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v6.8
  1// SPDX-License-Identifier: GPL-2.0-only
  2/*
  3 * SpanDSP - a series of DSP components for telephony
  4 *
  5 * echo.c - A line echo canceller.  This code is being developed
  6 *          against and partially complies with G168.
  7 *
  8 * Written by Steve Underwood <steveu@coppice.org>
  9 *         and David Rowe <david_at_rowetel_dot_com>
 10 *
 11 * Copyright (C) 2001, 2003 Steve Underwood, 2007 David Rowe
 12 *
 13 * Based on a bit from here, a bit from there, eye of toad, ear of
 14 * bat, 15 years of failed attempts by David and a few fried brain
 15 * cells.
 16 *
 17 * All rights reserved.
 
 
 
 
 
 
 
 
 
 
 
 
 
 18 */
 19
 20/*! \file */
 21
 22/* Implementation Notes
 23   David Rowe
 24   April 2007
 25
 26   This code started life as Steve's NLMS algorithm with a tap
 27   rotation algorithm to handle divergence during double talk.  I
 28   added a Geigel Double Talk Detector (DTD) [2] and performed some
 29   G168 tests.  However I had trouble meeting the G168 requirements,
 30   especially for double talk - there were always cases where my DTD
 31   failed, for example where near end speech was under the 6dB
 32   threshold required for declaring double talk.
 33
 34   So I tried a two path algorithm [1], which has so far given better
 35   results.  The original tap rotation/Geigel algorithm is available
 36   in SVN http://svn.rowetel.com/software/oslec/tags/before_16bit.
 37   It's probably possible to make it work if some one wants to put some
 38   serious work into it.
 39
 40   At present no special treatment is provided for tones, which
 41   generally cause NLMS algorithms to diverge.  Initial runs of a
 42   subset of the G168 tests for tones (e.g ./echo_test 6) show the
 43   current algorithm is passing OK, which is kind of surprising.  The
 44   full set of tests needs to be performed to confirm this result.
 45
 46   One other interesting change is that I have managed to get the NLMS
 47   code to work with 16 bit coefficients, rather than the original 32
 48   bit coefficents.  This reduces the MIPs and storage required.
 49   I evaulated the 16 bit port using g168_tests.sh and listening tests
 50   on 4 real-world samples.
 51
 52   I also attempted the implementation of a block based NLMS update
 53   [2] but although this passes g168_tests.sh it didn't converge well
 54   on the real-world samples.  I have no idea why, perhaps a scaling
 55   problem.  The block based code is also available in SVN
 56   http://svn.rowetel.com/software/oslec/tags/before_16bit.  If this
 57   code can be debugged, it will lead to further reduction in MIPS, as
 58   the block update code maps nicely onto DSP instruction sets (it's a
 59   dot product) compared to the current sample-by-sample update.
 60
 61   Steve also has some nice notes on echo cancellers in echo.h
 62
 63   References:
 64
 65   [1] Ochiai, Areseki, and Ogihara, "Echo Canceller with Two Echo
 66       Path Models", IEEE Transactions on communications, COM-25,
 67       No. 6, June
 68       1977.
 69       https://www.rowetel.com/images/echo/dual_path_paper.pdf
 70
 71   [2] The classic, very useful paper that tells you how to
 72       actually build a real world echo canceller:
 73	 Messerschmitt, Hedberg, Cole, Haoui, Winship, "Digital Voice
 74	 Echo Canceller with a TMS320020,
 75	 https://www.rowetel.com/images/echo/spra129.pdf
 76
 77   [3] I have written a series of blog posts on this work, here is
 78       Part 1: http://www.rowetel.com/blog/?p=18
 79
 80   [4] The source code http://svn.rowetel.com/software/oslec/
 81
 82   [5] A nice reference on LMS filters:
