DSP 72

DIGITAL SIGNAL PROCESSING LABORATORY 21071A0486
1. Circular Convolution
Aim: To find the circular convolution of x(n) = {1, 1, 2, 2} and h(n) = {1,2,3}
Software Used: MATLAB R2022A
Program:
clc; clear all; x = [1 1 2

2]; h = [1 2 3 ]; a =
length(x); b =
length(h); c =
max(a,b); x1 =[x
zeros(1,(c-a))]; h1 =[h
zeros(1,(c-b))]; z =
zeros(c,c); for i = 1:c
z(:,i) = circshift(h1',i-1);
end
y = z*x1';
disp('Circular convolved sequence is y(n) = ');
disp(y');
subplot(3,1,1); stem(x);
xlabel('Samples');
ylabel('Amplitude');
title('Input Sequence');
axis([0 c+1 0 5]);
subplot(3,1,2); stem(h);
xlabel('Samples');
title('Impulse Response Sequence');
axis([0 c+1 0 5]);
subplot(3,1,3);
stem(y'); xlabel('Samples');
title('Circular Convolved Sequence'); axis([0
c+1 0 12]);
VNRVJIET Page 1 Output: Circular convolved sequence is y(n) = {11 , 9 , 7, 9}

Graph:
Result:
Circular convolution is performed for x(n) = {1, 1, 2, 2} and h(n) = {1,2,3} in MATLAB. y(n) is
obtained as {11 , 9 , 7, 9}. The theoretical/mathematical and practical values obtained are
same.
Linear Convolution
Aim: To find the linear convolution of x(n) = {1, 2, 3, 4} and h(n) = {-3, 2, 1}
Program:
clc; clear all; x

= [1 2 3 4]; h
= [-3 2 1]; a =
length(x); b =
length(h); c =
a+b-1; y =
conv(x,h);
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if (c == length(y)) disp('Linear Convolved sequence

is y(n) = '); disp(y);
else disp('Error');
end
subplot(3,1,1); stem(x);
xlabel('Samples');
title('Input Sequence');
axis([0 5 -4 5]);
subplot(3,1,2); stem(h);
xlabel('Samples');
title('Impulse Response Sequence');
axis([0 5 -4 5]);
subplot(3,1,3); stem(y');
xlabel('Samples');
title('Linear Convolved Sequence');
axis([0 c+1 -6 12]);
Output: Linear Convolved sequence is y(n) = {-3, -4, -4, -4, 11, 4}
Graph:
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Result:
Linear convolution is performed for x(n) = {1, 2, 3, 4} and h(n) = {-3, 2, 1} in MATLAB. y(n) is
obtained as {-3, -4, -4, -4, 11, 4} . The theoretical/mathematical and practical values
obtained are same.
2. Generation of DTMF Signals

Aim: To generate DTMF (Dual-tone Multi-frequency) signal for a given input number and
plot the corresponding signal
Program:
clc;
num = input('Enter number ','s');
fs = 800; T
= 1;
t = 0:1/fs:T;
x = 2*pi*[697,770,852,941]; y =
2*pi*[1209,1336,1477,1633]; %DTMF
matrix as row and column vector
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n = length(num)/2;
for i=1:length(num) switch
num(i)
case'1'
b=sin(x(1)*t)+sin(y(1)*t);
case'2'
case'3'
case'4'
case'5'
case'6'
case'7'
case'8'
case'9'
case'*'
case'0'
case'#'
case'A'
case'B'
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case'C'
case'D'
otherwise
fprintf('invalid number');
end
sound(b); subplot(n,2,i); plot(t,b);

xlabel('Samples');
ylabel('Amplitude'); title(['Plot for
' num2str(num(i))]); pause(1);
end
Output:
Input Number: 9553251597

A unique sound for each input number and corresponding plot is observed.
Graph:
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Result:
DTMF signals and corresponding sounds are generated for each number in the input string.
Respective plots are observed.
3. Discrete Fourier Transform (DFT) and Inverse

