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  1/*
  2 * SpanDSP - a series of DSP components for telephony
  3 *
  4 * echo.c - A line echo canceller.  This code is being developed
  5 *          against and partially complies with G168.
  6 *
  7 * Written by Steve Underwood <steveu@coppice.org>
  8 *         and David Rowe <david_at_rowetel_dot_com>
  9 *
 10 * Copyright (C) 2001, 2003 Steve Underwood, 2007 David Rowe
 11 *
 12 * Based on a bit from here, a bit from there, eye of toad, ear of
 13 * bat, 15 years of failed attempts by David and a few fried brain
 14 * cells.
 15 *
 16 * All rights reserved.
 17 *
 18 * This program is free software; you can redistribute it and/or modify
 19 * it under the terms of the GNU General Public License version 2, as
 20 * published by the Free Software Foundation.
 21 *
 22 * This program is distributed in the hope that it will be useful,
 23 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 24 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 25 * GNU General Public License for more details.
 26 *
 27 * You should have received a copy of the GNU General Public License
 28 * along with this program; if not, write to the Free Software
 29 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
 30 */
 31
 32/*! \file */
 33
 34/* Implementation Notes
 35   David Rowe
 36   April 2007
 37
 38   This code started life as Steve's NLMS algorithm with a tap
 39   rotation algorithm to handle divergence during double talk.  I
 40   added a Geigel Double Talk Detector (DTD) [2] and performed some
 41   G168 tests.  However I had trouble meeting the G168 requirements,
 42   especially for double talk - there were always cases where my DTD
 43   failed, for example where near end speech was under the 6dB
 44   threshold required for declaring double talk.
 45
 46   So I tried a two path algorithm [1], which has so far given better
 47   results.  The original tap rotation/Geigel algorithm is available
 48   in SVN http://svn.rowetel.com/software/oslec/tags/before_16bit.
 49   It's probably possible to make it work if some one wants to put some
 50   serious work into it.
 51
 52   At present no special treatment is provided for tones, which
 53   generally cause NLMS algorithms to diverge.  Initial runs of a
 54   subset of the G168 tests for tones (e.g ./echo_test 6) show the
 55   current algorithm is passing OK, which is kind of surprising.  The
 56   full set of tests needs to be performed to confirm this result.
 57
 58   One other interesting change is that I have managed to get the NLMS
 59   code to work with 16 bit coefficients, rather than the original 32
 60   bit coefficents.  This reduces the MIPs and storage required.
 61   I evaulated the 16 bit port using g168_tests.sh and listening tests
 62   on 4 real-world samples.
 63
 64   I also attempted the implementation of a block based NLMS update
 65   [2] but although this passes g168_tests.sh it didn't converge well
 66   on the real-world samples.  I have no idea why, perhaps a scaling
 67   problem.  The block based code is also available in SVN
 68   http://svn.rowetel.com/software/oslec/tags/before_16bit.  If this
 69   code can be debugged, it will lead to further reduction in MIPS, as
 70   the block update code maps nicely onto DSP instruction sets (it's a
 71   dot product) compared to the current sample-by-sample update.
 72
 73   Steve also has some nice notes on echo cancellers in echo.h
 74
 75   References:
 76
 77   [1] Ochiai, Areseki, and Ogihara, "Echo Canceller with Two Echo
 78       Path Models", IEEE Transactions on communications, COM-25,
 79       No. 6, June
 80       1977.
 81       http://www.rowetel.com/images/echo/dual_path_paper.pdf
 82
 83   [2] The classic, very useful paper that tells you how to
 84       actually build a real world echo canceller:
 85	 Messerschmitt, Hedberg, Cole, Haoui, Winship, "Digital Voice
 86	 Echo Canceller with a TMS320020,
 87	 http://www.rowetel.com/images/echo/spra129.pdf
 88
 89   [3] I have written a series of blog posts on this work, here is
 90       Part 1: http://www.rowetel.com/blog/?p=18
 91
 92   [4] The source code http://svn.rowetel.com/software/oslec/
 93
 94   [5] A nice reference on LMS filters:
 95	 http://en.wikipedia.org/wiki/Least_mean_squares_filter
 96
 97   Credits:
 98
 99   Thanks to Steve Underwood, Jean-Marc Valin, and Ramakrishnan
100   Muthukrishnan for their suggestions and email discussions.  Thanks
101   also to those people who collected echo samples for me such as
102   Mark, Pawel, and Pavel.
103*/
104
105#include <linux/kernel.h>
106#include <linux/module.h>
107#include <linux/slab.h>
108
109#include "echo.h"
110
111#define MIN_TX_POWER_FOR_ADAPTION	64
112#define MIN_RX_POWER_FOR_ADAPTION	64
113#define DTD_HANGOVER			600	/* 600 samples, or 75ms     */
114#define DC_LOG2BETA			3	/* log2() of DC filter Beta */
115
116/* adapting coeffs using the traditional stochastic descent (N)LMS algorithm */
117
118#ifdef __bfin__
119static inline void lms_adapt_bg(struct oslec_state *ec, int clean, int shift)
120{
121	int i, j;
122	int offset1;
123	int offset2;
124	int factor;
125	int exp;
126	int16_t *phist;
127	int n;
128
129	if (shift > 0)
130		factor = clean << shift;
131	else
132		factor = clean >> -shift;
133
134	/* Update the FIR taps */
135
136	offset2 = ec->curr_pos;
137	offset1 = ec->taps - offset2;
138	phist = &ec->fir_state_bg.history[offset2];
139
140	/* st: and en: help us locate the assembler in echo.s */
141
142	/* asm("st:"); */
143	n = ec->taps;
144	for (i = 0, j = offset2; i < n; i++, j++) {
145		exp = *phist++ * factor;
146		ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
147	}
148	/* asm("en:"); */
149
150	/* Note the asm for the inner loop above generated by Blackfin gcc
151	   4.1.1 is pretty good (note even parallel instructions used):
152
153	   R0 = W [P0++] (X);
154	   R0 *= R2;
155	   R0 = R0 + R3 (NS) ||
156	   R1 = W [P1] (X) ||
157	   nop;
158	   R0 >>>= 15;
159	   R0 = R0 + R1;
160	   W [P1++] = R0;
161
162	   A block based update algorithm would be much faster but the
163	   above can't be improved on much.  Every instruction saved in
164	   the loop above is 2 MIPs/ch!  The for loop above is where the
165	   Blackfin spends most of it's time - about 17 MIPs/ch measured
166	   with speedtest.c with 256 taps (32ms).  Write-back and
167	   Write-through cache gave about the same performance.
168	 */
169}
170
171/*
172   IDEAS for further optimisation of lms_adapt_bg():
173
174   1/ The rounding is quite costly.  Could we keep as 32 bit coeffs
175   then make filter pluck the MS 16-bits of the coeffs when filtering?
176   However this would lower potential optimisation of filter, as I
177   think the dual-MAC architecture requires packed 16 bit coeffs.
178
179   2/ Block based update would be more efficient, as per comments above,
180   could use dual MAC architecture.
181
182   3/ Look for same sample Blackfin LMS code, see if we can get dual-MAC
183   packing.
184
185   4/ Execute the whole e/c in a block of say 20ms rather than sample
186   by sample.  Processing a few samples every ms is inefficient.