 83	 https://en.wikipedia.org/wiki/Least_mean_squares_filter
 84
 85   Credits:
 86
 87   Thanks to Steve Underwood, Jean-Marc Valin, and Ramakrishnan
 88   Muthukrishnan for their suggestions and email discussions.  Thanks
 89   also to those people who collected echo samples for me such as
 90   Mark, Pawel, and Pavel.
 91*/
 92
 93#include <linux/kernel.h>
 94#include <linux/module.h>
 95#include <linux/slab.h>
 96
 97#include "echo.h"
 98
 99#define MIN_TX_POWER_FOR_ADAPTION	64
100#define MIN_RX_POWER_FOR_ADAPTION	64
101#define DTD_HANGOVER			600	/* 600 samples, or 75ms     */
102#define DC_LOG2BETA			3	/* log2() of DC filter Beta */
103
104/* adapting coeffs using the traditional stochastic descent (N)LMS algorithm */
105
106static inline void lms_adapt_bg(struct oslec_state *ec, int clean, int shift)
107{
108	int i;
109
110	int offset1;
111	int offset2;
112	int factor;
113	int exp;
114
115	if (shift > 0)
116		factor = clean << shift;
117	else
118		factor = clean >> -shift;
119
120	/* Update the FIR taps */
121
122	offset2 = ec->curr_pos;
123	offset1 = ec->taps - offset2;
124
125	for (i = ec->taps - 1; i >= offset1; i--) {
126		exp = (ec->fir_state_bg.history[i - offset1] * factor);
127		ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
128	}
129	for (; i >= 0; i--) {
130		exp = (ec->fir_state_bg.history[i + offset2] * factor);
131		ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
132	}
133}
134
135static inline int top_bit(unsigned int bits)
136{
137	if (bits == 0)
138		return -1;
139	else
140		return (int)fls((int32_t) bits) - 1;
141}
142
143struct oslec_state *oslec_create(int len, int adaption_mode)
144{
145	struct oslec_state *ec;
146	int i;
147	const int16_t *history;
148
149	ec = kzalloc(sizeof(*ec), GFP_KERNEL);
150	if (!ec)
151		return NULL;
152
153	ec->taps = len;
154	ec->log2taps = top_bit(len);
155	ec->curr_pos = ec->taps - 1;
156
157	ec->fir_taps16[0] =
158	    kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
159	if (!ec->fir_taps16[0])
160		goto error_oom_0;
161
162	ec->fir_taps16[1] =
163	    kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
164	if (!ec->fir_taps16[1])
165		goto error_oom_1;
166
167	history = fir16_create(&ec->fir_state, ec->fir_taps16[0], ec->taps);
168	if (!history)
169		goto error_state;
170	history = fir16_create(&ec->fir_state_bg, ec->fir_taps16[1], ec->taps);
171	if (!history)
172		goto error_state_bg;
173
174	for (i = 0; i < 5; i++)
175		ec->xvtx[i] = ec->yvtx[i] = ec->xvrx[i] = ec->yvrx[i] = 0;
176
177	ec->cng_level = 1000;
178	oslec_adaption_mode(ec, adaption_mode);
179
180	ec->snapshot = kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
181	if (!ec->snapshot)
182		goto error_snap;
183
184	ec->cond_met = 0;
185	ec->pstates = 0;
186	ec->ltxacc = ec->lrxacc = ec->lcleanacc = ec->lclean_bgacc = 0;
187	ec->ltx = ec->lrx = ec->lclean = ec->lclean_bg = 0;
188	ec->tx_1 = ec->tx_2 = ec->rx_1 = ec->rx_2 = 0;
189	ec->lbgn = ec->lbgn_acc = 0;
190	ec->lbgn_upper = 200;
191	ec->lbgn_upper_acc = ec->lbgn_upper << 13;
192
193	return ec;
194
195error_snap:
196	fir16_free(&ec->fir_state_bg);
197error_state_bg:
198	fir16_free(&ec->fir_state);
199error_state:
200	kfree(ec->fir_taps16[1]);
201error_oom_1:
202	kfree(ec->fir_taps16[0]);
203error_oom_0:
204	kfree(ec);
205	return NULL;
206}
207EXPORT_SYMBOL_GPL(oslec_create);
208
209void oslec_free(struct oslec_state *ec)
210{
211	int i;
212
213	fir16_free(&ec->fir_state);
214	fir16_free(&ec->fir_state_bg);
215	for (i = 0; i < 2; i++)
216		kfree(ec->fir_taps16[i]);
217	kfree(ec->snapshot);
218	kfree(ec);
219}
220EXPORT_SYMBOL_GPL(oslec_free);
221
222void oslec_adaption_mode(struct oslec_state *ec, int adaption_mode)
223{
224	ec->adaption_mode = adaption_mode;
225}
226EXPORT_SYMBOL_GPL(oslec_adaption_mode);
227
228void oslec_flush(struct oslec_state *ec)
229{
230	int i;
231
232	ec->ltxacc = ec->lrxacc = ec->lcleanacc = ec->lclean_bgacc = 0;
233	ec->ltx = ec->lrx = ec->lclean = ec->lclean_bg = 0;
234	ec->tx_1 = ec->tx_2 = ec->rx_1 = ec->rx_2 = 0;
235
236	ec->lbgn = ec->lbgn_acc = 0;
237	ec->lbgn_upper = 200;
238	ec->lbgn_upper_acc = ec->lbgn_upper << 13;
239
240	ec->nonupdate_dwell = 0;
241
242	fir16_flush(&ec->fir_state);
243	fir16_flush(&ec->fir_state_bg);
244	ec->fir_state.curr_pos = ec->taps - 1;
245	ec->fir_state_bg.curr_pos = ec->taps - 1;
246	for (i = 0; i < 2; i++)
247		memset(ec->fir_taps16[i], 0, ec->taps * sizeof(int16_t));
248
249	ec->curr_pos = ec->taps - 1;
250	ec->pstates = 0;
251}
252EXPORT_SYMBOL_GPL(oslec_flush);
253
254void oslec_snapshot(struct oslec_state *ec)
255{
256	memcpy(ec->snapshot, ec->fir_taps16[0], ec->taps * sizeof(int16_t));
257}
258EXPORT_SYMBOL_GPL(oslec_snapshot);
259
260/* Dual Path Echo Canceller */
261
262int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
263{
264	int32_t echo_value;
265	int clean_bg;
266	int tmp;
267	int tmp1;
268
269	/*
270	 * Input scaling was found be required to prevent problems when tx
271	 * starts clipping.  Another possible way to handle this would be the
272	 * filter coefficent scaling.
