DSPLAB

LABORATORY MANUAL

DIGITAL SIGNAL PROCESSING

III B.TECH -II Semester (ECE)

A.Y 2014-2015

Prepared by

J.NARENDER
Asst Professor
ECE Dept

MARRI EDUCATIONAL SOCIETY’S GROUP OF INSTITUTIONS

MARRI LAXMAN REDDY INSTITUTE OF TECHNOLOGY & MANAGEMENT

(Approved by AICTE, New Delhi & Affiliated JNTU, Hyderabad

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MARRI EDUCATIONAL SOCIETY’S GROUP OF INSTITUTIONS

MARRI LAXMAN REDDY INSTITUTE OF TECHNOLOGY & MANAGEMENT
(Approved by AICTE, New Delhi & Affiliated JNTU, Hyderabad)
Dundigal, Quthbullapur (M), Hyderabad – 43

DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING

DIGITAL SIGNAL PROCESSING LAB

List of Experiments:

1. Generation of sinusoidal waveform /signal based on recursive difference equations.

2. To find DFT/IDFT of given DT signal.
3. To find frequency response of a given system (Transfer function /Differential equation
form).
4. Implementation of FFT of given sequence.
5. Determination of power spectrum of a given signal.
6. Implementation of LP FIR filter for a given sequence.
7. Implementation of HP FIR filter for a given sequence.
8. Implementation of LP IIR filter for a given sequence.
9. Implementation of HP IIR filter for a given sequence.
10. Generation of sinusoidal signal through filtering.
11. Generation of DTMF signals.
12. Implementation of Decimation process.
13. Implementation of Interpolation process.
14 Implementation of I/D sampling rate converters.
15. Removal of noise by autocorrelation / cross correlation.
16. Extraction of periodic signal masked by noise using correlation.
17. Verification of winer-khinchine relations.
18. Checking a random process for stationary in wide sense.

FG

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INSTRUCTIONS TO THE STUDENT FJHINSTRUCTIONS

TO THE STUDENT
S TO THE STUDENT
1. Students are required to attend all labs.

2. Students will work individually in hardware laboratories and in computer

laboratories.

3. While coming to the lab bring the lab manual cum observation book, record etc.

4. Take only the lab manual, calculator (if needed) and a pen or pencil to the work
area.

5. Before coming to the lab, prepare the pre-lab questions. Read through the lab
experiment to familiarize yourself with the components and assembly sequence.

6. Utilize 3 hours time properly to perform the experiment (both in software and
hardware) and note down the readings properly. Do the calculations, draw the
graph and take signature from the instructor.

7. If the experiment is not completed in the prescribed time, the pending work has to
be done in the leisure hour or extended hours.

8. You have to submit the completed record book according to the deadlines set up
by your instructor.

9. For practical subjects there shall be a continuous evaluation during the semester
for 25 sessional marks and 50 end examination marks.

10. Of the 25 marks for internal, 15 marks shall be awarded for day-to-day work and
10 marks to be awarded by conducting an internal laboratory test.

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INDEX

SL.NO. EXPERIMENT NAME PAGENO.

1 Generation of sinusoidal waveform /signal based on recursive difference 5

equations.

2 To find DFT/IDFT of given DT signal. 16

3 To find frequency response of a given system (Transfer function 22

/Differential equation form).
4 Implementation of FFT of given sequence. 26
.
5 Determination of power spectrum of a given signal. 29

6 Implementation of LP FIR filters for a given sequence. 32

7 Implementation of HP FIR filters for a given sequence. 38

8 Implementation of LP IIR filters for a given sequence. 44

9 Implementation of HP IIR filters for a given sequence. 48
10 Generation of sinusoidal signal through filtering. 52
11 Generation of DTMF signals. 56
.
12 Implementation of Decimation process. 62

13 Implementation of Interpolation process. 63

14 Implementation of I/D sampling rate converters. 69

15 Removal of noise by autocorrelation / cross correlation. 73
16 Extraction of periodic signal masked by noise using correlation. 75

17 Impulse response of first order and second order systems. 77

DSP Processor 80

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EXPERMENT NO-1

GENERATION OF BASIC SIGNALS USING MATLAB

AIM: -
To write a “MATLAB” Program to generate various signals such as unit impulse,
unit step, unit ramp, sinusoidal, exponential growing signal, exponential decaying signal,
cosine signal.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

One of the more useful functions in the study of linear systems is the "unit impulse
function." An ideal impulse function is a function that is zero everywhere but at the origin,
where it is infinitely high. However, the area of the impulse is finite. This is, at first hard to
visualize but we can do so by using the graphs shown below.

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Key Concept: Sifting Property of the Impulse

If b>a, then

Example: Another integral problem

Assume a<b, and evaluate the integral

Solution:

We now that the impulse is zero except at t=0 so

And

Unit Step Function

The unit step function and the impulse function are considered to be fundamental
functions in engineering, and it is strongly recommended that the reader becomes very
familiar with both of these functions.

The unit step function, also known as the Heaviside function, is defined as such:

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Sometimes, u(0) is given other values, usually either 0 or 1. For many applications, it is
irrelevant what the value at zero is. u(0) is generally written as undefined.

Derivative

The unit step function is level in all places except for a discontinuity at t = 0. For this
reason, the derivative of the unit step function is 0 at all points t, except where t = 0. Where t =
0, the derivative of the unit step function is infinite.

The derivative of a unit step function is called an impulse function. The impulse function will
be described in more detail next.

Integral

The integral of a unit step function is computed as such:

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Sinusoidal Signal Generation

The sine wave or sinusoid is a mathematical function that describes a smooth repetitive
oscillation. It occurs often in pure mathematics, as well as physics, signal processing,
electrical engineering and many other fields. Its most basic form as a function of time (t)

Where:
• A, the amplitude, is the peak deviation of the function from its center position.
• ω, the angular frequency, specifies how many oscillations occur in a unit time
interval, in radians per second
• φ, the phase, specifies where in its cycle the oscillation begins at t = 0.
A sampled sinusoid may be written as:

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-

%unit step signals%

clc;
clear all;
close all;
disp('unit step signals');
N=input('enter the no of samples');
x=ones(1,N);
stem(x);
xlabel('time');
ylabel('amplitude');
title('unit step sequence');

% sinusoidal signals%

clc;
clear all;
close all;
disp('sinusoidal signals');
N=input('enter the no of samples');

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n=0:1:N;
x=sin(n);
stem(x);
xlabel('time');
ylabel('amplitude');
title('sinusoidal sequence');

% unit ramp signals%

clc;
clear all;
close all;
disp('unit ramp signals');
N=input('enter the no of samples');
n=0:1:N;
x=n;
stem(x);
xlabel('time');
ylabel('amplitude');
title('unit ramp sequence');

% unit impulse signal %

clc;
clear all;
close all;
disp('unit impuse signal');
N=input ('enter the no of samples');
n=-N:1:N;
x=[zeros(1,N) 1 zeros(1,N)];
stem(n,x);
xlabel('time');
ylabel('amplitude');
tittle('impulse sequence');

% exponentialsignals%

clc;
clear all;
close all;
disp('exponential signals');
N=input('enter the no of samples');
n=0:1:N;
n=-N:1:N;
a=0.5;
x=a.^n;
stem(x);
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xlabel('time');
ylabel('amplitude');
title('exponential sequence');

OUTPUT:-

unit step signals

enter the no of samples6

unit step sequence

1

0.9

0.8

0.7

0.6
amplitude

0.5

0.4

0.3

0.2

0.1

0
1 1.5 2 2.5 3 3.5 4 4.5 5 5.5 6
time

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sinusoidal signals
enter the no of samples6

sinusoidal sequence
1

0.8

0.6

0.4

0.2
amplitude

-0.2

-0.4

-0.6

-0.8

-1
1 2 3 4 5 6 7
time

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unit ramp signals

enter the no of samples6

unit ramp sequence

6

4
amplitude

0
1 2 3 4 5 6 7
time

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unit impuse signal

enter the no of samples6

impulse sequence
1

0.9

0.8

0.7

0.6
amplitude

0.5

0.4

0.3

0.2

0.1

0
-6 -4 -2 0 2 4 6
time

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exponential signals
enter the no of samples6

exponential sequence
70

60

50

40
amplitude

30

20

10

0
0 2 4 6 8 10 12 14
time

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RESULT:-

Thus the MATLAB program for generation of all signals was performed and the
output was verified.

EXERCISE PROGRAM:-

1. Write program to get Discrete time Sinusoidal Signal?

2. Write program to get Fourier Transform of Sinusoidal Signal?

3. Write program to get Inverse Fourier Transform of Sinusoidal Signal?

4. Write Program for the following Function

Y=exp (-2*∏*f*t)+exp(-8*∏*f*)

Y= ((exp (-1.56∏f)*Sin(2∏f)+cos(2∏f)?
5. Write a matlab program for generating u(n)-u(n-1)?
6. Write program to get Discrete time co-Sinusoidal Signal?
7. Write program to get Discrete time saw tooth Signal?
8. Write program to get Discrete time triangular Signal?
9. Write program to get addition of two sinusoidal sequences?
10. Write program to get exponential sequence?

VIVA QUESTIONS:-

1. Define Signal?
2. Define determistic and Random Signal?
3. Define Delta Function?
4. What is Signal Modeling?
5. Define Periodic and a periodic Signal?

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EXPERMENT NO-2

DFT/IDFT OF A SEQUENCE WITHOUT USING THE INBUILT FUNCTIONS

AIM:-
To find the DFT& IDFT of a sequence without using the inbuilt functions.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

Given a sequence of N samples f(n), indexed by n = 0..N-1, the Discrete Fourier

Transform (DFT) is defined as F(k), where k=0..N-1:

F(k) are often called the 'Fourier Coefficients' or 'Harmonics'.

The sequence f(n) can be calculated from F(k) using the Inverse Discrete Fourier Transform
(IDFT):

In general, both f(n) and F(k) are complex.

The DFT is the most important discrete transform, used to perform Fourier analysis in
many practical applications.[1] In digital signal processing, the function is any quantity or
signal that varies over time, such as the pressure of a sound wave, a radio signal, or daily
temperature readings, sampled over a finite time interval (often defined by a window
function). In image processing, the samples can be the values of pixels along a row or column
of a raster image. The DFT is also used to efficiently solve partial differential equations, and
to perform other operations such as convolutions or multiplying large integers.