Discrete Fourier Transform (IDFT)
Aim: 1. To find the Discrete Fourier transform (DFT) and Inverse Discrete Fourier transform
(IDFT) of the given discrete sequence [ 0 1 1 2 ]
2. To find the DFT and IDFT of a sinusoidal input signal and plot it’s corresponding
magnitude and phase response
Program:
1. DFT and IDFT of a discrete sequence:
clc;
clear all;
x = input('Enter the input sequence');
N = length(x);
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X = zeros(1,N);
for k = 1:N w =
2*pi*(k-1)/N; for
n = 1:N
X(k)= X(k)+x(n)*exp(-1j*w*(n-1)); end
end
Y = fft(x);
fprintf('DFT using code \n'); disp(X);
fprintf('DFT using inbuilt command \n'); disp(Y);
x1 = zeros(1,N); for n = 1:N w =

2*pi*(n-1)/N; for k = 1:N x1(n)=
x1(n)+X(k)*exp(1j*w*(k-1));
end
x1(n) = abs(x1(n))/N;
end
y1 = ifft(Y);
fprintf('Inverse DFT using code \n'); disp(x1);
fprintf('Inverse DFT using inbuilt command \n');
disp(y1);
Output:
Enter the input sequence [ 0 1 1 2]
DFT using code

4.0000 + 0.0000i -1.0000 + 1.0000i -2.0000 - 0.0000i -1.0000 - 1.0000i
DFT using inbuilt command

4.0000 + 0.0000i -1.0000 + 1.0000i -2.0000 + 0.0000i -1.0000 - 1.0000i
Inverse DFT using code

0.0000 1.0000 1.0000 2.0000
Inverse DFT using inbuilt command

0 1 1 2
2. DFT and IDFT of a sinusoidal input signal:
clc;
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clear all; fs
= 1000;
fm = input('Enter sine frequency');
T = 1; n = 0:1/fs:T;
x = sin(2*pi*fm/fs*n);
N = length(x);
X = zeros(1,N);
for k = 1:N w =
2*pi*(k-1)/N; for
n1 = 1:N
X(k)= X(k)+x(n1)*exp(-1j*w*(n1-1)); end
end
x1 = zeros(1,N); for
n1 = 1:N w =
2*pi*(n1-1)/N; for k
= 1:N
x1(n1)= x1(n1)+X(k)*exp(1j*w*(k-1));
end x1(n1) =
x1(n1)/N; end
Y = fft(x); y1
= ifft(Y);
subplot(3,2,1); plot(n,x);
xlabel('samples');
ylabel('amplitude');
title('Input sine');
subplot(3,2,2); plot(abs(Y));
xlabel('samples');
title('FFT');
axis([-100 1100 0 600]);
subplot(3,2,3); plot(n,X);
xlabel('samples');
title('DFT');
axis([-0.5 1.5 0 6]);
subplot(3,2,4); plot(phase(Y));
xlabel('samples');
title('Phase response');
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axis([-100 1100 -5 10]);
subplot(3,2,5); plot(n,x1);
xlabel('samples');
title('Inverse DFT');
subplot(3,2,6); plot(y1);
xlabel('samples');
title('Inverse FFT');
Output: Enter sine frequency 3000 Graph:
Result:
DFT and Inverse DFT of a given sequence is found using both the theoretical formula and
inbuilt command. Both the results are observed to be same. Magnitude and Phase
response of sinusoidal input are plotted.
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4. Power Spectral Density

Aim: To find power spectral densities of given signals.
Software used: MATLAB R2022A
Program :
Method 1:
clc; clear
all; close
all;
fs=8000;
f1=1000;
f2=2000;
N=200;
n=0:N-1;
x=sin(2*pi*f1*n/fs)+2*sin(2*pi*f2*n/fs);
xf=abs(fft(x))/N; p=xf.*xf; f=(0:N-
1)*fs/N;
y=angle(xf);
subplot(3,1,1); plot(n,x);
xlabel('Samples');
title('Input Sine Signal');
subplot(3,1,2);
plot(f,p); xlabel('Frequency');
ylabel('Magnitude');
title(['Two sided PSD at ',num2str(f1),' and at ',num2str(f2),'with refernce to ',num2str(fs/2)]);
subplot(3,1,3); plot(f,10*log(p));
xlabel('frequency in Hz');
ylabel('magnitude in dB');
title('PSD in log scale');
Output: fs=8000, f1=1000, f2=2000
Graph:
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Method 2 :
clc; clear all;

