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
lms_adapt_bg(struct oslec_state * ec,int clean,int shift)118 static 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
top_bit(unsigned int bits)147 static 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
oslec_create(int len,int adaption_mode)155 struct 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
207 error_snap:
208 fir16_free(&ec->fir_state_bg);
209 error_state_bg:
210 fir16_free(&ec->fir_state);
211 error_state:
212 kfree(ec->fir_taps16[1]);
213 error_oom_1:
214 kfree(ec->fir_taps16[0]);
215 error_oom_0:
216 kfree(ec);
217 return NULL;
218 }
219 EXPORT_SYMBOL_GPL(oslec_create);
220
oslec_free(struct oslec_state * ec)221 void 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 }
232 EXPORT_SYMBOL_GPL(oslec_free);
233
oslec_adaption_mode(struct oslec_state * ec,int adaption_mode)234 void oslec_adaption_mode(struct oslec_state *ec, int adaption_mode)
235 {
236 ec->adaption_mode = adaption_mode;
237 }
238 EXPORT_SYMBOL_GPL(oslec_adaption_mode);
239
oslec_flush(struct oslec_state * ec)240 void 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 }
264 EXPORT_SYMBOL_GPL(oslec_flush);
265
oslec_snapshot(struct oslec_state * ec)266 void oslec_snapshot(struct oslec_state *ec)
267 {
268 memcpy(ec->snapshot, ec->fir_taps16[0], ec->taps * sizeof(int16_t));
269 }
270 EXPORT_SYMBOL_GPL(oslec_snapshot);
271
272 /* Dual Path Echo Canceller */
273
oslec_update(struct oslec_state * ec,int16_t tx,int16_t rx)274 int16_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) {
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 }
542 EXPORT_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
oslec_hpf_tx(struct oslec_state * ec,int16_t tx)566 int16_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 }
596 EXPORT_SYMBOL_GPL(oslec_hpf_tx);
597
598 MODULE_LICENSE("GPL");
599 MODULE_AUTHOR("David Rowe");
600 MODULE_DESCRIPTION("Open Source Line Echo Canceller");
601 MODULE_VERSION("0.3.0");
602