Signal Processing Laboratory Manual 21ECL58A BMS Institute of Technology and Management

BMS INSTITUTE OF TECHNOLOGY AND MANAGEMENT

AVALAHALLI, DODDABALLAPUR MAIN ROAD, BANGALORE-64

DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

SIGNAL PROCESSING LABORATORY: 21ECL58A

DEPARTMENT VISION

Be a pioneer in Providing Quality Education in Electronics, Communication and

Allied Engineering fields to serve as Valuable Resource for Industry and Society.

DEPARTMENT MISSION

1. Impart Sound Theoretical Concepts and Practical Skills through innovative

Pedagogy.
2. Promote Inter-disciplinary Research.
3. Inculcate Professional Ethics.

Course co-ordinators (2022-23)

1. Dr. Anna Merine George, Assistant Professor, ECE

2. Dr. Vijayalakshmi G V, Associate Professor, ECE
3. Dr. Deepa N Reddy, Associate Professor, ECE

Laboratory Instructor

1. Mr. Byrareddy A B

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Signal Processing Laboratory Manual 21ECL58A BMS Institute of Technology and Management

B.E ELECTRONICS & COMMUNICATION ENGINEERING

Choice Based Credit System (CBCS)
SEMESTER -V
Signal Processing Laboratory (0:0:2) 1
Common to ECE and ETE
(Effective from the academic year 2021-2021)

Course Code 21ECL58A CIE Marks 50

Teaching Hours/Week (L: T: P) 0:0:2 SEE Marks 50

Total Number of Contact Hours 26 Exam Hours 03

Course Objectives:
1. Simulate discrete time signals and verify of sampling theorem.
2. Compute convolution, correlation and verify its properties.
3. Find solutions to difference equations and determine the response to impulse,
step, and sinusoidal inputs.
4. Compute DFT using inbuilt functions and analyse the properties.
5. Compute and display the filtering operations and compare with the theoretical
values.
6. Familiarity with DSP kits and implement basic operations of signals & systems.

PART-A: Simulation Experiments

Following Experiments to be done using MATLAB:
1. Verification of sampling theorem in time domain and frequency domain.
2. Linear and circular convolution of two given sequences, Commutative,
distributive,

3. and associative property of convolution.

4. Solving a given difference equation.
5. Auto and cross correlation of two sequences and verification of their
properties.
6. Computation of N point DFT of a given sequence and to plot the
magnitude and phase spectrum (using DFT equation and verify it
by built-in routine).
7. (i) Verification of DFT properties (like Linearity and Parseval’s theorem,
etc.)
(ii) DFT computation of square pulse and Sinc function etc.
8. Design and implementation of Low pass and High pass FIR filter
to meet the desired specifications (using different window
techniques).

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9. Design and implementation of a digital IIR filter (Low pass and High pass)
to meet given specifications.

PART-B: Experiments on DSK

1. Obtain the Linear convolution of two sequences.
2. Compute Circular convolution of two sequences
3. Compute the N-point DFT of a given sequence.
4. Determine the Impulse response of first order/second order system.

Course Outcomes: Students will be able to

CO1: Write a code to carryout various basic operations on discreate signals and
verify them using MATLAB/OCTAVE software.
CO2: Simulate the programs and execute them on the DSP Starter Kit using Code
Composer Studio Software tool.
CO3: Write the report for the conducted experiment.
CO4: Write open ended experiment to analyse 1-D /2-D signals.

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SIGNAL PROCESSING LABORATORY

List of Experiments
PART –A
1. Verification of sampling theorem in time domain and frequency domain.
2. Linear and circular convolution of two given sequences, Commutative,
distributive, and associative property of convolution.
3. Solving a given difference equation.
4. Auto and cross correlation of two sequences and verification of their
properties
5. Computation of N point DFT of a given sequence and to plot magnitude and
phase spectrum (using DFT equation and verify it by built-in routine).
6. (i) Verification of DFT properties (like Linearity and Parseval’s theorem,
etc.)
(ii) DFT computation of square pulse and Sinc function etc.
7. Design and implementation of Low pass and High pass FIR filter to meet
the desired specifications (using different window techniques)
8. Design and implementation of a digital IIR filter (Low pass and High pass)
to meet given specifications.

PART-B
1. Obtain the Linear convolution of two sequences.
2. Compute Circular convolution of two sequences.
3. Compute the N-point DFT of a given sequence.
4. Determine the Impulse response of first order /second order system.

PART-C
Open end experiment: Creation of GUI based signal generator with time
domain and frequency domain plots.

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MATLAB

MATLAB (matrix laboratory) is a programming language developed by

MathWorks. It started out as a matrix programming language where linear algebra
programming was simple. It can be run both under interactive sessions and as a
batch job. MATLAB is a paradigm numerical environment and proprietary
programming language developed by Math Works.
MATLAB allows matrix manipulations, plotting of functions and data,
implementation of algorithms, creation of user interfaces, and interfacing with
programs written in other languages, including C, C++, C#, Java, Fortran and Python.

Although MATLAB is intended primarily for numerical computing, an optional

toolbox uses the MuPAD symbolic engine, allowing access to symbolic
computing abilities. An additional package, Simulink, adds graphical multi-domain
simulation and model-based design for dynamic and embedded systems.

As of 2017, MATLAB has roughly one million users across industry and
academia. MATLAB users come from various backgrounds of engineering, science,
and economics. There are various toolboxes in MATLAB like signal processing
toolbox, image processing toolbox, neural network toolbox, statistics toolbox,
financial toolbox, Bioinformatics toolboxes.

The Matlab screen has 4 important windows along with Editor (where programs
can be written and stored). These are Command window where we type
commands and outputs/error messages are seen, Command history window
where past commands are remembered(we can re-run any command by clicking
on the commands),Workspace window which stores all the variables used and
Current Directory window shows the files in current directory.

In this semester students are required to register in Math works website using
their official email ID and download the MATLAB software from the following link.

https://www.mathworks.com/academia/tah-portal/bms-institute-of-technology-
31463638.html

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The installation might take minimum of 15-20 GB of your internet data as

well as same amount of system memory.

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FIG 1 MATLAB WINDOWS

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FIG 2 SCREEN COMPONENTS

PART A

EXPERIMENTS USING MATLAB/OCTAVE

EXPERIMENT NO 1
Verification of Sampling Theorem

Aim: To verify Sampling theorem for a signal of given frequency.

Theory:
● Sampling is a process of converting a continuous time signal (analog
signal) x(t) into a discrete time signal x[n], which is represented as a
sequence of numbers. (A/D converter)
● Converting back x[n] into analog (resulting in is the process of
reconstruction. (D/A converter)
● Techniques for reconstruction-(i) ZOH (zero order hold)
interpolation results in a staircase waveform, is implemented by
MATLAB plotting function stairs (n, x), (ii) FOH (first order hold)
where the adjacent samples are joined by straight lines is
implemented by MATLAB plotting function plot (n, x),(iii) spline
interpolation, etc.
● For to be exactly the same as x(t), sampling theorem in the
generation of x(n) from x(t) is used. The sampling frequency fs
determines the spacing between samples.

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● Aliasing-A high frequency signal is converted to a lower frequency,

results due to under sampling. Though it is undesirable in ADCs, it
finds practical applications in stroboscope and sampling
oscilloscopes.

Algorithm:
1. Input the desired frequency fmax (for which sampling theorem is to
be verified).
2. Generate an analog signal xt of frequency fmax for comparison.
3. Generate over sampled, Nyquist & Under sampled discrete time
signals.
4. Plot the waveforms and hence prove sampling theorem.

MATLAB Implementation:

5. Step 1: MATLAB can generate only discrete time signals. For an

approximate analog signal xt, choose the spacing between the
samples to be very small (≈0), say 50µs = 0.00005. Next choose the
time duration, say xt exists for 0.05seconds.(final in program) (for
low frequency say <1000 Hz choose 0.05 sec, for higher choose
0.01 secs or lesser as appropriate).
Now begin with the vector that represents the time base
t = 0:0.00005:0.05;
The colon (:) operator in MATLAB creates a vector, in the above case a
time vector running from 0 to 0.05 in steps of 0.00005. The semicolon (;)
tells MATLAB not to display the result.

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Given t, the analog signal xt of frequency fd is generated as

(sin(ωt)=sin(2πft))-
xt=sin(2*pi*fmax*t);
pi is recognized as 3.14 by MATLAB.

Step 2: To illustrate oversampling condition, choose sampling frequency

fs0=2.2*fd. For this sampling rate T0=1/fs0, generate the time vector as n1
= 0:T0:0.05; & over sampled discrete time signal x1=sin(2*pi*fmax*n1);
[Alternately let n1 be a fixed number of samples, say n=0:10; &
x1=sin(2*pi*n*fd/fs0);]
The discrete signal is converted back to analog signal (reconstructed) using
the MATLAB plot function (FOH). In one plot both the analog signal
(approximate one in blue color) and the reconstructed signal (in red) are
plotted for comparison

Step 3: Repeat step 2 for different sampling frequencies, i.e.,

fs=1.3*fd&fs=2*fd for under sampling and Nyquist sampling conditions.

