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IIR filter design question

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neoflash

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IIR filters are all based on sampling inputs and do the z-domain transformation in digital FFs.

Thus, in frequency domain of the H(s) of the IIR filter, we will repeat the low frequency curve in the period of sampling frequency. For example, the curve of a first order low pass filter will repeat itself in frequency domain due to the sampling.

As the result, we are getting different H(s) between digital and analog versions of filter. What is done to fix it?
 

Hi neoflash,

As in any discrete-time system (not only IIR filters) the sampler that converts from continuous time to discrete time must be preceded by an anti-alias filter (except in the case you are sure that there cannot be frequency components able to produce alias). After sampling, the system is unable to distinguish between the different aliased frequency bands.
Regards

Z
 

Do you mean that although there will be additional passband for discrete IIR low pass filter in Fsample, the anti-aliasing filter will remove signal energy there thus the passband will not play any role in the transfer function?

Please help me confirm my thoughts, thank you!
 

neoflash said:
Do you mean that although there will be additional passband for discrete IIR low pass filter in Fsample, the anti-aliasing filter will remove signal energy there thus the passband will not play any role in the transfer function?
Click to expand...

Yes Neoflash, this is correct.
For standard (lowpass) sampling, the input signal has to be limited in frquency to the interval (-Fs/2, Fs/2), where Fs is the sampling frequency.
Passband sampling is also used. In that case the anti-aliasing filter is a passband one.
The frequency axis is wrapped on the unit circle. The intervals (k*Fs -Fs/2, k*Fs+Fs/2) for all integer k are superimposed all together by the sampling process. The anti-aliasing filter must assure that there is signal only the interval of interest.
Regards

Z
 

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    neoflash

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