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DSP NEW

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Exp.

No:1
Date:
GENERATION OF SIGNALS
AIM:
To write a Matlab program and generate the following discrete time
signals.
1. Unit step
2. Unit Impluse
3. Unit ramp
4. Experintial
5. sine and cosine.
SOFTWARE REQUIRED:
MATLAB version 7.2
THEORY:
A Discrete time signal is discrete in time and continuous in amplitude.
Unit Step u(n)
The amplitude of the signal is one for
0 ≥ n
and zero for
0 ≤ n

¹
'
¹


·
0 , 0
0 , 1
) (
n for
n for
n u
Unit Ramp r(n)
This is the signal whose amplitude increases linearly with time
¹
'
¹


·
0 , 0
0 ,
) (
n for
n for n
n r
Exponential e(n)
It varies exponentially with time
¹
'
¹


·
0 , 0
0 , ) exp(
) (
n for
n for n
n e
ALGORITHM:
Step 1: Start the program
Step 2: Define the amplitude sequence
Step 3: Give the necessary label
Step 4: Draw the amplitude versus time period graph
Step 5: Stop the process
The above steps are repeated for the generation of all the above
mentioned signals by altering the time instant and the amplitude values.
Program:
%UNIT IMPULSE
s=1,d=0;
subplot(2,2,1);
stem(d,s);
grid on;
xlabel('n---->');
ylabel('Amp---->');
title('UNIT IMPULSE');

%UNIT IMPULSE
s=[zeros(1,10) 1 zeros(1,10)];
subplot(2,2,2);
plot(s);
grid on;
xlabel('n---->');
ylabel('Amp---->');
title('UNIT IMPULSE');

%UNIT IMPULSE
s=[zeros(1,10) 1 zeros(1,10)];
d=-10:1:10;
subplot(2,2,3);
stem(d,s);
grid on;
xlabel('n---->');
ylabel('Amp---->');
title('UNIT IMPULSE');

%UNIT IMPULSE
s=[zeros(1,1000) 1 zeros(1,1000)];
d=-1000:1:1000;
subplot(2,2,4);
plot(d,s);
grid on;
xlabel('n---->');
ylabel('Amp---->');
title('UNIT IMPULSE');
%UNIT STEP SEQUENCE
clear all;
N=5;
X=ones(1,N);
n=0:1:N-1;
figure,subplot(2,2,1);
plot(n,X);
xlabel('n---->'),ylabel('X(n)---->');
title('UNIT STEP SEQUENCE');

%UNIT STEP SEQUENCE
x=-50:50;
y=[zeros(1,50) ones(1,51)];
subplot(2,2,2);stem(x,y);
xlabel('n---->'),ylabel('X(n)---->');
title('UNIT STEP SEQUENCE');

%UNIT RAMP
n=1:1:20;
subplot(2,2,3);
plot(n,n);
grid on;
xlabel('n---->');
ylabel('Amp---->');
title('UNIT RAMP');

%GROWING EXPONENTIAL SEQUENCE 1
n=0:1:10;
x1=exp(n);
subplot(2,2,4);
plot(n,x1);
xlabel('n---->');
ylabel('x1(n)---->');
title('GROWING EXPONENTIAL 1');
%GROWING EXPONENTIAL SEQUENCE 2
x2=-50:50;
y=[zeros(1,50) exp([0:0.1:5])];
figure
subplot(2,2,1);
stem(x2,y);
xlabel('n---->');
ylabel('x2(n)---->');
title('GROWING EXPONENTIAL 2');

%SINUSOIDAL SIGNAL
n=0:.5:(5*pi);
x3=sin(n);
subplot(2,2,3);
plot(n,x3);
xlabel('n---->');
ylabel('x3(n)---->');
title('SINE WAVE');
RESULT:
Thus a MATLAB program is written for generating of signals and
output is obtained.
Exp.No.2
Date:
LINEAR AND CIRCULAR CONVOLUTION
AIM:
To do linear and circular convolution of the discrete time sequence
and plot the sequences.
SOFTWARE REQUIRED:
MATLAB version 7.2
THEORY:
LINEAR CONVOLUTION
It is a generation used to find output and LDI / DT system in
time domain.
Let x(n) be the input to the LTI system and H(n) is the impulse
response of the LTI system then the output y(n) of the system is given
by

·
·
− ⋅ ·
∗ ·
1
0
) ( ) ( ) (
) ( ) ( ) (
µ
k
k n x k h n y
n h n x n y
The above equation is called linear convolution equation. This
equation is involved using convolution.
Let n
1
be the sequence of input
n
2
be the sequence of impulse response
The length of the output sequence
1
2 1 3
− + · n n n
CIRCULAR CONVOLUTION:
In circular convolution it is necessary the sequence must be same. It
is similar to linear convolution except the shifting is performed in circular
convolution.

·
·
− ⋅ ·
∗ ·
1
0
2 1
) ( ) ( ) (
) ( ) ( ) (
µ
k
k n x k x n y
n h n x n y
ALGORITHM:
Step 1: Start the program
Step 2: Get the first sequence
Step 3: Get the second sequence
Step 4: Convolution of both the sequence
Step 5: Plot the sequence
Step 6: Stop the process
LINEAR CONVOLUTION
clc;
clear all;
close all;
x=input('enter the x sequence');
n1=length(x);
o1=input('enter the origin of x');
h=input('enter the h sequence' );
n2=length(h);
o2=input('enter the origin of h');
o3=o1+o2;
yt=o3+n1+n2-2;
y=conv(x,h);
t=o3:1:yt;
figure ,stem(t,y);
tx=o1:1:n1+o1-1;
ty=o2:1:n2+o2-1;
figure , subplot(2,1,1),stem(tx,x);
subplot(2,1,2),stem(ty,h);
INPUT COMMAND :
enter the x sequence[ ]
enter the origin of x0
enter the h sequence[ ]
enter the origin of h-1
CIRCULAR CONVOLUTION
clc;
clear all;
close all;
x1=input('enter the x1 sequence');
n1=length(x1);
o1=input('enter the origin of x1');
x2=input('enter the x2 sequence' );
n2=length(x2);
o2=input('enter the origin of x2');
if (n1>n2)
x2=[x2 zeros(1,n1-n2)];
N=n1;
elseif (n1<n2)
x1=[x1 zeros(1,n2-n1)];
N=n2;
end
N=max(n1,n2);
y=cconv(x1,x2,N);
yt=0:1:N-1;
tx1=o1:1:n1-1;
tx2=o2:1:n2-1;
figure,stem(tx1,x1);
figure,stem(tx2,x2); HC
figure,stem(yt,y);
INPUT COMMAND :
enter the x1 sequence[ ]
enter the origin of x10
enter the x2 sequence[ ]
enter the origin of x20
RESULT:
Thus a MATLAB program for Linear and Circular Convolution has
been written and output is verified.
Exp.No.4
Date:
SPECTRUM ANALYSIS USING FFT
AIM:
To write a Matlab program to find spectrum analysis of discrete and
continuous time FFT
SOFTWARE REQUIRED:
MATLAB version 7.2
THEORY:
Fourier transform used for frequency analysis of the signal. It is given
as


