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[Link] OF RANDOM DATA AT A GIVEN DATA RATE

Aim:-
To generate the random data at a given data rate.
Software Required:- Pc with MATLAB 7.0 or above
Theory:-
A maximum length sequence (MLS) is a type of pseudorandom binary
sequence.
They are bit sequences generated using maximal linear feedback shift
registers and are so called because they are periodic and reproduce every
binary sequence that can be reproduced by the shift registers (i.e., for
length-m registers they produce a sequence of length 2m − 1). A MLS is also
sometimes called an n-sequence or an m-sequence. MLSs are spectrally
flat, with the exception of a near-zero DC term.
These sequences may be represented as coefficients of irreducible
polynomials in a polynomial ring over Z/2Z.
Practical applications for MLS include measuring impulse responses (e.g., of
room reverberation). They are also used as a basis for deriving pseudo-
random sequences in digital communication systems that employ direct-
sequence spread spectrum and frequency-hopping spread spectrum
transmission systems, and in the efficient design of some MRI experiments
Generation of maximum length sequences
Figure 1: The next value of register a3 in a feedback shift register of
length 4 is determined by the modulo-2 sum of a0 and a1.
MLS are generated using maximal linear feedback shift registers. An MLS-
generating system with a shift register of length 4 is shown in Fig. 1. It can
be expressed using the following recursive relation:
where n is the time index, k is the bit register position, and + represents
modulo-2 addition.

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As MLS are periodic and shift registers cycle through every possible binary
value (with the exception of the zero vector), registers can be initialized to
any state, with the exception of the zero vector.
A Gold code, also known as Gold sequence, is a type of binary
sequence, used in telecommunication (CDMA) and satellite navigation
(GPS). Gold codes are named after Robert Gold. Gold codes have bounded
small cross-correlations within a set, which is useful when multiple devices
are broadcasting in the same range. A set of Gold code sequences consists
of 2n − 1 sequences each one with a period of 2n − 1.
A set of Gold codes can be generated with the following steps. Pick two
maximum length sequences of the same length 2n − 1 such that their
absolute cross-correlation is less than or equal to 2(n + 2) / 2, where n is the
size of the LFSR used to generate the maximum length sequence (Gold '67).
The set of the 2n − 1 exclusive-ors of the two sequences in their various
phases (i.e. translated into all relative positions) is a set of Gold codes. The
highest absolute cross-correlation in this set of codes is 2(n + 2) / 2 + 1 for
even n and 2(n + 1) / 2 + 1 for odd n.
The exclusive or of two Gold codes from the same set is another Gold
code in some phase.
If l is odd, t = 2(l+1)/2 + 1, and
If l is even, t = 2(l+2)/2 + 1.
Thus, a Gold sequence formally is an arbitrary phase of a sequence in
the set G(u,v) defined by
G(u,v)= {u,v,u * v, u * Tv, u *T2 v, U * T(N-1) v}
Tk denotes the operator which shifts vectors cyclically to the left by k
places, * is the exclusive OR operator and u, v are m-sequences of period
generated by different primitive binary polynomials.

Matlab Program:-

clc;
clear all;
m=input('enter the size of shift registers m =');
N=2^m-1;
tt=-N+1:1:N;
x=ones(1,m);

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x1=x;
taps1=input('enter the taps1 =');
taps2=input('enter the taps2 =');
for i=1:N
y1(i)=x1(m);
x2=xor12(x1,taps1);
x1=[x2 x1(1:m)];
end
%disp(y1);
%%%Test-1
T1=sum(y1);
disp('total number of of 1s in maximalseq1 1s');
disp(T1);
if(T1==2^(m-1))
disp('maximal sequence is exist between the selected taps1');
else
disp('maximal sequence is NOT exist between the selected taps1');
end
%%%test2
sy1=2*y1-1;
s1=correlation11(sy1,sy1);
figure(1);
plot(tt,s1);
title('Auto correlation of Maximal sequence1');
xlabel('no of samples---->');
ylabel('magnitude---->');
grid;
%%%2nd maximal sequences
for i=1:N
y2(i)=x(m);
x2=xor12(x,taps2);
x=[x2 x(1:m)];
end
%disp(y2);
%%%Test-1
T2=sum(y2);
disp('total number of of 1s in maximalseq2 1s');
disp(T2);
if(T2==2^(m-1))
disp('maximal sequence is exist between the selected taps2');
else
disp('maximal sequence is NOT exist between the selected taps2');
end
%%%test2
sy2=2*y2-1;
s2=correlation11(sy2,sy2);
figure(2);
plot(tt,s2);
title('Auto correlation of Maximal sequence2');
xlabel('no of samples---->');

