters, adjusted to provide sufficient
bass rolloff to meet the above requirement to protect hybrids from things that
upset them. The difference between successive samples produces a lousy HPF, and
then a suitably placed pole flattens things out. The final result is a nicely
rolled off bass end. The filtering is implemented with extended fractional
precision, which noise shapes things, giving very clean DC removal.
*/
int16_t oslec_hpf_tx(struct oslec_state * ec, int16_t tx)
{
int tmp, tmp1;
if (ec->adaption_mode & ECHO_CAN_USE_TX_HPF) {
tmp = tx << 15;
#if 1
/* Make sure the gain of the HPF is 1.0. The first can still saturate a little under
impulse conditions, and it might roll to 32768 and need clipping on sustained peak
level signals. However, the scale of such clipping is small, and the error due to
any saturation should not markedly affect the downstream processing. */
tmp -= (tmp >> 4);
#endif
ec->tx_1 += -(ec->tx_1 >> DC_LOG2BETA) + tmp - ec->tx_2;
tmp1 = ec->tx_1 >> 15;
if (tmp1 > 32767)
tmp1 = 32767;
if (tmp1 < -32767)
tmp1 = -32767;
tx = tmp1;
ec->tx_2 = tmp;
}
return tx;
}
EXPORT_SYMBOL_GPL(oslec_hpf_tx);
MODULE_LICENSE("GPL");
MODULE_AUTHOR("David Rowe");
MODULE_DESCRIPTION("Open Source Line Echo Canceller");
MODULE_VERSION("0.3.0");

172
OSLEC/echo/echo.h

@ -0,0 +1,172 @@
/*
* SpanDSP - a series of DSP components for telephony
*
* echo.c - A line echo canceller. This code is being developed
* against and partially complies with G168.
*
* Written by Steve Underwood <steveu@coppice.org>
* and David Rowe <david_at_rowetel_dot_com>
*
* Copyright (C) 2001 Steve Underwood and 2007 David Rowe
*
* All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2, as
* published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* $Id: echo.h,v 1.9 2006/10/24 13:45:28 steveu Exp $
*/
#ifndef __ECHO_H
#define __ECHO_H
/*! \page echo_can_page Line echo cancellation for voice
\section echo_can_page_sec_1 What does it do?
This module aims to provide G.168-2002 compliant echo cancellation, to remove
electrical echoes (e.g. from 2-4 wire hybrids) from voice calls.
\section echo_can_page_sec_2 How does it work?
The heart of the echo cancellor is FIR filter. This is adapted to match the
echo impulse response of the telephone line. It must be long enough to
adequately cover the duration of that impulse response. The signal transmitted
to the telephone line is passed through the FIR filter. Once the FIR is
properly adapted, the resulting output is an estimate of the echo signal
received from the line. This is subtracted from the received signal. The result
is an estimate of the signal which originated at the far end of the line, free
from echos of our own transmitted signal.
The least mean squares (LMS) algorithm is attributed to Widrow and Hoff, and
was introduced in 1960. It is the commonest form of filter adaption used in
things like modem line equalisers and line echo cancellers. There it works very
well. However, it only works well for signals of constant amplitude. It works
very poorly for things like speech echo cancellation, where the signal level
varies widely. This is quite easy to fix. If the signal level is normalised -
similar to applying AGC - LMS can work as well for a signal of varying
amplitude as it does for a modem signal. This normalised least mean squares
(NLMS) algorithm is the commonest one used for speech echo cancellation. Many
other algorithms exist - e.g. RLS (essentially the same as Kalman filtering),
FAP, etc. Some perform significantly better than NLMS. However, factors such
as computational complexity and patents favour the use of NLMS.
A simple refinement to NLMS can improve its performance with speech. NLMS tends
to adapt best to the strongest parts of a signal. If the signal is white noise,
the NLMS algorithm works very well. However, speech has more low frequency than
high frequency content. Pre-whitening (i.e. filtering the signal to flatten its
spectrum) the echo signal improves the adapt rate for speech, and ensures the
final residual signal is not heavily biased towards high frequencies. A very
low complexity filter is adequate for this, so pre-whitening adds little to the
compute requirements of the echo canceller.
An FIR filter adapted using pre-whitened NLMS performs well, provided certain
conditions are met:
- The transmitted signal has poor self-correlation.
- There is no signal being generated within the environment being
cancelled.
The difficulty is that neither of these can be guaranteed.
If the adaption is performed while transmitting noise (or something fairly
noise like, such as voice) the adaption works very well. If the adaption is
performed while transmitting something highly correlative (typically narrow
band energy such as signalling tones or DTMF), the adaption can go seriously
wrong. The reason is there is only one solution for the adaption on a near
random signal - the impulse response of the line. For a repetitive signal,
there are any number of solutions which converge the adaption, and nothing
guides the adaption to choose the generalised one. Allowing an untrained
canceller to converge on this kind of narrowband energy probably a good thing,
since at least it cancels the tones. Allowing a well converged canceller to
continue converging on such energy is just a way to ruin its generalised
adaption. A narrowband detector is needed, so adapation can be suspended at
appropriate times.
The adaption process is based on trying to eliminate the received signal. When
there is any signal from within the environment being cancelled it may upset
the adaption process. Similarly, if the signal we are transmitting is small,
noise may dominate and disturb the adaption process. If we can ensure that the
adaption is only performed when we are transmitting a significant signal level,
and the environment is not, things will be OK. Clearly, it is easy to tell when
we are sending a significant signal. Telling, if the environment is generating
a significant signal, and doing it with sufficient speed that the adaption will
not have diverged too much more we stop it, is a little harder.
The key problem in detecting when the environment is sourcing significant
energy is that we must do this very quickly. Given a reasonably long sample of
the received signal, there are a number of strategies which may be used to
assess whether that signal contains a strong far end component. However, by the
time that assessment is complete the far end signal will have already caused
major mis-convergence in the adaption process. An assessment algorithm is
needed which produces a fairly accurate result from a very short burst of far
end energy.
\section echo_can_page_sec_3 How do I use it?
The echo cancellor processes both the transmit and receive streams sample by
sample. The processing function is not declared inline. Unfortunately,
cancellation requires many operations per sample, so the call overhead is only
a minor burden.
*/
#include "fir.h"
#include "oslec.h"
/*!
G.168 echo canceller descriptor. This defines the working state for a line
echo canceller.
*/
struct oslec_state {
int16_t tx, rx;
int16_t clean;
int16_t clean_nlp;
int nonupdate_dwell;
int curr_pos;
int taps;
int log2taps;
int adaption_mode;
int cond_met;
int32_t Pstates;
int16_t adapt;
int32_t factor;
int16_t shift;
/* Average levels and averaging filter states */
int Ltxacc, Lrxacc, Lcleanacc, Lclean_bgacc;
int Ltx, Lrx;
int Lclean;
int Lclean_bg;
int Lbgn, Lbgn_acc, Lbgn_upper, Lbgn_upper_acc;
/* foreground and background filter states */
fir16_state_t fir_state;
fir16_state_t fir_state_bg;
int16_t *fir_taps16[2];
/* DC blocking filter states */
int tx_1, tx_2, rx_1, rx_2;
/* optional High Pass Filter states */
int32_t xvtx[5], yvtx[5];
int32_t xvrx[5], yvrx[5];
/* Parameters for the optional Hoth noise generator */
int cng_level;
int cng_rndnum;
int cng_filter;
/* snapshot sample of coeffs used for development */
int16_t *snapshot;
};
#endif /* __ECHO_H */