−∞ ·


⋅ ·
n
n
e n x n x DFT
π
) ( ) ( ,
Here r is the frequency and has image of 0 to 27. These are several
methods to find DFT and IDFT. FFT is fast and efficient method. It is a
finite duration sequence and is obtained by sampling transfer function
) (

e x
at N equally spaced points over an interval ‘ω’.
π
π
ω
ω
2 0 ,
2
) ( ) (
3
≤ ≤ · · k
N
k
e x k x
IDFT, ∑

·
− ≤ ≤

⋅ ·
1
0
2
1 0 , ) (
1
) (
N
k
N Kn
N n e K X
N
n X
π
ALGORITHM:
Step 1: Start the program
Step 2: Initialize the give sequence
Step 3: Assign the length of the sequence
Step 4: Plot the sequence of magnitude and phase.
Step 5: Stop the process
FFT (DISCRETE)
clear all
x=input('enter the sequence');
N=length(x);
y=fft(x,N);
k=0:1:N-1;
Magy=abs(y);
Angy=angle(y);
Subplot(2,1,1),stem(k,Magy);
Xlabel(‘k’),ylabel(‘Magnitude of y’);
Subplot(2,1,2), stem(k,Angy);
Xlabel(‘k’),ylabel(‘Angle of y’);
INPUT:
Enter the sequence[ ]

FFT PSD (DISCRETE)
clear all
x=input('enter the sequence');
N=length(x);
y=fft(x,N)
k=0:1:N-1;
Magy=abs(y)
Angy=angle(y)
p=y.*conj(y)/N
Subplot(2,1,1),stem(k,Magy);
xlabel('k'),ylabel('Magnitude of y');
Subplot(2,1,2), stem(k,Angy);
xlabel('k'),ylabel('Angle of y');
figure,
stem(k,p);
INPUT:
Enter the sequence[ ]
FFT PSD(CONTINOUS)
clear all
fs=input('Enter the Sampling Frequency');
N=input('Enter the length of DFT');
f=input('Enter the frequency of the input signal');
k=0:1:N-1;
x=input('Enter the input signal');
%x=sin(2*pi*f*k*(1/fs));
y=fft(x,N)
Magy=abs(y)
Angy=angle(y)
p=y.*conj(y)/N
Subplot(2,1,1),stem(k,Magy);
xlabel('k'),ylabel('Magnitude of y');
Subplot(2,1,2), stem(k,Angy);
xlabel('k'),ylabel('Angle of y');figure,
stem(k,p);
INPUT:
Enter the Sampling Frequency
Enter the length of DFT
Enter the frequency of the input signal
Enter the input signal
FFT PSD (CONTINOUS)—VERIFICATION USING
AUTOCORRELATION
clear all
fs=input('Enter the Sampling Frequency');
N=input('Enter the length of DFT');
f=input('Enter the frequency of the input signal');
k=0:1:N-1;
x=input('Enter the input signal');
%x=sin(2*pi*f*k*(1/fs));
y=fft(x,N)
p=xcorr(x,x)
y1=fft(p,N)
Magy=abs(y)
Angy=angle(y)
p=y.*conj(y)/N
Subplot(2,1,1),stem(k,Magy);
xlabel('k'),ylabel('Magnitude of y');
Subplot(2,1,2), stem(k,Angy);
xlabel('k'),ylabel('Angle of y');
figure,
stem(k,y1);
RESULT
Thus a MATLAB program for Spectrum analysis using FFT has been
written and output has been verified.
Exp.No.3
Date:
AUTOCORRELATION AND CROSS CORRELATION
AIM:
To find the autocorrelation of given signal and cross correlation of the
same signal.
SOFTWARE REQUIRED:
MATLAB version 7.2
THEORY:
Autocorrelation of the given functions is calculated using the formula,
1 1
) 1 ( ) 1 ( ) ( ) (
≤ ≤ −
− ≤ ≤ − − − ⋅ ·


−∞ ·
n
N n N where n k x k x R
k
xx
Cross correlation of the given function is calculated using the formula,
1 1
) 1 ( ) 1 ( ) ( ) ( ) (
≤ ≤ −
− ≤ ≤ − − − ⋅ ·


−∞ ·
n
N n N where n k y k x n R
k
xy
ALGORITHM:
Step 1: Start the program
Step 2: Enter the program
Step 3: Run the program
Step 4: Calculate the cross correlation of sequence
Step 5: Calculate the autocorrelation of sequence
Step 6: Plot both the sequence
Step 7: Stop the process
Program:
AUTO CORRELATION
clc;
clear all;
close all;
x=input('Enter the input sequence');
y=xcorr(x)
xl=length(x);
xt=0:1:xl-1;
stem(xt,x);
xlabel('n---->');
ylabel('Amplitude');
title('Input sequence');
yt=-(xl-1):1:(xl-1);
figure,
stem(yt,y);
xlabel('n---->');
ylabel('Amplitude');
title ('Output sequence');
INPUT :
Enter the input sequence[ ]
CROSS CORRELATION
clc;
clear all;
close all;
x=input('Enter the first input sequence');
y=input('Enter the second input sequence');
z=xcorr(x,y)
x1=length(x);
xt=0:1:x1-1;
subplot(2,1,1),stem(xt,x);
xlabel('n---->');
ylabel('Amplitude');
title('first Input sequence');
subplot(2,1,2),stem(xt,y);
xlabel('n---->');
ylabel('Amplitude');
title('second Input sequence');
yt=-(x1-1):1:(x1-1);
figure,
stem(yt,z);
xlabel('n---->');
ylabel('Amplitude');
title('Output sequence');

INPUT:
Enter the first input sequence[ ]
Enter the second input sequence[ ]
RESULT
Thus a MATLAB program for autocorrelation and cross correlation
has been written and output has been verified.
Exp.No.6a FILTER DESIGN ANALYSIS –IIR FILTER
Date:
DESIGN OF BUTTERWORTH IIR FILTER
AIM:
To design the IIR filter and plot the frequency response.
SOFTWARE REQUIRED:
MATLAB version 7.2
THEORY:
Digital filters are used for two digital purposes
(i) Separating a signal that has been combined.
(ii) Distortion of signal that has been distorted in same way.
IIR filters are an efficient way of achieving along impulse response
without having to perform a long convolution. They execute very rapidly
but have less performance and flexibility than other digital filter.
Since the impulse response is composed of decaying exponential
signal, these signal filters are called as IIR filters. They are called as receiver
filter because digital filter are made up of recursion. Recursion filters are
previously calculated values from the output besides part from the output.
These filters are defined as with correspondence with analog filter between
butterworth and chebyshev filter.
The impulse response of these filter are composed of sinusoidal that
exponentially decay in amplitude of the input is a simple impulse. In
principle, this mares the impulse response infinitely long. However, the
amplitude eventually drop below the round of noise of the system. The
remaining samples are ignored because of the characterization of filter are
called IIR filter.
ALGORITHM:
Step 1: Start the program
Step 2: Assign the given specification to variables
Step 3: Compute the order of the filter using in-built
function
Step 4: Computer the function of co-efficient using
appropriate function
Step 5: Plot the frequency response of IIR filter
Step 6: Stop the process
IIR LOW PASS FILTER WITH BUTTERWORTH
CHARACTERISTICS (using Impulse Invariant
method)
Design an IIR low pass filter using Impulse invariant method and
MATLAB. The low pass filter with Butterworth characteristic is
required to meet the following specifications :
Cutoff frequency = 150Hz
Sampling frequency = 1.28KHz
Filter order = 2
a) Determine, using the impulse invariant method and
MATLAB,
i) The coefficients, poles and zeros of the
frequency-scaled analog filter.
ii)The coefficients, poles and zeros of the IIR
discrete filter. Write down its transfer function.
b) Plot the magnitude-frequency response and the pole-
zero diagram of the discrete filter.
---------------------------------------------------------
%BUTTERWORTH LPF USING IMPULSE INVARIANT METHOD

