DSP LAB Merged

Dr.K.V.
SUBBA REDDY COLLEGE OF ENGINEERING

FOR WOMEN
(Approved by AICTE, Affiliated to JNTUA)
NH -44, Opposite Dupadu Railway Station , Kurnool-518218
LABORATORY MANUAL OF
DIGITAL SIGNAL PROCESSING LAB

For
III.B.Tech - II Sem
Prepared by
K PRASAD BABU M.Tech (Ph.D)
Associate Professor
SYED NOORULLAH M.Tech
Assistant Professor
DEPARTMENT OF
ELECTRONICS & COMMUNICATIONS ENGINEERING
INDEX
S.No List Of Experiments Page No

1 Generation Of Various Signals & Sequences 1
2 Find Average Power And Energy Of A Given Signal 10
3 Convolution And Correlation Of Signals 12
4 DTFT Of A Given Signal 18
5 N-Point FFT Algorithm 21
Design Of FIR Filter Using Windowing Technique And Verify The
6 Frequency Response Of The Filter 23
Design Of IIR Filter Using Any Of The Available Methods And Verify The
7 Frequency Response Of The Filter 28
8 Generation Of Sine Wave And Square Wave 31

9 Convolution 33
10 N-Point Fast Fourier Transform 37
11 FIR Filter Design 39
12 IIR Filter Design 41
Add –on Experiments
1 Implementation of Decimation Process 42
2 Implementation of Interpolation Process 44
DSP Lab Manual
1. GENERATION OF VARIOUS SIGNALS & SEQUENCES

AIM: To generate random Signal and plot the same as waveform showing all the
Specification.
APPARATUS: Computer
MATLAB 7.8.0.347(R2009a) Software.
PROCEDURE:
1. Open the MATLAB software by double clicking the ICON on desktop.
2. Open the new M-file by using file menu
3. Write the program in new file
4. Click on save and run the icon
5. Perform error check which displayed on command window
6. Plot the waveforms which displays on figure window.
7. Note down the values, which displays on the work space.
PROGRAM:
t=-10:0.1:10
%--------------------Unit Impulse---------------------%
x1=1;
x2=0;
x =x1.*(t==0) %generate Unit Impulse Signal
figure(1), stem(t,x); % plot the generated Unit Impulse Signal
xlabel('t'); ylabel('x(t)'); title('Unit impulse Signal');
%------------------ Unit Step signal------------------%
x1 =1;
x2 = 0;
x = x1.*(t>=0)+x2.*(t<0); %generate Unit Step Signal
figure(2); stem(t,x); %Plot the generated unit step signal
xlabel('t'); ylabel('x(t)'); title('Unit Step Signal');
%---------------------- Ramp Signal------------------%
x1 = t;
x2 = 0;
x = x1.*(t>=0)+x2.*(t<0); %generate unit ramp signal
figure(3);stem(t,x); % plot the generated ramp signal
xlabel('t'); ylabel('x(t)'); title(' Ramp Signal');
%----------------------rectangular pulse------------%
t1=-10:0.01:10
a7=rectpuls(t1/10)
figure(4),stem(t1,a7), grid on,axis([-10 10 -2 2])
xlabel('time')
ylabel('magnitude')
title('rectangular pulse')
%-------------------- triangular pulse---------------%
t2=-10:0.1:10
Dr K V Subba Reddy College Engineering For Women Page 1

DSP Lab Manual
a8=tripuls(t2)
figure(5),stem(t2,a8), grid on
title('triangular pulse')
xlabel('time')
ylabel('magnitude')
%--------------- Sinc Signal----------------------%
x = sinc(t); %generate sinc signal
figure(6);stem(t,x); % plot the generated sinc signal
xlabel('t'); ylabel('x(t)'); title('sinc Signal');
%-----------------Sinusoidal Signal----------------%
T = 2; %declare time period
F = 1/T; %compute frequency
x = sin(2*pi*F*t); %generate sinusoidal signal
figure(7);stem(t,x); % plot the generated sinusoidal signal
xlabel('t'); ylabel('x(t)'); title('sinusoidal Signal');
%---------------- Sawtooth signal-----------------%
fs = 10000;
t = 0:1/fs:1.5;
x = sawtooth(2*pi*50*t); %generate sawtooth
signal
figure(8);stem(t,x); axis([0 0.2 -1 1]); %plot the generated sawtooth
signal
xlabel('t'); ylabel('x(t)'); title('sawtooth Signal');

