SRM VALLIAMMAI ENGINEERING COLLEGE

(An Autonomous Institution)

SRM NAGAR, KATTANKULATHUR – 603 203.

DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

LABORATORY MANUAL

EC8562 – DIGITAL SIGNAL PROCESSING LABORATORY

Regulation - 2017

Semester/Branch : V semester ECE

Academic Year : 2020 -21 (ODD)
Prepared By : Dr. J. Mohan, Associate Professor
Dr. C. Amali, A.P (Sel.G)
Mr. A. Anbarasan, A.P (O.G)

SRM VEC/ECE/LM/EC8562/DSPL/2020-21/ODD 1
 SRM VALLIAMMAI ENGINEERING COLLEGE
(An Autonomous Institution)
SRM Nagar, Kattankulathur – 603 203

DEPARTMENT OF ELECTRONICS AND COMMUNICATION

ENGINEERING

VISION OF THE INSTITUTE

Educate to excel in social transformation

MISSION OF THE INSTITUTE

• To contribute to the development of human resources in the form of professional

engineers and managers of international excellence and competence with high
motivation and dynamism, who besides serving as ideal citizen of our country will
contribute substantially to the economic development and advancement in their chosen
areas of specialization.
• To build the institution with international repute in education in several areas at
several levels with specific emphasis to promote higher education and research through
strong institute-industry interaction and consultancy.

VISION OF THE DEPARTMENT

To excel in the field of electronics and communication engineering and to develop

highly competent technocrats with global intellectual qualities.

MISSION OF THE DEPARTMENT

M1: To educate the students with the state of art technologies to compete
internationally, able to produce creative solutions to the society`s needs,
conscious to the universal moral values, adherent to the professional ethical code
M2: To encourage the students for professional and software development career
M3: To equip the students with strong foundations to enable them for continuing
education and research.

SRM VEC/ECE/LM/EC8562/DSPL/2020-21/ODD 2
 SRM VALLIAMMAI ENGINEERING COLLEGE
(An Autonomous Institution)
SRM Nagar, Kattankulathur – 603 203

PROGRAM OUTCOMES

1. Engineering knowledge: Apply the knowledge of mathematics, science, engineering fundamentals, and
an engineering specialization to the solution of complex engineering problems.
2. Problem analysis: Identify, formulate, review research literature, and analyze complex engineering
problems reaching substantiated conclusions using first principles of mathematics, natural sciences, and
engineering sciences.
3. Design/development of solutions: Design solutions for complex engineering problems and design system
components or processes that meet the specified needs with appropriate consideration for the public health and
safety, and the cultural, societal, and environmental considerations.
4. Conduct investigations of complex problems: Use research-based knowledge and research methods including
design of experiments, analysis and interpretation of data, and synthesis of the information to provide valid
conclusions.
5. Modern tool usage: Create, select, and apply appropriate techniques, resources, and modern engineering and
IT tools including prediction and modelling to complex engineering activities with an understanding of the
limitations.
6. The engineer and society: Apply reasoning informed by the contextual knowledge to assess societal, health,
safety, legal and cultural issues and the consequent responsibilities relevant to the professional engineering
practice.
7. Environment and sustainability: Understand the impact of the professional engineering solutions in societal
and environmental contexts, and demonstrate the knowledge of, and need for sustainable development.
8. Ethics: Apply ethical principles and commit to professional ethics and responsibilities and norms of the
engineering practice.
9. Individual and team work: Function effectively as an individual, and as a member or leader in diverse teams,
and in multidisciplinary settings.
10. Communication: Communicate effectively on complex engineering activities with the engineering
community and with society at large, such as, being able to comprehend and write effective reports and design
documentation, make effective presentations, and give and receive clear instructions.
11. Project management and finance: Demonstrate knowledge and understanding of the engineering and
management principles and apply these to one‟s own work, as a member and leader in a team, to manage
projects and in multidisciplinary environments.
12. Life-long learning: Recognize the need for, and have the preparation and ability to engage in independent
and life-long learning in the broadest context of technological change.

PROGRAM SPECIFIC OUTCOME (PSOs)

PSO1: Ability to apply the acquired knowledge of basic skills, mathematical foundations, principles of
electronics, modelling and design of electronics based systems in solving engineering Problems.
PSO2: Ability to understand and analyze the interdisciplinary problems for developing innovative sustained
solutions with environmental concerns.
PSO3: Ability to update knowledge continuously in the tools like MATLAB, NS2, XILINIX and technologies
like VLSI, Embedded, Wireless Communications to meet the industry requirements.
PSO4: Ability to manage effectively as part of a team with professional behaviour and ethics.

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 SYLLABUS LTP
C004
2

EC8562 DIGITAL SIGNAL PROCESSING LABORATORY

OBJECTIVES:
The student should be made:
• To perform basic signal processing operations such as Linear Convolution, Circular Convolution,
Auto Correlation, Cross Correlation and Frequency analysis in MATLAB
• To implement FIR and IIR filters in MATLAB and DSP Processor
• To study the architecture of DSP processor
• To design a DSP system to demonstrate the Multi-rate and Adaptive signal processing concepts.

LIST OF EXPERIMENTS:

MATLAB / EQUIVALENT SOFTWARE PACKAGE

1. Generation of elementary Discrete-Time sequences
2. Linear and Circular convolutions
3. Auto correlation and Cross Correlation
4. Frequency Analysis using DFT
5. Design of FIR filters (LPF/HPF/BPF/BSF) and demonstrates the filtering operation
6. Design of Butterworth and Chebyshev IIR filters (LPF/HPF/BPF/BSF) and demonstrate the
filtering operations.

DSP PROCESSOR BASED IMPLEMENTATION

1. Study of architecture of Digital Signal Processor.
2. Perform MAC operation using various addressing modes.
3. Generation of various signals and random noise.
4. Design and demonstration of FIR Filter for Low pass, High pass, Band pass and
Band Stop Filtering.
5. Design and demonstration of Butter worth and Chebyshev IIR Filters for Low pass, High
Pass, Band pass and Band stop filtering
6. Implement an Up-sampling and Down-sampling operation in DSP Processor

TOTAL: 60 PERIODS
OUTCOMES:
At the end of the course, the student should be able to:
• Carryout basic signal processing operations
• Demonstrate their abilities towards MATLAB based implementation of various DSP systems
• Analyze the architecture of a DSP Processor
• Design and Implement the FIR and IIR Filters in DSP Processor for performing filtering
operation over real-time signals
• Design a DSP system for various applications of DSP

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 LIST OF EXPERIMENTS

CYCLE I

1 Generation of elementary Discrete-Time sequences

2 Linear and Circular Convolutions
3 Auto correlation and Cross Correlation
4 Frequency Analysis using DFT
5 Design of FIR filters (LPF/HPF/BPF/BSF) and demonstrates the filtering operation
6 Design of Butterworth and Chebyshev IIR filters (LPF/HPF/BPF/BSF) and demonstrate the filtering
operations
CYCLE II

7 Study of architecture of Digital Signal Processor.

8 MAC operation using various addressing modes.
9 Generation of various signals and random noise.
10 Design and demonstration of FIR Filter for Low pass, High pass, Band pass and Band stop Filtering.
11 Design and demonstration of Butter worth and Chebyshev IIR Filters for Low pass, High pass, Band
pass and Band stop filtering.
12 Implement an Up-sampling and Down-sampling operation in DSP Processor.
TOPIC BEYOND SYLLABUS

13 Study of Image processing tool box.

14 Generation of Dual Tone Multi Frequency Signals (DTMF) using DFT.

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Ex No: Date:

1. Generation of sequences (Functional & Random) & correlation

Aim:
To write a MATLAB program to generate unit step, unit impulse, unit ramp, exponential, sine
waveform, cosine waveform, random signal and correlation.

Prerequisite Lab Questions:

1. Define Continuous Time and Discrete Time Signal?

2. How to represent Unit step, impulse and ramp function?
3. Compare stem & Plot functions.
4. Define Energy & Power signals.
5. Give an example for real time signals.

Description:
Input : Enter the sequence length for the generation of signals.

(a) The unit sample sequence (Unit Impulse) is

δ (n) = 1 for n = 0
= 0 for n ≠ 0

(b) The unit step sequence is

u (n) = 1 for n ≥ 0
= 0 for n < 0
(c) The unit ramp sequence is denoted as ur (n) is
ur (n) = n for n ≥ 0
= 0 for n < 0
(d) The exponential signal is a sequence of the
form x(n) = en for all „n‟
(e) Sine signal x(t) = sinωt , for all t values.
(f) Sine sequensce x(n) = sinωn , for all n values.
(g) Cosine signal x(t) = cosωt , for all t values.
(h) Cosine signal x(t) = cosωn , for all n values.

Output: Plot of all the above mentioned signals.

Algorithm:

1. Start the program.

2. Generate unit step, unit impulse, unit ramp, exponential, sine waveform, cosine waveform,
random signal and correlation.
3. Execute the program.
4. Plot the output for each sequence.
5. Stop the program.

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 Program:

%Program for Unit Impulse Signal

clc;
clear all;
t=-3:1:3;
y=[zeros(1,3),ones(1,1),zeros(1,3)];
figure(1);
subplot(2,2,1);
stem(t,y);
ylabel('amplitude --->');
xlabel('t --->');
title('Unit impulse signal');

%Program for Unit Step Signal

t=0:5;
y=[ones(1,6)];
subplot(2,2,2);
stem(t,y);
ylabel('amplitude --->');
xlabel('t --->');
title('Unit Step Signal');

%Program for Ramp Signal

t=0:5;
y=t;
subplot(2,2,3);
stem(t,y);
ylabel('amplitude --->');
xlabel('t --->');
title('Ramp Signal');

%Program for Exponential Signal

t=0:0.1:5;
y=exp(t);
subplot(2,2,4);
stem(t,y);
ylabel('amplitude --->');
xlabel('t --->');
title('Exponential Signal');
figure(2);
%Program for Sine
waveform t=-pi:0.01:pi;
y=sin(t);
subplot(2,2,1);
plot(t,y);
ylabel('amplitude --->');
xlabel('t --->');
title('Sine waveform');
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 %Program for Cosine
waveform t=-pi:0.01:pi;
y=cos(t);
subplot(2,2,2);
plot(t,y);
ylabel('amplitude --->');
xlabel('t --->');
title('Cosine waveform');

%Program for Discrete Sine waveform

t=-pi:0.1:pi;
y=sin(t);
subplot(2,2,3);
stem(t,y);
ylabel('amplitude --->');
xlabel('t --->');
title('Discrete Sine waveform');

%Program for Discrete Cosine

waveform t=-pi:0.1:pi;
y=cos(t);
subplot(2,2,4);
stem(t,y);
ylabel('amplitude --->');
xlabel('t --->');
title('Discrete Cosine waveform');

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Model Graph:

Unit impulse signal Unit Step Signal

1 1

amplitude

amplitude
0.5
0.5

0
-4 -2 0 2 4 0
0 2 4 6
t --->
t --->
Ramp Signal Exponential Signal
6
150

4 100
amplitude

amplitude
2 50

0 0
0 2 4 6 0 2 4 6
t ---> t --->

Sine waveform Cosine waveform

1 1

0.5 0.5
amplitude

amplitude

0 0

-0.5 -0.5

-1
-4 -2 0 2 4 -1
-4 -2 0 2 4
t --->
t --->
Discrete Sine waveform
Discrete Cosine waveform
1
1

