diff options
Diffstat (limited to 'drivers/staging/echo')
| -rw-r--r-- | drivers/staging/echo/Kconfig | 9 | ||||
| -rw-r--r-- | drivers/staging/echo/Makefile | 1 | ||||
| -rw-r--r-- | drivers/staging/echo/TODO | 5 | ||||
| -rw-r--r-- | drivers/staging/echo/echo.c | 674 | ||||
| -rw-r--r-- | drivers/staging/echo/echo.h | 187 | ||||
| -rw-r--r-- | drivers/staging/echo/fir.h | 216 | ||||
| -rw-r--r-- | drivers/staging/echo/oslec.h | 94 |
7 files changed, 0 insertions, 1186 deletions
diff --git a/drivers/staging/echo/Kconfig b/drivers/staging/echo/Kconfig deleted file mode 100644 index f1d41ea9cd4..00000000000 --- a/drivers/staging/echo/Kconfig +++ /dev/null @@ -1,9 +0,0 @@ -config ECHO - tristate "Line Echo Canceller support" - default n - ---help--- - This driver provides line echo cancelling support for mISDN and - Zaptel drivers. - - To compile this driver as a module, choose M here. The module - will be called echo. diff --git a/drivers/staging/echo/Makefile b/drivers/staging/echo/Makefile deleted file mode 100644 index 7d4caac12a8..00000000000 --- a/drivers/staging/echo/Makefile +++ /dev/null @@ -1 +0,0 @@ -obj-$(CONFIG_ECHO) += echo.o diff --git a/drivers/staging/echo/TODO b/drivers/staging/echo/TODO deleted file mode 100644 index 72a311a5a9c..00000000000 --- a/drivers/staging/echo/TODO +++ /dev/null @@ -1,5 +0,0 @@ -TODO: - - send to lkml for review - -Please send patches to Greg Kroah-Hartman <greg@kroah.com> and Cc: Steve -Underwood <steveu@coppice.org> and David Rowe <david@rowetel.com> diff --git a/drivers/staging/echo/echo.c b/drivers/staging/echo/echo.c deleted file mode 100644 index 9597e9523ca..00000000000 --- a/drivers/staging/echo/echo.c +++ /dev/null @@ -1,674 +0,0 @@ -/* - * 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. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2, as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ - -/*! \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. - http://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, - http://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: - http://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 */ - -#ifdef __bfin__ -static inline void lms_adapt_bg(struct oslec_state *ec, int clean, int shift) -{ - int i; - int offset1; - int offset2; - int factor; - int exp; - int16_t *phist; - int n; - - if (shift > 0) - factor = clean << shift; - else - factor = clean >> -shift; - - /* Update the FIR taps */ - - offset2 = ec->curr_pos; - offset1 = ec->taps - offset2; - phist = &ec->fir_state_bg.history[offset2]; - - /* st: and en: help us locate the assembler in echo.s */ - - /* asm("st:"); */ - n = ec->taps; - for (i = 0; i < n; i++) { - exp = *phist++ * factor; - ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15); - } - /* asm("en:"); */ - - /* Note the asm for the inner loop above generated by Blackfin gcc - 4.1.1 is pretty good (note even parallel instructions used): - - R0 = W [P0++] (X); - R0 *= R2; - R0 = R0 + R3 (NS) || - R1 = W [P1] (X) || - nop; - R0 >>>= 15; - R0 = R0 + R1; - W [P1++] = R0; - - A block based update algorithm would be much faster but the - above can't be improved on much. Every instruction saved in - the loop above is 2 MIPs/ch! The for loop above is where the - Blackfin spends most of it's time - about 17 MIPs/ch measured - with speedtest.c with 256 taps (32ms). Write-back and - Write-through cache gave about the same performance. - */ -} - -/* - IDEAS for further optimisation of lms_adapt_bg(): - - 1/ The rounding is quite costly. Could we keep as 32 bit coeffs - then make filter pluck the MS 16-bits of the coeffs when filtering? - However this would lower potential optimisation of filter, as I - think the dual-MAC architecture requires packed 16 bit coeffs. - - 2/ Block based update would be more efficient, as