%----------------- square signal---------------------%
t=(0:0.001:10);
sq_wave=square(4*pi*t); %generate square signal
figure(9); stem(t,sq_wave); axis([0 5 -2 2]) %plot the
generated square signal
xlabel('t'); ylabel('x(t)'); title('square signal');
%---------------------periodic triangle------------%
T=20;
x=linspace(0,5*T,10000);
y1=min(0.5,mod(x,T)/T);
y2=min(0.5,1-mod(x,T)/T);
figure(10),stem(x,y1+y2),grid on
xlabel('time')
ylabel('magnitude')
title('periodic triangle wave')
tc = gauspuls('cutoff',50e3,0.6,[],-40);
t = -tc : 1e-6 : tc;
yi = gauspuls(t,50e3,0.6);
figure(11),stem(t,yi),grid on
xlabel('time')
ylabel('magnitude')
title('Aperiodic triangle wave')
%---------------------Random Bipolar Sequence ------------%

%Nb is the number of bits to be transmitted
Nb=10;
% Generate Nb bits randomly
b=rand(1,Nb)>0.5;
%Rb is the bit rate in bits/second
Rb=10000;
fs=50; % For T=0.2s fs =50Hz%

NRZ_out=[];

DSP Lab Manual
%Vp is the peak voltage +v of the waveform
Vp=1;
%Here we encode input bitstream as Bipolar NRZ-L waveform
for index=1:size(b,2)
if b(index)==1
NRZ_out=[NRZ_out [1 1 1 1 1 1 1 1 1 1]*Vp];
elseif b(index)==0
NRZ_out=[NRZ_out [1 1 1 1 1 1 1 1 1 1]*(-Vp)];
end
end
figure(12);
stem((1:Nb*10)/10,NRZ_out);
xlabel('bits-->');
ylabel('Amplitude (volts)-->');
title(' Bipolar Random data for 10kbps NRZ-L encoded bit stream');
OUTPUT:
Unit impulse Signal
1
0.9
0.8
0.7
0.6
x(t)
0.5
0.4
0.3
0.2
0.1
0
-10 -8 -6 -4 -2 0 2 4 6 8 10
t

DSP Lab Manual
Unit Step Signal

1
0.9
0.8
0.7
0.6
x(t)
0.5
0.4
0.3
0.2
0.1
0
-10 -8 -6 -4 -2 0 2 4 6 8 10
t
Ramp Signal
10
6
x(t)
0
-10 -8 -6 -4 -2 0 2 4 6 8 10
t

DSP Lab Manual
rectangular pulse
2
1.5
0.5
magnitude
-0.5
-1
-1.5
-2
-10 -8 -6 -4 -2 0 2 4 6 8 10
time
triangular pulse
1
0.9
0.8
0.7
0.6
magnitude
0.5
0.4
0.3
0.2
0.1
0
-10 -8 -6 -4 -2 0 2 4 6 8 10
time

DSP Lab Manual
sinc Signal
1
0.8
0.6
0.4
x(t)
0.2
-0.2
-0.4
-10 -8 -6 -4 -2 0 2 4 6 8 10
t
sinusoidal Signal
1
0.8
0.6
0.4
0.2
x(t)
-0.2
-0.4
-0.6
-0.8
-1
-10 -8 -6 -4 -2 0 2 4 6 8 10
t

DSP Lab Manual
sawtooth Signal
1
0.8
0.6
0.4
0.2
x(t)
-0.2
-0.4
-0.6
-0.8
-1
0 0.02 0.04 0.06 0.08 0.1 0.12 0.14 0.16 0.18 0.2
t
square signal
2
1.5
0.5
x(t)
-0.5
-1
-1.5
-2
0 0.5 1 1.5 2 2.5 3 3.5 4 4.5 5
t

DSP Lab Manual
periodic triangle wave

1
0.9
0.8
0.7
0.6
magnitude
0.5
0.4
0.3
0.2
0.1
0
0 10 20 30 40 50 60 70 80 90 100
time
Aperiodic triangle wave

1
0.8
0.6
0.4
magnitude
0.2
-0.2
-0.4
-0.6
-0.8
-4 -3 -2 -1 0 1 2 3 4
time -5
x 10

DSP Lab Manual
Bipolar Random data for 10kbps NRZ-L encoded bit stream

1
0.8
0.6
0.4
Amplitude (volts)-->
0.2
-0.2
-0.4
-0.6
-0.8
-1
0 1 2 3 4 5 6 7 8 9 10
bits-->
RESULT: Generations of various sequence and random sequence & plot the same as
waveform showing all the Specification is done using MATLAB.

DSP Lab Manual
2. FIND AVERAGE POWER AND ENERGY OF A GIVEN SIGNAL

AIM: To Compute the Average power and energy of the given signal.
APPARATUS: Computer
PROCEDURE:
PROGRAM:
%-------------------computation of energy and power---------------%
t=-10:0.1:10;
x=10*sin(10*pi*t);To=10
xsq=x.^2;
E = trapz(t,xsq);
P = E/(2*To);
disp([' Energy, E = ', num2str(E),' Joules']);
disp([' Power, P = ', num2str(P),' Watts']);

DSP Lab Manual
RESULT: Computed the Average power and energy of the given signal done using
MATLAB.

