Practical No.

Aim :- To study basic concepts of MATLAB.

Theory:-

1. Starting a MATLAB Session

 On Microsoft Windows platforms, start the MATLAB program by doubleclicking
the MATLAB shortcut on your Windows desktop.
 On Apple Macintosh platforms, start MATLAB by double-clicking the MATLAB
icon in the Applications folder.
 On UNIX platforms, start MATLAB by typing matlab at the operating system
prompt.
When you start MATLAB, by default, MATLAB automatically loads all the program
files provided by The MathWorks for MATLAB and other MathWorks products. You do
not have to start each product you want to use.
There are alternative ways to start MATLAB, and you can customize MATLAB startup.
For example, you can change the folder in which MATLAB starts or automatically
execute MATLAB statements upon startup.

2.The Desktop
When you start MATLAB, the desktop appears, containing tools (graphical user
interfaces) for managing files, variables, and applications associated with MATLAB. The
following illustration shows the default desktop. You can customize the arrangement of
tools and documents to suit your needs. For more information about the desktop tools, see
Desktop Tools and Development Environment.
3.Quitting the MATLAB Program

To end your MATLAB session, select File > Exit MATLAB in the desktop, or
type quit in the Command Window. You can run a script file named finish.m
each time MATLAB quits that, for example, executes functions to save the
workspace.
Confirm Quitting
MATLAB can display a confirmation dialog box before quitting. To set this
option, select File > Preferences > General > Confirmation Dialogs, and
select the check box for Confirm before exiting MATLAB.

Result :- Thus we have studied the basic concepts of the MATLAB

 Practical No.2

Aim:Operations on sinusoidal sequence. Experimental ,ramp,unit.

Theory: A continuous-time signal x(t) is said to be periodic with period T if it satisfy the
condition
x (t+T) = x(t) for all t -∞<t<+∞ ………….eq1
A signal is periodic if the above condition is not satisfied for at least one value of T.
The smallest value of T that satisfies the above condition is known as fundamental period.
Complex exponential and sinusoidal signals are examples for continuous-time periodic signals.
Consider a sinusoidal signal
x(t)=Asin(wt+θ) ……………………eq2
Where A is tha amplitude, w is tha frequency in radians per second (rad/sec), and θ the phase in
radians. The frequency f0 in hertz is given by f0 = w /2π or periodic signal we have
X(t+T)=x(t) ………………eq3
For x(t)=Asin(wt+θ) ………………eq4
X (t+T) = Asin [w(t+T)+ θ)]
Equation 5 and 2 are equal if
wT = 2π , T =2π/ w
Thu the sinusoidal is periodic with period 2π/ w The sum of two periodic signals x1(t) and x2(t)
with periods T1 and T2 respectively may or may not be periodic, then the ratio T1/T2 can be
written as the ratio a/b of two integers a and b. If T1/T2 = a/b, then bT1 = aT2, and since a and b
are integers, x1(t) and x2(t) are periodic with period bT1. If a and b are co prime (i. e. a and b
have no common integer factors other than 1) then T = bT1 is the fundamental period of the sum
of two signals.

Program:
Unit:
n=input ('Enter the length of the step sequence N='); % Get the length of the require
sequence from the user
t=0:n-1; % defines the time axis
y=ones(1,n); % defines an 1 x n matrix which is filled with ones
stem(t,y); %displays the data as lines
ylabel ('Amplitude'); % name the Y axis
xlabel ('Time Index'); %Name the x axis
TITLE ('Unit Step Signal'); % Giving the title for the plot.
Output:

Result:Thus implementation of sinusoidal sequences is implemented successfully.

 Practical No.3

Aim : Write a program for linear convolution and deconvolution

Theory:
Convolution is a formal mathematical operation, just as multiplication, addition, and integration.
Addition takes two numbers and produces a third number, while convolution takes two signals
and produces a third signal. Convolution is used in the mathematics of many fields, such as
probability and statistics. In linear systems, convolution is used to describe the relationship
between three signals of interest: the input signal, the impulse response, and the output signal.
In this equation, x1(k), x2(n-k) and y(n) represent the input to and output from the system at time
n. Here we could see that one of the input is shifted in time by a value every time it is multiplied
with the other input signal. Linear Convolution is quite often used as a method of implementing
filters of various types.

Program:

%linear convolution program

clc;
clear all;
close all;
disp('linear convolution program');

x=input('enter i/p x(n):');

m=length(x);
h=input('enter i/p h(n):');
n=length(h);

x=[x,zeros(1,n)];
subplot(2,2,1), stem(x);
title('i/p sequencce x(n)is:');
xlabel('---->n');
ylabel('---->x(n)');grid;

h=[h,zeros(1,m)];
subplot(2,2,2), stem(h);
title('i/p sequencce h(n)is:');
xlabel('---->n');
ylabel('---->h(n)');grid;
disp('convolution of x(n) & h(n) is y(n):');
y=zeros(1,m+n-1);
for i=1:m+n-1
y(i)=0;
for j=1:m+n-1
if(j<i+1)
y(i)=y(i)+x(j)*h(i-j+1);
end
end
end y
subplot(2,2,[3,4]),stem(y);
title('convolution of x(n) & h(n) is :');
xlabel('---->n');
ylabel('---->y(n)');grid;

Output:

Result:Thus we have implemented the linear convolution.

 Practical no: 4

Aim: To verify Circular Convolution .

Equipment required:

Software: MATLAB 7.5

Theory: Circular convolution is another way of finding the convolution sum of two input
signals. It resembles the linear convolution, except that the sample values of one of the input
signals is folded and right shifted before the convolution sum is found. Also note that circular
convolution could also be found by taking the DFT of the two input signals and finding the
product of the two frequency domain signals. The Inverse DFT of the product would give the
output of the signal in the time domain which is the circular convolution output. The two input
signals could have been of varying sample lengths. But we take the DFT of higher point, which
ever signals levels to. For example, if one of the signal is of length 256 and the other spans 51
samples, then we could only take 256-point DFT. So the output of IDFT would be containing
256 samples instead of 306 samples, which follows N1+N2 – 1 where N1 & N2 are the lengths
256 and 51 respectively of the two inputs. Thus the output which should have been 306 samples
long is fitted into 256 samples. The256 points end up being a distorted version of the correct
signal. This process is called circular convolution.

Program:

clc;

clear all;

close all;

disp('circular convolution program');

x=input('enter i/p x(n):');

m=length(x);

h=input('enter i/p sequence h(n)');

n=length(h);

subplot(2,2,1), stem(x);

title('i/p sequence x(n)is:');

xlabel('---->n');
ylabel('---->x(n)');

grid;

subplot(2,2,2), stem(h);

title('i/p sequence h(n)is:');

xlabel('---->n');

ylabel('---->h(n)');

grid;

disp('circular convolution of x(n) & h(n) is y(n):');

if(m-n~=0)

if(m>n)

h=[h,zeros(1,m-n)];

n=m;

end

x=[x,zeros(1,n-m)];

m=n;

end

y=zeros(1,n);

y(1)=0;

a(1)=h(1);

for j=2:n

a(j)=h(n-j+2);

end

%circular convolution
for i=1:n

y(1)=y(1)+x(i)*a(i);

end

for k=2:n

y(k)=0;

% circular shift

for j=2:n

x2(j)=a(j-1);

end

x2(1)=a(n);

for i=1:n

if(i<n+1)

a(i)=x2(i);

y(k)=y(k)+x(i)*a(i);

end

end

end

y subplot(2,2,[3,4]),stem(y);

title('convolution of x(n) & h(n) is:');

xlabel('---->n');

ylabel('---->y(n)'); grid;
OUTPUT:

Result: Hence Circular Convolution is verified successfully using MATLAB.

 Practical no :5

Aim: To perform DFT computation

Theory:
In mathematics, the discrete Fourier transform (DFT) converts a finite sequence of equally-
spaced samples of a function into a same-length sequence of equally-spaced samples of the
discrete-time Fourier transform (DTFT), which is a complex-valued function of frequency. The
interval at which the DTFT is sampled is the reciprocal of the duration of the input sequence. An
inverse DFT is a Fourier series, using the DTFT samples as coefficients of complex sinusoids at
the corresponding DTFT frequencies. It has the same sample-values as the original input
sequence. The DFT is therefore said to be a frequency domain representation of the original
input sequence. If the original sequence spans all the non-zero values of a function, its DTFT is
continuous (and periodic), and the DFT provides discrete samples of one cycle. If the original
sequence is one cycle of a periodic function, the DFT provides all the non-zero values of one
DTFT cycle.
The DFT is the most important discrete transform, used to perform Fourier analysis in many
practical applications.[1] In digital signal processing, the function is any quantity or signal that
varies over time, such as the pressure of a sound wave, a radio signal, or daily temperature
readings, sampled over a finite time interval (often defined by a window function). In image
processing, the samples can be the values of pixels along a row or column of a raster image. The
DFT is also used to efficiently solve partial differential equations, and to perform other
operations such as convolutions or multiplying large integers.
Since it deals with a finite amount of data, it can be implemented in computersby numerical
algorithms or even dedicated hardware. These implementations usually employ efficient fast
Fourier transform (FFT) algorithms; so much so that the terms "FFT" and "DFT" are often used
interchangeably. Prior to its current usage, the "FFT" initialism may have also been used for the
ambiguous term "finite Fourier transform".

