DIGITAL SIGNAL PROCESSING LAB

EXPERIMENT-1 : GENERATION OF VARIOUS SIGNALS AND SEQUENCES

AIM: TO GENERATE VARIOUS SIGNALS AND SEQUENCES USING MATLAB
SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE
PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
PROGRAM :
%STEP SEQUENCE
N=input('enter the length of sequence in even number');
x=ones(1,N);
n=0:1:N-1;
stem(n,x);
axis([-1 N -1 2]);
title('step sequence');
xlabel('samples');
ylabel('amplitude');
OUTPUT : ENTER THE LENGTH OF SEQUENCE IN EVEN NUMBER: 20

%IMPULSE SEQUENCE
clc;
clear all;
close all;
N=input('enter the length of sequence in even number');
N1=N/2;
x= [zeros(1,N1) ones(1,1) zeros(1,N1)];
n= -N1:1:N1;
stem(n,x);
axis([-N1 N1 -1 2]);
title('impulse sequence');
xlabel('samples');
ylabel('amplitude');
OUTPUT : ENTER THE LENGTH OF SEQUENCE IN EVEN NUMBER: 20

%RAMP SEQUENCE
clc;
clear all;
close all;
N=input('enter the length of sequence in even number');
n=0:1:N-1;
x=n;
stem(n,x);
axis([-1 N -1 N]);
title('Ramp sequence');
xlabel('samples');
ylabel('amplitude');
grid on
OUTPUT : ENTER THE LENGTH OF SEQUENCE IN EVEN NUMBER: 20

%EXPONENTIAL SEQUENCE
clc;
clear all;
close all;
N=input('enter the length of sequence in even number');
n=0:1:N-1;
x=exp(-n);
stem(n,x);
axis([-1 N -1 2]);
title('exponential sequence');
xlabel('samples');
ylabel('amplitude');
grid on
OUTPUT : ENTER THE LENGTH OF SEQUENCE IN EVEN NUMBER: 20

%SAWTOOTH SEQUENCE
clc;
clear all;
close all;
F=input('enter frequency');
f=6;
fs=10*f;
n=0:1/fs:1;
s=sawtooth(2*pi*4*n);
stem(n,s);
title('sawtooth sequence');
xlabel('time');
ylabel('amplitude');
OUTPUT : ENTER THE FREQUENCY: 20
EXPERIMENT-2 : GENERATION OF SUM OF SINUSOIDAL SIGNALS
AIM: TO GENERATE SUM OF SINUSOIDAL SIGNALS
SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE
PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
PROGRAM :
clc;
clear all;
close all;
n=-4:0.5:4;
y1=sin(n);
subplot(2,2,1);
stem(n,y1,'filled');
grid on;
xlabel('samples');
ylabel('magnitude');
title('sinusoidal signal');
y2=sin(2*n);
subplot(2,2,2);
stem(n,y2,'filled');
grid on;
xlabel('samples');
ylabel('magnitude');
title('sinusoidal signal');
y3=y1+y2;
subplot(2,1,2);
stem(n,y3,'filled');
xlabel('samples');
ylabel('magnitude');
title('sum of sinusoidal signal');
EXPERIMENT-3 : IMPLEMENTATION AND VERIFICATION OF LINEAR AND CIRCULAR CONVOLUTION
AIM: TO IMPLEMENT AND VERIFY OF LINEAR AND CIRCULAR CONVOLUTION
SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE
PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
A)LINEAR CONVOLUTION PROGRAM :
clc;
clear all;
close all;
disp('enter the length of first sequence m=');
m=input(' ');
disp('enter the length of first sequence x(m)=');
for i=1:m;
x(i)=input(' ');
end
disp('enter the length of second sequence n=');
n=input(' ');
disp('enter the length of second sequence h(n)= ');
for j=1:n;
h(j)=input(' ');
end
y=conv(x,h);
figure;
subplot(3,1,1);
stem(x);
ylabel('amplitude');
xlabel('n');
title('x(n) vs n');
subplot(3,1,2);
stem(h);
ylabel('amplitude');
xlabel('n');
title('h(n)vs n');
subplot(3,1,3);
stem(y);
ylabel('amplitude');
xlabel('n');
title('y(n) vs n');
disp('linear convolution of x(n) and h(n) is y');
OUTPUT: enter the length of first sequence m=4
enter the length of first sequence x(m)= 1
2
3
4
enter the length of second sequence n=
5
enter the length of second sequence h(n)=