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PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-

%DFT%
clc;
clear all;
close all;
a=input ('enter the input sequence');
N=length(a);
disp('length of input sequence is ');
N
for k=1:N;
x(k)=0;
for i=1:N;
x(k)=x(k)+a(i)*exp((-j*pi*2/N)*((i-1)*(k-1)));
end;
end;
k=1:N;
disp('the output is');
x(k)
subplot(2,1,1);
stem(k,abs(x(k)));
grid;
xlabel ('discrete frequency');
ylabel('magnitude');
title('magnitude response of dft');
subplot(2,1,2);
stem(angle(x(k))*180/(pi));
grid;
xlabel('discrete frequency');
ylabel('phase angle');
title('phase response of dft');

%IDFT%

clc;
clear all;
close all;
a=input('enter the input sequence');

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disp('the length of input sequence is');

N=length(a);
N
for n=1:N;
x(n)=0;
for k=1:N;
x(n)=x(n)+a(k)*exp((j*pi*2*(n-1)*(k-1)/N));
end;
end;
n=1:N;
x=1/N*x(n);
disp('the output is');
x(n)
stem(n,abs(x));
grid;
xlabel('discrete time');
ylabel('magnitude');
title('magnitude response of the idft');
grid;

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OUTPUT:-

enter the input sequence[1 2 3 4]

length of input sequence is

N=

the output is

ans =

10.0000 -2.0000 + 2.0000i -2.0000 - 0.0000i -2.0000 - 2.0000i

magnitude response of dft

10
magnitude

0
1 1.5 2 2.5 3 3.5 4
discrete frequency
phase response of dft
200

100
phase angle

-100

-200
1 1.5 2 2.5 3 3.5 4
discrete frequency

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enter the input sequence[10 -2+2j -2 -2-2j]

the length of input sequence is

N=

the output is

ans =

1.0000 2.0000 + 0.0000i 3.0000 - 0.0000i 4.0000 - 0.0000i

magnitude response of the idft
4

3.5

2.5
magnitude

1.5

0.5

0
1 1.5 2 2.5 3 3.5 4
discrete time

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RESULT:-

DFT&IDFT of a given discrete time signal are executed using mat lab software.

EXERCISE PROGRAM:-

1. Write a matlab program to find the circular convolution of two

sequences?
2. Write a matlab program to find the circular convolution of
x1(n)={2,3,-1,2};x2(n)={-1,2,-1,2}?
3. Write a matlab program to find the circular convolution of
x1(n)={1,-1,2,3}; x2(n)={2,0,1,1}?
4. Write a matlab program to find the circular convolution of
x1(n)={1,1,-1,2}; x2(n)={0,1,2,3}?
5. Write a matlab program to find the DFT of x(n) ={1 1 1 1 0 0
0 0}?
6. Write a matlab program to find the DFT of x(n) ={1 2 1 2}?
7. Write a matlab program to find the DFT of x(n) ={1 0 -1 0}?
8. Write a matlab program to find the IDFT of X(k) ={1,1,-2j,-
1,1+2j }?
9. Write a matlab program to find the IDFT of X(k) ={1 0 1 0}?
10. Write a matlab program to find the to compare circular and
linear convolution of two sequences?

VIVA QUESTIONS:-

1. Define Symmetric and Anti-Symmetric Signals?

2. Define Continuous and Discrete Time Signals?
3. What are the Different types of representation of discrete time signals?
4. What are the Different types of Operation performed on signals?
5. Define DFT .How DFT can be calculated in matrix form?

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EXPERMENT NO-3

STUDY FREQUENCY RESPONSE OF SECOND ORDER SYSTEM

AIM: -
To study frequency response of second order system using MATLAB.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

Second order systems are the systems or networks which contain two or
more storage elements and have describing equations that are second order
differential equations.

The frequency response of second order filters is characterised by three

filter parameters: the gain k, the corner frequency and the quality factor Q.

A second order filter is a circuit that has a transfer function of the form:

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

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PROGRAM:-

%frequency response of differential equation %

clc ;
clear all;
b=[1,4];
a=[1,-5];
w=-2*pi:pi/8:2*pi;
[h]=freqz(b,a,w);
subplot(2,1,1);
stem(w,abs(h));
xlabel('freq/w');
ylabel('magnitude');
grid;
title('magnitude response of diffenrtial equataion');
subplot(2,1,2);
stem(w,angle(h));
xlabel('freq/w');
ylabel('phase in rad');
grid;
title('phase response of diffenrtial equataion');

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OUTPUT:-

magnitude response of diffenrtial equataion

1.5
magnitude

0.5

0
-8 -6 -4 -2 0 2 4 6 8
freq/w
phase response of diffenrtial equataion
4

2
phase in rad

-2

-4
-8 -6 -4 -2 0 2 4 6 8
freq/w

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RESULT:-

Hence the frequency response is executed by using MATLAB.

EXERCISE PROGRAM:-

1. Write a matlab program to find the frequency response of the

following difference equation y(n)-7y(n-1)+9y(n-2)=3x(n)-2x(n-1)?
2. Write a matlab program to find the frequency response of the
following difference equation 3y(n)+5y(n-1)=9x(n)?
3. Write a matlab program to find the frequency response of the
following difference equation 9 y(n)-2y(n-1)+7y(n-2)-3y(n-3)=6x(n)+x(n-1)?
4. Write a matlab program to find the frequency response of the
following difference equation 8y(n)+6y(n-1)=4x(n)+2x(n-1)?
5. Write a matlab program to find the frequency response of the
following difference equation 3y(n)-8y(n-1)+9y(n-2)=9x(n)+5x(n-1) ?
6. Write a matlab program to find the frequency response of the
following difference equation 6y(n)-5y(n-1)=9x(n)+5x(n-1) -7x(n-2)?
7. Write a matlab program to find the frequency response of the
following difference equation 9y(n)-8y(n-1)+2y(n-2)=9x(n)-3x(n-1) ?
8. Write a matlab program to find the frequency response of the
following difference equation 2y(n)-8y(n-1)=9x(n)+5x(n-1) ?
9. Write a matlab program to find the frequency response of the
following difference equation 9y(n)-8y(n-1)+9y(n-2)=9x(n)+5x(n-1)-x(n-2) ?
10. Write a matlab program to find the frequency response of the
following difference equation 3y(n)-8y(n-1)=7x(n)-3x(n-1) ?

VIVA QUESTIONS:-

1. What is the commend to find phase angle?

2. What is the commend to find frequency response?
3. What is transition band?
4. What is the formula for Z-transform?
5. What is the relationship b/w impulse response& frequency response?

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EXPERMENT NO-4

IMPLEMENTATION OF FFT OF GIVEN SEQUENCE

AIM: -

Implementation of FFT of given sequence.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

A fast Fourier transform (FFT) is an algorithm to compute the discrete Fourier

transform (DFT) and its inverse. Fourier analysis converts time (or space) to frequency and
vice versa; an FFT rapidly computes such transformations by factorizing the DFT matrix into
a product of sparse (mostly zero) factors.

PROCEDURE:-

 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-

%fft%
clc;
clear all;
close all;
xn=input('enter the input sequence');
N=input('enter the number of samples');
n=0:1:N-1;
xk=fft(xn,N);
k=0:1:N-1;
subplot(2,1,1);
stem(k,abs(xk));
xlabel('frq/w');

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ylabel('magnitude');
title('magnitude response of fft');
subplot(2,1,2);
stem(k,angle(xk));
xlabel('frq/w');
ylabel('phase');
title('phase response of fft');

OUTPUT:-

enter the input sequence[1 2 3 4]

enter the number of samples8
magnitude response of fft
10
magnitude

0
0 1 2 3 4 5 6 7
frq/w
phase response of fft
4

2
phase

-2

-4
0 1 2 3 4 5 6 7
frq/w

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RESULT:-

Hence the FFT of a given input sequence is performed & executed by using
MATLAB.

EXERCISE PROGRAM:-

1. Write a matlab program to find the cross correlation using FFT?

2. Write a matlab program to find the DFT of x(n) ={1 1 0 0 0 0 0 0}?
3. Write a matlab program to find the DFT of x(n) ={1 1 1 1 1 0 0 0}?
4. Write a matlab program to find the DFT of x(n) ={1 0 1 0 1 0 1 0}?
5. Write a matlab program to find the IDFT of X(k) ={1,1+j,-2j,1+2j ,-j, +j}?
6. Write a matlab program to find the IDFT of X(k) ={1,0,-2j,-1,+2j,-7j }?
7. Write a matlab program to find the IDFT of X(k) ={1,1,-2j,-1,1+2j }?
8. Write a matlab program to find the IDFT of X(k) ={1,1+j,-2j,-1+-j,1+2j }?
9. Write a matlab program to find the DFT of x(n) ={1 1 0 0 1 1 0 0}?
10. Write a matlab program to find the DFT of x(n) ={1 0 0 0 1 0 0 0}?

VIVA QUESTIONS:-

1. Whether linear convolution equation is a difference equation?

2. Whether DFT is a linear transform?
3. What is the difference between circular convolution & linear
convolution?
4. Can you implement linear convolution using circular convolution?
5. How FFT algorithms are classified?

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EXPERMENT NO-5

DETERMINATION OF POWER SPECTRUM

AIM: -

To obtain power spectrum of given signal using MATLAB.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

The power spectrum of a time-series x(t) describes how the variance of the data x(t) is
distributed over the frequency components into which x(t) may be decomposed. This
distribution of the variance may be described either by a measure µ or by a statistical
cumulative distribution function S(f) = the power contributed by frequencies from 0 up to f.

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-

% power spectrum %
clc;
clear all;
close all;
f1=input('enter the first frequencey f1=');
f2=input('enter the second frequencey f2=');
fs=input('enter the sampling frequencey fs=');
t=0:1/fs:1;
x=2*sin(2*pi*f1*t)+3*sin(2*pi*f2*t)+rand(size(t));
psd1=abs(fft(x).^2);
subplot(2,1,1);
plot(t*fs,10*log(psd1));

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xlabel('frequency');
ylabel('magnitude');
title('psd using square magnitude method');
psd2=abs(fft(xcorr(x),length(t)));
subplot(2,1,2);
plot(t*fs,10*log(psd2));
xlabel('frequency');
ylabel('magnitude');
title('psd using auto corelation method');
OUTPUT:-

enter the first frequencey f1=200

enter the second frequencey f2=400
enter the sampling frequencey fs=1000

psd using square magnitude method

150
magnitude

100

50

0
0 100 200 300 400 500 600 700 800 900 1000
frequency
psd using auto corelation method
140
magnitude

120

100

80
0 100 200 300 400 500 600 700 800 900 1000
frequency

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RESULT:-

Hence the power spectral density is performed & executed by using MATLAB.

