DSPRelated.com
Forums

How to design and apply a frequency filter on top of a previous filter to get a given filter?

Started by n.maisonneuve October 21, 2009
Hello everybody,

I have a microphone with a given internal frequency filter IF(x) that
I found experimentally in lab.
I have around 10 points defining this filter from 100 hz to 16 Khz. (I
can improve the granularity if required)

I want to apply a  A weighting filter AF(x) on the source ofthe
microphone.
but the output will be wrong if I apply directly the A filter from the
output of the microphone due to its internal filter.

How can I correct that?
How can I design and apply an additional filter that correct this
internal filter to give me a response similar to a A weighting filter
as if the source was not filtered?

I though about manually subtracting  IF from AF to get a correct
filter CF and apply it.
CF(x)=AF(x) - IF(x).

So I will have a function with also 10 points. but how do i apply CF
to a temporal signal?
and is it the right approach?

Thanks in advance for your help!

Nicolas
On Oct 21, 6:33&#4294967295;am, "n.maisonneuve" <n.maisonne...@gmail.com> wrote:
> Hello everybody, > > I have a microphone with a given internal frequency filter IF(x) that > I found experimentally in lab. > I have around 10 points defining this filter from 100 hz to 16 Khz. (I > can improve the granularity if required) > > I want to apply a &#4294967295;A weighting filter AF(x) on the source ofthe > microphone. > but the output will be wrong if I apply directly the A filter from the > output of the microphone due to its internal filter. > > How can I correct that? > How can I design and apply an additional filter that correct this > internal filter to give me a response similar to a A weighting filter > as if the source was not filtered? > > I though about manually subtracting &#4294967295;IF from AF to get a correct > filter CF and apply it. > CF(x)=AF(x) - IF(x). > > So I will have a function with also 10 points. but how do i apply CF > to a temporal signal? > and is it the right approach? > > Thanks in advance for your help! > > Nicolas
If your filters are expressed in frequency domain then you should multiply them in that domain. This is an equalising filter, you might look up Wiener filetsr that are often used to undo the effect of a known filter. One way to apply the filter is to take its FT and convolve, there is a disussion of this on our web site: www.bores.com/courses/intro/filters/4_fir.htm Chris ============== Chris Bore BORES Signal Processing www.bores.com