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 */ 105 106 static 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 } 134 135 static 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 } 142 143 struct 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; 194 195 error_snap: 196 fir16_free(&ec->fir_state_bg); 197 error_state_bg: 198 fir16_free(&ec->fir_state); 199 error_state: 200 kfree(ec->fir_taps16[1]); 201 error_oom_1: 202 kfree(ec->fir_taps16[0]); 203 error_oom_0: 204 kfree(ec); 205 return NULL; 206 } 207 EXPORT_SYMBOL_GPL(oslec_create); 208 209 void 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 } 220 EXPORT_SYMBOL_GPL(oslec_free); 221 222 void oslec_adaption_mode(struct oslec_state *ec, int adaption_mode) 223 { 224 ec->adaption_mode = adaption_mode; 225 } 226 EXPORT_SYMBOL_GPL(oslec_adaption_mode); 227 228 void 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 } 252 EXPORT_SYMBOL_GPL(oslec_flush); 253 254 void oslec_snapshot(struct oslec_state *ec) 255 { 256 memcpy(ec->snapshot, ec->fir_taps16[0], ec->taps * sizeof(int16_t)); 257 } 258 EXPORT_SYMBOL_GPL(oslec_snapshot); 259 260 /* Dual Path Echo Canceller */ 261 262 int16_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 } 530 EXPORT_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 */ 553 554 int16_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 } 584 EXPORT_SYMBOL_GPL(oslec_hpf_tx); 585 586 MODULE_LICENSE("GPL"); 587 MODULE_AUTHOR("David Rowe"); 588 MODULE_DESCRIPTION("Open Source Line Echo Canceller"); 589 MODULE_VERSION("0.3.0"); 590