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ylabel('magnitude---->')
grid;
if(T1==T2)
%module-2 of 2'maximalseq is given as gold sequences(y)
y=xor(y1,y2);
g1=2*y-1;
%%%to check 3-value cross correlation function
x=correlation11(sy1,sy2);
figure(3);
plot(tt,x);
title('cross correlation of MLS1&MLS2');
xlabel('no of samples---->');
ylabel('magnitude---->')
grid;
%%% autocorrelation of gold sequences
xx=correlation11(g1,g1);
figure(4);
plot(tt,xx);
title('auto correlation of gold sequenceses');
xlabel('no of samples---->');
ylabel('magnitude---->')
grid;
disp('gold sequences is exist between the selected taps');
else
disp('gold sequences is NOT exist between the selected taps');
end

%%%gold sequence testing by find corss correlation function

c=input('enter the MLsequence1 or MLSequence2 c=');
if(c==1)
seq=sy1;
else
seq=sy2;
end
nn=input('enter the No shift we want nn=');
yz=2*y-1;
yz1=circshift(yz,[0,nn]);
Y=correlation11(seq,yz1);
%disp(Y);
figure(5);
plot(tt ,Y);
title('3-value checking of goldsequence ');
xlabel('No of samples ');
ylabel('magnitude');
grid;

xor12 sub function:

function z=xor12(s,t)
n=length(t);

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c(1)=xor(s(t(1)),s(t(2)));
for i=1:n-2
c(i+1)=xor(s(t(i+2)),c(i));
end
z=c(length(c));
end
correlation sub function:-
function res=correlation11(x,y)
%%%right shift operation
zzz=0;
for i=1:length(x)
rx=circshift(x,[1,i]);
z=y.*rx;
zz=sum(z);
zzz=[zzz zz];
end
zzz(:,1)=[];

%%%left shift operation

zzz1=0;
for i=1:length(x)
rx=circshift(x,[1,-i]);
z=y.*rx;
zz=sum(z);
zzz1=[zzz1 zz];
end
zzz1(:,1)=[];
total=[(zzz1) zzz];
res=total;
end

Output Results:-
enter the size of shift registers m =7
enter the taps1 =[7 1]
enter the taps2 =[7 3]
total number of of 1s in maximalseq1 1s
64
maximal sequence is exist between the selected taps1
total number of of 1s in maximalseq2 1s
64
maximal sequence is exist between the selected taps2
gold sequences is exist between the selected taps
enter the MLsequence1 or MLSequence2 c=1
enter the No shift we want nn=23

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Auto correlation of Maximal sequence1
140

120

100

80

magnitude---->
60

40

20

-20
-150 -100 -50 0 50 100 150
no of samples---->

Auto correlation of Maximal sequence1

140

120

100
magnitude---->

80

60

40

20

-20
-150 -100 -50 0 50 100 150
no of samples---->

cross correlation of MLS1&MLS2

15

10

5
magnitude---->

-5

-10

-15

-20
-150 -100 -50 0 50 100 150
no of samples---->

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auto correlation of gold sequenceses
140

120

100

magnitude---->
80

60

40

20

-20
-150 -100 -50 0 50 100 150
no of samples---->

3-value checking of goldsequence

15

10

0
magnitude

-5

-10

-15

-20
-150 -100 -50 0 50 100 150
No of samples

Result:-
Thus the random data has been generated at a given data rate and the
outputs have been verified.

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2. Simulation of Rayleigh Fading Channel incorporating speed

of the mobile and power delay profile

Aim:- To Simulate the Rayleigh Fading Channel incorporating speed of

mobile and power delay profile.

Software Required:- Pc with MATLAB 7.0 or above.

Theory:-
Rayleigh fading is a statistical model for the effect of a propagation
environment on a radio signal, such as that used by wireless devices.
Rayleigh fading models assume that the magnitude of a signal that has
passed through such a transmission medium (also called a communications
channel) will vary randomly, or fade, according to a Rayleigh distribution —
the radial component of the sum of two uncorrelated Gaussian random
variables.
Rayleigh fading is viewed as a reasonable model for tropospheric and
ionospheric signal propagation as well as the effect of heavily built-up urban
environments on radio signals. Rayleigh fading is most applicable when
there is no dominant propagation along a line of sight between the
transmitter and receiver. If there is a dominant line of sight, Rician fading
may be more applicable.
Rayleigh fading is a reasonable model when there are many objects in
the environment that scatter the radio signal before it arrives at the receiver.
The Central limit Therom holds that, if there is sufficiently much scatter, the
channel impulse response will be well-modeled as Gaussian process
irrespective of the distribution of the individual components. If there is no
dominant component to the scatter, then such a process will have zero mean
and phase evenly between 0 and 2π radians. The envelope of the channel
response will therefore be Rayleigh.
Calling this random variable R, it will have a probability density function.
Where Ω = E(R2).
Often, the gain and phase elements of a channel's distortion are
conveniently represented as a complex number. In this case, Rayleigh fading