EXERCISE PROGRAM:-

1. Write a matlab program for power spectrum estimate using Welch

method?
2. Write a matlab program to plot the frequency response of a first order
system?
3. Write a matlab program to plot the frequency response of the system?
4. Write a matlab program to generate the periodic and aperiodic
sequences?
5. Write a matlab program to demonstrate the property of digital
frequency?
6. Write a matlab program to illustrate the concept of aliasing?
7. Write a matlab program to plot magnitude and phase response of first
order lowpass filter?
8. Write a matlab program to plot magnitude and phase response of first
order highpass filter?
9. Write a matlab program to plot magnitude and phase response of
second order bandpass filter?
10. Write a matlab program to plot magnitude and phase response of
second order bandstop filter?

VIVA QUESTIONS:-

1. Give the expressions for finding the Average power of a signal/sequence?

2. Give the expressions for finding the energy of a signal/sequence?
3. What is power spectrum?
4. Why there are two peaks in the magnitude spectrum of sine wave?
5. What is spectogram?

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EXPERMENT NO-6

IMPLEMENTATION OF LP FIR FILTERS

AIM: -

Implementation of Low Pass FIR filters for given sequence.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2.Windows XP SP2.

THEORY:-

A Finite Impulse Response (FIR) filter is a discrete linear time-invariant system

whose output is based on the weighted summation of a finite number of past inputs. An FIR
transversal filter structure can be obtained directly from the equation for discrete-time
convolution.

In this equation, x(k) and y(n) represent the input to and output from the filter at time n. h(n-
k) is the transversal filter coefficients at time n. These coefficients are generated by using
FDS (Filter Design Software or Digital filter design package).

FIR – filter is a finite impulse response filter. Order of the filter should be specified.
Infinite response is truncated to get finite impulse response. Placing a window of finite length
does this. Types of windows available are Rectangular, Barlett, Hamming, Hanning,
Blackmann window etc. This FIR filter is an all zero filter.

Dept ECE Page 32

 DSPLAB

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-

% LOW PASS FILTER %

clc;
clear all;
close all;
rp=input('enter the pass band ripple:rp=');
rs=input('enter the stop band ripple:rs=');
fp=input('enter the pass band freq :fp=');
fs=input('enter the stop band freq:fp=');
f=input('enter the sampling freq:f=');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(rp*rs))-13;
den=14.6*(fs-fp)/f;
n=ceil(num/den);
n1=n+1;
if(rem(n,2)~=0);
n1=n;
n=n-1;
end;
c=input('enter the type of window function 1.rectangular 2.trangular 3.kaiser:n=');
if(c==1);
y=rectwin(n1);
disp('rectangular window filter response');
end;
if(c==2);
y=triang(n1);
disp('triangular window filter response');
end;
if(c==3);
y=kaiser(n1);

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 DSPLAB

disp('kaiser window filter response');

end;
% low pass filter %
b=fir1(n,wp,y);
[n,o]=freqz(b,1,256);
m=20*log10(abs(n));
plot(o/pi,m);
xlabel('normalised freq output i');
ylabel('gain in db');
title('low pass filter');

OUTPUT:-

enter the pass band ripple:rp=0.02

enter the stop band ripple:rs=0.01
enter the pass band freq :fp=1000
enter the stop band freq:fp=1500
enter the sampling freq:f=10000
enter the type of window function 1.rectangular 2.trangular 3.kaiser:n=1
rectangular window filter response

low pass filter

10

-10

-20

-30
gain in db

-40

-50

-60

-70

-80

-90
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
normalised freq output i

Dept ECE Page 34

 DSPLAB

enter the pass band ripple:rp=0.02

enter the stop band ripple:rs=0.01
enter the pass band freq :fp=1000
enter the stop band freq:fp=1500
enter the sampling freq:f=10000
enter the type of window function 1.rectangular 2.trangular 3.kaiser:n=2
triangular window filter response

low pass filter

0

-5

-10

-15
gain in db

-20

-25

-30

-35

-40

-45
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
normalised freq output i

Dept ECE Page 35

 DSPLAB

enter the pass band ripple:rp=0.02

enter the stop band ripple:rs=0.01
enter the pass band freq :fp=1000
enter the stop band freq:fp=1500
enter the sampling freq:f=10000
enter the type of window function 1.rectangular 2.trangular 3.kaiser:n=3
kaiser window filter response

low pass filter

10

-10

-20

-30
gain in db

-40

-50

-60

-70

-80

-90
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
normalised freq output i

Dept ECE Page 36

 DSPLAB

RESULT:-

The implementation of Butterworth Low passes FIR Filters Completed.

EXERCISE PROGRAM:-

1. Write a matlab program to design FIR low pass filter using

hamming/hanning window?
2. Write a matlab program to design FIR low pass filter using hamming&
blackman window?
3. Write a matlab program to design FIR low pass filter with cutoff
frequency .5pi using frequency sampling method?
4. Write a matlab program to plot the frequency response of low pass
filter using Kaiser window for different values of beeta?
5. Write a matlab program to design FIR low pass filter for the given
specifications using Kaiser window?
6. Write a matlab program to design a 25-tap Hilbert transformer using
Bartlett and hamming windows and plot their frequency response?
7. Write a matlab program to design a 25-tap lowpass filter with cutoff
frequency .5pi radians using rectangular and hamming windows and plot their
frequency response?
8. Write a matlab program to design a 25-tap highpass filter with cutoff
frequency .5pi radians using rectangular and Blackman windows and plot their
frequency response?
9. Write a matlab program to design a 25-tap bandpass filter with cutoff
frequency .25pi & .75pi radians using rectangular and hamming windows and plot
their frequency response?
10. Write a matlab program to design a 25-tap bandstop filter with cutoff
frequency .25pi & .75pi radians using rectangular and hamming windows and plot
their frequency response?

VIVA QUESTIONS:-

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 DSPLAB

1. Give the expression for finding the magnitude & phase response of FIR filter?
2. Compare different windows & their characteristics?
3. How FIR filters are designed using rectangular window?
4. What is Gibbs phenomenon? How it can be reduced?
5. Compare FIR&IIR filters?

EXPERMENT NO-7

IMPLEMENTATION OF HP FIR FILTERS

AIM: -

Implementation of High Pass FIR filter for given sequence.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

A Finite Impulse Response (FIR) filter is a discrete linear time-invariant system whose
output is based on the weighted summation of a finite number of past inputs. An FIR
transversal filter structure can be obtained directly from the equation for discrete-time
convolution.

In this equation, x(k) and y(n) represent the input to and output from the filter at time n. h(n-
k) is the transversal filter coefficients at time n. These coefficients are generated by using
FDS (Filter Design Software or Digital filter design package).

FIR – filter is a finite impulse response filter. Order of the filter should be specified.
Infinite response is truncated to get finite impulse response. Placing a window of finite length
does this. Types of windows available are Rectangular, Barlett, Hamming, Hanning,

Dept ECE Page 38

 DSPLAB

Blackmann window etc. This FIR filter is an all zero filter.

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-

% high pass filter %

clc;
clear all;
close all;
rp=input('enter the pass band ripple:rp=');
rs=input('enter the stop band ripple:rs=');
fp=input('enter the pass band freq :fp=');
fs=input('enter the stop band freq:fp=');
f=input('enter the sampling freq:f=');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(rp*rs))-13;
den=14.6*(fs-fp)/f;
n=ceil(num/den);
n1=n+1;
if(rem(n,2)~=0);
n1=n;
n=n-1;
end;
c=input('enter the type of window function 1.rectangular 2.trangular 3.kaiser:n=');
if(c==1);
y=rectwin(n1);
disp('rectangular window filter response');
end;
if(c==2);
y=triang(n1);

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 DSPLAB

disp('triangular window filter response');

end;
if(c==3);
alpha=input('enter the alpha value=');
y=kaiser(n1,alpha);
disp('kaiser window filter response');
end;
% high pass filter %
b=fir1(n,wp,'high',y);
[n,o]=freqz(b,1,256);
m=20*log10(abs(n));
plot(o/pi,m);
xlabel('normalised freq output i');
ylabel('gain in db');
title('high pass filter');

OUTPUT:-

enter the pass band ripple:rp=0.02

enter the stop band ripple:rs=0.01
enter the pass band freq :fp=1000
enter the stop band freq:fp=1500
enter the sampling freq:f=10000
enter the type of window function 1.rectangular 2.trangular 3.kaiser:n=1
rectangular window filter response

Dept ECE Page 40

 DSPLAB

high pass filter

10

-10

-20
gain in db

-30

-40

-50

-60
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
normalised freq output i

enter the pass band ripple:rp=0.02

enter the stop band ripple:rs=0.01
enter the pass band freq :fp=1000
enter the stop band freq:fp=1500
enter the sampling freq:f=10000
enter the type of window function 1.rectangular 2.trangular 3.kaiser:n=2
triangular window filter response

Dept ECE Page 41

 DSPLAB

high pass filter

0

-5

-10
gain in db

-15

-20

-25

-30
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
normalised freq output i

enter the pass band ripple:rp=0.02

enter the stop band ripple:rs=0.01
enter the pass band freq :fp=1000
enter the stop band freq:fp=1500
enter the sampling freq:f=10000
enter the type of window function 1.rectangular 2.trangular 3.kaiser:n=3
enter the alpha value=5
Dept ECE Page 42
 DSPLAB

kaiser window filter response

high pass filter
10

-10

-20

-30
gain in db

-40

-50

-60

-70

-80

-90
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
normalised freq output i

RESULT:-

The implementation of Butterworth High passes FIR Filters Completed.

