Design of adaptive equalizer

Using LMS Algorithm

Project Report

Under the guidance of

Prof S Meher
Dept of Electronics and Communication Engineering
National Institute of Technology, Rourkela

4/20/2010
DSP LAB
ABHIJEET SAHU : 107EC019 SOUMYAJYOTI BEHERA : 107EC032

1
Aim of the project: Design of an adaptive equalizer using LMS(least mean square)
algorithm.

INTRODUCTION:

One of the most important advantages of the digital transmission systems for voice, data and
video communications is their higher reliability in noise environment in comparison with that of
their analog counterparts. Unfortunately most often the digital transmission of information is
accompanied with a phenomenon known as inter symbol interference (ISI). Briefly this means
that the transmitted pulses are smeared out so that pulses that correspond to different symbols are
not separable. Obviously for a reliable digital transmission system it is crucial to reduce the
effects of ISI. And one of the possible solutions is the implementation of equalizer based on filter
with finite impulse response (FIR) employing the least mean square (LMS) algorithm for
adjusting its coefficients.

ISI problem is resolved by channel equalization in which the aim is to construct an equalizer
such that the impulse response of the channel/equalizer combination is as close to z-d as possible,
where d is a delay. Frequently the channel parameters are not known in advance and moreover
they may vary with time, in some applications significantly. Hence, it is necessary to use the
adaptive equalizers, which provide the means of tracking the channel characteristics.
BLOCK DIAGRAM OF DIGITAL TRANSMISION SYSTEM USING CHANNEL EQUALIZATION

The source block transmits Binary Phase Shift Keying (BPSK) symbols u[n] with
equal probability. The channel block introduces inter symbol interference (ISI) using a finite
impulse response (FIR) type of channel model. The transfer function of channel CK is

Ck = C0+C1Z-1+C2 Z-2+……….+CL-1ZL-1
L = no of channel filter coefficients
At the output of the channel, a noise sequence v[n] is added. This noise is assumed to be additive
white Gaussian noise (AWGN) with variance Ω . The sum of the channel output and the noise
sequence forms the received signal x[n] , which is fed into the equalizer block. The equalizer
block performs the equalization of the channel, which is a transversal filter as shown in fig 2.

LMS ALGORITHM:

As in this project work we have designed an adaptive equalizer using the mean square error
(MSE) criterion. The minimization of the cost function can be performed adaptively by applying
the stochastic gradient (SG) or the least mean square (LMS) algorithm.

The LMS (least mean square) algorithm is an approximation of the steepest descent
algorithm which uses an instantaneous estimate of the gradient vector of a cost function. The
estimate of the gradient is based on sample values of the tap-input vector and an error signal.
The algorithm iterates over each coefficient in the filter, moving it in the direction of the
approximated gradient. For the LMS algorithm it is necessary to have a reference signal d[n]
representing the desired filter output. The difference between the reference signal and the actual
output of the transversal filter is the error signal:
e[n] = u[n-d] – y[n]*wk[n]

The task of the LMS algorithm is to find a set of filter coefficients c that minimize the
expected value of the quadratic error signal, i.e., to achieve the least mean squared error
(thus the name). The squared error and its expected value are:
(e[n])2 = {u[n-d] – wkT[n]y[n]}2
E(e2) = E(u2) - E(2uwTy) + E(wTyyTw)
= E(u2) – 2wTE(uy) + wTE(yyT)w

the squared error e2 is a quadratic function of the coefficient vector c, and

thus has only one (global) minimum (and no other (local) minima), that theoretically could
be found if the correct expected values in the above equation are known. In the LMS algorithm,
however, a very short-term estimate is used by only taking into account the current samples:
E(ux) ~ dx, and E(yyT) ~ yyT, leading to an update equation for the filter coefficients:

wk[n+1] = wk[n] + µy[n]e[n]T

where µ is the step size parameter.

The dependence of the stability bound on the signal power is exploited in the normalized LMS
algorithm by normalizing the step-size according to the signal power:
wk[n+1] = wk[n] + µy[n]e[n]T / € + |y[n]|2
€: small positive constant

The mean square error (MSE) for the filter in the kth time instant is defined as :
MSEk = E(ek2)
In simulations the expectation is evaluated through averaging over T independent runs, in order
to be able to view the convergence of the equalizer as a function of the time. After the
convergence of the adaptive equalizer, we can compute the final MSE by time averaging if the
process is ergodic . However, it is important that the initial transient of the equalizer is discarded.
Otherwise, we will end up with a too large value of the MSE.