187*/
188
189#else
190static inline void lms_adapt_bg(struct oslec_state *ec, int clean, int shift)
191{
192	int i;
193
194	int offset1;
195	int offset2;
196	int factor;
197	int exp;
198
199	if (shift > 0)
200		factor = clean << shift;
201	else
202		factor = clean >> -shift;
203
204	/* Update the FIR taps */
205
206	offset2 = ec->curr_pos;
207	offset1 = ec->taps - offset2;
208
209	for (i = ec->taps - 1; i >= offset1; i--) {
210		exp = (ec->fir_state_bg.history[i - offset1] * factor);
211		ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
212	}
213	for (; i >= 0; i--) {
214		exp = (ec->fir_state_bg.history[i + offset2] * factor);
215		ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
216	}
217}
218#endif
219
220static inline int top_bit(unsigned int bits)
221{
222	if (bits == 0)
223		return -1;
224	else
225		return (int)fls((int32_t) bits) - 1;
226}
227
228struct oslec_state *oslec_create(int len, int adaption_mode)
229{
230	struct oslec_state *ec;
231	int i;
232
233	ec = kzalloc(sizeof(*ec), GFP_KERNEL);
234	if (!ec)
235		return NULL;
236
237	ec->taps = len;
238	ec->log2taps = top_bit(len);
239	ec->curr_pos = ec->taps - 1;
240
241	for (i = 0; i < 2; i++) {
242		ec->fir_taps16[i] =
243		    kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
244		if (!ec->fir_taps16[i])
245			goto error_oom;
246	}
247
248	fir16_create(&ec->fir_state, ec->fir_taps16[0], ec->taps);
249	fir16_create(&ec->fir_state_bg, ec->fir_taps16[1], ec->taps);
250
251	for (i = 0; i < 5; i++)
252		ec->xvtx[i] = ec->yvtx[i] = ec->xvrx[i] = ec->yvrx[i] = 0;
253
254	ec->cng_level = 1000;
255	oslec_adaption_mode(ec, adaption_mode);
256
257	ec->snapshot = kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
258	if (!ec->snapshot)
259		goto error_oom;
260
261	ec->cond_met = 0;
262	ec->Pstates = 0;
263	ec->Ltxacc = ec->Lrxacc = ec->Lcleanacc = ec->Lclean_bgacc = 0;
264	ec->Ltx = ec->Lrx = ec->Lclean = ec->Lclean_bg = 0;
265	ec->tx_1 = ec->tx_2 = ec->rx_1 = ec->rx_2 = 0;
266	ec->Lbgn = ec->Lbgn_acc = 0;
267	ec->Lbgn_upper = 200;
268	ec->Lbgn_upper_acc = ec->Lbgn_upper << 13;
269
270	return ec;
271
272error_oom:
273	for (i = 0; i < 2; i++)
274		kfree(ec->fir_taps16[i]);
275
276	kfree(ec);
277	return NULL;
278}
279EXPORT_SYMBOL_GPL(oslec_create);
280
281void oslec_free(struct oslec_state *ec)
282{
283	int i;
284
285	fir16_free(&ec->fir_state);
286	fir16_free(&ec->fir_state_bg);
287	for (i = 0; i < 2; i++)
288		kfree(ec->fir_taps16[i]);
289	kfree(ec->snapshot);
290	kfree(ec);
291}
292EXPORT_SYMBOL_GPL(oslec_free);
293
294void oslec_adaption_mode(struct oslec_state *ec, int adaption_mode)
295{
296	ec->adaption_mode = adaption_mode;
297}
298EXPORT_SYMBOL_GPL(oslec_adaption_mode);
299
300void oslec_flush(struct oslec_state *ec)
301{
302	int i;
303
304	ec->Ltxacc = ec->Lrxacc = ec->Lcleanacc = ec->Lclean_bgacc = 0;
305	ec->Ltx = ec->Lrx = ec->Lclean = ec->Lclean_bg = 0;
306	ec->tx_1 = ec->tx_2 = ec->rx_1 = ec->rx_2 = 0;
307
308	ec->Lbgn = ec->Lbgn_acc = 0;
309	ec->Lbgn_upper = 200;
310	ec->Lbgn_upper_acc = ec->Lbgn_upper << 13;
311
312	ec->nonupdate_dwell = 0;
313
314	fir16_flush(&ec->fir_state);
315	fir16_flush(&ec->fir_state_bg);
316	ec->fir_state.curr_pos = ec->taps - 1;
317	ec->fir_state_bg.curr_pos = ec->taps - 1;
318	for (i = 0; i < 2; i++)
319		memset(ec->fir_taps16[i], 0, ec->taps * sizeof(int16_t));
320
321	ec->curr_pos = ec->taps - 1;
322	ec->Pstates = 0;
323}
324EXPORT_SYMBOL_GPL(oslec_flush);
325
326void oslec_snapshot(struct oslec_state *ec)
327{
328	memcpy(ec->snapshot, ec->fir_taps16[0], ec->taps * sizeof(int16_t));
329}
330EXPORT_SYMBOL_GPL(oslec_snapshot);
331
332/* Dual Path Echo Canceller */
333
334int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
335{
336	int32_t echo_value;
337	int clean_bg;
338	int tmp, tmp1;
339
340	/*
341	 * Input scaling was found be required to prevent problems when tx
342	 * starts clipping.  Another possible way to handle this would be the
343	 * filter coefficent scaling.
