DSP LAB -17ECL57

VSM’s
Somashekhar R Kothiwale Institute of
Technology, Nipani-591237

Department of Electronics &

Communication Engg

DSP LAB

18ECL57

BY

Prof.GOVIND M R

1 Dept Of ECE,VSMSRKIT
DSP LAB -17ECL57

INTRODUCTION TO MATLAB
MATLAB is a software package for high performance numerical computation
and visualization provides an interactive environment with hundreds of built in
functions for technical computation, graphics and animation. The MATLAB
name stands for MATrixLABoratory.
The diagram shows the main features and capabilities of MATLAB.

MATLAB

Computations External interface

Graphics
 Linear Algebra. Interface with C and
 2-D graphics.
 Signal processing.
 3-D graphics. FORTRAN Programs
 Polynomials
 Animation.
&interpolation.
 Quadrature
solution of ODES.

Toolbox

 Signal Processing
 Image Processing
 Statistics
 Control system
 Neutral networks
 Communications
and many more….

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DSP LAB -17ECL57

At its core, MATLAB is essentially a set (a ―toolbox‖) of routines (called ―m files‘

or ―mex files‖) that sit on your computer and a window that allows you to
create new variables with names (e.g. voltage and time) and process those
variables with any of those routines (e.g. plot voltage against time, find the
largest voltage, etc.)
MATLAB® is a high-level language and interactive environment that enables
you to perform computationally intensive tasks faster than with traditional
programming languages such as C, C++, and Fortran.
Key Features:
 High-level language for technical computing.
 Development environment for managing code, files, and data.
 Interactive tools for iterative exploration, design, and problem solving.
 Mathematical functions for linear algebra, statistics, Fourier analysis,
 Filtering, optimization, and numerical integration.
 2-D and 3-D graphics functions for visualizing data.
 Tools for building custom graphical user interfaces.
 Functions for integrating MATLAB based algorithms with external
applications and languages, such as C, C++, Fortran, Java, COM, and
Microsoft® Excel®.
MATLAB Windows:
MATLAB works with through three basic windows
Command Window: This is the main window .It is characterized by
MATLAB command prompt(>>) when you launch the application program
MATLAB puts you in this window all commands including those for user-
written programs ,are typed in this window at the MATLAB prompt.
Graphics window: The output of all graphics commands typed in the
command window are flushed to the graphics or figure window, a separate
gray window with white background color the user can create as many
windows as the system memory will allow.
Edit window: This is where you write edit, create and save your own
programs in files called M files.

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Input-output:
MATLAB supports interactive computation taking the input from the screen
and flushing the output to the screen. In addition it can read input files and
write output files.

Data Type: The fundamental data-type in MATLAB is the array. It

encompasses several distinct data objects- integers, real numbers, matrices.
Character strings, structures and cells. There is no need to declare variables as
real or complex, MATLAB automatically sets the variable to be real.
Dimensioning: Dimensioning is automatic in MATLAB. No dimension
statements are required for vectors or arrays. We can find the dimensions of an
existing matrix or a vector with the size and length commands.
Basic Instructions in Matlab
1) T = 0:1:10
This instruction indicates a vector T which as initial value 0 and final value
10 with an increment of 1.
Therefore,
T=

0 1 2 3 4 5 6 7 8 9 10
2) T= Zeros (2, 3)
The above instruction creates a vector of two rows and three columns whose
values are zero.
T=

0 0 0
0 0 0

3) T=Ones (3,2)
The above instruction creates a vector of three rows and two columns whose
values are one
T=
1 1
1 1
1 1

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4) a = [1 2 3] b = [4 5 6]

a.*b= [41018], which is the result of individual elements.

5) Matrix and Array Operations

MATLAB allows you to process all of the values in a matrix using a single
arithmetic operator or function.
To create a matrix that has multiple rows, separate the rows with semicolons.
 a = [1 2 3; 4 5 6; 7 8 10]
a=
1 2 3
4 5 6
7 8 10
 a + 10
ans =
11 12 13
14 15 16
17 18 20
 sin(a)
ans =
0.8415 0.9093 0.1411
-0.7568 -0.9589 -0.2794
0.6570 0.9894 -0.5440
 To transpose a matrix, use a single quote ('):
a'
ans =
1 4 7
2 5 8
3 6 10

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You can perform standard matrix multiplication, which computes the inner
products between rows and columns, using the * operator. For example,
confirm that a matrix times its inverse returns the identity matrix:

 p = a*inv(a)
p=
1.0000 0 -0.0000
0 1.0000 0
0 0 1.0000
Notice that p is not a matrix of integer values. MATLAB stores numbersas
floating-point values, and arithmetic operations are sensitive to small
differences between the actual value and its floating-point representation. You
can display more decimal digits using the format command:
 Format long:
p = a*inv(a)
p=
1.000000000000000 0 -0.000000000000000
0 1.000000000000000 0
0 0 0.999999999999998
Reset the display to the shorter format using format short format affects only
the display of numbers, not the way MATLAB computes or saves them. To
perform element-wise multiplication rather than matrix multiplication, use
the.* operator:
 p = a.*a
p=
1 4 9
16 25 36
49 64 100

The matrix operators for multiplication, division, and power each have a
corresponding array operator that operates element-wise. For example, raise
each element of a to the third power:

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 a.^3
ans =
1 8 27
64 125 216
343 512 1000
Concatenation:
Concatenation is the process of joining arrays to make larger ones. In fact, you
made your first array by concatenating its individual elements. The pair of
square brackets [] is the concatenation operator.

 A = [a,a]
A=
1 2 3 1 2 3
4 5 6 4 5 6
7 8 10 7 8 10

Concatenating arrays next to one another using commas is called horizontal

concatenation. Each array must have the same number of rows. Similarly,
when the arrays have the same number of columns, you can concatenate
vertically using semicolons.

 A = [a; a]
A=

1 2 3

4 5 6

7 8 10

1 2 3

4 5 6

7 8 10

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Complex Numbers:

Complex numbers have both real and imaginary parts, where the imaginary
unit is the square root of –1.

 sqrt(-1)
ans = 0 + 1.0000i

To represent the imaginary part of complex numbers, use either i or j.

c = [3+4i, 4+3j, -i, 10j]

c = 3.0000 + 4.0000i 4.0000 + 3.0000i 0 - 1.0000i 0

+10.0000i

Array Indexing:

Every variable in MATLAB is an array that can hold many numbers. When you
want to access selected elements of an array, use indexing.
For example, consider the 4-by-4 magic square A:
 A = magic(4)
A=
16 2 3 13

5 11 10 8

9 7 6 12

4 14 15 1

There are two ways to refer to a particular element in an array. The most
common way is to specify row and column subscripts, such as
 A(4,2)
ans = 14
 Less common, but sometimes useful, is to use a single subscript that
traverses down each column in order:
 A(8)
ans = 14

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 Using a single subscript to refer to a particular element in an array is called

linear indexing.
 If you try to refer to elements outside an array on the right side of an
assignment statement, MATLAB throws an error.
 test = A(4,5)
 Attempted to access A(4,5); index out of bounds because size(A)=[4,4].
However, on the left side of an assignment statement, you can specify
elements outside the current dimensions. The size of the array increases to
accommodate the newcomers.
 A(4,5) = 17
Example:
A=
16 2 3 13 0

5 11 10 8 0

9 7 6 12 0

4 14 15 1 17

To refer to multiple elements of an array, use the colon operator, which allows
you to specify a range of the form start:end. For example, list the elements in
the first three rows and the second column of A:

 A(1:3,2)
ans =2

11

The colon alone, without start or end values, specifies all of the elements in
that dimension. For example, select all the columns in the third row of A:
 A(3,:)
ans = 9 7 6 12 0
The colon operator also allows you to create an equally spaced vector of
values using the more general form start:step:end.