MATLAB PROGRAM AND SIMULATION :

clc
clear all
close all

N = 1000; %simulation length

M = 5; %channel length

T = 50; %number of independent trials

cir = zeros(1,2*M-1); %cascade impulse response

MSE_vec = zeros(T,N-3); %Mean Square Error vector

MSE_f=zeros(1,N-3); %final MSE vector

o_p=zeros(1,N); %output from the equalizer in Tth trial

for j = 1 : T

u = sign(randn(1,N)); %input signal

c = randn(M,1); %channel to be equalized

c = c / norm(c);

z = filter(c,1,u); %channel output

SNR = 30; %Signal to Noise Ratio

var_v = var(z) * 10^(-SNR/10); %additive noise to the channel output
v = var_v^0.5 * randn(1,N);
 y = z + v; %input to the equalizer

%-----------------------------LMS channel equalization-----------------------%

w = zeros(M,1);
y_regressor = zeros(1,M);
step =.12;
epsilon = 10^(-6);
MSE=0;
for k = 4 : N
y_regressor = [y(k) y_regressor(1:M-1)];
e = u(k-3) - y_regressor * w;
w = w + step * y_regressor' * e / (y_regressor * y_regressor' + epsilon);

if j==T
o_p(1,k)=y_regressor * w;
end

MSE=MSE+(abs(e).^2);
MSE_avg=MSE/(k-3);
MSE_vec(j,:)=[MSE_avg MSE_vec(j,1:N-4)];

end
MSE_vec(j,:)=fliplr(MSE_vec(j,:));
q=[4:1:N];
figure(1);
plot(q,MSE_vec(j,:));title('mean square error for different trial');xlabel('time at any instant ');ylabel('MSE');
hold on;
cir = cir + conv(w,c)';
end
hold off

figure(2);
subplot(2,2,1);
plot(q,MSE_vec(j,:)); title('MSE in final trial');
xlabel('Time in samples');ylabel('MSE');
hold on;

o_p_b=sign(o_p); %converting to binary(+1/-1)

n=[1:N];
plot(n,v); %noise in the final trial
hold off;

temp=ones(T,1);
for j=1:N-3
MSE_f(1,j)=MSE_f(1,j)+((MSE_vec(:,j)'*temp)/T);
end
subplot(2,2,2);
plot(q,MSE_f);
title('MSE in final trial'); xlabel('time at any instant');ylabel('MSE');
%-----------------------FOR SIGNALS IN FINAL TRIAL-------------------------%
figure(3);
subplot(2,2,1);
stem(c);title('channel impulse response');xlabel('time');ylabel('magnitude');

subplot(2,2,2);
stem(w);title('equalizer impulse response');xlabel('time');ylabel('magnitude');

subplot(2,2,3);
stem(cir/T);
title('cascade channel-equalizer impulse response');ylabel('magnitude');xlabel('taps');

figure(4);
subplot(2,2,1);
n_i=n(4:N);
stem(n_i,u(1:N-3));xlabel('time');title('input signal');ylabel('magnitude');

subplot(2,2,2);
stem(n,z);xlabel('time');title('output after passing through channel');ylabel('magnitude');

subplot(2,2,3);
n_o=n(1:N-3);
stem(n_o,o_p(1:N-3));xlabel('time');title('Final output');ylabel('magnitude');

subplot(2,2,4);
stem(n_o,o_p_b(1:N-3));xlabel('time');title('Binary final output');ylabel('magnitude');

CODE DESCRIPTION :

In the following MATLAB code, we took a random digital signal and passéd through a channel
of length M=5. The output from the channel was passed through a noise signal, which corrupts
the original signal. To retrieve the original signal we used the LMS algorithm to design the
adaptive equalizer whose coefficients were dependant on the difference of the delayed original
signal and the output from the equalizer i.e. the error signal. Until and unless the error function
gets almost reduced to zero the output signal can’t be equal to the input signal. The efficiency of
this algorithm is judged by the degree of convergence i.e. the decrease of error function with the
succeeding iterations. The mean square error found from the very last iteration gives the best
result compared to its predecessor. So we checked the various MSE values for different iterations
and checked the convergence. Finally we convolved the channel with the equalizer to get the
cascade impulse response.
Shown below are the following plots made out from the above code, to verify the way the
adaptive equalizer helped us to retrieve the corrupt data.
Parameters used:

Parameter Value
No of independent trials T 50
No of transmitted symbols N 1000
No of filter coefficients M 5
SNR 30
Step size .12
Channel coefficients random
Figure 1: Impulse responses

Figure 2: Mean Square Error

 Figure 3: Discrete Data output

CONCLUSION:

The basic goal of this project was to design an adaptive equalizer using an algorithm which
provides us a high convergence rate in order to meet the requirements for short training time and
good tracking properties. So we used the LMS algorithm to design it and analyzed the mean
square error which reduced finally to provide us the intended output with a certain delay. The
channel and the noise signals were taken randomly to realize the real time environment and were
equalized properly as the corrupted signals were recovered.

LMS algorithm is largely used due to its unique properties:

1. Simplicity of implementation
2. Model independent
3. Higher degree of convergence of the error function
One of the many applications of LMS is acoustic echo cancellation.