clc;
Fs=1280; %sampling freq
fc=150; %cutoff freq
WC=2*pi*fc; %cutoff freq in radian
N=2;
[b,a]=butter(N,WC,'s')%create an analog filter
%[b,a] = butter(n,Wn,'s')designs an order n lowpass analog Butterworth
filter with angular cutoff frequency Wn rad/s. It returns the filter coefficients
in the length n+1 row vectors b and a,in descending powers of s, derived
from this transfer function:
H(s)=[b(1)s^n+b(2)s^n-1+....+b(n+1)]/
[s^n+a(2)s^n-1+....+a(n+1)]
%[b,a] = butter(n,Wn) designs an order n lowpass digital Butterworth filter
with normalized cutoff frequency Wn. It returns the filter coefficients in
length n+1 row vectors b and a, with coefficients in descending powers of z.
H(z)=[b(1)+b(2)z^-1+....+b(n+1)z^-n]/
[1+a(2)z^-1+....+a(n+1)z^-n]
[z,p,k]=butter(N,WC,'s')%returns the zeros and poles in length n column
vectors z(zeros) and p(poles), and the gain in the scalar k.
[bz,az]=impinvar(b,a,Fs)%determine the coeff of IIR filter
%[bz,az] = impinvar(b,a,fs) creates a digital filter with numerator and
denominator coefficients bz and az, respectively, whose impulse response is
equal to the impulse response of the analog filter with coefficients b and
a,scaled by 1/fs. If you leave out the argument fs,or specify fs as the empty
vector [],it takes the default value of 1 Hz.
subplot(2,1,1)
[H,f]=freqz(bz,az,512,Fs);%[h,f] = freqz(b,a,f,fs) returns the frequency
response vector h and the corresponding frequency vector f for the digital
filter whose transfer function represented in the vectors b and a, respectively.
The frequency response vector h is calculated at the frequencies(in Hz)
supplied in the vector f. The vector f can be any length. For this syntax, the
frequency response is calculated using the sampling frequency specified by
the scalar fs (in hertz). The frequency vector f has values ranging from 0 to
fs/2Hz.
plot(f,20*log10(abs(H))),grid
xlabel('Frequency(Hz)')
ylabel('Magnitude Response(dB)')
subplot(2,1,2)%plot ploe-zero diagram
zplane(bz,az)
zz=roots(bz) %Determine poles and zeros
pz=roots(az)
output : BUTTERWORTH LPF USING IMPULSE INVARIANT
METHOD

b =
1.0e+005 *
0 0 8.8826
a =
1.0e+005 *
0.0000 0.0133 8.8826
z =
Empty matrix: 0-by-1
p =
1.0e+002 *
-6.6643 + 6.6643i
-6.6643 - 6.6643i
k =
8.8826e+005
bz =
0 0.3078 0
az =
1.0000 -1.0308 0.3530
zz =
0
pz =
0.5154 + 0.2955i
0.5154 - 0.2955i
IIR LOW PASS FILTER WITH BUTTERWORTH
CHARACTERISTICS (using Bilinear Z-Transform method)
Design an IIR low pass filter using BiLinear Z-Transform method and
MATLAB. The low pass filter with Butterworth characteristic is
required to meet the following specifications :
Cutoff frequency = 150Hz
Sampling frequency = 1.28KHz
Filter order = 2
a) Determine, using the BiLinear Z-Transform method
and MATLAB,
i) The coefficients, poles and zeros of the
discrete filter. Write down its transfer function.
b) Plot the magnitude-frequency response and the pole-
zero diagram of the discrete filter.
---------------------------------------------------------
%BUTTERWORTH LPF USING BLT METHOD
clc;
Fs=1280;%sampling freq
fc=150;%cutoff freq
N=2;

FN=Fs/2;
Fc=fc/FN;%normalized cutoff frequency
[bz,az]=butter(N,Fc)%create an analog filter %[b,a] = butter(n,Wn) designs
an order n lowpass digital Butterworth filter with normalized cutoff
frequency Wn. It returns the filter coefficients in length n+1 row vectors b
and a, with coefficients in descending powers of z.
H(z)=[b(1)+b(2)z^-1+....+b(n+1)z^-n]/
[1+a(2)z^-1+....+a(n+1)z^-n]
[z,p,k]=butter(N,Fc)%returns the zeros and poles in length n column vectors
z(zeros) and p(poles), and the gain in the scalar k.
subplot(2,1,1)
[H,f]=freqz(bz,az,512,Fs);%[h,f] = freqz(b,a,f,fs) returns
the frequency response vector h and the corresponding frequency vector f for
the digital filter whose transfer function represented in the vectors b and a,
respectively. The frequency response vector h is calculated at the
frequencies(in Hz) supplied in the vector f. The vector f can be any length.
For this syntax, the frequency response is calculated using the sampling
frequency specified by the scalar fs (in hertz). The frequency vector f has
values ranging from 0 to fs/2Hz.
plot(f,abs(H)),grid
xlabel('Frequency(Hz)')
ylabel('Magnitude Response(dB)')
subplot(2,1,2)%plot ploe-zero diagram
zplane(bz,az)
zz=roots(bz); %Determine poles and zeros
pz=roots(az);
Output: BUTTERWORTH LPF USING BLT METHOD
bz =
0.0878 0.1756 0.0878
az =
1.0000 -1.0048 0.3561
z =
-1
-1
p =
0.5024 + 0.3220i
0.5024 - 0.3220i
k =
0.0878
IIR BAND PASS FILTER WITH BUTTERWORTH
CHARACTERISTICS (using Bilinear Z-Transform method)
Design an IIR Band pass filter using Bilinear Z-Transform method and
MATLAB. The Band pass filter with Butterworth characteristic is
required to meet the following specifications :
Pass band = 200–300Hz
Sampling frequency = 2KHz
Filter order = 8
a) Determine, using the BiLinear Z-Transform method
and MATLAB,
i) The coefficients, poles and zeros of the
Discrete filter. Write down its transfer function.
b) Plot the magnitude-frequency response and the pole-
zero diagram of the discrete filter.
---------------------------------------------------------
%BUTTERWORTH BPF USING BLT METHOD
Fs=2000;
FN=Fs/2;%Sampling freq
fc1=200/FN;
fc2=300/FN;
[b,a]=butter(4,[fc1,fc2]) %create/digitize analog filter %Filter order used in
the m-file is half that specified in the design problem : for BPF and BSF, the
order is 2N.
[z,p,k]=butter(4,[fc1,fc2])
subplot(2,1,1) %plot magnitude-frequency response
[H,f]=freqz(b,a,512,Fs);
plot(f,abs(H))
xlabel('Frequency (Hz)')
ylabel('Magnitude Response')
subplot(2,1,2) %plot pole-zero diagram
zplane(b,a)
Output: BUTTERWORTH BPF USING BLT METHOD
b =
Columns 1 through 7
0.0004 0 -0.0017 0 0.0025 0 -0.0017
Columns 8 through 9
0 0.0004
a =
Columns 1 through 7
1.0000 -5.1408 13.1256 -20.9376 22.6982 -17.0342 8.6867
Columns 8 through 9
-2.7672 0.4383
z =
1
1
1
1
-1
-1
-1
-1
p =
0.5601 + 0.7475i
0.5601 - 0.7475i
0.5800 + 0.6286i
0.5800 - 0.6286i
0.6656 + 0.5628i
0.6656 - 0.5628i
0.7647 + 0.5648i
0.7647 - 0.5648i
k =
4.1660e-004
IIR HIGH PASS FILTER WITH BUTTERWORTH
CHARACTERISTICS (using Bilinear Z-Transform method)
Design an IIR high pass Butterworth filter with the following
specifications using BiLinear Z-Transform method and MATLAB.
Passband = 2-4KHz
Stopband = 0-500Hz
Passband ripple = 3dB
Stopband attenuation = 20dB
Sampling frequency = 8KHz
a) Pass and stop band edge frequencies
a) Determine, using the BiLinear Z-Transform method
and MATLAB,
i) The order,coefficients, poles and zeros of the
discrete filter. Write down its transfer function.
b) Plot the magnitude-frequency response and the pole-
zero diagram of the discrete filter.
--------------------------------------------------------------------------------
%BUTTERWORTH HPF USING BLT METHOD

