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Diffstat (limited to 'drivers/misc/echo/echo.c')
| -rw-r--r-- | drivers/misc/echo/echo.c | 589 |
1 files changed, 0 insertions, 589 deletions
diff --git a/drivers/misc/echo/echo.c b/drivers/misc/echo/echo.c deleted file mode 100644 index 3c4eaba86576..000000000000 --- a/drivers/misc/echo/echo.c +++ /dev/null @@ -1,589 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * SpanDSP - a series of DSP components for telephony - * - * echo.c - A line echo canceller. This code is being developed - * against and partially complies with G168. - * - * Written by Steve Underwood <steveu@coppice.org> - * and David Rowe <david_at_rowetel_dot_com> - * - * Copyright (C) 2001, 2003 Steve Underwood, 2007 David Rowe - * - * Based on a bit from here, a bit from there, eye of toad, ear of - * bat, 15 years of failed attempts by David and a few fried brain - * cells. - * - * All rights reserved. - */ - -/*! \file */ - -/* Implementation Notes - David Rowe - April 2007 - - This code started life as Steve's NLMS algorithm with a tap - rotation algorithm to handle divergence during double talk. I - added a Geigel Double Talk Detector (DTD) [2] and performed some - G168 tests. However I had trouble meeting the G168 requirements, - especially for double talk - there were always cases where my DTD - failed, for example where near end speech was under the 6dB - threshold required for declaring double talk. - - So I tried a two path algorithm [1], which has so far given better - results. The original tap rotation/Geigel algorithm is available - in SVN http://svn.rowetel.com/software/oslec/tags/before_16bit. - It's probably possible to make it work if some one wants to put some - serious work into it. - - At present no special treatment is provided for tones, which - generally cause NLMS algorithms to diverge. Initial runs of a - subset of the G168 tests for tones (e.g ./echo_test 6) show the - current algorithm is passing OK, which is kind of surprising. The - full set of tests needs to be performed to confirm this result. - - One other interesting change is that I have managed to get the NLMS - code to work with 16 bit coefficients, rather than the original 32 - bit coefficents. This reduces the MIPs and storage required. - I evaulated the 16 bit port using g168_tests.sh and listening tests - on 4 real-world samples. - - I also attempted the implementation of a block based NLMS update - [2] but although this passes g168_tests.sh it didn't converge well - on the real-world samples. I have no idea why, perhaps a scaling - problem. The block based code is also available in SVN - http://svn.rowetel.com/software/oslec/tags/before_16bit. If this - code can be debugged, it will lead to further reduction in MIPS, as - the block update code maps nicely onto DSP instruction sets (it's a - dot product) compared to the current sample-by-sample update. - - Steve also has some nice notes on echo cancellers in echo.h - - References: - - [1] Ochiai, Areseki, and Ogihara, "Echo Canceller with Two Echo - Path Models", IEEE Transactions on communications, COM-25, - No. 6, June - 1977. - https://www.rowetel.com/images/echo/dual_path_paper.pdf - - [2] The classic, very useful paper that tells you how to - actually build a real world echo canceller: - Messerschmitt, Hedberg, Cole, Haoui, Winship, "Digital Voice - Echo Canceller with a TMS320020, - https://www.rowetel.com/images/echo/spra129.pdf - - [3] I have written a series of blog posts on this work, here is - Part 1: http://www.rowetel.com/blog/?p=18 - - [4] The source code http://svn.rowetel.com/software/oslec/ - - [5] A nice reference on LMS filters: - https://en.wikipedia.org/wiki/Least_mean_squares_filter - - Credits: - - Thanks to Steve Underwood, Jean-Marc Valin, and Ramakrishnan - Muthukrishnan for their suggestions and email discussions. Thanks - also to those people who collected echo samples for me such as - Mark, Pawel, and Pavel. -*/ - -#include <linux/kernel.h> -#include <linux/module.h> -#include <linux/slab.h> - -#include "echo.h" - -#define MIN_TX_POWER_FOR_ADAPTION 64 -#define MIN_RX_POWER_FOR_ADAPTION 64 -#define DTD_HANGOVER 600 /* 600 samples, or 75ms */ -#define DC_LOG2BETA 3 /* log2() of DC filter Beta */ - -/* adapting coeffs using the traditional stochastic descent (N)LMS algorithm */ - -static inline void lms_adapt_bg(struct oslec_state *ec, int clean, int shift) -{ - int i; - - int offset1; - int offset2; - int factor; - int exp; - - if (shift > 0) - factor = clean << shift; - else - factor = clean >> -shift; - - /* Update the FIR taps */ - - offset2 = ec->curr_pos; - offset1 = ec->taps - offset2; - - for (i = ec->taps - 1; i >= offset1; i--) { - exp = (ec->fir_state_bg.history[i - offset1] * factor); - ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15); - } - for (; i >= 0; i--) { - exp = (ec->fir_state_bg.history[i + offset2] * factor); - ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15); - } -} - -static inline int top_bit(unsigned int bits) -{ - if (bits == 0) - return -1; - else - return (int)fls((int32_t) bits) - 1; -} - -struct oslec_state *oslec_create(int len, int adaption_mode) -{ - struct oslec_state *ec; - int i; - const int16_t *history; - - ec = kzalloc(sizeof(*ec), GFP_KERNEL); - if (!ec) - return NULL; - - ec->taps = len; - ec->log2taps = top_bit(len); - ec->curr_pos = ec->taps - 1; - - ec->fir_taps16[0] = - kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL); - if (!ec->fir_taps16[0]) - goto error_oom_0; - - ec->fir_taps16[1] = - kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL); - if (!ec->fir_taps16[1]) - goto error_oom_1; - - history = fir16_create(&ec->fir_state, ec->fir_taps16[0], ec->taps); - if (!history) - goto error_state; - history = fir16_create(&ec->fir_state_bg, ec->fir_taps16[1], ec->taps); - if (!history) - goto error_state_bg; - - for (i = 0; i < 5; i++) - ec->xvtx[i] = ec->yvtx[i] = ec->xvrx[i] = ec->yvrx[i] = 0; - - ec->cng_level = 1000; - oslec_adaption_mode(ec, adaption_mode); - - ec->snapshot = kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL); - if (!ec->snapshot) - goto error_snap; - - ec->cond_met = 0; - ec->pstates = 0; - ec->ltxacc = ec->lrxacc = ec->lcleanacc = ec->lclean_bgacc = 0; - ec->ltx = ec->lrx = ec->lclean = ec->lclean_bg = 0; - ec->tx_1 = ec->tx_2 = ec->rx_1 = ec->rx_2 = 0; - ec->lbgn = ec->lbgn_acc = 0; - ec->lbgn_upper = 200; - ec->lbgn_upper_acc = ec->lbgn_upper << 13; - - return ec; - -error_snap: - fir16_free(&ec->fir_state_bg); -error_state_bg: - fir16_free(&ec->fir_state); -error_state: - kfree(ec->fir_taps16[1]); -error_oom_1: - kfree(ec->fir_taps16[0]); -error_oom_0: - kfree(ec); - return NULL; -} -EXPORT_SYMBOL_GPL(oslec_create); - -void oslec_free(struct oslec_state *ec) -{ - int i; - - fir16_free(&ec->fir_state); - fir16_free(&ec->fir_state_bg); - for (i = 0; i < 2; i++) - kfree(ec->fir_taps16[i]); - kfree(ec->snapshot); - kfree(ec); -} -EXPORT_SYMBOL_GPL(oslec_free); - -void oslec_adaption_mode(struct oslec_state *ec, int adaption_mode) -{ - ec->adaption_mode = adaption_mode; -} -EXPORT_SYMBOL_GPL(oslec_adaption_mode); - -void oslec_flush(struct oslec_state *ec) -{ - int i; - - ec->ltxacc = ec->lrxacc = ec->lcleanacc = ec->lclean_bgacc = 0; - ec->ltx = ec->lrx = ec->lclean = ec->lclean_bg = 0; - ec->tx_1 = ec->tx_2 = ec->rx_1 = ec->rx_2 = 0; - - ec->lbgn = ec->lbgn_acc = 0; - ec->lbgn_upper = 200; - ec->lbgn_upper_acc = ec->lbgn_upper << 13; - - ec->nonupdate_dwell = 0; - - fir16_flush(&ec->fir_state); - fir16_flush(&ec->fir_state_bg); - ec->fir_state.curr_pos = ec->taps - 1; - ec->fir_state_bg.curr_pos = ec->taps - 1; - for (i = 0; i < 2; i++) - memset(ec->fir_taps16[i], 0, ec->taps * sizeof(int16_t)); - - ec->curr_pos = ec->taps - 1; - ec->pstates = 0; -} -EXPORT_SYMBOL_GPL(oslec_flush); - -void oslec_snapshot(struct oslec_state *ec) -{ - memcpy(ec->snapshot, ec->fir_taps16[0], ec->taps * sizeof(int16_t)); -} -EXPORT_SYMBOL_GPL(oslec_snapshot); - -/* Dual Path Echo Canceller */ - -int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx) -{ - int32_t echo_value; - int clean_bg; - int tmp; - int tmp1; - - /* - * Input scaling was found be required to prevent problems when tx - * starts clipping. Another possible way to handle this would be the - * filter coefficent scaling. - */ - - ec->tx = tx; - ec->rx = rx; - tx >>= 1; - rx >>= 1; - - /* - * Filter DC, 3dB point is 160Hz (I think), note 32 bit precision - * required otherwise values do not track down to 0. Zero at DC, Pole - * at (1-Beta) on real axis. Some chip sets (like Si labs) don't - * need this, but something like a $10 X100P card does. Any DC really - * slows down convergence. - * - * Note: removes some low frequency from the signal, this reduces the - * speech quality when listening to samples through headphones but may - * not be obvious through a telephone handset. - * - * Note that the 3dB frequency in radians is approx Beta, e.g. for Beta - * = 2^(-3) = 0.125, 3dB freq is 0.125 rads = 159Hz. - */ - - if (ec->adaption_mode & ECHO_CAN_USE_RX_HPF) { - tmp = rx << 15; - - /* - * Make sure the gain of the HPF is 1.0. This can still - * saturate a little under impulse conditions, and it might - * roll to 32768 and need clipping on sustained peak level - * signals. However, the scale of such clipping is small, and - * the error due to any saturation should not markedly affect - * the downstream processing. - */ - tmp -= (tmp >> 4); - - ec->rx_1 += -(ec->rx_1 >> DC_LOG2BETA) + tmp - ec->rx_2; - - /* - * hard limit filter to prevent clipping. Note that at this - * stage rx should be limited to +/- 16383 due to right shift - * above - */ - tmp1 = ec->rx_1 >> 15; - if (tmp1 > 16383) - tmp1 = 16383; - if (tmp1 < -16383) - tmp1 = -16383; - rx = tmp1; - ec->rx_2 = tmp; - } - - /* Block average of power in the filter states. Used for - adaption power calculation. */ - - { - int new, old; - - /* efficient "out with the old and in with the new" algorithm so - we don't have to recalculate over the whole block of - samples. */ - new = (int)tx * (int)tx; - old = (int)ec->fir_state.history[ec->fir_state.curr_pos] * - (int)ec->fir_state.history[ec->fir_state.curr_pos]; - ec->pstates += - ((new - old) + (1 << (ec->log2taps - 1))) >> ec->log2taps; - if (ec->pstates < 0) - ec->pstates = 0; - } - - /* Calculate short term average levels using simple single pole IIRs */ - - ec->ltxacc += abs(tx) - ec->ltx; - ec->ltx = (ec->ltxacc + (1 << 4)) >> 5; - ec->lrxacc += abs(rx) - ec->lrx; - ec->lrx = (ec->lrxacc + (1 << 4)) >> 5; - - /* Foreground filter */ - - ec->fir_state.coeffs = ec->fir_taps16[0]; - echo_value = fir16(&ec->fir_state, tx); - ec->clean = rx - echo_value; - ec->lcleanacc += abs(ec->clean) - ec->lclean; - ec->lclean = (ec->lcleanacc + (1 << 4)) >> 5; - - /* Background filter */ - - echo_value = fir16(&ec->fir_state_bg, tx); - clean_bg = rx - echo_value; - ec->lclean_bgacc += abs(clean_bg) - ec->lclean_bg; - ec->lclean_bg = (ec->lclean_bgacc + (1 << 4)) >> 5; - - /* Background Filter adaption */ - - /* Almost always adap bg filter, just simple DT and energy - detection to minimise adaption in cases of strong double talk. - However this is not critical for the dual path algorithm. - */ - ec->factor = 0; - ec->shift = 0; - if (!ec->nonupdate_dwell) { - int p, logp, shift; - - /* Determine: - - f = Beta * clean_bg_rx/P ------ (1) - - where P is the total power in the filter states. - - The Boffins have shown that if we obey (1) we converge - quickly and avoid instability. - - The correct factor f must be in Q30, as this is the fixed - point format required by the lms_adapt_bg() function, - therefore the scaled version of (1) is: - - (2^30) * f = (2^30) * Beta * clean_bg_rx/P - factor = (2^30) * Beta * clean_bg_rx/P ----- (2) - - We have chosen Beta = 0.25 by experiment, so: - - factor = (2^30) * (2^-2) * clean_bg_rx/P - - (30 - 2 - log2(P)) - factor = clean_bg_rx 2 ----- (3) - - To avoid a divide we approximate log2(P) as top_bit(P), - which returns the position of the highest non-zero bit in - P. This approximation introduces an error as large as a - factor of 2, but the algorithm seems to handle it OK. - - Come to think of it a divide may not be a big deal on a - modern DSP, so its probably worth checking out the cycles - for a divide versus a top_bit() implementation. - */ - - p = MIN_TX_POWER_FOR_ADAPTION + ec->pstates; - logp = top_bit(p) + ec->log2taps; - shift = 30 - 2 - logp; - ec->shift = shift; - - lms_adapt_bg(ec, clean_bg, shift); - } - - /* very simple DTD to make sure we dont try and adapt with strong - near end speech */ - - ec->adapt = 0; - if ((ec->lrx > MIN_RX_POWER_FOR_ADAPTION) && (ec->lrx > ec->ltx)) - ec->nonupdate_dwell = DTD_HANGOVER; - if (ec->nonupdate_dwell) - ec->nonupdate_dwell--; - - /* Transfer logic */ - - /* These conditions are from the dual path paper [1], I messed with - them a bit to improve performance. */ - - if ((ec->adaption_mode & ECHO_CAN_USE_ADAPTION) && - (ec->nonupdate_dwell == 0) && - /* (ec->Lclean_bg < 0.875*ec->Lclean) */ - (8 * ec->lclean_bg < 7 * ec->lclean) && - /* (ec->Lclean_bg < 0.125*ec->Ltx) */ - (8 * ec->lclean_bg < ec->ltx)) { - if (ec->cond_met == 6) { - /* - * BG filter has had better results for 6 consecutive - * samples - */ - ec->adapt = 1; - memcpy(ec->fir_taps16[0], ec->fir_taps16[1], - ec->taps * sizeof(int16_t)); - } else - ec->cond_met++; - } else - ec->cond_met = 0; - - /* Non-Linear Processing */ - - ec->clean_nlp = ec->clean; - if (ec->adaption_mode & ECHO_CAN_USE_NLP) { - /* - * Non-linear processor - a fancy way to say "zap small - * signals, to avoid residual echo due to (uLaw/ALaw) - * non-linearity in the channel.". - */ - - if ((16 * ec->lclean < ec->ltx)) { - /* - * Our e/c has improved echo by at least 24 dB (each - * factor of 2 is 6dB, so 2*2*2*2=16 is the same as - * 6+6+6+6=24dB) - */ - if (ec->adaption_mode & ECHO_CAN_USE_CNG) { - ec->cng_level = ec->lbgn; - - /* - * Very elementary comfort noise generation. - * Just random numbers rolled off very vaguely - * Hoth-like. DR: This noise doesn't sound - * quite right to me - I suspect there are