DSP Lab Manual
3. CONVOLUTION AND CORRELATION OF SIGNALS

AIM: To Perform Convolution and Correlation (auto and cross correlation) of discrete
sequences without using built in functions for convolution and correlation operations
APPARATUS: Computer
PROCEDURE:
PROGRAM: Linear Convolution

% MATLAB program for linear convolution
%linear convolution program
clc;
clear all;
close all;
disp('linear convolution program');
x=input('enter the input secquence x(n):');
m=length(x);
h=input('enter the impulse response h(n):');
n=length(h);
x=[x,zeros(1,n)];
subplot(2,2,1), stem(x);
title('input sequence x(n)is:');
xlabel('---->n');
ylabel('---->x(n)');grid;
h=[h,zeros(1,m)];
subplot(2,2,2), stem(h);
title('impulse response h(n)is:');
xlabel('---->n');
ylabel('---->h(n)');grid;
disp('convolution of x(n) & h(n) is y(n):');
y=zeros(1,m+n-1);
for i=1:m+n-1
y(i)=0;
for j=1:m+n-1
if(j<i+1)
y(i)=y(i)+x(j)*h(i-j+1);
end
end
end

DSP Lab Manual
subplot(2,2,[3,4]),stem(y);
title('convolution of x(n) & h(n) is :');
xlabel('---->n');
ylabel('---->y(n)');grid;
PROGRAM: Circular Convolution

clc;
clear all;
close all;
x=input ('Enter the input sequence :');
h=input ('Enter the impulse response :');
subplot(3, 1,1);
stem(x);
title('input sequence');
xlabel('time----->');
ylabel('amplitude------>');
subplot(3, 1,2);
stem(h);
title('impulse response');
xlabel('time----->');
n1=length(x);
n2=length(h);
n3=max(n1,n2);
n4=n1-n2;
%======================================================
%Appending zeros if lengths of the sequences are not equal

if(n4>0)
h=[h,zeros(1,n4)];
else
x=[x,zeros(1,-n4)];
end
%======================================================
%======================================================
%Loop for circular convolution
for(n=1:n3);
y(n)=0;
for(i=1:n3);
j=n-i+1;
if(j<=0)
j=j+n3;
end
y(n)=y(n)+x(i)*h(j);
end
end
%======================================================
disp('circ convolution of given sequences is….');
disp(y);
subplot(3,1,3);
stem(y);
title('output sequence(circular convolution)');
xlabel('n----->');

DSP Lab Manual
PROGRAM: AutoCorrelation.
clc;
clear all;
close all;
x=input ('Enter the input sequence :');
disp(x);
stem(x);figure(1),subplot(2,2,1),
title('input samples');
xlabel('samples----->');
y = xcorr2(x);
disp('Auto correlation of the sequences is….');
disp(x);
stem(x);
title('input samples');
subplot(2,2,2),
disp(y);
stem(y);
title('Auto correlation');
PROGRAM: Cross Correlation.

clc;
clear all;
close all;
%% Inputs
% You can specify the inputs or can take them through command window
h= input ('enter the sequence h');
x= input ('enter the sequence x');
pp=h;qq=x;
% Plot the inputs
subplot(3,1,1); stem(h,'m'); title ('h');ylabel('amplitude')
% %
subplot(3,1,2); stem(x,'b'); title ('x');ylabel('amplitude')
%% calculate the cross correlation
l1= length (h); % calculate length of sequence h
l2= length (x); % calculate length of sequence x
l=abs (l1-l2); % calculate difference in lengths of sequence
if (l1 > l2)
x= [x zeros(1,l)];
else if(l2 > l)
h= [h zeros(1,l)];
end
end
h= [h zeros(1, max (l1, l2))];
for shift= 0:max(l1,l2);
new_x = [zeros(1, shift) x zeros(1, (max (l1 , l2))-shift)];
y(shift+1,:)= sum(h.* new_x);
end
y=y';
% Display the correlation sum
subplot(3,1,1); stem(pp,'m'); title ('h');ylabel('amplitude')
subplot(3,1,2); stem(qq,'b'); title ('x');ylabel('amplitude')
subplot(3,1,3); stem(y,'r'); title ('cross correlation'); xlabel('time');

DSP Lab Manual
OUTPUT:
linear convolution program

enter the input secquence x(n):[1 2 3 4 5]
enter the impulse response h(n):[2 1 3 4 5]
convolution of x(n) & h(n) is y(n):
input sequence x(n)is: impulse response h(n)is:
6 6
4 4
---->x(n)
---->h(n)
2 2
0 0
0 5 10 0 5 10
---->n ---->n
convolution of x(n) & h(n) is :
50
40
---->y(n)
30
20
10
0
1 2 3 4 5 6 7 8 9
---->n