344	 */
345
346	ec->tx = tx;
347	ec->rx = rx;
348	tx >>= 1;
349	rx >>= 1;
350
351	/*
352	 * Filter DC, 3dB point is 160Hz (I think), note 32 bit precision
353	 * required otherwise values do not track down to 0. Zero at DC, Pole
354	 * at (1-Beta) on real axis.  Some chip sets (like Si labs) don't
355	 * need this, but something like a $10 X100P card does.  Any DC really
356	 * slows down convergence.
357	 *
358	 * Note: removes some low frequency from the signal, this reduces the
359	 * speech quality when listening to samples through headphones but may
360	 * not be obvious through a telephone handset.
361	 *
362	 * Note that the 3dB frequency in radians is approx Beta, e.g. for Beta
363	 * = 2^(-3) = 0.125, 3dB freq is 0.125 rads = 159Hz.
364	 */
365
366	if (ec->adaption_mode & ECHO_CAN_USE_RX_HPF) {
367		tmp = rx << 15;
368
369		/*
370		 * Make sure the gain of the HPF is 1.0. This can still
371		 * saturate a little under impulse conditions, and it might
372		 * roll to 32768 and need clipping on sustained peak level
373		 * signals. However, the scale of such clipping is small, and
374		 * the error due to any saturation should not markedly affect
375		 * the downstream processing.
376		 */
377		tmp -= (tmp >> 4);
378
379		ec->rx_1 += -(ec->rx_1 >> DC_LOG2BETA) + tmp - ec->rx_2;
380
381		/*
382		 * hard limit filter to prevent clipping.  Note that at this
383		 * stage rx should be limited to +/- 16383 due to right shift
384		 * above
385		 */
386		tmp1 = ec->rx_1 >> 15;
387		if (tmp1 > 16383)
388			tmp1 = 16383;
389		if (tmp1 < -16383)
390			tmp1 = -16383;
391		rx = tmp1;
392		ec->rx_2 = tmp;
393	}
394
395	/* Block average of power in the filter states.  Used for
396	   adaption power calculation. */
397
398	{
399		int new, old;
400
401		/* efficient "out with the old and in with the new" algorithm so
402		   we don't have to recalculate over the whole block of
403		   samples. */
404		new = (int)tx * (int)tx;
405		old = (int)ec->fir_state.history[ec->fir_state.curr_pos] *
406		    (int)ec->fir_state.history[ec->fir_state.curr_pos];
407		ec->Pstates +=
408		    ((new - old) + (1 << (ec->log2taps - 1))) >> ec->log2taps;
409		if (ec->Pstates < 0)
410			ec->Pstates = 0;
411	}
412
413	/* Calculate short term average levels using simple single pole IIRs */
414
415	ec->Ltxacc += abs(tx) - ec->Ltx;
416	ec->Ltx = (ec->Ltxacc + (1 << 4)) >> 5;
417	ec->Lrxacc += abs(rx) - ec->Lrx;
418	ec->Lrx = (ec->Lrxacc + (1 << 4)) >> 5;
419
420	/* Foreground filter */
421
422	ec->fir_state.coeffs = ec->fir_taps16[0];
423	echo_value = fir16(&ec->fir_state, tx);
424	ec->clean = rx - echo_value;
425	ec->Lcleanacc += abs(ec->clean) - ec->Lclean;
426	ec->Lclean = (ec->Lcleanacc + (1 << 4)) >> 5;
427
428	/* Background filter */
429
430	echo_value = fir16(&ec->fir_state_bg, tx);
431	clean_bg = rx - echo_value;
432	ec->Lclean_bgacc += abs(clean_bg) - ec->Lclean_bg;
433	ec->Lclean_bg = (ec->Lclean_bgacc + (1 << 4)) >> 5;
434
435	/* Background Filter adaption */
436
437	/* Almost always adap bg filter, just simple DT and energy
438	   detection to minimise adaption in cases of strong double talk.
439	   However this is not critical for the dual path algorithm.
440	 */
441	ec->factor = 0;
442	ec->shift = 0;
443	if ((ec->nonupdate_dwell == 0)) {
444		int P, logP, shift;
445
446		/* Determine:
447
448		   f = Beta * clean_bg_rx/P ------ (1)
449
450		   where P is the total power in the filter states.
