I’m a little unsure of my maths here, but if i wanted to filter out, or attenuate a given frequency out of a signal how would i go about implementing it.
Do i need to perform some kind of time->frequency domain transform, and then work with the value… I’m just not sure how to filter a time domain signal, forgive the somewhat vague question wording, but this is completely outside my area.
assume that the value in pretend code is
float timedomain[x], where x = time, do i need to laplace transform the signal to work with it? (wish i had my DSP books with me here – might have to drive and get them)
- ArchfileX asked 11 years ago
I would just recommend looking at http://www.harmony-central.com/Computer/Programming/ or http://www.musicdsp.org/
I am using fmod, but I tried the Parametric EQ, I am writing this for some research, but its not possible to replicate the effect (in subjects) with the paramEQ, my supervisor is suggesting that i need to make more complex filters than a range of ParamEQ filters, so i’m trying to get a jump on it and learn a bit about implementing the filters themselves in time domain samples.
I suspect that the maths for this is likely to be well beyond me.
- ArchfileX answered 11 years ago
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