273	 */
274
275	ec->tx = tx;
276	ec->rx = rx;
277	tx >>= 1;
278	rx >>= 1;
279
280	/*
281	 * Filter DC, 3dB point is 160Hz (I think), note 32 bit precision
282	 * required otherwise values do not track down to 0. Zero at DC, Pole
283	 * at (1-Beta) on real axis.  Some chip sets (like Si labs) don't
284	 * need this, but something like a $10 X100P card does.  Any DC really
285	 * slows down convergence.
286	 *
287	 * Note: removes some low frequency from the signal, this reduces the
288	 * speech quality when listening to samples through headphones but may
289	 * not be obvious through a telephone handset.
290	 *
291	 * Note that the 3dB frequency in radians is approx Beta, e.g. for Beta
292	 * = 2^(-3) = 0.125, 3dB freq is 0.125 rads = 159Hz.
293	 */
294
295	if (ec->adaption_mode & ECHO_CAN_USE_RX_HPF) {
296		tmp = rx << 15;
297
298		/*
299		 * Make sure the gain of the HPF is 1.0. This can still
300		 * saturate a little under impulse conditions, and it might
301		 * roll to 32768 and need clipping on sustained peak level
302		 * signals. However, the scale of such clipping is small, and
303		 * the error due to any saturation should not markedly affect
304		 * the downstream processing.
305		 */
306		tmp -= (tmp >> 4);
307
308		ec->rx_1 += -(ec->rx_1 >> DC_LOG2BETA) + tmp - ec->rx_2;
309
310		/*
311		 * hard limit filter to prevent clipping.  Note that at this
312		 * stage rx should be limited to +/- 16383 due to right shift
313		 * above
314		 */
315		tmp1 = ec->rx_1 >> 15;
316		if (tmp1 > 16383)
317			tmp1 = 16383;
318		if (tmp1 < -16383)
319			tmp1 = -16383;
320		rx = tmp1;
321		ec->rx_2 = tmp;
322	}
323
324	/* Block average of power in the filter states.  Used for
325	   adaption power calculation. */
326
327	{
328		int new, old;
329
330		/* efficient "out with the old and in with the new" algorithm so
331		   we don't have to recalculate over the whole block of
332		   samples. */
333		new = (int)tx * (int)tx;
334		old = (int)ec->fir_state.history[ec->fir_state.curr_pos] *
335		    (int)ec->fir_state.history[ec->fir_state.curr_pos];
336		ec->pstates +=
337		    ((new - old) + (1 << (ec->log2taps - 1))) >> ec->log2taps;
338		if (ec->pstates < 0)
339			ec->pstates = 0;
340	}
341
342	/* Calculate short term average levels using simple single pole IIRs */
343
344	ec->ltxacc += abs(tx) - ec->ltx;
345	ec->ltx = (ec->ltxacc + (1 << 4)) >> 5;
346	ec->lrxacc += abs(rx) - ec->lrx;
347	ec->lrx = (ec->lrxacc + (1 << 4)) >> 5;
348
349	/* Foreground filter */
350
351	ec->fir_state.coeffs = ec->fir_taps16[0];
352	echo_value = fir16(&ec->fir_state, tx);
353	ec->clean = rx - echo_value;
354	ec->lcleanacc += abs(ec->clean) - ec->lclean;
355	ec->lclean = (ec->lcleanacc + (1 << 4)) >> 5;
356
357	/* Background filter */
358
359	echo_value = fir16(&ec->fir_state_bg, tx);
360	clean_bg = rx - echo_value;
361	ec->lclean_bgacc += abs(clean_bg) - ec->lclean_bg;
362	ec->lclean_bg = (ec->lclean_bgacc + (1 << 4)) >> 5;
363
364	/* Background Filter adaption */
365
366	/* Almost always adap bg filter, just simple DT and energy
367	   detection to minimise adaption in cases of strong double talk.
368	   However this is not critical for the dual path algorithm.
369	 */
370	ec->factor = 0;
371	ec->shift = 0;
372	if (!ec->nonupdate_dwell) {
373		int p, logp, shift;
374
375		/* Determine:
376
377		   f = Beta * clean_bg_rx/P ------ (1)
378
379		   where P is the total power in the filter states.
380
381		   The Boffins have shown that if we obey (1) we converge
382		   quickly and avoid instability.
383
384		   The correct factor f must be in Q30, as this is the fixed
385		   point format required by the lms_adapt_bg() function,
386		   therefore the scaled version of (1) is:
387
388		   (2^30) * f  = (2^30) * Beta * clean_bg_rx/P
389		   factor      = (2^30) * Beta * clean_bg_rx/P     ----- (2)
390
391		   We have chosen Beta = 0.25 by experiment, so:
392
393		   factor      = (2^30) * (2^-2) * clean_bg_rx/P
394
395		   (30 - 2 - log2(P))
396		   factor      = clean_bg_rx 2                     ----- (3)
397
398		   To avoid a divide we approximate log2(P) as top_bit(P),
399		   which returns the position of the highest non-zero bit in
400		   P.  This approximation introduces an error as large as a
401		   factor of 2, but the algorithm seems to handle it OK.
