/* aec.h
*
* Copyright (C) DFS Deutsche Flugsicherung (2004, 2005).
* All Rights Reserved.
* Author: Andre Adrian
*
* Acoustic Echo Cancellation NLMS-pw algorithm
*
* Version 0.3.1 Allow change of stability parameter delta
* Version 0.3 filter created with www.dsptutor.freeuk.com
*/
#ifndef _AEC_H /* include only once */
#include <string.h>
#include <stdlib.h>
#define WIDEB 2
#define RATE WIDEB*8000
// use double if your CPU does software-emulation of float
typedef float REAL;
/* dB Values */
const REAL M0dB = 1.0f;
const REAL M3dB = 0.71f;
const REAL M6dB = 0.50f;
const REAL M9dB = 0.35f;
const REAL M12dB = 0.25f;
const REAL M18dB = 0.125f;
const REAL M24dB = 0.063f;
/* dB values for 16bit PCM */
/* MxdB_PCM = 32767 * 10 ^(x / 20) */
const REAL M10dB_PCM = 10362.0f;
const REAL M20dB_PCM = 3277.0f;
const REAL M25dB_PCM = 1843.0f;
const REAL M30dB_PCM = 1026.0f;
const REAL M35dB_PCM = 583.0f;
const REAL M40dB_PCM = 328.0f;
const REAL M45dB_PCM = 184.0f;
const REAL M50dB_PCM = 104.0f;
const REAL M55dB_PCM = 58.0f;
const REAL M60dB_PCM = 33.0f;
const REAL M65dB_PCM = 18.0f;
const REAL M70dB_PCM = 10.0f;
const REAL M75dB_PCM = 6.0f;
const REAL M80dB_PCM = 3.0f;
const REAL M85dB_PCM = 2.0f;
const REAL M90dB_PCM = 1.0f;
const REAL MAXPCM = 32767.0f;
/* Design constants (Change to fine tune the algorithms */
/* The following values are for hardware AEC and studio quality
* microphone */
/* maximum NLMS filter length in taps. A longer filter length gives
* better Echo Cancellation, but maybe slower convergence speed and
* needs more CPU power (Order of NLMS is linear) */
#define NLMS_LEN (80*WIDEB*8)
/* minimum energy in xf. Range: M70dB_PCM to M50dB_PCM. Should be equal
* to microphone ambient Noise level */
const REAL NoiseFloor = M55dB_PCM;
/* Double Talk Detector Speaker/Microphone Threshold. Range <=1
* Large value (M0dB) is good for Single-Talk Echo cancellation,
* small value (M12dB) is good for Doulbe-Talk AEC */
const REAL GeigelThreshold = M6dB;
/* Double Talk Detector hangover in taps.
* Small Value (0ms = No DTD) is good for Single-Talk Echo cancellation,
* Large Value (30ms) is good for Double-Talk Echo cancellation.
*/
const int Thold = 30 * WIDEB * 8;
/* for Non Linear Processor. Range >0 to 1. Large value (M0dB) is good
* for Double-Talk, small value (M12dB) is good for Single-Talk */
const REAL NLPAttenuation = M12dB;
// Adrian soft decision DTD
// left point. X is ratio, Y is stepsize
const float STEPX1 = 1.0, STEPY1 = 1.0;
// right point. STEPX2=2.0 is good double talk, 3.0 is good single talk.
