after applying a filter -i design- to a real signal, it returns complex signal after ifft

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[x,fs]=audioread('some.wav');
n=length(x);
t = ((0:n-1)*(fs/n));
y=fft(x);
f = ((0:n-1)*(fs/n));
y0 = fftshift(y);
f0 = ((-n/2:n/2-1)*(fs/n));
noise = find(abs(y0) == max(abs(y0)));
noiseW= [2*pi*f0(noise(1)) 2*pi*f0(noise(2))];
z1=exp(1i*noiseW(1));
z2=exp(1i*noiseW(2));
z=exp(1i*2*pi*f0);
H=(1-z1*(z.^-1)).*(1-z2*(z.^-1));
out0=y0.*(H.');
out=ifftshift(out0);
signal=ifft(out);
audiowrite('output.wav',final,fs);
  2 Comments
Hamza AKYILDIZ
Hamza AKYILDIZ on 14 May 2021
some.wav file is a real signal. after i apply the filter H to it, i expect it to be still real after ifft.

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Accepted Answer

Paul Hoffrichter
Paul Hoffrichter on 14 May 2021
Your imaginary part only is non-zero due to the usual floating point roundoffs and truncations. Fix this using round:
signal=ifft(out);
signalRound = round(signal, 10);

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