EXERCISE PROGRAM:-

Dept ECE Page 43

 DSPLAB

1. Write a matlab program to design FIR high pass filter using

hamming/hanning window?
2. Write a matlab program to design FIR high pass filter using
hamming& blackman window?
3. Write a matlab program to design FIR high pass filter with cutoff
frequency .5pi using frequency sampling method?
4. Write a matlab program to plot the frequency response of high pass
filter using Kaiser window for different values of beeta?
5. Write a matlab program to design FIR high pass filter for the given
specifications using Kaiser window?
6. Write a matlab program to design a 25-tap Hilbert transformer using
Bartlett and hamming windows and plot their frequency response?
7. Write a matlab program to design a 25-tap high pass filter with cutoff
frequency .5pi radians using rectangular and hamming windows and plot their
frequency response?
8. Write a matlab program to design a 25-tap high pass filter with cutoff
frequency .5pi radians using rectangular and Blackman windows and plot their
frequency response?
9. Write a matlab program to design a 25-tap band pass filter with cutoff
frequency .25pi & .75pi radians using rectangular and hamming windows and plot
their frequency response?
10. Write a matlab program to design a 25-tap bands top filter with cutoff
frequency .25pi & .75pi radians using rectangular and hamming windows and plot
their frequency response?

VIVA QUESTIONS:-

1. What is FIR filter?

2. State properties of FIR filters?
3. Why FIR filters are inherently stable?
4. How linear phase is achieved in FIR filters?
5. Given an approximate formula for order of FIR filter?

EXPERMENT NO-8

IMPLEMENTATION OF LP IIR FILTERS

Dept ECE Page 44

 DSPLAB

AIM: -

Implementation of Low Pass IIR filters for given sequence.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

Infinite impulse response (IIR) is a property applying to many linear time-invariant

systems. Common examples of linear time-invariant systems are most electronic and digital
filters. Systems with this property are known as IIR systems or IIR filters, and are
distinguished by having an impulse response which does not become exactly zero past a
certain point, but continues indefinitely. This is in contrast to a finite impulse response in
which the impulse response h(t) does become exactly zero at times t > T for some finite T,
thus being of finite duration.

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-

% IIR LOW PASS FILTER %

clc;
clear all;
close all;
rp=input('enter the pass band ripple:rp=');
rs=input('enter the stop band ripple:rs=');
fp=input('enter the pass band frequency:fp=');
fs=input('enter the stop band frequency:fs=');
f=input('enter the sampling frequency:f=');
wp=2*fp/f;
ws=2*fs/f;
[N,wc]=buttord(wp,ws,rp,rs,'s');
[b,a]=butter(N,wc,'low','s');
w=0:0.01:pi;[n,o]=freqz(b,1,256);
[n,omega]=freq(b,a,w);
m=20*log10(abs(n));
subplot(2,1,1);
Dept ECE Page 45
 DSPLAB

plot(omega/pi,m);
xlabel('normalised frequency 0/pi');
ylabel('gain frequency in db');
title('magnitude response');
subplot(2,1,2);
plot(angle,(n));
xlabel('normalised frequency ');
ylabel('phase');
title('phase response');

OUTPUT:-

enter the pass band ripple:rp=0.15

Dept ECE Page 46
 DSPLAB

enter the stop band ripple:rs=60

enter the pass band frequency:fp=1500
enter the stop band frequency:fs=3000
enter the sampling frequency:f=7000

magnitude response
100
gain frequency in db

-100

-200

-300
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
normalised frequency 0/pi
phase response
4

2
phase

-2

-4
0 50 100 150 200 250 300 350
normalised frequency

RESULT:-

Thus IIR lowpass filter is designed using MATLAB.

Dept ECE Page 47

 DSPLAB

EXERCISE PROGRAM:-

1. Write a matlab program to generate IIR chebyshev analog lowpass filter?

2. Write a matlab program to design a Butterworth lowpass filter for the specifications?
3. Write a matlab program to design a Butterworth bandpass filter for the specifications?
4. Write a matlab program to design a Butterworth highpass filter for the specifications?
5. Write a matlab program to design a Butterworth bandreject filter for the
specifications?
6. Write a matlab program to design a chebyshev -I lowpass filter for the specifications?
7. Write a matlab program to design a chebyshev -II lowpass filter for the specifications?
8. Write a matlab program to design a chebyshev -I bandpass filter for the specifications?
9. Write a matlab program to design a chebyshev -II bandpass filter for the
specifications?
10. Write a matlab program to design a chebyshev -I high pass filter for the specifications?

VIVA QUESTIONS:-

1. What is the difference b/w analog and digital filter?

2. State the advantages & disadvantages of digital filters?
3. What are the different types of digital filters?
4. What are the characteristics of Butterworth filters?
5. How the s-plane is mapped to z-plane in impulse invariant transformation?

EXPERMENT NO-9

IMPLEMENTATION OF HP IIR FILTERS

Dept ECE Page 48

 DSPLAB

AIM: -
To implement the analog & digital High Pass IIR filter.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

Infinite impulse response (IIR) is a property applying to many linear time-invariant

systems. Common examples of linear time-invariant systems are most electronic and digital
filters. Systems with this property are known as IIR systems or IIR filters, and are
distinguished by having an impulse response which does not become exactly zero past a
certain point, but continues indefinitely. This is in contrast to a finite impulse response in
which the impulse response h(t) does become exactly zero at times t > T for some finite T,
thus being of finite duration.

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-
clc;
clear all;
close all;
disp('enter the sepecifications of iir filter');
rp=input('enter the pass band ripple:rp=');
rs=input('enter the stop band ripple:rs=');
wp=input('enter the pass band freq:wp=');
ws=input('enter the stop band freq:ws=');
fs=input('enter the sampling freq fs=');
w1=2*wp/fs;
w2=2*ws/fs;
[N,wc]=buttord(w1,w2,rp,rs,'s');
disp('freq resp of iir high pass filter is:');
[b,a]=butter(N,wc,'high','s');
w=0:0.001:pi;
[n,omega]=freqs(b,a,w);
m=20*log10(abs(n));
subplot(2,1,1);

Dept ECE Page 49

 DSPLAB

plot(omega/pi,m);
xlabel('normalised freq');
ylabel('gain');
title('magnitude response');
subplot(2,1,2);
plot(angle(n));
xlabel('normalised freq');
ylabel('phase');
title('phase response');

OUTPUT:-

enter the sepecifications of iir filter

enter the pass band ripple:rp=0.15
Dept ECE Page 50
 DSPLAB

enter the stop band ripple:rs=60

enter the pass band freq:wp=1500
enter the stop band freq:ws=3000
enter the sampling freq fs=7000
freq resp of iir high pass filter is:

magnitude response
500

0
gain

-500

-1000
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
normalised freq
phase response
4

2
phase

-2

-4
0 500 1000 1500 2000 2500 3000 3500
normalised freq

RESULT:-

Thus IIR lowpass filter is designed using MATLAB.

Dept ECE Page 51

 DSPLAB

EXERCISE PROGRAM:-

1. Write a matlab program to generate IIR chebyshev analog highpass filter?

2. Write a matlab program to design a Butterworth highpass filter for the specifications?
3. Write a matlab program to design a Butterworth bandpass filter for the specifications?
4. Write a matlab program to design a Butterworth highpass filter for the specifications?
5. Write a matlab program to design a Butterworth bandreject filter for the
specifications?
6. Write a matlab program to design a chebyshev -I highpass filter for the specifications?
7. Write a matlab program to design a chebyshev -II highpass filter for the specifications?
8. Write a matlab program to design a chebyshev -I highpass filter for the specifications?
9. Write a matlab program to design a chebyshev -II highpass filter for the specifications?
10. Write a matlab program to design a chebyshev -I high pass filter for the specifications?

VIVA QUESTIONS:-

1. What are the steps in designing the IIR filters?

2. State the disadvantages of impulse invariant transformation?
3. Why impulse invariant transformation is not suitable for design of high pass filters?
4. What is frequency relationship for bilinear transformation?
5. What is the frequency relationship for bilinear transformation?

EXPERMENT NO-10

SINUSOIDAL SIGNAL THROUGH FILTERING

Dept ECE Page 52

 DSPLAB

AIM: -

Generation of Sine Wave & Illustration of the Sampling Process in the Time Domain.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

Sinusoidal Signal Generation

The sine wave or sinusoid is a mathematical function that describes a smooth repetitive
oscillation. It occurs often in pure mathematics, as well as physics, signal processing,
electrical engineering and many other fields. Its most basic form as a function of time (t)
where:
• A, the amplitude, is the peak deviation of the function from its center position.
• ω, the angular frequency, specifies how many oscillations occur in a unit time
interval, in radians per second
• φ, the phase, specifies where in its cycle the oscillation begins at t = 0.
A sampled sinusoid may be written as:

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-

% Generation of Sine Wave & Illustration of the Sampling Process in the Time Domain

Dept ECE Page 53

 DSPLAB

clc;
t=
0:0.0005:1; a
= 10
f = 13;
xa =
a*sin(2*pi*f*t);
subplot(2,1,1)
plot(t,xa);grid
xlabel('Time,
msec');
ylabel('Amplitude')
;
title('Continuous-time signal
x_{a}(t)'); axis([0 1 -10.2 10.2])
subplot(2,1,2
); T = 0.01;
n = 0:T:1;
xs =
a*sin(2*pi*f*n); k
= 0:length(n)-1;
stem(k,xs);
grid
xlabel('Time index n');
ylabel('Amplitude');
title('Discrete-time signal
x[n]'); axis([0 (length(n)-1) -
10.2 10.2])

OUTPUT:-

Dept ECE Page 54

 DSPLAB

RESULT:-

Sinusoidal signal is generated by using MATLAB.

Dept ECE Page 55

 DSPLAB

EXERCISE PROGRAM:-

1. Write program to get Discrete time Sinusoidal Signal?

2. Write program to get Fourier Transform of Sinusoidal Signal?
3. Write program to get Inverse Fourier Transform of Sinusoidal Signal?
4. Write a matlab program for generating u(n)-u(n-1)?
5. Write program to get Discrete time co-Sinusoidal Signal?
6. Write program to get Discrete time saw tooth Signal?
7. Write program to get Discrete time triangular Signal?
8. Write program to get addition of two sinusoidal sequences?
9. Write program to get exponential sequence?
10. Write program to get Discrete time Co-Sinusoidal Signal?

VIVA QUESTIONS:-

1. Define sinusoidal signal?

2. Define C.T.S?
3. Define D.T.S?
4. Compare C.T.S & D.T.S?
5. . Draw the C.T.S & D.T.S diagrams?

EXPERMENT NO-11

DTMF SIGNAL GENERATION

Dept ECE Page 56

 DSPLAB

AIM: -

The objective of this program is To Generate Dual Tone Multiple Frequency (DTMF)
Signals.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

Dual Tone Multiple Frequency (DTMF) Signals.