MATLAB Program:

tfinal=0.05;
t=0:0.00005: tfinal;
fmax=input('Enter analog frequency');

%Define analog signal for comparison

xt=sin(2*pi*fmax*t);

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%Simulate condition for under sampling i.e., fs1<2*fmax

fs1=1.3*fmax;
%define the time vector
n1=0:1/fs1: tfinal;
%Generate the under sampled signal
xn1=sin(2*pi*n1*fmax);
%plot the analog & sampled signals
subplot(3,1,1);
plot(t,xt,'b',n1,xn1,'r*-');
title('Under sampling plot');

%Condition forNyquistplot
fs2=2*fmax;
n2=0:1/fs2:tfinal;
xn2=sin(2*pi*fmax*n2);
subplot(3,1,2);
plot(t,xt,'b',n2,xn2,'r*-');
title('Nyquist plot');
%Condition for Oversampling
fs3=10*fmax;
n3=0:1/fs3:tfinal;
xn3=sin(2*pi*fmax*n3);
subplot(3,1,3);
plot(t,xt,'b',n3,xn3,'r*-');
title('Oversampling plot');
xlabel('time');
ylabel('amplitude');

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legend('analog','discrete')

Result:
Enter analog frequency 200

Output Waveforms
Plots of a sampled sine wave of 200Hz

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Students are expected to use interp1q () function to connect the samples in

the sampled signal.

Plots of a sampled cosine wave of 200Hz

Inference:
1. From the under-sampling plot observe the aliasing effect. The
analog signal is of 200Hz (T=0.005s). The reconstructed (from
under sampled plot) is of a lower frequency. The alias frequency is
computed as fmax-fs1 = 200-1.3*200 = 200-260= -60Hz

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This is verified from the plot. The minus sign results in a 180˚ phase
shift.
(For example: 3kHz& 6kHz sampled at 5kHz result in aliases of
-2kHz (3k-5k) & 1kHz (6k-5k) respectively)
2. Sampling at the Nyquist rate results in samples sin(πn) which are
identically zero, i.e., we are sampling at the zero crossing points and
hence the signal component is completely missed. This can be
avoided by adding a small phase shift to the sinusoid. The above
problem is not seen in cosine waveforms (except cos(90n)). A
simple remedy is to sample the analog signal at a rate higher than
the Nyquist rate. The Fig1.2 shows the result due to a cosine signal
(x1=cos(2*pi*fmax*n1);
3. The over sampled plot shows a reconstructed signal almost similar
to that of the analog signal. Using low pass filtering the wave form
can be further smoothened.

Open ended Experiments:

Generate sine/cosine waveforms of multiple frequencies, say

x1=sin(2*pi*fd1*n1)+ sin(2*pi*fd2*n2); where fd1, fd2 are 2 different
frequencies

VIVA Questions
1. Define plot, stem, and subplot in MATLAB.
2. List all the functions used in the above program.
3. Briefly explain sampling theorem.
4. What is aliasing?

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EXPERIMENT NO 2
A. Linear Convolution of Two Given Sequences

Aim: To obtain convolution of two finite duration sequences.

Theory:
● The output y[n] of a LTI (linear time invariant) system can be
obtained by convolving the input x[n] with the system’s impulse
response h[n].
● The convolution sum is

● x[n] and h[n] can be both finite or infinite duration sequences.

● Even if one (or both) of the sequences is infinite (say,

), we can analytically evaluate the convolution

formula to get a functional form (closed form solution) of y[n].
● If both the sequences are of finite duration, then we can use the
MATLAB function ‘conv’ to evaluate the convolution sum to obtain
the output y[n]. Convolution is implemented as polynomial
multiplication (refer MATLAB help).

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● The length of y[n] = xlength + hlength -1.

● The conv function assumes that the two sequences begin at n=0 and
is invoked by y=conv(x,h).
● Even if one of the sequences begin at other values of n, say n=-3,or
n=2; then we need to provide a beginning and end point to y[n]
which are ybegin=xbegin+hbegin and yend=xend+hend
respectively.

Algorithm:
1. Input the two sequences as x1, x2
2. Convolve both to get output y.
3. Plot the sequences.
MATLAB Implementation:
MATLAB recognizes index 1 to positive maximum. Index 0 is also
not recognized. The timing information for a sequence is provided by
another vector, say n=-3:5; creates a vector with values from -3 to 5 with
an increment of 1.
During plotting, the time vector size and the sequence size should be
the same, i.e., the number of elements in the sequence x1 and in its time
vector n1 should be the same. Similarly for x2 and y.

MATLAB Program:
Level 1 Program (both sequences are right sided)
%Mainpart of computation
x1=input('Enter first sequence')
x2=input('Enter second sequence')
y=conv(x1, x2);

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n1=0:length(x1)-1;
n2=0:length(x2)-1;
n=0:length(y)-1;
disp('linear con of x1 & x2 is y=');
disp(y);
%Graphical display part
subplot (2,1,1);
stem(n,y);
xlabel('time index n');
ylabel('amplitude ');
title ('convolution output');
subplot(2,2,3);
stem(n1,x1);
xlabel('time index n');
ylabel('amplitude ');
title('plot of x1');
subplot(2,2,4);
stem(n2,x2);
xlabel('time index n');
ylabel('amplitude ');
title('plot of x2');

Result
Enter first sequence [1 2 3 4]
x1 = 1 2 3 4
Enter second sequence [2 3 2 4]
x2 = 2 3 2 4

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Linear con of x1 & x2 is y= 2 7 14 25 26 20 16

Output Waveforms

Level 2 Program (Two sided sequences)

%Main part of computation
x1=[1 2 3 2 1 3 4] %first sequence

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n1=-3:3 %time vector of first seq

x2=[ 2 -3 4 -1 0 1] %second seq
n2=-1:4 %time vector of second seq
%Add first elements of time vector
ybegin=n1(1)+n2(1);
%Add last elements of time vector
yend=n1(length(x1))+n2(length(x2));
ny=[ybegin:yend];
y=conv(x1, x2) ;
disp('linear con of x1 & x2 is y=');
disp(y);

%Graphical display with time info

subplot (2,2,1);
stem(ny,y);
xlabel('time index n');
ylabel('amplitude ');
title('convolution output');
subplot(2,2,3);
stem(n1,x1);
xlabel('time index n');
ylabel('amplitude ');
title('plot of x1');
subplot(2,2,4);
stem(n2,x2);
xlabel('time index n');
ylabel('amplitude ');

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title('plot of x2');
Result
x1 =1 2 3 2 1 3 4
n1 =-3 -2 -1 0 1 2 3
x2 =2 -3 4 -1 0 1
n2 =-1 0 1 2 3 4
linear con of x1 & x2 is y= 2 1 4 2 6 9 3 2 15 -3 3 4
Output waveforms

Properties of Convolution:
a). Commutative property
clear all;
close all;

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x1=[1 2 4 4];
x2=[1 2 7 2];
y1=conv(x1,x2);
y2=conv(x2,x1);
if y1==y2
disp('Commutative property is satisfied');
else
disp('Commutative property is not satisfied');
end
b). Associative property
clear all;
close all;
x1=[1 2 3 4];
x2=[1 2 1 2];
x3=[3 4 2 1];
y1=conv(x1,x2);
y2=conv(y1,x3);
y3=conv(x2,x3);
y4=conv(x1,y3);
if y2==y4
disp('Associative property is satisfied');
else
disp('Associative property is not satisfied');
end
c). Distributive property
clear all;
close all;
x1=[1 2 3 4 ];
x2=[1 2 1 2];
x3=[3 4 2 1];
y1=x2+x3;
y2=conv(x1,y1);

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y3=conv(x1,x2)+conv(x1,x3);
if y2==y3
disp('distributive property is satisfied');
else
disp('distributive property is not satisfied');
end

Open ended Experiments:

Find the convolution of
a. x[n]=[1 2 3] and y[n]=[1 2 3 4]

VIVA Questions :
1. Explain how linear convolution can be obtained.
2. Explain different properties of convolution
3. Give the different methods for performing convolution

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B. Circular Convolution of Two Given Sequences

Aim: To obtain circular convolution of two finite duration sequences.
Theory:
● As seen in the last experiment, the output y[n] of a LTI (linear time
invariant) system can be obtained by convolving the input x[n] with
the system’s impulse response h[n].
● The above linear convolution is generally applied to periodic
sequences. Whereas the Circular Convolution is used to study the
interaction of two signals that are periodic.
● The linear convolution sum is

.
● To compute this equation, the linear convolution involves the
following operations:

o Folding- fold h[k] to get h[-k] ( for y[0])

o Multiplication – vk[n] = x[k] × h[n-k] (both sequences are
multiplied sample by sample)
o Addition- Sum all the samples of the sequence vk[n] to obtain
y[n]
o Shifting – the sequence h[-k] to get h[n-k] for the next n.