Fs=8000;%Sampling frequency
Ap=3;%Attenuation in passband
As=20;%Attenuation in stopband
Wp=2000/4000
Ws=500/4000
[N,wc]=buttord(Wp,Ws,Ap,As) %Determine filter order
[zz,pz,kz]=butter(N,2000/4000,'high') %Digitize filter
[b,a]=butter(N,2000/4000,'high')
subplot(2,1,1) %plot magnitude-frequency response
[H,f]=freqz(b,a,512,Fs);
plot(f,abs(H));
xlabel('Frequency (Hz)')
ylabel('Magnitude Response')
subplot(2,1,2) %plot pole-zero diagram
zplane(b,a)
Output : BUTTERWORTH HPF USING BLT METHOD
Wp =
0.5000
Ws =
0.1250
N =
2
wc =
0.3567
zz =
1
1
pz =
0.0000 + 0.4142i
0.0000 - 0.4142i
kz =
0.2929
b =
0.2929 -0.5858 0.2929
a =
1.0000 -0.0000 0.1716
IIR BAND STOP FILTER WITH BUTTERWORTH
CHARACTERISTICS (using Bilinear Z-Transform method)
Design an IIR Band stop Butterworth filter with the following
specifications using BiLinear Z-Transform method and MATLAB.
Lower Passband = 0-50Hz
Upper Passband = 450-500Hz
Stopband = 200-300Hz
Passband ripple = 3dB
Stopband attenuation = 20dB
Sampling frequency = 1KHz
a) Pass and stop band edge frequencies
a) Determine, using the BiLinear Z-Transform method
and MATLAB,
i) The order,coefficients, poles and zeros of the
discrete filter. Write down its transfer function.
b) Plot the magnitude-frequency response and the pole-
zero diagram of the discrete filter.
--------------------------------------------------------------------------------
%BUTTERWORTH BSF USING BLT METHOD

Fs=1000;%Sampling frequency
Ap=3;%Attenuation in passband
As=20;%Attenuation in stopband
Wp=[50/500, 450/500]%Bandedge frequencies
Ws=[200/500, 300/500]
[N,wc]=buttord(Wp,Ws,Ap,As) %Determine filter order
[zz,pz,kz]=butter(N,Ws,'stop') %Digitize filter
[b,a]=butter(N,Ws,'stop')
subplot(2,1,1) %plot magnitude-frequency response
[H,f]=freqz(b,a,512,Fs);
plot(f,abs(H));
xlabel('Frequency (Hz)')
ylabel('Magnitude Response')
subplot(2,1,2) %plot pole-zero diagram
zplane(b,a)
Output : BUTTERWORTH BSF USING BLT METHOD
Wp =
0.1000 0.9000
Ws =
0.4000 0.6000
N =
2
wc =
0.2461 0.7539
zz =
0.0000 + 1.0000i
0.0000 - 1.0000i
0.0000 + 1.0000i
0.0000 - 1.0000i
pz =
-0.1884 + 0.7791i
-0.1884 - 0.7791i
0.1884 + 0.7791i
0.1884 - 0.7791i
kz =
0.6389
b =
0.6389 -0.0000 1.2779 -0.0000 0.6389
a =
1.0000 -0.0000 1.1430 -0.0000
RESULT
Thus, the Butterworth Low pass and high pass IIR filter was designed
and frequency response was plotted using MATLAB.
Exp.No.6b
Date:
DESIGN OF CHEBYSHEV TYPE I FILTER
AIM:
To design Chebyshev type I IIR filter and plot the frequency response.
SOFTWARE REQUIRED:
MATLAB version 7.2
THEORY:
These are two types of Chebyshev filters. Type I Chebyshev filters are
all pole filters that exhibit eqrilripple behavior is pass band and a monolonic
characteristics in stop band. The magnitude sequence of N
th
order is
expressed as,
( ) .. ,......... 2 , 1
1
2 2
2
·

,
_

¸
¸
Σ
·

N
rp
r
C H
r H
N
Pass band filter allows only a band of frequencies r
1
to r
2
to pass and
stops all other frequencies. Band reject filter rejects all the frequencies
between r
1
to r
2
and allows other frequencies.
ALGORITHM:
Step 1: Start the program
Step 2: Compute the stop-band and pass-band attenuation
Step 3: Computer the order
Step 4: Find the transfer function co-efficient
Step 5: Plot the magnitude response of respective filter
RESULT
Thus, the ideal characteristics of Chebyshev type I filter was designed
and frequency was plotted.
Exp.No.5 FILTER DESIGN ANALYSIS –FIR FILTER
Date:
DESIGN OF FIR FILTERS USING WINDOW TECHNIQUES
AIM:
To design FIR filter using various windows techniques and plot the
frequency response.
SOFTWARE REQUIRED:
MATLAB version 7.2
THEORY:
HAMMING WINDOW:
Hamming window is most commonly used window in speech
processing. This window also has bell like shape. Its first and last samples
are not zero. Its show that the sketch and magnitude response of this
window. It has reduced side lobes but slightly increase main lobe. The side
lobes are higher than the Blackmann window.
KAISER WINDOW:
Kaiser window allow separate control of width of the main lobe
and attenuation of side lobe. If wk(n) has two parameter the length (mt) and
shape B. It can be selected independently in Kaiser Window. Its first and
last samples are not zero. The shape of the window depends upon values of
M and β. It shows the magnitude response of Kasier Window. The shape of
main lobe and side lobe can be adjusted by selection of ‘M’ and ‘β’. It is
most commonly used in digital filter.
HANNING WINDOW:
Hanning Window has shape simplar to those of Blackmann and
Hamming. Its first and last samples are zero. It is given as,
¹
¹
¹
'
¹
− ·