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-

% Dual Tone Multiple Frequency (DTMF) Signals.

clc;
clearall;
closeall;

number=input('enter a phone number with no

spaces','s'); %number=1;
fs=8192; % fs is the sampling Frequency
T=0.5; % T stores how for how long a tone will be played
x= 2*pi*[697 770 852 941];

y= 2*pi*[1209 1336 1477

1633]; t=[0:1/fs:T]'
tx=[sin(x(1)*t),sin(x(2)*t),sin(x(3)*t),sin(x(4)*t)]/2;

Dept ECE Page 57

 DSPLAB

ty=[sin(y(1)*t),sin(y(2)*t),sin(y(3)*t),sin(y(4)*t)
]/2; for k=1:length(number)

switch
number(k)
case '1'

tone =
tx(:,1)+ty(:,1);
sound(tone);
stem(tone);
case '2'

tone =
tx(:,1)+ty(:,2);
sound(tone);
stem(tone);

case '3'

tone =
tx(:,1)+ty(:,3);
sound(tone);
stem(tone);
case '4'

tone = tx(:,2)+ty(:,1); sound(tone); stem(tone); otherwise

disp('invalid number');
end
pause(2.70
)
end;

OUTPUT:-

Input: 01234

Dept ECE Page 58

 DSPLAB

Dept ECE Page 59

 DSPLAB

Dept ECE Page 60

 DSPLAB

RESULT:-

Dual Tone Multiple Frequency (DTMF) Signals are generated by using MAT LAB.
Dept ECE Page 61
 DSPLAB

EXERCISE PROGRAM:-

1. Write a matlab program to generate a sine wave with amplitude = 3, frequency 20Hz?
2. Write a matlab program to generate a cos wave with amplitude = 3, frequency 20Hz?
3. Write a matlab program to generate a triangular wave with amplitude = 8, frequency
10Hz?
4. Write a matlab program to generate a square wave with amplitude = 2, frequency
10kHz?
5. Write a matlab program to generate a sinc wave with amplitude = -8, frequency5Khz?
6. Write a matlab program to generate a sine wave with amplitude = 7, frequency 29Hz.
7. Write a matlab program to generate a cos wave with amplitude = 9, frequency 50Hz.
8. Write a matlab program to generate a triangular wave with amplitude = 24, frequency
100Hz.
9. Write a matlab program to generate a square wave with amplitude = 12, frequency
10kHz.
10. Write a matlab program to generate a sinc wave with amplitude = 5, frequency5Khz.

VIVA QUESTIONS:-

1. Define Signal?
2. Define determistic and Random Signal?
3. Define Delta Function?
4. What is Signal Modeling?
5. Define Periodic and a periodic Signal?

EXPERMENT NO-12

INTERPOLATION
Dept ECE Page 62
 DSPLAB

AIM: -
The objective of this program is To Perform up sampling on the Given Input Sequence.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

Up sampling is interpolation, applied in the context of digital signal processing and

sample rate conversion. When up sampling is performed on a sequence of samples of a
continuous function or signal, it produces an approximation of the sequence that would have
been obtained by sampling the signal at a higher rate (or density, as in the case of a
photograph). For example, if compact disc audio is up sampled by a factor of 5/4, the resulting
sample-rate increases from 44,100 Hz to 55,125 Hz.

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-

%interpolation%
clc;
clear all;
close all;
N=input('enter sample value');
n=0:N-1;
L=input('enter up sampling factor');
x=sin(2*pi*0.043*n)+sin(2*pi*0.031*n);
y=interp(x,L);
subplot(2,1,1);
stem(n,x(1:N));
xlabel('time');
ylabel('amp');
title('input sequence');
t=0:(N*L)-1;
subplot(2,1,2);
stem(t,y(1:N*L));
xlabel('time');

Dept ECE Page 63

 DSPLAB

ylabel('amp');
title('output sequence');

OUTPUT:-

enter sample value50

Dept ECE Page 64
 DSPLAB

enter up sampling factor

input sequence
2

1
amp

-1

-2
0 5 10 15 20 25 30 35 40 45 50
time
output sequence
2

1
amp

-1

-2
0 50 100 150 200 250
time

RESULT:-

This MATLAB program has been written to perform interpolation on the Given

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Input Sequence.

EXERCISE PROGRAM:-

1. Write a matlab program to illustrate the effect of anti-aliasing filter?

2. Write a matlab program to illustration of upsampling?
3. Write a matlab program to illustration of downsampling?
4. Write a matlab program to illustration of effect of upsampling in frequency
domain?
5. Write a matlab program to illustration of effect of downsampling in frequency
domain?
6. Write a matlab program to illustrate the concept of aliasing?
7. Write a matlab program to plot magnitude response of comb filter?
8. Write a matlab program to plot magnitude response of allpass filter?
9. Write a matlab program to plot magnitude & unwrapped phase response of two
transfer functins?
10. Write a matlab program to design a filter that eliminates high frequency
component in a CT signal?

VIVA QUESTIONS:-

1. How aliasing can be avoided?

2. Which type of interpolation is used to reconstruct the signal?
3. What is aliasing?
4. Define interpolation?
5. What is pre-alias filter?

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EXPERMENT NO-13

DECIMATION

AIM: -

The objective of this program is To Perform Decimation on the Given Input Sequence.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

In digital signal processing, decimation is the process of reducing the sampling rate of a
signal. Complementary to interpolation, which increases sampling rate, it is a specific case of sample
rate conversion in a multi-rate digital signal processing system. Decimation utilizes filtering to
mitigate aliasing distortion, which can occur when simply down sampling a signal. A system
component that performs decimation is called a decimator.

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window
PROGRAM:-

%decimation%
clc;
clear all;
close all;
N=input('enter sample value');
n=0:N-1;
m=input('enter down sampling factor');
x=sin(2*pi*0.043*n)+sin(2*pi*0.031*n);
y=decimate(x,m,'fir');
subplot(2,1,1);
stem(n,x(1:N));
xlabel('time');
ylabel('amp');
title('input sequence');
t=0:(N/m)-1;
subplot(2,1,2);
stem(t,y(1:N/m));
xlabel('time');
ylabel('amp');
title('output sequence');

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OUTPUT:-

enter sample value65

enter down sampling factor3

input sequence
2

1
amp

-1

-2
0 10 20 30 40 50 60 70
time
output sequence
2

1
amp

-1

-2
0 2 4 6 8 10 12 14 16 18 20
time

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RESULT:-

This MATLAB program has been written to perform Decimation on the Given
Input Sequence.

EXERCISE PROGRAM:-

1. Write a matlab program to illustrate the effect of anti-aliasing filter?

2. Write a matlab program to illustration of upsampling?
3. Write a matlab program to illustration of downsampling?
4. Write a matlab program to illustration of effect of upsampling in frequency domain?
5. Write a matlab program to illustration of effect of downsampling in frequency domain?
6. Write a matlab program to illustrate the concept of aliasing?
7. Write a matlab program to plot magnitude response of comb filter?
8. Write a matlab program to plot magnitude response of allpass filter?
9. Write a matlab program to plot magnitude & unwrapped phase response of two transfer
functins?
10. Write a matlab program to design a filter that eliminates high frequency component in a
CT signal?

VIVA QUESTIONS:-

1. Define decimation?
2. Define multi rate signal processing?
3. What are the effects of coefficient quantization in FIR filters?
4. What is quantization process?
5. What is transmultiplexer? What is its use?

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 DSPLAB

EXPERMENT NO-14

IMPLEMENTATION OF I/D SAMPLING RATE CONVERTERS

AIM: -

To study sampling rate conversion by a rational form using MATLAB.

SOFTWARE REQURIED:-

1. MATLAB R2010a.
2. Windows XP SP2.

THEORY:-

"Up sampling" is the process of inserting zero-valued samples between original samples to
increase the sampling rate. (This is called "zero-stuffing".) Up sampling adds to the original signal
undesired spectral images which are centered on multiples of the original sampling rate.

"Interpolation", in the DSP sense, is the process of up sampling followed by filtering. (The
filtering removes the undesired spectral images.) As a linear process, the DSP sense of interpolation is
somewhat different from the "math" sense of interpolation, but the result is conceptually similar: to
create "in-between" samples from the original samples. The result is as if you had just originally
sampled your signal at the higher rate.

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-

% interpolation/dismation sampling %

clc; clear all;close all;

N=input('enter the sample value');
n=0:N-1;
l=input('enter up sampling factor');
m=input('enter down sampling factor');
x=sin(2*pi*0.043*n)+sin(2*pi*0.03*n);
y=resample(x,l,m);
subplot(2,1,1);
stem(n,x(1:N));
xlabel('time');
ylabel('amplitude');
title('input sequence');
t=0:(N*l/m)-1;

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subplot(2,1,2);
stem(t,y(1:N*l/m));
xlabel('time');
ylabel('amplitude');
title('input sampling sequence');

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OUTPUT:-

enter the sample value30

enter up sampling factor7
enter down sampling factor2

input sequence
2

1
amplitude

-1

-2
0 5 10 15 20 25 30
time
input sampling sequence
2

1
amplitude

-1

-2
0 20 40 60 80 100 120
time

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RESULT:-

Thus sampling rate conversion by a rational form is performed using MATLAB.

EXERCISE PROGRAM:-

1. Write a matlab program to illustrate the effect of anti-aliasing filter?

2. Write a matlab program to illustration of upsampling?
3. Write a matlab program to illustration of downsampling?
4. Write a matlab program to illustration of effect of upsampling in frequency domain?
5. Write a matlab program to illustration of effect of downsampling in frequency domain?
6. Write a matlab program to illustrate the concept of aliasing?
7. Write a matlab program to plot magnitude response of comb filter?
8. Write a matlab program to plot magnitude response of allpass filter?
9. Write a matlab program to plot magnitude & unwrapped phase response of two transfer
functins?
10. Write a matlab program to design a filter that eliminates high
frequency component in a CT signal?

VIVA QUESTIONS:-

1. What is multi rate signal processing?

2. What is the need for anti-imaging filter after up sampling a signal?
3. What is the need for anti-imaging filter prior to down sampling?
4. Define down sampling?
5. What is meant by up sampling?

