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Your question is not very specific, nor the answer will be. For example what do you mean by 'standard filter parameters'? I assume that you want to make given speech signals sound like ones being reco …

This is probably the craziest, though important question on DSP SE ;) Answer is, because they: look like peak: and tend to have a shelf either in the beginning or the end of frequency range: …

Like I already mentioned in the comment, you might find MATLAB's detrend function useful. Generally it is useful for removal of linear (or piecewise linear) trend: y = detrend(x); Another solution, …

There is a very nice relation between trigonometric functions and exponential function: $$e^{i \theta} = \cos(\theta) + i \sin(\theta)$$ $$\cos(\theta) = \dfrac{e^{i\theta}+e^{-i\theta}}{2} $$ $$\s …

You have a typo in: result(i) = ( b(2)/a(1)) * audio(i); That should be: result(i) = ( b(1)/a(1)) * audio(i);

Basically you never want to use the Transfer Function representation (with b and a) and rather use the Zeros-Poles-Gain (z,p,k). This will allow you to avoid the numerical errors. In your case you mig …

You can try to window the filter with some window function, i.e. Gaussian, to get rid of small coefficients (taper them). Although it won't really work very well and you might really want to think abo …

This approach with notch filter with not work. All clicks are impulse-like sounds and we know that an impulse has frequency content at almost every frequency. What you are trying to do, by applying th …

In case of non-recursive filters (FIR) you use only coefficients stored in b vector. … I suggest you to set a to 1 if you use non-recursive FIR filters. …

This is the FIR filter, although it looks like an IIR. If you calculate the coefficients you get finite impulse response: $h[n]=\delta[n]$ This happens due to zero-pole cancellation: $Y(z)-0.5Y(z)z^{- …

I believe that if you are already using measurements with sweep sine, then the whole procedure of obtaining frequency response is pretty straightforward. Let me describe whole procedure from the begin …

Assuming that you have knowledge of DSP you claim, it is a very simple task. First of all convert the difference block into proper filter representation (negative powers of $z$ reflect delay of samp …

Filters do have a delay (a lag) since they do not act immediately on your signal. …

I suggest you to use fvtool in MATLAB: fvtool(b,a)

The same author, concludes that all-pole filters are easier to manipulate than all-zero filters, due to their filter length. … Morevover, author mentioned that all-pole filters are sufficient approximation than using raw impulse response data. …