,
_

¸
¸


·
otherwise
M n
M
n
n w
HN
, 0
1 ....... .......... , 1 , 0 ,
1
2
cos 1
2
1
) (
π
This window is commonly used for spectrum analysis, speech and
music processing. In the magnitude response of Hanning window, we
observe that is has narrow main, but few side lobes are significant. Then
side lobes are reduced rapidly
ALGORITHM:
Step 1: Get the pass band and stop band ripples
Step 2: Get the pass band and stop band edge frequencies
Step 3: Get the sampling frequency.
Step 4: Calculated the order of filter
Step 5: Find the Window co-efficient
Step 6: Draw the magnitude and phase response
.
FIR BAND PASS FILTER USING WINDOW
Determine the coefficients and plot the magnitude-frequency response
of a bandpass FIR filter, using an appropriate window that meets the
following specifications : passband = 150-250Hz,transition width =
50Hz,passband ripple = 0.1dB,stopband
Attenuation = 60dB,sampling frequency = 1kHz.

FS=1000; % Sampling frequency
FN=FS/2;%Nyqiust frequency
N=73;%Filter length
beta=5.65;%Kaiser window ripple parameter
fc1=125/FN;
fc2=275/FN;%Normalised cutoff frequencies
FC=[fc1 fc2];%Band edge frequency vector
k_window=kaiser(N,beta)
stem(k_window);grid on
xlabel('Sample number n')
ylabel('Amplitude (dB)')
figure,
hn=fir1(N-1,FC,kaiser(N,beta))%Obtain windowed filter coefficients
stem(hn);grid on
xlabel('Sample number n')
ylabel('Impulse response of designed filter(dB)')
figure,
[H,f]=freqz(hn,1,512,FS);%Compute frequency response
mag=20*log10(abs(H));
plot(f,mag),grid on
xlabel('Frequency (Hz)')
ylabel('Magnitude response, Gain (dB)')
FIR LOWPASS FILTER USING WINDOW
Obtain the coefficients of an FIR lowpass filter to meet the specifications
given using the window method: passband edge frequency
=1.5kHz,transition width = 0.5kHz,stopband attenuation <
50dB,sampling frequency = 5kHz.


clc;
FS=input('enter the sampling freq'); % Sampling frequency
delF=input('enter the transition width');
N=round((3.1*FS)/delF)%Filter length
FC=input('enter the cutoff freq');
FC1=FC/FS;
stem(hanning(N)),grid on
xlabel('Sample number n')
ylabel('Amplitude (dB)')
figure,
hn=fir1(N-1,FC1,hanning(N))%Obtain windowed filter coefficients
stem(hn),grid on
xlabel('Sample number n')
ylabel('Impulse response of designed filter(dB)')
figure,
[H,f]=freqz(hn,1,512,FS)%Compute frequency response
mag=20*log10(abs(H));
plot(f,mag),grid on
xlabel('Frequency (Hz)')
ylabel('Magnitude response (dB)')
Input :
enter the sampling freq5000
enter the transition width500
enter the cutoff freq1500
FIR HIGH PASS FILTER USING WINDOW
Obtain the coefficients of an FIR highpass filter to meet the
specifications given using the window method: passband edge frequency
= 1.2kHz, transition width = 0.5kHz,stopband attenuation >
50dB,sampling frequency = 8kHz.

clc;
FS=input('enter the sampling freq');% Sampling frequency
N=input('enter the length of the filter');
delF=round((3.3*FS)/N)%transition width
FC=input('enter the cutoff freq');
FC1=FC/FS;
stem(hamming(N)),grid on
xlabel('Sample number n')
ylabel('Amplitude (dB)')
figure,
hn=fir1(N-1,FC1,'high',hamming(N))%Obtain windowed filter coefficients
stem(hn),grid on
xlabel('Sample number n')
ylabel('Impulse response of designed filter(dB)')
figure,
[H,f]=freqz(hn,1,512,FS)%Compute frequency response
mag=20*log10(abs(H));
plot(f,mag),grid on
xlabel('Frequency (Hz)')
ylabel('Magnitude response (dB)')
Input :
enter the sampling freq8000
enter the transition width500
enter the cutoff freq1200
FIR BAND STOPFILTER USING WINDOW
Determine the coefficients and plot the magnitude-frequency response
of a bandstop FIR filter, using an appropriate window that meets the
following specifications : passband = 150-250Hz,filter length =
27,passband ripple = 0.1dB,stopband
Attenuation = 20dB,sampling frequency = 1kHz.

FS=1000; % Sampling frequency
FN=FS/2;%Nyqiust frequency
N=27;%Filter length
fc1=125/FN;
fc2=275/FN;%Normalised cutoff frequencies
FC=[fc1 fc2];%Band edge frequency vector
r_window=boxcar(N)
stem(r_window);grid on
xlabel('Sample number n')
ylabel('Amplitude (dB)')
figure,
hn=fir1(N-1,FC,'stop',boxcar(N))%Obtain windowed filter coefficients
stem(hn);grid on
xlabel('Sample number n')
ylabel('Impulse response of designed filter(dB)')
figure,
[H,f]=freqz(hn,1,512,FS);%Compute frequency response
mag=20*log10(abs(H));
plot(f,mag),grid on
xlabel('Frequency (Hz)')
ylabel('Magnitude response, Gain (dB)')


