Filter odd or even harmonics with notch or inverse notch filter Hi i had the following question.
I have a signal containing a 200Hz sine wave and it's odd and even harmonics (no other frequencys or disturbing signals are contained).
What i'm looking for is a kind of filter which is able to seperate the odd harmonics from the even ones.
The end goal is to be able to determine the amplitude and phase of each single of the odd harmonics as accurate as possible.
I know that you normally would use an FFT for this purpose but i'm trying to figure out if this wouldn't also be possible with a filter instead.
So i thought about two options:

Use a parallel circuit of notch filters (one for each harmonic to be filtered) to cut out the even harmonics i don't want to use and seperate the odd harmonics from eachother.

Do exactly the opposite and use &quot;inverse notch filters&quot; to filter out the odd harmonics i want to use and seperate them from eachother by using a parallel circuit of &quot;inverse notch filters&quot; (one for each harmonic i want to filter out and take amplitude and phase from).


I've added a little sketch as png to make things clearer.

The problem is that it's very difficult to find understandable information about notch filter or inverse notch filter. (Inverse notch filter is not the same as a peak filter)
For notch filters i only find spurious and short information like these:
https://zone.ni.com/reference/en-XX/help/371325F-01/lvdfdtconcepts/notch_peak_filters/
https://www.onesdr.com/2019/11/21/fm-notch-filters-why-you-need-one-with-most-sdrs/
https://www.youtube.com/watch?v=tpAA5eUb6eo
And the only source of information i found about inverse notch-filter was this paper i mentioned (on top of the second page of the paper):
https://www.scirp.org/pdf/_2013060615443504.pdf
The problem is i would need way more information about both filter types and how derive them from simpler filter types and calculate their transfer function and Z-transform.
So my question is (if i'm not completely on the wood path) which of the both approaches described you would prefer and can you give me some good sources of clear and understandable information about the two mentioned filter types and thier implementation in hardware ?
Our would you recommend me a completely different filter type for this task ?
Greetings
What you are looking for are what we, in the audio space, call comb filters.  Comb filters may or may not have a feedback path, just like FIR and IIR filters. In fact, there is a generalized theory about designing comb filters based simply on designing any digital filter.
In fact, you can think of every LTI digital filter as being a comb filter where the teeth of the comb are spaced apart by the sampling frequency.  To space the teeth apart by a frequency spacing much less than the sampling frequency, you need a delay element that is longer than a single-sample delay ($z^{-1}$).  If a delay with precision of an integer number of samples is good enough to tune your comb filter, good for you.  But if it isn't, you need to deal with interpolation and precision delay.