DSP Lab Manual
Enter the input sequence :[1 2 3 4 5]

Enter the impulse response :[1 3 5 7 9]
circ convolution of given sequences is….
75 85 85 75 55
input sequence
amplitude------>
0
1 1.5 2 2.5 3 3.5 4 4.5 5
time----->
impulse response
amplitude------>
10
0
1 1.5 2 2.5 3 3.5 4 4.5 5
time----->
output sequence(circular convolution)
amplitude------>
100
50
0
1 1.5 2 2.5 3 3.5 4 4.5 5
n----->

DSP Lab Manual
Auto Correlation
Enter the input sequence :[1 2 3 4 5 6]
1 2 3 4 5 6
Auto correlation of the sequences is….
1 2 3 4 5 6
6 17 32 50 70 91 70 50 32 17 6
amplitude------> input samples amplitude------> Auto correlation
6 100
4
50
2
0 0
0 2 4 6 0 5 10 15
samples-----> samples----->
Cross Correlation
enter the sequence h[1 3 5 7 9]
enter the sequence x[1 2 3 4 5]
h
10
amplitude
0
1 1.5 2 2.5 3 3.5 4 4.5 5
x
5
amplitude
0
1 1.5 2 2.5 3 3.5 4 4.5 5
cross correlation
100
50
0
1 1.5 2 2.5 3 3.5 4 4.5 5 5.5 6
time
RESULT: Performed Convolution and Correlation (auto and cross correlation) of discrete
sequences without using built in functions for convolution and correlation operations done
using MATLAB.

DSP Lab Manual
4. DTFT OF A GIVEN SIGNAL

AIM: To Perform the DTFT of given sequence using MATLAB.
APPARATUS: Computer
PROCEDURE:
PROGRAM:
% Write a MATLAB program to find the DTFT of a given sequence.
x=input('Enter the sequence : ')
N=length(x)
n=0:N-1
K=0:N-1
wn=exp(-j*2*pi/N)
nK=n'*K
wNnK=wn.^nK
df=x*wNnK
subplot(3,1,1)
stem(x,abs(df))
title('DTFT')
xlabel('K')
ylabel('Magnitude')
subplot(3,1,2)
stem(x,angle(df))
title('DTFT')
xlabel('K')
OUTPUT:
Enter the sequence : [1 2 3 4]
x= 1 2 3 4
N= 4
n= 0 1 2 3
K=
0 1 2 3
wn =
3.1328e-008

DSP Lab Manual
nK =
0 0 0 0
0 1 2 3
0 2 4 6
0 3 6 9
wNnK =
1.0000 1.0000 1.0000 1.0000
1.0000 0.0000 0.0000 0.0000
1.0000 0.0000 0.0000 0.0000
1.0000 0.0000 0.0000 0.0000
df =
10.0000 1.0000 1.0000 1.0000

wn =
3.1921e+007
nK =
0 0 0 0
0 1 2 3
0 2 4 6
0 3 6 9
wNnK =
1.0e+067 *
0.0000 0.0000 0.0000 0.0000
0.0000 0.0000 0.0000 0.0000
0.0000 0.0000 0.0000 0.0000
0.0000 0.0000 0.0000 3.4406
inversdf =
1.0e+067 *
0.0000 0.0000 0.0000 3.4406

DSP Lab Manual
DTFT
Magnitude 10
0
1 1.5 2 2.5 3 3.5 4
K
DTFT
1
phase
-1
1 1.5 2 2.5 3 3.5 4
K
67
x 10 Inverse DTFT
4
Magnitude
0
1 1.5 2 2.5 3 3.5 4
N
RESULT: Performed the DTFT of given sequence done using MATLAB.

DSP Lab Manual
5. N-POINT FFT ALGORITHM

AIM: To Perform N-Point FFT algorithm sequence using MATLAB.
APPARATUS: Computer
PROCEDURE:
PROGRAM:
%fast fourier transform
clc;
clear all;
close all;
tic;
x=input('enter the sequence');
n=input('enter the length of fft');
%compute fft
disp('fourier transformed signal');
X=fft(x,n)
subplot(1,2,1);stem(x);
title('i/p signal');
xlabel('n --->');
ylabel('x(n) -->');grid;
subplot(1,2,2);stem(X);
title('fft of i/p x(n) is:');
xlabel('Real axis --->');
ylabel('Imaginary axis -->');grid;
OUTPUT:
enter the sequence [1 2 3 4 4 3 2 1]
enter the length of fft8
fourier transformed signal
X=
Columns 1 through 5

DSP Lab Manual
20.0000 -5.8284 - 2.4142i 0 -0.1716 - 0.4142i 0

Columns 6 through 8
-0.1716 + 0.4142i 0 -5.8284 + 2.4142i
i/p signal fft of i/p x(n) is:
4 2.5
2
3.5
1.5
3
1
Imaginary axis -->

2.5
0.5
x(n) -->
2 0
-0.5
1.5
-1
1
-1.5
0.5
-2
0 -2.5
0 2 4 6 8 0 2 4 6 8
n ---> Real axis --->
RESULT: Performed the FFT of given sequence done using MATLAB.