HAMMING WINDOW
clc:
Clear all;
Close all;
fs=input (‘stop band frequency of , fs=’) ;
fp=input (‘pass band frequency of, fp=’) ;
f=input (‘sampling frequency, f=’) ;
dels=input(‘stop band error tolerance, dels=’) ;
delp=input (‘pass band error tolerance, delp=’) ;
wp=(2*fp) /f ;
ws=(2*fs) / f ;
rp=-20*log10 (1-delp) ;
rs=-20*log10 (dels)
num=-20log(sqrt(delp*dels))-13 ;
dem=14.6*(fs-fp) / f ;
n=cell (num / dem) ;
n1=n+1 ;
if (rem (n,2)=0)
n1=n ;
n=n-1 ;
end
y=hamming (n1) ;
% low pass filter
b=firl (n, wp, y) ;
[h,w]=freqz (b,1,512) ;
m=20*log(abs (h) ;
subplot (2,2,1);
plot(1/pi,m) ;
grid ;
ylabel (‘Gain in dB’) ;
xlabel (‘Normalised Freq’) ;
title (‘Response of LPF’);
%High pass filter
b=firl (n, wp, ‘high’ , y) ;
[h, w]=freqz (b,1,512) ;
m=20*log(abs(h)) ;
subplot(2,2,2) ;
plot(w/pi,m) ;
grid ;
ylabel (‘Gain in dB’) ;
xlabel (‘Normalised Freq’) ;
title (‘Response of HPF’) ;
%Band pass filter
wn=[wp ws ] ;
b=firl (n,wn, ‘bandpass’, y) ;
[h,w]=freqz (b,1,512) ;
m=20*log(abs(h));
subplot (2,2,3)
plot(w/pi,m) ;
grid ;
ylabel (‘Gain in dB’) ;
xlabel (‘Normalised Freq’) ;
title (‘Response of HPF’) ;
%Band pass filter
wn=[wp ws ] ;
b=firl (n,wn, ‘stop’, y) ;
[h,w]=freqz (b,1,512) ;
m=20*log(abs(h));
subplot (2,2,4)
plot(w/pi,m) ;
grid ;
ylabel (‘Gain in dB’) ;
xlabel (‘Normalised Freq’) ;
title (‘Response of HPF’) ;
gtext (‘Frequency response Hamming window’) ;
KAISER WINDOW
clc:
Clear all;
Close all;
fs=input (‘stop band frequency of , fs=’) ;
fp=input (‘pass band frequency of, fp=’) ;
f=input (‘sampling frequency, f=’) ;
ds=input(‘stop band error tolerance, dels=’) ;
dp=input (‘pass band error tolerance, delp=’) ;
wp=(2*fp) /f ;
ws=(2*fs) / f ;
rp=-20*log10 (1-dp) ;
rs=-20*log10 (ds)
num=-20log(sqrt(dp*ds))-13 ;
dem=14.6*(fs-fp) / f ;
n=cell (num / dem) ;
n1=n+1 ;
if (rem (n,2)=0)
n1=n ;
n=n-1 ;
end
y=kaiser (n, 0.75) ;
% low pass filter
b=firl (n, wp, ‘low’ y) ;
[h,w]=freqz (b,1,512) ;
m=20*log(abs (h) ;
subplot (2,2,1);
plot(w/pi,m) ;
grid ;
ylabel (‘Gain in dB’) ;
xlabel (‘Normalised Freq’ (hz)*) ;
title (‘Frequency Response of LPF using Kaiser window’);
%High pass filter
b=firl (n, wp, ‘high’ , y) ;
[h, w]=freqz (b,1,512) ;
m=20*log(abs(h)) ;
subplot(2,2,2) ;
plot(w/pi,m) ;
grid ;
ylabel (‘Gain in dB’) ;
xlabel (‘Normalised Freq’) ;
title (‘Response of HPF using Kaiser window’) ;
%Band pass filter
wn=[wp ws ] ;
b=firl (n,wp, ‘bandpass’, y) ;
[h,w]=freqz (b,1,512) ;
m=20*log(abs(h));
subplot (2,2,3)
plot(w/pi,m) ;
grid ;
ylabel (‘Gain in dB’) ;
xlabel (‘Normalised Freq’) ;
title (‘Response of BPF using Kaiser window’) ;
%Band pass filter
wn=[wp ws ] ;
b=firl (n,wn, ‘stop’, y) ;
[h,w]=freqz (b,1,512) ;
m=20*log(abs(h));
subplot (2,2,4)
plot(w/pi,m) ;
grid ;
ylabel (‘Gain in dB’) ;
xlabel (‘Normalised Freq’) ;
title (‘Response of BSF using Kaiser window’) ;
gtext (‘Frequency response Hamming window’) ;
HANNING WINDOW
clc:
Clear all;
Close all;
fs=input (‘stop band frequency of , fs=’) ;
fp=input (‘pass band frequency of, fp=’) ;
f=input (‘sampling frequency, f=’) ;
ds=input(‘stop band error tolerance, dels=’) ;
dp=input (‘pass band error tolerance, delp=’) ;
wp=(2*fp) /f ;
ws=(2*fs) / f ;
rp=-20*log10 (1-delp) ;
rs=-20*log10 (dels)
num=-20log(sqrt(delp*dels))-13 ;
dem=14.6*(fs-fp) / f ;
n=cell (num / dem) ;
n1=n+1 ;
if (rem (n,2)~=0)
n1=n ;
n=n-1 ;
end
y=hanning (n1) ;
% low pass filter
b=firl (n, wp, y) ;
[h,w]=freqz (b,1,512) ;
m=20*log(abs (h) ;
subplot (2,2,1);
plot(w/pi,m) ;
grid ;
ylabel (‘Gain in dB’) ;
xlabel (‘Normalised Freq’) ;
title (‘Response of LPF using Hanning window’);
%High pass filter
b=firl (n, wp, ‘high’ , y) ;
[h, w]=freqz (b,1,512) ;
m=20*log(abs(h)) ;
subplot(2,2,2) ;
plot(w/pi,m) ;
grid ;
ylabel (‘Gain in dB’) ;
xlabel (‘Normalised Freq’) ;
title (‘Response of HPF using Hanning window’) ;
%Band pass filter
wn=[wp ws ] ;
b=firl (n,wn, ‘bandpass’, y) ;
[h,w]=freqz (b,1,512) ;
m=20*log(abs(h));
subplot (2,2,3)
plot(w/pi,m) ;
grid ;
ylabel (‘Gain in dB’) ;
xlabel (‘Normalised Freq’) ;
title (‘Response of BPF using Hanning window’) ;
%Band pass filter
wn=[wp ws ] ;
b=firl (n,wn, ‘stop’, y) ;
[h,w]=freqz (b,1,512) ;
m=20*log(abs(h));
subplot (2,2,4)
plot(w/pi,m) ;
grid ;
ylabel (‘Gain in dB’) ;
xlabel (‘Normalised Freq’) ;
title (‘Response of BSF using Hanning window’) ;
RESULT :
Thus the program for FIR filters using various window are generated
and frequency response is plotted
Exp.No. 7
Date:
MULTIRATE FILTERS
AIM:
To program upsampling and down sampling of a signal using
MATLAB program.
SOFTWARE REQUIRED:
MATLAB Version 7.2
THEORY:
ALGORITHM:
Step1: Start the process
Step2: Enter the program for upsampling and downsampling in a
sequence
Step3: Run the program
Step4: Enter the length of the sequence that is to be upsampled and
downsampled.
Step5: Enter the number of sampling factor
Step6: Plot the output
PROGRAM:
A continuous time signal is characterized by the following equation
X(t)= A cos (2πf1t) + B Cos (2πf2t)
a) Generate with aid of Matlab a discrete time equivalent of the signal.
Assume a sampling frequency of 1KHz, f1= 50Hz, f2= 100 Hz and
the ratio of the amplitudes of the frequency components,(A/B) = 1.5
b) Interpolate the discrete time signal by a factor of 4.
c) Decimate the output of the interpolater in step (b) by a factor of 4
d) Plot the original, interpolated and decimated discrete time singals
MULTIRATE FILTERS
fs=1000; % sampling frequency
A=1.5; % relative amplitudes
B=1;
f1=50; % signal frequencies
f2=100;
t=0:1/fs:1; % time vector
x=A*cos(2*pi*f1*t)+B*cos(2*pi*f2*t); %generate signal
y=interp(x,4); %interpolate signal by 4
stem(x(1:25)) % plot original signal
xlabel('Discrete time, nT')
ylabel('Input signal level')
figure,
stem(y(1:100)) % plot interpolated signal
xlabel('Discrete time, 4 x nT')
ylabel('Interpolated output signal level')
figure,
y1=decimate(y,4);
stem(y1(1:25)) % plot decimated signal
xlabel('Discrete time, nT')
ylabel('Decimated output signal level')
RESULT:
Thus, a program for Multirate filters (upsampling and
downsampling)of a discrete time signal is performed and output is obtained.
Exp.No.8
Date:
ADAPTIVE FILTERS
AIM:
To design an Adaptive filter to identify an unknown 32 order FIR
filter using 500 iterations using MATLAB program.
SOFTWARE REQUIRED:
MATLAB Version 7.2
THEORY:
PROGRAM:
ADAPTIVE FILTER
Use 500 iterations of an adapting filter system to identify an unknown
32nd-order FIR filter.
x = randn(1,500); % Input to the filter
b = fir1(31,0.5); % FIR system to be identified
n = 0.1*randn(1,500); % Observation noise signal
d = filter(b,1,x)+n; % filter the signal through the
unknown system to get the
DESIRED SIGNAL.
mu = 0.008; % LMS step size.
ha = adaptfilt.lms(32,mu);
[y,e] = filter(ha,x,d); determine the unknown system.
subplot(3,1,1);plot(1:500,d);
title('Desired signal');
xlabel('Time Index'); ylabel('Signal Value');
subplot(3,1,2);plot(1:500,y);
title('Output signal');
xlabel('Time Index'); ylabel('Signal Value');
subplot(3,1,3);plot(1:500,e);
title('Error signal');
xlabel('Time Index'); ylabel('Signal Value');
figure,
subplot(2,1,1); plot(1:500,[d;y;e]);
title('System Identification of an FIR Filter');
legend('Desired','Output','Error');
xlabel('Time Index'); ylabel('Signal Value');
subplot(2,1,2); stem([b.',ha.coefficients.']);
legend('Actual','Estimated');
xlabel('Coefficient #'); ylabel('Coefficient Value');
grid on;
Result:
Thus an unknown FIR is identified using an adaptive filter system.
Exp.No.10
Date:
ECHO CANCELLATION
AIM:
To design a filter for noise cancellation.
SOFTWARE REQUIRED:
MATLAB Version 7.2
THEORY:
Noise or Interference Cancellation :
In noise cancellation, adaptive filters let you remove noise from a signal in
real time. Here, the desired signal, the one to clean up, combines noise and
desired information. To remove the noise, feed a signal n'(k) to the adaptive
filter that represents noise that is correlated to the noise to remove from the
desired signal. Using an Adaptive Filter to Remove Noise from an Unknown
System
So long as the input noise to the filter remains correlated to the unwanted
noise accompanying the desired signal, the adaptive filter adjusts its
coefficients to reduce the value of the difference between y(k) and d(k),
removing the noise and resulting in a clean signal in e(k). Notice that in this
application, the error signal actually converges to the input data signal,
rather than converging to zero.
S(K)+n(k)
n^(k)
+
Adaptive
Filter
ADAPTIVE FILTER
Design an Adaptive RLS filter to extract useful information from a noisy
signal. The information bearing signal is a sine wave of 0.055
cycles/sample,that is corrupted by additive white gaussian noise.
Assume the use of two microphones for this adaptive noise
cancellation system with the noisy input signal as the input to a primary
microphone, while a secondary microphone receives noise that is
uncorrelated to the information bearing signal, but is correlated to the noise
picked up by the primary microphone. The noise that corrupts the sine wave
is a lowpass filtered version of (correlated to) the noise picked up by the
secondary microphone.
%The information bearing signal is a sine wave of 0.055 cycles/sample.

