audiostuff: intercom/aec.h comparison

comparison intercom/aec.h @ 2:13be24d74cd2

import intercom-0.4.1

author	Peter Meerwald <pmeerw@cosy.sbg.ac.at>
date	Fri, 25 Jun 2010 09:57:52 +0200
parents
children	c6c5a16ce2f2

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-:9cadc470e3da
+:13be24d74cd2
+/* aec.h
+*
+* Copyright (C) DFS Deutsche Flugsicherung (2004, 2005).
+* All Rights Reserved.
+* Author: Andre Adrian
+*
+* Acoustic Echo Cancellation Leaky 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
+*/
+#ifndef _AEC_H                  /* include only once */
+// 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 */
+/* NLMS filter length in taps (samples). 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  (100*WIDEB*8)
+/* Vector w visualization length in taps (samples).
+* Must match argv value for wdisplay.tcl */
+#define DUMP_LEN  (40*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;
+/* Leaky hangover in taps.
+*/
+const int Thold = 60 * WIDEB * 8;
+// 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;
+const float ALPHAFAST = 1.0f / 100.0f;
+const float ALPHASLOW = 1.0f / 20000.0f;
+/* Ageing multiplier for LMS memory vector w */
+const REAL Leaky = 0.9999f;
+/* 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;
+/* 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;
+/* 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
+// Adrian soft decision DTD (Double Talk Detector)
+REAL dfast, xfast;
+REAL dslow, xslow;
+// 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
+// AES
+float aes_y2;                 // not in use!
+// w vector visualization
+REAL ws[DUMP_LEN];            // tap weights sums
+int fdwdisplay;               // TCP file descriptor
+int dumpcnt;                  // wdisplay output counter
+/* 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);
+void AEC::leaky();
+/* 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);
+public:
+// variables are public for visualization
+int hangover;
+float stepsize;
+AEC();
+/* 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_);
+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_;
+};
+void AEC::openwdisplay();
+void AEC::setaes(float aes_y2_) {
+aes_y2 = aes_y2_;
+};
+double AEC::max_dotp_xf_xf(double u);
+};
+#define _AEC_H
+#endif

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comparison intercom/aec.h @ 2:13be24d74cd2