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![Jesper Juhl](/assets/img/avatar_default.png)
Our convention is one (statement and) variable per line. Enforce this in drivers/staging/echo/echo.[ch] . Signed-off-by: Jesper Juhl <jj@chaosbits.net> Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
662 lines
19 KiB
C
662 lines
19 KiB
C
/*
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* SpanDSP - a series of DSP components for telephony
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*
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* echo.c - A line echo canceller. This code is being developed
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* against and partially complies with G168.
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*
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* Written by Steve Underwood <steveu@coppice.org>
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* and David Rowe <david_at_rowetel_dot_com>
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*
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* Copyright (C) 2001, 2003 Steve Underwood, 2007 David Rowe
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*
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* Based on a bit from here, a bit from there, eye of toad, ear of
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* bat, 15 years of failed attempts by David and a few fried brain
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* cells.
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*
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* All rights reserved.
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License version 2, as
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* published by the Free Software Foundation.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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*/
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/*! \file */
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/* Implementation Notes
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David Rowe
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April 2007
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This code started life as Steve's NLMS algorithm with a tap
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rotation algorithm to handle divergence during double talk. I
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added a Geigel Double Talk Detector (DTD) [2] and performed some
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G168 tests. However I had trouble meeting the G168 requirements,
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especially for double talk - there were always cases where my DTD
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failed, for example where near end speech was under the 6dB
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threshold required for declaring double talk.
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So I tried a two path algorithm [1], which has so far given better
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results. The original tap rotation/Geigel algorithm is available
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in SVN http://svn.rowetel.com/software/oslec/tags/before_16bit.
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It's probably possible to make it work if some one wants to put some
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serious work into it.
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At present no special treatment is provided for tones, which
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generally cause NLMS algorithms to diverge. Initial runs of a
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subset of the G168 tests for tones (e.g ./echo_test 6) show the
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current algorithm is passing OK, which is kind of surprising. The
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full set of tests needs to be performed to confirm this result.
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One other interesting change is that I have managed to get the NLMS
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code to work with 16 bit coefficients, rather than the original 32
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bit coefficents. This reduces the MIPs and storage required.
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I evaulated the 16 bit port using g168_tests.sh and listening tests
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on 4 real-world samples.
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I also attempted the implementation of a block based NLMS update
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[2] but although this passes g168_tests.sh it didn't converge well
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on the real-world samples. I have no idea why, perhaps a scaling
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problem. The block based code is also available in SVN
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http://svn.rowetel.com/software/oslec/tags/before_16bit. If this
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code can be debugged, it will lead to further reduction in MIPS, as
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the block update code maps nicely onto DSP instruction sets (it's a
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dot product) compared to the current sample-by-sample update.
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Steve also has some nice notes on echo cancellers in echo.h
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References:
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[1] Ochiai, Areseki, and Ogihara, "Echo Canceller with Two Echo
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Path Models", IEEE Transactions on communications, COM-25,
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No. 6, June
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1977.
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http://www.rowetel.com/images/echo/dual_path_paper.pdf
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[2] The classic, very useful paper that tells you how to
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actually build a real world echo canceller:
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Messerschmitt, Hedberg, Cole, Haoui, Winship, "Digital Voice
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Echo Canceller with a TMS320020,
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http://www.rowetel.com/images/echo/spra129.pdf
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[3] I have written a series of blog posts on this work, here is
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Part 1: http://www.rowetel.com/blog/?p=18
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[4] The source code http://svn.rowetel.com/software/oslec/
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[5] A nice reference on LMS filters:
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http://en.wikipedia.org/wiki/Least_mean_squares_filter
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Credits:
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Thanks to Steve Underwood, Jean-Marc Valin, and Ramakrishnan
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Muthukrishnan for their suggestions and email discussions. Thanks
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also to those people who collected echo samples for me such as
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Mark, Pawel, and Pavel.
