[Mne_analysis] Low pass filtering questions

[Krieger, Donald N.]

Thanks for responding, Alex.

Please bear with a further question:
If the low-pass is set at 40 Hz, then the cos^2 drop off is applied beginning at the 40 hz components and falling to zero at 45 ?  Or is it applied beginning at something like 38 hz so that the fourier coefficients are attenuated by a fact of 2 at 40 ?

And pardon this stupid question: Does this produce a gain of 0.0 for all higher frequencies in the filtered signal above 45?  Or is there some kind of ringing which occurs?

Thanks again.

Don

Don Krieger, Ph.D.
Department of Neurological Surgery
University of Pittsburgh

________________________________________
From: Alexandre Gramfort [gramfort at nmr.mgh.harvard.edu]
Sent: Thursday, March 14, 2013 6:31 AM
To: Krieger, Donald N.
Cc: mne_analysis at nmr.mgh.harvard.edu
Subject: Re: [Mne_analysis] Low pass filtering questions

hi Don,

MNE uses what is called an overlap-add approach. Indeed fft is applied to
buffers of 2048 samples by default. The filter is applied in the
frequency domain
using a transition band of 5Hz. The benefit is that it is much faster
and you only
need to store a fraction of the data in memory. FYI the MNE-Python code
uses it too but also supports IIR filters.

HTH
Alex

On Tue, Mar 12, 2013 at 4:19 PM, Krieger, Donald N. <kriegerd at upmc.edu> wrote:
> Hi Everyone,
>
>
>
> I use the low pass filtering build into mne_process_raw().
>
> By default, it appears to use a window of 2048 points and a taper of 5 Hz.
>
>
>
> I would like to make sure I understand what the filter is doing.
>
> I assume that these mean the following:
>
>
>
> 2048 points at a time are run through an FFT.
>
> A cosine^2 window is applied to the fourier coefficients beginning at a
> point where, close to the selected ½ amplitude point, for the entire set of
> coefficients?
>
> Or perhaps the fourier coefficients are tapered down to zero over a 5 hz
> interval?
>
> I’m definitely hazy on this part.
>
>
>
> The resulting “filtered” fourier coefficients are then run through an
> inverse fourier transform to replace the original time domain signal with a
> low pass filtered one.
>
>
>
> What is the correct understanding of this?
>
> And is there information about the transfer function of the filter?
>
>
>
> For instance, what is the fall off per octave depending on the selected ½
> amplitude point and width of the taper?
>
>
>
> Thanks for any help you can provide.
>
>
>
> Regards,
>
>
>
> Don
>
>
>
> Don Krieger, Ph.D.
>
> Department of Neurological Surgery
>
> University of Pittsburgh
>
>
>
>
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