1
2
3
4
5
linear convolution of x(n) and h(n) is y

B)CIRCULAR CONVOLUTION PROGRAM :

clc;
clear all;
close all;
x=input('enter the input sequence x(n)=');
h=input('enter the impulse response sequence h(n)=');
n1=length(x);
n2=length(h);
y=cconv(x,h,max(n1,n2));
n=length(y);
disp('input sequence x(n):');
disp(x);
disp('input response h(n):');
disp(h);
disp('output of discrete LTI system is');
disp(y);
disp('press enter for graphs');
pause
b=0:1:n1-1;
a=stem(b,x);
title('input');
xlabel('time');
ylabel('amplitude');
pause
b=0:1:n2-1;
a=stem(b,h);
title('impulse');
xlabel('time');
ylabel('amplitude');
pause
b=0:1:n-1;
a=stem(b,y);
title('output');
xlabel('time');
ylabel('amplitude');

OUTPUT:
enter the input sequence x(n)=
[1 2 3 4]
enter the impulse response sequence h(n)=
[1 2 3 4 5]
input sequence x(n):
1 2 3 4

input response h(n):

1 2 3 4 5

output of discrete LTI system is

35 35 30 20 30

press enter for graphs

EXPERIMENT-4 : IMPLEMENT AND VERIFY AUTO AND CROSS CORRELATION
AIM: TO IMPLEMENT AND VERIFY AUTO AND CROSS CORRELATION
SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE
PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
PROGRAM :
clc;
clear all;
close all;
%two input sequences
x=input('enter the sequence');
h=input('enter the impulse sequence');
subplot(2,2,1);
stem(x);
xlabel('n');
ylabel('x(n)');
title('input sequence');
subplot(2,2,2);
stem(h);
xlabel('n');
ylabel('h(n)');
title('impulse sequence');
%auto correlation
z=xcorr(x,x);
subplot(2,2,3);
stem(z);
disp(z);
xlabel('n');
ylabel('z(n)');
title('auto correlation');
%CROSS CORRELATION
y=xcorr(x,h);
subplot(2,2,4);
stem(y);
disp(y);
xlabel('n');
ylabel('y(n)');
title('cross correlation');
OUTPUT:

enter the sequence

[1 2 3 4 5]
enter the impulse sequence
[1 2 3 4 5 6]
AUTO-5.0000 14.0000 26.0000 40.0000 55.0000 40.0000 26.0000 14.0000
5.0000

CROSS-6.0000 17.0000 32.0000 50.0000 70.0000 55.0000 40.0000 26.0000

14.0000 5.0000 0.0000

EXPERIMENT-5 : IMPLEMENT AND VERIFY DFT OF A SIGNAL

AIM: TO IMPLEMENT AND VERIFY DFT OF A SIGNAL
SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE
PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
PROGRAM :
clc;
clear all;
close all;
a = input('enter the sequence = ');
N = length(a);
disp('The length of the sequence');
y = zeros(1, N); % Initialize y before using it
for k = 1:N
y(k)=0
for i = 1:N
y(k) = y(k) + a(i) * exp([-2*pi*j/N]*[(i-1)*(k-1)]);N
end
end
k = 1:N;
disp('the result is:');y
figure(1);
subplot(2,1,1);
stem(k, abs(y(k)));
grid;
xlabel('sample values n----->');
ylabel('Amplitude------->');
title('magnitude response of the DFT of the given sequence');
subplot(2,1,2);
stem(angle(y) * 180 / pi);
xlabel('sample values n');
ylabel('phase');
title('phase response of the DFT of the given sequence');
OUTPUT:

enter the sequence =

[1 2 3 4]
N=4
y = 10.0000 + 0.0000i -2.0000 + 2.0000i 0.0000 + 0.0000i 0.0000 + 0.0000i