Program:

Matlab Code for DFT

x = [2 3 -1 4];
N = length(x);
X = zeros(4,1)
for k = 0:N-1
for n = 0:N-1
X(k+1) = X(k+1) + x(n+1)*exp(-j*pi/2*n*k)
end
end
t = 0:N-1
subplot(311)
stem(t,x);
xlabel('Time (s)');
ylabel('Amplitude');
title('Time domain - Input sequence')
subplot(312)
stem(t,X)
xlabel('Frequency');
ylabel('|X(k)|');
title('Frequency domain - Magnitude response')
subplot(313)
stem(t,angle(X))
xlabel('Frequency');
ylabel('Phase');
title('Frequency domain - Phase response')
X % to check |X(k)|
angle(X) % to check phase

Output:

X=
8.0000 + 0.0000i
3.0000 + 1.0000i
-6.0000 - 0.0000i3.0000 - 1.0000i
ans =
0
0.3218
-3.1416
-0.3218

The response $X[k]$ is same above calculated. The plots are:

Result: Thus we performed DFT computation.
 Practical No:6

Aim: Cross Correlation and Auto Correlation

Theory:

N=1024; % Number of samples

f1=1; % Frequency of the sinewave
FS=200; % Sampling Frequency
n=0:N-1; % Sample index numbers
x=sin(2*pi*f1*n/FS); % Generate the signal, x(n)
t=[1:N]*(1/FS); % Prepare a time axis
subplot(2,1,1); % Prepare the figure
plot(t,x); % Plot x(n)
title('Sinwave of frequency 1000Hz [FS=8000Hz]');
xlabel('Time, [s]');
ylabel('Amplitude');
grid;
Rxx=xcorr(x); % Estimate its autocorrelation
subplot(2,1,2); % Prepare the figure
plot(Rxx); % Plot the autocorrelation
grid;
title('Autocorrelation function of the sinewave');
xlable('lags');
ylabel('Autocorrelation');

function [Rxx]=autom(x)
% [Rxx]=autom(x)
% This function Estimates the autocorrelation of the sequence
of
% random variables given in x as: Rxx(1), Rxx(2),…,Rxx(N),
where N is
% Number of samples in x.
N=length(x);
Rxx=zeros(1,N);
for m=1: N+1
for n=1: N-m+1
Rxx(m)=Rxx(m)+x(n)*x(n+m-1);
end;
end;
To use the above function, you need to save this under autom.m and then it may be
used as follows:
N=1024;
f1=1;
FS=200;
n=0:N-1;
x=sin(2*pi*f1*n/FS);
t=[1:N]*(1/FS);
subplot(2,1,1);
plot(t,x);
title('Sinwave of frequency 1000Hz [FS=8000Hz]');
xlabel('Time, [s]');
ylabel('Amplitude');
grid;
Rxx=autom(x);
subplot(2,1,2);
plot(Rxx);
grid;
title('Autocorrelation function of the sinewave');
xlabel('lags');
ylabel('Autocorrelation');
Conclusion: Thus we performed practical successfully.
 Practical No: 7

Aim:To study and design FIR filter.

Theory:
In signal processing, a finite impulse response (FIR) filter is a filter whose impulse response (or
response to any finite length input) is of finite duration, because it settles to zero in finite time.
This is in contrast to infinite impulse response (IIR) filters, which may have internal feedback
and may continue to respond indefinitely (usually decaying).
The impulse response (that is, the output in response to a Kronecker delta input) of an Nth-order
discrete-time FIR filter lasts exactly N + 1 samples (from first nonzero element through last
nonzero element) before it then settles to zero.

FIR filters can be discrete-time or continuous-time, and digital or analog.

An FIR filter has a number of useful properties which sometimes make it preferable to an infinite
impulse response (IIR) filter. FIR filters:
 Require no feedback. This means that any rounding errors are not compounded by summed
iterations. The same relative error occurs in each calculation. This also makes
implementation simpler.
 Are inherently stable, since the output is a sum of a finite number of finite multiples of the

