Hi all,
I have some basic questions regarding a low-pass filter. I have never studied
audio signal processing that's why my questions might seem trivial for most
of you.
So I have to design a low-pass filter. Here is the way I do it. I have my matrix
with the signals signalMatrix. I create a FFT so I do :
newVector = fftsfift(fft(signalMatrix));
Here is my first question : what do I get now ? I mean when I plot this vector
what is the X-axis ? I suppose the Y-axis is the frequency range. I think the
X-axis is not the time because it is symmetric ... And I don't understand
why is it (always?) symmetric ?
The other group of questions relates to the low-pass filter. Here's the way
I do it : I parse the newVector and I change every frequency that is greater
then my cut-off frequency to the cut-off frequency. Then I do an ifft(newVector)
to transform it back to a signalMatrix. Is this a low-pass filter ?
Could you please help my out ?
If there are some books that explain this for the newcomers in this filed would
recommend please share it with me ?
And yes this is a collage project and I want to understand how things work not
just do it for the sake of a lab.
Thank you !
Attila