402
403		   Come to think of it a divide may not be a big deal on a
404		   modern DSP, so its probably worth checking out the cycles
405		   for a divide versus a top_bit() implementation.
406		 */
407
408		p = MIN_TX_POWER_FOR_ADAPTION + ec->pstates;
409		logp = top_bit(p) + ec->log2taps;
410		shift = 30 - 2 - logp;
411		ec->shift = shift;
412
413		lms_adapt_bg(ec, clean_bg, shift);
414	}
415
416	/* very simple DTD to make sure we dont try and adapt with strong
417	   near end speech */
418
419	ec->adapt = 0;
420	if ((ec->lrx > MIN_RX_POWER_FOR_ADAPTION) && (ec->lrx > ec->ltx))
421		ec->nonupdate_dwell = DTD_HANGOVER;
422	if (ec->nonupdate_dwell)
423		ec->nonupdate_dwell--;
424
425	/* Transfer logic */
426
427	/* These conditions are from the dual path paper [1], I messed with
428	   them a bit to improve performance. */
429
430	if ((ec->adaption_mode & ECHO_CAN_USE_ADAPTION) &&
431	    (ec->nonupdate_dwell == 0) &&
432	    /* (ec->Lclean_bg < 0.875*ec->Lclean) */
433	    (8 * ec->lclean_bg < 7 * ec->lclean) &&
434	    /* (ec->Lclean_bg < 0.125*ec->Ltx) */
435	    (8 * ec->lclean_bg < ec->ltx)) {
436		if (ec->cond_met == 6) {
437			/*
438			 * BG filter has had better results for 6 consecutive
439			 * samples
440			 */
441			ec->adapt = 1;
442			memcpy(ec->fir_taps16[0], ec->fir_taps16[1],
443			       ec->taps * sizeof(int16_t));
444		} else
445			ec->cond_met++;
446	} else
447		ec->cond_met = 0;
448
449	/* Non-Linear Processing */
450
451	ec->clean_nlp = ec->clean;
452	if (ec->adaption_mode & ECHO_CAN_USE_NLP) {
453		/*
454		 * Non-linear processor - a fancy way to say "zap small
455		 * signals, to avoid residual echo due to (uLaw/ALaw)
456		 * non-linearity in the channel.".
457		 */
458
459		if ((16 * ec->lclean < ec->ltx)) {
460			/*
461			 * Our e/c has improved echo by at least 24 dB (each
462			 * factor of 2 is 6dB, so 2*2*2*2=16 is the same as
463			 * 6+6+6+6=24dB)
464			 */
465			if (ec->adaption_mode & ECHO_CAN_USE_CNG) {
466				ec->cng_level = ec->lbgn;
467
468				/*
469				 * Very elementary comfort noise generation.
470				 * Just random numbers rolled off very vaguely
471				 * Hoth-like.  DR: This noise doesn't sound
472				 * quite right to me - I suspect there are some
473				 * overflow issues in the filtering as it's too
474				 * "crackly".
475				 * TODO: debug this, maybe just play noise at
476				 * high level or look at spectrum.
477				 */
478
479				ec->cng_rndnum =
480				    1664525U * ec->cng_rndnum + 1013904223U;
481				ec->cng_filter =
482				    ((ec->cng_rndnum & 0xFFFF) - 32768 +
483				     5 * ec->cng_filter) >> 3;
484				ec->clean_nlp =
485				    (ec->cng_filter * ec->cng_level * 8) >> 14;
486
487			} else if (ec->adaption_mode & ECHO_CAN_USE_CLIP) {
488				/* This sounds much better than CNG */
489				if (ec->clean_nlp > ec->lbgn)
490					ec->clean_nlp = ec->lbgn;
491				if (ec->clean_nlp < -ec->lbgn)
492					ec->clean_nlp = -ec->lbgn;
493			} else {
494				/*
495				 * just mute the residual, doesn't sound very
496				 * good, used mainly in G168 tests
497				 */
498				ec->clean_nlp = 0;
499			}
500		} else {
501			/*
502			 * Background noise estimator.  I tried a few
503			 * algorithms here without much luck.  This very simple
504			 * one seems to work best, we just average the level
505			 * using a slow (1 sec time const) filter if the
506			 * current level is less than a (experimentally
507			 * derived) constant.  This means we dont include high
508			 * level signals like near end speech.  When combined
509			 * with CNG or especially CLIP seems to work OK.
510			 */
511			if (ec->lclean < 40) {
512				ec->lbgn_acc += abs(ec->clean) - ec->lbgn;
513				ec->lbgn = (ec->lbgn_acc + (1 << 11)) >> 12;
514			}
515		}
516	}
517
518	/* Roll around the taps buffer */
519	if (ec->curr_pos <= 0)
520		ec->curr_pos = ec->taps;
521	ec->curr_pos--;
522
523	if (ec->adaption_mode & ECHO_CAN_DISABLE)
524		ec->clean_nlp = rx;
525
526	/* Output scaled back up again to match input scaling */
527
528	return (int16_t) ec->clean_nlp << 1;
529}
530EXPORT_SYMBOL_GPL(oslec_update);
531
532/* This function is separated from the echo canceller is it is usually called
533   as part of the tx process.  See rx HP (DC blocking) filter above, it's
534   the same design.