signal = sin(2*pi*0.055*(0:1000-1)');

subplot(3,1,1),plot(0:199,signal(1:200));
grid; axis([0 200 -2 2]);
title('The information bearing signal');

%The noise picked up by the secondary microphone is the input for the RLS
adaptive filter.The noise that corrupts the sine wave is a lowpass filtered
version of (correlated to) this noise.The sum of the filtered noise and the
information bearing signal is the desired signal for the adaptive filter.

nvar = 1.0; % Noise variance
noise = randn(1000,1)*nvar; % White noise

subplot(3,1,2),plot(0:999,noise);
title('Noise picked up by the secondary microphone');
grid; axis([0 1000 -4 4]);

%The noise corrupting the information bearing signal is a filtered version of
'noise':

nfilt = fir1(31,0.5); % 31st order Low pass FIR filter
fnoise = filter(nfilt,1,noise); % Filtering the noise

%"Desired signal" for the adaptive filter (sine wave + filtered noise):

d = signal+fnoise;
subplot(3,1,3),plot(0:199,d(1:200));
grid; axis([0 200 -4 4]);
title('Desired input to the Adaptive Filter = Signal + Filtered Noise');

%Set and initialize RLS adaptive filter parameters and values:

M = 32; % Filter order
lam = 1; % Exponential weighting factor
delta = 0.1; % Initial input covariance estimate
w0 = zeros(M,1); % Initial tap weight vector
P0 = (1/delta)*eye(M,M); % Initial setting for the P matrix
Zi = zeros(M-1,1); % FIR filter initial states

%Running the RLS adaptive filter for 1000 iterations. The plot shows the
convergence of the adaptive filter response to the response of the FIR filter.

Hadapt = adaptfilt.rls(M,lam,P0,w0,Zi);
Hadapt.PersistentMemory = true;
[y,e] = filter(Hadapt,noise,d);
H = abs(freqz(Hadapt,1,64));
H1 = abs(freqz(nfilt,1,64));

wf = linspace(0,1,64);
figure,
plot(wf,H,wf,H1);
xlabel('Normalized Frequency (\times\pi rad/sample)');
ylabel('Magnitude');
legend('Adaptive Filter Response','Required Filter Response');
grid;
axis([0 1 0 2]);

%As the adaptive filter converges, the filtered noise should be completely
subtracted from the "signal + noise" signal and the error signal 'e' should
contain only the original signal.