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is exhibited by the assumption that the real and imaginary parts of the
response are modelled by independent and identically distributed zero-mean
Gaussian processes so that the amplitude of the response is the sum of two
such processes.
JAKE’S Model Diagram:-

Jakes' model:-
Jakes popularized a model for Rayleigh fading based on summing
sinusoids. Let the scatterers be uniformly distributed around a circle at
angles αn with k rays emerging from each scatterer. The Doppler shift on ray
n is and, with M such scatterers, the Rayleigh fading of the kth waveform
over time t can be modeled as: .
Here, and the and are model parameters with usually set to zero,
chosen so that there is no cross-correlation between the real and imaginary
parts of R(t): and used to generate multiple waveforms. If a single-path
channel is being modeled, so that there is only one waveform then can be
zero. If a multipath, frequency-selective channel is being modeled so that
multiple waveforms are needed, Jakes suggests that uncorrelated
waveforms are given by: .

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In fact, it has been shown that the waveforms are correlated among
themselves — they have non-zero cross-correlation — except in special
circumstances. The model is also deterministic (it has no random element to
it once the parameters are chosen). A modified Jakes' model chooses
slightly different spacing for the scatterers and scales their waveforms using
Walsh-Hadamard sequences to ensure zero cross-correlation. Setting
And, Results in the following model, usually termed the Dent model or the
modified Jakes model:
The weighting functions Ak(n) are the kth Walsh-Hadamard sequence in
n. Since these have zero cross-correlation by design, this model results in
uncorrelated waveforms. The phases can be initialized randomly and have
no effect on the correlation properties. The Fast Walsh transform can be
used to efficiently generate samples using this model.
The Jakes' model also popularized the Doppler spectrum associated
with Rayleigh fading, and, as a result, this Doppler spectrum is often termed
Jakes' spectrum.

Program:-

% IMPLEMENTATION OF RAYLEIGH FADING CHANNEL

%INCORPORATING SPEED OF MOBILE AND POWER DELAY PROFILE

Clear all;
v=input('MU speed in m/s....5, 10, 15, 20, 25,..>');
numpaths = 10; %number of paths
Fc = 900e6; %carrier frequency
Fs = 4*Fc; %sampling frequency
Ts = 1/Fs; %sampling period
t = [0:Ts:1999*Ts]; %time array
wc = 2*pi*Fc; %radian frequency
%v = 25; %vehicle speed in m/s

ray = zeros(1,length(t)); %received signal

a=1;

for i = 1:numpaths
wd = 2*pi*v*Fc*cos(unifrnd(0,2*pi))/3e8;
ray = ray + a.*cos((wc+wd)*t+unifrnd(0,2*pi,1,length(t)));
end

[rayi rayq] = demod(ray,Fc,Fs,'qam'); %demodulated signal

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env_ray = sqrt(rayi.^2+rayq.^2); %envelope of received signal
subplot(2,2,1)
plot(t*1e9,ray)%plots the rf signal
title('rf signal')
xlabel('time ns')
ylabel('volt')
xlim([0 20])
subplot(2,2,2)
plot(t*1e9,rayi)%plots the inphase component
xlabel('time ns')
ylabel('volt')
xlim([0 20])
title('inphase component')
subplot(2,2,3)
plot(t*1e9,rayq)%plots the quadrature component
title('quadrature component')
xlabel('time ns')
ylabel('volt')
xlim([0 20])
subplot(2,2,4)
plot(t*1e9,env_ray)%plots the envelope
title('envelope')
xlabel('time ns')
ylabel('volt')
xlim([0 20])

%computation of the outage probabilities

power_ray=env_ray.^2;
powerdB=10*log10(power_ray);
mean_power=10*log10(mean(env_ray.^2))
MK=length(env_ray)
for k=1:20;
pow(k)=mean_power-2*k; %threshold power
kps=pow(k);
ratio=10^(kps/10)/10^(mean_power/10);
poutth(k)=1-exp(-ratio);%theoretical outage
count=0;
for ku=1:MK;
power=powerdB(ku);%power in dB
if power <= kps
count=count+1;
else
end;
end;
poutsim(k)=count/MK;%outage probability simulated

end;
figure
plot(pow,poutth,':',pow,poutsim,'--');
xlabel('thrshold power dBm')

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ylabel('outage probability')
legend('poutth','poutsim')