0.5
0.5
amplitude

amplitude

0
0

-0.5
-0.5

-1
-4 -2 0 2 4 -1
-4 -2 0 2 4
t --->
t --->

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 %Program for Random Signal
t = rand(1,10);
subplot(2,1,1);
plot(t);
ylabel('amplitude --->');
xlabel('t --->');
title('Random Signal');
disp('t=');
disp(t);
a=2;
b=3;
t1 = a + (b-a).*rand(100,1);
subplot(2,1,2);
plot(t1)
ylabel('amplitude --->');
xlabel('t --->');
title('Random Signal');
disp('t1=');
disp(t1);

Random Signal
1
amplitude

0.5

0
1 2 3 4 5 6 7 8 9 10
t --->
Random Signal
3
amplitude

2.5

2
0 10 20 30 40 50 60 70 80 90 100
t --->

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 % Program for computing cross-correlation of the sequences x[1 2 3 4] and h[4 3 2 1]
clc;
clear all;
close all;
x=input('enter the 1st sequence');
h=input('enter the 2nd sequence');
y= xcorr(x,h);
subplot(3,1,1);
stem(x);
ylabel('Amplitude --.');
xlabel('(a) n --.');
title('input sequence');
subplot(3,1,2);
stem(h);
ylabel('Amplitude --.');
xlabel('(b) n --.');
title('impulse sequence');
subplot(3,1,3);
stem(y);
ylabel('Amplitude --.');
xlabel('(c) n --.');
title('Cross correlated
sequence'); disp('The resultant
signal is y='); disp(y)

Output:
enter the 1st sequence[1 2 3 4]
enter the 2nd sequence[4 3 2 1]
The resultant signal is y=
1.0000 4.0000 10.0000 20.0000 25.0000 24.0000 16.0000

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Model Graph:

input sequence
4
Amplitude
2

0
1 1.5 2 2.5 3 3.5 4
(a) n --.
impulse sequence
4
Amplitude

0
1 1.5 2 2.5 3 3.5 4
(b) n --.
Cross correlated sequence
40
Amplitude

20

0
1 2 3 4 5 6 7
(c) n --.

%Program for computing autocorrelation function x[1 2 3 4]

x=input('enter the sequence');
y=xcorr(x,x);
subplot(2,1,1);
stem(x);
ylabel('Amplitude --.');
xlabel('(a) n --.');
title('original signal');
subplot(2,1,2);
stem(y);
ylabel('Amplitude --.');
xlabel('(a) n --.');
title ('Auto correlated
sequence'); disp('The resultant
signal is y='); disp(y)

Output:
enter the sequence[1 2 3 4]
The resultant signal is y=
4.0000 11.0000 20.0000 30.0000 20.0000 11.0000 4.0000

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Model Graph

original signal
4

Amplitude 3

0
1 1.5 2 2.5 3 3.5 4
(a) n --.
Auto correlated sequence
30

20
Amplitude

10

0
1 2 3 4 5 6 7
(a) n --.

Postlab Questions:

1. Modify Program (b) to generate a delayed unit step sequence u[n] with an advance of “n”
samples. Run the modified program and display the sequence generated.
2. What is the subplot command used to plot 4 figures in 2 rows and 2 columns?
3. How can you compute the average power of the generated sinusoidal sequence?
4. Write the syntax to determine auto correlation of the given signal x[n].
5. Give the examples for 1-D, 2-D & 3-D signals.

Result:
Thus a program to generate sequence for unit step, unit impulse, unit ramp, exponential, sine,
cosine and correlation signal using MATLAB was written and executed and the corresponding simulation
output is plotted.

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Ex No: Date:

2a. LINEAR CONVOLUTION OF TWO SEQUENCES

Aim:
To write a program to find out the linear convolution of two sequence using Matlab.

Prerequisite Lab Questions:

1. Define Convolution.
2. What is zero padding ?
3. List the properties of linear convolution.
4. What is the MATLAB function used to perform linear & circular convolution?
5. How will you determine the maximum length of the convolution of the sequences?
Description:

Input: Give the input x(n) and h(n) in matrix form.

Output: y(n), using the formula



y(n)   x(k )h(n  k)

k 

Algorithm:

1. Start the program.

2. Get the two input sequence x(n) and h(n).
3. Flip the h vector to the right or left direction.
4. Calculate the length of the input and output sequence.
5. Append zeros to the input to get the same length as output.
6. Sum the product of x(n) and shifted sequence of h(n-k).
7. Execute the program, display the result and verify theoretically.
8. Plot the graph for the input and output sequence.
9. Stop the pogram.

Program:
% Program for Linear Convolution
clc;
clear all;
close all;
%x=[1 1 1 1];
%h=[4 5 6 7];
x=input('enter the value x=');
h=input('enter the value h=');
subplot(3,1,1);
stem(x);
title('Input x(n)');
ylabel('x(n) --->');
xlabel('t --->');
subplot(3,1,2);
stem(h);
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 title('Input h(n)');
ylabel('h(n) --->');
xlabel('t --->');
h1=fliplr(h);
n1=length(x);
n2=length(h);
n=n1+n2-1;
x=[zeros(1,n2-1),x];
h1=[h1,zeros(1,n1-1)];
for i=1:n
new=[zeros(1,i-1),h1(1:n-i+1)];
out(i)=x*new';
end
disp('output=');
disp(out);
subplot(3,1,3);
t=0:1:n-1;
stem(t,out);
title('Output y(n)');
ylabel('y(n) --->');
xlabel('t --->');

MODEL GRAPH
Input x(n)
1

0.5
x(n)

0
1 1.5 2 2.5 3 3.5 4
t --->
Input h(n)
10

5
h(n)

0
1 1.5 2 2.5 3 3.5 4
t --->
Output y(n)

20
y(n)

10

0
0 1 2 3 4 5 6 7
t --->

THEORETICAL CALCULATION
∞

𝑦(𝑛) = ∑ 𝑥(𝑘)ℎ(𝑛 − 𝑘)
𝑘=−∞

Enter the value x = [1 1 1 1]

Enter the value h = [4 5 6 7]
Output = 4 9 15 22 18 13 7

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 EX NO: 2b CIRCULAR CONVOLUTION
Aim:
To write a program to find out the circular convolution of two sequences using MATLAB.

Description:
Input: Give the input x(n) and h(n) in matrix form.

Output: y(n), using the formula

y(n)  x(n)  h(n)

N 1
y(n) 
x(k)h(n  k)
k 0

Algorithm:
1. Start the program.
2. Get the two input sequence x(n) and h(n) and compute the length of the sequence.
3. Append zeros to x(n) or h(n) to make both sequence of same length.
4. Rotate the sequence h(-k), „n‟ times. If „n‟ is positive shift the sequence in anticlockwise direction
else in the clockwise direction.
5. Determine the product of sequence x(n) and shifted h(n) sequence and sum it.
6. Execute the program, display the result and verify it theoretically.
7. Plot the output graph for each sequence.
8. Stop the process.

Program:
% Program for Circular Convolution
clc;
clear all;
close all;
%g=[1,2,3,4];
%h=[2,4,6];
x=input('enter the
sequence'); h=input('enter
the sequence'); N1=length(x);
N2=length(h);
subplot(3,1,1);
n=0:1:N1-1;
stem(n,x);
title('x(n)');
ylabel('amplitude --->');
xlabel('time --->');
subplot(3,1,2);
n=0:1:N2-1;
stem(n,h);
title('h(n)');
ylabel('amplitude --->');

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 xlabel('time --->');
N=max(N1,N2);
if(N1<=N);
x1=[x,zeros(1,N-N1)];
end
if(N2<=N);
h1=[h,zeros(1,N-N2)];
end
for n=1:N
y(n)=0;
for i=1:N
j=n-i+1;
if(j<=0)
j=N+j;
end
y(n)=y(n)+x1(i)*h1(j);
end
end
disp('y=');
disp(y);
subplot(3,1,3);
n=0:1:N-1;
stem(n,y);
title('output y(n)');
ylabel('amplitude --->');
xlabel('time --->');

MODEL GRAPH
g(n)
4
amplitude

0
0 0.5 1 1.5 2 2.5 3
time --->
h(n)
10
amplitude

0
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
time --->
output y(n)
40
amplitude

20

0
0 0.5 1 1.5 2 2.5 3
time --->

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THEORITICAL CALCULATION

Using graphical or Matrix method

y(n)  x(n)  h(n)

N 1
y(n) 
x(k)h(n  k)
k 0

Enter the sequence g = [1,2,3,4]

Enter the sequence h = [2,4,6]

Output Sequence is = 36 32 20 32

Matrix Method:
x(0) x(N-1) x(n-2) ….. x(1) h(0) y(0)
x(1) x(0) x(N-1)… x(2) h(1) y(1)
x(2) x(1) x(0)… x(N-1) h(2) y(2)
. . .
. . = . . .
.
x(N-1) x(N-2) x(N-3).. x(0) h(N-1) y(N-1)

Postlab Questions :

1. What is the difference between Linear convolution & circular convolution?

2. State convolution property of a signal.
3. What are sectioned convolutions & what is the need of it?
4. Write the applications of convolution.
5. What is the relation between convolution in time domain and frequency domain ?

Result:

Thus a program to find out the linear and circular convolution of two sequences using Matlab was
written and executed.

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Ex No: Date:
3(a). SPECTRUM ANALYSIS USING DFT

Aim:
To analyze the amplitude and phase response of the DFT and IDFT signal using MATLAB.

Prerequisite Lab Questions:

1. Write the analysis and synthesis equation for DFT?

2. What do you mean by the property of shifting of DFT?
3. Find the value of twiddle factor for N=8, K=2 .
4. How many multiplications and additions are required to compute N point DFT ?
5. What is the relationship between DFT and Z-transform?

Algorithm:

1. Start the program.

2. Get the two input sequence x (n) and type of the DFT.
3. Compute the Discrete Fourier Transform (DFT) of the given sequence.
4. Find out the magnitude and phase response of the given sequence.
5. Execute the program, display the result and verify it.
6. Plot the output graph for magnitude and phase.
7. Compute the Inverse Discrete Fourier Transform (IDFT) of the given sequences.
8. Display the results verify it and plot the result.
9. Stop the process.

