Add the rt linux 4.1.3-rt3 as base

Import the rt linux 4.1.3-rt3 as OPNFV kvm base.

It's from git://git.kernel.org/pub/scm/linux/kernel/git/rt/linux-rt-devel.git linux-4.1.y-rt and
the base is:

commit 0917f823c59692d751951bf5ea699a2d1e2f26a2
Author: Sebastian Andrzej Siewior <bigeasy@linutronix.de>
Date:   Sat Jul 25 12:13:34 2015 +0200

    Prepare v4.1.3-rt3

    Signed-off-by: Sebastian Andrzej Siewior <bigeasy@linutronix.de>

We lose all the git history this way and it's not good. We
should apply another opnfv project repo in future.

Change-Id: I87543d81c9df70d99c5001fbdf646b202c19f423
Signed-off-by: Yunhong Jiang <yunhong.jiang@intel.com>

1 files changed, 674 insertions, 0 deletions

diff --git a/kernel/drivers/misc/echo/echo.c b/kernel/drivers/misc/echo/echo.c
new file mode 100644
index 000000000..9597e9523
--- /dev/null
+++ b/kernel/drivers/misc/echo/echo.c
@@ -0,0 +1,674 @@
+/*
+ * 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");