per comments above, - could use dual MAC architecture. - - 3/ Look for same sample Blackfin LMS code, see if we can get dual-MAC - packing. - - 4/ Execute the whole e/c in a block of say 20ms rather than sample - by sample. Processing a few samples every ms is inefficient. -*/ - -#else -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); - } -} -#endif - -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 == 0)) { - 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"); diff --git a/drivers/staging/echo/echo.h b/drivers/staging/echo/echo.h deleted file mode 100644 index 9b08c63e636..00000000000 --- a/drivers/staging/echo/echo.h +++ /dev/null @@ -1,187 +0,0 @@ -/* - * 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 Steve Underwood and 2007 David Rowe - * - * All rights reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2, as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ - -#ifndef __ECHO_H -#define __ECHO_H - -/* -Line echo cancellation for voice - -What does it do? - -This module aims to provide G.168-2002 compliant echo cancellation, to remove -electrical echoes (e.g. from 2-4 wire hybrids) from voice calls. - -How does it work? - -The heart of the echo cancellor is FIR filter. This is adapted to match the -echo impulse response of the telephone line. It must be long enough to -adequately cover the duration of that impulse response. The signal transmitted -to the telephone line is passed through the FIR filter. Once the FIR is -properly adapted, the resulting output is an estimate of the echo signal -received from the line. This is subtracted from the received signal. The result -is an estimate of the signal which originated at the far end of the line, free -from echos of our own transmitted signal. - -The least mean squares (LMS) algorithm is attributed to Widrow and Hoff, and -was introduced in 1960. It is the commonest form of filter adaption used in -things like modem line equalisers and line echo cancellers. There it works very -well. However, it only works well for signals of constant amplitude. It works -very poorly for things like speech echo cancellation, where the signal level -varies widely. This is quite easy to fix. If the signal level is normalised - -similar to applying AGC - LMS can work as well for a signal of varying -amplitude as it does for a modem signal. This normalised least mean squares -(NLMS) algorithm is the commonest one used for speech echo cancellation. Many -other algorithms exist - e.g. RLS (essentially the same as Kalman filtering), -FAP, etc. Some perform significantly better than NLMS. However, factors such -as computational complexity and patents favour the use of NLMS. - -A simple refinement to NLMS can improve its performance with speech. NLMS tends -to adapt best to the strongest parts of a signal. If the signal is white noise, -the NLMS algorithm works very well. However, speech has more low frequency than -high frequency content. Pre-whitening (i.e. filtering the signal to flatten its -spectrum) the echo signal improves the adapt rate for speech, and ensures the -final residual signal is not heavily biased towards high frequencies. A very -low complexity filter is adequate for this, so pre-whitening adds little to the -compute requirements of the echo canceller. - -An FIR filter adapted using pre-whitened NLMS performs well, provided certain -conditions are met: - - - The transmitted signal has poor self-correlation. - - There is no signal being generated within the environment being - cancelled. - -The difficulty is that neither of these can be guaranteed. - -If the adaption is performed while transmitting noise (or something fairly -noise like, such as voice) the adaption works very well. If the adaption is -performed while transmitting something highly correlative (typically narrow -band energy such as signalling tones or DTMF), the adaption can go seriously -wrong. The reason is there is only one solution for the adaption on a near -random signal - the impulse response of the line. For a repetitive signal, -there are any number of solutions which converge the adaption, and nothing -guides the adaption to choose the generalised one. Allowing an untrained -canceller to converge on this kind of narrowband energy probably a good thing, -since at least it cancels the tones. Allowing a well converged canceller to -continue converging on such energy is just a way to ruin its generalised -adaption. A narrowband detector is needed, so adapation can be suspended at -appropriate times. - -The adaption process is based on trying to eliminate the received signal. When -there is any signal from within the environment being cancelled it may upset -the adaption process. Similarly, if the signal we are transmitting is small, -noise may dominate and disturb the adaption process. If we can ensure that the -adaption is only performed when we are transmitting a significant signal level, -and the environment is not, things will be OK. Clearly, it is easy to tell when -we are sending a significant signal. Telling, if the environment is generating -a significant signal, and doing it with sufficient speed that the adaption will -not have diverged too much more we stop it, is a little harder. - -The key problem in detecting when the environment is sourcing significant -energy is that we must do this very quickly. Given a reasonably long sample of -the received signal, there are a number of strategies which may be used to -assess whether that signal contains a strong far end component. However, by the -time that assessment is complete the far end signal will have already caused -major mis-convergence in the adaption process. An assessment algorithm is -needed which produces a fairly accurate result from a very short burst of far -end energy. - -How do I use it? - -The echo cancellor processes both the transmit and receive streams sample by -sample. The processing function is not declared inline. Unfortunately, -cancellation requires many operations per sample, so the call overhead is only -a minor burden. -*/ - -#include "fir.h" -#include "oslec.h" - -/* - G.168 echo canceller descriptor. This defines the working state for a line - echo canceller. -*/ -struct oslec_state { - int16_t tx; - int16_t rx; - int16_t clean; - int16_t clean_nlp; - - int nonupdate_dwell; - int curr_pos; - int taps; - int log2taps; - int adaption_mode; - - int cond_met; - int32_t pstates; - int16_t adapt; - int32_t factor; - int16_t shift; - - /* Average levels and averaging filter states */ - int ltxacc; - int lrxacc; - int lcleanacc; - int lclean_bgacc; - int ltx; - int lrx; - int lclean; - int lclean_bg; - int lbgn; - int lbgn_acc; - int lbgn_upper; - int lbgn_upper_acc; - - /* foreground and background filter states */ - struct fir16_state_t fir_state; - struct fir16_state_t fir_state_bg; - int16_t *fir_taps16[2]; - - /* DC blocking filter states */ - int tx_1; - int tx_2; - int rx_1; - int rx_2; - - /* optional High Pass Filter states */ - int32_t xvtx[5]; - int32_t yvtx[5]; - int32_t xvrx[5]; - int32_t yvrx[5]; - - /* Parameters for the optional Hoth noise generator */ - int cng_level; - int cng_rndnum; - int cng_filter; - - /* snapshot sample of coeffs used for development */ - int16_t *snapshot; -}; - -#endif /* __ECHO_H */ diff --git a/drivers/staging/echo/fir.h b/drivers/staging/echo/fir.h deleted file mode 100644 index 7b9fabf1fea..00000000000 --- a/drivers/staging/echo/fir.h +++ /dev/null @@ -1,216 +0,0 @@ -/* - * SpanDSP - a series of DSP components for telephony - * - * fir.h - General telephony FIR routines - * - * Written by