Program:
% Program for Discrete Fourier Transform
clc;
clear all;
close all;
%x=[1 2 3 4 5 6 7 8];
%N=[8];
x=input('enter the input x=');
N=input('enter the order of dft
N='); k=0:1:N-1;
disp('Input Sequence x=');
disp(x);
XK=fft(x,N);
disp('XK=');
disp(XK);
R=real(XK);
disp('Real part=');
disp(R);
I=imag(XK);
disp('Imaginary part=');
disp(I);
mag=abs(XK);
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 disp('Magnitude=');
disp(mag);
ang=angle(XK);
disp('Angle=');
disp(ang);
xn=ifft(XK,N);
disp('IDFT xn=');
disp(xn);

figure(1);
subplot(2,1,1);
stem(x);
xlabel('time -->');
ylabel('Amplitude -->');
title('Given sequence');

subplot(2,1,2);
stem(R,'r*');
hold on;
stem(I);
legend('Real Part','Imaginary Part');
xlabel('time -->');
ylabel('Amplitude -->');
title('Discrete Fourier Transform')

figure(3);
subplot(2,2,1);
stem(k,mag);
xlabel('time -->');
ylabel('x(k) --->');
title('Magnitude response');

subplot(2,2,2);
stem(k,ang);
xlabel('time -->');
ylabel('ang x(k) --->');
title('Phase response');

subplot(2,2,3);
stem(k,xn);
xlabel('time -->');
ylabel('x(n)');
title('Inverse Discrete Fourier transform');
%Spectrum analysis
Fs=[10];
xdft = XK(1:N/2+1);
psdx = (1/(Fs*N)) * abs(xdft).^2;
psdx(2:end-1) = 2*psdx(2:end-1);
freq = 0:Fs/N:Fs/2;
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 subplot(2,2,4);
plot(freq,psdx);
grid on;
xlabel('hz -->');
ylabel('db');
title('Spectrum analysis DFT');
disp('Spectrum analysis DFT=');
disp(psdx);

Output:
enter the input x = [1 2 3 4 5 6 7 8]
enter the order of dft N=[8]
Input Sequence x = 1 2 3 4 5 6 7 8

XK=
36.0000 + 0.0000i
-4.0000 - 9.6569i
-4.0000 - 4.0000i
-4.0000 - 1.6569i
-4.0000 + 0.0000i
-4.0000 + 1.6569i
-4.0000 + 4.0000i
-4.0000 + 9.6569i

Real part= 36 -4 -4 -4 -4 -4 -4 -4

Imaginary part= 0 9.6569 4.0000 1.6569 0 -1.6569 -4.0000 -9.6569

Magnitude= 36.0000 10.4525 5.6569 4.3296 4.0000 4.3296 5.6569 10.4525

Angle= 0 1.9635 2.3562 2.7489 3.1416 -2.7489 -2.3562 -1.9635

IDFT xn = [1 2 3 4 5 6 7 8]
Spectrum analysis DFT= 16.2000 2.7314 0.8000 0.4686 0.2000

Model Graph
Given sequence
8

6
Amplitude

0
1 2 3 4 5 6 7 8
time -->
Discrete Fourier Transform
40
Real Part
Imaginary Part
20
Amplitude

-20
1 2 3 4 5 6 7 8
time -->

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 Magnitude response
Phase response
40
4

30 2

ang x(k)
20 0

x(k)
10
-2
0
0 2 4 6 -4
8 0 2 4 6 8
time --> time -->
Inverse Discrete Fourier
Spectrum analysis DFT
transform 8
20
6
15

4
x

10

d
2
5
0
0 2 4 6 8 0
0 2 4 6
time -->
hz -->

Postlab Questions :

1. Mention the applications of DFT?

2. What is the drawback in Fourier transform and how it can be avoided?
3. Write the 4-point DFT matrix representation.
4. How many multiplications and additions are required to compute DFT for N=512?
5. Distinguish between DFT and DTFT.

Result:

Thus a MATLAB program for Spectrum analysis of the DFT and IDFT signal was written, executed and
the graphs were plotted.
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Ex: No: Date:

3(b). FAST FOURIER TRANSFORM AND INVERSE FAST FOURIER TRANSFORM

(FFT & IFFT)

Aim:
To write a program to find out the Fast Fourier transform and Inverse Fast Fourier transform of the
sequence using MATLAB and display the magnitude and phase response.

Prerequisite Lab Questions:

1. What do you mean by the term “bit reversal” as applied to FFT?
2. How many multiplications and additions are required to compute N point DFT using radix 2
FFT?
3. What are the advantages of FFT algorithm over direct computation of DFT?
4. State and periodicity and symmetry property of FFT?
5. Define in place computation?

Description

Input : Enter input x (n) as a sequence

Output : X (k) can be obtained by using formula,

FFT
X (k)  N  1 kn
 x(n)
n N
0
IFFT
1 N1
x(n)   kn
 X (k)
Nk0  N


j2kn
N
 kn 
Algorithm: N
e
1. Start the program.
2. Get the two input sequence x (n) and type of the FFT.
3. Compute the fast Fourier transform (FFT) of the given sequence.
4. Find out the magnitude and phase response of the given sequence.
5. Execute the program, display the result and verify it.
6. Plot the output graph for magnitude and phase.
7. Compute the inverse fast Fourier transform (FFT) of the given sequences.
8. Display the results verify it and plot the result.
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 9. Stop the process.

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 Program:
% Program for FFT & IFFT
clc;
clear all;
close all;
%x=[2 2 2 2 1 1 1 1];
%N=[8];
x=input('enter the input x=');
N=input('enter the order of fft
N='); k=0:1:N-1;

disp('Input Sequence x=');

disp(x);

XK=fft(x,N);
disp('XK=');
disp(XK);

R=real(XK);
disp('Real part=');
disp(R);

I=imag(XK);
disp('Imaginary part=');
disp(I);

mag=abs(XK);
disp('Mag=');
disp(mag);

ang=angle(XK);
disp('angle=');
disp(ang);

xn=ifft(XK,N);
disp('IFFT xn=');
disp(xn);

figure(1);
stem(R,'r*');
hold on;
stem(I);
legend('Real Part','Imaginary Part');
xlabel('time -->');
ylabel('Amplitude -->');
title('Fast Fourier Transform')

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 figure(2);
subplot(2,2,1);
stem(x);
xlabel('time -->');
ylabel('Amplitude -->');
title('Given sequence');
subplot(2,2,2);
stem(k,mag);
xlabel('time -->');
ylabel('x(k) --->');
title('Magnitude response');

subplot(2,2,3);
stem(k,ang);
xlabel('time -->');
ylabel('ang x(k) --->');
title('Phase response');

subplot(2,2,4);
stem(k,xn);
xlabel('time -->');
ylabel('x(n)');
title('Inverse Fast Fourier transform');

Output:
enter the input x = [2 2 2 2 1 1 1 1]
enter the order of fft N = [8]
Input Sequence x = 2 2 2 2 1 1 1 1
XK=
12.0000 + 0.0000i
1.0000 + 2.4142i
0.0000 + 0.0000i
1.0000 + 0.4142i
0.0000 + 0.0000i
1.0000 - 0.4142i
0.0000 + 0.0000i
1.0000 - 2.4142i

Real part = 12 1 0 1 0 1 0 1

Imaginary part = 0 -2.4142 0 -0.4142 0 0.4142 0 2.4142

Mag = 12.0000 2.6131 0 1.0824 0 1.0824 0 2.6131

Angle = 0 -1.1781 0 -0.3927 0 0.3927 0 1.1781

IFFT xn = 2 2 2 2 1 1 1 1

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Model Graph:
Fast Fourier Transform
12
Real Part
Imaginary Part
10

6
Amplitude

-2

-4
1 2 3 4 5 6 7 8
time -->

Given sequence
Magnitude response
2
15

1.5
10
Amplitude

1
x(k)

0.5 5

0
0 2 0
4 6 8 0 2 4 6 8
time --> time -->
Phase response Inverse Fast Fourier transform
2 2

1 1.5
ang x(k)

0 1
x(

-1
0.5

-2
0 2 4 6 8 0
0 2 4 6 8
time --> time -->

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 2
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Postlab Questions:
1. How many multiplications and additions are involved in radix-2 FFT for N=1024 when compared to
DFT?
2. Draw the 2-point DIF-FFT butterfly structure.
3. Compare DIF and DIT FFT algorithm.
4. Why FFT is needed?
5. Mention the applications of FFT algorithm?

Result:

Thus a program to find out the fast Fourier (FFT) and inverse fast Fourier transform (IFFT) of the
sequence using Matlab was written and executed.

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Ex No: Date:
4. FIR FILTER DESIGN

Aim:
To write a program to design the FIR low pass, high pass, band pass and band stop filters and obtain the
frequency response of the filter using MATLAB.

Prerequisite Lab Questions:

1. Mention the advantages and disadvantages of FIR filters.
2. What are the classifications of filters?
3. List the techniques for designing FIR filters.
4. What are the features of windowing method?
5. What are the types of windows in FIR filter design?

Description:

Input: Give the cutoff frequency and order of the filter.

Output: 1) Find the impulse response hd (n) for the given cut-off frequency and
Order of the filter.
2) Decide the window function w (n) and compute the window
Coefficients.
3) Find the filter coefficients h (n) =hd (n)*w (n).
4) Plot the frequency response for Low pass, High pass, Band pass and
Band stop filters

Algorithm:

1. Start the program.

2. Get the cutoff frequency and order of the filter as input...
3. Find the ideal impulse response of the filter hd (n).
4. Select a window function w (n).
5. Find out the filter coefficients h (n) by multiplying hd (n) and w (n).
6. Compute the frequency response of the filter.
7. Repeat the process for high pass, band pass and band stop filters.
8. Execute the program, display the result and verify it.
9. Plot the output graph for hd(n),w(n),h(n)and frequency response of LPF,HPF,BPF and BSF.
10. Stop the process.

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Program:

% Ideal LowPass filter computation

% hd = ideal impulse response between 0 to M-1
% wc = cutoff frequency in radians
% M = length of the ideal filter
% Save as ideal_lp

% Function:
function [hd]=ideal_lp(wc,M)
alpha=(M-1)/2;
n=0:1:M-1;
N=n-alpha+eps; % add smallest number to avoid divide by zero
hd=sin(wc*N)./(pi*N);

Program:
%PROGRAM FOR LOW PASS FILTER
clc;
clear all;
close all;
M=7;
wc=1.2;
n=0:1:M-1;
hd=ideal_lp(wc,M);
w=hamming(M);
h=hd.*w';
disp('Filter coefficients for LPF=');

[M1,P]=freqz(h,1,2000);
mag=20*log10(abs(M1));

subplot(2,2,1);
stem(n,hd);
xlabel('n--->');
ylabel('hd(n)--->');
title('Ideal Impulse Response');
disp('hd=');
disp(hd);
subplot(2,2,2);
stem(n,w);
xlabel('n--->');
ylabel('w(n)--->');
title('Hamming Window');
disp('w=');
disp(w);
subplot(2,2,3);
stem(n,h);
xlabel('n--->');
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 ylabel('h(n)--->');
title('Actual Impulse Response');
disp('h=');
disp(h);
subplot(2,2,4);
plot(P/pi, mag);
grid on;
xlabel('Normalized frequency--->');
ylabel('Magnitude in db--->');
title('Frequency Response ');

Output:
Filter coefficients for LPF=

hd = -0.0470 0.1075 0.2967 0.3820 0.2967 0.1075 -0.0470

w = 0.0800 0.3100 0.7700 1.0000 0.7700 0.3100 0.0800

h = -0.0038 0.0333 0.2284 0.3820 0.2284 0.0333 -0.0038

Model Graph:

LOW PASS FILTER

Ideal Impulse Response Hamming Window

0.6 1

0.4
hd(n)--

w(n)--

0.2
0.5

-0.2
0 2 4 6 0
0 2 4 6
n--->
n--->
Actual Impulse Response Frequency Response
0.6 0

0.4
-50
Magnitude in

0.2
h(n)--

0 -100

-0.2
-150
0 2 4 6
0 0.5 1
n--->
Normalized frequency--->

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%PROGRAM FOR HIGH PASS FILTER
clc;
clear all;
close all;
M=11;
wc=1.2;
n=0:1:M-1;
hd=ideal_lp(pi,M)-ideal_lp(wc,M);
w=hann(M);
h=hd.*w';
disp('Filter coefficients for HPF=');