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6) Plot (t, m)
If m= [6 7 8 9]
t= [1 2 3 4]
This instruction will display a figure window which indicates the plot of
m versus t.

7) Stem (t,m)
This Instruction will display a figure window as shown below,

8) Subplot: This function divides the figure window into rows and
Columns.
Example: subplot(3,1,2)-----3-rows, 1-column, 2-location.

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WARM UP CYCLE: GENERATION OF SIGNALS

Aim: Generation of impulse signal.
clc;
clear all;
close all;
t=-5:1:5;
y=[zeros(1 ,5),ones(1,1),zeros(1,5)];
stem(t,y);
xlabel('Time');
ylabel('Amplitude');
title('Discrete time impulse function');

Aim: Generation of unit step signal.

clc;
clear all;
close all;
n=input('Enter the n value:');
t=0:1:n-1;
stem(t,y);
xlabel('Time');
ylabel('Amplitude');
title('Discrete time unit step function');

Aim: Generation of ramp signal.

clc;
clear all;
close all;
n=input('Enter the n value:' );
t=0:n;
stem(t,t,'r');
hold on;
plot(t,t,'k');
xlabel('Time');
ylabel('Amplitude');
title('Ramp function');
legend('Discrete','continuous');

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Aim:Generation of exponential signal.

clc;
clear all;
close all;
n=input('enter the N value:');
t=0:0.1:n;
a=input('enter the A value:');
y=exp(a*t);
stem(t,y,'y');
hold on;
plot(t,y,'r');
xlabel('Time');
ylabel('Amplitude');
title('Exponential function');
legend('Discrete','continuous');

Aim: Generation of sine wave y(n)=sin(2πfn).

clc;
clear all;
close all;
t=0:0.001:5;
f=1;
y=sin(2*pi*f*t);
plot(t,y,'g');
xlabel('Frequency');
ylabel('amplitude');
title('sine wave');

Aim: Generation of cosine wave y(n)=cos (2πfn).

clc;
clear all;
close all;
t=0:0.001:5;
f=1;
y=cos(2*pi*f*t);
plot(t,y,'g');
xlabel('Frequency');
ylabel('amplitude');
title('cosine wave');

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1. SAMPLING THEOREM
Aim: To verify sampling theorem for the following cases:
i) Under sampling ii) Right sampling iii) Over sampling

Program:
clc; % clears the command window
clear all;
close all;% clears the variables declared
k=input ('Enter the no of cycles = ');
a=input ('Enter the input signal amplitude= ');
fm=input ('Enter the input frequency = ');
t=0:1/(fm*fm):k/fm;
y=a*cos(2*pi*fm*t)
figure;
subplot(2,2,1)
plot(t,y)
grid on;
xlabel('------t');ylabel('amplitude');
title('The input signal');

fnq=2*fm

% under sampling
fs=3/4*fnq;
tx=0:1/fs:k/fm;
ys=a*cos(2*fm*pi*tx)
subplot(2,2,2);
stem(tx,ys);
hold on
plot(tx,ys,'r');xlabel('------tx');ylabel('amplitude');
title('The under sampling case');

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% Nyquist sampling
fs=fnq;
tx=0:1/fs:k/fm;
ys=a*cos(2*pi*fm*tx)
subplot(2,2,3);
stem(tx,ys);
hold on;
plot(tx,ys,'g');xlabel('------tx');ylabel('amplitude');
title('The RIGHT SAMPLING case');

% over sampling
fs=10*fnq;
tx=0:1/fs:k/fm;
ys=a*cos(2*pi*fm*tx)
subplot(2,2,4);
stem(tx,ys);
hold on;
plot(tx,ys,'r');xlabel('------tx');ylabel('amplitude');
title('The OVER SAMPLING case');

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2. Convolution and its properties

2a Aim: To find the Linear Convolution of given two sequences
Program:
clc;
clear all;
close all;

x=input('Enter the first input sequence x[n]= ');

xsi=input('Enter the starting index of x[n]= ')

h=input('Enter the first input sequence h[n]= ');

hsi=input('Enter the starting index of h[n]= ')

%To calculate the convolution of x[n] and h[n]%

y = conv(x,h)

%To generate the index arrays%

n1 = xsi:1:length(x)+xsi-1
n2 = hsi:1:length(h)+hsi-1
n3 = xsi+hsi:1:length(x)+length(h)+xsi+hsi-2

%To sketch the result%

subplot(311),stem(n1,x);
xlabel('n1----->'),ylabel('Input Sequence----->')
title ('Input Sequence x(n)')
subplot(312),stem(n2,h);
xlabel('n2----->'),ylabel('Impulse Response----->')
title ('Input Sequence h(n)')
subplot(313),stem(n3,y);
xlabel('n3----->'),ylabel('Output Sequence----->')
title ('Convolution of two sequences')

Result:

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2bAim:To find the Circular Convolution of given two

sequences

Program:
clc;
clear all;
close all;

x=input('Enter the first sequence x[n]= ');

h=input('Enter the second sequence h[n]=');

N = max(length(x),length(h));
Y=cconv(x,h,N) %y=xN h%

stem(Y);
xlabel('N----->'),ylabel('Amplitude----->')
title('Plot of circularly convoluted sequence');

Result:

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2c.Aim: To prove the commutative ,distributive and associative

property of convolution
1. Commutative property= aN b=bN a
2. Associative property= aN (bN c) = (aN b) N c
3. Distributive property= aN (b+c) = (aN b)+(aN c)
Program:
clc;
clear all;
close all;

a1=input('enter the a1 sequence a1=')

b1=input('enter the b1 sequence b1=')
c1=input('enter the c1 sequence c1=')

N1=max(length(a1),length(b1))
N=max(N1,length(c1))

a=[a1,zeros(1,N-length(a1))]
b=[b1,zeros(1,N-length(b1))]
c=[c1,zeros(1,N-length(c1))]

x=cconv(a,b,N1)
y=cconv(b,a,N1) %commutative property aN b=bN a
if x==y
disp('commutative property is proved')
else
disp('commutative property is not proved')
end

t=cconv(b,c,N)
y1=cconv(a,t,N) %associative property aN (bN c)=(aN b) N c
y2=cconv(x,c,N)
if y1==y2

disp('associative property is proved')

else
disp('associative property is not proved')
end

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t1= b + c
y3=cconv(a,t1,N)
t2=cconv(a,c,N) % distributive property aN (b+c)=(aN b)+(aN c)
y4= x + t2

if y3==y4

disp('distributive property is proved')

else
disp('distributive property is not proved')
end

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3. Auto correlation and its properties.

Aim:To find the Auto correlation of a given sequence and to verify
its properties.