DSP Lab Manual
6. DESIGN OF FIR FILTER USING WINDOWING TECHNIQUE

AND VERIFY THE FREQUENCY RESPONSE OF THE FILTER
AIM: To Perform Design of FIR filter using Windowing technique and verify the
frequency response of the filter
APPARATUS: Computer
PROCEDURE:
PROGRAM:
%fir filt design window techniques
clc;
clear all;
close all;
rp=input('enter passband ripple');
rs=input('enter the stopband ripple');
fp=input('enter passband freq');
fs=input('enter stopband freq');
f=input('enter sampling freq ');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(rp*rs))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n1=n+1;
if(rem(n,2)~=0)
n1=n;
n=n-1;
end
c=input('enter your choice of window function 1. rectangular 2. triangular
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');

DSP Lab Manual
end
if(c==3)
y=kaiser(n1);
disp('kaiser window filter response');
end
%LPF
b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);plot(o/pi,m);
title('LPF');
ylabel('Gain in dB-->');
xlabel('(a) Normalized frequency-->');
%HPF
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
title('HPF');
xlabel('(b) Normalized frequency-->');
%BPF
wn=[wp ws];
b=fir1(n,wn,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
title('BPF');
xlabel('(c) Normalized frequency-->');
%BSF
b=fir1(n,wn,'stop',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
title('BSF');
xlabel('(d) Normalized frequency-->');
OUTPUT:
enter passband ripple0.02
enter the stopband ripple0.01
enter passband freq1000
enter stopband freq1500
enter sampling freq 10000
enter your choice of window function 1. rectangular 2. triangular 3.kaiser:
1
Rectangular window filter response
>>

DSP Lab Manual
LPF HPF
50 20
0
Gain in dB-->
Gain in dB-->
0
-20
-50
-40
-100 -60
0 0.5 1 0 0.5 1
(a) Normalized frequency--> (b) Normalized frequency-->
BPF BSF
50 20
Gain in dB-->
0 Gain in dB--> 0
-50 -20
-100 -40
0 0.5 1 0 0.5 1
(c) Normalized frequency--> (d) Normalized frequency-->

2
Triangular window filter response
>>

DSP Lab Manual
LPF HPF
Gain in dB--> 0 0
Gain in dB-->
-20 -10
-40 -20
-60 -30
0 0.5 1 0 0.5 1
BPF BSF
20 5
0 0
Gain in dB-->
Gain in dB-->
-20 -5
-40 -10
-60 -15
0 0.5 1 0 0.5 1

3
kaiser window filter response
>>

DSP Lab Manual
LPF HPF
50 20
0
Gain in dB-->
Gain in dB-->
0
-20
-50
-40
-100 -60
0 0.5 1 0 0.5 1
BPF BSF
50 50
Gain in dB-->
0 Gain in dB--> 0
-50 -50
-100 -100
0 0.5 1 0 0.5 1
RESULT: Design of FIR filter using Windowing technique and verify the frequency response
of the filter done using MATLAB

DSP Lab Manual
7. DESIGN OF IIR FILTER USING ANY OF THE AVAILABLE

METHODS AND VERIFY THE FREQUENCY RESPONSE OF THE
FILTER
AIM: To Design of IIR filter using any of the available methods and Verify the frequency
response of the filter
APPARATUS: Computer
PROCEDURE:
PROGRAM:
% IIR filters LPF & HPF
clc;
clear all;
close all;
disp('enter the IIR filter design specifications');
rp=input('enter the passband ripple');
rs=input('enter the stopband ripple');
wp=input('enter the passband freq');
ws=input('enter the stopband freq');
fs=input('enter the sampling freq');
w1=2*wp/fs;w2=2*ws/fs;
[n,wn]=buttord(w1,w2,rp,rs,'s');
c=input('enter choice of filter 1. LPF 2. HPF \n ');
if(c==1)
disp('Frequency response of IIR LPF is:');
[b,a]=butter(n,wn,'low','s');
end
if(c==2)
disp('Frequency response of IIR HPF is:');
[b,a]=butter(n,wn,'high','s');
end
w=0:.01:pi;
[h,om]=freqs(b,a,w);
m=20*log10(abs(h));
an=angle(h);
figure,subplot(2,1,1);plot(om/pi,m);