535
536   Some soft phones send speech signals with a lot of low frequency
537   energy, e.g. down to 20Hz.  This can make the hybrid non-linear
538   which causes the echo canceller to fall over.  This filter can help
539   by removing any low frequency before it gets to the tx port of the
540   hybrid.
541
542   It can also help by removing and DC in the tx signal.  DC is bad
543   for LMS algorithms.
544
545   This is one of the classic DC removal filters, adjusted to provide
546   sufficient bass rolloff to meet the above requirement to protect hybrids
547   from things that upset them. The difference between successive samples
548   produces a lousy HPF, and then a suitably placed pole flattens things out.
549   The final result is a nicely rolled off bass end. The filtering is
550   implemented with extended fractional precision, which noise shapes things,
551   giving very clean DC removal.
552*/
553
554int16_t oslec_hpf_tx(struct oslec_state *ec, int16_t tx)
555{
556	int tmp;
557	int tmp1;
558
559	if (ec->adaption_mode & ECHO_CAN_USE_TX_HPF) {
560		tmp = tx << 15;
561
562		/*
563		 * Make sure the gain of the HPF is 1.0. The first can still
564		 * saturate a little under impulse conditions, and it might
565		 * roll to 32768 and need clipping on sustained peak level
566		 * signals. However, the scale of such clipping is small, and
567		 * the error due to any saturation should not markedly affect
568		 * the downstream processing.
569		 */
570		tmp -= (tmp >> 4);
571
572		ec->tx_1 += -(ec->tx_1 >> DC_LOG2BETA) + tmp - ec->tx_2;
573		tmp1 = ec->tx_1 >> 15;
574		if (tmp1 > 32767)
575			tmp1 = 32767;
576		if (tmp1 < -32767)
577			tmp1 = -32767;
578		tx = tmp1;
579		ec->tx_2 = tmp;
580	}
581
582	return tx;
583}
584EXPORT_SYMBOL_GPL(oslec_hpf_tx);
585
586MODULE_LICENSE("GPL");
587MODULE_AUTHOR("David Rowe");
588MODULE_DESCRIPTION("Open Source Line Echo Canceller");
589MODULE_VERSION("0.3.0");
v4.17
 
  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
118static inline void lms_adapt_bg(struct oslec_state *ec, int clean, int shift)
119{
120	int i;
121
122	int offset1;
123	int offset2;
124	int factor;
125	int exp;
126
127	if (shift > 0)
128		factor = clean << shift;
129	else
130		factor = clean >> -shift;
131
132	/* Update the FIR taps */
133
134	offset2 = ec->curr_pos;
135	offset1 = ec->taps - offset2;
136
137	for (i = ec->taps - 1; i >= offset1; i--) {
138		exp = (ec->fir_state_bg.history[i - offset1] * factor);
139		ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
140	}
141	for (; i >= 0; i--) {
142		exp = (ec->fir_state_bg.history[i + offset2] * factor);
143		ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
144	}
145}
146
147static inline int top_bit(unsigned int bits)
148{
149	if (bits == 0)
150		return -1;
151	else
152		return (int)fls((int32_t) bits) - 1;
153}
154
155struct oslec_state *oslec_create(int len, int adaption_mode)
156{
157	struct oslec_state *ec;
158	int i;
159	const int16_t *history;
160
161	ec = kzalloc(sizeof(*ec), GFP_KERNEL);
162	if (!ec)
163		return NULL;
164
165	ec->taps = len;
166	ec->log2taps = top_bit(len);
167	ec->curr_pos = ec->taps - 1;
168
169	ec->fir_taps16[0] =
170	    kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
171	if (!ec->fir_taps16[0])
172		goto error_oom_0;
173
174	ec->fir_taps16[1] =
175	    kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
176	if (!ec->fir_taps16[1])
177		goto error_oom_1;
178
179	history = fir16_create(&ec->fir_state, ec->fir_taps16[0], ec->taps);
180	if (!history)
181		goto error_state;
182	history = fir16_create(&ec->fir_state_bg, ec->fir_taps16[1], ec->taps);
183	if (!history)
184		goto error_state_bg;
185
186	for (i = 0; i < 5; i++)
187		ec->xvtx[i] = ec->yvtx[i] = ec->xvrx[i] = ec->yvrx[i] = 0;