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 DSPLAB

EXPERMENT NO-15

AUDIO APPLICATIONS

AIM: -

To perform audio applications such as to plot a time & frequency display of microphone plus a
cosine using DSP.

EQUIPMENTS:-

1. Operating system-windows XP.

2. Software –cc studio 3.
3. Software-mat lab 6.5.
4. DSK 6713DSP Trainer kit.
5. USB cable.
6. Power Supply.

PROGRAM:-

Spectrogram-rdts-mtl-frequency analysis of signals using RDTX-MATLAB

#include”dsk6713-aic23.h” //code-DSK support file

Unit32fs=dsk6713-aic23freq-8Khz; //set sampling rate
#include<rtdh.h> //RTDX support file
#include<math.h>
#include”hamming.cof” //Hamming window coefficients
#define pts256 //# of points for FFT
#define pi3.1415926538979
typedef struct {flot real,imag;}COMPLEX;
Void FFT(COMPLEX*Y,int n); //FFT protype
float iobuffer{PTS],iobuffer 1[PTS],a[PTS]; //input and output buffer
float x[PTS]; // intermediate buffer
short I; //general purpose index variable
int j,k,l,curr_block=0; // index variables
short buffercount=0; //number of new samples in iobuffer
short flag=0; //set to 1 by ISR when buffer is full
COMPLEX w[PTS]; //twiddle constants stored in w
COMPLEX samples [PTS]; //primary working buffer
RTDX_ Create output Channel(ochan); //create output channel C6x->PC

Main()
{
For(i=0;i<PTS;i++) //set up twiddle constants in w
{
w[i].real=cos(2*PI*i/512.0); //Re component of twiddle constants
w[i].imagl=sin(2*PI*i/512.0); //Im component of twiddle constants
}
Comm._intr(); //init DSK,code,McBSP
While (!RTDX_is Outpu Enabled(&ochan)) //wait forPC to enable RTDX

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 DSPLAB

Puts(“\n\n Wating…”); //while waiting

for(1=0;1<256;I++)
a[I]=cos(2*3.14*1500*1/8000);
for (k=0;k<5000;k++) //infinite loop
{
While (flag==0); //wait until iobuffer is full
Flag=0; //reset flag
For(i=0;I<PTS;i++) //swap buffer
{iobuffer1[i]=iobuffer[i]+a[i];
Samples[i].real=h[i]*iobuffer1[i]; //multiply by Hamming window coeffs
iobuffer 1[i]=x[i]; // process frame to iobuffer
}
For(i=0;i<PTS;i++)
Samples[i].imag=0.0 //imag components=0
FFT(samples,PTS); //call C-coded FFT function
For(i=0;i<PTS;i++) //compute square of FFT magnitude
{
X[i]=(samples[i].real*samp0les[i].real
+samples[i].imag*samples[i].imag)/16;//FFT data scaling
}
RTDX_write(&ochan,x1,sizeof(x1)/2); //send 128 samples to PC
} //end of infinite loop
} // end of main
Interrupt void c_int11() //ISR
{
Output_ sample ((short)(iobuffer[buffercount])); //out fromiobuffer
Iobuffer[[buffercount++]=(short)(input_sample()); //input to iobuffer
If(buffercount>=PTS) //if iobuffer full
{
Buffercount=0; /reinit buffercount
Flag=1; //reset flag

RESULT:-

Thus audio application is performed & spectrogram of an input signal is plotted using
MATLAB.

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 DSPLAB

EXPERMENT NO-16

NOISE REMOVAL

AIM: -

To add noise above 3kHz &then remove.

EQUIPMENTS:-

1. Operating system-windows XP.

2. Software –cc studio 3.
3. Software-mat lab 6.5.
4. DSK 6713DSP Trainer kit.
5. USB cable.
6. Power Supply.
7. CRO.
8. Function Generator.

THEORY:-

Adaptive filters are best used in cases where signal conditions or system parameters are slowly
Changing and the filter are to be adjusted to compensate for the change. The least mean squares
(LMS) criterion is such algorithm that can be used to provide the strategy for adjusting the filter
coefficient.
.
PROGRAM:-

#include “NCefg.h”
#include”dsk6713.h”
#include” dsk6713_aic23.h”
#define beta IE-13 //rate of convergence
#define N 30 //adaptive FIR filter length-vary this parameter & observe
float delay [N];
float w[N};
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_LEFTINVOL Left channel headphone volume*/ \
0x00d8, /*3 DSK6713_AIC23_RIGHTINVOL 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, /*7DSK6713_AIC23_DIGIF Digital audio interface format*/ \
0x0081, /*8DSK6713_AIC23_SAMPLERATE Sample rate control*/ \
0x0001, /*9DSK6713_AIC23_DIGACT Digital interface activation */ \
};
/*main()-Main code routine, Initializes BSL and generates tone*/

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Void main()
{
DSK6713_AIC23_codecHandle hCodec;
Int I_input,r_input,I_output,r_output,T;
/*Initialize the board support library,must be called first*/
DSK6713_init();
hCodec=DSK6713_AIC23_openCodec(0,&config); /*start the code C*/
DSK6713_AIC23_setfreq(hCOde c,1);
For(T=0:T<30:T++) //Initialize the adaptive FIR Coeffs=0
{ w[T]=0; //init buffer for weights
Delay[T]=0; ////init buffer for delay samples
}
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=(short int) adaptive_filter(l_input,r_input);
l_output=r_output;
While(!DSK6713_AIC23_write (hCodec.&l_output)); /*send output to the Left channel*/
While(!DSK6713_AIC23_write(hCodec.&r_output)); /*send output to the Right channel*/
}
\DSK6713_AIC23_closecodec(hcodec); /*close the codec*/
}

RESULT:-

Thus noise signal cancellation using adaptive filters is verified.

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 DSPLAB

EXPERMENT NO-17

IMPULSE RESPONSE

AIM: -

To find the impulse response of the given equation

y(n)-y(n-1+0.9y(n-2)=x(n)

SOFTWARE REQURIED:-

1.MATLAB R2010a.
2.Windows XP SP2.

THEORY:-

Second order systems are the systems or networks which contain two or more
storage elements and have describing equations that are second order differential
equations.

The frequency response of second order filters is characterised by three filter

parameters: the gain k, the corner frequency and the quality factor Q.

PROCEDURE:-
 Open MATLAB
 Open new M-file
 Type the program
 Save in current directory
 Compile and Run the program
 For the output see command window\ Figure window

PROGRAM:-

%To find the impulse response of discrete time system

%y(n)-y(n-1)+0.9y(n-2)=xn)%
clc;
clear all;
close all;
b=input('Enter the coefficeints of x(n):b=');
a=input('Enter the coefficeints of x(n):a=');
N=input('Enter the order of N =');
h=impz(b,a,N);
n=0:N-1;
subplot(2,1,1);
stem(n,h);
xlabel('discrete time');
ylabel('amplitude');
title('Impulse Response');
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 DSPLAB

subplot(2,1,2);
zplane(b,a);
xlabel('real axis');
ylabel('imaginary axis');
title('pole zero in z-plane');

OUTPUT:-

Enter the coefficeints of x(n):b=[1]

Enter the coefficeints of x(n):a=[1 -1 .9]
Enter the order of N =2

Impulse Response
1
amplitude

0.5

0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
discrete time
pole zero in z-plane
1
imaginary axis

0.5
2
0

-0.5

-1
-3 -2 -1 0 1 2 3
real axis

RESULT:-

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 DSPLAB

Hence the impulse response of the given system is performed.

EXERCISE PROGRAM:-

1. Write a matlab program to find the frequency response of the following

difference equation y(n)-7y(n-1)+9y(n-2)=3x(n)-2x(n-1)?
2. Write a matlab program to find the frequency response of the following
difference equation 3y(n)+5y(n-1)=9x(n)?
3. Write a matlab program to find the frequency response of the following
difference equation 9 y(n)-2y(n-1)+7y(n-2)-3y(n-3)=6x(n)+x(n-1)?
4. Write a matlab program to find the frequency response of the following
difference equation 8y(n)+6y(n-1)=4x(n)+2x(n-1)?
5. Write a matlab program to find the frequency response of the following
difference equation 3y(n)-8y(n-1)+9y(n-2)=9x(n)+5x(n-1) ?
6. Write a matlab program to find the frequency response of the following
difference equation 6y(n)-5y(n-1)=9x(n)+5x(n-1) -7x(n-2)?
7. Write a matlab program to find the frequency response of the following
difference equation 9y(n)-8y(n-1)+2y(n-2)=9x(n)-3x(n-1) ?
8. Write a matlab program to find the frequency response of the following
difference equation 2y(n)-8y(n-1)=9x(n)+5x(n-1) ?
9. Write a matlab program to find the frequency response of the following
difference equation 9y(n)-8y(n-1)+9y(n-2)=9x(n)+5x(n-1)-x(n-2) ?
10. Write a matlab program to find the frequency response of the following
difference equation 3y(n)-8y(n-1)=7x(n)-3x(n-1) ?

VIVA QUESTIONS:-

6. What is the commend to find phase angle?

7. What is the commend to find frequency response?
8. What is transition band?
9. What is the formula for Z-transform?
10. What is the relationship b/w impulse response& frequency response?

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 DSPLAB

1. INTRODUCTION TO DSP PROCESSORS

A signal can be defined as a function that conveys information, generally about the state or
behavior of a physical system. There are two basic types of signals viz Analog (continuous time
signals which are defined along a continuum of times) and Digital (discrete-time).

Remarkably, under reasonable constraints, a continuous time signal can be adequately

represented by samples, obtaining discrete time signals. Thus digital signal processing is an ideal
choice for anyone who needs the performance advantage of digital manipulation along with today‟s
analog reality.

Hence a processor which is designed to perform the special operations(digital

manipulations) on the digital signal within very less time can be called as a Digital signal processor.
The difference between a DSP processor, conventional microprocessor and a microcontroller are
listed below.

