I tried to implement a simple LP filter on an audio file, but I'm slightly confused by the apparent shifting of the signal in the time domain. What have I overlooked / misunderstood? Thanks.
filename = 'Raw_Wav_20180425_000226.wav';
[y,fs] = audioread(filename);
subplot(1,2,1)
plot(t,y)
xlabel('Time (s)')
ylabel('Amplitude')
title('Raw')
fmax1 = 40; %end of pass band
fmax2 = 60; %end of transition band
nt = length(y);
t = [0:nt-1]/fs;
y = y-mean(y);
dF = fs/nt;
f = -fs/2:dF:fs/2-dF;
yft = fftshift(fft(y));
I = find(abs(f)>fmax2);
yft(I) = 0;
I = find( abs(f)>fmax1 & abs(f)<=fmax2 );
yft(I) = yft(I) .* hanning(length(I));
y = ifftshift(ifft(yft));
subplot(1,2,2)
plot(t,y)
xlabel('Time (s)')
ylabel('Amplitude')
title('Post-filter')

 Akzeptierte Antwort

Sugar Daddy
Sugar Daddy am 12 Jun. 2020

0 Stimmen

As you have taken fft first and then fftshift, so if you want to do inverse of this, you need inverse of fftshift first as it is performed at the end. so the order of inverse must be
  1. ifftshift
  2. fft

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