input values, so can be no greater than times the largest value appearing in the input.
 Can easily be designed to be linear phase by making the coefficient sequence symmetric.
This property is sometimes desired for phase-sensitive applications, for example data
communications, seismology, crossover filters, and mastering.
The main disadvantage of FIR filters is that considerably more computation power in a general
purpose processor is required compared to an IIR filter with similar sharpness or selectivity,
especially when low frequency (relative to the sample rate) cutoffs are needed. However many
digital signal processors provide specialized hardware features to make FIR filters approximately
as efficient as IIR for many applications.
An FIR filter is designed by finding the coefficients and filter order that meet certain
specifications, which can be in the time-domain (e.g. a matched filter) and/or the frequency
domain (most common). Matched filters perform a cross-correlation between the input signal
and a known pulse-shape. The FIR convolution is a cross-correlation between the input signal
and a time-reversed copy of the impulse-response. Therefore, the matched-filter's impulse
response is "designed" by sampling the known pulse-shape and using those samples in reverse
order as the coefficients of the filter.
When a particular frequency response is desired, several different design methods are common:
1. Window design method
2. Frequency Sampling method
3. Weighted least squares design
4. Parks-McClellan method
Program:

clc;
clear all;
close all;
rp=input('Enter the passband ripple(rp):');
rs=input('Enter the stopband ripple(rs):');
fp=input('Enter the passband frequency(fp):');
fs=input('Enter the stopband
frequency(fs):');
f=input('Enter the sampling frequency(f):');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(rp*rs))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n1=n+1;
if(rem(n,2)~=0)
n1=n;
n=n-1;
end
y=boxcar(n1);
%Low pass filter
b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);
plot(m);
ylabel('Gain(db)->');
xlabel('(a)Normalised frequency->');
%High pass filter
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);
plot(m);
ylabel('Gain(db)->');
xlabel('(b)Normalised frequency->');
%Band pass filter
wn=[wp*ws];
b=fir1(n,wn,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,3);
plot(m);
ylabel('Gain(db)->');
xlabel('(c)Normalised frequency->');
%Band stop filter
wn=[wp*ws];
b=fir1(n,wn,'stop',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,4);
plot(m);
ylabel('Gain(db)->');
xlabel('(d)Normalised frequency->');

Output:

Result: Thus we have studied and designed FIR filter.

 Practical No. 8

Aim : To design analog band pass filter

Theory: A band-pass filter (also bandpass filter, BPF) is a device that passes frequencies within
a certain range and rejects (attenuates) frequencies outside that range. An example of
an analogue electronic band-pass filter is an RLC circuit (a resistor–inductor–capacitor circuit).
These filters can also be created by combining a low-pass filter with a high-pass filter.
Bandpass is an adjective that describes a type of filter or filtering process; it is to be
distinguished from passband, which refers to the actual portion of affected spectrum. Hence, one
might say "A dual bandpass filter has two passbands." A bandpasssignalis a signal containing a
band of frequencies not adjacent to zero frequency, such as a signal that comes out of a bandpass
filter. An ideal bandpass filter would have a completely flat passband (e.g. with no
gain/attenuation throughout) and would completely attenuate all frequencies outside the
passband. Additionally, the transition out of the passband would have brickwallcharacteristics.
The bandwidth of the filter is simply the difference between the upper and lower cutoff
frequencies. The shape factor is the ratio of bandwidths measured using two different attenuation
values to determine the cutoff frequencyOptical band-pass filters are common in photography
and theatre lighting work. These filters take the form of a transparent coloured film or sheet.
Q factor: A band-pass filter can be characterized by its Q factor. The Q-factor is the reciprocal of
the fractional bandwidth. A high-Q filter will have a narrow passband and a low-Q filter will
have a wide passband. These are respectively referred to as narrow-band and wide-band filters.

Program:

clear all;
clc;
rp = input(‘pass ripple freq’);
rs = input(‘stop ripple freq’);
fp = input(‘pass band freq’);
fs = input(‘stop band freq’);
f = input(‘sample freq’);
w1 = 2*fp/f;
w2= 2*fs/f;
[nJ= buttord(w1 ,w2,rp,rs);
wn= [w1,w2];
[b,a]= butter(n,wn,’bandpass’);
w=0:0.1:pi;
[h,p]= freqz(b,a,w);
g= 20*log10(abs(h));
A=angle(h); subplot (2,2,1);
plot(p/pi,g); ylabel(‘amp’);
xlabel(‘ferq’);
title(‘amp,freq’);
subplot (2,2,2);
plot(p/pi,A);
xlabel(‘normal. freq’);
ylabel(‘phase’);
title(‘normal. freq,phas&);

Command Window of Analog Band pass:

rp=0.5000
stop ripple freq4o
rs= 40
pass band freq2000
fp= 2000
stop band freq3 000
fs= 3000
sample freql0000
w1 =0.4000
w2= 0.6000
n= 9
wn = 0.4000 0.6000
b= 1.0e-003 *
Columns 1 through 4
0.0064 0 -0.0572 0
Columns 5 through 8
0.2287 0 -0.5336 0