Properties of Auto-correlation:
Property 1:Rxx(l) = Rxx(-l) (Symmetry property).
Property 2: |rxx(l)| ≤rxx(0) = Ex (Energy property).
Program:
clc;
clear all;
close all;
x=input('Enter the sequence x[n]= ');
xsi=input('enter the starting index=');
Rxx=xcorr(x,x)
energy=sum(x.^2)
c_i=ceil(length(Rxx)/2)
Rxx_0=Rxx(c_i)
if Rxx_0==energy
disp('Eneregy property proved.');
else
disp('EnergyProperty not proved');
end
Rxxf=fliplr(Rxx)
if Rxx==Rxxf
disp('It is even.');
else
disp('It is not even.');
end
n1= xsi:length(x)+xsi-1;
n2= -(length(x)-1):(length(x)-1);
subplot(3,1,1),stem(n1,x),xlabel('n1'),ylabel('amplitude');
subplot(3,1,2),stem(n2,Rxx),xlabel('n2'),ylabel('amplitude');
subplot(3,1,3),stem(n2,Rxxf),xlabel('n2'),ylabel('amplitude');

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4. Cross correlation and its properties.

Aim:To find the Cross-correlation of two given sequences and to
verify its properties.
Properties of Cross-correlation:
Property 1: Rxy(l) = Ryx(-l) (Symmetry property).
Property 2: |Rxy(l)| ≤ sqrt[Ex.Ey] (Energy property).

Program:
clc;
clear all;
close all;
x=input('Enter the first sequence=');
xsi=input('enter the starting index of x=');
xei=input('enter the ending index of x=');
y=input('Enter the second sequence= ');
ysi=input('enter the starting index of y=');
yei=input('enter the ending index of y=');
Ex=sum(x.^2);
Ey=sum(y.^2);
energy=sqrt(Ex*Ey);
Rxy=xcorr(x,y)
Ryx=xcorr(y,x)
Ryxf=fliplr(Ryx);
if Rxy==Ryxf
disp('Symmetry Property is proved');
else
disp('Symmetry property not proved');
end
n1=xsi:length(x)+xsi-1;
n2=ysi:length(y)+ysi-1;
n3=(xsi-yei):(xei-ysi)
subplot(1,4,1),stem(n1,x),xlabel('lag'),ylabel('amplitude');
subplot(1,4,2),stem(n2,y),xlabel('lag'),ylabel('amplitude');
subplot(1,4,3),stem(n3,Rxy),xlabel('lag'),ylabel('amplitude');
subplot(1,4,4),stem(n3,Ryxf),xlabel('lag'),ylabel('amplitude');
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5. Solution to Difference Equation

Aim:To solve the given difference equation (DE) with initial
conditions.
Program:
clc;
clear all;
close all;
a=input('Enter the coefficient of y')
b=input('Enter the coefficient of x')
xi=input('Enter the initial values x[-1],x[-2],..= ')
yi=input('Enter the initial values y[-1],y[-2],....= ')
x=input ('Enter the input sequence x= ');
zi=filtic(b,a,yi,xi);
disp('The response of the given system is,');
y = filter(b,a,x,zi)
n= 0:1:length(y)-1;
stem(n,y);
xlabel('n----->'),ylabel('amplitude----->');
title ('solution to a given differential equation with initial
conditions')

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6. Discrete Fourier Transformation

Aim: To compute N point DFT of a given sequence x[n] and to plot
magnitude and phase spectrum.

Program:
clc;
clear all;
close all;
x = input('enter the input seq=')
N = length(x);
X = zeros(N,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/N)
end
end

mag=abs(X)
phase=angle(X)*180/pi

t = 0:N-1
subplot(311)
stem(t,x);
xlabel('Time (s)');ylabel('Amplitude');
title('Time domain - Input sequence')
subplot(312)
stem(t,mag)
xlabel('Frequency');ylabel('|X(k)|');
title('Frequency domain - Magnitude response')
subplot(313)
stem(t,phase)
xlabel('Frequency');ylabel('phase(X(k))');
title('Frequency domain - phase response')

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with inbuilt function

clc;
clear all;
close all;
x=input('enter the first sequence=')
N=length(x);
X=fft(x,N)
mag=abs(X)
phase=angle(X)*180/pi

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))
ylabel('phase(X(k))');
title('Frequency domain - phase response');

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7. Discrete Fourier Transformation properties

7a.Aim: To verify the properties of DFT like parseval’s theorem and
circular convolution etc.

Linearity property: {a.x(n)+b.y(n)} a.X(K )+ b.Y(K)

Parseval‘s theorem:

Time shift property : x(n-m) WN-(m*k) X(K)

% Program To prove the Linearity property
clear all;
close all;

N=input('enter the value for N= ')

x1=input('enter the x1 input x1=')
x2=input('enter the x2 input x2=')
a=input('enter the x2 input a=')
b=input('enter the x2 input b=')

y1=a.*x1+b.*x1
Y1=fft(y1,N)
X1=fft(x1,N)
X2=fft(x2,N)
Y2=a.*X1+b.*X2

MagY1=abs(Y1)
MagY2=abs(Y2)
phaseY1=angle(Y1)*180/pi
phaseY2=angle(Y2)*180/pi

figure
k=0:1:(N-1)
subplot(2,2,1);stem(k,MagY1)
xlabel('N-point');ylabel('Magnitude of Y1')
subplot(2,2,3);stem(k,MagY2)
xlabel('N-point');ylabel('Magnitude of Y2')

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subplot(2,2,2);stem(k,phaseY1)
xlabel('N-point');ylabel('phase of Y2')
subplot(2,2,4);stem(k,phaseY2)
xlabel('N-point');ylabel('phase of Y2')
if Y1==Y2
disp('Linearity property satisfied for the given sequence')
else
disp('Linearity property not satisfied for the given
sequence')
end

% Program To prove the time shifting property of DFT

clc;
clear all;
close all;
x = input('enter the input sequence=')
N = length(x); %data length
W = exp(-j*pi*2/N)
X = fft(x);
subplot(1,2,1)
stem(x);xlabel('---------->N');
ylabel('amplitude');title('plot of input signal x(n)')
k = 0:N-1; % frequency index
m = 2; % number of shift
Y= W.^(-m*k) .* X;
y= ifft(Y)
mag=abs(y)
subplot(1,2,2)
stem(mag);xlabel('---------->N');
ylabel('amplitude');title('plot of input shifted signal x(n-
m)')

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% Program To prove the parsevals theorem

clc;
clear all;
close all;
x1=input('enter the 1st input seq=')
x2=input('enter the 2nd input seq=')
N=4;
X1=fft(x1,N)
X2=fft(x2,N)
y1=sum(x1.*conj(x2))
y2=sum(X1.*conj(X2))./N

if y1==y2
disp('parsvals theorem proved');
else
disp('parsevals theorem not proved')

end

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7b.Aim: DFT computation of square pulse and Sinc function etc.

a) DFT Computation of square pulse without built in function

clc;

clear all;

close all;

x=[zeros(1,32),ones(1,64),zeros(1,32)]

X=fft(x)

X1=fftshift(X)

subplot(2,1,1)

plot(x)

xlabel('time')

ylabel('amplitude')

title('time domain squer pulse')

subplot(2,1,2)

plot(abs(X1))

xlabel('frequency')

ylabel('magnitude')

title('forrier transform of sequence')