Output results:-
MU speed in m/s....5, 10, 15, 20, 25,..>20

mean_power =

9.8335

MK =
2000

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% jakes model %

clc;
clear all;
N0=8;
N=2*(2*N0+1);
t=250*10^-3;
t1=0:t/(2500):t;
fm=input('enter the dopplear spread frequency fm=');
wm=2*pi*fm;
gamma=1;
fc=900*10^6;
pw1=0;
for s=0:(t/(2500)):t;
xca=0;
xsa=0;
i=1:N0;
wn=wm.*cos(2*pi*i/N);
bt=(pi*i/N0);
for i=1:N0;
x=cos(wn(i)*s);
xca=xca+x.*2*cos(bt(i));
xsa=xsa+x.*2*sin(bt(i));
end
x1=0.707*cos(2*pi*fm*s);
x1ca=x1*2*cos(gamma);
x1sa=x1*2*sin(gamma);
xc=xca+x1ca;
xs=xsa+x1sa;
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y=xs.*sin(2*pi*fc*s)+xc.*cos(2*pi*fc*s);
%disp(y)

z=y.^2;
pw=10*log10(z);
pw1=[pw1 pw];

end

%disp(pw1);
plot(pw1(2:length(pw1)));
title(' rayleigh fading channel by using jakes model');
xlabel('samples');
ylabel('power (db)');
% axis([0 120 -50 20]);
grid;

Output results:-

enter the Doppler spread frequency fm=100Hz

rayleigh fading channel by using jakes model

20

10

-10

-20
power (db)

-30

-40

-50

-60

-70

-80
0 500 1000 1500 2000 2500 3000
samples

Result:-
Thus we have simulated the Rayleigh Fading channel incorporating speed
of the mobile and power delay profile.

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3. SIMULATION OF BPSK SYSTEM OVER AWGN CHANNEL &

FINDING ITS PERFORMANCE WITH BER

Aim: To implement Simulation of BPSK system over AWGN channel &

finding its performance with BER.

Software Required:- Pc with MATLB 7.0 or above

Theory:

BPSK (also sometimes called PRK, phase reversal keying, or 2PSK) is

the simplest form of phase shift keying (PSK). It uses two phases which are
separated by 180° and so can also be termed 2-PSK. It does not particularly
matter exactly where the constellation points are positioned, and in this figure
they are shown on the real axis, at 0° and 180°. This modulation is the most
robust of all the PSKs since it takes the highest level of noise or distortion to
make the demodulator reach an incorrect decision. It is, however, only able to
modulate at 1 bit/symbol and so is unsuitable for high data-rate applications.
The basic block diagram of BPSK system is shown in below

The general form for BPSK follows the equation:

This yields two phases, 0 and π. In the specific form, binary data is often
conveyed with the following signals:

for binary "0"

for binary "1"

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where fc is the frequency of the carrier-wave. Hence, the signal-space can be

represented by the single basis function.

where 1 is represented by and 0 is represented by .

The bit error rate (BER) of BPSK in AWGN can be calculated as:[5]

or

In this program first we generate some random message signal m and

then we convert that message in to binary bits. Based on that binary values
we generate s1,s2 signals.
In the next step we generate some additive white Gaussian noise. When
the modulated signal passes through the channel the noise will be added to
the signal.
The combination of message signal and noise are received at the
receiver. From these corrupted signals we extract the message signal using
bpsk demodulation. Then we calculate the error probability which is nothing
but the ratio of error bits to the total number of bits . Then we calculate the
error complementary function Q and we plot the graphs between probability
of error , Q function and energy of bits to the error probability.

Matlab Program:
clear all;

Eb_No = [1,2,3,4,5,6,7];

No = 1;

Eb = Eb_No * No ;

N=4;

symbol_number = 10000 ;

m = rand(1,symbol_number) ;

[i] = find(m>0.5) ;

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[j] = find(m<=0.5) ;

m(i) = 1 ;

m(j) = 0 ;

for k = 1:length(Eb)

E = 10^(Eb(k)/10) ;

E = sqrt(2*E/N) ;

s1 = ones(1,N)*E ;

s2 = ones(1,N)*(-E) ;

for i = 1:symbol_number

if m(i) == 0

message((i-1)*(N) + 1 : (i-1)*(N) + N) = s2 ;

else

message((i-1)*(N) + 1 : (i-1)*(N) + N) = s1 ;

end

end

received_signal = awgn(message,Eb(k)) ;

for j = 1:N

c1(j) = s1(5-j) ;

c2(j) = s2(5-j) ;

end

Z1 = conv(received_signal, c1) ;

Z2 = conv(received_signal, c2) ;

m_hat = zeros(1,symbol_number); % added by me

for i = 1:symbol_number

m_hat(i) = Z1(i*N - 1) > Z2(i*N - 1) ;

end

error = xor(m_hat,m) ;

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error_number = length(find(error>0))

Pb(k) = error_number/symbol_number ;

end

figure

semilogy(Eb_No,Pb);

title(' Error Prob.. of BPSK ');

xlabel(' Eb/No(dB) ')

ylabel(' Pb(Log Scale) ');

legend(' Prob.. of error ');