const float STEPX2 = 2.5, STEPY2 = 0;
/* Below this line there are no more design constants */
/* Exponential Smoothing or IIR Infinite Impulse Response Filter */
class IIR_HP {
REAL x;
public:
IIR_HP() {
x = 0.0f;
}
REAL highpass(REAL in) {
const REAL a0 = 0.01f; /* controls Transfer Frequency */
/* Highpass = Signal - Lowpass. Lowpass = Exponential Smoothing */
x += a0 * (in - x);
return in - x;
};
};
#if WIDEB==1
/* 17 taps FIR Finite Impulse Response filter
* Coefficients calculated with
* www.dsptutor.freeuk.com/KaiserFilterDesign/KaiserFilterDesign.html
*/
class FIR_HP_300Hz {
REAL z[18];
public:
FIR_HP_300Hz() {
memset(this, 0, sizeof(FIR_HP_300Hz));
}
REAL highpass(REAL in) {
const REAL a[18] = {
// Kaiser Window FIR Filter, Filter type: High pass
// Passband: 300.0 - 4000.0 Hz, Order: 16
// Transition band: 75.0 Hz, Stopband attenuation: 10.0 dB
-0.034870606, -0.039650206, -0.044063766, -0.04800318,
-0.051370874, -0.054082647, -0.056070227, -0.057283327,
0.8214126, -0.057283327, -0.056070227, -0.054082647,
-0.051370874, -0.04800318, -0.044063766, -0.039650206,
-0.034870606, 0.0
};
memmove(z + 1, z, 17 * sizeof(REAL));
z[0] = in;
REAL sum0 = 0.0, sum1 = 0.0;
int j;
for (j = 0; j < 18; j += 2) {
// optimize: partial loop unrolling
sum0 += a[j] * z[j];
sum1 += a[j + 1] * z[j + 1];
}
return sum0 + sum1;
}
};
#else
/* 35 taps FIR Finite Impulse Response filter
* Passband 150Hz to 4kHz for 8kHz sample rate, 300Hz to 8kHz for 16kHz
* sample rate.
* Coefficients calculated with
* www.dsptutor.freeuk.com/KaiserFilterDesign/KaiserFilterDesign.html
*/
class FIR_HP_300Hz {
REAL z[36];
public:
FIR_HP_300Hz() {
memset(this, 0, sizeof(FIR_HP_300Hz));
}
REAL highpass(REAL in) {
const REAL a[36] = {
// Kaiser Window FIR Filter, Filter type: High pass
// Passband: 150.0 - 4000.0 Hz, Order: 34
// Transition band: 34.0 Hz, Stopband attenuation: 10.0 dB
-0.016165324, -0.017454365, -0.01871232, -0.019931411,
-0.021104068, -0.022222936, -0.02328091, -0.024271343,
-0.025187887, -0.02602462, -0.026776174, -0.027437767,
-0.028004972, -0.028474221, -0.028842418, -0.029107114,
-0.02926664, 0.8524841, -0.02926664, -0.029107114,
-0.028842418, -0.028474221, -0.028004972, -0.027437767,
-0.026776174, -0.02602462, -0.025187887, -0.024271343,
-0.02328091, -0.022222936, -0.021104068, -0.019931411,
-0.01871232, -0.017454365, -0.016165324, 0.0
};
memmove(z + 1, z, 35 * sizeof(REAL));
z[0] = in;
REAL sum0 = 0.0, sum1 = 0.0;
int j;
for (j = 0; j < 36; j += 2) {
// optimize: partial loop unrolling
sum0 += a[j] * z[j];
sum1 += a[j + 1] * z[j + 1];
}
return sum0 + sum1;
}
};
#endif
/* Recursive single pole IIR Infinite Impulse response High-pass filter
*
* Reference: The Scientist and Engineer's Guide to Digital Processing
*
* output[N] = A0 * input[N] + A1 * input[N-1] + B1 * output[N-1]
*
* X = exp(-2.0 * pi * Fc)
* A0 = (1 + X) / 2
* A1 = -(1 + X) / 2
* B1 = X
* Fc = cutoff freq / sample rate
*/
class IIR1 {
REAL in0, out0;
REAL a0, a1, b1;
public:
IIR1() {
memset(this, 0, sizeof(IIR1));
}
void init(REAL Fc) {
b1 = expf(-2.0f * M_PI * Fc);
a0 = (1.0f + b1) / 2.0f;
a1 = -a0;
in0 = 0.0f;
out0 = 0.0f;
}