Output:

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B) DFT Computation of square pulse with built in function

clc;

clear all;

close all;

A=input('Enter the amplitude of signal pulse=')

f=input('enter the amplitude of squre pulse wave=')

dc=input('enter the duty cycle of squere wave=')

t=0:0.001:1

x=A*square(2*pi*f*t,dc)

X=fft(x)

X1=fftshift(X)

subplot(2,1,1)

plot(x)

xlabel('time')

ylabel('amplitude')

title('time domain squer pulse')

subplot(2,1,2)

plot(abs(X1))

xlabel('frequency')

ylabel('magnitude')

title('forrier transform of sequence'

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C) DFT Computation of sinc function

clc;

clear all;

close all;

n=-pi:0.005:pi

x=sinc(n)

X=fft(x)

X1=fftshift(X)

subplot(2,1,1)

plot(x)

xlabel('time')

ylabel('amplitude')

title('time domain sinc function')

subplot(2,1,2)

plot(abs(X1))

xlabel('frequency')

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ylabel('magnitude')

title('sinc transform of sequence')

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8. FIR Filter design using windowing technique

Aim:To Design a digital FIR low pass filter with following specifications.

a) Pass band cut-off frequency : ωP=__________rad

b) Stop band cut-off frequency : ωS=__________rad

c) Stop band attenuation : kS=__________dB

Choose an appropriate window function and determine impulse response and

provide a plot of frequency response of the designed filter.
Design steps:

1. Choose an appropriate window function from the stop band attenuation

using the following table.
Window function Stop band
Sl. No. K
w[n] attenuation

Rectangular(Boxcar)
1 21 dB 2
window

Triangular (Bartlett)
2 27 dB 4
window

3 Hanning window 44 dB 4

4 Hamming window 53 dB 4

5 Blackman window 75 dB 6

2. Determine the order of the FIR filter (number of coefficients) using the
relation;
2k
N≥
 p  s 
Note: Choose N as odd. If N is even select next odd integer.


3. Choose the cut-off frequency of the filter as ωC=ωP+
2

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DSP LAB -17ECL57

N  1
4. Choose α = so that linear phase is ensured.
2
5. The impulse response of the filter is given by

sin c n   
h(n) = *w(n); for 0<n<N-1 & n ≠ α
 n   
c
*w(n) ; for n = α

where w(n) is window function.

  j  N21     N 3 

e   
 
  N 1  2    
 N 1 
H(ω) =   * h 2    2hn  cos   n   2   
     n 0      
 
Design example:

Design a digital FIR low pass filter with following specifications.

a) Pass band cut-off frequency : 0.3π rad

b) Stop band cut-off frequency : 0.45π rad

c) Stop band attenuation : 50 dB

I step: Choose an appropriate window function from the stop band attenuation
specification.

As Ks = 50 dB select Hamming window.

(Note: Though Blackman window can be used it results in higher transition

bandwidth)

II step:Compute the order of the FIR filter (number of coefficients) using the
relation,

2k
N≥
 p  s  [K=4 for Hamming window]
N≥ 53.5 (choose N=55)

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
III step:Choose the cut-off frequency of the filter as ωC= ωP+ rad.
2

 N 1
IV step: Choose α =   so that linear phase is ensured.
 2 

V step: The impulse response of the filter is given by,

sin c n   
h(n) = *w(n) ; for0<n<N-1 & n ≠ α.
 n   
where w(n) is window function.

sin 0.3 n  27    2n 

* 0.54  0.46 cos  ;0<n<N-1 & n ≠ α
 n  27 
h(n) =
  N  1 
For ex : The impulse response coefficients are,
h(0)=0.00029114 = h(54)
h(1)= -5.9806*(10-4) = h(53)……..
Note: when n=α
c
i.e.h(27) = *w(27) =0.3*1=0.3

VI step: Frequency response can be obtained using equation,

  j  N21     N 3 

e    N 1
 
   2 
   N  1   
H(ω) =  h  
 *   2  n 0
  2 h n  cos  
 n     
      2   


To calculate the Phase, θ(ω) of the FIR filter:

For any N,

  N 1
      ; H    0
2  2 
3  N 1
      ; H    0
2  2 

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Program:
clc;
clear all;
close all;
wp= input('Enter the pass band edge (rad)= ');
ws= input('Enter the stop band edge (rad)= ');
ks= input('Enter the stop band attenuation (dB)= ');
%If 43<Ks<54 choose hamming window.
%To select N,order of filter.
N= (2*pi*4)./(ws-wp); % k=4 for Hamming window.
N= ceil(N); %To round-off N to the next integer.
r = rem(N,2); %Choose odd N.
if(r==0)
N=N+1;
end
wc=(wp+(ws-wp)/2)./pi
% To obtain h(n)
h= fir1(N-1,wc,hamming(N))
% Frequency response
freqz(h,1); % 1 is the normalized frequency
title('Frequency response of the lowpass digital FIR filter')
figure(2), stem(h);
title('Impulse response of the lowpass digital FIR filter')

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9. IIR low pass filter

BUTTERWORTH FILTER:

Program:
clc; clear all; close all;

rp=input('enter passband ripple'); % enter rp<3

rs=input('enter stop band ripple'); % enter rs > 20

fp=input('enter passband frequency');

fs=input('enter stopband frequency'); %fs > fp

f=input('enter the sampling frequency'); % f>>>2fp

w1= 2*fp/fs; %Compute normalized frequencies

w2=2*fs/f;

%%%%%LOW PASS FILTER

[n,wn]=buttord(w1,w2,rp,rs,'s'); % Compute
order and cutoff frequency
[b,a]=butter(n,wn,'s'); % Compute poles and
zeros of frequency response w=0:0.001:pi;

[h,om]=freqs(b,a,w);

M=20*log10(abs(h));

subplot(4,1,1);

plot(om/pi,M);

xlabel('Normalized frequency');

ylabel('Gain in dB');

title('Low pass filter');

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%%%%%HIGH PASS FILTER

[b,a]=butter(n,wn,'high','s');

w=0:0.001:pi;

[h,om]=freqs(b,a,w);

M=20*log10(abs(h));

subplot(4,1,2);

plot(om/pi,M);

xlabel('Normalized frequency');

ylabel('Gain in dB');

title('High pass filter');

%%%%%BAND PASS FILTER

[n,wn]=buttord(w1,w2,rp,rs,'s');

wn=[w1,w2];

[b,a]=butter(n,wn,'bandpass','s');

w=0:0.001:pi;

[h,om]=freqs(b,a,w);

M=20*log10(abs(h));

subplot(4,1,3);

plot(om/pi,M);

xlabel('Normalized frequency');

ylabel('Gain in dB');
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title('Band pass filter');

%%%%%BAND STOP FILTER

[n,wn]=buttord(w1,w2,rp,rs,'s');

wn=[w1,w2];

[b,a]=butter(n,wn,'stop','s');

w=0:0.001:pi;

[h,om]=freqs(b,a,w);

M=20*log10(abs(h));

subplot(4,1,4);

plot(om/pi,M);

xlabel('Normalized frequency');

ylabel('Gain in dB');

title(„Band stop pass filter');

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Features of TMS320C6713 DSK Kit