[M1,P]=freqz(h,1,2000);
mag=20*log10(abs(M1));

subplot(2,2,1);
stem(n,hd);
xlabel('n--->');
ylabel('hd(n)--->');
title('Ideal Impulse Response');
disp('hd=');
disp(hd);

subplot(2,2,2);
stem(n,w);
xlabel('n--->');
ylabel('w(n)--->');
title('Hanning Window');
disp('w=');
disp(w);

subplot(2,2,3);
stem(n,h);
xlabel('n--->');
ylabel('h(n)--->');
title('Actual Impulse Response');
disp('h=');
disp(h);

subplot(2,2,4);
plot(P/pi, mag);
grid on;
xlabel('Normalized frequency--->');
ylabel('Magnitude in db--->');
title('Frequency Response ');

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 Output:

Filter coefficients for HPF=

hd = 0.0178 0.0793 0.0470 -0.1075 -0.2967 0.6180 -0.2967 -0.1075 0.0470 0.0793
0.0178

w= 0 0.0955 0.3455 0.6545 0.9045 1.0000 0.9045 0.6545 0.3455 0.0955 0

h= 0 0.0076 0.0162 -0.0704 -0.2683 0.6180 -0.2683 -0.0704 0.0162 0.0076 0

Model Graph:

HIGH PASS FILTER

Ideal Impulse Hanning Window

Response 1
1

0.5
hd(n)--

w(n)--

0.5
0

-0.5 0
0 5 10 0 5 10
n---> n--->
Actual Impulse Response Frequency Response
1 50

0.5 0
Magnitude in
h(n)--

0 -50

-0.5 -100
0 5 10 0 0.5 1
n---> Normalized frequency--->

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%PROGRAM FOR BAND PASS FILTER
clc;
clear all;
close all;
M=11;
wc1=1.2;
wc2=2;
n=0:1:M-1;
hd=ideal_lp(wc2,M)-ideal_lp(wc1,M);
w=hann(M);
h=hd.*w';
disp('Filter coefficients for BPF=');

[M1,P]=freqz(h,1,2000);
mag=20*log10(abs(M1));

subplot(2,2,1);
stem(n,hd);
xlabel('n--->');
ylabel('hd(n)--->');
title('Ideal Impulse Response');
disp('hd=');
disp(hd);

subplot(2,2,2);
stem(n,w);
xlabel('n--->');
ylabel('w(n)--->');
title('Hanning Window');
disp('w=');
disp(w);

subplot(2,2,3);
stem(n,h);
xlabel('n--->');
ylabel('h(n)--->');
title('Actual Impulse Response');
disp('h=');
disp(h);

subplot(2,2,4);
plot(P/pi, mag);
grid on;
xlabel('Normalized frequency--->');
ylabel('Magnitude in db--->');
title('Frequency Response ');

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 Output:

Filter coefficients for BPF=

hd = -0.0168 0.1580 0.0173 -0.2280 -0.0072 0.2546 -0.0072 -0.2280 0.0173

0.1580 -0.0168

w= 0 0.0955 0.3455 0.6545 0.9045 1.0000 0.9045 0.6545 0.3455 0.0955 0

h= 0 0.0151 0.0060 -0.1492 -0.0065 0.2546 -0.0065 -0.1492 0.0060 0.0151 0

Model Graph:

BANDPASS FILTER

Ideal Impulse Hanning Window

0.4 Response 1

0.2
hd(n)--

w(n)--

0 0.5

-0.2

-0.4 0
0 5 10 0 5 10
n---> n--->
Actual Impulse Response Frequency Response
0.6 0

0.4 -20
Magnitude in
h(n)--

0.2 -40

0 -60

-0.2 -80
0 5 10 0 0.5 1
n---> Normalized frequency--->

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%PROGRAM FOR BAND STOP FILTER
clc;
clear all;
close all;
M=7;
wc1=1.2;
wc2=2;
n=0:1:M-1;
hd=ideal_lp(pi,M)-ideal_lp(wc2,M)+ideal_lp(wc1,M);
w=rectwin(M);
h=hd.*w';
disp('Filter coefficients for BSF=');

[M1,P]=freqz(h,1,2000);
mag=20*log10(abs(M1));

subplot(2,2,1);
stem(n,hd);
xlabel('n--->');
ylabel('hd(n)--->');
title('Ideal Impulse Response');
disp('hd=');
disp(hd);

subplot(2,2,2);
stem(n,w);
xlabel('n--->');
ylabel('w(n)--->');
title('Rectangular Window');
disp('w=');
disp(w);

subplot(2,2,3);
stem(n,h);
xlabel('n--->');
ylabel('h(n)--->');
title('Actual Impulse Response');
disp('h=');
disp(h);

subplot(2,2,4);
plot(P/pi, mag);
grid on;
xlabel('Normalized frequency--->');
ylabel('Magnitude in db--->');
title('Frequency Response ');

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 Output:

Filter coefficients for BSF=

hd = -0.0173 0.2280 0.0072 0.7454 0.0072 0.2280 -0.0173

w= 1 1 1 1 1 1 1

h = -0.0173 0.2280 0.0072 0.7454 0.0072 0.2280 -0.0173

Model Graph:
BANDSTOP FILTER

Ideal Impulse Rectangular Window

Response 1
1

0.5
hd(n)--

w(n)--
0.5
0

-0.5 0
0 2 4 0 2 4 6
6
n--->
n--->
Frequency Response
Actual Impulse Response
5
1
Magnitude in db---

0
0.5
h(n)--

-5
0
-10

-0.5 -15
0 4 0 0.5 1
Normalized frequency--->
2 6
n--->

Windows:
Theoretical calculation:

1. hd (n) =1/2π ∫ Hd(w) ejwn dw

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2. Select w (n)
3. Find h (n) = hd (n)w(n)

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Rectangular Window

1,
w(n)  0  n  M 1
0,
otherwise
Hanning Window
   2n 
0.5* 1 cos 0  n  M 1
, M
1
   
w(n) 
0,
otherwise

Hamming Window

  2n 
0.54  0.46*
cos , 0  n  M 1

  M 1
1. w(n) 
0, otherwise

Kaiser window

  1

w(n) 
I

 1  (n  ) / 
2
 2
/ I ( ) 0  n  M 1
 0  0

0 otherwise

M-1)/2 and I0 = the zeroth order Modified Bessel Function

If  p ,  s , Rp , and As are given then the following equations are needed for design.

Transition width  s   p
f  2

Filter order M  As  7.95 1

14.36f
 0.1102( A  8.7), As  50
Parameter  
A 21)0.4  s0.07886(  21),
A 21   50
0.5842( As s s

 0 As  21
Postlab questions:
1. What is meant by linear phase filters?
2. What is the condition for the impulse response of FIR filter to satisfy for constant and
Phase delay and for only constant group delay?
3. What are Gibbs oscillations?
4. Give the equation for rectangular, hamming and Kaiser Windows.
5. State the condition for linear phase characteristics in FIR.

4
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Result:
Thus a program to design the FIR low pass, high pass, band pass and band stop filters was
Written and response of the filter using MATLAB was executed.

4
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Ex No: Date:
5. IIR FILTER DESIGN
Aim:
To write a program to design the Butterworth low pass, high pass filters and find out the response of the
filter using MATLAB.

Prerequisite Lab Questions:

1. What are the requirements for an analog filter to be stable and causal?
2. What are the types of analog filters?
3. State the conditions to design a stable digital filters.
4. Write the relationship between analog and digital frequency in impulse invariant method?
5. Define Bilinear transformation.

Description:

Input :( a) Enter the pass band ripple and stop band

attenuation (b)Enter the passband and stop band
frequency

Output : (a) Find the order and cutoff frequency using bilinear and
impulse Invariant transformation
(b) Calculate the Analog filter coefficients H(s)
(c) Transform the Analog filter to digital filter and find H (z)
(d) Plot the responses for all the filters.
Algorithm:

1. Start the program.

2. Get the input sequences for pass band ripple, stop band ripple, pass band frequency, stop
band frequency and sampling frequency.
3. Convert the frequency into radians/sec.
4. Compute the order and cutoff frequency of the filter and display it.
5. Design the analog filter and find the coefficient.
6. Convert the analog filter into digital using bilinear or impulse invariant transform.
7. Find out the frequency response of the filter.
8. Compute the magnitude and phase response of the filter.
9. Execute the program, display the result and verify it.
10. Plot the output graph for each sequence for low pass and high pass filters.
11. Stop the process.

Program:
% function to find order of butterworth filter using impulse invariant transformation
% Save as butt_impord
function [N,Wc]=
butt_impord(w1,w2,a1,a2) l=sqrt((1/a2^2)-1)
e=sqrt((1/a1^2)-1)
num=log10(l/e)
den=log10(w2/w1)

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N
=
c
e
i
l
(
(
n
u
m
/
d
e
n
)
)

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 Wc=(w1)/((e)^(1/N))
%function to find order of butterworth filter using bilinear transformation Save is as butt_biord.
function [N1,Wc1]= butt_biord(w1,w2,a1,a2)
T=1;
wp1=(2/T)*tan(w1/2)
ws1=(2/T)*tan(w2/2)
lam=sqrt((1/a2^2)-1)
ep=sqrt((1/a1^2)-1)
num1=log10(lam/ep)
den1=log10(ws1/wp1)
N1=ceil((num1/den1))
Wc1=wp1/((ep)^(1/N1))

% Program for Butterworth Low Pass Filter

clc;
clear all;
close all;
%w1=0.2*pi;w2=0.6*pi;a1=0.8;a2=0.2;
w1=input('Enter the Passband frequency w1=');
w2=input('Enter the Stopband frequency w2=');
a1= input('Enter the Passband ripple a1=');
a2=input('Enter the Stopband ripple a2=');
[N,Wc]=butt_impord(w1,w2,a1,a2);
disp('LPF ANALOG FILTER TRANSFER FUNCTION');
[num,den]=butter(N,Wc,'s')
disp('LPF DIGITAL FILTER TRANSFER FUNCTION');
[b,a]=impinvar(num,den)
[mag,angle]=freqz(b,a,512);
magnitude=20*log10(abs(mag));
figure(1);
plot(angle/pi,magnitude);
grid on;
title('Low Pass Butterworth Filter');
xlabel('frequency --->');
ylabel('magnitude --->');

%Program for Butterworth High Pass Filter

[N1,Wc1]=butt_biord(w2,w1,a1,a2);
disp(' HPF ANALOG FILTER TRANSFER FUNCTION');
[num1,den1]=butter(N1,Wc1,'high','s')
disp('HPF DIGITAL FILTER TRANSFER FUNCTION');
[b1,a1]=bilinear(num1,den1,1)
[mag1,angle1]=freqz(b1,a1,512);
magnitude=20*log10(abs(mag1));
figure(2);
plot(angle1/pi,magnitude);
grid on;
title('High Pass Butterworth Filter');
xlabel('frequency --->');
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 ylabel('magnitude --->');
Output:
Enter the Passband frequency w1=[0.2*pi]
Enter the Stopband frequency w2=[0.6*pi]
Enter the Passband ripple a1=[0.8]
Enter the Stopband ripple

a2=[0.2] l = 4.8990 ,e = 0.7500

num = 0.8150 , den = 0.4771

N = 2 , Wc = 0.7255

LPF ANALOG FILTER TRANSFER

FUNCTION num = 0 0 0.5264

den = 1.0000 1.0260 0.5264

LPF DIGITAL FILTER TRANSFER FUNCTION

b= 0 0.3015 0

a = 1.0000 -1.0432 0.3584

T = 1, wp1 = 0.6498

ws1 = 2.7528, lam = 4.8990

ep = 0.7500

num1 = 0.8150, den1 = 0.6270

N1 = 2

Wc1 = 0.7504

HPF ANALOG FILTER TRANSFER FUNCTION

num1 = 1 0 0

den1 = 1.0000 1.0612 0.5631

HPF DIGITAL FILTER TRANSFER FUNCTION

b1 = 0.5983 -1.1966 0.5983

a1 = 1.0000 -1.0282 0.3651

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Model graph
Low Pass Butterworth Filter
0