451
452		   The Boffins have shown that if we obey (1) we converge
453		   quickly and avoid instability.
454
455		   The correct factor f must be in Q30, as this is the fixed
456		   point format required by the lms_adapt_bg() function,
457		   therefore the scaled version of (1) is:
458
459		   (2^30) * f  = (2^30) * Beta * clean_bg_rx/P
460		   factor      = (2^30) * Beta * clean_bg_rx/P     ----- (2)
461
462		   We have chosen Beta = 0.25 by experiment, so:
463
464		   factor      = (2^30) * (2^-2) * clean_bg_rx/P
465
466		   (30 - 2 - log2(P))
467		   factor      = clean_bg_rx 2                     ----- (3)
468
469		   To avoid a divide we approximate log2(P) as top_bit(P),
470		   which returns the position of the highest non-zero bit in
471		   P.  This approximation introduces an error as large as a
472		   factor of 2, but the algorithm seems to handle it OK.
473
474		   Come to think of it a divide may not be a big deal on a
475		   modern DSP, so its probably worth checking out the cycles
476		   for a divide versus a top_bit() implementation.
477		 */
478
479		P = MIN_TX_POWER_FOR_ADAPTION + ec->Pstates;
480		logP = top_bit(P) + ec->log2taps;
481		shift = 30 - 2 - logP;
482		ec->shift = shift;
483
484		lms_adapt_bg(ec, clean_bg, shift);
485	}
486
487	/* very simple DTD to make sure we dont try and adapt with strong
488	   near end speech */
489
490	ec->adapt = 0;
491	if ((ec->Lrx > MIN_RX_POWER_FOR_ADAPTION) && (ec->Lrx > ec->Ltx))
492		ec->nonupdate_dwell = DTD_HANGOVER;
493	if (ec->nonupdate_dwell)
494		ec->nonupdate_dwell--;
495
496	/* Transfer logic */
497
498	/* These conditions are from the dual path paper [1], I messed with
499	   them a bit to improve performance. */
500
501	if ((ec->adaption_mode & ECHO_CAN_USE_ADAPTION) &&
502	    (ec->nonupdate_dwell == 0) &&
503	    /* (ec->Lclean_bg < 0.875*ec->Lclean) */
504	    (8 * ec->Lclean_bg < 7 * ec->Lclean) &&
505	    /* (ec->Lclean_bg < 0.125*ec->Ltx) */
506	    (8 * ec->Lclean_bg < ec->Ltx)) {
507		if (ec->cond_met == 6) {
508			/*
509			 * BG filter has had better results for 6 consecutive
510			 * samples
511			 */
512			ec->adapt = 1;
513			memcpy(ec->fir_taps16[0], ec->fir_taps16[1],
514			       ec->taps * sizeof(int16_t));
515		} else
516			ec->cond_met++;
517	} else
518		ec->cond_met = 0;
519
520	/* Non-Linear Processing */
521
522	ec->clean_nlp = ec->clean;
523	if (ec->adaption_mode & ECHO_CAN_USE_NLP) {
524		/*
525		 * Non-linear processor - a fancy way to say "zap small
526		 * signals, to avoid residual echo due to (uLaw/ALaw)
527		 * non-linearity in the channel.".
528		 */
529
530		if ((16 * ec->Lclean < ec->Ltx)) {
531			/*
532			 * Our e/c has improved echo by at least 24 dB (each
533			 * factor of 2 is 6dB, so 2*2*2*2=16 is the same as
534			 * 6+6+6+6=24dB)
535			 */
536			if (ec->adaption_mode & ECHO_CAN_USE_CNG) {
537				ec->cng_level = ec->Lbgn;
538
539				/*
540				 * Very elementary comfort noise generation.
541				 * Just random numbers rolled off very vaguely
542				 * Hoth-like.  DR: This noise doesn't sound
543				 * quite right to me - I suspect there are some
544				 * overflow issues in the filtering as it's too
545				 * "crackly".
546				 * TODO: debug this, maybe just play noise at
547				 * high level or look at spectrum.