DSP Lab Manual
title('magnitude response of IIR filter is:');
xlabel('(a) Normalized freq. -->');
subplot(2,1,2);plot(om/pi,an);
title('phase response of IIR filter is:');
xlabel('(b) Normalized freq. -->');
ylabel('Phase in radians-->');
OUTPUT
enter the IIR filter design specifications
enter the passband ripple0.15
enter the stopband ripple60
enter the passband freq1500
enter the stopband freq3000
enter the sampling freq7000
enter choice of filter 1. LPF 2. HPF
1
Frequency response of IIR LPF is:
>>
magnitude response of IIR filter is:

0
Gain 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
(a) Normalized freq. -->
phase response of IIR filter is:
4
Phase in radians-->
-2
-4
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
(b) Normalized freq. -->
enter the IIR filter design specifications

enter the passband ripple0.15
enter the stopband ripple60
enter the passband freq1500
enter the stopband freq3000
enter the sampling freq7000
enter choice of filter 1. LPF 2. HPF

DSP Lab Manual
2
Frequency response of IIR HPF is:
>>
magnitude response of IIR filter is:

200
0
Gain in dB-->
-200
-400
-600
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
(a) Normalized freq. -->
phase response of IIR filter is:
4
Phase in radians-->
-2
-4
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
(b) Normalized freq. -->
RESULT: Design of IIR filter using any of the available methods and verify the frequency
response of the filter done using MATLAB.

DSP Lab Manual
1) GENERATION OF SINE WAVE AND SQUARE WAVE
AIM:- To generate a sine wave and square wave using C6713 simulator
EQUIPMENTS:-
Operating System - Windows XP

Software - CC STUDIO 3
DSK 6713 DSP Trainer kit.
USB Cable
Power supply
PROCEDURE:-
1. Open Code Composer Setup and select C6713 simulator, click save and quit
2. Start a new project using „Project-> New‟ pull down menu, save it in a separate
directory (C:\My projects) with file name sinewave.pjt
3. Create a new source file using File->New->Source file menu and save it in the project
folder(sinewave.c)
4. Add the source file (sinewave.c) to the project
ProjectAdd files to Project Select sinewave.c
5. Add the linker command file hello.cmd
Project -> Add files to Project (path: C:\CCstudio\tutorial\dsk6713\hello\hello.cmd)
6. Add the run time support library file rts6700.lib
ProjectAdd files to Project (path: C\CCStudio\cgtools\lib\rts6700.lib)
7. Compile the program using „projectCompile‟ menu or by Ctrl+F7
8. Build the program using „projectBuild‟ menu or by F7
9. Load the sinewave.out file (from project folder lcconv\Debug) using FileLoad Program
10. Run the program using „DebugRun or F5
11. To view the output graphically Select ViewGraphTime and Frequency
12. Repeat the steps 2 to 11 for square wave
PROGRAM:
#include <stdio.h>
#include <math.h>
float a[500];
void main()
{
int i=0;
for(i=0;i<500;i++)
{
a[i]=sin(2*3.14*10000*i);
}
}
PROGRAM: (b)
#include <stdio.h>
#include <math.h>

DSP Lab Manual
int a[1000];
void main()
{
int i,j=0;
int b=5;
for(i=0;i<10;i++)
{
for (j=0;j<=50;j++)
{
a[(50*i)+j]=b;
}
b=b*(-1) ;
}
}
OUTPUT:
RESULT: The sine wave and square wave has been obtained using C6713 simulator

DSP Lab Manual
2. LINEAR CONVOLUTION
AIM: To verify Linear Convolution using C6713 simulator
EQUIPMENTS:-

USB Cable
Power supply
PROCEDURE:
1. Open Code Composer Setup and select C6713 simulator, click save and quit
2. Start a new project using Project New pull down menu, save it in a separate directory
(C:\My projects) with file name linearconv.pjt
3. Create a new source file using File New Source file menu and save it in the project
folder (linearconv.c)
4. Add the source file (linearconv.c) to the project
Project Add files to Project Select linearconv.c
5. Add the linker command file hello.cmd
Project Add files to Project
(path: C:\CCstudio\tutorial\dsk6713\hello\hello.cmd)
6. Add the run time support library file rts6700.lib
Project Add files to Project
(Path: C\CCStudio\cgtools\lib\rts6700.lib)
7. Compile the program using project Compile menu or by Ctrl+F7
8. Build the program using project  Build menu or by F7
9. Load the linearconv.out file (from project folder impulse response\Debug) using
File Load Program
10. Run the program using „Debug Run or F5
11. To view the output graphically
Select ViewGraph Time and Frequency
12. Observe the values in the output window.
PROGRAM:
// Linear convolution program in c language using CC Studio