295
OSLEC/echo/fir.h

@ -0,0 +1,295 @@
/*
* SpanDSP - a series of DSP components for telephony
*
* fir.h - General telephony FIR routines
*
* Written by Steve Underwood <steveu@coppice.org>
*
* Copyright (C) 2002 Steve Underwood
*
* All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2, as
* published by the Free Software Foundation.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* $Id: fir.h,v 1.8 2006/10/24 13:45:28 steveu Exp $
*/
/*! \page fir_page FIR filtering
\section fir_page_sec_1 What does it do?
???.
\section fir_page_sec_2 How does it work?
???.
*/
#if !defined(_FIR_H_)
#define _FIR_H_
/*
Blackfin NOTES & IDEAS:
A simple dot product function is used to implement the filter. This performs
just one MAC/cycle which is inefficient but was easy to implement as a first
pass. The current Blackfin code also uses an unrolled form of the filter
history to avoid 0 length hardware loop issues. This is wasteful of
memory.
Ideas for improvement:
1/ Rewrite filter for dual MAC inner loop. The issue here is handling
history sample offsets that are 16 bit aligned - the dual MAC needs
32 bit aligmnent. There are some good examples in libbfdsp.
2/ Use the hardware circular buffer facility tohalve memory usage.
3/ Consider using internal memory.
Using less memory might also improve speed as cache misses will be
reduced. A drop in MIPs and memory approaching 50% should be
possible.
The foreground and background filters currenlty use a total of
about 10 MIPs/ch as measured with speedtest.c on a 256 TAP echo
can.
*/
#if defined(USE_MMX) || defined(USE_SSE2)
#include "mmx.h"
#endif
/*!
16 bit integer FIR descriptor. This defines the working state for a single
instance of an FIR filter using 16 bit integer coefficients.
*/
typedef struct {
int taps;
int curr_pos;
const int16_t *coeffs;
int16_t *history;
} fir16_state_t;
/*!
32 bit integer FIR descriptor. This defines the working state for a single
instance of an FIR filter using 32 bit integer coefficients, and filtering
16 bit integer data.
*/
typedef struct {
int taps;
int curr_pos;
const int32_t *coeffs;
int16_t *history;
} fir32_state_t;
/*!
Floating point FIR descriptor. This defines the working state for a single
instance of an FIR filter using floating point coefficients and data.
*/
typedef struct {
int taps;
int curr_pos;
const float *coeffs;
float *history;
} fir_float_state_t;
static __inline__ const int16_t *fir16_create(fir16_state_t * fir,
const int16_t * coeffs, int taps)
{
fir->taps = taps;
fir->curr_pos = taps - 1;
fir->coeffs = coeffs;
#if defined(USE_MMX) || defined(USE_SSE2) || defined(__bfin__)
fir->history = kcalloc(2 * taps, sizeof(int16_t), GFP_KERNEL);
#else
fir->history = kcalloc(taps, sizeof(int16_t), GFP_KERNEL);
#endif
return fir->history;
}
static __inline__ void fir16_flush(fir16_state_t * fir)
{
#if defined(USE_MMX) || defined(USE_SSE2) || defined(__bfin__)
memset(fir->history, 0, 2 * fir->taps * sizeof(int16_t));
#else