Qx = (1/2)*erfc((10.^(Eb/10))/sqrt(2)) ;

figure

semilogy(Eb_No, Pb, 'r+-', Eb_No, Qx) ;

title(' Error Prob.. of BPSK with comparision of Q(SQRT( 2Eb/No)) ');

xlabel(' Eb/No(dB) ')

ylabel(' Pb(Log Scale) ');

legend(' Prob.. of error ', ' Q(SQRT(2Eb/No)) ')

figure

plot(Eb_No,Pb,'r',Eb_No,Qx);

title(' Error Prob... of BPSK ')

xlabel(' Eb/No(dB) ')

ylabel(' Pb ');

legend(' Prob.. of Error ', 'Q(x) ');

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Output Results:

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Result:
Thus Simulation of BPSK system over AWGN channel & finding its
performance with BER

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[Link] EQUALIZATION AT THE RECEIVER TO

REMOVE ISI CAUSED DUE TO LOW CHANNEL BANDWIDTH

Aim:- To implement equalization at the receiver to remove ISI caused due

to low channel bandwidth.
Software Required:- Pc with MATLAB 7.0 or above
Theory:
A channel equalizer is a device that reduces the ISI and thus reduces
the error rate in the demodulated sequence. Channel distortion is caused
due to low channel bandwidth results in Inter symbol Interference(ISI). Many
communication channels including telephone channels, some radio
channels, may generally characterized as band limited linear filters. As a
result of distortion caused by non-ideal channel frequency response
characteristic a succession of pulses transmitted through the channel at
rates comparable to bandwidth are smearged to a point that they are no
longer distinguishable as well defined pulses at receiving terminal instead
they overlap causing Inter Symbol Interference.

Linear Equalizers:

The most common type of equalizers used to reduce ISI are adaptive
equalizers that update their parameters on periodic basis during the
transmission of data so they are capable of tracking slowly time varying
channel response.

Adaptive MSE Equalizer:

A communication system transmits binary symbols through an

ISI channel whose characteristic is given by vector x, as number of taps
2K+1.

K=[0.05 -0.063 0.088 -0.126 -0.25 0.9047 0.25 0.0126 0.038 0.088]

Channel output is the sequence

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where , n<0

is white Gaussian noise sequence with variance.

The equalizer is linear adaptive equalizer based on the MSE criterion.

Its output sequence is

where are equalizer coefficients.

To determine the error probability for the equalizer as a function of SNR

use the first 200 transmitted symbols to the equalizer with use of LMS
algorithm and a step size . Then performing the simulation for
10,000 transmitted symbols for each choice of variance.

For Instance, The gradient vector denoted as for the MSE criterion
found by taking the derivatives of MSE w.r.t each of 2K+1 coefficients is

Adaptive channel equalizers are required for channels whose

characteristic change with time. In such a case the ISI varies with time which
is tracked in channel response and update its coefficients to reduce the ISI.

The algorithm for adjusting the equalizer tap coefficients may be

expressed as

Where = estimate of gradient vector

= estimate of tap coefficient vector

In case of MSE criterion the gradient vector is also expressed as

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An estimate of gradient vector at the kth iteration

Where ; = difference between desired output from equalizer at the kth

time instant and the actual output z(kT) and denote the column vector of
2K+1 received signal values contained in equalizer at time instant K.

Error signal

Where

Adaptive algorithm for optimizing the tap coefficient based on MSE

criterion has the block diagram that adapts its tap coefficients according to

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The difference between desired output and actual output from

equalizer used to form the error signal .The error is scaled by step size
parameter and scaled error signal multiplies the received signal values

at (2K+1) taps. The products at the

(2K+1) taps are then added to previous values of taps coefficients to obtain
the updated tap coefficients. The computation is repeated at each new
symbol is received. Thus the equalizer coefficients are updated at the
symbol rate.

Initially the adaptive equalizer is trained by the transmission if known

pseudo-random sequence over the channel. At the demodulator the
equalizer employs the known signal to adjust its coefficients upon initial
adjustment the adaptive equalizers switches from a training mode to a
decision directed mode in which case the decisions at the output of the
detector are sufficiently reliable so that the error signal formed by computing
the difference between the detector output and the equalizer output.

Where = Output of detector.

Matlab Program:

%%% echo on
clear all;
close all;
N = 20000;
Nt = [1000000 1000000 1000000 1000000 1000000 5000000 10000000];
delta = 0.0045;
K = 5;
actual_isi = [0.05 -0.063 0.088 -0.126 -0.25 0.9047 0.25 0 0.126 0.038
0.088];
snr_db = 2:2:14;
snr = 10.^(snr_db/10);
l_snr = size(snr,2);
var = 1/2./snr;
sigma = sqrt(var);
pe = zeros(1,l_snr);
for idx = 1:l_snr
% [A] the training sequence
training_s = ones(1,N);
for i = 1:N
if(rand<0.5)
training_s(i) = -1 ;
end