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● The circular convolution sum is

where the index implies circular shifting operation and

implies folding the sequence circularly.

● Steps for circular convolution are the same as the usual convolution,
except all index calculations are done "mod N" = "on the wheel".
o Plot f [m] and h [−m] as shown in Fig. 4.1.
(use f(m) instead of x(k))
o Multiply the two sequences
o Add to get y[m]
o "Spin" h[−m] n times Anti Clockwise
(counter-clockwise) to get h[n-m].
x[n] and h[n] can be both finite or infinite duration sequences. If infinite
sequences, they should be periodic, and the N is chosen to be at least equal
to the period. If they are finite sequences N is chosen as >= to
max(xlength, hlength). Whereas in linear convolution N>=
xlength+hlength-1.
● Say x[n]={-2,3,1,1} and N = 5, then the x[-1] and x[-2] samples
are plotted at x[N-1] = x[-4] and x[N-2] =x[-3] places. Now x[n] is
entered as x[n] = {1, 1, 0,-2,3}.

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Fig. 4.1 Plotting of f(m) and h(-m) for circular convolution

Algorithm:
1. Input the two sequences as x and h.
2. Circularly convolve both to get output y.
3. Plot the sequences.

MATLAB Implementation:
MATLAB recognizes index 1 to be positive maximum. Index 0 is
not recognized. Hence in the below program wherever y, x and h
sequences are accessed, the index is added with +1. The modulo index
calculation for circular convolution is carried out using the function -
MOD Modulus (signed remainder after division). MOD( x, y) is x -
y.*floor(x./y) if y ~= 0. By convention, MOD(x,0) is x. The input x and y
must be real arrays of the same size, or real scalars. MOD( x, y) has the
same sign as y while REM( x, y) has the same sign as x. MOD( x, y) and
REM( x, y) are equal if x and y have the same sign, but differ by y if x and
y have different signs.

MATLAB Program:
Program to find circular convolution of two sequences by
padding zeroes

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clc;
clear all;
close all;
g=input('Enter the first sequence');
h=input('Enter the second sequence');
N1=length(g);
N2=length(h);
N=max(N1,N2);
N3=N1-N2;
if(N3>=0)
h=[h,zeros(1,N3)];
else
g=[g,zeros(1,-N3)];
end
for n=0:N-1
y(n+1)=0;

for k=0:N-1
i=mod((n-k),N); %calculation of x index
if i<0
i=i+N;
end %end of ‘if’
y(n+1)=y(n+1)+h(k+1)*g(i+1);
end %end of inner ‘for loop’
end %end of inner ‘for loop’
disp('The resultant signal is');y
figure(1);

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subplot(3,1,1);
stem(g);
title('The first sequence is');
grid on;
subplot(3,1,2);
stem(h);
title('The second sequence is');
grid on;
subplot(3,1,3);
stem(y);
title('The circularly convolved sequence is');
grid on;

Result:
Enter the first sequence [1 1 1 1]
Enter the second sequence [1 2 3 4]
The resultant signal is
y =10 10 10 10

Output Waveform

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2 Circular convolution using Matrix method

clc;
clear all;
close all;
X=input('ENTER THE FIRST SEQUENCE');
H=input('ENTER THE SECOND SEQUENCE IN MATRIX FORMAT');
Y=X*H;
disp(‘circular convolution’);
disp(Y);
stem(Y);
xlabel(‘n');
ylabel('Y(n)');
grid on;
Result:

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Enter the first sequence [1 1 1 1]

Enter the second sequence in matrix format[1 4 3 2;2 1 4 3;3 2 1 4;4 3 2 1]
Circular convolution is: 10 10 10 10

Open ended Experiments:

Find the circular convolution of x[n] = [1 3 5 6] and y[n] = [1 2 ]

VIVA Questions :
1. Write A Program to find the circular convolution of two sequences
using Matrix Method
2. Explain how circular convolution can be obtained.

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3. Give the different methods for performing circular convolution.

EXPERIMENT NO 3
Solving a Given Difference Equation

Aim: To obtain the impulse response/step response of a system

described by the given difference equation

Theory:

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● A difference equation with constant coefficients describes a LTI

system. For example the difference equation y[n] + 0.8y[n-2] +
0.6y[n-3] = x[n] + 0.7x[n-1] + 0.5x[n-2] describes a LTI system of
order 3. The coefficients 0.8, 0.7, etc. are all constant i.e., they are
not functions of time (n). The difference equation
y[n]+0.3ny[n-1]=x[n] describes a time varying system as the
coefficient 0.3n is not constant.
● The difference equation can be solved to obtain y[n], the output for
a given input x[n] by rearranging as y[n] = x[n] + 0.7x
[n-1]+0.5x[n-2]- 0.8y[n-2]- 0.6y[n-3] and solving.
● The output depends on the input x[n]
o With x[n]= δ[n], an impulse, the computed output y[n] is the
impulse response.
o If x[n]=u[n], a step response is obtained.
o If x[n] = cos(wn) is a sinusoidal sequence, a steady state
response is obtained (wherein y[n] is of the same frequency
as x[n], with only an amplitude gain and phase shift-refer
Fig.7.3).
o Similarly for any arbitrary sequence of x[n], the
corresponding output response y[n] is computed.
● The difference equation containing past samples of output, i.e.,
y[n-1], y[n-2], etc. leads to a recursive system, whose impulse
response is of infinite duration (IIR). For such systems the impulse
response is computed for a large value of n, say n=100 (to
approximate n=∞). The MATLAB function filter is used to compute
the impulse response/ step response/ response to any given x[n].

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Note: The filter function evaluates the convolution of an infinite

sequence (IIR) and x[n], which is not possible with conv function
(remember conv requires both the sequences to be finite).
● The difference equation having only y[n] and present and past
samples of input (x[n], x[n-k]), represents a system whose impulse
response is of finite duration (FIR). The response of FIR systems
can be obtained by both the ‘conv’ and ‘filter’ functions. The filter
function results in a response whose length is equal to that of the
input x[n], whereas the output sequence from conv function is of a
longer length (xlength + hlength-1).
Algorithm:
1. Input the two sequences as a and b representing the coefficients of y
and x.
2. If IIR response, then input the length of the response required (say
100, which can be made constant).
3. Compute the output response using the ‘filter’ command.
4. Plot the input sequence & impulse response (and also step response,
etc if required).

MATLAB Implementation:
MATLAB has an inbuilt function ‘filter’ to solve difference equations
numerically, given the input and difference equation coefficients (b, a).
y=filter (b,a,x)

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where x is the input sequence, y is the output sequence which is of same

length as x.
Given a difference equation a0y[n]+a1y[n-1]+a2y[n-2]=b0x[n]+b2x[n-2], the
coefficients are written in a vector as b=[b0 0 b2] and a=[a0 a1 a2]. Note the
zero in b (x[n-1] term is missing).
Also remember a0, the coefficient of y[n] should always be 1.
For impulse response x[n] = {1, 0, 0, 0, ----} the number of zeros = the
length of the IIR response required (say 100 implemented by function
zeros(1,99)).
For step response x[n]={1,1,1,1,----} the number of one’s = the length of
the IIR response required-1 (say 100 implemented by function
ones(1,100).
Similarly for any given x[n] sequence, the response y[n] can be calculated.

Given Problem
1) Difference equation y(n) - y(n-1) + 0.9y(n-2) = x(n);
Calculate impulse response h(n) and also step response at n=0,…..,100
2) Plot the steady state response y[n] to x[n]=cos(0.05πn)u(n), given
y[n]-0.8y[n-1]=x[n]
MATLAB Program:
%To find Impulse Response
N=input ('Length of response required=');
b=[1]; %x[n] coefficient
a=[1,-1,0.9]; %y coefficients
%impulse input
h=[1,zeros(1,N-1)];
%time vector for plotting

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n=0:1:N-1;
%impulse response
y=filter (b,a,h);

%plot the waveforms

subplot (2,1,1);
stem(n,h);
title('impulse input');
xlabel('n');
ylabel('impulse)');
subplot(2,1,2);
stem(n,y);
title('impulse response');
xlabel('n');
ylabel('y(n)

Result:
Length=100
Output Waveform

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%To find step response–

N=input ('Length of response required=');
b=[1]; %x[n] coefficient
a=[1,-1,0.9]; %y coefficients
h=[ones(1,N)]; %step input
n=0:1:N-1; %time vector for plotting
y=filter(b,a,h); %step response

%plot the waveforms

subplot(2,1,1);

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stem(n,h);
title('step input');
xlabel('n');
ylabel('u(n)');
subplot(2,1,2);
stem(n,y);
title('step response');
xlabel('n');
ylabel('y(n)');
Result:
Length=100
Output Waveform

%To find steady state response

%y[n]-0.8y[n-1]=x[n]

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N=input('Length of response required=');

b=[1]; %x[n] coefficient
a=[1,-0.8]; %y coefficients
n=0:1:N-1; %time vector
h=cos(0.05*pi*n); % sinusoidal input
y=filter(b,a,h); %steady state response
subplot(2,1,1);
stem(n,h);
title('steady state input');
xlabel('n');
ylabel('u(n)');
subplot(2,1,2);
stem(n,y);
title('steady state response');
xlabel('n');
ylabel('y(n)');
Result:
Length=100

Output Waveform

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Open ended Experiments :

Given y[n] = y[n-1]+y[n-2]+x[n-1] find
a. Impulse response
b. Step response
c. Sinusoidal response

VIVA Questions
1. Explain the Matlab function filter
2. How to find various responses using filter function

EXPERIMENT NO 4

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A. Auto correlation of a Given Sequence and Verification of its

Properties.