-5

-10

-15

-20
magnitude

-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
frequency --->

High Pass Butterworth Filter

0

-20

-40

-60

-80
magnitude

-100

-120

-140

-160

-180

-200
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
frequency --->

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Theoretical calculation

Bilinear Transformation

wp1=(2/T)*tan(w1/2)
ws1=(2/T)*tan(w2/2)
lam=sqrt((1/a2^2)-1)
ep=sqrt((1/a1^2)-1)
N=log10(lam/ep) / log10(ws1/wp1);
Wc1=wp1/((ep)^(1/N1));

To find H (Z) replace„s‟ by

S= 2 (Z-1)
T (Z+1)

Impulse Invariant Transformation

l=sqrt((1/a2^2)-1)
e=sqrt((1/a1^2)-1)
N=log10(l/e) / log10(w2/w1); ;
Wc=(w1)/((e)^(1/N));
H(S) is same

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Postlab questions:
1. What are the properties of Butterworth low pass filters?
2. Give the magnitude function of Butterworth filter.
3. What is meant by conformal mapping?
4. Define warping effect?
5. What are the advantages and disadvantages of bilinear transformations?

Result:
Thus a program to design the Butterworth low pass and high pass filter using Matlab was written and
response of the filter was executed.

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 Ex No: Date:

6. MULTIRATE FILTERS

Aim:
Write a program to perform the decimation and interpolation on the given input sequence operation using
Matlab.

Prerequisite Lab Questions:

1. What are the needs for multirate signal processing?
2. Define Decimator and Interpolator.
3. How aliasing effect is overcome in decimation?
4. Write the time domain relationship for upsampling and downsampling.
5. List the advantages of MDSP.

Description:

Input: Give the input signal

Output: Find the decimated and interpolated output signal for the given number of samples.

Algorithm:
1. Start the program.
2. Get the amplitude of the input signal.
3. Get the decimation and interpolation factor.
4. Perform the decimation and interpolation operations using Matlab functions.
5. Plot the output signals.
6. Stop the process.

Program:
%Program for Decimate signal

clc;
clear all;
close all;
t=0:0.00025:1; % Time vector
x=sin(2*pi*30*t)+sin(2*pi*60*t);
r=4; % Factor of 4
y=decimate(x,r);

%View the original signal

figure(1);
subplot(2,1,1);
stem(x(1:120));
xlabel('n--->');
ylabel('Amplitude--->');

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 title('Original Signal');

%View the decimated signal

subplot(2,1,2);
stem(y(1:30));
xlabel('n--->');
ylabel('Amplitude--->');
title('Decimated Signal');

%Program for Interpolated signal

t=0:0.001:1; % Time vector

x=sin(2*pi*30*t)+sin(2*pi*60*t);
r=4; % Factor of 4
y=interp(x,r);
%View the Original signal
figure(2);
subplot(2,1,1);
stem(x(1:30));
xlabel('n--->');
ylabel('Amplitude--->');
title('Original Signal');

%View the Interpolated Signal

subplot(2,1,2);
stem(y(1:120));
xlabel('n--->');
ylabel('Amplitude--->');
title('Interpolated Signal');

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Model graph:

Original Signal
2

1
Amplitude---

-1

-2
0 20 40 60 80 100 120
n--->
Decimated Signal
2

1
Amplitude---

-1

-2
0 5 10 15 20 25 30
n--->

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 Original Signal
2

Amplitude---
0

-1

-2
0 5 10 15 20 25 30
n--->
Interpolated Signal
2

1
Amplitude---

-1

-2
0 20 40 60 80 100 120
n--->

Postlab Questions:

1. What are the effects of upsampling?

2. List the applications of multirate signal processing?
3. What are sampling rate converters?
4. Define sub band coding of signals.
5. For the given signal x[n]= {1,3,4,5,6,7,8,9,1,-2} . Decimate and interpolate by a factor of 2.

Result:
Thus a program to perform decimation and interpolation the input signal was written using MATLAB
and plotted.

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Ex No: Date:
7. EQUALIZATION

Aim:

To Design an Least Mean Square (LMS) channel equalizer using MATLAB.

Prerequisite Lab Questions:

1. Define adaptive filtering?
2. What are the two steps in adaptive filtering?
3. Whether FIR or IIR filters are used for adaptive filtering?
4. List the classifications of algorithms for adaptive filtering.
5. What is the need for adaptive equalization?
THEORY:
The Least Mean Square algorithm or non-recursive adaptive filter is routinely used in filtering
applications that range from adaptive equalizers to adaptive noise control systems. The various reasons for its
priority, the stability are controlled. There are simple and efficient algorithms to adjust the filter coefficients.
Finally the performance of these algorithms is well understood in terms of convergence and stability. For an
LMS algorithm having p+1 coefficient require P+1 multiplications and p additions to calculate the output. The
simplicity of this algorithm comes from the update equation. And most importantly the data and LMS weight
vector are statistically independent. The LMS algorithm is the most commonly used nonrecursive channel
equalizer technique.

ALGORITHM:
1. The Input values are all set
2. For input generate normally distributed random matrix.
3. For filter generate uniform distributed pseudo random matrix.
4. The filter coefficients are determined.
5. The input vector is filtered with the specified vector and stored in another vector.
6. Now all the outputs and vectors are initialized.
7. Now the LMS adaptation curve is determined using the equation
w=w+mu(Re(e(n)*conj(in))-iImg(e(n)*conj(in)))
8. Finally the generated output waveform is plotted in a semilog graph sheet

% ****************************************************
% LEAST MEAN SQUARE ALGORITHM
% ****************************************************
clear all;
close all;
randn('state',sum(100*clock)); % Returns a matrix with pseudorandom values in normal
distribution
rand('state',sum(100*clock)); % Returns a matrix with pseudorandom values in uniform
distribution
numpoints=5000; % Sampling points
numtaps = 10; % No of filter coefficients
Mu=0.01;
x=randn(numpoints,1)+i*randn(numpoints,1); % Complex Random Input Signal
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 h=rand(numtaps,1); % filter transfer function
h=h /max(h);
d = filter(h,1,x); % digital filter with num coeff h and den coeff 1 and vector x
w=[];
y=[];
in=[];
e=[];
w=zeros(numtaps+1,1)+i*zeros(numtaps+1,1); % initialize the w matrix
for n=numtaps+1:numpoints
in=x(n:-1:n-numtaps);
y(n)=w'*in;
e(n)=d(n)-y(n); % Calculates the error value
w=w+Mu*(real(e(n)*conj(in))-i*imag(real(e(n)*conj(in))));
end
figure(10);
semilogy(abs(e));grid on;
title(['LMS Adaptation Learning Curve using Mu=',num2str(Mu)]);
xlabel('Iteration Number');
ylabel('Output estimation error in db');

OUTPUT

Post lab Questions:

1. List the applications of adaptive filtering.
2. What do you mean by adaptive noise cancellation?
3. Define mean square error criterion.
4. What is in-phase and quadrature signals?
5. What are the characteristics of LMS algorithm?

RESULT:
Thus the Least Mean Square (LMS) channel equalizer was designed and the generated output sequence
was plotted using MATLAB software.

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Ex No: Date:

8. STUDY OF ARCHITECTURE OF DIGITAL SIGNAL PROCESSOR

Aim:
To Study the architecture of TMS320C5416 Digital Signal Processor Starter Kit (DSK).

Description:
The TMS320VC5416 fixed-point, digital signal processor (DSP) (hereafter referred to as the device
unless otherwise specified) is based on an advanced modified Harvard architecture that has one program
Memory bus and three data memory buses. This processor provides an arithmetic logic unit (ALU) with a High
degree of parallelism, application-specific hardware logic, on-chip memory, and additional on-chip Peripherals.
The basis of the operational flexibility and speed of this DSP is a highly specialized instruction Set. Separate
program and data spaces allow simultaneous access to program instructions and data, providing, a high degree
of parallelism. Two read operations and one write operation can be performed in a single cycle. Instructions
with parallel store and application-specific instructions can 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. The device also includes
the control mechanisms to manage interrupts, repeated
Operations, and function calls.
Architecture:
The ‟54x DSPs use an advanced, modified Harvard architecture that maximizes processing power by
maintaining one program memory bus and three data memory buses. These processors also provide an
arithmetic logic unit (ALU) that has a high degree of parallelism, application-specific hardware logic, on-chip
memory, and additional on-chip peripherals. These DSP families also provide a highly specialized instruction
set, which is the basis of the operational flexibility and speed of these DSPs. Separate program and data spaces
allow simultaneous access to program instructions and data, providing the high degree of parallelism. Two
reads and one write operation can be performed in a single cycle. Instructions with parallel store and
application-specific instructions can 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 included are the control mechanisms to
manage interrupts, repeated operations, and function calls.

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Architecture of TMS320VC5416 Fixed-Point Digital Signal Processor:

Central Processing Unit (CPU)

The CPU is common to all C54x™ devices. The C54x CPU contains:
40-bit arithmetic logic unit (ALU)
Two 40-bit accumulators
Barrel shifter
17 × 17-bit multiplier
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 40-bit adder
Compare, select, and store unit (CSSU)
Data address generation unit
Program address generation unit
Arithmetic Logic Unit (ALU)
The C54x 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,
T Two 16-bit words from data memory
32-bit word from data memory
40-bit word from either accumulator
The ALU can also function as two 16-bit ALUs and perform two 16-bit operations
simultaneously.
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.
Barrel Shifter
The C54x 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). The
multiplier has two inputs: one input is selected from T, a data-memory operand, or accumulator A; the
other is selected from program memory, data memory, accumulator A, or an immediate value.
The fast, on-chip multiplier allows the C54x DSP to perform operations efficiently such as
convolution, correlation, and filtering. In addition, the multiplier and ALU together execute
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 multiply/accumulate (MAC) computations and ALU operations in parallel in a single instruction cycle.
This function is used in determining the Euclidian distance and in implementing symmetrical and LMS
filters, which are required for complex DSP algorithms.
Compare, Select, and Store Unit (CSSU)
The compare, select, and store unit (CSSU) performs maximum comparisons between the
accumulator‟s high and low word, allows both the test/control flag bit (TC) in status register ST0 and
the transition register (TRN) to keep their transition histories, and selects the larger word in the
accumulator to store into data memory. The CSSU also accelerates Viterbi-type butterfly computations
with optimized on-chip hardware.
Status Registers (ST0, ST1)
The status registers ST0 and ST1 contain the status of the various conditions and modes for the
C54x devices. ST0 contains the flags (OVA, OVB, C, and TC) produced by arithmetic operations and
bit manipulations, in addition to the DP and the ARP fields. ST1 reflects the status of modes and
instructions executed by the processor.