Microprocessor or General Purpose Processor such as Intel xx86 or Motorola 680xx

family Contains - only CPU
-No RAM
-No ROM -
No I/O ports
-No Timer
Microcontroller such as 8051 family
Contains - CPU
- RAM
- ROM
-I/O ports
- Timer &
- Interrupt circuitry
Some Micro Controllers also contain A/D, D/A and Flash Memory

DSP Processors such as Texas instruments and Analog Devices

Contains
- CPU - RAM - ROM - I/O ports - Timer
Optimized for
- Fast - Extended - Dual operand - Zero - Circular
arithmetic precision fetch overhead loop buffering

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The basic features of a DSP Processor are

Feature Use
Fast-Multiply accumulate Most DSP algorithms, including filtering,
transforms, etc. are multiplication- intensive

Multiple – access Many data-intensive DSP operations require reading a

program instruction and multiple data items during
memory each
architecture instruction cycle for best performance

Specialized addressing
modes Efficient handling of data arrays and first-in, first-out
buffers in memory
Specialized program
control Efficient control of loops for many iterative DSP
algorithms. Fast interrupt handling for frequent I/O
operations.
On-chip peripherals and On-chip peripherals like A/D converters allow for
I/O small
low cost system designs. Similarly I/O interfaces
interfaces tailored
for common peripherals allow clean interfaces to off-
chip
I/O devices.

ARCHITECTUREOF6713 DSPPROCESSOR

This chapter provides an overview of the architectural structure of the TMS320C67xx DSP,
which comprises the central processing unit (CPU), memory, and on-chip peripherals. The C67xE
DSPs use an advanced modified Harvard architecture that maximizes processing power with eight
buses. Separate program and data spaces allow simultaneous access to program instructions and
data, providing a high degree of parallelism. For example, three reads and one write can be
performed in a single cycle. Instructions with parallel store and application-specific instructions
fully utilize this architecture. In addition, data can be transferred between data and program spaces.
Such Parallelism supports a powerful set of arithmetic, logic, and bit-manipulation operations that
can all be performed in a single machine cycle. Also, the C67xx DSP includes the control
mechanisms to manage interrupts, repeated operations, and function calling.

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 DSPLAB

Bus Structure

The C67xx DSP architecture is built around eight major 16-bit buses (four program/data
buses and four address buses):
_ The program bus (PB) carries the instruction code and immediate operands from program
memory.
_ Three data buses (CB, DB, and EB) interconnect to various elements, such as the CPU, data
address generation logic, program address generation logic, on-chip peripherals, and data memory.
_ The CB and DB carry the operands that are read from data memory.
_ The EB carries the data to be written to memory.
_ Four address buses (PAB, CAB, DAB, and EAB) carry the addresses needed for instruction
execution.

The C67xx DSP can generate up to two data-memory addresses per cycle using the two
auxiliary register arithmetic units (ARAU0 and ARAU1). The PB can carry data operands stored in
program space (for instance, a coefficient table) to the multiplier and adder for multiply/accumulate
operations or to a destination in data space for data move instructions (MVPD and READA). This
capability, in conjunction with the feature of dual-operand read, supports the execution of single-
cycle, 3-operand instructions such as the FIRS instruction. The C67xx DSP also has an on-chip
bidirectional bus for accessing on-chip peripherals. This bus is connected to DB and EB through the
bus exchanger in the CPU interface. Accesses that use this bus can require two or more cycles for
reads and writes, depending on the peripheral’s structure.

Central Processing Unit (CPU)

The CPU is common to all C67xE devices. The C67x CPU contains:

_ 40-bit arithmetic logic unit

(ALU) _ Two 40-bit accumulators
_ Barrel shifter
_ 17 × 17-bit multiplier
_ 40-bit adder
_ Compare, select, and store unit
(CSSU) _ Data address generation unit
_ Program address generation unit

Arithmetic Logic Unit (ALU)

The C67x DSP performs 2s-complement arithmetic with a 40-bit arithmetic logic unit
(ALU) and two 40-bit accumulators (accumulators A and B). The ALU can also perform Boolean
operations. The ALU uses these inputs:

_ 16-bit immediate value

_ 16-bit word from data memory
_ 16-bit value in the temporary register,
_ Two 16-bit words from data memory
_ 32-bit word from data memory
_ 40-bit word from either accumulator

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 DSPLAB

Accumulators
Accumulators A and B store the output from the ALU or the multiplier/adder block. They
can also provide a second input to the ALU; accumulator A can be an input to the multiplier/adder.
Each accumulator is divided into three parts:
Guard bits (bits 39–32) High-
order word (bits 31–16) Low-
order word (bits 15–0)
Instructions are provided for storing the guard bits, for storing the high- and the low- order
accumulator words in data memory, and for transferring 32-bit accumulator words in or out of data
memory. Also, either of the accumulators can be used as temporary storage for the other.

Barrel Shifter

The C67x DSP barrel shifter has a 40-bit input connected to the accumulators or to data
memory (using CB or DB), and a 40-bit output connected to the ALU or to data memory (using
EB). The barrel shifter can produce a left shift of 0 to 31 bits and a right shift of 0 to 16 bits on the
input data. The shift requirements are defined in the shift count field of the instruction, the shift
count field (ASM) of status register ST1, or in temporary register T (when it is designated as a shift
count register).The barrel shifter and the exponent encoder normalize the values in an accumulator
in a single cycle. The LSBs of the output are filled with 0s, and the MSBs can be either zero filled
or sign extended, depending on the state of the sign-extension mode bit (SXM) in ST1. Additional
shift capabilities enable the processor to perform numerical scaling, bit extraction, extended
arithmetic, and overflow prevention operations.
Multiplier/Adder Unit

The multiplier/adder unit performs 17 _ 17-bit 2s-complement multiplication with a 40- bit
addition in a single instruction cycle. The multiplier/adder block consists of several elements: a
multiplier, an adder, signed/unsigned input control logic, fractional control logic, a zero detector, a
rounder (2s complement), overflow/saturation logic, and a 16-bit temporary storage register (T).

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 DSPLAB

2. VERIFY LINEAR CONVOLUTION

AIM: -

To perform linear convolution of two sequences

EQUIPMENT REQUIRED:-

Operating System – Windows XP

Constructor - Simulator
Software - CCStudio 3.3 & MATLAB 7.5

THEORY:-

Convolution is a formal mathematical operation, just as multiplication, addition, and

integration. Addition takes two numbers and produces a third number, while convolution takes two
signals and produces a third signal. Convolution is used in the mathematics of many fields, such as
probability and statistics. In linear systems, convolution is used to describe the relationship between
three signals of interest: the input signal, the impulse response, and the output signal.

In this equation, x1(k), x2(n-k) and y(n) represent the input to and output from the system at
time n. Here we could see that one of the input is shifted in time by a value every time it is
multiplied with the other input signal. Linear Convolution is quite often used as a method of
implementing filters of various types.

PROCEDURE:
1) Generate the first input sequence ‘x’.
2) Plot the sequence in discrete form. [Make use of stem( )]
3) Give some relevant names to x-axis and y-axis.
4) Generate second input sequence ‘y’.
5) Repeat steps (2) and (3) for second sequence ‘y’.
6) Use the inbuilt function ‘conv( )’ to compute linear convolution of ‘x’ and ‘y’.
z = conv(x,y)
7) Plot the output sequence ‘z’. [Repeat steps 2 and 3]
8) Make use of subplot ( ) to plot the inputs and output sequences in a single window.

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 DSPLAB

PROGRAM:

// Linear convolution program in c language using CCStudio

#include<stdio.h>
int x[15],h[15],y[15];
main()
{
int i,j,m,n;
printf("\n enter value for m");
scanf("%d",&m);
printf("\n enter value for n");
scanf("%d",&n);
printf("Enter values for i/p x(n):\n");
for(i=0;i<m;i++)
scanf("%d",&x[i]);
printf("Enter Values for i/p h(n)
\n"); for(i=0;i<n; i++)
scanf("%d",&h[i]);

// padding of zeros
for(i=m;i<=m+n-1;i++)
x[i]=0; for(i=n;i<=m+n-
1;i++)
h[i]=0;
/* convolution operation
*/ for(i=0;i<m+n-1;i++)
{
y[i]=0;
for(j=0;j<=i;j++)
{
y[i]=y[i]+(x[j]*h[i-j]);
}
}
//displaying the o/p
for(i=0;i<m+n-1;i++)
printf("\n The Value of output y[%d]=%d",i,y[i]);

Dept ECE Page 87

 DSPLAB

OUTPUT:

Enter value for m 4

Enter value for n 4
123
Enter values for i/p 4
123
Enter Values for n 4
The Value of output y[0]=1
The Value of output y[1]=4
The Value of output y[2]=10
The Value of output y[3]=20
The Value of output y[4]=25
The Value of output y[5]=24
The Value of output y[6]=16

Dept ECE Page 88

 DSPLAB

RESULT:-

linear convolution of two sequences using CCStudio 3.3 is obtained.

VIVA QUESTIONS:-

1. What is the purpose of using convolution?

2. Give the formula for calculating linear convolution?
3. What are the properties of convolution?
4. What is meant by discrete convolution?
5. Define linear system and give example?

Dept ECE Page 89

 DSPLAB

3. VERIFY CIRCULAR CONVOLUTION.

AIM:-
To perform circular convolution of two sequences using MATLAB & Code
Composer Studio3.1.

EQUIPMENT REQUIRED:-

Operating System – Windows XP

Constructor - Simulator
Software - CCStudio 3.3 & MATLAB 7.5

THEORY:-

Circular convolution is another way of finding the convolution sum of two input signals. It
resembles the linear convolution, except that the sample values of one of the input signals is folded
and right shifted before the convolution sum is found. Also note that circular convolution could
also be found by taking the DFT of the two input signals and finding the product of the two
frequency domain signals. The Inverse DFT of the product would give the output of the signal in
the time domain which is the circular convolution output. The two input signals could have been of
varying sample lengths. But we take the DFT of higher point, which ever signals levels to. For eg.
If one of the signal is of length 256 and the other spans 51 samples, then we could only take

256 point DFT. So the output of IDFT would be containing 256 samples instead of 306
samples, which follows N1+N2 – 1 where N1 & N2 are the lengths 256 and 51 respectively of the
two inputs. Thus the output which should have been 306 samples long is fitted into 256 samples.
The
256 points end up being a distorted version of the correct signal. This process is called circular
convolution.