t=0:100;
fm=1000;
fs=8000;
fo=fm/fs;
x=sin(2*pi*fo*t);
N=length(x);
n=0:N-1; f=(0:N-
1)*fs/N; k=n;
wf=exp(-j*2*pi/N)
nk=k'*n;
wfnk=wf.^nk;
X=x*wfnk;
X1=conj(X);
Z=fft(x);
Z1=conj(Z);
Y=(X.*X1)/N;
W=(Z.*Z1)/N;
subplot(2,1,1); plot(f,Y);
subplot(2,1,2);
plot(f,W);
Output: fs=8000,fm=1000
Graph:
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Result: Therefore, power spectral densities of given signals is found and plotted.
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5. Impulse Response and Step Response

Aim: To compute the impulse response and step response of a system based on it’s
coefficients and plot the response
Program:
a) Impulse Response:
clc; b = [1 2 4]; a = [1
-1.5 1.5 -1 0.5];
n=0:1:20;
x=zeros(1,length(n));
x(find(n==0))=1;
y=filter(b,a,x);
[z,t]=impz(b,a,length(n));
subplot(3,1,1) stem(n,x);
xlabel('time');
title('Unit Impulse Input');
axis([-0.5 20 0 2]);
subplot(3,1,2)
stem(n,y); xlabel('time');
title('Impulse Response (Filtered output)');
subplot(3,1,3) stem(t,z); xlabel('time');

ylabel('amplitude'); title('Impulse Response
(Using Built-in Func)');
b) Step Response:
clc; b = [1 1 2]; a = [1
-1.5 1.5 -1 0.5];
n = 0:20; x =
ones(1,length(n)); y =
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filter(b,a,x); [z,t] =
stepz(b,a,length(n));
subplot(3,1,1);
stem(n,x); xlabel('time');
ylabel('amplitude')
title('Input Step Signal');
axis([0 20 0 2]);
subplot(3,1,2);
stem(n,y); xlabel('time');
title('Step Response (Using Filter)');
subplot(3,1,3); stem(t,z); xlabel('time');

ylabel('amplitude'); title('Step Response (Using
Built-in command)'); Output: Respective
response plots are observed
Graphs:
Impulse Response Plot
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Step Response Plot
Result: For the given filter coefficients the step and impulse responses of the system are
calculated both using the procedure and the inbuilt command.
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6. Interpolation and Decimation

Aim: To perform interpolation and decimation on a given sampled signal.
Program:
a) Interpolation:
clc;
clear all;
fs=input('enter sampling frequency');

f=input('enter frequency of signal');
L=input('enter interpolation factor');
t=0:1/fs:1;
x=sin(2*pi*f*t);
N=length(x); n=0:N-1;
m=0:(N*L)-1;
x1=zeros(1,L*N);
j=1:L:N*L; x1(j)=x;
f1=fir1(34,0.48,'low');
output=2*filtfilt(f1,1,x1);
y=interp(x,L);
subplot(3,1,1); stem(n,x);
xlabel('samples');
title('Input signal');
subplot(3,1,2);
stem(m,output); axis
([0 200 -1 1]);
xlabel('samples');
title('Interpolated signal');
subplot(3,1,3);
stem(m,y); axis ([0
200 -1 1]);
xlabel('samples');
title('Interpolated signal using inbuilt command');
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b) Decimation:
clc;
clear all;

fm=input('enter frequency of signal');
D=input('enter decimation factor');
t=0:1/fs:1;
x=sin(2*pi*fm*t);
N=length(x); n=0:N-1;
m=0:(N/D);
x2=zeros(1,length(m));
j=1:D:N; x2=x(j);
v=decimate(x,D,'fir');
subplot(3,1,1); stem(n,x);
xlabel('samples');
title('Input signal');
subplot(3,1,2); stem(m,x2);
xlabel('samples');
title('Decimated signal');
subplot(3,1,3); stem(m,v); axis([0 50 -1.2 1.2]);

xlabel('samples'); ylabel('amplitude');
title('Decimated signal (Using inbuilt
command)');
Output:
Input Signal Frequency: 10