The C6713 DSK is a low-cost standalone development platform that enables
users to evaluate and develop applications for the TI C67xx DSP family. The
DSK also serves as a hardware reference design for the TMS320C6713 DSP.
Key features:
 A Texas Instruments TMS320C6713 DSP operates at 225 MHz and
delivers up to 1800 million instructions per second (MIPs).
 4 user accessible LEDs and DIP switches.
 Software board configuration through registers implemented in CPLD.
 Single voltage power supply (+5V).
 Embedded JTAG support via USB.
 High-quality 24-bit stereo codec.
 Four 3.5mm audio jacks for microphone, line in, speaker and line out.
 512K words of Flash and 16 MB SDRAM.
 Expansion port connector for plug-in modules.
 On-board standard IEEE JTAG interface.
 A complete Integrated Development Environment (IDE), an efficient
optimizing C/C++ compiler assembler, linker, debugger, an a advanced
editor with Code Maestro™ technology for faster code creation, data
visualization, a profiler and a flexible project manager.
 DSP/BIOS™ real-time kernel.
 Target error recovery software.
 DSK diagnostic tool.
 "Plug-in" ability for third-party software for additional functionality.

Fig.1 Block Diagram of TMS320C6713 DSK

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DSP PROCESSOR ARCHITECTURE AND APPLICATION

 Memory architecture designed for streaming data, using DMA extensively.
 Separate program and data memories (Harvard architecture).
 Special SIMD (single instruction, multiple data) operations.
 Special arithmetic operations, such as fast multiply-accumulates (MACs).
Many fundamental DSP algorithms, such as FIR filters or the Fast Fourier
transform (FFT) depend heavily on multiply-accumulate performance.
 Bit-reversed addressing, a special addressing mode useful only for
calculating FFTs.
 Deliberate exclusion of a memory management unit. DSPs frequently use
multitasking operating systems, but have no support for virtual memory or
memory protection. Operating systems that use virtual memory require
more time for context switching among processes, which increases latency.

Applications
The main applications of DSP are audio signal processing, audio compression,
digital image processing, video compression, speech processing, speech
recognition and digital communications. Specific examples are speech
compression and transmission in digital mobile phones, equalisation of sound
in Hi-Fi equipment, weather forecasting, economic forecasting, seismic data
processing, analysis and control of industrial processes, computer-generated
animations in movies, medical imaging such as CAT scans and MRI, image
manipulation, and audio effects for use with electric guitar amplifiers. A
further application is very low frequency (VLF) reception with a PC soundcard.

Difference between Microcontroller and DSP Processor

General purpose microprocessor and microcontrollers follow Von Neumann
architecture (or CISC architecture) where Data Memory and Program Memory
have common data and address bus. Instruction execution cycles may be
different for different instructions (depending on the type of operation).
Recent high speed microcontrollers have RISC architecture where most
instructions take a single cycle for execution.

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Digital Signal Processors follow Harvard architecture where Data memory and
Program Memory have separate data and address buses. Multiply and
accumulate, Program memory fetch and data memory write are all executed in
a single cycle.

Microcontrollers are primarily used in control-oriented applications that are

interrupt driven sensing and controlling external events. DSPs, meanwhile, are
traditionally found in systems that require the precision processing of analog
signals. As today‘s systems gain in complexity, Microcontrollers are being given
some signal processing capabilities while more DSPs are capable of executing
real-time event-driven functions.

Procedure for execution in TMS3206713 Simulator:

 Open CCS Studio Setup3.1

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 Select 67xx Family

 Select Simulator Platform

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DSP LAB -17ECL57

 Select 6713 Device cycle accurate simulator

 Now click on “<<Add” Button and then click on Save & Quit button.
 Next Click on “Yes” Button to open CCS studio.

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DSP LAB -17ECL57

 Project→New→ProjectName→Location(C:\CCStudio_v3.1\MyProjects\)→Pr
ojecttype (.out Executable) →Target (TMS320C67xx).

 Click on File→New→Source File.

 Write the code in a new source file. Save it in the project folder with .c

43 Dept Of ECE,VSMSRKIT
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file format.
 Add this to the project. Project will be having .pjt extension. Right click
on.pjt file created, add the .c file that you have written.
 Two other files are to be added to project. One is library file (*.lib) and
other is Linker command file (*.cmd).
 Add rts6713.lib
C:\CCStudio_v3.1\C6000\cgtools\lib\rts6700.lib
 Add hello.cmd
C:\CCStudio_v3.1\tutorial\dsk6713\hello1\hello.cmd
 Compile the program using the „Project-compile‟ pull down menu or by
clicking the shortcut icon.
 Build the program using the „Project-Build‟ pull down menu or by
clicking the shortcut icon.
 Load the .out file (which will be in the debug folder where the project
is stored)using the „File-load program‟ pull down menu.
 Run the program using „Debug - Run‟ pull down menu or by clicking
the shortcut icon.

Procedure for execution in TMS320DSK6713 kit:

 CCS Studio Setup v3.1 →Family (67xx)→Platform

(dsk)→Endianness→Little endian→Add it to panel. Click on
6713dsk,save and quit.
 Connect the power card to the DSP kit.
 Connect the data cable → USB from PC
 After getting the project window, DEBUG→ CONNECT.
 Rest of the procedure is same as compared to simulator running.

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1. LINEAR CONVOLUTION
Aim: To compute Linear Convolution of the two given sequences:

Program:
#include<stdio.h>
int y[20];
main()
{

int x[]={ };
int h[]={ };
int l= ,n,k;
for(n=0;n<l;n++)
{

y[n]=0;
for(k=0;k<=n;k++)
y[n]+=x[k]*h[n-k];

}
Printf(“\n Linear convolution output is\n”);
For(n=0;n<l;n++)
Printf(“y[%d]=%d”,n,y[n]);
}

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2. CIRCULAR CONVOLUTION

Aim: C-program to compute theCircular Convolution for two given

sequences of length four:

Program:
#include<stdio.h>

int y[20];
main()
{

int x[]={ };
int h[]={};

int l=4,n,k;
int a[20],b[20],c[20],d[20];

y[0]=0;
a[0]=h[0];

for(k=1;k<l;k++) /*Folding h(n) to h(-n).*/

a[k]=h[l-k];

for(n=0;n<l;n++) /*Circular convolution.*/

y[0]=y[0]+x[n]*a[n];

y[1]=0;

b[0]=a[l-1];

for(k=1;k<l;k++) /*Folding h(n) to h(-n).*/

b[k]=a[k-1];

for(n=0;n<l;n++) /*Circular convolution.*/

y[1]=y[1]+x[n]*b[n];

y[2]=0;
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DSP LAB -17ECL57

c[0]=b[l-1];

for(k=1;k<l;k++) /*Folding h(n) to h(-n).*/

c[k]=b[k-1];

for(n=0;n<l;n++) /*Circular convolution.*/

y[2]=y[2]+x[n]*c[n];

y[3]=0;

d[0]=c[l-1];

for(k=1;k<l;k++) /*Folding h(n) to h(-n).*/

d[k]=c[k-1];

for(n=0;n<l;n++) /*Circular convolution.*/

y[3]=y[3]+x[n]*d[n];

Printf(“\n The output of circular convolution is\n”);

for(n=0;n<l;n++)

Printf(“y[%d]=%d\n”,n,y[n]);

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DSP LAB -17ECL57

3. Computation of n- point dft of a given sequence

Aim: To compute N-point DFT of a given sequence

Program:
#include<stdio.h>
#include<math.h>
main()
{
int N = ,n = 0, k = 0;
int x[] = { };
floatsumRe=0,sumIm=0,costerm=0,sinterm=0,pi= 3.1416;
for(k=0 ; k<N ; k++)
{
sumRe = 0;
sumIm = 0;
for (n=0; n<N ; n++)
{
costerm = cos(2*pi*k*n/N);
sinterm = sin(2*pi*k*n/N);
sumRe = sumRe + x[n] * costerm;
sumIm = sumIm - x[n] * sinterm;
}
printf("X[%d]= %7.3f %7.3fj \n", k,sumRe,sumIm );
}
}

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4. IMPULSE RESPONSE
4a) Aim: To compute Impulse Response of the system defined by
its difference equation of first order.