548				 */
549
550				ec->cng_rndnum =
551				    1664525U * ec->cng_rndnum + 1013904223U;
552				ec->cng_filter =
553				    ((ec->cng_rndnum & 0xFFFF) - 32768 +
554				     5 * ec->cng_filter) >> 3;
555				ec->clean_nlp =
556				    (ec->cng_filter * ec->cng_level * 8) >> 14;
557
558			} else if (ec->adaption_mode & ECHO_CAN_USE_CLIP) {
559				/* This sounds much better than CNG */
560				if (ec->clean_nlp > ec->Lbgn)
561					ec->clean_nlp = ec->Lbgn;
562				if (ec->clean_nlp < -ec->Lbgn)
563					ec->clean_nlp = -ec->Lbgn;
564			} else {
565				/*
566				 * just mute the residual, doesn't sound very
567				 * good, used mainly in G168 tests
568				 */
569				ec->clean_nlp = 0;
570			}
571		} else {
572			/*
573			 * Background noise estimator.  I tried a few
574			 * algorithms here without much luck.  This very simple
575			 * one seems to work best, we just average the level
576			 * using a slow (1 sec time const) filter if the
577			 * current level is less than a (experimentally
578			 * derived) constant.  This means we dont include high
579			 * level signals like near end speech.  When combined
580			 * with CNG or especially CLIP seems to work OK.
581			 */
582			if (ec->Lclean < 40) {
583				ec->Lbgn_acc += abs(ec->clean) - ec->Lbgn;
584				ec->Lbgn = (ec->Lbgn_acc + (1 << 11)) >> 12;
585			}
586		}
587	}
588
589	/* Roll around the taps buffer */
590	if (ec->curr_pos <= 0)
591		ec->curr_pos = ec->taps;
592	ec->curr_pos--;
593
594	if (ec->adaption_mode & ECHO_CAN_DISABLE)
595		ec->clean_nlp = rx;
596
597	/* Output scaled back up again to match input scaling */
598
599	return (int16_t) ec->clean_nlp << 1;
600}
601EXPORT_SYMBOL_GPL(oslec_update);
602
603/* This function is separated from the echo canceller is it is usually called
604   as part of the tx process.  See rx HP (DC blocking) filter above, it's
605   the same design.
606
607   Some soft phones send speech signals with a lot of low frequency
608   energy, e.g. down to 20Hz.  This can make the hybrid non-linear
609   which causes the echo canceller to fall over.  This filter can help
610   by removing any low frequency before it gets to the tx port of the
611   hybrid.
612
613   It can also help by removing and DC in the tx signal.  DC is bad
614   for LMS algorithms.
615
616   This is one of the classic DC removal filters, adjusted to provide
617   sufficient bass rolloff to meet the above requirement to protect hybrids
618   from things that upset them. The difference between successive samples
619   produces a lousy HPF, and then a suitably placed pole flattens things out.
620   The final result is a nicely rolled off bass end. The filtering is
621   implemented with extended fractional precision, which noise shapes things,
622   giving very clean DC removal.
623*/
624
625int16_t oslec_hpf_tx(struct oslec_state *ec, int16_t tx)
626{
627	int tmp, tmp1;
628
629	if (ec->adaption_mode & ECHO_CAN_USE_TX_HPF) {
630		tmp = tx << 15;
631
632		/*
633		 * Make sure the gain of the HPF is 1.0. The first can still
634		 * saturate a little under impulse conditions, and it might
635		 * roll to 32768 and need clipping on sustained peak level
636		 * signals. However, the scale of such clipping is small, and
637		 * the error due to any saturation should not markedly affect
638		 * the downstream processing.
639		 */
640		tmp -= (tmp >> 4);
641
642		ec->tx_1 += -(ec->tx_1 >> DC_LOG2BETA) + tmp - ec->tx_2;
643		tmp1 = ec->tx_1 >> 15;
644		if (tmp1 > 32767)
645			tmp1 = 32767;
646		if (tmp1 < -32767)
647			tmp1 = -32767;
648		tx = tmp1;
649		ec->tx_2 = tmp;
650	}
651
652	return tx;
653}
654EXPORT_SYMBOL_GPL(oslec_hpf_tx);
655
656MODULE_LICENSE("GPL");
657MODULE_AUTHOR("David Rowe");
658MODULE_DESCRIPTION("Open Source Line Echo Canceller");
659MODULE_VERSION("0.3.0");