Steve Underwood <steveu@coppice.org> - * - * Copyright (C) 2002 Steve Underwood - * - * All rights reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2, as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ - -#if !defined(_FIR_H_) -#define _FIR_H_ - -/* - Blackfin NOTES & IDEAS: - - A simple dot product function is used to implement the filter. This performs - just one MAC/cycle which is inefficient but was easy to implement as a first - pass. The current Blackfin code also uses an unrolled form of the filter - history to avoid 0 length hardware loop issues. This is wasteful of - memory. - - Ideas for improvement: - - 1/ Rewrite filter for dual MAC inner loop. The issue here is handling - history sample offsets that are 16 bit aligned - the dual MAC needs - 32 bit aligmnent. There are some good examples in libbfdsp. - - 2/ Use the hardware circular buffer facility tohalve memory usage. - - 3/ Consider using internal memory. - - Using less memory might also improve speed as cache misses will be - reduced. A drop in MIPs and memory approaching 50% should be - possible. - - The foreground and background filters currenlty use a total of - about 10 MIPs/ch as measured with speedtest.c on a 256 TAP echo - can. -*/ - -/* - * 16 bit integer FIR descriptor. This defines the working state for a single - * instance of an FIR filter using 16 bit integer coefficients. - */ -struct fir16_state_t { - int taps; - int curr_pos; - const int16_t *coeffs; - int16_t *history; -}; - -/* - * 32 bit integer FIR descriptor. This defines the working state for a single - * instance of an FIR filter using 32 bit integer coefficients, and filtering - * 16 bit integer data. - */ -struct fir32_state_t { - int taps; - int curr_pos; - const int32_t *coeffs; - int16_t *history; -}; - -/* - * Floating point FIR descriptor. This defines the working state for a single - * instance of an FIR filter using floating point coefficients and data. - */ -struct fir_float_state_t { - int taps; - int curr_pos; - const float *coeffs; - float *history; -}; - -static inline const int16_t *fir16_create(struct fir16_state_t *fir, - const int16_t *coeffs, int taps) -{ - fir->taps = taps; - fir->curr_pos = taps - 1; - fir->coeffs = coeffs; -#if defined(__bfin__) - fir->history = kcalloc(2 * taps, sizeof(int16_t), GFP_KERNEL); -#else - fir->history = kcalloc(taps, sizeof(int16_t), GFP_KERNEL); -#endif - return fir->history; -} - -static inline void fir16_flush(struct fir16_state_t *fir) -{ -#if defined(__bfin__) - memset(fir->history, 0, 2 * fir->taps * sizeof(int16_t)); -#else - memset(fir->history, 0, fir->taps * sizeof(int16_t)); -#endif -} - -static inline void fir16_free(struct fir16_state_t *fir) -{ - kfree(fir->history); -} - -#ifdef __bfin__ -static inline int32_t dot_asm(short *x, short *y, int len) -{ - int dot; - - len--; - - __asm__("I0 = %1;\n\t" - "I1 = %2;\n\t" - "A0 = 0;\n\t" - "R0.L = W[I0++] || R1.L = W[I1++];\n\t" - "LOOP dot%= LC0 = %3;\n\t" - "LOOP_BEGIN dot%=;\n\t" - "A0 += R0.L * R1.L (IS) || R0.L = W[I0++] || R1.L = W[I1++];\n\t" - "LOOP_END dot%=;\n\t" - "A0 += R0.L*R1.L (IS);\n\t" - "R0 = A0;\n\t" - "%0 = R0;\n\t" - : "=&d"(dot) - : "a"(x), "a"(y), "a"(len) - : "I0", "I1", "A1", "A0", "R0", "R1" - ); - - return dot; -} -#endif - -static inline int16_t fir16(struct fir16_state_t *fir, int16_t sample) -{ - int32_t y; -#if defined(__bfin__) - fir->history[fir->curr_pos] = sample; - fir->history[fir->curr_pos + fir->taps] = sample; - y = dot_asm((int16_t *) fir->coeffs, &fir->history[fir->curr_pos], - fir->taps); -#else - int i; - int offset1; - int offset2; - - fir->history[fir->curr_pos] = sample; - - offset2 = fir->curr_pos; - offset1 = fir->taps - offset2; - y = 0; - for (i = fir->taps - 1; i >= offset1; i--) - y += fir->coeffs[i] * fir->history[i - offset1]; - for (; i >= 0; i--) - y += fir->coeffs[i] * fir->history[i + offset2]; -#endif - if (fir->curr_pos <= 0) - fir->curr_pos = fir->taps; - fir->curr_pos--; - return (int16_t) (y >> 15); -} - -static inline const int16_t *fir32_create(struct fir32_state_t *fir, - const int32_t *coeffs, int taps) -{ - fir->taps = taps; - fir->curr_pos = taps - 1; - fir->coeffs = coeffs; - fir->history = kcalloc(taps, sizeof(int16_t), GFP_KERNEL); - return fir->history; -} - -static inline void fir32_flush(struct fir32_state_t *fir) -{ - memset(fir->history, 0, fir->taps * sizeof(int16_t)); -} - -static inline void fir32_free(struct fir32_state_t *fir) -{ - kfree(fir->history); -} - -static inline int16_t fir32(struct fir32_state_t *fir, int16_t sample) -{ - int i; - int32_t y; - int offset1; - int offset2; - - fir->history[fir->curr_pos] = sample; - offset2 = fir->curr_pos; - offset1 = fir->taps - offset2; - y = 0; - for (i = fir->taps - 1; i >= offset1; i--) - y += fir->coeffs[i] * fir->history[i - offset1]; - for (; i >= 0; i--) - y += fir->coeffs[i] * fir->history[i + offset2]; - if (fir->curr_pos <= 0) - fir->curr_pos = fir->taps; - fir->curr_pos--; - return (int16_t) (y >> 15); -} - -#endif diff --git a/drivers/staging/echo/oslec.h b/drivers/staging/echo/oslec.h deleted file mode 100644 index f4175360ce2..00000000000 --- a/drivers/staging/echo/oslec.h +++ /dev/null @@ -1,94 +0,0 @@ -/* - * OSLEC - A line echo canceller. This code is being developed - * against and partially complies with G168. Using code from SpanDSP - * - * Written by Steve Underwood <steveu@coppice.org> - * and David Rowe <david_at_rowetel_dot_com> - * - * Copyright (C) 2001 Steve Underwood and 2007-2008 David Rowe - * - * All rights reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2, as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - * - */ - -#ifndef __OSLEC_H -#define __OSLEC_H - -/* Mask bits for the adaption mode */ -#define ECHO_CAN_USE_ADAPTION 0x01 -#define ECHO_CAN_USE_NLP 0x02 -#define ECHO_CAN_USE_CNG 0x04 -#define ECHO_CAN_USE_CLIP 0x08 -#define ECHO_CAN_USE_TX_HPF 0x10 -#define ECHO_CAN_USE_RX_HPF 0x20 -#define ECHO_CAN_DISABLE 0x40 - -/** - * oslec_state: G.168 echo canceller descriptor. - * - * This defines the working state for a line echo canceller. - */ -struct oslec_state; - -/** - * oslec_create - Create a voice echo canceller context. - * @len: The length of the canceller, in samples. - * @return: The new canceller context, or NULL if the canceller could not be - * created. - */ -struct oslec_state *oslec_create(int len, int adaption_mode); - -/** - * oslec_free - Free a voice echo canceller context. - * @ec: The echo canceller context. - */ -void oslec_free(struct oslec_state *ec); - -/** - * oslec_flush - Flush (reinitialise) a voice echo canceller context. - * @ec: The echo canceller context. - */ -void oslec_flush(struct oslec_state *ec); - -/** - * oslec_adaption_mode - set the adaption mode of a voice echo canceller context. - * @ec The echo canceller context. - * @adaption_mode: The mode. - */ -void oslec_adaption_mode(struct oslec_state *ec, int adaption_mode); - -void oslec_snapshot(struct oslec_state *ec); - -/** - * oslec_update: Process a sample through a voice echo canceller. - * @ec: The echo canceller context. - * @tx: The transmitted audio sample. - * @rx: The received audio sample. - * - * The return value is the clean (echo cancelled) received sample. - */ -int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx); - -/** - * oslec_hpf_tx: Process to high pass filter the tx signal. - * @ec: The echo canceller context. - * @tx: The transmitted auio sample. - * - * The return value is the HP filtered transmit sample, send this to your D/A. - */ -int16_t oslec_hpf_tx(struct oslec_state *ec, int16_t tx); - -#endif /* __OSLEC_H */ |