Sampling Frequency: 100
Interpolation Factor: 2
Decimation Factor: 2
Graphs:
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Interpolation
Decimation
Result: For a given signal with a specific sampling frequency Interpolation and Decimation
are performed to increase and decrease the sampling rates respectively.
7. Generation of Sinusoidal signal through Filtering

(LCCDE approach)
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Aim: To generate sinusoidal signal from an impulse signal by using filtering and LCCDE
algorithm approach
Program:
clc
clear all t=0:0.01:10;
x=zeros(1,length(t));
x(find(t==0))=1;
w=input('Enter the value of w');
%a: Coefficients of Y[z]

%b: Coefficients of X[z]
a=[1 -2*cos(w) 1]; b=[0
sin(w) 0];
y=zeros(1,length(t));
x1=0; x2=0;
y1=0; y2=0;
%Generation of Sinusoidal Signal using LCCDE approach for

n=1:length(x)
y(n)=b(1)*x(n)+b(2)*x1+b(3)*x2-a(2)*y1-a(3)*y2;
x2=x1; x1=x(n); y2=y1; y1=y(n);
end
%Plot the input signal, generated sinusoidal signal and the response.
subplot(3,1,1); plot(t,x,'lineWidth',2); xlabel('Time');
title('Input Signal (impulse)');
subplot(3,1,2);
plot(t,y); xlabel('Time');
title('Sinusoidal signal (Sine) generated');
yi=filter(b,a,x);
subplot(3,1,3);
plot(t,yi); xlabel('Time');
title('Using inbuilt function');
Output: w = 0.1
Graphs:
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Result: Thus, sinusoidal signal is generated based on the input frequency using LCCDE
coefficients and is also verified with inbuilt function.
8. Implementation of Filters using IIR

(Butterworth and Chebyshev)
Aim: To plot the frequency response of Butterworth and Chebyshev filters as Low pass and
High pass filters and verify the same with a given input signal
Program:
1) High Pass and Low Pass Filter design:
clc; clear
all; close
all;
fs=8000;
N=1024;
n=0:N-1;
f1=500;
f2=2000;
x=3*sin(2*pi*f1*n/fs)+cos(2*pi*f2*n/fs); X=abs(fft(x,N))/N;
f=(0:1:N/2-1)*fs/N;
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subplot(3,1,1);
plot(f,X(1:N/2));
xlabel('frequency');
ylabel('amplitude'); title('input
signal spectrum');
%filter parameters
wp=1000;
ws=1500; rp=3;
rs=20; fF=fs/2;
wpa=wp/fF;
wsa=ws/fF;
%filter design
[n,wc]=buttord(wpa,wsa,rp,rs);
[b,a]=butter(n,wc,'high');
[h,f_hz]=freqz(b,a,512,fs);
subplot(3,1,2);
plot(f_hz,20*log10(abs(h)));
xlabel('frequency'); ylabel('gain
in db');
title('High pass filter response');
y=filter(b,a,x);
y1=abs(fft(y,N))/N;
subplot(3,1,3);
plot(f,y1(1:N/2));
xlabel('frequency'); ylabel('gain
in db');
title('filtered signal spectrum');
Output: Low pass and high pass filters designed for given parameters
Graphs:
1) Low pass Filter response
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2) High Pass filter response:
2) Butterworth filter design
clc; clear
all;
close all;
wp=input('enter passband frequency');

ws=input('enter stopband frequency');
rp=input('enter passband ripple');
rs=input('enter stopband ripple');
wpn=2*wp/fs; wsn=2*ws/fs;
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[n,wc]=buttord(wpn,wsn,rp,rs);
[b,a]=butter(n,wc); [h,w]=freqz(b,a);
h1=20*log10(abs(h));
w1=angle(h);
subplot(2,1,1); plot(w/pi,h1);
title('magnitude plot');
xlabel('freq');
ylabel('magnitude H(W)');
subplot(2,1,2);
plot(w/pi,w1); title('phase
plot'); xlabel('freq');
ylabel('phase');
Output and Graphs:
1) enter passband frequency :