Program:
#include<stdio.h>

#include<math.h>
float y[8];
void main()

float x[8];
int num= ;
x[-1]=0;
y[-1]= ;
for(n=0;n<num;n++)

{
if(n==0)
x[n]=1; else
x[n]=0;

y[n]= ;

printf("y[%d]=%f\n",n,y[n]);

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4b) Aim: To compute Impulse Response of the system defined by

its difference equation of second order.

Program:
#include<stdio.h>

#include<math.h>

float y[8];

void main()

float x[8];

intnum= ;

x[-1]=0;

y[-1]= ;

y[-2]= ;

for(n=0;n<num;n++)

if (n==0)

x[n]=1;

else

x[n]=0;

y[n]= ;

printf("y[%d]=%f\n",n,y[n]);

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DSP LAB -17ECL57

5. FIR FILTER
Aim: Realization of an FIR filter (any type) to meet given
specifications.The input can be a signal from function generator /
speech signal.

Procedure:
(1) Connect CRO to the Socket provided for LINE OUTsocket.
(2) Connect a Signal generator to the LINE IN socket.
(3) Switch on the signal generator with a sine wave of frequency >20 Hz and
Vp-p=1v.
(4) Now switch on the DSK and bring up code composer studio on the PC.
(5) Create a new project with name Ex: FIR.pjt or IIR.pjt.
(6) From the File menu newDSP/BIOS Configurationselect
“dsk6713.cdb” and save it as “fir.cdb”
(7) Add “fir.cdb” to the current project.
(8) Add the given “CODEC.C” file to the current project which has the main
function and calls the other necessary routines.
(9) Add the library file “dsk6713bsl.lib” to the current project
path”c:\ccstudio\C6000\dsk6713\lib\dsk6713bsl.lib”
(10) Add new file to the project and save as fir.c and write the source code.
(11) Copy header files “dsk6713.h” and “dsk6713_aic23.h” from
C:\CCStudio_v3.1\c6000\dsk6713\include and paste it in the
Current project folder.
(12) Compile the program.
(13) Build the program.
(14) Load the generated object file (*.out) on to target board.
(15) Run the program
(16) Observe the waveform that appears on the CRO screen.
(17) Vary the frequency on function generator to see the response of filter.

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CODEC.C
#include "fircfg.h"

#include "c:\ccstudio_v3.1\c6000\dsk6713\include\dsk6713.h"

#include"c:\ccstudio_v3.1\c6000\dsk6713\include\dsk6713_aic23.h"

extern signed int FILTER(signed int x);

DSK6713_AIC23_Config config = {\

0x0017, /* 0 DSK6713_AIC23_LEFTINVOL Left line input channel volume */\

0x0017, /* 1 DSK6713_AIC23_RIGHTINVOL Right line input channel volume */\

0x00d8, /* 2 DSK6713_AIC23_LEFTHPVOL Left channel headphone volume */\

0x00d8, /* 3 DSK6713_AIC23_RIGHTHPVOL Right channel headphone volume */\

0x0011, /* 4 DSK6713_AIC23_ANAPATH Analog audio path control */\

0x0000, /* 5 DSK6713_AIC23_DIGPATH Digital audio path control */\

0x0000, /* 6 DSK6713_AIC23_POWERDOWN Power down control

*/\

0x0043, /* 7 DSK6713_AIC23_DIGIF Digital audio interface format */ \

0x0081, /* 8 DSK6713_AIC23_SAMPLERATE Sample rate control

*/\

0x0001 /* 9 DSK6713_AIC23_DIGACT Digital interface activation */ \

};

/*main () - Main code routine, initializes BSL and generates tone */

void main()

DSK6713_AIC23_CodecHandle hCodec;

Uint32 l_input, r_input,l_output, r_output;

/* Initialize the board support library, must be called first */

DSK6713_init();

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/* Start the codec */

hCodec = DSK6713_AIC23_openCodec(0, &config);

DSK6713_AIC23_setFreq(hCodec, 3);

while(1)

{ /* Read a sample to the left channel */

while (!DSK6713_AIC23_read(hCodec, &l_input));

/* Read a sample to the right channel */

while (!DSK6713_AIC23_read(hCodec, &r_input));

l_output=FILTER(l_input);

r_output=l_output;

/* Send a sample to the left channel */

while (!DSK6713_AIC23_write(hCodec, l_output));

/* Send a sample to the right channel */

while (!DSK6713_AIC23_write(hCodec, r_output));

/* Close the codec */

DSK6713_AIC23_closeCodec(hCodec);

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FIR FILTER

Calculation:

I. Clearly specify the filter specifications.

Eg: Order = 30;
Sampling Rate = 8000 samples/sec.
Cut off Freq. = 400 Hz.

II. Compute the cut-off frequency Wc.

Eg: Wc= 2* π * fc/ Fs
= 2* π * 400/8000
= 0.1*π

III. Compute the desired Impulse Response hd(n) using particular

Window.
Eg: b_rect1=fir1(order, Wc, boxcar(31));{For Low-pass Filter.}
Eg:b_rect1= fir1(order, Wc, 'high',boxcar(31));{For High-pass
Filter.}

IV. Convolve input sequence with truncated Impulse Response

x (n)*h (n).