Auxiliary Registers (AR0–AR7)

The eight 16-bit auxiliary registers (AR0–AR7) can be accessed by the CPU and modified by the
auxiliary register arithmetic units (ARAUs). The primary function of the auxiliary registers is to
generate 16-bit addresses for data space. However, these registers can also act as general-purpose
registers or counters.
Transition Register (TRN)
The 16-bit transition (TRN) register holds the transition decision for the path to new metrics to
perform the Viterbi algorithm. The CMPS (compare select max and store) instruction updates the
contents of TRN register on the basis of the comparison between the accumulator high word and the
accumulator low word.

Temporary Register (T)

The temporary (T) register has many uses. For example, it may hold:
One of the multiplicands for multiply and multiply/accumulate instructions (For more details
about the T register and the processes of multiplication.
A dynamic (execution-time programmable) shift count for instructions with shift operation such
as the ADD, LD, and SUB instructions.
A dynamic bit address for the BITT instruction.
Branch metrics used by the DADST and DSADT instructions for ACS operation of Viterbi
decoding.
In addition, the EXP instruction stores the exponent value computed into T register, and then the
NORM instruction uses the T register value to normalize the number.
Stack-Pointer Register (SP)
The 16-bit stack-pointer register (SP) contains the address of the top of the system stack. The SP
always points to the last element pushed onto the stack. The stack is manipulated by interrupts, traps,
calls, returns, and the PSHD, PSHM, POPD, and POPM instructions. Pushes and pops of the stack
predecrement and postincrement, respectively, the 16-bit value in the stack pointer.

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Circular-Buffer Size Register (BK)
The ARAUs use16-bit circular-buffer size register (BK) in circular addressing to specify the data
block size. For information on BK and circular addressing.
Block-Repeat Registers (BRC, RSA, REA)
The 16-bit block-repeat counter (BRC) register specifies the number of times a block of code is
to repeat when a block repeat is performed. The 16-bit blockrepeat start address (RSA) register contains
the starting address of the block of program memory to be repeated. The 16-bit block-repeat end address
(REA) register contains the ending address of the block of program memory to be repeated.

Memory-Mapped Registers
The 64K words of data memory space include the device‟s memory-mapped registers, which
reside in data page 0 (data addresses 0000h–007Fh).
The peripheral registers are used as control and data registers in peripheral circuits. These
registers reside within addresses 0020h–005F and reside on a dedicated peripheral bus structure. For a
list of peripherals on a particular C54x device.
The scratch-pad RAM block (60h–7Fh in data memory) includes 32 words of DARAM for
variable storage that helps avoid fragmenting the large RAM block.The data memory space contains
memory-mapped registers for the CPU and the on-chip peripherals. Simplifying access to them. The
memory-mapped access provides a convenient way to save and restore the registers for context switches
and to transfer information between the accumulators and the other registers.

Data Memory
The data memory space addresses up to 64K of 16-bit words. The device automatically accesses
the On-chip RAM when addressing within its bounds. When an address is generated outside the RAM
bounds, the device automatically generates an external access.
The advantages of operating from on-chip memory are as follows:
· Higher performance because no wait states are required
· Higher performance because of better flow within the pipeline of the central arithmetic logic
unit (CALU)
· Lower cost than external memory
· Lower power than external memory
The advantage of operating from off-chip memory is the ability to access a larger address space.
Program Memory
Software can configure their memory cells to reside inside or outside of the program address
map. When the cells are mapped into program space, the device automatically accesses them when their
addresses are within bounds. When the program-address generation (PAGEN) logic generates an
address outside its Bounds, the device automatically generates an external access.
The advantages of operating from on-chip memory are as follows:
· Higher performance because no wait states are required
· Lower cost than external memory
· Lower power than external memory
The advantage of operating from off-chip memory is the ability to access a larger address space.

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Overview of the TMS320VC5416 DSK
The TMS320VC5416 DSK is a stand-alone development and evaluation module. It allows
evaluators to examine certain characteristics of the C5416 digital signal processor (DSP) to determine if
it meets their application requirements. Furthermore, the module is an excellent platform to develop and
run software for the TMS320VC5416 family of processors.
The DSK allows full speed verification of VC5416 code. With 64K words of on board RAM
memory, 256K words of on board Flash ROM, and a Burr Brown PCM 3002 stereo codec, the board can
solve a variety of problems as shipped. Three expansion connectors are provided for interfacing to
evaluation circuitry not provided on the as shipped configuration.
To simplify code development and shorten debugging time, a special version of Code Composer
Studio is shipped with the board.
Key Features of the TMS320VC5416 DSK
The VC5416 DSK has the following features:
• VC5416 operating at 16-160 MHz
• On board USB JTAG controller with plug and play drivers
• 64K words of on board RAM
• 256K words of on board Flash ROM
• 3 Expansion Connectors (Memory Interface, Peripheral Interface, and Host Port Interface)
• On board IEEE 1149.1 JTAG Connection for Optional Emulation Debug
• Burr Brown PCM 3002 Stereo Codec
• +5 volt operation
Code Composer Studio - 3.1v
INTRODUCTION TO CODE COMPOSER STUDIO

Code Composer is the DSP industry's first fully integrated development environment (IDE) with
DSP-specific functionality. With a familiar environment liked MS-based C++TM, Code Composer lets
you edit, build, debug, profile and manage projects from a single unified environment. Other unique
features include graphical signal analysis, injection/extraction of data signals via file I/O, multi-
processor debugging, automated testing and customization via a C-interpretive scripting language and
much more.
Initial setup to work on Code Composer Studio:
 To execute using Simulator Open Setup CCStudio_v3.1 add C5416 Device Simulator Save and
Quit.
(OR)
 To Execute using DSP Processor Kit -DSK Diagnostic
UtilityStart Check for PASS in the display and continue the
Procedure
 Select 5416 dskCCStudio_v3.1DebugConnect

Procedure to work on Code Composer Studio:

To create the New Project

Project  New  (C:\CCStudio_v3.1\MyProjects\projectname.pjt)
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 To Create a Source file
File  New  Type the code (Save & give file name, Eg: filename.c or
filename.asm). (C:\CCStudio_v3.1\MyProjects\projectname.pjt\
filename.c)

To Add Source files to Project

Project  Add files to Project  filename.c

To add library & cmd:

Project  Add files to Project rts_EXT.lib
Library files: (Path: C:\ CCStudio\C5400\cgtools\lib\rts_EXT.lib)
Note: Select Object & Library from all files
Project  Add files to Project c5416dsk.cmd
CMD file – Which is common for all non-real time programs.
(Path: Path: C:\ CCStudio\tutorial\dsk5416\shared\c5416dsk.cmd)
Note: Select Linker Command file (*.cmd) in Type of files

To Compile:
Project  Compile

To Build:
Project Build
If there is more number of warnings after build, then
Project Build Options  Advanced (in Category) Use Far Calls (- mf)

Procedure to Load and Run program:

Load the program to DSK: File  Load program  project name. Out (C:\CCStudio_v3.1\
MyProjects\projectname.pjt\Debug\projectname.out)

To execute project:
Debug  Run.

To Plot Graph:
View  Graph (OR Right click in the program Insert Graph)
Select proper Properties and specify the address.

Result:

Thus the architecture of TMS320C5416 Digital Signal Processor Starter Kit (DSK) and programming
steps for Code Composer Studio was studied.

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 Ex No: Date:
9. MAC OPERATION USING VARIOUS ADRESSING MODES

Aim:
To perform MAC operation using various addressing modes of TMS320C5416 DSP processor.

Prerequisite Lab Questions:

1. What are the classifications of DSP Processors?
2. Compare fixed and floating point Processors?
3. Brief the different types of addressing mode present in the TMS320C5416?
4. Define program counter.
5. What is the architectural difference between conventional and DSP proceesors?

Algorithm:

1. Start the program.

2. Store the first number from X memory to AR2.
3. Store the second number from Y memory to AR3.
4. Store the immediate address to AR4 to store the output.
5. Add both the numbers and store the result in accumulator A.
6. Store the lower order word to the memory location 1500 and higher order word to 1501 and
increment the memory location
7. Load the content pointed by AR2 to the temporary register
8. Multiply the Content pointed by AR2 with the temporary register and store the result in B register.
9. Store the lower order word and higher order word of B register to the memory location pointed by
AR4
10. Stop the program execution.

Program:

; Program for Addition

.global _main
.mmregs
.data
X: .word 50h
Y: .word 20h
.text
_main
STM X,AR2
STM Y,AR3
STM #1500h,AR4
ADD *AR2,*AR3,A
STL A,*AR4+
STH A,*AR4
.end

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Output:
0x1500: 0x0000
0x1501: 0x0070

; Program for Subtraction

.global _main
.mmregs
.data
X: .word 60h
Y: .word 20h
.text
_main
STM X,AR2
STM Y,AR3
STM #1500h,AR4
SUB *AR2,*AR3,A
STL A,*AR4+
STH A,*AR4
.end

Output:
0x1500: 0x0000
0x1501: 0x0040

; Program for Multiplication

.global _main
.mmregs
.data
X: .word 30h
Y: .word 20h
.text
_main
STM X,AR2
STM Y,AR3
STM #1500h,AR4
MPY *AR2,*AR3,A
STL A,*AR4+
STH A,*AR4
.end

Output:
0x1500: 0x0600

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 ; Program for Division

.global _main
.mmregs
.data
X: .word 90h
Y: .word 30h
.text
_main
STM X,AR2
STM Y,AR3
STM #1500h,AR4
LD *AR2+,A
RPT #15
SUBC *AR2,A
STL A,*AR4+
STH A,*AR4
.end

Output:
0x1500: 0x0003

Postlab Questions:
1. Give an example for immediate and indirect addressing modes?
2. What do you mean by bit reversal addressing mode?
3. What are the registers available for indirect addressing mode?
4. What are barrel shifters?
5. What is arithmetic and logical shift in DSP Processors?

Result:
Thus the assembly coding for MAC operation using various addressing modes of DSP processor
TMS320C5416 was written and executed.

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Ex No: Date:
10. LINEAR CONVOLUTION
Aim:
Write a program to find out the linear convolution of two sequence using TMS320C5416 processor.

Prerequisite Lab Questions:

1. Define Convolution.
2. What is zero padding ?
3. List the properties of linear convolution.

Description:

Input: Give the input x (n) and h (n) in matrix form.

Output: y (n), using the formula



y(n)   x(k )h(n  k)

k 
Algorithm:
1. Start the program.
2. Get the two input sequence x (n) and h (n) with zero appended.
3. Sum the product of x (n) and shifted sequence of h (n-k).
4. Execute the program, display the result and verify theoretically.
5. Stop the process.