Aim: To obtain autocorrelation of the given sequence and verify its

properties.

Theory:
● Correlation is mathematical technique which indicates whether 2
signals are related and in a precise quantitative way how much they are
related. A measure of similarity between a pair of energy signals x[n]
and y[n] is given by the cross correlation sequence rxy[l] defined by

● The parameter ‘l’ called ‘lag’ indicates the time shift between the pair.
● Autocorrelation sequence of x[n] is given by

● Some of the properties of autocorrelation are enumerated below

o The autocorrelation sequence is an even function i.e.,
o At zero lag, i.e., at l=0, the sample value of the autocorrelation
sequence has its maximum value (equal to the total energy of the signal

εx) i.e., .

This is verified in Fig. 5.1, where the autocorrelation of the rectangular

pulse (square) has a maximum value at l=0. All other samples are of

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lower value. Also the maximum value = 11 = energy of the pulse

[12+12+12..].
o A time shift of a signal does not change its autocorrelation sequence.
For example, let y[n]=x[n-k]; then ryy[l] = rxx[l] i.e., the autocorrelation
of x[n] and y[n] are the same regardless of the value of the time shift k.
This can be verified with a sine and cosine sequences of same
amplitude and frequency will have identical autocorrelation functions.
For power signals the autocorrelation sequence is given by

and for periodic signals with period N

it is and this rxx[l] is also

o Periodic with N. This is verified in Fig. 5.3 where we use the

periodicity property of the autocorrelation sequence to determine the
period of the periodic signal y[n] which is x[n] (=cos(0.25*pi*n))
corrupted by an additive uniformly distributed random noise of
amplitude in the range [-0.5 0.5]

Algorithm:
1. Input the sequence as x.
2. Use the ‘xcorr’ function to get auto correlated output r.
3. Plot the sequences.

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MATLAB Implementation:
MATLAB has the inbuilt function XCORR(A), when A is a vector,
is the auto-correlation sequence. If A is of length M vector (M>1), then the
xcorr function returns the length 2*M-1 auto-correlation sequence. The
zeroth lag of the output correlation is in the middle of the sequence at
element M.
XCORR(...,MAXLAG) computes the (auto/cross) correlation over
the range of lags: -MAXLAG to MAXLAG, i.e., 2*MAXLAG+1 lags. If
missing, default is MAXLAG = M-1.
[C,LAGS] = XCORR(...) returns a vector of lag indices (LAGS).

MATLAB Programs
%Computation of Autocorrelation of a rectangular Sequence
x = input ('Type in the reference sequence = ');
% Compute the correlation sequence
n = length(x)-1;
r = conv(x, fliplr(x));
disp('autocorrelation sequence r=');
disp(r);
%plot the sequences
subplot (2,1,1)
stem (x);
title ('square sequence');
subplot
(2,1,2)
k = -n: n;
stem (k, r);

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title('autocorrelation output');
xlabel('Lag index');
ylabel('Amplitude');

Result:
Autocorrelation sequence r = 1.0000 2.0000 3.0000 4.0000 5.0000
6.0000 7.0000 8.0000 9.0000 10.0000 11.0000 10.0000 9.0000
8.0000 7.0000 6.0000 5.0000 4.0000 3.0000 2.0000 1.0000

% Computation of Auto -correlation sequence using xcorr function

x = input('Type in the reference sequence = ');

% Compute the correlation sequence
[r,lag]=xcorr(x,x);
disp('auto correlation output is =');
disp(r);
stem(lag,r);
xlabel('Lag index');
ylabel('Amplitude');

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Output Waveform

Inference: Following Autocorr properties are verified

1) Max peak at zero lag (l=0)
2) The autocorrseq is an even function

Exercise: Find the autocorrelation of x[n] = cos (nPi/2) for n=0, 1, 2, 3

VIVA Questions
1. Explain the differences between convolution and correlation
2. List the methods for performing correlation
3. Which are the Mat lab functions used to perform correlation
4. Explain the following functions
a. rand(1,N)-0.5
b. fliplr(Y)
c. xcorr(Y)

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B. Cross Correlation of a Given Sequence and Verification of its

Properties.
Aim: To obtain cross correlation of the given sequence and verify its
properties.
Theory:
Cross Correlation has been introduced in the last experiment. Comparing
the equations for the linear convolution and cross correlation we find that

. i.e., convolving the

reference signal with a folded version of sequence to be shifted (y[n])
results in cross correlation output. (Use ‘fliplr’ function for folding the
sequence for correlation).
● The properties of cross correlation are 1) the cross correlation
sequence sample values are upper bounded by the inequality

2) The cross correlation of two sequences x[n] and y[n]=x[n-k]

shows a peak at the value of k. Hence cross correlation is employed
to compute the exact value of the delay k between the 2 signals.
Used in radar and sonar applications, where the received signal
reflected from the target is the delayed version of the transmitted
signal (measure delay to determine the distance of the target).
3) The ordering of the subscripts xy specifies that x[n] is the
reference sequence that remains fixed in time, whereas the sequence
y[n] is shifted w.r.t x[n]. If y[n] is the reference sequence then
. Hence ryx[l] is obtained by time reversing the
sequence rxy[l].

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Algorithm:
1. Input the sequence as x and y.
2. Use the ‘xcorr’ function to get cross correlated output r.
3. Plot the sequences.

MATLAB Implementation:
MATLAB has the inbuilt function XCORR: Say C = XCORR(A,B), where
A and B are length M vectors (M>1), returns the length 2*M-1
cross-correlation sequence C. If A and B are of different length, the
shortest one is zero-padded. Using convolution to implement correlation,
the instruction is FLIPLR Flip matrix in left/right direction. FLIPLR(X)
returns X with row preserved and columns flipped in the left/right
direction. X = 1 2 3 becomes 3 2 1.

MATLAB Programs
% Computation of Cross-correlation Sequence using folded sequence
and convolution
x = input('Type in the reference sequence = ');
y = input('Type in the second sequence = ');
% Compute the correlation sequence
n1 = length(x)-1;
n2 = length(y)-1;
r = conv(x,fliplr(y));
disp(‘Cross correlation output is =’);
disp(r);
k = (-n1):n2;%time vector for plotting
stem(k,r);

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xlabel('Lag index');
ylabel('Amplitude');

Result:
Type in the reference sequence = [1 3 -2 1 2 -1 4 4 2]
Type in the second sequence = [2 -1 4 1 -2 3]
Cross correlation output is =
3 7 -11 14 13 -15 28 6 -2 21 12 12 6 4
Output Waveforms

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% Computation of Cross-correlation sequence using xcorr function

x = input('Type in the reference sequence = ');

y = input('Type in the second sequence = ');
% Compute the correlation sequence
[r,lag]=xcorr(x,y);
disp('Cross correlation output is =');
disp(r);
stem(lag,r);
xlabel('Lag index');
ylabel('Amplitude');
Result:

Type in the reference sequence = [1 3 -2 1 2 -1 4 4 2]

Type in the second sequence = [2 -1 4 1 -2 3]
Verification: with xcorr(x,y), the output is
{-0.0000 , -0.0000 , 0.0000, 3.0000, 7.0000, -11.0000, 14.0000,
13.0000, -15.0000, 28.0000, 6.0000, -2.0000, 21.0000, 12.0000,
12.0000, 6.0000 , 4.0000}

%Verification of the properties

x1=[1,2,3,-1,0,0];
x2=[0,0,1,2,3,-1]; %=xr[n-2],i.e., k=2
>>xcorr(x1,x2)
ans = {-1,1,5,15,5,1-1,0,0}

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Inference:
Strong peak of 15 at lag = -2 implies the delay between xi and x2 is 2.
Also peak =15=energy of xi (1+22+32+ (-1)2) implies that both xi and x2
are strongly correlated.
%consider the below sequences
xr=[1,2,3,-1];
x2=[3,-1,1,2];
xcorr(xr,x2) ans ={2,5,7,2,2,10,-3}
xcorr(x2,xr) ans ={-3,10,2,2,7,5,2}

Inference:
Strong peak of 10 at lag = 2 implies the delay between xr and x2 is 2, but
since 10<15, it implies that xr and x2 are uncorrelated slightly (may be due
to noise, etc). is verified.