188
189	ec->cng_level = 1000;
190	oslec_adaption_mode(ec, adaption_mode);
191
192	ec->snapshot = kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
193	if (!ec->snapshot)
194		goto error_snap;
195
196	ec->cond_met = 0;
197	ec->pstates = 0;
198	ec->ltxacc = ec->lrxacc = ec->lcleanacc = ec->lclean_bgacc = 0;
199	ec->ltx = ec->lrx = ec->lclean = ec->lclean_bg = 0;
200	ec->tx_1 = ec->tx_2 = ec->rx_1 = ec->rx_2 = 0;
201	ec->lbgn = ec->lbgn_acc = 0;
202	ec->lbgn_upper = 200;
203	ec->lbgn_upper_acc = ec->lbgn_upper << 13;
204
205	return ec;
206
207error_snap:
208	fir16_free(&ec->fir_state_bg);
209error_state_bg:
210	fir16_free(&ec->fir_state);
211error_state:
212	kfree(ec->fir_taps16[1]);
213error_oom_1:
214	kfree(ec->fir_taps16[0]);
215error_oom_0:
216	kfree(ec);
217	return NULL;
218}
219EXPORT_SYMBOL_GPL(oslec_create);
220
221void oslec_free(struct oslec_state *ec)
222{
223	int i;
224
225	fir16_free(&ec->fir_state);
226	fir16_free(&ec->fir_state_bg);
227	for (i = 0; i < 2; i++)
228		kfree(ec->fir_taps16[i]);
229	kfree(ec->snapshot);
230	kfree(ec);
231}
232EXPORT_SYMBOL_GPL(oslec_free);
233
234void oslec_adaption_mode(struct oslec_state *ec, int adaption_mode)
235{
236	ec->adaption_mode = adaption_mode;
237}
238EXPORT_SYMBOL_GPL(oslec_adaption_mode);
239
240void oslec_flush(struct oslec_state *ec)
241{
242	int i;
243
244	ec->ltxacc = ec->lrxacc = ec->lcleanacc = ec->lclean_bgacc = 0;
245	ec->ltx = ec->lrx = ec->lclean = ec->lclean_bg = 0;
246	ec->tx_1 = ec->tx_2 = ec->rx_1 = ec->rx_2 = 0;
247
248	ec->lbgn = ec->lbgn_acc = 0;
249	ec->lbgn_upper = 200;
250	ec->lbgn_upper_acc = ec->lbgn_upper << 13;
251
252	ec->nonupdate_dwell = 0;
253
254	fir16_flush(&ec->fir_state);
255	fir16_flush(&ec->fir_state_bg);
256	ec->fir_state.curr_pos = ec->taps - 1;
257	ec->fir_state_bg.curr_pos = ec->taps - 1;
258	for (i = 0; i < 2; i++)
259		memset(ec->fir_taps16[i], 0, ec->taps * sizeof(int16_t));
260
261	ec->curr_pos = ec->taps - 1;
262	ec->pstates = 0;
263}
264EXPORT_SYMBOL_GPL(oslec_flush);
265
266void oslec_snapshot(struct oslec_state *ec)
267{
268	memcpy(ec->snapshot, ec->fir_taps16[0], ec->taps * sizeof(int16_t));
269}
270EXPORT_SYMBOL_GPL(oslec_snapshot);
271
272/* Dual Path Echo Canceller */
273
274int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
275{
276	int32_t echo_value;
277	int clean_bg;
278	int tmp;
279	int tmp1;
280
281	/*
282	 * Input scaling was found be required to prevent problems when tx
283	 * starts clipping.  Another possible way to handle this would be the
284	 * filter coefficent scaling.
285	 */
286
287	ec->tx = tx;
288	ec->rx = rx;
289	tx >>= 1;
290	rx >>= 1;
291
292	/*
293	 * Filter DC, 3dB point is 160Hz (I think), note 32 bit precision
294	 * required otherwise values do not track down to 0. Zero at DC, Pole
295	 * at (1-Beta) on real axis.  Some chip sets (like Si labs) don't
296	 * need this, but something like a $10 X100P card does.  Any DC really
297	 * slows down convergence.
298	 *
299	 * Note: removes some low frequency from the signal, this reduces the
300	 * speech quality when listening to samples through headphones but may
301	 * not be obvious through a telephone handset.
302	 *
303	 * Note that the 3dB frequency in radians is approx Beta, e.g. for Beta
304	 * = 2^(-3) = 0.125, 3dB freq is 0.125 rads = 159Hz.
305	 */
306
307	if (ec->adaption_mode & ECHO_CAN_USE_RX_HPF) {
308		tmp = rx << 15;
309
310		/*
311		 * Make sure the gain of the HPF is 1.0. This can still
312		 * saturate a little under impulse conditions, and it might
313		 * roll to 32768 and need clipping on sustained peak level
314		 * signals. However, the scale of such clipping is small, and
315		 * the error due to any saturation should not markedly affect
316		 * the downstream processing.