EXPERIMENT-6 : IMPLEMENT AND VERIFY FFT OF A SIGNAL

AIM: TO IMPLEMENT AND VERIFY FFT OF A SIGNAL
SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE
PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
PROGRAM :
clc;
clear all;
close all;
N=8;
m=8;
a=input('enter the input sequence');
n=0:1:N-1;
subplot(2,2,1);
stem(n,a);
xlabel('Time signal n');
ylabel('amplitude');
title('input signal');
d=fft(a,m);
disp(d);
k=0:1:N-1;
subplot(2,2,2);
stem(k,abs(d));
xlabel('frequency index k');
ylabel('magnitude');
title('magnitude response');
subplot(2,2,3);
stem(k,angle(d));
xlabel('frequency index k');
ylabel('phase');
title('phase response');
OUTPUT: enter the input sequence

[1 1 0 0 1 1 0 0]
FFT- 4.0000 + 0.0000i 0.0000 + 0.0000i 2.0000 - 2.0000i 0.0000 + 0.0000i
0.0000 + 0.0000i 0.0000 + 0.0000i 2.0000 + 2.0000i 0.0000 + 0.0000i

EXPERIMENT-7 : DESIGN OF ANALOG FILTERS

AIM: TO DESIGN ANALOG FILTERS
SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE
PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
PROGRAM : A)LPF
clear all;
close all;
f=100:20:8000;
fb=900;
k=length(f);
for i=1:k;
m(i)=1/sqrt(1+(f(i)/fb)^2);
mag(i)=20*log10(m(i));
end
figure;
semilogx(f,mag);
title('magnitude response of analog of LPF');
xlabel('frequency');
ylabel('magnitude in db');
grid on;
OUTPUT:

B)HPF
clear all;
close all;
f=100:20:8000;
fl=400;
k=length(f);
for i=1:k;
m(i)=1/sqrt(1+(fl/f(i))^2);
mag(i)=20*log10(m(i));
end
figure;
semilogx(f,mag);
title('magnitude response of analog of HPF');
xlabel('frequency');
ylabel('magnitude in db');
grid on;
EXPERIMENT-8 : DESIGN OF IIR BUTTERWORTH FILTER AND VERIFY FREQUENCY RESPONSE
AIM: TO DESIGN IIR FILTER
SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE
PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
PROGRAM :
A)LPF
clc;
clear all;
close all;
format long;
rp=input('enter pass band ripple');
rs=input('enter the stopband ripple');
wp=input('enter the passband frequency');
ws=input('enter the stopband frequency');
fs=input('enter the sampling frequency');
w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=buttord(w1,w2,rp,rs,'s');
[z,p,k]=butter(n,wn);
[b,a]=butter(n,wn,'s');
w=0:0.01:pi;
[h,om]=freqs(b,a,w);
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('magnitude plot of lpf iir');
ylabel('gain in db');
xlabel('(a)normalised freq');
subplot(2,1,2);
plot(om/pi,an);
title('phase response of lpf iir filter');
ylabel('phase in db');
xlabel('(b)normalised freq');
OUTPUT:

enter pass band ripple 0.6

enter the stopband ripple 60
enter the passband frequency 1500
enter the stopband frequency 3000
enter the sampling frequency 7000
B)HPF
clc;
clear all;
close all;
format long;
rp=input('enter pass band ripple');
rs=input('enter the stopband ripple');
wp=input('enter the passband frequency');
ws=input('enter the stopband frequency');
fs=input('enter the sampling frequency');
w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=buttord(w1,w2,rp,rs,'s');
[z,p,k]=butter(n,wn);
[b,a]=butter(n,wn,'high','s');
w=0:0.01:pi;
[h,om]=freqs(b,a,w);
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
title('magnitude plot of hpf iir');
ylabel('gain in db');
xlabel('(a)normalised freq');
subplot(2,1,2);
plot(om/pi,an);
title('phase response of hpf iir filter');
ylabel('phase in db');
xlabel('(b)normalised freq');
OUTPUT:
enter pass band ripple 0.2
enter the stopband ripple 40
enter the passband frequency 2000
enter the stopband frequency 3500
enter the sampling frequency 8000
EXPERIMENT-9 : DESIGN OF FIR FILTER USING WINDOWING TECHNIQUE
AIM: TO DESIGN FIR FILTER USING WINDOWING TECHNIQUE
SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE
PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
PROGRAM :