REAL highpass(REAL in) {
REAL out = a0 * in + a1 * in0 + b1 * out0;
in0 = in;
out0 = out;
return out;
}
};
/* Recursive two pole IIR Infinite Impulse Response filter
* Coefficients calculated with
* http://www.dsptutor.freeuk.com/IIRFilterDesign/IIRFiltDes102.html
*/
class IIR2 {
REAL x[2], y[2];
public:
IIR2() {
memset(this, 0, sizeof(IIR2));
}
REAL highpass(REAL in) {
// Butterworth IIR filter, Filter type: HP
// Passband: 2000 - 4000.0 Hz, Order: 2
const REAL a[] = { 0.29289323f, -0.58578646f, 0.29289323f };
const REAL b[] = { 1.3007072E-16f, 0.17157288f };
REAL out =
a[0] * in + a[1] * x[0] + a[2] * x[1] - b[0] * y[0] - b[1] * y[1];
x[1] = x[0];
x[0] = in;
y[1] = y[0];
y[0] = out;
return out;
}
};
// Extention in taps to reduce mem copies
#define NLMS_EXT (10*8)
// block size in taps to optimize DTD calculation
#define DTD_LEN 16
class AEC {
// Time domain Filters
IIR_HP acMic, acSpk; // DC-level remove Highpass)
FIR_HP_300Hz cutoff; // 150Hz cut-off Highpass
REAL gain; // Mic signal amplify
IIR1 Fx, Fe; // pre-whitening Highpass for x, e
IIR1 Fxx, Fdd; // pre-whitening Highpass
// Geigel DTD (Double Talk Detector)
REAL max_max_x; // max(|x[0]|, .. |x[L-1]|)
int hangover;
// optimize: less calculations for max()
REAL max_x[NLMS_LEN / DTD_LEN];
int dtdCnt;
int dtdNdx;
// Different ageing Near-End Far-End Ratio DTD
REAL dfast, xfast;
REAL dslow, xslow;
// H-infinity max
double max_max_u; // max(|x[0]|, .. |x[L-1]|)
// optimize: less calculations for max()
REAL max_u[NLMS_LEN / DTD_LEN];
int c_max_u;
int i_max_u;
// NLMS-pw
REAL x[NLMS_LEN + NLMS_EXT]; // tap delayed loudspeaker signal
REAL xf[NLMS_LEN + NLMS_EXT]; // pre-whitening tap delayed signal
REAL w[NLMS_LEN]; // tap weights
int j; // optimize: less memory copies
double dotp_xf_xf; // double to avoid loss of precision
float delta; // noise floor to stabilize NLMS
public:
AEC();
/* Double-Talk Detector
*
* in d: microphone sample (PCM as REALing point value)
* in x: loudspeaker sample (PCM as REALing point value)
* return: from 0 for doubletalk to 1.0 for single talk
*/
float dtd(REAL d, REAL x);
/* Acoustic Echo Suppression
*/
float AEC::aes(REAL d);
/* Normalized Least Mean Square Algorithm pre-whitening (NLMS-pw)
* The LMS algorithm was developed by Bernard Widrow
* book: Haykin, Adaptive Filter Theory, 4. edition, Prentice Hall, 2002
*
* in d: microphone sample (16bit PCM value)
* in x_: loudspeaker sample (16bit PCM value)
* in stepsize: NLMS adaptation variable
* return: echo cancelled microphone sample
*/
REAL nlms_pw(REAL d, REAL x_, float stepsize);
/* Acoustic Echo Cancellation and Suppression of one sample
* in d: microphone signal with echo
* in x: loudspeaker signal
* return: echo cancelled microphone signal
*/
int AEC::doAEC(int d_, int x_, int enable);
float AEC::getambient() {
return dfast;
};
void AEC::setambient(float Min_xf) {
dotp_xf_xf -= delta; // subtract old delta
delta = (NLMS_LEN-1) * Min_xf * Min_xf;
dotp_xf_xf += delta; // add new delta
};
void AEC::setgain(float gain_) {
gain = gain_;
};
double AEC::max_dotp_xf_xf(double u);
};
#define _AEC_H
#endif
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