400 enter stopband frequency :
700 enter passband ripple : 0.3
enter stopband ripple : 20 enter
sampling frequency : 2000
2) enter passband frequency :

500 enter stopband frequency :
800 enter passband ripple : 0.1
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3) Cheybyshev Type 1 Filter

Design:
clc; clear
all;
close all;
wp=input('enter passband frequency');

ws=input('enter stopband frequency');
rp=input('enter passband ripple'); rs=input('enter
stopband ripple');
wpn=2*wp/fs; wsn=2*ws/fs;
[n,wc]=cheb1ord(wpn,wsn,rp,rs);
[b,a]=cheby1(n,rp,wc);
[h,w]=freqz(b,a); h1=20*log10(abs(h));
w1=angle(h);
subplot(2,1,1); plot(w/pi,h1);
title('Chebyschev Filter I magnitude plot');
xlabel('freq');
ylabel('magnitude H(W)');
subplot(2,1,2);
plot(w/pi,w1); title('phase
plot'); xlabel('freq');
ylabel('phase');
Output and Graphs:
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1) enter passband frequency : 400

enter stopband frequency : 700
enter passband ripple : 0.3
2) enter passband frequency : 500

enter stopband frequency : 800
enter passband ripple : 0.1
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Result: Butterworth and Chebyshev Filters are designed and verified for specific input
signals. Their frequency responses are plotted.
8. Implementation of Filters using FIR

(Windowing Techniques)
Aim: To verify the functionality and filtering characteristics of the following windowing
functions in FIR filter design
1) Rectangular Window
2) Blackman Window
3) Hamming Window
4) Triangular Window Software used: MATLAB
R2022A Program:
1) Rectangular Window function:
N=input('enter order of filter');

wc=input('enter the cutoff frequency');
ws=input('enter sampling frequency');
wcd=2*wc/ws;
n=0:N;
wr=rectwin(N+1); [hr,
w]=freqz(wr,1,512);
hlp=fir1(N,wcd,'low',wr);
[hlpz,w1]=freqz(hlp,1,512);
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subplot(2,2,1);
stem(n,wr); xlabel('time');
ylabel('frequency');
title('rectangular window function');
subplot(2,2,3);
plot(w/pi,20*log10(abs(hr))); xlabel('frequency');
ylabel('amplitude'); title('frequency response of
rectangular window function');
subplot(2,2,2);
plot(w1/pi,20*log10(abs(hlpz)));
xlabel('frequency'); ylabel('amplitude');
title('frequency response of designed fir filter');
subplot(2,2,4);
plot(w1/pi,angle(hlpz));
xlabel('phase'); ylabel('amplitude');
title('phase response of designed fir filter');
Output and Graphs:
order N=5
order N=50
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2) Blackman Window function
N=input('enter order of filter'); wc=input('enter

the cutoff frequency'); ws=input('enter
sampling frequency');
wcd=2*wc/ws; n=0:N;
wr=blackman(N+1); [hr,
w]=freqz(wr,1,512);
subplot(2,2,1); stem(n,wr);
xlabel('time'); ylabel('frequency');
title('blackman window function');
subplot(2,2,3); plot(w/pi,20*log10(abs(hr)));
xlabel('frequency'); ylabel('amplitude'); title('frequency
response of blackman window function');
subplot(2,2,2);
subplot(2,2,4); plot(w1/pi,angle(hlpz));
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title('phase response of designed fir

filter');
Output and Graphs:
order of the filter N=5
order of the filter is N=50
3) Hamming Window function
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N=input('enter order of filter');