A)USING RECTANGULAR WINDOW

clc;
clear all;
close all;
rp=input('Enter the passband ripple(rp):');
rs=input('Enter the stopband ripple(rs):');
fp=input('Enter the passband frequency(fp):');
fs=input('Enter the stopband frequency(fs):');
f=input('Enter the sampling frequency(f):');
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
y=boxcar(n1);
%Low pass filter
b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,1,1);
plot(m);
ylabel('Gain(db)->');
xlabel('(a)Normalised frequency->');
%High pass filter
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,1,2);
plot(m);
ylabel('Gain(db)->');
xlabel('(b)Normalised frequency->');
OUTPUT:
Enter the passband ripple(rp):
0.01
Enter the stopband ripple(rs):
0.04
Enter the passband frequency(fp):
1000
Enter the stopband frequency(fs):
2000
Enter the sampling frequency(f):
5000

B)USING KAISER WINDOW

clc;
clear all;
format long;
rp=input('Enter the passband ripple(rp):');
rs=input('Enter the stopband ripple(rs):');
fp=input('Enter the passband frequency(fp):');
fs=input('Enter the stopband frequency(fs):');
f=input('Enter the sampling frequency(f):');
beta=input('Enter the beta value:');
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
y=kaiser(n1,beta);
%Low pass filter
b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,1,1);
plot(o/pi,m);
ylabel('Gain(db)->');
xlabel('(a)Normalised frequency->');
title('KAISER WINDOW')
%High pass filter
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,1,2);
plot(o/pi,m);
ylabel('Gain(db)->');
xlabel('(b)Normalised frequency->');

OUTPUT:
Enter the passband ripple(rp):
0.01
Enter the stopband ripple(rs):
0.04
Enter the passband frequency(fp):
1000
Enter the stopband frequency(fs):
2000
Enter the sampling frequency(f):
5000
Enter the beta value:
2

EXPERIMENT-10: COMPUTE THE DECIMATION AND INTERPOLATION OF GIVEN SIGNAL

AIM: TO COMPUTE THE DECIMATION AND INTERPOLATION OF GIVEN SIGNAL

SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE
PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
PROGRAM :

a)decimation
clc;
clear all;
close 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)
stem (m,y);
title('Output Sequence');
xlabel('Time index n');
ylabel('Amplitude');
output:
enter Down-sampling factor: 3
enter number of samples :100

b)interpolation
clc;
clear all;
close all;
L=input('enter the upsampling factor');
N=input('enter the length of the input signal'); % Length should be greater than 8
f1=input('enter the frequency of first sinusoidal');
f2=input('enter the frequency of second sinusoidal');
n=0:N-1;
x=sin(2*pi*f1*n)+sin(2*pi*f2*n);
y=interp(x,L);
subplot(2,1,1)
stem(n,x(1:N))
title('input sequence');
xlabel('time(n)');
ylabel('amplitude');
subplot(2,1,2)
m=0:N*L-1;
stem(m,y(1:N*L))
title('output sequence ');
xlabel('time(n)');
ylabel('amplitude');
output:
enter the upsampling factor
4
enter the length of the input signal
12
enter the frequency of first sinusoidal
5
enter the frequency of second sinusoidal
10

EXPERIMENT-11: DESIGN OF IIR CHEBYSHEV FILTER AND VERIFY FREQUENCY RESPONSE

AIM: TO DESIGN IIR CHEBYSHEV FILTER
SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE
PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
PROGRAM :
A)CHEBYSHEV TYPE 1 LOW PASS ANALOG FILTER
clc;
close all;
clear all;
format long
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]=cheb1ord(w1,w2,rp,rs,'s');
[b,a]=cheby1(n,rp,wn,'s');
W=0:.01:pi;
[h,om]=freqs(b,a,W);
M=20*log10(abs(h));
An=angle(h);
subplot(2,1,1);
plot(om/pi,M);
ylabel('Gain in dB --.');
xlabel('(a) Normalised frequency --.');
title('magnitude response')
subplot(2,1,2);
plot(om/pi,An);
xlabel('(b) Normalised frequency --.');
ylabel('Phase in radians --.');
title('phase response')
output:
enter the passband ripple...0.34
enter the stopband ripple...34
enter the passband freq...1400
enter the stopband freq...1600
enter the sampling freq...10000

