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Somebody help me to explain this code. Also mention about how to obtain snr from this code.

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Aiswarya Babu
Aiswarya Babu on 14 Jun 2021
Answered: Arthi Sathyamurthi on 27 Jul 2021
%%1) Load the 'audio_sample.wav' file in MATLAB
[sample_data, sample_rate] = audioread('test.wav');
% a. Plot the signal in time and the amplitude of its frequency
% components using the FFT.
sample_period = 1/sample_rate;
t = (0:sample_period:(length(sample_data)-1)/sample_rate);
subplot(2,2,1)
plot(t,sample_data)
title('Time Domain - Unfiltered Sound')
xlabel('Time (seconds)')
ylabel('Amplitude')
xlim([0 t(end)])
m = length(sample_data); % Original sample length.
n = pow2(nextpow2(m)); % Transforming the length so that the number of
% samples is a power of 2. This can make the transform computation
% significantly faster,particularly for sample sizes with large prime
% factors.
y = fft(sample_data, n);
f = (0:n-1)*(sample_rate/n);
amplitude = abs(y)/n;
subplot(2,2,2)
plot(f(1:floor(n/2)),amplitude(1:floor(n/2)))
title('Frequency Domain - Unfiltered Sound')
xlabel('Frequency')
ylabel('Amplitude')
% b. Listen to the audio file.
% sound(sample_data, sample_rate)
%%2) Filter the audio sample data to remove noise from the signal.
Fs = sample_rate; % Sampling Frequency (Hz)
Fn = Fs/2; % Nyquist Frequency (Hz)
Wp = 40/Fn; % Passband Frequency (Normalised)
Ws = 150/Fn; % Stopband Frequency (Normalised)
Rp = 3; % Passband Ripple (dB)
Rs = 60; % Stopband Ripple (dB)
[n,Ws] = cheb2ord(Wp,Ws,Rp,Rs); % Filter Order
[z,p,k] = cheby2(n,Rs,Ws,'low'); % Filter Design
[soslp,glp] = zp2sos(z,p,k); % Convert To Second-Order-Section For Stability
figure(3)
freqz(soslp, 2^16, Fs) % Filter Bode Plot
filtered_sound = filtfilt(soslp, glp, sample_data);
sound(filtered_sound, sample_rate)
t1 = (0:sample_period:(length(filtered_sound)-1)/sample_rate);
subplot(2,2,3)
plot(t1,filtered_sound)
title('Time Domain - Filtered Sound')
xlabel('Time (seconds)')
ylabel('Amplitude')
xlim([0 t1(end)])
m1 = length(sample_data); % Original sample length.
n1 = pow2(nextpow2(m1)); % Transforming the length so that the number of
% samples is a power of 2. This can make the transform computation
% significantly faster,particularly for sample sizes with large prime
% factors.
y1 = fft(filtered_sound, n1);
f = (0:n1-1)*(sample_rate/n1);
amplitude = abs(y1)/n1;
subplot(2,2,4)
plot(f(1:floor(n1/2)),amplitude(1:floor(n1/2)))
title('Frequency Domain - Filtered Sound')
xlabel('Frequency')
ylabel('Amplitude')

Answers (1)

Arthi Sathyamurthi
Arthi Sathyamurthi on 27 Jul 2021
This code performs the FFT on the audio signal, Chebyshev filters the same and performs FFT on the filtered signal. And further, since no information was given about the signal snr must be obtained, you can use the SNR function of MATLAB as snr(x, y) to return the signal-to-noise ratio (SNR) in decibels of a signal, x, by computing the ratio of its summed squared magnitude to that of the noise, y.

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