How to create low pass filter for audio file?

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

Answer 1

[Star Strider]

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.

[Daemian]

Thanks!!

[Star Strider]

My pleasure!

[Daemian]

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]

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.

Answer 2

[Tengsim Momin]

how to filter eeg signal with(0.1db-80db) low pass filter using fdatool