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*/
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#include <linux/kernel.h>
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#include <linux/module.h>
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#include <linux/slab.h>
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#include "echo.h"
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#define MIN_TX_POWER_FOR_ADAPTION 64
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#define MIN_RX_POWER_FOR_ADAPTION 64
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#define DTD_HANGOVER 600 /* 600 samples, or 75ms */
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#define DC_LOG2BETA 3 /* log2() of DC filter Beta */
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/* adapting coeffs using the traditional stochastic descent (N)LMS algorithm */
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#ifdef __bfin__
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static inline void lms_adapt_bg(struct oslec_state *ec, int clean, int shift)
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{
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int i;
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int j;
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int offset1;
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int offset2;
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int factor;
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int exp;
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int16_t *phist;
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int n;
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if (shift > 0)
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factor = clean << shift;
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else
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factor = clean >> -shift;
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/* Update the FIR taps */
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offset2 = ec->curr_pos;
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offset1 = ec->taps - offset2;
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phist = &ec->fir_state_bg.history[offset2];
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/* st: and en: help us locate the assembler in echo.s */
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/* asm("st:"); */
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n = ec->taps;
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for (i = 0, j = offset2; i < n; i++, j++) {
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exp = *phist++ * factor;
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ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
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}
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/* asm("en:"); */
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/* Note the asm for the inner loop above generated by Blackfin gcc
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4.1.1 is pretty good (note even parallel instructions used):
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R0 = W [P0++] (X);
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R0 *= R2;
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R0 = R0 + R3 (NS) ||
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R1 = W [P1] (X) ||
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nop;
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R0 >>>= 15;
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R0 = R0 + R1;
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W [P1++] = R0;
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A block based update algorithm would be much faster but the
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above can't be improved on much. Every instruction saved in
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the loop above is 2 MIPs/ch! The for loop above is where the
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Blackfin spends most of it's time - about 17 MIPs/ch measured
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with speedtest.c with 256 taps (32ms). Write-back and
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Write-through cache gave about the same performance.
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*/
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}
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/*
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IDEAS for further optimisation of lms_adapt_bg():
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1/ The rounding is quite costly. Could we keep as 32 bit coeffs
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then make filter pluck the MS 16-bits of the coeffs when filtering?
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However this would lower potential optimisation of filter, as I
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think the dual-MAC architecture requires packed 16 bit coeffs.
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2/ Block based update would be more efficient, as per comments above,
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could use dual MAC architecture.
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3/ Look for same sample Blackfin LMS code, see if we can get dual-MAC
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packing.
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4/ Execute the whole e/c in a block of say 20ms rather than sample
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by sample. Processing a few samples every ms is inefficient.
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*/
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#else
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static inline void lms_adapt_bg(struct oslec_state *ec, int clean, int shift)
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{
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int i;
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int offset1;
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int offset2;
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int factor;
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int exp;
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if (shift > 0)
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factor = clean << shift;
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else
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factor = clean >> -shift;
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/* Update the FIR taps */
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offset2 = ec->curr_pos;
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offset1 = ec->taps - offset2;
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for (i = ec->taps - 1; i >= offset1; i--) {
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exp = (ec->fir_state_bg.history[i - offset1] * factor);
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ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
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}
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for (; i >= 0; i--) {
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exp = (ec->fir_state_bg.history[i + offset2] * factor);
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ec->fir_taps16[1][i] += (int16_t) ((exp + (1 << 14)) >> 15);
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}
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}
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#endif
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static inline int top_bit(unsigned int bits)
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{
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if (bits == 0)
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return -1;
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else
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return (int)fls((int32_t) bits) - 1;
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}
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struct oslec_state *oslec_create(int len, int adaption_mode)
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{
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struct oslec_state *ec;
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int i;
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ec = kzalloc(sizeof(*ec), GFP_KERNEL);
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if (!ec)
