MATLAB Simulation with Filtering Audio Recording

One of the continuing research issues in the realm of speech recognition and audio analysis is the extraction of singing voices from music. This issue has numerous applications throughout the music world, such as identifying music structure, recognizing lyrics, and recognizing singers. In this task we have to design the filter to enhance the vocal or to remove the vocal from the song so that we have to make the karaoke effect. In 2nd part we modulate the signal of 8000kHz and demodulate the signal with the presence of noise signal. In 3rd part we have to simulate the sound localization using multiple microphones. Although numerous studies have been done here on separation of voice from background, there has been less research on the singing voice. Signal processing is done via audio circuits, which effectively convert sound waves into electrical signals that can then be amplified, filtered, or mixed. These signals can be recorded and recreated as well. One component of this system is audio filters, which act as amplifiers or passive circuits with different frequency responses. These filters, like microphones and speakers, are vital components of an audio system's foundation. They have the ability to amplify or attenuate a wide spectrum of frequencies from an audio input. A frequency of the radio carrier is altered in accordance with the loudness of the incoming audio stream to create a frequency modulated signal. The new radio frequency signal travels from up and down frequency as the audio signal gets modulated it onto radio frequency carrier. The quantity by which the signal rises and falls is critical. It's called the deviation, and it's usually expressed as a number of kilohertz. 

% clear data and figures
clc
clear
close all
% Read Audio
[y,Fs] = audioread('Song.mp3');
% extract Sample from song
y=y(1:1.5*10^6,1);
% timet=0:(1/Fs):(1/Fs)*(numel(y)-1);
% Plot Sample
figure
plot(0:(1/Fs):(1/Fs)*(numel(y)-1),y)
grid on
xlabel('time')
ylabel('signal')
title('Input Source Signal')
% playing Song
sound(y,Fs)
pause(t(end)+5)

The first step in MATLAB audio signal processing is to import the audio recording into MATLAB. This can be done using the built-in audioread function, which supports a wide range of audio file formats such as WAV, MP3, and FLAC. The function returns the audio data as a vector and the sample rate of the audio file.

Once the audio data is imported, it's time to apply filtering techniques to the signal. Filtering is the process of altering the frequency content of a signal, and there are several different types of filters that can be applied to an audio signal, such as high-pass, low-pass, band-pass, and notch filters.

To apply a filter to an audio signal, MATLAB provides several built-in functions such as fir1, fir2, and iirnotch. These functions allow you to design and apply digital filters to an audio signal. For example, the fir1 function can be used to design a low-pass filter with a specific cutoff frequency, and the filter function can be used to apply the filter to the audio data.

Once the filtering process is complete, the filtered audio signal can be visualized using the built-in plot function. This function plots the amplitude of the audio signal over time, allowing you to see the effect of the filter on the waveform.