Codes:
%Sampling Theorem
clear all;
close all;
clc;
f=input(‘Enter frequency’);
%T=1/f;
fs1=input(‘Enter the sampling frequency’);
t=0:0.1:100;
t1=0:10:500;
x=sin(2*3.14*f*t); %2*pi*f*t
subplot(2,1,1);
plot(t,x);
y=sin(2*3.14*f*t1/fs1);
subplot(2,1,2);
stem(t1,y);
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A headphone mike would have done a better job for recording
If you take f=100Hz, fs=10Hz, you are voilating Niquist Theorem as fs should be >= 2*f, right?
my friend, you really need a better microphone. asap.
This is incorrect
#conversion of analog to digital signal with time t=nTs is sampling theorem ie fs>=2fm.