some - * overflow issues in the filtering as it's too - * "crackly". - * TODO: debug this, maybe just play noise at - * high level or look at spectrum. - */ - - ec->cng_rndnum = - 1664525U * ec->cng_rndnum + 1013904223U; - ec->cng_filter = - ((ec->cng_rndnum & 0xFFFF) - 32768 + - 5 * ec->cng_filter) >> 3; - ec->clean_nlp = - (ec->cng_filter * ec->cng_level * 8) >> 14; - - } else if (ec->adaption_mode & ECHO_CAN_USE_CLIP) { - /* This sounds much better than CNG */ - if (ec->clean_nlp > ec->lbgn) - ec->clean_nlp = ec->lbgn; - if (ec->clean_nlp < -ec->lbgn) - ec->clean_nlp = -ec->lbgn; - } else { - /* - * just mute the residual, doesn't sound very - * good, used mainly in G168 tests - */ - ec->clean_nlp = 0; - } - } else { - /* - * Background noise estimator. I tried a few - * algorithms here without much luck. This very simple - * one seems to work best, we just average the level - * using a slow (1 sec time const) filter if the - * current level is less than a (experimentally - * derived) constant. This means we dont include high - * level signals like near end speech. When combined - * with CNG or especially CLIP seems to work OK. - */ - if (ec->lclean < 40) { - ec->lbgn_acc += abs(ec->clean) - ec->lbgn; - ec->lbgn = (ec->lbgn_acc + (1 << 11)) >> 12; - } - } - } - - /* Roll around the taps buffer */ - if (ec->curr_pos <= 0) - ec->curr_pos = ec->taps; - ec->curr_pos--; - - if (ec->adaption_mode & ECHO_CAN_DISABLE) - ec->clean_nlp = rx; - - /* Output scaled back up again to match input scaling */ - - return (int16_t) ec->clean_nlp << 1; -} -EXPORT_SYMBOL_GPL(oslec_update); - -/* This function is separated from the echo canceller is it is usually called - as part of the tx process. See rx HP (DC blocking) filter above, it's - the same design. - - Some soft phones send speech signals with a lot of low frequency - energy, e.g. down to 20Hz. This can make the hybrid non-linear - which causes the echo canceller to fall over. This filter can help - by removing any low frequency before it gets to the tx port of the - hybrid. - - It can also help by removing and DC in the tx signal. DC is bad - for LMS algorithms. - - This is one of the classic DC removal filters, adjusted to provide - sufficient bass rolloff to meet the above requirement to protect hybrids - from things that upset them. The difference between successive samples - produces a lousy HPF, and then a suitably placed pole flattens things out. - The final result is a nicely rolled off bass end. The filtering is - implemented with extended fractional precision, which noise shapes things, - giving very clean DC removal. -*/ - -int16_t oslec_hpf_tx(struct oslec_state *ec, int16_t tx) -{ - int tmp; - int tmp1; - - if (ec->adaption_mode & ECHO_CAN_USE_TX_HPF) { - tmp = tx << 15; - - /* - * Make sure the gain of the HPF is 1.0. The first can still - * saturate a little under impulse conditions, and it might - * roll to 32768 and need clipping on sustained peak level - * signals. However, the scale of such clipping is small, and - * the error due to any saturation should not markedly affect - * the downstream processing. - */ - tmp -= (tmp >> 4); - - ec->tx_1 += -(ec->tx_1 >> DC_LOG2BETA) + tmp - ec->tx_2; - tmp1 = ec->tx_1 >> 15; - if (tmp1 > 32767) - tmp1 = 32767; - if (tmp1 < -32767) - tmp1 = -32767; - tx = tmp1; - ec->tx_2 = tmp; - } - - return tx; -} -EXPORT_SYMBOL_GPL(oslec_hpf_tx); - -MODULE_LICENSE("GPL"); -MODULE_AUTHOR("David Rowe"); -MODULE_DESCRIPTION("Open Source Line Echo Canceller"); -MODULE_VERSION("0.3.0"); |