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return NULL;
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ec->taps = len;
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ec->log2taps = top_bit(len);
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ec->curr_pos = ec->taps - 1;
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for (i = 0; i < 2; i++) {
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ec->fir_taps16[i] =
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kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
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if (!ec->fir_taps16[i])
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goto error_oom;
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}
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fir16_create(&ec->fir_state, ec->fir_taps16[0], ec->taps);
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fir16_create(&ec->fir_state_bg, ec->fir_taps16[1], ec->taps);
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for (i = 0; i < 5; i++)
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ec->xvtx[i] = ec->yvtx[i] = ec->xvrx[i] = ec->yvrx[i] = 0;
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ec->cng_level = 1000;
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oslec_adaption_mode(ec, adaption_mode);
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ec->snapshot = kcalloc(ec->taps, sizeof(int16_t), GFP_KERNEL);
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if (!ec->snapshot)
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goto error_oom;
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ec->cond_met = 0;
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ec->Pstates = 0;
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ec->Ltxacc = ec->Lrxacc = ec->Lcleanacc = ec->Lclean_bgacc = 0;
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ec->Ltx = ec->Lrx = ec->Lclean = ec->Lclean_bg = 0;
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ec->tx_1 = ec->tx_2 = ec->rx_1 = ec->rx_2 = 0;
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ec->Lbgn = ec->Lbgn_acc = 0;
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ec->Lbgn_upper = 200;
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ec->Lbgn_upper_acc = ec->Lbgn_upper << 13;
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return ec;
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error_oom:
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for (i = 0; i < 2; i++)
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kfree(ec->fir_taps16[i]);
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kfree(ec);
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return NULL;
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}
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EXPORT_SYMBOL_GPL(oslec_create);
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void oslec_free(struct oslec_state *ec)
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{
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int i;
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fir16_free(&ec->fir_state);
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fir16_free(&ec->fir_state_bg);
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for (i = 0; i < 2; i++)
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kfree(ec->fir_taps16[i]);
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kfree(ec->snapshot);
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kfree(ec);
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}
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EXPORT_SYMBOL_GPL(oslec_free);
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void oslec_adaption_mode(struct oslec_state *ec, int adaption_mode)
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{
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ec->adaption_mode = adaption_mode;
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}
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EXPORT_SYMBOL_GPL(oslec_adaption_mode);
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void oslec_flush(struct oslec_state *ec)
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{
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int i;
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ec->Ltxacc = ec->Lrxacc = ec->Lcleanacc = ec->Lclean_bgacc = 0;
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ec->Ltx = ec->Lrx = ec->Lclean = ec->Lclean_bg = 0;
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ec->tx_1 = ec->tx_2 = ec->rx_1 = ec->rx_2 = 0;
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ec->Lbgn = ec->Lbgn_acc = 0;
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ec->Lbgn_upper = 200;
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ec->Lbgn_upper_acc = ec->Lbgn_upper << 13;
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ec->nonupdate_dwell = 0;
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fir16_flush(&ec->fir_state);
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fir16_flush(&ec->fir_state_bg);
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ec->fir_state.curr_pos = ec->taps - 1;
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ec->fir_state_bg.curr_pos = ec->taps - 1;
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for (i = 0; i < 2; i++)
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memset(ec->fir_taps16[i], 0, ec->taps * sizeof(int16_t));
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ec->curr_pos = ec->taps - 1;
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ec->Pstates = 0;
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}
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EXPORT_SYMBOL_GPL(oslec_flush);
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void oslec_snapshot(struct oslec_state *ec)
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{
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memcpy(ec->snapshot, ec->fir_taps16[0], ec->taps * sizeof(int16_t));
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}
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EXPORT_SYMBOL_GPL(oslec_snapshot);
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/* Dual Path Echo Canceller */
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int16_t oslec_update(struct oslec_state *ec, int16_t tx, int16_t rx)
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{
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int32_t echo_value;
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int clean_bg;
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int tmp;
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int tmp1;
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/*
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* Input scaling was found be required to prevent problems when tx
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* starts clipping. Another possible way to handle this would be the
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* filter coefficent scaling.
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*/
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ec->tx = tx;
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ec->rx = rx;
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tx >>= 1;
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rx >>= 1;
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/*
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* Filter DC, 3dB point is 160Hz (I think), note 32 bit precision
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* required otherwise values do not track down to 0. Zero at DC, Pole
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* at (1-Beta) on real axis. Some chip sets (like Si labs) don't
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* need this, but something like a $10 X100P card does. Any DC really
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* slows down convergence.
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*
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* Note: removes some low frequency from the signal, this reduces the
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* speech quality when listening to samples through headphones but may
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* not be obvious through a telephone handset.
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*
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* Note that the 3dB frequency in radians is approx Beta, e.g. for Beta
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* = 2^(-3) = 0.125, 3dB freq is 0.125 rads = 159Hz.
