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staging: echo: move to drivers/misc/

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The code is clean, there are users of it, so it doesn't belong in
staging anymore, move it to drivers/misc/.

Cc: Steve Underwood <steveu@coppice.org>
Cc: David Rowe <david@rowetel.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
This commit is contained in:
Greg Kroah-Hartman
2014-02-28 14:08:42 -08:00
parent dc93c85235
commit 6e2055a9e5
11 changed files with 2 additions and 8 deletions
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9
drivers/misc/echo/Kconfig Normal file
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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
drivers/misc/echo/Makefile Normal file
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obj-$(CONFIG_ECHO) += echo.o

674
drivers/misc/echo/echo.c Normal file
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/*
* 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.
*/
/*! \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>
#include <linux/module.h>
#include <linux/slab.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 */
/* adapting coeffs using the traditional stochastic descent (N)LMS algorithm */
#ifdef __bfin__
static inline void lms_adapt_bg(struct oslec_state *ec, int clean, int shift)
{
int i;
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; i < n; i++) {
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
static inline int top_bit(unsigned int bits)
{
if (bits == 0)
return -1;
else
return (int)fls((int32_t) bits) - 1;
}
struct oslec_state *oslec_create(int len, int adaption_mode)
{
struct oslec_state *ec;
int i;
const int16_t *history;
ec = kzalloc(sizeof(*ec), GFP_KERNEL);
if (!ec)
return NULL;
ec->taps = len;
ec->log2taps = top_bit(len);
ec->curr_pos = ec->taps - 1;
ec->fir_taps16[0] =
kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
if (!ec->fir_taps16[0])
goto error_oom_0;
ec->fir_taps16[1] =
kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
if (!ec->fir_taps16[1])
goto error_oom_1;
history = fir16_create(&ec->fir_state, ec->fir_taps16[0], ec->taps);
if (!history)
goto error_state;
history = fir16_create(&ec->fir_state_bg, ec->fir_taps16[1], ec->taps);
if (!history)
goto error_state_bg;
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_snap;
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_snap:
fir16_free(&ec->fir_state_bg);
error_state_bg:
fir16_free(&ec->fir_state);
error_state:
kfree(ec->fir_taps16[1]);
error_oom_1:
kfree(ec->fir_taps16[0]);
error_oom_0:
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;
int 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) on 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;
/*
* 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);
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 - 1))) >> 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) &&
/* (ec->Lclean_bg < 0.875*ec->Lclean) */
(8 * ec->lclean_bg < 7 * ec->lclean) &&
/* (ec->Lclean_bg < 0.125*ec->Ltx) */
(8 * ec->lclean_bg < 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
* overflow 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 separated 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;
int tmp1;
if (ec->adaption_mode & ECHO_CAN_USE_TX_HPF) {
tmp = tx << 15;
/*
* 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);
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");

187
drivers/misc/echo/echo.h Normal file
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/*
* 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.
*/
#ifndef __ECHO_H
#define __ECHO_H
/*
Line echo cancellation for voice
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.
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.
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;
int16_t 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;
int lrxacc;
int lcleanacc;
int lclean_bgacc;
int ltx;
int lrx;
int lclean;
int lclean_bg;
int lbgn;
int lbgn_acc;
int lbgn_upper;
int lbgn_upper_acc;
/* foreground and background filter states */
struct fir16_state_t fir_state;
struct fir16_state_t fir_state_bg;
int16_t *fir_taps16[2];
/* DC blocking filter states */
int tx_1;
int tx_2;
int rx_1;
int rx_2;
/* optional High Pass Filter states */
int32_t xvtx[5];
int32_t yvtx[5];
int32_t xvrx[5];
int32_t 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 */