B)CHEBYSHEV TYPE 2 HIGH PASS ANALOG FILTER

clc;
close all;
clear all;
format long
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]=cheb2ord(w1,w2,rp,rs,'s');
[b,a]=cheby2(n,rs,wn,'high','s');
w=0:.01:pi;
[h,om]=freqs(b,a,w);
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
ylabel('Gain in dB --.');
xlabel('(a) Normalised frequency --.');
title('magnitude response');
subplot(2,1,2);
plot(om/pi,an);
xlabel('(b) Normalised frequency --.');
ylabel('Phase in radians --.');
title('phase response');
output:
enter the passband ripple...0.34
enter the stopband ripple...34
enter the passband freq...1400
enter the stopband freq...1600
enter the sampling freq...10000

EXPERIMENT-12: Real time implementation of an audio signal using a digital signal

processor.
AIM: To Real time implementation of an audio signal using a digital signal processor.
SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE
PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
PROGRAM :

a)LPF
clc;
close all;
clear all;
% LPF
[dataIn,Fs]=audioread('mary_trumpet_adsr-38724.mp3');
% Filter the signal
fc=500; % make higher to hear higherfrequnecies
% design a Butterworth filter
[b,a]=butter(6,fc/Fs);
subplot(2,1,1);
stem(a);
xlabel('normalised frequency');
ylabel('magnitude in db');
title('magnitude response of audio signal');
subplot(2,1,2);
stem(b);
xlabel('normalised frequency');
ylabel('phase in db');
title('phase response of audio signal');
% Apply the BUtterworth filter
filtered_signal=filter(b,a,dataIn);
% Play the sound
player=audioplayer(filtered_signal,Fs);
play(player);

b)hpf
clc;
close all;
clear all;
% HPF
[dataIn,Fs]=audioread('mary_trumpet_adsr-38724.mp3');
% Filter the signal
fc=500; % make higher to hear higherfrequnecies
% design a Butterworth filter
[b,a]=butter(6,fc/(Fs/2));
subplot(2,1,1);
stem(a);
xlabel('normalised frequency');
ylabel('magnitude in db');
title('magnitude response of audio signal');
subplot(2,1,2);
stem(b);
xlabel('normalised frequency');
ylabel('phase in db');
title('phase response of audio signal');
% Apply the BUtterworth filter
filtered_signal=filter(b,a,dataIn);
% Play the sound
player=audioplayer(filtered_signal,Fs);
play(player);

EXPERIMENT-13: Compute the correlation coefficient for the two given audio signals of same
length using a digital signal processor.
AIM: To Compute the correlation coefficient for the two given audio signals of same length using a
digital signal processor.

SOFTWARE REQUIRED : PC INSTALLED WITH MATLAB SOFTWARE

PROCEDURE: 1.OPEN MATLAB SOFTWARE
2.OPEN NEW FILE
3.TYPE THE PROGRAM
4.RUN THE PROGRAM
5.CHECK OUTPUT IN COMMAND WINDOW AND OUTPUT GRAPHS
PROGRAM :
clc;
close all;
[x,~]=audioread("mary_trumpet_adsr-38724.mp3");
[y,fs]=audioread("Machine-Gun-Automatic-Fire-A-www.fesliyanstudios.com.mp3");
x1=length(x);
y1=length(y);
d=corrcoef(x1,y1);
disp('corrcoef is');d
fc=100;
[b,a]=butter(6,fc/fs);
subplot(2,1,1);
stem(a);
xlabel('normalised frequency');
ylabel('magnitude in db');
title('magnitude response of audio signal');
subplot(2,1,2);
stem(b);
xlabel('normalised frequency');
ylabel('phase in db');
title('phase response of audio signal');
output:
correlation coefficient is d =1