Open ended Experiments: Find the cross correlation of x[n]=[1 2 0

1] and y[n]=[2 2 1 1]

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EXPERIMENT NO 5
Computation of N point DFT of a Given Sequence and to Plot
Magnitude and Phase Spectrum.

Aim: To compute DFT of the given sequence.

Theory:
● Discrete Fourier Transform (DFT) is used for performing frequency
analysis of discrete time signals. DFT gives a discrete frequency
domain representation whereas the other transforms are continuous
in frequency domain.
● The N point DFT of discrete time signal x[n] is given by the
equation

Where N is chosen such that , where L=length of x[n].

● The inverse DFT allows us to recover the sequence x[n] from the
frequency samples.

● X(k) is a complex number (remember ejw=cosw + jsinw). It has both

magnitude and phase which are plotted versus k. These plots are
magnitude and phase spectrum of x[n]. The ‘k’ gives us the
frequency information.
● Here k=N in the frequency domain corresponds to sampling
frequency (fs). Increasing N, increases the frequency resolution, i.e.,
it improves the spectral characteristics of the sequence. For example

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if fs=8 kHz and N=8 point DFT, then in the resulting spectrum, k=1
corresponds to 1 kHz frequency. For the same fs and x[n], if N=80
point DFT is computed, then in the resulting spectrum, k=1
corresponds to 100Hz frequency. Hence, the resolution in frequency
is increased.
● Since , increasing N to 8 from 80 for the same x[n] implies
x[n] is still the same sequence (<8), the rest of x[n] is padded with
zeros. This implies that there is no further information in time
domain, but the resulting spectrum has higher frequency resolution.
This spectrum is known as ‘high density spectrum’ (resulting from
zero padding x[n]). Instead of zero padding, for higher N, if more
number of points of x[n] are taken (more data in time domain), then
the resulting spectrum is called a ‘high resolution spectrum’.
Algorithm:
1. Input the sequence for which DFT is to be computed.
2. Input the length of the DFT required (say 4, 8, >length of the
sequence).
3. Compute the DFT using the ‘fft’ command.
4. Plot the magnitude & phase spectra.

MATLAB Implementation:
MATLAB has an inbuilt function ‘FFT’ which computes the Discrete
Fourier transform.
FFT(X) is the discrete Fourier transform (DFT) of vector X. For length
N input vector x, the DFT is a length N vector X, with elements N.

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FFT(X,N) is the N-point FFT, padded with zeros if X has less than N
points and truncated if it has more.
The magnitude spectrum is computed using the function ABS Absolute
value.
ABS(X) is the absolute value of the elements of X. When X is complex,
ABS(X) is the complex modulus (magnitude) of the elements of X.
The phase spectrum is computed using the function ANGLE Phase angle.
ANGLE (H) returns the phase angles, in radians, of a matrix with complex
elements.

Matlab Program:
N-point FFT and IFFT of a given sequence

clc;
clear all;
close all;
x=input ('Enter the sequence x (n)');
N=length(x);
y=fft(x,N);
disp(y);
mag=abs(y);
phase=angle(y);
subplot(2,2,1);
stem(x);
grid on;
title('Input Data Sequence x(n)');
subplot(2,2,3);

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stem(mag);
grid on;
title('Magnitude Plot');
subplot(2,2,2);
stem(phase);
grid on;
title('Phase plot');
z=ifft(y,N);
mag1=real(z);
subplot(2,2,4);
stem(mag1);
grid on;
title('Signal Sequence constituted from Spectrum');

Result:
Enter the sequence x(n) [0.5 0.5 0.5 0.5 0 0.5 0.5 0.5]
3.5000 0.5000 -0.5000 0.5000 -0.5000 0.5000 -0.5000
0.5000
Output waveform

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DFT of a given sequence

To find the N-point DFT of a sequence using a function file dft
clc:
xn=input(‘enter the sequence x(n)’);
N=input(‘enter the number of points of computation’);
y=dft(xn,N);
disp(y);
amp=abs(y);
ph=angle(y);
subplot(2,1,1)
stem(amp);
xlabel (‘MAGNITUDE’);
ylabel (‘K’);
title (‘magnitude plot’);
stem (ph*180/pi);
xlabel(‘phase’);
ylabel(‘K’);

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title(‘phase plot’);

%function to find dft

function[xk] =dft(xn,N)
n=[0:1:N-1];
k=[0:1:N-1];
WN=exp(-j*2*pi/N);
nk=n`*k;
WNnk=WN.^nk;
xk=xn*WNnk;
Result:
Enter the sequence x(n)=[1 1 1 1 0 0 0 0]
Xk=4.000 1.0000-2.4142i -0.0000-0.0000i 0.0000-0.0000i
1.0000+0.4142i 0.0000-0.0000i

IDFT of a given sequence

To find the N-point IDFT of a sequence using a function file idft

clc:
xk=input(‘enter the sequence x(k)’);
N=input(‘enter the number of points of computation’);
y=idft(xk,N);
disp(y);
amp=abs(y);
ph=angle(y);
subplot(2,1,1)

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stem(amp);
xlabel(‘MAGNITUDE’);
ylabel(‘K’);
title(‘magnitude plot’);
stem(ph*180/pi);
xlabel(‘phase’);
ylabel(‘K’);
title(‘phase plot’);

%function to find idft

function[xn] =idft(xk,N)
n=[0:1:N-1];
k=[0:1:N-1];
WN=exp(-j*2*pi/N);
nk=n`*k;
WNnk=WN.^(-nk);
xn=(xk*WNnk)/N;

Result:
Enter the sequence Xk=[4, 1-2.4142i ,0,1-0.4142i,0, 1.0000+0.4142i ,
0,1+2.4142i]
x(n)=[1 1 1 1 0 0 0 0]

Inputs -
Run1: Give 4 point sequence and find 4 point DFT
Run2: Give 8 point sequence and find 8 point DFT

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Run 3: Give a Sinusoidal Signal with a frequency 250 Hz with fs = 8

KHz and sample it for 64 points and perform 64 point DFT on it.
Run 4: Give a Sinusoidal Signal with a frequency 250 Hz and 500 Hz
with fs = 8 KHz and sample it for 64 point and perform 64 point DFT on
it.

Open ended Experiments :

1. Find the 4 point DFT of x[n]=cosnπ/4

2. Find the 3 point IDFT of X[K]=[1-j2,-1,1+j2]

VIVA Questions:
1. Explain DFT
2. What are the methods of finding DFT
3. Explain differences between DFT and FFT
4. Explain how functions can be declared in MATLB.
5. Explain MATLAB function fft(x)
6. How can user create a MATLAB function

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EXPERIMENT NO 6
A. Verification of DFT properties (Linearity and Parseval’s
Theorem)
Matlab code for Linearity properly &Parseval's Theorem
i) Verificaton of Parseval's Theorem

Program

x1=[1 2 3 4]
X1=fft(x1,4);
en1=sum(x1.^2);
en2=(1/4)*(sum(abs(X1.^2)))
if en1==en2
disp('ParsevalTheoreem Proved')
disp('Energy =')
disp(en2)
end

Result

x1 =1 2 3 4

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en2 = 30
Parseval Theorem Proved
Energy =30

ii) Verification of Linearity Property

Program
x1=[1 2 3 4]
x2=[1 1 1 1]
a1=a2=1;
term1=a1.*x1+a2.*x2;
LHS=fft(term1,4);
term2=fft(a1.*x1);
term3=fft(a2.*x2);
RHS=term2+term3;
if(LHS==RHS)
disp('Lineraity Property Proved')
else
disp('Non Linear')
end

Result
x1=1 2 3 4

x2 =1 1 1 1

Linearity Property Proved

B. DFT computation of Square Pulse

Program

x1=[zeros(1,8),ones(1,16),zeros(1,8)]
xk=fft(x1)
xk2=fftshift(xk);
subplot(2 ,1, 1)
stem(x1)
subplot(2 ,1, 2)

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plot(abs(xk2));

Result

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C. DFT computation of Sinc Pulse

x=-pi:0.05:pi;
x1=sinc(x)
xk=fft(x1)
xk2=fftshift(xk);
subplot(2 ,1, 1)
stem(x1)
subplot(2 ,1, 2)
plot(abs(xk2));

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Result

EXPERIMENT NO 7

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Design and Implementation of FIR Filter to Meet Given Specifications

Aim: To design and implement a FIR filter for given specifications.