216
drivers/misc/echo/fir.h Normal file
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/*
* 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.
*/
#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.
*/
/*
* 16 bit integer FIR descriptor. This defines the working state for a single
* instance of an FIR filter using 16 bit integer coefficients.
*/
struct fir16_state_t {
int taps;
int curr_pos;
const int16_t *coeffs;
int16_t *history;
};
/*
* 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.
*/
struct fir32_state_t {
int taps;
int curr_pos;
const int32_t *coeffs;
int16_t *history;
};
/*
* Floating point FIR descriptor. This defines the working state for a single
* instance of an FIR filter using floating point coefficients and data.
*/
struct fir_float_state_t {
int taps;
int curr_pos;
const float *coeffs;
float *history;
};
static inline const int16_t *fir16_create(struct fir16_state_t *fir,
const int16_t *coeffs, int taps)
{
fir->taps = taps;
fir->curr_pos = taps - 1;
fir->coeffs = coeffs;
#if 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(struct fir16_state_t *fir)
{
#if defined(__bfin__)
memset(fir->history, 0, 2 * fir->taps * sizeof(int16_t));
#else
memset(fir->history, 0, fir->taps * sizeof(int16_t));
#endif
}
static inline void fir16_free(struct fir16_state_t *fir)
{
kfree(fir->history);
}
#ifdef __bfin__
static inline int32_t dot_asm(short *x, short *y, int len)
{
int dot;
len--;
__asm__("I0 = %1;\n\t"
"I1 = %2;\n\t"
"A0 = 0;\n\t"
"R0.L = W[I0++] || R1.L = W[I1++];\n\t"
"LOOP dot%= LC0 = %3;\n\t"
"LOOP_BEGIN dot%=;\n\t"
"A0 += R0.L * R1.L (IS) || R0.L = W[I0++] || R1.L = W[I1++];\n\t"
"LOOP_END dot%=;\n\t"
"A0 += R0.L*R1.L (IS);\n\t"
"R0 = A0;\n\t"
"%0 = R0;\n\t"
: "=&d"(dot)
: "a"(x), "a"(y), "a"(len)
: "I0", "I1", "A1", "A0", "R0", "R1"
);
return dot;
}
#endif
static inline int16_t fir16(struct fir16_state_t *fir, int16_t sample)
{
int32_t y;
#if defined(__bfin__)
fir->history[fir->curr_pos] = sample;
fir->history[fir->curr_pos + fir->taps] = sample;
y = dot_asm((int16_t *) fir->coeffs, &fir->history[fir->curr_pos],
fir->taps);
#else
int i;
int offset1;
int offset2;
fir->history[fir->curr_pos] = sample;
offset2 = fir->curr_pos;
offset1 = fir->taps - offset2;
y = 0;
for (i = fir->taps - 1; i >= offset1; i--)
y += fir->coeffs[i] * fir->history[i - offset1];
for (; i >= 0; i--)
y += fir->coeffs[i] * fir->history[i + offset2];
#endif
if (fir->curr_pos <= 0)
fir->curr_pos = fir->taps;
fir->curr_pos--;
return (int16_t) (y >> 15);
}
static inline const int16_t *fir32_create(struct fir32_state_t *fir,
const int32_t *coeffs, int taps)
{
fir->taps = taps;
fir->curr_pos = taps - 1;
fir->coeffs = coeffs;
fir->history = kcalloc(taps, sizeof(int16_t), GFP_KERNEL);
return fir->history;
}
static inline void fir32_flush(struct fir32_state_t *fir)
{
memset(fir->history, 0, fir->taps * sizeof(int16_t));
}
static inline void fir32_free(struct fir32_state_t *fir)
{
kfree(fir->history);
}
static inline int16_t fir32(struct fir32_state_t *fir, int16_t sample)
{
int i;
int32_t y;
int offset1;
int offset2;
fir->history[fir->curr_pos] = sample;
offset2 = fir->curr_pos;
offset1 = fir->taps - offset2;
y = 0;
for (i = fir->taps - 1; i >= offset1; i--)
y += fir->coeffs[i] * fir->history[i - offset1];
for (; i >= 0; i--)
y += fir->coeffs[i] * fir->history[i + offset2];
if (fir->curr_pos <= 0)
fir->curr_pos = fir->taps;
fir->curr_pos--;
return (int16_t) (y >> 15);
}
#endif

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/*
* OSLEC - A line echo canceller. This code is being developed
* against and partially complies with G168. Using code from SpanDSP
*
* Written by Steve Underwood <steveu@coppice.org>
* and David Rowe <david_at_rowetel_dot_com>
*
* Copyright (C) 2001 Steve Underwood and 2007-2008 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.
*
*/
#ifndef __OSLEC_H
#define __OSLEC_H
/* Mask bits for the adaption mode */
#define ECHO_CAN_USE_ADAPTION 0x01
#define ECHO_CAN_USE_NLP 0x02
#define ECHO_CAN_USE_CNG 0x04
#define ECHO_CAN_USE_CLIP 0x08
#define ECHO_CAN_USE_TX_HPF 0x10
#define ECHO_CAN_USE_RX_HPF 0x20
#define ECHO_CAN_DISABLE 0x40
/**
* oslec_state: G.168 echo canceller descriptor.
*
* This defines the working state for a line echo canceller.
*/
struct oslec_state;
/**
* oslec_create - Create a voice echo canceller context.
* @len: The length of the canceller, in samples.
* @return: The new canceller context, or NULL if the canceller could not be
* created.
*/
struct oslec_state *oslec_create(int len, int adaption_mode);
/**
* oslec_free - Free a voice echo canceller context.
* @ec: The echo canceller context.
*/
void oslec_free(struct oslec_state *ec);
/**
* oslec_flush - Flush (reinitialise) a voice echo canceller context.
* @ec: The echo canceller context.
*/
void oslec_flush(struct oslec_state *ec);
/**
* oslec_adaption_mode - set the adaption mode of a voice echo canceller context.
* @ec The echo canceller context.
* @adaption_mode: The mode.
*/
void oslec_adaption_mode(struct oslec_state *ec, int adaption_mode);
void oslec_snapshot(struct oslec_state *ec);
/**
* oslec_update: Process a sample through a voice echo canceller.
* @ec: The echo canceller context.
* @tx: The transmitted audio sample.
* @rx: The received audio sample.
*
* The return value is the clean (echo cancelled) received sample.
*/
int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx);
/**
* oslec_hpf_tx: Process to high pass filter the tx signal.
* @ec: The echo canceller context.
* @tx: The transmitted auio sample.
*
* The return value is the HP filtered transmit sample, send this to your D/A.
*/
int16_t oslec_hpf_tx(struct oslec_state *ec, int16_t tx);
#endif /* __OSLEC_H */