wcd=2*wc/ws; n=0:N;
wr=hamming(N+1); [hr,
w]=freqz(wr,1,512);
xlabel('time'); ylabel('frequency');
title('hamming window function');
subplot(2,2,3); plot(w/pi,20*log10(abs(hr)));
xlabel('frequency'); ylabel('amplitude'); title('frequency
response of hamming window function');
subplot(2,2,2);
subplot(2,2,4); plot(w1/pi,angle(hlpz));
title('phase response of designed fir
filter');
Output:
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4) Triangular Window Function
clc; clear
all; close
all;
N=input('enter order of the filter');
wcd=2*wc/ws; n=0:N;
wr=triang(N+1); [hr
w]=freqz(wr,1,512);
title('rectangular window function');
xlabel('time'); ylabel('amplitude');
subplot(2,2,3);
plot(w/pi,20*log10(abs(hr)));
title('frequency response of rectangular window function');
xlabel('frequency'); ylabel('amplitude'); subplot(2,2,2);
plot(w/pi,20*log10(abs(hlpz))); title('frequency response
of designed FIR LPF');
ylabel('amplitude'); subplot(2,2,4);
plot(w/pi,angle(hlpz)); title('
response of designed FIR LPF');
Outputs and Graphs:

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1) enter order of the filter10

enter the cutoff frequency100
enter sampling frequency2000
2) enter order of the filter5

enter the cutoff frequency600
enter sampling frequency3000
Result: Therefore, frequency responses of various window functions are observed.
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CODE COMPOSER STUDIO STEPS

• Start CCS setup by double clicking on the CCS desktop icon.
New project creation:

• To create a new project, choose new CSS project from the project menu.
• Then Select the target as C671X Floating point DSP and click on
TMS320C6713 Enter the project name.
Then Click on Finish.
New Target configuration:
• In View Option, select Target configuration.
• Click on the project folder and select New Target configuration.
• Then change the filename as 6713.ccxml.
• Click on Finish.
• In basic window, select Connection as Texas Instruments Simulator.
select Device as C67xx CPU Cycle Accurate Simulator, little Endian. -Click
on Save.
• Then in Advanced window, select TMS320C67XX.
• Then in CPU Properties window click on browse button and open
DSK6713.gel file.
-Click on Save.
Launch Target configuration:
• Click on the User defined folder and select C713.ccxml.
Click on Launch target configuration. Then
click on the stop button.
• Then a CSS Editor window will be opened.
Click on your project and then click on main.c to type the programs. Execution:
• To compile a program, right click on the project name. Select build
configuration and then click on build all.
• To run a program, click on the debug option in run menu.
Click on the run option again to display the output in the console window.
1.1 LINEAR CONVOLUTION
AIM: To Perform Linear Convolution of two sequences using C-Code
SOFTWARE: PC and Code Composer Studio
PROGRAM:
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#include<stdio.h>
void main()
{ int
a[9]={1,2,3,4,5}; int
b[9]={1,1,1,1,1};
int m=5,n=5;
int c[10],i,j;
for(i=0;i<m+n-1;i++)
{ c[i]=0;
for(j=0;j<=i;j++)
c[i]=c[i]+(a[j]*b[i-j]);
}
for(i=0;i<m+n-1;i++) printf
("%d", c[i]);
}
OUTPUT :
1 3 6 10 15 14 12 9 5
1.2 LINEAR CONVOLUTION
AIM : To Perform Linear Convolution of two sequences using C-Code
PROGRAM:
#include<stdio.h>
#define MAX1 10
#define MAX2 10
void main()
{
int a[MAX1],b[MAX2],i,j,M,N,temp;
printf("ENTER THE RANGE OF THE ARRAY A:(MAX1=10)\n"); scanf("%d",&M);
printf("ENTER THE RANGE OF THE ARRAY B:(MAX2=10)\n"); scanf("%d",&N);
printf("ENTER THE ELEMENTS OF THE ARRAY A:\n"); for(i=0;i<M;i++)
scanf("%d",&a[i]);
printf("ENTER THE ELEMENTS OF THE ARRAY B:\n"); for(i=0;i<N;i++)
scanf("%d",&b[i]);
printf("THE VALUES OF THE ARRAY (A CONV B):\n");
for(i=0;i<(M+N-1);i++)
VNRVJIET Page 35
{ temp=0;
for(j=0;j<=i;j++
)
{
if((j<M)&&((i-j)<N)) temp+=a[j]*b[i-
j];
}
printf("%d ",temp);
}
}
SIMULATION RESULTS:
ENTER THE RANGE OF THE ARRAY A:(MAX1=10)