#include<stdio.h>
int x[15],h[15],y[15];
main()
{
int i,j,m,n;
DSP Lab Manual
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]);
}
OUTPUT:-
enter value for m 4

enter value for n 4
Enter values for i/p 1234
Enter Values for n 1234
The Value of output y[0]=1

RESULT: Thus linear convolution of two sequences using C6713 simulator

DSP Lab Manual
CIRCULAR CONVOLUTION
AIM : To write a program in ‘C’ to compute circular convolution between two discrete time
sequences.
APPARATUS:
 Operating System - Windows XP

 Software - CC STUDIO 3
 DSK 6713 DSP Trainer kit.
 USB Cable
 Power supply
PROGRAM
#include<stdio.h>
#include<math.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");
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];
DSP Lab Manual
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]);
TEST DATA enter the length of the first sequence :2

enter the length of the second sequence :2
enter the first sequence :1 2
enter the second sequence :1 2
EXPECTED OUTPUT: The circular convolution is :54
RESULT: Thus the Circular convolution between two discrete time sequences is performed
using C6713 simulator

DSP Lab Manual
3. N-POINT FAST FOURIER TRANSFORM
AIM: To find the DFT of a sequence using FFT algorithm using C6713 simulator
EQUIPMENTS:
 Operating System - Windows XP
 Software - CC STUDIO 3
 DSK 6713 DSP Trainer kit.
 USB Cable
 Power supply
PROCEDURE:
1) Open Code Composer Studio, make sure the DSP kit is turned on.
2) Start a new project using ‘Project-new ‘ pull down menu, save it in a
3) separate directory(c:\ti\myprojects) with name “FFT.pjt”.
4) Add the source files “fft256.c“ and “fft.C” in the project using
5) ‘Projectadd files to project’ pull down menu.
6) Add the linker command file “hello.cmd”.
7) Add the rts file “rts6700.lib” .
8) Compile the program using the ‘Project-compile’ pull down menu or by
9) clicking the shortcut icon on the left side of program window.
10) Load the program in program memory of DSP chip using the ‘File-load program’ pull
down menu.
11) Run the program and observe output using graph utility.
PROGRAM
// N-point Fast Fourier Transform (FFT) algorithm program in c language using CC

Studio //
#include<stdio.h>
#include<math.h>
void main ()
{
short N=8;
short X[8]={1,2,3,4,4,3,2,1};//test data
float pi=3.1416;
float sumRe=0, sumlm=0;//init real/imag components
float cosine=0,sine=0;//initialize cosine/sine components
//output Real and imaginary components
float out_real[8]={0.0},out_imag[8]={0.0};
int n=0, k=0;
for(k=0;k<N;k++)

DSP Lab Manual
{
sumRe=0;
sumlm=0;
for(n=0;n<N;n++)
{
cosine=cos(2*pi*k*n/N);
sine=sin(2*pi*k*n/N);
sumRe=sumRe+X[n]*cosine;
sumlm=sumlm-X[n]*sine;
}
out_real[k]=sumRe;
out_imag[k]=sumlm;
printf("[%d]%7.3f%7.3f\n",k,out_real[k],out_imag[k]);
}
}
OUTPUT:
RESULT: N-Point FFT program was written and implemented using C6713 simulator

DSP Lab Manual
4. FIR FILTER DESIGN
AIM: To verify FIR filter using C6713 simulator
EQUIPMENTS:-

USB Cable
Power supply
PROGRAM:
#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: 1.rect 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;
}
}

DSP Lab Manual
OUTPUT:
Rectangular Window:
Triangular Window ;
RESULT: Here we designed FIR filter using C6713 simulator

DSP Lab Manual
5. IIR FILTER DESIGN

AIM: To verify IIR filter using C6713 simulator.
EQUIPMENTS:-

USB Cable
Power supply
PROGRAM:
#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;
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);
}

DSP Lab Manual
OUTPUT FOR IIR FILTER LPF

enter order of filter 2
enter the cutoff freq 50
enter the choice for IIR filter 1. LPF 2.HPF: 1
OUTPUT FOR IIR FILTER HPF

enter order of filter 2
enter the cutoff freq 50
enter the choice for IIR filter 1. LPF 2.HPF: 2
RESULT: Here we designed IIR filter using C6713 simulator

DSP Lab Manual
ADDITIONAL EXPERIMENTS
Experiment No: 1
Implementation of Decimation Process
AIM: To implement the decimation of given sequence by factor M by using MAT LAB
Software.
APPARATUS: (i) MATLAB Software

(ii) Computer
PROGRAM:
clc;
close all;
clear all;
M = input('enter Down-sampling factor : ');
N = input('enter number of samples :');
n = 0:N-1;
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));
title('Input Sequence');
xlabel('Time index n');
ylabel('Amplitude');
subplot(2,1,2);
m = 0:(N/M)-1
stem(m,y(1:N /M));
title('Output Sequence');
xlabel('Time index n');ylabel('Amplitude');
OUTPUT:
enter Down-sampling factor : 3
enter number of samples :100

DSP Lab Manual
Input Sequence
2
1
Amplitude
-1
-2
0 10 20 30 40 50 60 70 80 90 100
Time index n
Output Sequence
2
1
Amplitude
-1
-2
0 5 10 15 20 25 30 35
Time index n
RESULT: The decimator for a given sequence is observed for chosen factor M. Hence the
theory and practical is verified.