Theory:
There are two types of systems – Digital filters (perform signal filtering in
time domain) and spectrum analyzers (provide signal representation in the
frequency domain). The design of a digital filter is carried out in 3 steps-
specifications, approximations and implementation.

DESIGNING AN FIR FILTER (using window method):

Method 1: Given the order N, cutoff frequency fc, sampling frequency fs

and the window.
● Step 1: Compute the digital cut-off frequency Wc (in the range -π
<Wc< π, with π corresponding to fs/2) for fc and fs in Hz. For
example let fc=400Hz, fs=8000Hz
Wc = 2*π* fc / fs = 2* π * 400/8000 = 0.1* π radians
For MATLAB the Normalized cut-off frequency is in the range 0
and 1, where 1 corresponds to fs/2 (i.e.,fmax)). Hence to use the
MATLAB commands
wc = fc / (fs/2) = 400/(8000/2) = 0.1
Note: if the cut off frequency is in radians then the normalized
frequency is computed as wc = Wc / π
● Step 2: Compute the Impulse Response h(n) of the required FIR
filter using the given Window type and the response type (low pass,
band pass, etc). For example given a rectangular window, order

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N=20, and a high pass response, the coefficients (i.e., h[n] samples)
of the filter are computed using the MATLAB inbuilt command
‘fir1’ as
h =fir1(N, wc , 'high', boxcar(N+1));
Note: In theory we would have calculated h[n]=hd[n]×w[n], where
hd[n] is the desired impulse response (low pass/ high pass, etc given
by the sinc function) and w[n] is the window coefficients. We can
also plot the window shape as stem(boxcar(N)).
Plot the frequency response of the designed filter h(n) using the
freqz function and observe the type of response (lowpass / highpass
/bandpass).

Method 2:
Given the pass band (wp in radians) and Stop band edge (ws in
radians) frequencies, Pass band ripple Rp and stopband attenuation
As.
● Step 1: Select the window depending on the stopband attenuation
required. Generally if As>40 dB, choose Hamming window. (Refer
table )
● Step 2: Compute the order N based on the edge frequencies as
Transition bandwidth = tb=ws-wp;
N=ceil (6.6*pi/tb);
● Step 3: Compute the digital cut-off frequency Wc as
Wc=(wp+ws)/2
Now compute the normalized frequency in the range 0 to 1 for
MATLAB as
wc=Wc/pi;

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Note: In step 2 if frequencies are in Hz, then obtain radian

frequencies (for computation of tb and N) aswp=2*pi*fp/fs,
ws=2*pi*fstop/fs, where fp, fstop and fs are the pass band, stop
band and sampling frequencies in Hz
● Step 4: Compute the Impulse Response h(n) of the required FIR
filter using N, selected window, type of response(low/high,etc)
using ‘fir1’ as in step 2 of method 1.

IMPLEMENTATION OF THE FIR FILTER

1. Once the coefficients of the FIR filter h[n] are obtained, the next
step is to simulate an input sequence x[n], say input of 100, 200 &
400 Hz (with sampling frequency of fs), each of 20/30 points.
Choose the frequencies such that they are >, < and = to fc.
2. Convolve input sequence x[n] with Impulse Response, i.e., x (n)*h
(n) to obtain the output of the filter y[n]. We can also use the ‘filter’
command.
3. Infer the working of the filter (low pass/ high pass, etc).

MATLAB IMPLEMENTATION
FIR1 Function
B = FIR1(N,Wn) designs an N'th order low pass FIR digital filter and
returns the filter coefficients in length N+1 vector B. The cut-off frequency
Wn must be between 0 <Wn< 1.0, with 1.0 corresponding to half the
sample rate. The filter B is real and has linear phase, i.e., even symmetric
coefficients obeying B(k) = B(N+2-k), k = 1,2,...,N+1.
If Wn is a two-element vector, Wn = [W1 W2], FIR1 returns an order N
band pass filter with pass band W1< W < W2. B = FIR1(N,Wn,'high')

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designs a high pass filter. B = FIR1(N,Wn,'stop') is a band stop filter if Wn

= [W1 W2]. If Wn is a multi-element vector, Wn = [W1 W2 W3 W4
W5 ... WN], FIR1 returns an order N multiband filter with bands
0 < W < W1, W1 < W < W2, ..., WN < W < 1.
FREQZ Digital filter frequency response. [H,W] = FREQZ(B,A,N) returns
the N-point complex frequency response vector H and the N-point
frequency vector W in radians/sample of the filter whose numerator and
denominator coefficients are in vectors B and A. The frequency response
is evaluated at N points equally spaced around the upper half of the unit
circle. If N isn't specified, it defaults to 512.
For FIR filter enter A=1 and B = h[n] coefficients. Appropriately choose N
as 128, 256, etc

Matlab Program:
1 RECTANGULAR WINDOW
clc;
clear all;
close all;
lp=input('Enter the pass band ripple:');
ls=input('Enter the stop band ripple:');
fp=input('Enter the pass band frequency:');
fs=input('Enter the stop band frequency:');
f=input('Enter the sampling frequency:');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(lp*ls))-13;
dem=14.6*(fs-fp)/f;

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n=ceil(num/dem);
n1=n+1;
if (rem(n,2)~=0)
n1=n;
n=n-1;
end
y=boxcar(n1);% or y=rect(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(o/pi,m);
ylabel('Gain in db-->');
xlabel('(a)Normalised Frequency-->');
grid on;
% 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(o/pi,m);
ylabel('Gain in db-->');
xlabel('(b)Normalised Frequency-->');
grid on;

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Result

Enter the pass band ripple: 0.05

Enter the stop band ripple: 0.04
Enter the pass band frequency: 1500
Enter the stop band frequency: 2000
Enter the sampling frequency: 9000

Output waveform

2 HAMMING WINDOW
clc;
clear all;
close all;
lp=input('Enter the passband ripple:');
ls=input('Enter the stopband ripple:');
fp=input('Enter the passband frequency:');
fs=input('Enter the stopband frequency:');
f=input('Enter the sampling frequency:');

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wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(lp*ls))-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=hamming(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(o/pi,m);
ylabel('Gain in db-->');
xlabel('(a)Normalised Frequency-->');
grid on;

% High Pass Filter

b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);

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plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(b)Normalised Frequency-->');
grid on;

Result:
Enter the passband ripple: 0.05
Enter the stopband ripple: 0.001
Enter the passband frequency: 1200
Enter the stopband frequency: 1700
Enter the sampling frequency: 9000
Output waveform

3 BARTLET
clc;
clear all;
close all;
lp=input('Enter the passband ripple:');
ls=input('Enter the stopband ripple:');
fp=input('Enter the passband frequency:');
fs=input('Enter the stopband frequency:');

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f=input('Enter the sampling frequency:');

wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(lp*ls))-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=bartlett(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(o/pi,m);
ylabel('Gain in db-->');
xlabel('(a)Normalised Frequency-->');
grid on;

% High Pass Filter

b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));

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subplot(2,2,2);
plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(b)Normalised Frequency-->');
grid on;

Result:
Enter the passband ripple: 0.05
Enter the stopband ripple: 0.04
Enter the passband frequency: 1500
Enter the stopband frequency: 2000
Enter the sampling frequency: 9000
Output Waveform

4 HANNING
clc;
clear all;
close all;
lp=input('Enter the passband ripple:');
ls=input('Enter the stopband ripple:');

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fp=input('Enter the passband frequency:');

fs=input('Enter the stopband frequency:');
f=input('Enter the sampling frequency:');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(lp*ls))-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=hanning(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(o/pi,m);
ylabel('Gain in db-->');
xlabel('(a)Normalised Frequency-->');
grid on;

% High Pass Filter

b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);

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m=20*log10(abs(h));
subplot(2,2,2);
plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(b)Normalised Frequency-->');
grid on;

Result
Enter the passband ripple: 0.03
Enter the stopband ripple: 0.01
Enter the passband frequency: 1400
Enter the stopband frequency: 2000
Enter the sampling frequency: 8000

Output Waveform

6 KAISER
clc;
clearall;
closeall;
lp=input('Enter the passband ripple:');
ls=input('Enter the stopband ripple:');

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fp=input('Enter the passband frequency:');

fs=input('Enter the stopband frequency:');
f=input('Enter the sampling frequency:');
beta=input('Enter the beta value:');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(lp*ls))-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=kaiser(n1,beta);

% Low Pass Filter

b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m);
ylabel('Gain in db-->');
xlabel('(a)Normalised Frequency-->');
gridon;

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% 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(o/pi,m);
ylabel('Gain in db-->');
xlabel('(b)Normalised Frequency-->');
gridon;

Result:
Enter the passband ripple: 0.05
Enter the stopband ripple: 0.04
Enter the passband frequency: 1500
Enter the stopband frequency: 2000
Enter the sampling frequency: 9000
Enter the beta value: 5.8

Output Waveform

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VIVA Questions:
1. What is a Filter? What are its specifications?
2. Mention the types of filters.
3. Compare analog and digital filters.
4. Define a FIR filter.
5. What are the important properties of FIR filter?
6. What are the window techniques used?
7. Why the Kaiser window is important?
8. How do you calculate the length of Kaiser Window?
9. What are the functions used in designing a FIR filter in Matlab

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EXPERIMENT NO 8
Design and Implementation of IIR filter to meet given specifications.