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Program:
#define ORDER 30

float h[] ={0.0161,0.0208,0.0062,-0.0169,-0.0276,

-0.0132,0.0176,0.0383, 0.0256,-0.0181,

-0.0598,-0.0553,0.0184,0.1410, 0.2558,

0.3025, 0.2558,0.1410,0.0184,-0.0553,

-0.0598,-0.0181,0.0256,0.0383, 0.0176,

-0.0132,-0.0276,-0.0169,0.0062,0.0208,

0.0161};

static short in_buffer[100];

signedint FILTER(signed int x)

int i=0;

signed long output=0;

in_buffer[0] = x;

for(i=ORDER;i>0;i--)

in_buffer[i] = in_buffer[i-1];

for(i=0;i<ORDER+1;i++)

output = output + h[i] * in_buffer[i];

return(output);

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Result:
 The output plot is observed on the CRO.
 This behaves as a low-pass filter.
 See the effect of filtering by giving a sinusoidal input to DSP kit and
slowly varying its frequency.
 Observe the cutoff frequency of the filter.
 Change the filter co-efficients for different types of window & Cut-
off frequencies.
(Refer FIR filter design part from experiment-10 in part A-MATLAB)

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Additional Programs

1. Interference Suppression
Aim: To add noise above 3kHz and then remove; Interference
suppression using 400 Hz tone.
Procedure:

1. Generate a 400Hz tone corrupted by a 4kHz sinusoid for a 0.0052 second

duration.
2. Design a FIR LPF of order 30,and assign the coefficients to variable b[ ].
3. Set up the loop for the output variable (for the number of samples
required).
4. Assign sum=0.
5. Find the sum=b0x[n] + b1x[n-1] + …….. + b30x[n-30];inner loop.
6. Assign y[n]=sum, end of output loop.
7. Verify the results using matlab.
Program:
#include<stdio.h>
float x[101]={0, 0.6506,1.0764 ,1.1384,0.8365,0.3090,
-0.2197,-0.5253, -0.4693,-0.0520,0.5878,
1.2252, 1.6356, 1.6800, 1.3583, 0.8090,
0.2565,-0.0747,-0.0462, 0.3420, 0.9511,
1.5564, 1.9333, 1.9432, 1.5858, 1.0000,
0.4102, 0.0411, 0.0312, 0.3808, 0.9511,
1.5176, 1.8559, 1.8274, 1.4321, 0.8090,
0.1827,-0.2221,-0.2665, 0.0496, 0.5878,
1.1236, 1.4328, 1.3768, 0.9559, 0.3090,
-0.3391,-0.7637,-0.8257,-0.5250,-0.0000,
0.5250, 0.8257, 0.7637, 0.3391,-0.3090,
-0.9559,-1.3768,-1.4328,-1.1236,-0.5878,
-0.0496, 0.2665, 0.2221,-0.1827,-0.8090,
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-1.4321,-1.8274,-1.8559,-1.5176,-0.9511,
-0.3808,-0.0312,-0.0411,-0.4102,-1.0000,
-1.5858,-1.9432,-1.9333,-1.5564,-0.9511,
-0.3420, 0.0462, 0.0747,-0.2565,-0.8090,
-1.3583,-1.6800,-1.6356,-1.2252,-0.5878,
0.0520, 0.4693, 0.5253, 0.2197,-0.3090,
-0.8365,-1.1384,-1.0764,-0.6506,0.0000};
float h[31]={0.0048,0.0054,0.0072,0.0101,0.0141,
0.0188,0.0243,0.0301,0.0361,0.0419,
0.0474,0.0523,0.0563,0.0593,0.0612,
0.0618,0.0612,0.0593,0.0563,0.0523,
0.0474,0.0419,0.0361,0.0301,0.0243,
0.0188,0.0141,0.0101,0.0072,0.0054,
0.0048};
float y[101]={0.0};
void main()
{
int N=30,k=0,n=0;
float sum;
for (n=0;n<101;n++)
{
sum=0;
for(k=0;k<N;k++)
if(n-k>=0)
{
sum=sum+h[k]*x[n-k];
}
y[n]=sum;
}
for(n=0;n<101;n++)

printf("y[%d] = %f \n",n,y[n]);
}

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MATLAB program to generate the filter coefficients and to verify the

result:
clc;
clear all;
close all;
t=0:0.000025:0.0025;
x=sin(2*pi*400*t)+sin(2*pi*4000*t)
b=fir1(30,800/40000)
y=filter(b,1,x);
plot(t,x,'b',t,y,'r')
legend('Corrupted Signal', 'Original Signal');

Result:
The 4kHz interfering tone is successfully suppressed and is
observed in the plot and also the obtained result is verified in the
MATLAB.

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2. MATLAB program to verify the circular convolution property of

DFT

Circular convolution property: x(n)©y(n) X(k)Y(k)

% program to prove circular convolution property of DFT

clc;
clear all;
close all;
x1=input('enter the 1st input seq=')
x2=input('enter the 2nd input seq=')
N= input('enter the value for N=')
y1=cconv(x1,x2,N)
X1=fft(x1,N)
X2=fft(x2,N)
H=X1.*X2
y2=ifft(H,N)

if y1 == y2
disp('circular convolution property is proved');
else
disp('circular convolution property is not proved');

end

subplot(2,1,1)
stem(y1);
xlabel('N------>'),ylabel('amplitude')

subplot(2,1,2)
stem(y2);
xlabel('N------>'),ylabel('amplitude')

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REFERENCES
[1]. Digital signal processing using MATLAB –SanjeetMitra, TMH,2001.
[2]. Digital signal processing using MATLAB - J. G. Proakis&Ingale, MGH,
2000.
[3]. Digital Signal Processors, B. Venkataramani and Bhaskar, TMH,2002.
[4]. A.Gilat, MATLAB: An Introduction with Applications, John Wiley &
Sons, Inc. 2004.
[5]. R. Pratap, Getting Started with MATLAB 7: A Quick Introduction for
Scientists and Engineers, Oxford University Press, 2009.
[6]. D. Hanselman and B. Little_eld, Mastering MATLAB 5: A
Comprehensive Tutorial and Reference,Prentice Hall, 1998.
[7]. B. R. Hunt, R. L. Lipsman, and J. M. Rosenberg (with K. R. Coombes,
J. E. Osborn, and G. J.Stuck), A Guide to MATLAB: for beginners and
experienced users, Cambridge University Press 2001.
[8] RulphChassaing, “DSP Applications Using C and the TMS320C6X
DSK”, John Wiley, New York, 2002

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VIVA
1) What is DSP? Why do we need DSP?
2) List the merits and demerits of DSP
3) What are the applications of DSP?
4) What is sampling theorem?
5) What is aliasing effect?
6) Define Quantization.
7) Explain the principle of operation of analog to digital conversion of the
signal with a neat diagram.
8) Explain the significance of Nyquist rate and aliasing during the
sampling of continuous time signals.
9) What do you mean by process of reconstruction?
10) What are techniques of reconstructions?
11) What do you mean Aliasing? What is the condition to avoid aliasing
for sampling?
12) Write the conditions of sampling.How many types of sampling there?
13) Explain the statement: t= 0:0.000005:0.05
In the above example what does colon (: ) and semicolon (; ) denotes.
14) What is a) Undersampling b) nyquist plot c) Oversampling.
15) What is MATLAB? What are the applications of MATLAB?
16) What is the use of command ‗legend‘?
17) Distinguish between ‗plot‘ and ‗stem‘ functions.
18) Explain the function ‗subplot‘?
19) What is convolution? What are the properties of convolution?
20) What is linear convolution?
21) Distinguish between Linear convolution and circular convolution.
22) How will you obtain linear convolution from circular convolution?
23) Explain how convolution syntax built in function works.
24) What is the total output length of linear convolution sum?
25) What is an LTI system?

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26) Describe impulse response of a function.

