First time asking a question on here as I have hit that brick wall now.
I am trying to filer an audio signal that I have, for example I only want to know between 100 and 1000 Hz, instead of the whole thing. I can try a limit of the axis, but of course the data is still there when I further process it.
I have read through many, many, many examples on here and can't seem to figure it out at all. I have looked at bandpass filtering, something to do with butter....
My code uses a FFT to look at the full frequency spectrum, with the audio clip available at http://bit.ly/2RlPl6p . This is just the FFT, I haven't included any filering bits.
File = 'Loud.wav';
[x,fs] = audioread(File);
lpad = 8*length(x);
xdft = fft(x,lpad);
xdft = xdft(1:lpad/2+1);
xdft = xdft/length(x);
xdft(2:end-1) = 2*xdft(2:end-1);
freq = 0:fs/lpad:fs/2;
Any help at all would be greatly appreciatied!
(I am not a student doing homework)
This is the wav file stats.