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1

Regarding the prewarping, it gets significantly challenging to match the analog shape as we approach the Nyquist boundary since the bilinear transform warps infinity in s to z=-1 in z (Nyquist). Prewarping implies the filter design is being mapped from s to z which wouldn't be my first choice for filter design unless I specifically needed to "copy the ...


1

is it possible to have a digital filter with a sampling frequency of 44.410kHz and a Fc = 20kHz? Yes it is possible... . I use MIM (Magnitude Invariance Method) for this. Here's plot showing 2nd order LPF responses for fc=20kHz-25kHz (1kHz step) @ 44.kHz sampling: MIM method works well for even lower sample rate implementations but, there are also issues ...


0

I suppose that the $n$ points (hereafter, the chunk) precisely divide the base period (which you don't know yet) evenly. You are lucky. You can verify that by concatenating the $n$ discrete samples (or possibly $n-1$ consecutive ones) by an integer number $K$ of (periodic) repetitions (copy-paste, several times, hereafter the chain), with $Kn\ge m$ (or $...


2

One potential option would be to split the lowpass filter into the FIR part and the pureley recursive IIR part. $$H(z) = H_1(z) \cdot H_2(z) = \frac{1}{1 - a1z^{-1} - a2z^{-2} - a3z^{-3} - a4z^{-4}} \cdot \left [ b0 + b1z^{-1} + b2z^{-2} + b3z^{-3} + + b4z^{-4} \right ]$$ You still have to apply the recursive part to each sample in the upsampled domain, but ...


1

filtfilt() applies forwards, then backwards filtering using any iir filter you feed it. As such, it only applies to finite length file processing, not realtime stream processing. fir1 is a method to design fir filters that can subsequently be applied using filter, conv or (unusually) filtfilt.


5

Kalman filters really aren't that special, and you seem to be missing the point of a Kalman filter. A Kalman filter is really just a generally time-varying, generally IIR, generally multi-input multi-output filter that's been designed using a specific procedure. Can we deem that traditional filters such as FIR and low-pass filter are designed to be used ...


0

There are 2 steps for the solution Step 1 - Design some kind of low-pass filter to measure the DC component. In order to perform step 1 you need to answer a few questions : What do you mean by DC? For example, measure the DC for the last 10 seconds? What is the input sampling rate? What is the desired output sampling rate? You need to provide this ...


0

You're not actually mixing two complex exponentials, because otherwise you'd get a complex exponential with the sum of the frequencies of the individual complex exponentials. That was already mentioned in a comment by Marcus Müller. What you are doing is compute a signal with real part $\cos(\omega_0t)\cos(\omega_1t)$ and with imaginary part $\sin(\omega_0t)\...


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