PROGRAM FOR LINEAR CONVOLUTION

//PROGRAM FOR LINEAR CONVOLUTION

#include<stdio.h>
int x[10]; /*={1 2 3 4 0 0 0} Input Signal Samples*/
int h[10]; /*={1 2 3 4 0 0 0} Impulse Response */
/*At the end of input sequences pad 'M' and 'N' no. of zero's*/
int y[10];
int m=4; /*Lenght of input samples sequence x(n)*/
int n=4; /*Lenght of impulse response h(n) */
int i,j;
main()
{
printf(" enter the first sequence\n");
for(i=0;i<m+n-1;i++)
scanf("%d",&x[i]);
printf(" enter the second sequence\n");
for(j=0;j<m+n-1;j++)
scanf("%d",&h[j]);
for(i=0;i<m+n-1;i++)
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 { y[i]=
0;
for(j=0;j<=i;j++)
y[i]+=x[j]*h[i-j];
}
printf("output\n");
for(i=0;i<m+n-1;i++)
printf("%d\n",y[i]);
}

Result:

Enter the first sequence

1234000
Enter the second sequence
1111000
Output:
1
3
6
10
9
7
4

Procedure for Plot:

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MODEL GRAPH

Postlab Questions :

1. State convolution property of a signal.

2. What are sectioned convolutions & what is the need of it?
3. Write the applications of convolution.

Result:

Thus a program to find out the linear convolution of two sequences using TMS320C5416 processor was
written and executed.

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Ex No: Date:
11. CIRCULAR CONVOLUTION

Aim:
Write a program to find out the circular convolution of two sequences using TMS320C5416 processor.

Prerequisite Lab Questions:

1. What is the MATLAB function used to perform linear & circular convolution?
2. How will you determine the maximum length of the convolution of the sequences?
3. List the methods used to find the circular convolution.

Description:
Input: Give the input x (n) and h (n) in matrix form.

Output: y (n), using the formula

y(n)  x(n)  h(n)

N 1
y(n) 
x(k)h(n  k)
k 0
Algorithm:

1. Start the program.

2. Get the two input sequence x (n) and h (n) and compute the length of the sequence.
3. Append zeros to x (n) or h (n) to make both sequence of same length.
4. Rotate the sequence h (-k), „n‟ times. If „n‟ is positive shift the sequence in anticlockwise direction
else in the clockwise direction.
5. Determine the product of sequence x (n) and shifted h (n) sequence and sum it.
6. Execute the program, display the result and verify it theoretically.
7. Plot the output graph for each sequence.
8. Stop the process.

PROGRAM FOR CIRCULAR CONVOLUTION

//PROGRAM FOR CIRCULAR CONVOLUTION

#include<stdio.h>
int m,n,x[30],h[30],y[30],i,j,temp[30],k,x2[30],a[30];
void main()
{
printf(" enter the length of the first sequence\n");

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 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];
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]);
}

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Model Graph:

Output:

enter the length of the first sequence

5
enter the length of the second sequence
4
enter the first sequence
13579
enter the second sequence
2442
the circular convolution is
76 60 44 48 72

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Properties to Plot Graph:

Postlab Questions:

1. What is the difference between Linear convolution & circular convolution?

2. What is the relation between convolution in time domain and frequency domain ?
3. Give the significance of using convolution process in digital signal processing.

Result:

Thus a program to find out the circular convolution of two sequences using TMS320C5416 processor
was written and executed.

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Ex No: Date:
12. FFT IMPLEMENTATION

Aim:
To write a program to perform Discrete Fourier transform using TMS320C5416 DSP Processor.

Prerequisite Lab Questions:

1. Write the analysis and synthesis equation for DFT?

2. What do you mean by the property of shifting of DFT?
3. Find the value of twiddle factor for N=8, K=2 .
4. How many multiplications and additions are required to compute N point DFT ?
5. What is the relationship between DFT and Z-transform?
Algorithm:

1. Start the program.

2. Declare the variables
3. Generate the input sequence
4. Using DFT, calculate the real and imaginary part of X (k) for k=0, 1... N-1
5. Calculate the Magnitude of X (k).
6. Print the Result.
7. Stop the Process.

Program:

//Program for DFT

#include<stdio.h>
#include<math.h>
#define pi 3.1415
#define N 8
float X[N];
int i,k,n;
float xr[N],xi[N];
float XR[N],XI[N];
main()
{

for (i=0;i<=N-1;i++)
{
xr[i]=sin(2*pi*10*i/8.0);
xi[i]=0;
}
for (k=0;k<N;k++)
{

XR[k]=0;
XI[k]=0;

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 for (n=0;n<N; n++)
{
XR[k]+=(xr[n]*cos(2*pi*k*n/N))+(xi[n]*sin(2*pi*k*n/N));
XI[k]+=(xr[n]*sin(2*pi*k*n/N))-(xi[n]*cos(2*pi*k*n/N));
}

X[k]=sqrt((XR[k]*XR[k])+(XI[k]*XI[k]));
}
printf("X(k) and magnitude of X(k)\n");

for (k=0;k<N;k++)
{
printf("\n X[%d] = %f + j %f \n",k,XR[k],XI[k]);
printf(" magnitude of X(%d) = %f\n",k,X[k]);
}
}

Model Graph:

Graph Properties:

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Sample Values:
X(k) and magnitude of X(k)

X[0] = 0.000926 + j 0.000000

magnitude of X(0) = 0.000926

X[1] = 0.000927 + j -0.000795

magnitude of X(1) = 0.001221

X[2] = -0.002035 + j 3.999998

magnitude of X(2) = 3.999999

X[3] = 0.000925 + j 0.000531

magnitude of X(3) = 0.001066

X[4] = 0.000924 + j -0.000373

magnitude of X(4) = 0.000997

X[5] = 0.000923 + j -0.001582

magnitude of X(5) = 0.001831

X[6] = -0.005004 + j -3.999995

magnitude of X(6) = 3.999999

X[7] = 0.000927 + j 0.001844

magnitude of X(7) = 0.002064

Calculation:
Formula:
nk
X (k)   M  j 2
e n
;0   N 1
x(n) k
k 0

Postlab Questions :

1. Mention the applications of DFT?

2. What is the drawback in Fourier transform and how it can be avoided?
3. Write the 4-point DFT matrix representation.
4. How many multiplications and additions are required to compute DFT for N=512?
5. Distinguish between DFT and DTFT.

Result:

Thus a program was written to perform Discrete Fourier Transform and executed using DSP processor.

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 Ex No: Date:
13. WAVEFORM GENERATION
Aim:
To write a program to generate Sine wave and Ramp wave form using TMS320C5416 DSP Processor.

Algorithm:

1. Start the program.

2. Declare the output buffer to store the result.
3. Assign 0 to accumulator
4. Increment the variable by one and store the value in the output buffer
5. After the counter is incremented to 40, then decrement the value of the variable one by one from
a negative going ramp
6. Again decrement the value from a negative going ramp in negative scale until the value
becomes maximum negative value.
7. Decrement the maximum negative value one by one until the value of the variable becomes zero.
Program:

; Program for Triangular Wave Generation

.global _main
.mmregs
.def start
.data
.text

_main

start STM #2000h,AR4

LD #0h,0,A

BEGIN STM #40,BRC

RPTB RAMP
STL A,*AR4+
LD #1h,0,B

RAMP ADD B,0,A

STM #40,BRC
RPTB RAMP1
STL A,*AR4+
LD #1h,0,B

RAMP1 SUB B,0,A

STM #40,BRC

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 RPTB RAMP2
STL A,*AR4+
LD #1h,0,B

RAMP2 SUB B,0,A

STM #40,BRC
RPTB RAMP3
STL A,*AR4+
LD #1h,0,B

RAMP3 ADD B,0,A

STL
A,*AR4+ LD
#1h,0,A SUB
#1h,0,A
BC BEGIN,ANEQ
.end

Model Graph:

Graph Property:

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Program:

//Program for Sine Wave Generation

#include <stdio.h>
#include<math.h>
#define FREQ 1000

float a[128];
main()
{
int i=0;

for(i=0;i<128;i++)
{
a[i]=sin(2*3.14*FREQ*i/24000);
printf("%f",a[i]);
}
}

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Model Graph:

Graph Property:

Result:
Thus the program was written to generate a Ramp signal, Sine signal and executed using DSP processor.

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Ex No: Date:
14. IIR AND FIR IMPLEMENTATION

Aim:
To write a program to implement Low pass filter using Finite Impulse Response in TMS320C5416 DSP
Processor.

Prerequisite Lab Questions:

1. What are the properties of FIR filter?
2. When cascade from realization is preferred in FIR filters?
3. What is the reason that FIR filter is always stable?
4. What are the desirable characteristics of the windows?

Description:
Input: Give the cutoff frequency and order of the filter

Output: 1. Find the ideal impulse response hd (n) of the filter order.
2. Find the window coefficients w (n) using the appropriate window
Function.
3. Find the filter coefficients h (n) =hd (n)*w (n).
4. Plot hd (n), w (n) and h (n).
Algorithm:

1. Start the program.

2. Get the cutoff frequency and order as an input...
3. Select a window function w (n).
4. Find the ideal impulse response of the filter hd (n).
5. Find the filter coefficients h (n) by multiplying hd (n) and w (n).
6. Execute the program, display the result and verify it.
7. Plot the graph for hd (n), w (n) and h (n) for the low pass filter.
8. Stop the process.

Program:

//Program for Fir

#include<stdio.h>
#include<math.h>
#define pi 3.14
int n,N;
float hd[10],alpha,w[10],h[10],wc;
main()
{
printf("enter N value");
scanf("%d",&N);
printf("enter wc value");

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 scanf("%f",&wc);
for(n=0;n<N;n++)
{
/*Hamming window function*/
w[n]=0.54-0.46*(cos((2*pi*n)/(N-1)));
printf("w[%d]= %f \n",n,w[n]);
}
alpha=(N-1)/2;
/* Impulse function */
for(n=0;n<N;n++)
{
if(n!=alpha)
hd[n]=sin(wc*(n-alpha))/(pi*(n-alpha));
else
hd[n]=wc/3.14;
printf("hd[%d]= %f\n",n,hd[n]);
}
/*Filter coefficients*/
for(n=0;n<N;n++)
{
h[n]=hd[n]*w[n];
printf("h[%d]= %f\n",n,h[n]);
}

Output:

enter N value11
enter wc value1.2

window coefficients:
w[0]= 0.080000 w[1]= 0.167766 w[2]= 0.397574 w[3]= 0.681730
w[4]= 0.911803 w[5]= 0.999999 w[6]= 0.912664 w[7]= 0.683123
w[8]= 0.398967 w[9]= 0.168629 w[10]= 0.080002

Ideal impulse response:

hd[0]= -0.017797 hd[1]= -0.079312 hd[2]= -0.046977 hd[3]= 0.107558
hd[4]= 0.296828 hd[5]= 0.382166 hd[6]= 0.296828 hd[7]= 0.107558
hd[8]= -0.046977 hd[9]= -0.079312 hd[10]= -0.017797

Low pass filter coefficients:

h[0]= -0.001424 h[1]= -0.013306 h[2]= -0.018677 h[3]= 0.073325
h[4]= 0.270648 h[5]= 0.382165 h[6]= 0.270904 h[7]= 0.073475
h[8]= -0.018742 h[9]= -0.013374 h[10]= 0.000025

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Model Graph:

Graph Property:

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Theoretical calculation:

1. hd (n)=1/2π ∫ Hd(w)ejwn dw
2. Select w (n)
3. Find h (n) = hd (n) w (n)

Rectangular Window

1,
w(n)  0  n  M 1
0, otherwise
Hanning Window
   2n 
0.5* 1 cos , 0  n  M 1
   M 1
w(n) 
0,
otherwise

Hamming Window

  2n 
0.54  0.46*
cos , 0  n  M 1

  M 1
2. w(n) 
0, otherwise


Postlab Questions:

1. What is the necessary and sufficient condition for linear phase characteristics in
FIR filter?
2. What is the principle of designing FIR filter using frequency sampling method?
3. What is the principle of designing FIR filter using frequency sampling method?
4. What are the disadvantages of' Fourier series method?