Aim: To design and implement an IIR filter for given specifications.

Theory:
There are two methods of stating the specifications as illustrated in
previous program. In the first program, the given specifications are directly
converted to digital form and the designed filter is also implemented. In
the last two programs the butterworth and chebyshev filters are designed
using bilinear transformation (for theory verification).
Method I: Given the order N, cutoff frequency fc, sampling frequency fs
and the IIR filter type (butterworth, cheby1, cheby2).
● Step 1: Compute the digital cut-off frequency Wc (in the range -π
<Wc< π, with π corresponding to fs/2) for fc and fs in Hz. For example
let fc=400Hz, fs=8000Hz
Wc = 2*π* fc / fs = 2* π * 400/8000 = 0.1* π radians
For MATLAB the Normalized cut-off frequency is in the range 0 and 1,
where 1 corresponds to fs/2 (i.e.,fmax)). Hence to use the MATLAB
commands
wc = fc / (fs/2) = 400/(8000/2) = 0.1
Note: if the cut off frequency is in radians then the normalized
frequency is computed as wc = Wc / π
● Step 2: Compute the Impulse Response [b,a] coefficients of the
required IIR filter and the response type (lowpass, bandpass, etc) using
the appropriate butter, cheby1, cheby2 command. For example given a
butterworth filter, order N=2, and a high pass response, the coefficients

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[b,a] of the filter are computed using the MATLAB inbuilt command
‘butter’ as [b,a] =butter(N, wc , 'high');
Method 2:
Given the pass band (Wp in radians) and Stop band edge (Ws in
radians) frequencies, Pass band ripple Rp and stopband attenuation
As.
● Step 1: Since the frequencies are in radians divide by π to obtain
normalized frequencies to get wp=Wp/pi and ws=Ws/pi
If the frequencies are in Hz (note: in this case the sampling
frequency should be given), then obtain normalized frequencies as
wp=fp/(fs/2), ws=fstop/(fs/2), where fp, fstop and fs are the
passband, stop band and sampling frequencies in Hz
● Step 2: Compute the order and cut off frequency as
[N, wc] = BUTTORD(wp, ws, Rp, Rs)
● Step 3: Compute the Impulse Response [b,a] coefficients of the
required IIR filter and the response type as [b,a] =butter(N, wc ,
'high');
IMPLEMENTATION OF THE IIR FILTER
1. Once the coefficients of the IIR filter [b,a] are obtained, the next
step is to simulate an input sequence x[n], say input of 100, 200 &
400 Hz (with sampling frequency of fs), each of 20/30 points.
Choose the frequencies such that they are >, < and = to fc.
2. Filter the input sequence x[n] with Impulse Response, to obtain the
output of the filter y[n] using the ‘filter’ command.
3. Infer the working of the filter (low pass/ high pass, etc).

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MATLAB IMPLEMENTATION
BUTTORD Butterworth filter order selection.
[N, Wn] = BUTTORD (Wp, Ws, Rp, Rs) returns the order N of the lowest
order digital Butterworth filter that loses no more than Rp dB in the
passband and has at least Rs dB of attenuation in the stopband. Wp and
Ws are the passband and stopband edge frequencies, normalized from 0 to
1 (where 1 corresponds to pi radians/sample). For example,
Lowpass: Wp = .1, Ws = .2 Highpass: Wp = .2, Ws = .1
Bandpass: Wp = [.2 .7], Ws = [.1 .8] Bandstop: Wp = [.1 .8], Ws = [.2 .7]
BUTTORD also returns Wn, the Butterworth natural frequency (or, the "3
dB frequency") to use with BUTTER to achieve the specifications.
[N, Wn] = BUTTORD(Wp, Ws, Rp, Rs, 's') does the computation for an
analog filter, in which case Wp and Ws are in radians/second.
When Rp is chosen as 3 dB, the Wn in BUTTER is equal to Wp in
BUTTORD.

BUTTER Butterworth digital and analog filter design.

[B,A] = BUTTER(N,Wn) designs an Nth order lowpass digital
Butterworth filter and returns the filter coefficients in length N+1 vectors
B (numerator) and A (denominator). The coefficients are listed in
descending powers of z. The cutoff frequency Wn must be 0.0 <Wn< 1.0,
with 1.0 corresponding to half the sample rate. If Wn is a two-element
vector, Wn = [W1 W2], BUTTER returns an order 2N bandpass filter with
passband W1< W < W2. [B,A] = BUTTER(N,Wn,'high') designs a
highpass filter. [B, A] = BUTTER (N,Wn,'stop') is a bandstop filter if Wn
= [W1 W2].

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BUTTER (N,Wn,'s'), BUTTER (N,Wn,'high','s') and BUTTER

(N,Wn,'stop','s') design analog Butterworth filters. In this case, Wn is
in [rad/s] and it can be greater than 1.0.

MATLAB PROGRAMS
Butterworth Filters
1.Butterworth LPF
clc;
clear all;
close all;
lp=input('Enter the passband attenuation:');
ls=input('Enter the stopband attenuation:');
wp=input('Enter the passband frequency:');
ws=input('Enter the stopband frequency:');
fs=input('Enter the sampling frequency:');
w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=buttord(w1,w2,lp,ls);disp(n);disp(wn)
[b,a]=butter(n,wn);
w=0:0.01:pi;
[h,om]=freqz(b,a,w);
m=20*log10(abs(h));
an=angle(h);
figure(1);
subplot(2,1,1);
title('Butterworth Low Pass Filter');
plot(om/pi,m);
grid on;

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ylabel('Gain in db-->');
xlabel('(a)Normalised frequency-->');
subplot(2,1,2);
plot(om/pi,an);
xlabel('(b)Normalised frequency-->');
ylabel('Phase in radians-->');
grid on;
Result:
Enter the passband attenuation: 0.5
Enter the stopband attenuation: 50
Enter the passband frequency: 1200
Enter the stopband frequency: 2400
Enter the sampling frequency: 10000
Output Wave form

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2 Butterworth HPF
clc;
clear all;
close all;
format long
lp=input('Enter the passband attenuation:');
ls=input('Enter the stopband attenuation:');
wp=input('Enter the passband frequency:');
ws=input('Enter the stopband frequency:');
fs=input('Enter the sampling frequency:');
w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=buttord(w1,w2,lp,ls); disp(n);disp(wn)
[b,a]=butter(n,wn,'high');
w=0:0.01:pi;
[h,om]=freqz(b,a,w);
m=20*log10(abs(h));
an=angle(h);
figure(1);
subplot(2,1,1);
title('Butterworth Low Pass Filter');
plot(om/pi,m);
grid on;
ylabel('Gain in db-->');
xlabel('(a)Normalised frequency-->');
subplot(2,1,2);
plot(om/pi,an);

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xlabel('(b)Normalised frequency-->');
ylabel('Phase in radians-->');
grid on
Result:
Enter the passband attenuation: 0.5
Enter the stopband attenuation: 50
Enter the passband frequency: 1200
Enter the stopband frequency: 2400
Enter the sampling frequency: 10000

Output Wave form

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CHEBYSHEV FILTERS
1 Chebyshev LPF
clc;
close all;
clear all;
format long
rp=input('enter the passband ripple');
rs=input('enter the stopband ripple');
wp=input('enter the passband frequency');
ws=input('enter the stopband frequency');
fs=input('enter the sampling frequency');
w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=cheb1ord(w1,w2,rp,rs);
[b,a]=cheby1(n,rp,wn);
w=0:0.01:pi;
[h,om]=freqz(b,a,w);
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);plot(om/pi,m);
ylabel('gain in db-->');
xlabel('(a) normalised frequency-->');
subplot(2,1,2);plot(om/pi,an);
xlabel('(b) normalised frequency-->');
ylabel('phase in radians-->');

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Result:
enter the passband ripple 0.2
enter the stopband ripple 45
enter the passband frequency 1300
enter the stopband frequency 1500
enter the sampling frequency 10000

Output Waveform

2 Chebyshev HPF
clc;
close all;
clear all;
rp=input('enter the passband ripple');
rs=input('enter the stopband ripple');
wp=input('enter the passband frequency');
ws=input('enter the stopband frequency');
fs=input('enter the sampling frequency');

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w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=cheb1ord(w1,w2,rp,rs);
[b,a]=cheby1(n,rp,wn,'high');
w=0:0.01:pi;
[h,om]=freqz(b,a,w);
m=20*log10(abs(h));

an=angle(h);
subplot(2,1,1);plot(om/pi,m);
ylabel('gain in db-->');
xlabel('(a) normalised frequency-->');
subplot(2,1,2);plot(om/pi,an);
xlabel('(b) normalised frequency-->');
ylabel('phase in radians-->');

Result:
enter the passband ripple 0.3
enter the stopband ripple 60
enter the passband frequency 1500
enter the stopband frequency 2000
enter the sampling frequency 9000

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Output Waveform

VIVA Questions:
1. Define an IIR filter.
2. Compare Impulse invariance and bilinear transformations.
3. How do you convert analog filter prototype to a digital filter?
4. What are the functions used in MATLAB for designing a digital
Butterworth and Chebyshev low pass filter using BLT?
5. Draw typical responses of Chebyshev filter for order odd & even.
6. Compare FIR & IIR filters.

