%%1) Load the 'audio_sample.wav' file in MATLAB [sample_data, sample_rate] = audioread('audio_sample.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 Representation - 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 Representation - 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. order = 7; [b,a] = butter(order,1000/(sample_rate/2),'low'); filtered_sound = filter(b,a,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 Representation - 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 Representation - Filtered Sound') xlabel('Frequency') ylabel('Amplitude')
Fs = sample_rate; % Sampling Frequency (Hz) Fn = Fs/2; % Nyquist Frequency (Hz) Wp = 1000/Fn; % Passband Frequency (Normalised) Ws = 1010/Fn; % Stopband Frequency (Normalised) Rp = 1; % Passband Ripple (dB) Rs = 150; % 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)
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