DSP Lab Manual
Experiment No: 2
Implementation of Interpolation Process
AIM: To implement the interpolation for given sequence by factor L.
APPARATUS: (i) MATLAB Software
(ii) Computer
PROGRAM:
clc;
close all;
clear all;
L = input('Up-sampling factor = ');
N = input('enter number of samples :');
n = 0:N-1;
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));
title('Input Sequence');
subplot(2,1,2);
m = 0:(N*L)-1;
stem(m,y(1:N*L));
title('Output Sequence');
OUTPUT:
Up-sampling factor = 3
enter number of samples :100

DSP Lab Manual
Input Sequence
2
1
Amplitude
-1
-2
0 10 20 30 40 50 60 70 80 90 100
Time index n
Output Sequence
2
1
Amplitude
-1
-2
0 50 100 150 200 250 300
Time index n
RESULT: The interpolation of given sequence is observed for factor value L. Hence the
theory and practical are verified.

DSP LAB Merged

Uploaded by

Document Information

Original Description:

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

DSP LAB Merged

Uploaded by

Copyright:

Available Formats

Dr.K.V.

SUBBA REDDY COLLEGE OF ENGINEERING

DIGITAL SIGNAL PROCESSING LAB

S.No List Of Experiments Page No

8 Generation Of Sine Wave And Square Wave 31

1. GENERATION OF VARIOUS SIGNALS & SEQUENCES

MATLAB 7.8.0.347(R2009a) Software.

1. Open the MATLAB software by double clicking the ICON on desktop.

2. Open the new M-file by using file menu

3. Write the program in new file

4. Click on save and run the icon

5. Perform error check which displayed on command window

6. Plot the waveforms which displays on figure window.

7. Note down the values, which displays on the work space.

Dr K V Subba Reddy College Engineering For Women Page 1

xlabel('t'); ylabel('x(t)'); title('sawtooth Signal');

%---------------------Random Bipolar Sequence ------------%

fs=50; % For T=0.2s fs =50Hz%

Dr K V Subba Reddy College Engineering For Women Page 2

Dr K V Subba Reddy College Engineering For Women Page 3

Unit Step Signal

Dr K V Subba Reddy College Engineering For Women Page 4

Dr K V Subba Reddy College Engineering For Women Page 5

Dr K V Subba Reddy College Engineering For Women Page 6

Dr K V Subba Reddy College Engineering For Women Page 7

periodic triangle wave

Aperiodic triangle wave

Dr K V Subba Reddy College Engineering For Women Page 8

Bipolar Random data for 10kbps NRZ-L encoded bit stream

Dr K V Subba Reddy College Engineering For Women Page 9

2. FIND AVERAGE POWER AND ENERGY OF A GIVEN SIGNAL

MATLAB 7.8.0.347(R2009a) Software.

1. Open the MATLAB software by double clicking the ICON on desktop.

2. Open the new M-file by using file menu

3. Write the program in new file

4. Click on save and run the icon

5. Perform error check which displayed on command window

6. Plot the waveforms which displays on figure window.

7. Note down the values, which displays on the work space.

Dr K V Subba Reddy College Engineering For Women Page 10

Dr K V Subba Reddy College Engineering For Women Page 11

3. CONVOLUTION AND CORRELATION OF SIGNALS

MATLAB 7.8.0.347(R2009a) Software.

1. Open the MATLAB software by double clicking the ICON on desktop.

2. Open the new M-file by using file menu

3. Write the program in new file

4. Click on save and run the icon

5. Perform error check which displayed on command window

6. Plot the waveforms which displays on figure window.

7. Note down the values, which displays on the work space.

PROGRAM: Linear Convolution

Dr K V Subba Reddy College Engineering For Women Page 12

PROGRAM: Circular Convolution

%Appending zeros if lengths of the sequences are not equal

Dr K V Subba Reddy College Engineering For Women Page 13

PROGRAM: Cross Correlation.

Dr K V Subba Reddy College Engineering For Women Page 14

linear convolution program

Dr K V Subba Reddy College Engineering For Women Page 15

Enter the input sequence :[1 2 3 4 5]

Dr K V Subba Reddy College Engineering For Women Page 16

Dr K V Subba Reddy College Engineering For Women Page 17

4. DTFT OF A GIVEN SIGNAL