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PART II

EXPERIMENTS USING DSP PROCESSOR

CCS INTRODUCTION
Procedure
1. Double click on CCS icon

2. Create a new project: project 🡪 New

Type the project name (x.pjt) or (USN) & store in my projects folder

To open an existing project: project 🡪 open

3. Files to be added to the project: project 🡪 Add Files to project

a. c: \ CCStudio \ c6000 \ cgtools \ lib \ rts6700.lib

<Select type of file as: library>
b. c: \ CCStudio \ tutorial \ dsk6713 \ hello1 \ hello.cmd
< Select type of file as: Linker Command file>

4. File 🡪 New 🡪 source file 🡪 same as xx.c and compile

( Select type of file as: C/C++ file before saving) & add this C file to
your project

5. Project 🡪 Build. (obtain Build successful)

6. To execute ; File 🡪 load program (select pjtname.out file)

7. To Run : Debug 🡪 Run

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8. Observe the output on the stdout window or waveforms using graph or

CRO

9. To view waveforms View 🡪 Graph

changes in graph property dialog box to be made

a. Display type 🡪 Dual (to observe 2 waveforms)

b. Title 🡪 User defined

c. Start address – upper display 🡪 x (user defined variable array names

used in C program. Note: x, y should be made global to be used by

graph)

d. Start address – lower display 🡪 y

e. Acquisition Buffer size 🡪 depends on the array size

f. Display Buffer size 🡪 verify given output data size and change

g. DSP data type – 32-bit Floating pt

h. Auto scale 🡪 ON (if off 🡪 choose max yvalue=1)

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EXPERIMENT NO 1
Linear convolution for the given sequences
Aim: To perform linear convolution for the given sequences
The linear convolution sum is

C Program

# include<stdio.h>
# include<math.h>
float y[10];
main()
{
//the two sequences can be of different lengths, no need to zero pad for this
program
float h[4] = { 2,2,2,2};
float x[4] ={1,2,3,4};
intxlen=4;
inthlen=4;
int N=xlen+hlen-1;
intk,n;
for(n=0;n<N;n++) //outer loop for y[n] array
{ y[n]=0;
for(k=0;k<hlen;k++) //inner loop for computing each y[n] point
{ if (((n-k)>=0) & ((n-k)<xlen))
y[n]=y[n]+h[k]*x[n-k]; //compute output
} //end of inner for loop
printf("%f\t", y[n]);

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} //end of outer for loop

} //end of main

Exercise: Find the linear convolution of x[n]=[2,1,2,1] and

h[n]=[1,2,3,4]

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EXPERIMENT NO 2
Circular convolution of two given sequences

Aim: To perform circular convolution of two given sequences

The circular convolution sum is

C Program
# include<stdio.h>
# include<math.h>
float y[10]; //output sequence
main()
{ //input the two sequences, if of uneven lengths zero pad the smaller one
such that both are of same size
float x[5]={1,2,3,4,5};
float h[5]={2,1,3,4,5};
int N=5; // N=max of xlen and hlen//
intk,n,i;
for(n=0;n<N;n++) //outer loop for y[n] array
{ y[n]=0;
for(k=0;k<N;k++) //inner loop for computing each y[n]
point
{ i=(n-k)%N; //compute the index modulo N
if(i<0) //if index is <0, say x[-1], then convert to
x[N-1]
i=i+N;
y[n]=y[n]+h[k]*x[i]; //compute output

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} //end of inner for loop

printf("%f\t",y[n]);
} //end of outer for loop
} //end of main

Open ended Experiments :

Find the circular convolution of x[n]=[2 3 1 1] and h[n]=[1 3 5 3]

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EXPERIMENT NO 3
Computation of N- Point DFT of a given sequence

Aim: To compute the N (=4/8/16) point DFT of the given sequence

Theory:
The N point DFT of discrete time signal x[n] is given by the equation

Where N is chosen such that , where L=length of x[n]. To

implement using C program we use the expression

and allot memory space for real and

imaginary parts of the DFT X(k)

C Program
#include <stdio.h>
#include <math.h>
main()
{float y[16]; //for 8 point DFT to store real & imaginary
float x[4]={1,3,2,5}; //input only real sequence
float w;
int n,k,k1,N=8,xlen=4;
for(k=0;k<2*N;k=k+2)
{y[k]=0; y[k+1]=0; //initialize real &image parts
k1=k/2; //actual k index
for(n=0;n<xlen;n++)
{w=-2*3.14*k1*n/N; //careful about minus sign
y[k]=y[k]+x[n]*cos(w);

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y[k+1]=y[k+1]+x[n]*sin(w);
}
printf("%f+j%f \n",y[k],y[k+1]);
}
}//end of main

MATLAB verification
>> x=[1 3 2 5];
>>fft(x,8) 11.0000 -0.4142 - 7.6569i -1.0000 + 2.0000i
2.4142 - 3.6569i -5.0000 2.4142 + 3.6569i -1.0000 - 2.0000i
-0.4142 + 7.6569i

Open ended Experiments: Find the 8 point DFT of x[n]=[0.5,0,0.5,0]

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Signal Processing Laboratory Manual 21ECL58A BMS Institute of Technology and Management

EXPERIMENT NO 4
Impulse response of the given system

Aim: To find the Impulse response of the given first order / second
order system
Theory:
A linear constant coefficient difference equation representing a second
order system is given by

For a first order system the coefficients a2 and b2 are zero.

Since the difference equation contains past samples of output, i.e.,
y[n-1], y[n-2], its impulse response is of infinite duration (IIR). For
such systems the impulse response is computed for a large value of N,
say N=100 (to approximate n=∞).
Impulse response implies x[n]= δ[n], an impulse, and the initial conditions
x[-1], x[-2], y[-1] and y[-2] are zero.
(referexpt 2 &7of part A - h=impz(b,a,N); %impulse response verification
in MATLAB)
To implement in C the following algorithm is used.

Algorithm
1. Input the coefficients b0 b1 b2, a1 a2 (Say the coefficients from a 2nd
order Butterworth filter. Note coefficient of y[n] is a0=1 always)
2. Generate the impulse input of size n (1st element=1, rest all=0)
3. Compute the output as

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4. Repeat step 3 for n=0 through N (say 50,100, etc)

5.
C Program
#include <stdio.h>
float x[60],y[60];
main()
{float a1,a2,b0,b1,b2;
inti,j,N=20;
a1= -1.1430; a2= 0.4128;
b0=0.0675;b1=0.1349;b2=0.0675;
//generate impulse input
x[0]=1;
for(i=1;i<N;i++)
x[i]=0;
//generate output
for(j=0;j<N;j++)
{y[j]=b0*x[j];
if(j>0)
y[j]=y[j]+b1*x[j-1]-a1*y[j-1];
if ((j-1)>0)
y[j]=y[j]+b2*x[j-2]-a2*y[j-2];
printf("%f \t",y[j]);
}
}//end of main
Note: For a first order system just enter a2=0 and b2=0.
Open ended Experiments:
Find the impulse response of the given difference equation

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a. y[n]-1/9y[n-2]=x[n-1]
b. y[n]-3/4y[n-1]+1/8y[n-2]=x[n]+1/2x[n-1]
Open Ended Experiment

In this open end experiments students are expected to write programs to

create wave forms such as square wave, triangular wave and saw tooth
wave. Students may write separate programs for each waveform.

Mini Project
Students are expected to create a GUI based signal generator using
MATLAB. The signal generator will have three buttons to select three
signals ie., sine wave, square wave and triangular wave. Upon the
selection the wave form in time domain and frequency domain needs to be
displayed graphically. There should be a control to change the amplitude
and frequency of the signals.

Questions:
1. Why do we need SP processors?
2. What is the difference between Fixed and floating point
processors?
3. Explain the architectural features of DSP processor.
4. How do perform signed arithmetic on DSP processor?
5. What are the advantages and disadvantages of DSP processor?

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