317		 */
318		tmp -= (tmp >> 4);
319
320		ec->rx_1 += -(ec->rx_1 >> DC_LOG2BETA) + tmp - ec->rx_2;
321
322		/*
323		 * hard limit filter to prevent clipping.  Note that at this
324		 * stage rx should be limited to +/- 16383 due to right shift
325		 * above
326		 */
327		tmp1 = ec->rx_1 >> 15;
328		if (tmp1 > 16383)
329			tmp1 = 16383;
330		if (tmp1 < -16383)
331			tmp1 = -16383;
332		rx = tmp1;
333		ec->rx_2 = tmp;
334	}
335
336	/* Block average of power in the filter states.  Used for
337	   adaption power calculation. */
338
339	{
340		int new, old;
341
342		/* efficient "out with the old and in with the new" algorithm so
343		   we don't have to recalculate over the whole block of
344		   samples. */
345		new = (int)tx * (int)tx;
346		old = (int)ec->fir_state.history[ec->fir_state.curr_pos] *
347		    (int)ec->fir_state.history[ec->fir_state.curr_pos];
348		ec->pstates +=
349		    ((new - old) + (1 << (ec->log2taps - 1))) >> ec->log2taps;
350		if (ec->pstates < 0)
351			ec->pstates = 0;
352	}
353
354	/* Calculate short term average levels using simple single pole IIRs */
355
356	ec->ltxacc += abs(tx) - ec->ltx;
357	ec->ltx = (ec->ltxacc + (1 << 4)) >> 5;
358	ec->lrxacc += abs(rx) - ec->lrx;
359	ec->lrx = (ec->lrxacc + (1 << 4)) >> 5;
360
361	/* Foreground filter */
362
363	ec->fir_state.coeffs = ec->fir_taps16[0];
364	echo_value = fir16(&ec->fir_state, tx);
365	ec->clean = rx - echo_value;
366	ec->lcleanacc += abs(ec->clean) - ec->lclean;
367	ec->lclean = (ec->lcleanacc + (1 << 4)) >> 5;
368
369	/* Background filter */
370
371	echo_value = fir16(&ec->fir_state_bg, tx);
372	clean_bg = rx - echo_value;
373	ec->lclean_bgacc += abs(clean_bg) - ec->lclean_bg;
374	ec->lclean_bg = (ec->lclean_bgacc + (1 << 4)) >> 5;
375
376	/* Background Filter adaption */
377
378	/* Almost always adap bg filter, just simple DT and energy
379	   detection to minimise adaption in cases of strong double talk.
380	   However this is not critical for the dual path algorithm.
381	 */
382	ec->factor = 0;
383	ec->shift = 0;
384	if ((ec->nonupdate_dwell == 0)) {
385		int p, logp, shift;
386
387		/* Determine:
388
389		   f = Beta * clean_bg_rx/P ------ (1)
390
391		   where P is the total power in the filter states.
392
393		   The Boffins have shown that if we obey (1) we converge
394		   quickly and avoid instability.
395
396		   The correct factor f must be in Q30, as this is the fixed
397		   point format required by the lms_adapt_bg() function,
398		   therefore the scaled version of (1) is:
399
400		   (2^30) * f  = (2^30) * Beta * clean_bg_rx/P
401		   factor      = (2^30) * Beta * clean_bg_rx/P     ----- (2)
402
403		   We have chosen Beta = 0.25 by experiment, so:
404
405		   factor      = (2^30) * (2^-2) * clean_bg_rx/P
406
407		   (30 - 2 - log2(P))
408		   factor      = clean_bg_rx 2                     ----- (3)
409
410		   To avoid a divide we approximate log2(P) as top_bit(P),
411		   which returns the position of the highest non-zero bit in
412		   P.  This approximation introduces an error as large as a
413		   factor of 2, but the algorithm seems to handle it OK.
414
415		   Come to think of it a divide may not be a big deal on a
416		   modern DSP, so its probably worth checking out the cycles
417		   for a divide versus a top_bit() implementation.
418		 */
419
420		p = MIN_TX_POWER_FOR_ADAPTION + ec->pstates;
421		logp = top_bit(p) + ec->log2taps;
422		shift = 30 - 2 - logp;
423		ec->shift = shift;
424
425		lms_adapt_bg(ec, clean_bg, shift);
426	}
427
428	/* very simple DTD to make sure we dont try and adapt with strong
429	   near end speech */
430
431	ec->adapt = 0;
432	if ((ec->lrx > MIN_RX_POWER_FOR_ADAPTION) && (ec->lrx > ec->ltx))
433		ec->nonupdate_dwell = DTD_HANGOVER;
434	if (ec->nonupdate_dwell)
435		ec->nonupdate_dwell--;
436
437	/* Transfer logic */
438
439	/* These conditions are from the dual path paper [1], I messed with
440	   them a bit to improve performance. */
441
442	if ((ec->adaption_mode & ECHO_CAN_USE_ADAPTION) &&
443	    (ec->nonupdate_dwell == 0) &&
444	    /* (ec->Lclean_bg < 0.875*ec->Lclean) */
445	    (8 * ec->lclean_bg < 7 * ec->lclean) &&
446	    /* (ec->Lclean_bg < 0.125*ec->Ltx) */
447	    (8 * ec->lclean_bg < ec->ltx)) {
448		if (ec->cond_met == 6) {
449			/*
450			 * BG filter has had better results for 6 consecutive
451			 * samples
452			 */
453			ec->adapt = 1;
454			memcpy(ec->fir_taps16[0], ec->fir_taps16[1],
455			       ec->taps * sizeof(int16_t));
456		} else
457			ec->cond_met++;
458	} else
459		ec->cond_met = 0;
460
461	/* Non-Linear Processing */
462
463	ec->clean_nlp = ec->clean;
464	if (ec->adaption_mode & ECHO_CAN_USE_NLP) {
465		/*
466		 * Non-linear processor - a fancy way to say "zap small
467		 * signals, to avoid residual echo due to (uLaw/ALaw)
468		 * non-linearity in the channel.".