4
ENTER THE RANGE OF THE ARRAY B:(MAX2=10)
3
ENTER THE ELEMENTS OF THE ARRAY A:
1234
ENTER THE ELEMENTS OF THE ARRAY B:
123
THE VALUES OF THE ARRAY (A CONV B):
1 4 10 16 17 12
VNRVJIET Page 36
AIM:
2.1 CIRCULAR CONVOLUTION
To Perform Circular Convolution of two sequences using C-Code
PROGRAM:
#include<stdio.h>
#define MAX 10
void main( )
{
int a[MAX],b[MAX],i,j,M,N,temp,k,c[MAX],x,z;
int max(int ,int );
printf("ENTER THE RANGE OF THE ARRAY A:(MAX=10)\n");
scanf("%d",&M);
printf("ENTER THE RANGE OF THE ARRAY B:(MAX=10)\n");
scanf("%d",&N);
printf("ENTER THE ELEMENTS OF THE ARRAY A:\n");
for(i=0;i<M;i++)
scanf("%d",&a[i]);
printf("ENTER THE ELEMENTS OF THE ARRAY B:\n");
for(i=0;i<N;i++) scanf("%d",&b[i]); k=max(M,N);
if(k==M) {
for(i=N;i<M;i++
) b[i]=0;
for(i=0;i<M;i++)
c[i]=b[M-1-i];
} else {
for(i=M;i<N;i++
) a[i]=0;
for(i=0;i<N;i++)
c[i]=a[N-i-1];
}
printf("THE VALUES OF THE ARRAY (A CIRCULAR CONV B):\n");
for(i=0;i<k;i++)
{ temp=c[0];
c[0]=c[k-1];
for(z=k-1;z>=2;z--
)
VNRVJIET Page 37
{ c[z]=c[z-1]; }
c[1]=temp;
x=0;
for(j=0;j<k;j++
) { if(k==M)
x=x+a[j]*c[j];
else
x=x+b[j]*c[j];
}
printf("%d ",x);
}}
int max(int l,int p)
{ if(l>p)
return(l);
else
return(p)
;
}
SIMULATED RESULTS:
ENTER THE RANGE OF THE ARRAY A:(MAX=10)

4
ENTER THE RANGE OF THE ARRAY B:(MAX=10)

3

1
2
3
4

1
2
3
THE VALUES OF THE ARRAY (A CIRCULAR CONV B):

18 16 10 16
3.1 AUTO & CROSS CORRELATION
To Perform correlation of two sequences using C-Code
VNRVJIET Page 38
AIM:
PROGRAM:
#include<stdio.h>
#define MAX 10
void main()
{
int a[MAX],b[MAX],i,j,M,N,temp;
printf("ENTER THE RANGE OF THE ARRAY A:(MAX=10)\n");
scanf("%d",&M);
printf("ENTER THE RANGE OF THE ARRAY B:(MAX=10)\n");
scanf("%d",&N);
printf("ENTER THE ELEMENTS OF THE ARRAY A:\n");
for(i=0;i<M;i++)
scanf("%d",&a[i]);
printf("ENTER THE ELEMENTS OF THE ARRAY B:\n"); for(i=N-
1;i>=0;i--)
scanf("%d",&b[i]);
printf("THE VALUES OF THE ARRAY (A CONV B):\n");
for(i=0;i<(M+N-1);i++)
{ temp=0;
for(j=0;j<=i;j++
)
{
if((j<M)&&((i-j)<N)) temp+=a[j]*b[i-
j];
} printf("%d
",temp);
}
}
SIMULATED RESULTS:
ENTER THE RANGE OF THE ARRAY A:(MAX=10)

