How to obtain the Fourier transform of a filtered signal?
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I am working on a project in which i have to analyze the walking signals of the person. I obtained the signals with the help of an accelerometer. Then I applied low pass butterworth filter of cut-off frequency 0.2Hz and sampling frequency 1Hz. I have obtained the filtered signals in time domain, but i need the signals in frequency domain. From the help of this site I got to know the syntax of fft i.e Y = fft(x) I applied it to my filtered signals and obtained a very unusual result. Please help me in performing fourier transform on the signals. I am attaching my results below.
the fig attached consists of the x component of my filtered signal(i have three columns of readings-x,y and z) and the signal achieved after fft.
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Adam
2014년 8월 1일
편집: Adam
2014년 8월 1일
fft gives a complex result. That is what you are seeing in that figure - real vs imaginary on the x and y axes. You can view the spectrum using the abs function.
e.g.
y = fft(x, nFFT);
plot( 2*abs(Y(1:nFFT/2+1) ) );
Obviously you can also calculate the frequencies to plot those on the x-axis. The Matlab help page for fft gives an example of this.
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