469		 */
470
471		if ((16 * ec->lclean < ec->ltx)) {
472			/*
473			 * Our e/c has improved echo by at least 24 dB (each
474			 * factor of 2 is 6dB, so 2*2*2*2=16 is the same as
475			 * 6+6+6+6=24dB)
476			 */
477			if (ec->adaption_mode & ECHO_CAN_USE_CNG) {
478				ec->cng_level = ec->lbgn;
479
480				/*
481				 * Very elementary comfort noise generation.
482				 * Just random numbers rolled off very vaguely
483				 * Hoth-like.  DR: This noise doesn't sound
484				 * quite right to me - I suspect there are some
485				 * overflow issues in the filtering as it's too
486				 * "crackly".
487				 * TODO: debug this, maybe just play noise at
488				 * high level or look at spectrum.
489				 */
490
491				ec->cng_rndnum =
492				    1664525U * ec->cng_rndnum + 1013904223U;
493				ec->cng_filter =
494				    ((ec->cng_rndnum & 0xFFFF) - 32768 +
495				     5 * ec->cng_filter) >> 3;
496				ec->clean_nlp =
497				    (ec->cng_filter * ec->cng_level * 8) >> 14;
498
499			} else if (ec->adaption_mode & ECHO_CAN_USE_CLIP) {
500				/* This sounds much better than CNG */
501				if (ec->clean_nlp > ec->lbgn)
502					ec->clean_nlp = ec->lbgn;
503				if (ec->clean_nlp < -ec->lbgn)
504					ec->clean_nlp = -ec->lbgn;
505			} else {
506				/*
507				 * just mute the residual, doesn't sound very
508				 * good, used mainly in G168 tests
509				 */
510				ec->clean_nlp = 0;
511			}
512		} else {
513			/*
514			 * Background noise estimator.  I tried a few
515			 * algorithms here without much luck.  This very simple
516			 * one seems to work best, we just average the level
517			 * using a slow (1 sec time const) filter if the
518			 * current level is less than a (experimentally
519			 * derived) constant.  This means we dont include high
520			 * level signals like near end speech.  When combined
521			 * with CNG or especially CLIP seems to work OK.
522			 */
523			if (ec->lclean < 40) {
524				ec->lbgn_acc += abs(ec->clean) - ec->lbgn;
525				ec->lbgn = (ec->lbgn_acc + (1 << 11)) >> 12;
526			}
527		}
528	}
529
530	/* Roll around the taps buffer */
531	if (ec->curr_pos <= 0)
532		ec->curr_pos = ec->taps;
533	ec->curr_pos--;
534
535	if (ec->adaption_mode & ECHO_CAN_DISABLE)
536		ec->clean_nlp = rx;
537
538	/* Output scaled back up again to match input scaling */
539
540	return (int16_t) ec->clean_nlp << 1;
541}
542EXPORT_SYMBOL_GPL(oslec_update);
543
544/* This function is separated from the echo canceller is it is usually called
545   as part of the tx process.  See rx HP (DC blocking) filter above, it's
546   the same design.
547
548   Some soft phones send speech signals with a lot of low frequency
549   energy, e.g. down to 20Hz.  This can make the hybrid non-linear
550   which causes the echo canceller to fall over.  This filter can help
551   by removing any low frequency before it gets to the tx port of the
552   hybrid.
553
554   It can also help by removing and DC in the tx signal.  DC is bad
555   for LMS algorithms.
556
557   This is one of the classic DC removal filters, adjusted to provide
558   sufficient bass rolloff to meet the above requirement to protect hybrids
559   from things that upset them. The difference between successive samples
560   produces a lousy HPF, and then a suitably placed pole flattens things out.
561   The final result is a nicely rolled off bass end. The filtering is
562   implemented with extended fractional precision, which noise shapes things,
563   giving very clean DC removal.
564*/
565
566int16_t oslec_hpf_tx(struct oslec_state *ec, int16_t tx)
567{
568	int tmp;
569	int tmp1;
570
571	if (ec->adaption_mode & ECHO_CAN_USE_TX_HPF) {
572		tmp = tx << 15;
573
574		/*
575		 * Make sure the gain of the HPF is 1.0. The first can still
576		 * saturate a little under impulse conditions, and it might
577		 * roll to 32768 and need clipping on sustained peak level
578		 * signals. However, the scale of such clipping is small, and
579		 * the error due to any saturation should not markedly affect
580		 * the downstream processing.
581		 */
582		tmp -= (tmp >> 4);
583
584		ec->tx_1 += -(ec->tx_1 >> DC_LOG2BETA) + tmp - ec->tx_2;
585		tmp1 = ec->tx_1 >> 15;
586		if (tmp1 > 32767)
587			tmp1 = 32767;
588		if (tmp1 < -32767)
589			tmp1 = -32767;
590		tx = tmp1;
591		ec->tx_2 = tmp;
592	}
593
594	return tx;
595}
596EXPORT_SYMBOL_GPL(oslec_hpf_tx);
597
598MODULE_LICENSE("GPL");
599MODULE_AUTHOR("David Rowe");
600MODULE_DESCRIPTION("Open Source Line Echo Canceller");
601MODULE_VERSION("0.3.0");