How to create low pass filter for audio file?

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Daemian
Daemian 2015-2-24
回答： Tengsim Momin 2016-3-28
How do i create a low pass filter for an audio file?
I would like to keep 20 Hz and below Sampling rate 8000 Hz
Thanks

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Star Strider
Star Strider 2015-2-24
You gave enough information for the Signal Processing Toolbox (with a bit of help from me) to be able to design your filter:
Fs = 8000; % Sampling Frequency (Hz)
Fn = Fs/2; % Nyquist Frequency
Fco = 20; % Passband (Cutoff) Frequency
Fsb = 30; % Stopband Frequency
Rp = 1; % Passband Ripple (dB)
Rs = 10; % Stopband Ripple (dB)
[n,Wn] = buttord(Fco/Fn, Fsb/Fn, Rp, Rs); % Filter Order & Wco
[b,a] = butter(n,Wn); % Lowpass Is Default Design
[sos,g] = tf2sos(b,a); % Second-Order-Section For STability
figure(1)
freqz(sos, 2048, Fs) % Check Filter Performance
Then use the filtfilt function to filter your signal.
I chose a Butterworth design, but if you want a steeper cutoff, consider using a Chebychev design instead. You may want to experiment with the design to get the exact result you want with respect to your signal. See the relevant documentation for details.
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Daemian
Daemian 2015-2-24
Hi, after i filter if i want to convert it to ftt, do i do it this way?
Fs = 8000; % Sampling Frequency (Hz)
Fn = Fs/2; % Nyquist Frequency
Fco = 20; % Passband (Cutoff) Frequency
Fsb = 30; % Stopband Frequency
Rp = 1; % Passband Ripple (dB)
Rs = 10; % Stopband Ripple (dB)
[n,Wn] = buttord(Fco/Fn, Fsb/Fn, Rp, Rs); % Filter Order & Wco
[b,a] = butter(n,Wn); % Lowpass Is Default Design
[sos,g] = tf2sos(b,a); % Second-Order-Section For STability
%figure(1) %freqz(sos, 2048, Fs) % Check Filter Performance yd = filtfilt(sos, g, myRecording); % Filter ‘xd’ To Get ‘y’
yd_len = length(yd); % Length Of ‘wav’ File Data
Ts = 1/Fs; % Sampling Time
Tv = linspace(0, yd_len-1, yd_len)*Ts; % Time Vector
y_fft = abs(fft(yd)); %Retain Magnitude
Nsamps = length(yd);
y_fft = y_fft(1:Nsamps/2); %Discard Half of Points
f = Fs*(0:Nsamps/2-1)/Nsamps; %Prepare freq data for plot
plot(handles.normalfft,f, y_fft)
xlim(handles.normalfft,[8 12])
Star Strider
Star Strider 2015-2-24
I’m not certain what you’re doing. The usual procedure is to divide the output of the fft function by the length of the vector to normalise the amplitudes. I also don’t understand your frequency vector ‘f’ calculation (it doesn’t look correct to me), but if it works for you, go for it!
I follow the amplitude and frequency vector calculations in the documentation for the fft function, rather than creating my own calculations. It’s easier, and it works. I suggest you do that.

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Tengsim Momin
Tengsim Momin 2016-3-28
how to filter eeg signal with(0.1db-80db) low pass filter using fdatool

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