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*/
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if (ec->adaption_mode & ECHO_CAN_USE_RX_HPF) {
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tmp = rx << 15;
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/*
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* Make sure the gain of the HPF is 1.0. This can still
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* saturate a little under impulse conditions, and it might
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* roll to 32768 and need clipping on sustained peak level
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* signals. However, the scale of such clipping is small, and
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* the error due to any saturation should not markedly affect
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* the downstream processing.
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*/
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tmp -= (tmp >> 4);
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ec->rx_1 += -(ec->rx_1 >> DC_LOG2BETA) + tmp - ec->rx_2;
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/*
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* hard limit filter to prevent clipping. Note that at this
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* stage rx should be limited to +/- 16383 due to right shift
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* above
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*/
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tmp1 = ec->rx_1 >> 15;
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if (tmp1 > 16383)
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tmp1 = 16383;
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if (tmp1 < -16383)
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tmp1 = -16383;
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rx = tmp1;
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ec->rx_2 = tmp;
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}
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/* Block average of power in the filter states. Used for
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adaption power calculation. */
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{
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int new, old;
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/* efficient "out with the old and in with the new" algorithm so
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we don't have to recalculate over the whole block of
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samples. */
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new = (int)tx * (int)tx;
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old = (int)ec->fir_state.history[ec->fir_state.curr_pos] *
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(int)ec->fir_state.history[ec->fir_state.curr_pos];
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ec->Pstates +=
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((new - old) + (1 << (ec->log2taps - 1))) >> ec->log2taps;
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if (ec->Pstates < 0)
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ec->Pstates = 0;
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}
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/* Calculate short term average levels using simple single pole IIRs */
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ec->Ltxacc += abs(tx) - ec->Ltx;
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ec->Ltx = (ec->Ltxacc + (1 << 4)) >> 5;
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ec->Lrxacc += abs(rx) - ec->Lrx;
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ec->Lrx = (ec->Lrxacc + (1 << 4)) >> 5;
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/* Foreground filter */
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ec->fir_state.coeffs = ec->fir_taps16[0];
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echo_value = fir16(&ec->fir_state, tx);
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ec->clean = rx - echo_value;
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ec->Lcleanacc += abs(ec->clean) - ec->Lclean;
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ec->Lclean = (ec->Lcleanacc + (1 << 4)) >> 5;
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/* Background filter */
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echo_value = fir16(&ec->fir_state_bg, tx);
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clean_bg = rx - echo_value;
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ec->Lclean_bgacc += abs(clean_bg) - ec->Lclean_bg;
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ec->Lclean_bg = (ec->Lclean_bgacc + (1 << 4)) >> 5;
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/* Background Filter adaption */
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/* Almost always adap bg filter, just simple DT and energy
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detection to minimise adaption in cases of strong double talk.
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However this is not critical for the dual path algorithm.
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*/
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ec->factor = 0;
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ec->shift = 0;
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if ((ec->nonupdate_dwell == 0)) {
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int P, logP, shift;
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/* Determine:
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f = Beta * clean_bg_rx/P ------ (1)
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where P is the total power in the filter states.
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The Boffins have shown that if we obey (1) we converge
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quickly and avoid instability.
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The correct factor f must be in Q30, as this is the fixed
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point format required by the lms_adapt_bg() function,
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therefore the scaled version of (1) is:
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(2^30) * f = (2^30) * Beta * clean_bg_rx/P
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factor = (2^30) * Beta * clean_bg_rx/P ----- (2)
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We have chosen Beta = 0.25 by experiment, so:
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factor = (2^30) * (2^-2) * clean_bg_rx/P
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(30 - 2 - log2(P))
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factor = clean_bg_rx 2 ----- (3)
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To avoid a divide we approximate log2(P) as top_bit(P),
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which returns the position of the highest non-zero bit in
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P. This approximation introduces an error as large as a
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factor of 2, but the algorithm seems to handle it OK.
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Come to think of it a divide may not be a big deal on a
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modern DSP, so its probably worth checking out the cycles
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for a divide versus a top_bit() implementation.
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*/
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P = MIN_TX_POWER_FOR_ADAPTION + ec->Pstates;
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logP = top_bit(P) + ec->log2taps;
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shift = 30 - 2 - logP;
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ec->shift = shift;
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|
|
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");
|