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end
% The channel output
y = filter(actual_isi,1,training_s);
noise = zeros(1,N);
for i = 1:2:N
noise(i) = random('Normal',0,sigma(idx));
noise(i+1) = noise(i);
end
y = y +noise;
% The equalization part follows
estimated_c = [0 0 0 0 0 1 0 0 0 0 0]; % initial estimate of isi
for k = 1:N-2*K
y_k = y(k:k+2*K);
z_k = estimated_c*y_k';
e_k = training_s(k) - z_k;
estimated_c = estimated_c + delta*e_k*y_k;
end
% (B) the transmitted information sequence
info = ones(1,Nt(idx));
for i = 1:Nt(idx)
if(rand<0.5)
info(i) = -1;
end
end
% The channeled output
y = filter(actual_isi,1,info);
noise = sigma(idx)*randn(1,Nt(idx));
y = y + noise;
% The equalization part
count = 0;
err_count = 0;
z_k_vec = ones(1,Nt(idx) - 2*K);
for k = 1:Nt(idx)-2*K;
y_k = y(k:k+2*K);
z_k = estimated_c*y_k';
if z_k<0
z_k_vec(k) = -1 ;
end
err_count = err_count + 0.5*abs(info(k)-z_k_vec(k));
end
pe(idx) = err_count/length(z_k_vec);
clear y; clear noise
end
% Plot the results
semilogy(snr_db,pe)
grid
ylabel(' pe ')
xlabel(' snr(db) ')

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Output Results:

Result:
Implementing equalization at the receiver to remove ISI caused
due to low channel bandwidth is simulated and results are verified.

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5. IMPLEMENTATION OF RAKE RECEIVER AND FINDING ITS

PERFORMANCE THROUGH BER CURVE.

Aim:- To implement Rake Receiver and to find its performance through

BER curve.
Software Required:- Pc with MATLAB 7.0 or above.
Theory:-
A rake receiver is a radio receiver designed to counter the effects of
multipath fading. It does this by using several "sub-receivers" called fingers,
that is, several correlators each assigned to a different multipath component.
Each finger independently decodes a single multipath component; at a later
stage the contribution of all fingers are combined in order to make the most
use of the different transmission characteristics of each transmission path.
This could very well result in higher signal-to-noise ratio (or Eb/N0) in a
multipath environment than in a "clean" environment.
The multipath channel through which a radio wave transmits can be
viewed as transmitting the original (line of sight) wave plus a number of
multipath components. Multipath components are delayed copies of the
original transmitted wave traveling through a different echo path, each with a
different magnitude and time-of-arrival at the receiver. Since each
component contains the original information, if the magnitude and time-of-
arrival (phase) of each component is computed at the receiver (through a
process called channel estimation), then all the components can be added
coherently to improve the information reliability.
The rake receiver is so named because it reminds the function of a
garden rake, each finger collecting symbol energy similarly to how tines on a
rake collect leaves.
Rake receivers are common in a wide variety of CDMA and W-CDMA
radio devices such as mobile phones and wireless LAN equipment.

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Rake Receiver Diagram:-

Rake receiver:-
If, in a mobile radio channel reflected waves arrive with small relative
time delays, self interference occurs. Direct Sequence (DS) Spread
Spectrum is often claimed to have particular properties that makes it less
vulnerable to multipath reception. In particular, the rake receiver architecture
allows an optimal combining of energy received over paths with different. It
avoids wave cancellation (fades) if delayed paths arrive with phase
differences and appropriately weighs signals coming in with different signal-
to-noise ratios.
The rake receiver consists of multiple correlators, in which the receive
signal is multiplied by time-shifted versions of a locally generated code
sequence. The intention is to separate signals such that each finger only
sees signals coming in over a single (resolvable) path. The spreading code
is chosen to have a very small autocorrelation value for any nonzero time
offset. This avoids crosstalk between fingers. In practice, the situation is less
ideal. It is not the full periodic autocorrelation that determines the crosstalk
between signals in different fingers, but rather two partial correlations, with
contributions from two consecutive bits or symbols. It has been attempted to
find sequences that have satisfactory partial correlation values, but the
crosstalk due to partial (non-periodic) correlations remains substantially
more difficult to reduce than the effects of periodic correlations.

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The rake receiver is designed to optimally detect a DS-CDMA signal
transmitted over a dispersive multipath channel. It is an extension of the
concept of the matched filter.
In a multipath channel, delayed reflections interfere with the direct
signal. However, a DS-CDMA signal suffering from multipath dispersion can
be detected by a rake receiver. This receiver optimally combines signals
received over multiple paths.
Like a garden rake, the rake receiver gathers the energy received over
the various delayed propagation paths. According to the maximum ratio
combining principle, the SNR at the output is the sum of the SNRs in the
individual branches, provided that
 we assume that only AWGN is present (no interference)
 codes with a time offset are truly orthogonal

