--- /dev/null
+/*
+ * 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");
Code Review / kvmfornfv.git / blobdiff