# Filtering an Audio File in MATLAB

I am trying to process an audio file in Matlab by filtering out all frequencies except those within $\pm 25\ Hz$ of $523\ Hz$ (as well as its harmonics up to the Nyquist).

The resulting waveform should look like the green wave displayed below (blue being the original):

However, my resulting waveform is entirely zero:

Here is the simple code I wrote:

n = length(s);
f = linspace(-n/2, n/2, n)/(n*Fs);
S = fftshift(fft(s)/n);

filter = zeros(size(S));
f0 = 523;
while (f0<(Fs/2))
filter(abs(f-f0)<25)=1;
filter(abs(f+f0)<25)=1;
f0 = f0+523;
end

s_filter = real(ifft(fftshift(S.*filter)*n));
s_filter = int16(s_filter);

Tspan = n/Fs;
t = linspace(0,n,n)/n*Tspan;

plot(t,s, t, s_filter);

The problem is that the filter array returns only zeros. But what exactly causes that?

Any help would be greatly appreciated.

• Any reason not to use a standard filter design command, such as fir2? – MBaz May 10 '17 at 13:55
• The most simple way is to make the filter by constructing an array which is full of zeros and only contains 1s for the frequencies we want to keep. Pre-defined functions like fir2 are beyond the scope of my DSP course, so I am trying to do it using this simple approach. – Merin May 10 '17 at 14:27
• did you realize that your filter is zero everywhere? – firion May 10 '17 at 14:32
• Yes, as I mentioned in my post this is the problem. But I can't see why it would be zero everywhere; the while loop should introduce 1s for positive and negative harmonics of the 523 Hz frequency up to the Nyquist. – Merin May 10 '17 at 14:38