27) What is the difference between convolution and filter?
28) Where to use command ‗filter‘ or ‗impz‘, and what is the difference
between these two?
29) What is the use of function command ‗deconv‘?
30) What is the difference between linear and circular convolution?
31) What do you mean by statement subplot (3,3,1).
32) What do you mean by command ―mod‖ and where it is used?
33) What do you mean by Autocorrelation and Crosscorrelation
sequences?
34) What is the difference between Autocorrelatio and Crosscorrelation?
35) List all the properties of autocorrelation and Crosscorrelaion
sequence.
36) Where we use the inbuilt function ‗xcorr‘ and what is the purpose of
using this function?
37) How to calculate output of DFT using MATLAB?
38) Distinguish between DFT and DTFT.
39) Where DFT is used?
40) What is the difference between DFT and IDFT?
41) What do you mean by built in function ‗abs‘ and where it is used?
42) What do you mean by phase spectrum and magnitude spectrum/give
comparison.
43) How to compute maximum length N for a circular convolution using
DFT and IDFT. (What is the command?).
44) Explain the statement- y=x1.*x2
45) What is FFT?
46) Why FFT is needed?
47) What is meant by Radix-2 FFT?
48) What is decimation-in-time algorithm?
49) What is decimation in frequency algorithm?
50) What are the differences and similarities between DIF and DIT
algorithm?

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51) How many stages are there for 8-point DFT?

52) How many multiplication terms are required for doing DFT by
expressional method and FFT method?
53) What is the main advantage of FFT?
54) What are the applications of FFT algorithms?
55) What do you mean by ‗filtic‘ command, explain.
56) How to calculate output length of the linear and circular convolution?
57) What do you mean by built in function ‗fliplr‘ and where we need to
use this?
58) What is steady state response?
59) Which built in function is used to solve a given difference equation?
60) Explain the concept of difference equation.
61) What is filter?
62) What are the types of digital filter according to their impulse
response?
63) What is FIR Filter?
64) Write the procedure for designing FIR filters
65) Write the characteristics of FIR filter.
66) What are the advantages and disadvantages and applications of FIR
FILTER?
67) What is GIBBS phenomenon?
68) Write the desirable characteristics of frequency response of window
functions.
69) Write the characteristics features of rectangular window.
70) List merits and demerits of rectangular window.
71) What do you understand by linear phase response?
72) What are the advantages of Kaiser Window?
73) What is the need for employing window technique for FIR filter design?
74) What is windowing and why it is necessary?
75) What is the necessary and sufficient condition for linear phase
characteristic in FIR filter?
76) Define IIR filter.

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77) What are the methods available for designing analog IIR filter?
78) Mention the importance of IIR filter.
79) What is warping effect?
80) Compare analog and digital filter.
81) What is FIR and IIR filter define, and distinguish between these two.
82) Explain the command – N=ceil(6.6*pi/tb)
83) What is the matlab command for Hamming window? Explain.
84) What do you mean by cut-off frequency?
85) What are the differences between FIR and IIR filters?
86) What are the differences between Butterworth and Chebyshevfilters?
87) What do you mean by command ‗butter‘ and ‗cheby1‘?
88) Explain the command in detail-[N,wc]=buttord(2*fp/fs,2*fstp/fs,rp,As)
89) What are the Features of TMS320C6713 DSK Kit?
90) What are the Difference between the Microcontroller and the DSP
Processor?
91) What is CCS? Explain in detail to execute a program using CCS.
92) Why do we need of CCS?
93) How to execute a program using 'dsk' and 'simulator'?
94) Which IC is used in CCS? Explain the dsk, dsp kit.
95) What do you mean by noise?
96) Explain the program for linear convolution for your given sequence.
97) Why we are using command 'float' in CCS programs.
98) Where we use 'float' and where we use 'int'?
99) Explain the command: i=(n-k)%N
100) Explain the entire CCS program in step by step of execution.

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DIGITAL SIGNAL PROCESSING LABORATORY

Question Bank
A. Programs based on MATLAB implementation:
1) By using MATLAB implementation sample a band limited continuous time
signal band limited to frn = ………….Hz i) fs< 2fm ii) fs = 2fm iii) fs ≤2fm

2) Using MATLAB Compute the linear convolution of two given finite length of
sequences. Given: x(n)=[……………. ] and h(n)=[ ……………]

3) Using MATLAB Compute the circular convolution of two given finite length
of sequences. Given: x(n)=[ ………………..] and h(n)=[……………….]

4) Using MATLAB verify the associative, commutative and distributive

properties of convolution for the sequences. Given: x(n)=[………………] and
h(n)=[………………..]

5) Using MATLAB Compute the autocorrelation of a reference sequence and

verify its properties. Given: x(n)=[ …………………].

6) Using MATLAB Compute the cross correlation of two given finite length of
sequences and verify its properties. Given: Ref sequence x(n)=[……………]
and h(n)=[……………..]

7) Obtain the complete response of the system described by the following

difference equation …………………for the input x(n)=[ ……….] With initial
conditions …………………. Use MATLAB as a tool.

8) For a given sequence x(n)=[…………..].Compute the N-point DFT using

MATLAB. Plot magnitude and phase spectrum.(with equation method
and verify using built in function)

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9) Verify the DFT properties for given sequences. Given: x(n)=[………………]

and x2(n)=[………………..]
10) Compute the DFT for square pulse and sinc function.

11) Using MATLAB design a low-pass FIR filter with the following
Specifications:
a) Pass band cut-off frequency : ωP=__________rad
b) Stop band cut-off frequency : ωS=__________rad
c) Stop band attenuation : kS=__________dB
Plot the Frequency, phase and impulse response of the designed filter.

12) Using MATLAB, design a low-pass IIR Butterworth filter with the
following specifications.
kP = Pass band attenuation(dB) =
kS = Stop band attenuation( dB) =
fP = Pass band edge(Hz) =
fS = Stop band edge(Hz) =
1
FS = = Sampling rate (samples/sec) =
TS
Plot the Magnitude Frequency response of the low-pass Butterworth
filter.

13) Using MATLAB ,design alow-pass IIR Chebyshev type I filter with the
Following specifications.
kP = Pass band attenuation(dB) =
kS = Stop band attenuation( dB) =
fP = Pass band edge(Hz) =
fS = Stop band edge(Hz) =
1
FS = = Sampling rate (samples/sec) =
TS

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B. Programs based on DSP Processor TMS320C6713 (Code Composer

Studio):
1) Write a C program to compute linear convolution of two sequences for DSP
Processor (TMS320C671 3).Given: x (n) = […………] and h(n) = […………….]
Execute the program on Simulator/Emulator (DSP Starter kit - Board).

2) Write a C program to compute circular convolution of two sequences for

DSP Processor (TMS320C6713).Given: x(n)=[ ………………..] and
h(n)=[………….].
Execute the program on Simulator/Emulator (DSP Starter kit - Board).

3) Write a C program to find the impulse response of first and second order
system for DSP Processor (TMS320C6713). Assume zero initial conditions
for an initially relaxed system.
Given: First order equation:……………………….
Second order equation: ……………………
Execute the program on Simulator/Emulator (DSP Starter kit - Board).

4) Write a C program to compute N — Point DFT of a sequence for DSP

Processor (TMS320C6713).Given: x(n) = […………………………………….]
Execute the program on Simulator/Emulator (DSP Starter kit - Board).

5) Write a C program to realize the lowpass/highpass FIR filter to meet given

specifications. The input can be a signal from function generator / speech
signal for DSP Processor (TMS320C6713).
Given the the filter specifications.
 Order (N).
 Sampling Rate.
 Cut off Freq.

68 Dept Of ECE,VSMSRKIT