Result:

Thus a program was written to implement Low pass filter using Finite Impulse Response and executed
using DSP processor.
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Ex No: Date:

15. FINITE WORD LENGTH EFFECT

Aim:
To study the functions of finite word length effect in fixed point DSP systems..

Apparatus Required:
Hardware : Personal Computer & TMS320C67XX kit
Software : Code Composer Studio version4

Prerequisite Lab Questions:

1. What is the difference between rounding and tradeoff?
2. Define finite word length effect
3. What is Limit cycle oscillation?
4. What do you mean by dead band?
5. What is the need for scaling?

Program:
function ADCNoiseGain=ADCNoise(b,a,n,FM) [B,A] = sos2tf([b a]);
%form A(z) and B(z) [h,t] = impz(B,A,n);
ADCNoiseGain = sum(h.^2)/12.0;
fprintf('ADC noise gain is %f\n\n',ADCNoiseGain);
if FM~=1
fprintf('ADC noise is %g^2*%g*q^2\n',[FM ADCNoiseGain]);
else
fprintf('ADC noise is %g*q^2\n',ADCNoiseGain);
end function
CoeffQuantizeErr(b,a,maxbits,ftype,f,Fs)
%COEFFICIENT QUANTIZATION ERROR ANALYSIS
n=256;
for nbits=2:maxbits [B,A]=QuantizeCoeff(b,a,nbits);
[B,A] = sos2tf([B A]);
[h,w] = freqz(B,A,n);
amag = abs(h);
amag = amag/max(amag); fprintf('\n\nnbits\tstage\tA1\tA2\tradius1\
tangle1\tradiu s2\tangle2\n'); for nbits=2:maxbits
[B,A]=QuantizeCoeff(b,a,nbits);
for i=1:size(b,1) r1 = sqrt(abs(A(i,3)));
angle1 = 180/pi*acos(A(i,2)/(-2.0*r1));
r2 = sqrt(abs(a(i,3)));
angle2 = 180/pi*acos(a(i,2)/(-2.0*r2));
fprintf('%d\t%d\t%-7.4f\t%-7.4f\t%-7.4f\t%-7.2f\t%- 7.4f\t%-7.2f\n', nbits, i,A(i,2) , A(i,3),
r1,angle1,r2,angle2);
end
end format;

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 function ScaleFactor(b,a,nstep,size,structure) norm0 = DirectScale(b,a,0,nstep);
norm1 = DirectScale(b,a,1,nstep);
norm2 = DirectScale(b,a,2,size);
else norm0 = CanonicScale(b,a,0,nstep);
norm1 = CanonicScale(b,a,1,nstep);
norm2 = CanonicScale(b,a,2,size);
end disp('L1-norms of the second order sections:');
disp(norm0); disp('L2-norms of the second order sections:');
disp(norm1); disp('Loo-norms of the second order sections:');
disp(norm2);
function s = DirectScale(b,a,iopt,n)
if(iopt>=3) s = ones(1,size(b,1));
return;
else
A = 1; B = 1;
for i=1:size(b,1)
%loop for each
stage A =
conv(A,a(i,:));
B = conv(B,b(i,:));
s(i) = GetScaleFactor(B,A,iopt,n);
end
end function
s = CanonicScale(b,a,iopt,n)
if(iopt>=3)
s = ones(1,size(b,1));
return;
else
A = 1;
B = 1;
for i=1:size(b,1)
A = conv(A,a(i,:));
if i>1 B = conv(B,b(i-1,:));
end
s(i) = GetScaleFactor(B,A,iopt,n);
end
end

Postlab Questions:
1. What are the advantages of floating point arithmetic?
2. What are the methods to prevent overflow?
3. What are the two kinds of limit cycles?
4. What is the relationship between truncation error e and the bits b for representing a decimal
into binary?
5. What are the three quantization errors to finite word length registers in digital filters?

Result:
Thus the function of finite word length effect in fixed point DSP processor is studied.
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Ex No: Date:

16. INTRODUCTION TO IMAGE PROCESSING TOOLBOX

Aim:
To study the basic operations in image processing using MATLAB program.

Prerequisite Lab Questions:

1. Define image processing.
2. What is a digital image?
3. List the applications of image processing techniques.
4. How will you load and display an image?
5. How will you define 2-D DFT of an image?

Description:
Input: Give input images from MATLAB edit window.

Output: Perform various image processing techniques given and display it in different Figure
window.

Algorithm:

Load an image in edit window and perform the following operations:

1. Reading and displaying an image
2. Improving the contrast of an image
3. Addition of images
4. Displaying an color image
5. Converting a color image to Gray scale image

Program:

% PROGRAM FOR READING AND DISPALYING AN IMAGE

clc;
clear all;
close all;
I = imread('pout.tif');
figure, imshow(I)

% PROGRAM FOR IMPROVING THE CONTRAST OF AN IMAGE

figure, imhist(I)
I2 = histeq(I);
figure, imshow(I2)

%PROGRAM FOR ADDING THE IMAGES

I = imread('rice.png');
J = imread('cameraman.tif');

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 K = imadd(I,J);
figure, imshow(K)
% PROGRAM FOR CONVERTING COLOR IMAGE TO GRAY SCALE
I = imread('board.tif');
J = rgb2gray(I);
figure, imshow(I), figure, imshow(J);

Sample Output

ORIGINAL IMAGE

IMAGE HISTOGRAM
1600

1400

1200

1000

800

600

400

200

0 50 100 150 200 250

HISTOGRAM EQUALIZED IMAGE ADDITION OF IMAGES

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 COLOR IMAGE GRAY SCALE IMAGE

Postlab Questions:

1. Mention the applications of image processing techniques.

2. How will you improve the contrast of an image?
3. What are the various mathematical operations done in an image?
4. Define spatial and frequency domain?
5. How to perform 2-D DFT of an image?

Result:

Thus a MATLAB program for studying the image processing tool box was written, executed and the
graphs were plotted.

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Ex No: Date:

17. GENERATION AND DETECTION OF DTMF SIGNAL USING DFT

Aim:
To study the applications of Dual Tone Multi- Frequency (DTMF) signals in telephones and to generate
and visualize the DTMF signals using MATLAB program.

Prerequisite Lab Questions:

1. What are the applications of DTMF signals?
2. List the various low and high frequency signals used for DTMF generation.
3. How DFT is used for DTMF generation?
4. Draw the basic symbols in a telephone key pad.
5. What do you mean by DTMF generation and detection?

Description:
Input: Generate sine input signal for low and high frequencies.
Output:
i. Represent the generated low and high frequency signals using DFT.
ii. Generate the Matlab code for the generation and visualization of the DTMF tones

Algorithm:
1. Start the program.
2. Generate the sinusoidal signal with low frequency group f = 697 Hz, 770 Hz, 852 Hz, 941 Hz
and high frequencies h= 1209 Hz, 1336 Hz, 1477Hz.
3. Set the sampling frequency 8 KHz, N = 800, No of samples at a spacing of 100ms.
4. Generate and visualize the DFT of the corresponding low and high frequency signal.
5. Stop the program.

Theory:

Dual-tone Multi-Frequency (DTMF) signaling is the basis for voice communications control and is
widely used worldwide in modern telephony to dial numbers and configure switchboards. It is also used in
systems such as in voice mail, electronic mail and telephone banking.
Generating DTMF Tones. A DTMF signal consists of the sum of two sinusoids - or tones - with frequencies
taken from two mutually exclusive groups. These frequencies were chosen to prevent any harmonics from being
incorrectly detected by the receiver as some other DTMF frequency. Each pair of tones contains one frequency
of the low group (697 Hz, 770 Hz, 852 Hz, 941 Hz) and one frequency of the high group (1209 Hz, 1336 Hz,
1477Hz) and represents a unique symbol. The frequencies allocated to the push-buttons of the telephone pad are
shown below:

Matlab Program
i. Generation of the twelve frequency pairs

Clear all;
Close all;
symbol = {'1','2','3','4','5','6','7','8','9','*','0','#'};

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 lfg = [697 770 852 941]; % Low frequency group
hfg = [1209 1336 1477]; % High frequency group
f = [];
for c=1:4,
for r=1:3,
f = [ f [lfg(c);hfg(r)] ];
end
end

ans =

697 1209
697 1336
697 1477
770 1209
770 1336
770 1477
852 1209
852 1336
852 1477
941 1209
941 1336
941 1477

Matlab code for the generation and visualization of the DTMF tones

Fs = 8000; % Sampling frequency 8 kHz

N = 800; % Tones of 100 ms
t = (0:N-1)/Fs; % 800 samples at Fs
pit = 2*pi*t;

tones = zeros(N,size(f,2));
for toneChoice=1:12,
% Generate tone
tones(:,toneChoice) = sum(sin(f(:,toneChoice)*pit))';
% Plot tone
subplot(4,3,toneChoice),plot(t*1e3,tones(:,toneChoice));
title(['Symbol "', symbol{toneChoice},'":
[',num2str(f(1,toneChoice)),',',num2str(f(2,toneChoice)),']'])
set(gca, 'XLim', [0 25]);
ylabel('Amplitude');
if toneChoice>9, xlabel('Time (ms)'); end
end
set(gcf, 'Color', [1 1 1], 'Position', [1 1 1280 1024])
annotation(gcf,'textbox', 'Position',[0.38 0.96 0.45 0.026],...
'EdgeColor',[1 1 1],...
'String', '\bf Time response of each tone of the telephone pad', ...
'FitBoxToText','on');

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MODEL GRAPH
Time response of each tone of the telephone pad

Symbol "1": [697,1209]

2 Symbol "2": [697,1336] Symbol "3": [697,1477]
2 2
Amplitu

Amplitu

Amplitu
0 0 0

-2
0 5 10 15 20 -2 -2
25 0 5 10 15 20 0 5 10 15 20 25
25

Symbol "4": [770,1209]

2 Symbol "5": [770,1336] Symbol "6": [770,1477]
2 2
Amplitu

Amplitu

Amplitu
0 0 0

-2
0 5 10 15 20 -2 -2
25 0 5 10 15 20 0 5 10 15 20 25
25

Symbol "7": [852,1209]

2 Symbol "8": [852,1336] Symbol "9": [852,1477]
2 2
Amplitu

Amplitu

Amplitu
0 0 0

-2
0 5 10 15 20 -2 -2
25 0 5 10 15 20 0 5 10 15 20 25
25

Symbol "*": [941,1209]

2 Symbol "0": [941,1336] Symbol "#": [941,1477]
2 2
Amplitu

Amplitu

Amplitu

0 0 0

-2 -2
0 5 10 15 20 25 0 5 -2
10 15 20 25 0 5 10 15 20 25
Time (ms) Time (ms) Time (ms)

Postlab Questions:
1. What are the applications of DSP systems?
2. List the procedure for generation of DTMF signals.
3. Define Goertzel algorithm for DFT estimation.
4. What are the types of multiple access in communication systems?
5. Define OFDM.

Result:

Thus a program to generate and visualize DTMF tones for telephone communication using Matlab was
written and executed.
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