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 !
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