figure,plot(0:499,signal(1:500),0:499,e(1:500)); grid;
axis([0 500 -4 4]);
title('Original information bearing signal and the error signal');
legend('Original Signal','Error Signal');
RESULT:
Thus an Adaptive filter has been designed for noise or interference
rejection from information bearing signal.
Exp.No :9
Date:
EQUALIZATION
AIM:
To Demonstrate and perform equalization using Mat lab program
SOFTWARE REQUIRED:
MATLAB Version 7.2
THEORY:
EQUALIZATION
Time-dispersive channels can cause intersymbol interference (ISI).
For example, in a multipath scattering environment, the receiver sees
delayed versions of a symbol transmission, which can interfere with other
symbol transmissions.An equalizer attempts to mitigate ISI and thus
improve the receiver's performance.
Channel equalization is a simple way of mitigating the detrimental
effects caused by a frequency-selective and/or dispersive communication
link between sender and receiver. For this demonstration, all signals are
assumed to have a digital baseband representation.During the training phase
of channel equalization, a digital signal s[n] that is known to both the
transmitter and receiver is sent by the transmitter to the receiver. The
received signal x[n] contains two signals:the signal s[n] filtered by the
channel impulse response,and an unknown broadband noise signal v[n]. The
goal is to filter x[n] to remove the inter-symbol interference (ISI) caused by
the dispersive channel and to minimize the effect of the additive noise
v[n].Ideally, the output signal would closely follow a delayed version of the
transmitted signal s[n].
EQUALIZATION USING MATLAB :
Equalizing a signal using Communications Toolbox involves these steps:
>> Create an equalizer object that describes the equalizer class and the
adaptive algorithm that you want to use. An equalizer object is a type of
MATLAB variable that contains information about the equalizer, such
as the name of the equalizer class, the name of the adaptive algorithm, and
the values of the weights.
>> Adjust properties of the equalizer object, if necessary, to tailor it to your
needs. For example, you can change the number of weights or the values of
the weights.
>> Apply the equalizer object to the signal you want to equalize, using the
equalize method of the equalizer object.
PROGRAM:
Design an equalizer of the type shown in figure given below. H(z) represents
a communication channel with the following numerator and denominator
coefficients respectively : b = exp(j*pi/5)*[0.2 0.7 0.9];
a = [1 -0.7 0.4].Write a MATLAB script that uses
a RLS algorithm based adaptive filter to construct an equalizer of order
l=20.Use RLS forgetting factor lambda = 0.99.
The input signal takes one of sixteen different values given by all possible
combinations of {-3, -1, 1, 3} + j*{-3, -1, 1, 3}, where j = sqrt(-1).Generate
a sequence of 5000 such symbols, where each one is
equiprobable. Assume a complex Gaussian noise signal for the additive
noise. And use only the first 2000 samples for training.
%TRANSMITTED INPUT SIGNAL

%[The input signal takes one of sixteen different values given by all possible
combinations of {-3, -1, 1, 3} + j*{-3, -1, 1, 3}, where j = sqrt(-1).Let's
generate a sequence of 5000 such symbols, where each one is
equiprobable.]

ntr = 5000;
j = sqrt(-1);
s = sign(randn(1,ntr)).*(2+sign(randn(1,ntr)))+...
j*sign(randn(1,ntr)).*(2+sign(randn(1,ntr)));
plot(s,'o');
axis([-4 4 -4 4]);
axis('square');
+
Delay
Z
-m
Unknown
System , H(z)
Adaptive
Filter, W(z)
xlabel('Re\{s(n)\}');
ylabel('Im\{s(n)\}');
title('Input signal constellation');

%TRANSMISSION CHANNEL

%[The transmission channel is defined by the channel impulse response and
the noise characteristics.]

b = exp(j*pi/5)*[0.2 0.7 0.9];
a = [1 -0.7 0.4];
% Transmission channel filter
channel = dfilt.df2t(b,a);
% Impulse response
hFV = fvtool(channel,'Analysis','impulse');
legend(hFV, 'Transmission channel');
set(hFV, 'Color', [1 1 1])

%------------------------------------------------

%RECEIVED SIGNAL

%[The received signal x[n] is generated by the transmitted signal s[n]
filtered by the channel impulse response with additive noise v[n]. We shall
assume a complex Gaussian noise signal for the additive noise.]

sig = sqrt(1/16*(4*18+8*10+4*2))/sqrt(1000)*norm(impz(channel));
v = sig*(randn(1,ntr) + j*randn(1,ntr))/sqrt(2);
x = filter(channel,s) + v;
figure,plot(x,'.');
xlabel('Re\{x[n]\}');
ylabel('Im\{x[n]\}');
axis([-40 40 -40 40]);
axis('square');
title('Received signal x[n]');
set(gcf, 'Color', [1 1 1])

%-------------------------------------------------

%TRAINING SIGNAL

%[The training signal is a shifted version of the original transmitted signal
s[n].This signal would be known to both the transmitter and receiver.]

d = [zeros(1,10) s(1:ntr-10)];

%--------------------------------------------------

%TRAINED EQUALIZATION

%[To obtain the fastest convergence, we shall use the conventional version
of a recursive least-squares estimator. Only the first 2000 samples are used
for training. The output signal constellation shows clusters of values
centered on the sixteen different symbol values--an indication that
equalization has been achieved.]

P0 = 100*eye(20);
lam = 0.99;
h = adaptfilt.rls(20,lam,P0);
ntrain = 1:2000;
[y,e] = filter(h,x(ntrain),d(ntrain));
figure,plot(y(1001:2000),'.');
xlabel('Re\{y[n]\}');
ylabel('Im\{y[n]\}');
axis([-5 5 -5 5]);
axis('square');
title('Equalized signal y[n]');
set(gcf, 'Color', [1 1 1])
RESULT:
Thus Equalization is performed using Matlab and output is verified
Exp.No.10
Date:
SAMPLING AND ALIASING
AIM:
To a sample of continuous time signal and to show the effect of
aliasing.
SOFTWARE REQUIRED:
MATLAB version 7.2
THEORY:
Sampling is the acquisition of Act waves as a discrete time interval
and is a fundamental concept in a real time signal processing.
SAMPLING THEOREM:
If the highest frequency component in a signal in junction is unknown,
then the signal should be sampled at a rate of atleast 2 tmax for the samples
to describe the signal completely. When F is sampling freqeuncy or rate.
Thus if f max is an analog signal 4KHZ, then the signal should be sampled
at 8 KHZ or more.
The superimposition of high frequency component on low frequency
is called aliasing.
ALGORITHM:
Step 1: Start the process
Step 2: Calculate sampling frequency and sampling time
for fs > 2wm,
f = 2wm, f < 2wm
Step 3: Plot the graph
Step 4: Observe aliasing effect
Step 5: Store the result
Step 6: Stop the process
PROGRAM:
SAMPLING AND ALIASING
clc;
clear all ;
close all ;
fm=input (‘enter the signal frequency, fm = ‘) ;
t=[0 : . 02 : 1 ] ;
x=cos 92*pi*fm*t) ;
ts1 = .2 ;
ts2 = .1 ;
ts3 = .04 ;
t1 = [0 : ts1 : 1] :
x1 = cos (2*pi*t1*fm) ;
subplot (2,2,1) ;
plot (t,x)
xlabel (‘time’) ;
ylabel (‘x(t)’) ;
title (‘inputsignal’) ;
subplot (2,2,2) ;
stem (t1, x1) ;
xlabel (‘n’) ;
ylabel (‘x(n)’) ;
title (‘sampled sequence with fs<2fm’) ;
hold on ;
plot (t,x,’r : ‘ ) ;
hold off ;
t2=[0 : ts2 : 1] ;
x2=cos(2*pi*t2*fm) ;
subplot(2,2,3)
stem(t2,x2) ;
xlabel (‘n’) ;
ylabel (‘x(n)’) ;
title (‘sampled sequence with fs=2fm’) ;
hold on ;
plot (t,x,’r’:’) ;
hold off
t3=[0 :ts3 : 1] ;
x3=cos(2*pi*t3*fm) ;
subplot (2,2,4)
stem(t3,x3) ;
xlabel (‘n’) ;
ylabel (‘x(n)’) ;
hold on ;
plot (t,x,’r’:’) ;
hold off
title (‘sampling sequence with fs=2fm’) ;
gtext(‘sampling &* effect of aliasing’ ) ;
RESULT
Thus a simple program for sampling and aliasing was written and
output has been obtained successfully.

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