Mercurial > hg > audiostuff
view intercom/aec.cpp @ 2:13be24d74cd2
import intercom-0.4.1
author | Peter Meerwald <pmeerw@cosy.sbg.ac.at> |
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date | Fri, 25 Jun 2010 09:57:52 +0200 |
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/* aec.cpp * * Copyright (C) DFS Deutsche Flugsicherung (2004, 2005). * All Rights Reserved. * * Acoustic Echo Cancellation NLMS-pw algorithm * * Version 0.3 filter created with www.dsptutor.freeuk.com * Version 0.3.1 Allow change of stability parameter delta * Version 0.4 Leaky Normalized LMS - pre whitening algorithm */ #include <stdio.h> #include <stdlib.h> #include <math.h> #include <string.h> #include <unistd.h> #include "oss.h" #include "aec.h" #include "intercomd.h" #include "tcp.h" /* Vector Dot Product */ REAL dotp(REAL a[], REAL b[]) { REAL sum0 = 0.0, sum1 = 0.0; int j; for (j = 0; j < NLMS_LEN; j += 2) { // optimize: partial loop unrolling sum0 += a[j] * b[j]; sum1 += a[j + 1] * b[j + 1]; } return sum0 + sum1; } AEC::AEC() { hangover = 0; memset(x, 0, sizeof(x)); memset(xf, 0, sizeof(xf)); memset(w, 0, sizeof(w)); j = NLMS_EXT; delta = 0.0f; setambient(NoiseFloor); dfast = dslow = M75dB_PCM; xfast = xslow = M80dB_PCM; gain = 1.0f; Fx.init(2000.0f/RATE); Fe.init(2000.0f/RATE); aes_y2 = M0dB; fdwdisplay = -1; dumpcnt = 0; memset(ws, 0, sizeof(ws)); } // Adrian soft decision DTD // (Dual Average Near-End to Far-End signal Ratio DTD) // This algorithm uses exponential smoothing with differnt // ageing parameters to get fast and slow near-end and far-end // signal averages. The ratio of NFRs term // (dfast / xfast) / (dslow / xslow) is used to compute the stepsize // A ratio value of 2.5 is mapped to stepsize 0, a ratio of 0 is // mapped to 1.0 with a limited linear function. inline float AEC::dtd(REAL d, REAL x) { float stepsize; // fast near-end and far-end average dfast += ALPHAFAST * (fabsf(d) - dfast); xfast += ALPHAFAST * (fabsf(x) - xfast); // slow near-end and far-end average dslow += ALPHASLOW * (fabsf(d) - dslow); xslow += ALPHASLOW * (fabsf(x) - xslow); if (xfast < M70dB_PCM) { return 0.0; // no Spk signal } if (dfast < M70dB_PCM) { return 0.0; // no Mic signal } // ratio of NFRs float ratio = (dfast * xslow) / (dslow * xfast); // begrenzte lineare Kennlinie const float M = (STEPY2 - STEPY1) / (STEPX2 - STEPX1); if (ratio < STEPX1) { stepsize = STEPY1; } else if (ratio > STEPX2) { stepsize = STEPY2; } else { // Punktrichtungsform einer Geraden stepsize = M * (ratio - STEPX1) + STEPY1; } return stepsize; } inline void AEC::leaky() // The xfast signal is used to charge the hangover timer to Thold. // When hangover expires (no Spk signal for some time) the vector w // is erased. This is my implementation of Leaky NLMS. { if (xfast >= M70dB_PCM) { // vector w is valid for hangover Thold time hangover = Thold; } else { if (hangover > 1) { --hangover; } else if (1 == hangover) { --hangover; // My Leaky NLMS is to erase vector w when hangover expires memset(w, 0, sizeof(w)); } } } void AEC::openwdisplay() { // open TCP connection to program wdisplay.tcl fdwdisplay = socket_async("127.0.0.1", 50999); }; inline REAL AEC::nlms_pw(REAL d, REAL x_, float stepsize) { x[j] = x_; xf[j] = Fx.highpass(x_); // pre-whitening of x // calculate error value // (mic signal - estimated mic signal from spk signal) REAL e = d; if (hangover > 0) { e -= dotp(w, x + j); } REAL ef = Fe.highpass(e); // pre-whitening of e // optimize: iterative dotp(xf, xf) dotp_xf_xf += (xf[j] * xf[j] - xf[j + NLMS_LEN - 1] * xf[j + NLMS_LEN - 1]); if (stepsize > 0.0) { // calculate variable step size REAL mikro_ef = stepsize * ef / dotp_xf_xf; // update tap weights (filter learning) int i; for (i = 0; i < NLMS_LEN; i += 2) { // optimize: partial loop unrolling w[i] += mikro_ef * xf[i + j]; w[i + 1] += mikro_ef * xf[i + j + 1]; } } if (--j < 0) { // optimize: decrease number of memory copies j = NLMS_EXT; memmove(x + j + 1, x, (NLMS_LEN - 1) * sizeof(REAL)); memmove(xf + j + 1, xf, (NLMS_LEN - 1) * sizeof(REAL)); } // Saturation if (e > MAXPCM) { return MAXPCM; } else if (e < -MAXPCM) { return -MAXPCM; } else { return e; } } int AEC::doAEC(int d_, int x_) { REAL d = (REAL) d_; REAL x = (REAL) x_; // Mic Highpass Filter - to remove DC d = acMic.highpass(d); // Mic Highpass Filter - cut-off below 300Hz d = cutoff.highpass(d); // Amplify, for e.g. Soundcards with -6dB max. volume d *= gain; // Spk Highpass Filter - to remove DC x = acSpk.highpass(x); // Double Talk Detector stepsize = dtd(d, x); // Leaky (ageing of vector w) leaky(); // Acoustic Echo Cancellation d = nlms_pw(d, x, stepsize); if (fdwdisplay >= 0) { if (++dumpcnt >= (WIDEB*8000/10)) { // wdisplay creates 10 dumps per seconds = large CPU load! dumpcnt = 0; write(fdwdisplay, ws, DUMP_LEN*sizeof(float)); // we don't check return value. This is not production quality!!! memset(ws, 0, sizeof(ws)); } else { int i; for (i = 0; i < DUMP_LEN; i += 2) { // optimize: partial loop unrolling ws[i] += w[i]; ws[i + 1] += w[i + 1]; } } } return (int) d; }