PROCEDURE:-

1. Generate the first input sequence „x‟.

2. Plot the sequence in discrete form. [Make use of stem( )]
3. Give some relavent names to x-axis and y-axis.
4. Generate second input sequence „h‟.
5. Repeat steps (2) and (3) for second sequence „h‟.
6. Find out the length of first sequence and store it in variable „n1‟. [Make use of length ( )]
7. Similarly find out the length of second sequence and store it in variable „n2‟.
9. Make the length of smaller sequence equal to the larger sequence by padding zeros to it.
11. Display and plot the convolved sequence. [Repeat steps (2) and (3)]
12. Make use of subplot to plot the inputs and output sequences in a single window.
13. Compare theoretical and practical values.

Dept ECE Page 90

 DSPLAB

PROGRAM USING CODE COMPOSER STUDIO 3.3

/* program to implement
circular convolution */
#include<stdio.h>
int
m,n,x[30],h[30],y[30],i,j,
k,x2[30],a[30]; void
main()
{
printf(" Enter the length of the
first sequence\n");
scanf("%d",&m);
printf(" Enter the length of the second sequence\n");
scanf("%d",&n);
printf(" Enter the first sequence\n");
for(i=0;i<m;i++)
scanf("%d",&x[i]);
printf(" Enter the second sequence\n");
for(j=0;j<n;j++)
scanf("%d",&h[j]);

if(m-n!=0) /*If length of both sequences are not equal*/

{
if(m>n) /* Pad the smaller sequence with zero*/
{
for(i=n;i<m;i++)
h[i]=0;
n=m;
}
for(i=m;i<n;i++)
x[i]=0;
m=n;
}
y[0]=0;
a[0]=h[0];

Dept ECE Page 91

 DSPLAB

for(j=1;j<n;j++) /*folding h(n) to h(-n)*/

a[j]=h[n-j];
/*Circular convolution*/
for(i=0;i<n;i++)
y[0]+=x[i]*a[i];
for(k=1;k<n;k++)
{

y[k]=0
;
/*circular shift*/
for(j=1;j<n;j++)
x2[j]=a[j-1];

x2[0]=a[n-1];
for(i=0;i<n;i++)
{
a[i]=x2[i];
y[k]+=x[i]*x2[i];
}
} /*displaying the result*/
printf(" The circular convolution is\n");
for(i=0;i<n;i++)
printf("%d \t",y[i]);

Dept ECE Page 92

 DSPLAB

OUTPUT:-

Enter the length of the first sequence 4

Enter the length of the second sequence 3
Enter the first sequence 1234
Enter the second sequence 123
The circular convolution is 18 16 10 16

Dept ECE Page 93

 DSPLAB

RESULT:-

Circular Convolution of two sequences using CC Studio3.3 is obtained.

VIVA QUESTIONS:-

1. What is the different between Circular and Linear convolution?

2. What is the function in MATLAB used for padding zeros to a sequence? If your sequence
is, x = [1 2 3 4] and you want to pad zeros to it. How can you do that in MATLAB?
3. What is the use of following functions in MATLAB:
i. length( )
ii. max( )
iii. min( )
4. Give the steps to get the result of linear convolution from the method of
circular convolution?
5. What is the circular convolution?

Dept ECE Page 94

 DSPLAB

4. DESIGN FIR FILTER (LP/HP) USING WINDOWING TECHNIQUE

FIR FILTERS (LP/HP) USING WINDOWING TECHNIQUE

a) Using Rectangular Window b) Using Bartlett Window c) Using Kaiser Window

AIM:-
To design of FIR( LP/HP) filters using rectangular window.

EQUIPMENT REQUIRED:-

Operating System – Windows XP Constructor

- Simulator
Software - CCStudio 3 & MATLAB 7.5

THEORY:

A Finite Impulse Response (FIR) filter is a discrete linear time-invariant system whose
output is based on the weighted summation of a finite number of past inputs. An FIR transversal
filter structure can be obtained directly from the equation for discrete-time convolution.

In this equation, x(k) and y(n) represent the input to and output from the filter at time n.
h(n-k) is the transversal filter coefficients at time n. These coefficients are generated by using FDS
(Filter Design Software or Digital filter design package).
FIR – filter is a finite impulse response filter. Order of the filter should be specified. Infinite
response is truncated to get finite impulse response. placing a window of finite length does this.
Types of windows available are Rectangular, Barlett, Hamming, Hanning, Blackmann window etc.
This FIR filter is an all zero filter.

PROCEDURE:-

1. Enter the passband ripple (rp) and stopband ripple (rs).

2. Enter the passband frequency (fp) and stopband frequency (fs).
3. Enter the sampling frequency (f).
4. Calculate the analog passband edge frequency (wp) and stop band edge frequency (ws)
wp=2*fp/f ws=2*fs/f
5. Calculate the order of the filter using the following
formula, (-20log10 (rp.rs) –13)
n=
(14.6 (fs-fp)/f).

[Use „ceil( )‟ for rounding off the value of „n‟ to the nearest integer] if „n‟ is
an odd number, then reduce its value by „1‟.
6. Generate (n+1)th point window coefficients.For example boxcar(n+1) generates a rectangular
window. y=boxcar(n+1)
7. Design an nth order FIR filter using the previously generated (n+1) length window
function. b=fir1(n,wp,y)

Dept ECE Page 95

 DSPLAB

8. Find the frequency response of the filter by using „freqz( )‟ function. [h,o]=freqz(b,a,k) This
function returns k-point complex frequency response vector „h‟ and k-point frequency vector „o‟
in radians/samples of the filter.

H(eiw)= B(ejw) = b(1)+b(2)e-jw+…………..b(m+1)e-jmw

A(ejw) a(1)+a(2)e-jw+………….a(n+1)e-jnw
Where a, b are vectors containing the denominator and numerator
coefficients. Here a=1.
9. Calculate the magnitude of the frequency response in decibels (dB). m= 20*log10(abs(h))
10. Plot the magnitude response [magnitude in dB Vs normalized frequency (o/pi)]
11. Give relevant names to x- and y- axes and give an appropriate title for the plot.

PROGRAM USING CODE COMPOSER STUDIO 3.3:

#include<stdio.h>
#include<math.h>
#define pi 3.1415 int
n,N,c; float
wr[64],wt[64];
void main()
{
printf("\n enter no. of
samples,N= :");
scanf("%d",&N);

printf("\n enter choice of window function\n 1.rect \n 2. triang

\n c= :"); scanf("%d",&c);
printf("\n elements of window function
are:"); switch(c)
{
case 1:
for(n=0;n<=N-
1;n++)
{
wr[n]=1;
printf(" \n wr[%d]=%f",n,wr[n]);
}
break
; case
2:
for(n=0;n<=N-1;n++)
{
wt[n]=1-(2*(float)n/(N-1));
printf("\n
wt[%d]=%f",n,wt[n]);
}
break;
}
}

Dept ECE Page 96

 DSPLAB

RESULT: -

By using windowing techniques (Bartlett, Rectangular, Blackman) the filters are designed.

VIVA QUESTIONS:-

1. What are the uses of function ceil and for?

2. Define boxcar
3. Define Kaiser
4. Define Bartlett
5. What is an FIR system? Compare FIR and IIR system?

Dept ECE Page 97

 DSPLAB

5. IMPLEMENT IIR FILTER (LOW PASS & HIGH PASS)

IIR FILTER (LP/HP) USING BUTTER WORTH (ANLOG & DIGITAL) FILTERS &
CHEBYSHEV (DIGITAL) TYPE – I & II FILTERS

AIM:-
To design of Butterworth Digital (low pass & high pass) filter.

EQUIPMENTS:-

Operating System – Windows XP Constructor

- Simulator
Software - CCStudio 3 & MATLAB 7.5

THEORY:-

The IIR filter can realize both the poles and zeroes of a system because it has a rational transfer
function, described by polynomials in z in both the numerator and the denominator:

Eq.2
The difference equation for such a system is described by the following:

Eq .3
M and N are order of the two polynomials bk and ak are the filter coefficients. These
filter coefficients are generated using FDS (Filter Design software or Digital Filter design
package).

IIR filters can be expanded as infinite impulse response filters. In designing IIR filters, cutoff
frequencies of the filters should be mentioned. The order of the filter can be estimated using butter
worth polynomial. That’s why the filters are named as butter worth filters. Filter coefficients can be
found and the response can be plotted.

PROCEDURE:

1. Enter the pass band ripple (rp) and stop band ripple (rs).
2. Enter the pass band frequency (fp) and stop band frequency (fs).
3. Get the sampling frequency (f).
4. Calculate the analog pass band edge frequencies, w1 and w2.
w1 = 2*fp/f w2 = 2*fs/f
5. Calculate the order and 3dB cutoff frequency of the analog filter. [Make use of the following
function] [n,wn]=buttord(w1,w2,rp,rs,’s’)
6. Design an nth order analog lowpass Butter worth filter using the following
statement.
[b,a]=butter(n,wn,’s’)
7. Find the complex frequency response of the filter by using ‘freqs( )’
function

Dept ECE Page 98

 DSPLAB

PROGRAM USING CODE COMPOSER STUDIO 3.3:

//iirfilters
#include<stdio.h>
#include<math.h>
int
i,w,wc,c,N
; float
H[100];
float
mul(float,
int);
void main()
{
printf("\n
enter order of
filter ");
scanf("%d",&
N);
printf("\n enter
the cutoff freq
");
scanf("%d",&
wc);
printf("\n enter the choice for IIR
filter 1. LPF 2.HPF ");
scanf("%d",&c);
switch(c)
{
case 1:
for(w=0;w<100;w++)
{
H[w]=1/sqrt(1+mul((w/(float)wc),2*N));
printf("H[%d]=%f\n",w,H[w]);
}
break;

Dept ECE Page 99

 DSPLAB

case 2:
for(w=0;w<=100;w++)
{
H[w]=1/sqrt(1+mul((float)wc/w,2*N));
printf("H[%d]=%f\n",w,H[w]);
}
break;
}
}
float mul(float a,int x)
{
for(i=0;i<x-1;i++)
a*=a;
return(a);
}

OUTPUT:-

Dept ECE Page 100

MLRITM DSP LAB

RESULT: -
Thus IIR (LP/HP) filter is designed using low pass Butterworth and chebyshev filters
technique and verified using the DSP processor.
VIVA QUESTIONS:-
1. What are the properties of chebyshev filter?
2. Define signal flow graph?
3. Draw the signal flow graph of first order digital filter?
4. What is advantage of cascade realization?
5. What is the main disadvantage of direct-form realization?

Department of Electronics and Communication Engineering Page 101