Matlab Program:
clc;
clear all;
e=1;
snrdb=0:1:20;
n=1000;
L=length(snrdb);N=7;code=[1 1 1 -1 -1 1 -1];G=7;chipsperslot=7;
bfad=randn(1,10);%[0.1 -0.2 0.3];
bfadt=conj(fliplr(bfad));%[0.3 -0.2 0.1];

t=0:ts:((n*tb/2)-ts);
sp=10;
for ii=1:L
pe1=0;
snr=10^((ii-1)/10);
sigmao=sqrt(sum(bfad.*bfadt)*G/snr);
%%%data generater
d=rand(1,n)>.5;
d1=2*d-1;
%%%%SPREADING THE DATA
% out=conv(d1,code);
out=[];
for i=1:n
out=[out d1(i)*code];
end

K=length(bfad);

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mfdata=conv(out,bfad);
% %% AWGN CHANNEL
w=sigmao.*randn(1,length(mfdata));
rx=w+mfdata;
% %%AT RECEIVER SIDE
%%%matched filter
mfdata1=conv(rx,conj(fliplr(bfad)));
%delay=length(code)+length(bfad);
getdata=mfdata1(3:length(mfdata1)-2);
%getdata=mfdata1(delay:delay+chipsperslot);
%getdata=getdata(K:length(code):length(getdata));
rdata=getdata;
for i=1:n
rdata(i)= getdata(((K)+(i-1)*N));
end
rd=rdata>0;
for i=1:n
if(rd(i)==d(i))
else
pe1=pe1+1;
end
end

pe2(ii)=pe1;
%%probability error in simulated system
BER(ii)=pe2(ii)/(n);
%%probability error in theoretical function
geta=G*(sum(bfad.*bfadt)/sigmao^2);
qf4(ii)=0.5*(1-sqrt(geta/(1+geta)));
end
disp(pe2);
disp(qf4);
disp('bit error rate is');
disp(BER);
semilogy(snrdb,BER,'-o',snrdb,qf4,'-+');
hold all;
%axis([0 14 10e-7 10e-1]);
legend(' praticalBER','theoreticalBER');
title('BER performance of RAKE Receiver')
xlabel('snrdb--->');
ylabel('Pe--->');
grid;

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Output Result:-

Result:-

Thus we have implemented the Rake Receiver and its performance has
been calculated through BER curve.

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6. IMPLEMENTATION OF LMS ALGORITHM TO ESTIMATE

THE ORIGINAL DATA WHEN IT IS CORRUPTED BY NOISE
AND CHANNEL

Aim:- To implement the LMS algorithm to estimate the original data when it
is corrupted by noise and channel.
Software Required:- Pc with MATLAB 7.0 or above.
Theory:-
Adaptive Noise Cancellation (ANC) and its Applications. Adaptive noise
Cancellation is an alternative technique of estimating signals corrupted by
additive noise or interference. Its advantage lies in that, with no apriori
estimates of signal or noise, levels of noise rejection are attainable that
would be difficult or impossible to achieve by other signal processing
methods of removing noise.

Its cost, inevitably, is that it needs two inputs - a primary input containing
the corrupted signal and a reference input containing noise correlated in
some unknown way with the primary noise. The reference input is
adaptively filtered and subtracted from the primary input to obtain the signal
estimate. Adaptive filtering before subtraction allows the treatment of
inputs that are deterministic or stochastic, stationary or time-variable.

For the application of ANC, it is desired to choose an algorithm which is

computationally very fast. Taking this into consideration, LMS algorithm
became an obvious choice over RLS.

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In the given program we designed the LMS algorithm with order K = 5 so

we require total 11 tap weights and actual_isi, sigma and delta are taken as
fixed. With these specifications first we generate the information signal info
then with the help of gngauss function some random noise will be generated
and we add that noise to the info then we get corrupted signal which is called
as primary signal.

Matlab Program:
echo on

N = 500 ;
K=5;
actual_isi = [0.05 -0.063 0.088 -0.126 -0.25 0.9047 0.25 0 0.126 0.038 0.088] ;
sigma = 0.01 ;
delta = 0.115 ;
Num_of_realizations = 1000 ;
mse_av = zeros(1,N-2*K) ;
for j = 1:Num_of_realizations
% the information sequence
for i = 1:N
if(rand<0.5)
info(i) = -1 ;
else
info(i) = 1 ;
end
echo off ;

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end
if(j == 1) ;
echo on ;
end
% the channel output
y = filter(actual_isi,1,info);
for i = 1:2:N, [noise(i) noise(i+1)] = gngauss(sigma) ; end;
y = y+noise ;
% Now the equalization part follows
estimated_c = [ 0 0 0 0 0 1 0 0 0 0 0] ; % initial estimate of ISI
for k = 1:N-2*K,
y_k = y(k:k+2*K) ;
z_k = estimated_c*y_k.' ;
e_k = info(k) - z_k ;
estimated_c = estimated_c + delta*e_k*y_k ;
mse(k) = e_k^2 ;
echo off ;
end ;
if(j == 1) ; echo on ; end
mse_av = mse_av + mse ;
echo off ;
end ;
echo on ;
mse_av = mse_av/Num_of_realizations ; % mean-squared error versus
iteraions
% plotting commands follow.
figure
plot(y)
ylabel(' data corrupted by noise and channel ')
xlabel(' number of iterations ')
figure
plot(mse_av)
ylabel(' average mean square error ')
xlabel(' number of iterations ')

gngauss sub function:

function [gsrv1, gsrv2] = gngauss(m,sgma)

% [gsrv1, gsrv2] = gngauss(m,sgma)

% [gsrv1, gsrv2] = gngauss(sgma)
% [gsrv1, gsrv2] = gngauss

% GNGAUSS generates two independent Gaussian random variables with

% mean m and standard deviation sgma. If one of the input
% arguments is missing, it takes the mean as 0.
% If neither the mean nor the variance is given, it generates two
% standard Gaussian random variables.

if nargin == 0,

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m = 0 ; sgma = 1 ;
elseif nargin ==1,
sgma = m; m =0 ;
end ;
u = rand ; % a uniform random variable in (0,1)
z = sgma*(sqrt(2*log(1/(1-u)))) ; % a Rayleigh distributed random variable
u = rand ; % another uniform random variable in (0,1)
gsrv1 = m + z*cos(2*pi*u) ;
gsrv2 = m + z*sin(2*pi*u) ;

Output Result:-

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Result:-

Thus we have implemented the LMS algorithm to estimate the original

data when it is corrupted by noise and channel.

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7. IMPLEMENTATION OF R.L.S ALGORITHM TO

ESTIMATE THE ORIGINAL DATA WHEN IT IS
CORRUPTED BY NOISE AND CHANNEL

Aim: To implement R.L.S algorithm to estimate the original data when it is

corrupted by noise & channel.

Software Required: Pc with Matlab 7.0 or above

THEORY: Another popular noise cancellation adaptive filter involves

using the recursive least squares method (RLS). Unlike the LMS method,
the RLS method finds the filter coefficients that minimize a weighted linear
least squares function relating to the input signals instead of trying to
reduce the mean square error. It is deterministic (i.e. no randomness is
involved of future states of the system).The RLS method is faster than the
LMS method at the expense of more complex computations

The idea behind RLS filters is to minimize a cost function

by appropriately selecting the filter coefficients , updating the filter
as new data arrives. The error signal and desired signal are
defined in the negative feedback diagram below:

parameters:

p = filter order
forgetting factor
value to initialize p(0)
n = length of the desired signal
Initialization:

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w = [ w(0) = 0 ,w(1) = 0 ,.......,w(p) = 0]
computations:

Matlab Program:
% Recursive Least Squares
% Importing Data
% Vectors must have same length and sampling frequency
% signal = source input
% noise = noise to add to source input
% ref = reference signal (e.g. delayed noise)
% lambda = forgetting factor,
% M = filter order
% delta = initial value for P; use a value in the 10^-3 range

% Output arguments:
% xi = output
% w = final filter coefficients

% Initialization
clear all;
close all;
clc;
lambda = 1;
M =5;
delta = .0005;
w=zeros(M,1); k=zeros(M,1);
t=1:.025:5;
signal=5*sin(2*3.*t);
ref=5*sin(2*50*3.*t+3/20);
noise=random('normal',0,t);
primary = signal + noise;
subplot(5,1,1);
plot(t,signal);
ylabel('signal');
subplot(5,1,2);

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plot(t,noise);
ylabel('noise');

subplot(5,1,3);
plot(t,primary);
ylabel('primary');
% eye(M) gives an MxM identity matrix
% dividing by delta replaces the 1's in the identity matrix with delta^-1
P=eye(M)/delta;

% Input Signal Length

N=length(ref);
% Set output as desired signal
xi=primary;

% Loop, RLS
for n=M:N
uvec=ref(n:-1:n-M+1);
k=lambda^(-1)*P*uvec'/(1+lambda^(-1)*uvec*P*uvec');
% xi is output
xi(n)=primary(n)-w'*uvec';
% Recursive equation to minimize cost function (difference between the
% desired and input signals)
w=w+k*conj(xi(n));
P=lambda^(-1)*P-lambda^(-1)*k*uvec*P;

end

% Calculate Mean Squared Error

MSE = xi - signal(1:length(xi));
MSE = MSE.^2;
subplot(5,1,4);
plot(t,xi);
ylabel('noise cancelled output');

% Time to use for plotting in time domain

% 0-5 seconds with 1/fs increments

subplot(5,1,5);
plot(t(1:length(MSE)), MSE)
xlabel('Time (seconds)')
ylabel('Error')
title('Mean Squared Error')

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OUTPUT RESULTS:

Result:

Thus R.L.S algorithm to estimate the original data when it is

corrupted by noise & channel is implemented by using matlab software.

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