4
ENTER THE RANGE OF THE ARRAY B:(MAX=10)
3
1256
VNRVJIET Page 39

891
THE VALUES OF THE ARRAY (A CONV B):
1 11 31 67 94 48
4.1 DFT
AIM: To Perform DFT using C-Code
PROGRAM:
#include<stdio.h>
#include<math.h>
int N,k,n,i;
float pi=3.1416,sumre=0, sumim=0,out_real[8]={0.0}, out_imag[8]={0.0};
int x[32]; void main(void)
{
printf(" enter the length of the sequence\n");
scanf("%d",&N); printf("
enter the sequence\n");
for(i=0;i<N;i++)
scanf("%d",&x[i]);
for(k=0;k<N;k++)
{
sumre=0;
sumim=0;
for(n=0;n<N;n++)
{
sumre=sumre+x[n]* cos(2*pi*k*n/N);
sumim=sumim-x[n]* sin(2*pi*k*n/N);
}
out_real[k]=sumre; out_imag[k]=sumim;
printf("X([%d])=\t%f\t+\t%fi\n",k,out_real[k],out_imag[k])
;
}
}
SIMULATED RESULTS:
VNRVJIET Page 40
AIM:
enter the length of the sequence

4 enter the
sequence
1467
X([0])= 18.000000 + 0.000000i
X([1])= -4.999938 + 3.000043i
X([2])= -4.000000 + 0.000096i
X([3])= -5.000184 + -2.999868i
5 .GENERATION OF SINE WAVE
To Generate sine wave using C-Code
PROGRAM:
#include<stdio.h>
#include<math.h>
float a[100];
main()
{
int i;
for(i=0;i<11;i++)
{ a[i]=
sin(2*3.14*5*i/100);
printf("%f",a[i]);
}
}
SIMULATED RESULTS:
0.0000000
.3088660
.5875280
.8087360
.9508591
.0000000
.9513510
VNRVJIET Page 41
.8096720
.5888160
.3103800
.001593
VNRVJIET Page 42

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DIGITAL SIGNAL PROCESSING LABORATORY 21071A0486

Software Used: MATLAB R2022A

clc; clear all; x = [1 1 2

VNRVJIET Page 1 Output: Circular convolved sequence is y(n) = {11 , 9 , 7, 9}

Software Used: MATLAB R2022A

clc; clear all; x

if (c == length(y)) disp('Linear Convolved sequence

2. Generation of DTMF Signals

Software Used: MATLAB R2022A

for i=1:length(num) switch

sound(b); subplot(n,2,i); plot(t,b);

Input Number: 9553251597

3. Discrete Fourier Transform (DFT) and Inverse

Software Used: MATLAB R2022A

1. DFT and IDFT of a discrete sequence:

x1 = zeros(1,N); for n = 1:N w =

Enter the input sequence [ 0 1 1 2]

DFT using code

DFT using inbuilt command

Inverse DFT using code

Inverse DFT using inbuilt command

2. DFT and IDFT of a sinusoidal input signal:

axis([-100 1100 -5 10]);

Output: Enter sine frequency 3000 Graph:

4. Power Spectral Density

Software used: MATLAB R2022A

Output: fs=8000, f1=1000, f2=2000

clc; clear all;

5. Impulse Response and Step Response

Software used: MATLAB R2022A

subplot(3,1,3) stem(t,z); xlabel('time');

subplot(3,1,3); stem(t,z); xlabel('time');

Impulse Response Plot

Step Response Plot

6. Interpolation and Decimation

Software used: MATLAB R2022A

fs=input('enter sampling frequency');

fs=input('enter sampling frequency');

subplot(3,1,3); stem(m,v); axis([0 50 -1.2 1.2]);

Input Signal Frequency: 10

7. Generation of Sinusoidal signal through Filtering

Software used: MATLAB R2022A

%a: Coefficients of Y[z]

%Generation of Sinusoidal Signal using LCCDE approach for

8. Implementation of Filters using IIR

Software used: MATLAB R2022A

1) High Pass and Low Pass Filter design:

1) Low pass Filter response

2) High Pass filter response:

2) Butterworth filter design

wp=input('enter passband frequency');

Output and Graphs:

1) enter passband frequency :

2) enter passband frequency :

3) Cheybyshev Type 1 Filter

wp=input('enter passband frequency');

Output and Graphs:

1) enter passband frequency : 400

2) enter passband frequency : 500

8. Implementation of Filters using FIR

1) Rectangular Window function:

N=input('enter order of filter');

Output and Graphs: