33
votes
Accepted
How many taps does an FIR filter need?
Citing Bellanger's classic Digital Processing of Signals – Theory and Practice, the point is not where your cut-off frequency is, but how much attenuation you need, how much ripple in the signal you ...
21
votes
Accepted
Why do digital filters work?
Consider a discrete-time input signal of the form:
$$ x[n] = \cos(\omega_0 n) ~~~,~~~-\infty < n < \infty, ~~~~~ n\in \mathcal{Z}$$
where the radian frequency $\omega_0$ is set between 0 and $\...
19
votes
Accepted
Transfer function of second order notch filter
For digital notch filters, I like to use the following form for a notch filter at DC ( $ \omega $=0):
$$ H(z) = \frac{1+a}{2}\frac{(z-1)}{(z-a)} $$
where $a$ is a real positive number < 1. The ...
15
votes
How many taps does an FIR filter need?
For a quick and very practical estimate, I like fred harris' rule-of-thumb:
$$ N_{taps} = \frac{Atten}{22*B_T}$$
where:
Atten is the desired attenuation in dB,
$B_T$ is the normalized transition ...
12
votes
What is the maximum output of an IIR filter?
I would like to know,if there is any way to estitmate what could be the maximum limits of the output
Yes. It's
$$y_{max} = x_{max} \cdot \sum_{n = 0}^{\infty} |h[n]| $$
I.e. the maximum gain is the ...
12
votes
Accepted
Why don't unit circle poles lead to infinite amplitude response for Butterworth lowpass?
You've made an understandable mistake. You are probably looking at this picture:
That is not the unit circle, and it isn't even in the $z$ domain.
What you are looking at is the locations of the ...
11
votes
Accepted
Exponential weighted moving average time constant
If I understood you correctly, you want to compute the value of $\alpha$ that results in a specified 3dB cut-off frequency for an exponentially weighted moving average filter. If you square your last ...
11
votes
Digital filter coefficients from low-pass to high-pass
You can apply a so-called all-pass transformation to a discrete-time low-pass prototype filter in order to convert it to other standard filters (such as high-pass, band-pass, and band-stop). This is ...
9
votes
Accepted
Is a high-passed signal the same as a signal minus a low-passed signal?
In general you can't simply subtract a low-pass filtered version of a signal from the original one to obtain a high-pass filtered signal. The reason is as follows. What you're actually doing is ...
8
votes
Precise Centre frequency of an All-pole digital filter
For a filter consisting of a complex conjugate pair of poles, the $z$-domain transfer function is:
$$H(z) = \frac{a}{\left(1-r(\cos{\theta}-i\sin{\theta})z^{-1}\right)\left(1-r(\cos{\theta}+i\sin{\...
8
votes
Accepted
What is the largest "safe" order for the digital Butterworth filter of a given signal?
One cause is that higher order Butterworth filters have poles closer to the unit circle. This nearby infinite gain point increases the likelihood of numerical instabilities. (e.g. rounding/...
8
votes
Accepted
How Come the Low Pass Filter in Sobel Operator Isn't Normalized?
The answer is simple, the Sobel Filter is a composition of Lows Pass Filter (LPF) and High Pass Filter (HPF). The composition is done by convolution.
Now, indeed the LPF presented above $ {\left[ 1, 2,...
7
votes
Accepted
Complex low pass filters
A low pass filter has a frequency response that satisfies
$$|H(\omega)|\approx 0,\quad |\omega|>\omega_c\tag{1}$$
where $\omega_c$ is the cut-off frequency. A complex low pass filter must also ...
7
votes
How many taps does an FIR filter need?
Adding to the accepted answer, a few additional references. I won't write the formulas which can be involved. Those formulae mostly yield rule-of-thumbs or approximations to start from. You can fiddle ...
7
votes
Accepted
Protect an IIR filter from being reverse-engineered
Short answer:
You can't. If an attacker can insert a signal that covers the whole bandwidth (e.g. a white signal, or at least one that has no spectral zeros) into the system (and he can do that over ...
7
votes
Accepted
3 dB cut-off frequency of first-order IIR high-pass filter
A discrete-time first-order high pass filter with unity gain at Nyquist and a zero at DC is described by the following difference equation:
$$y[n]=\frac{1+\alpha}{2}\big(x[n]-x[n-1]\big)+\alpha y[n-1],...
6
votes
Accepted
Estimating an Image Filter from Examples
I will talk about the Linear Spatially Invariant (LSI) Model.
Since the Filter (System) is LSI it can be described in the domain of Fourier Transform:
$$ Y \left( \omega \right) = H \left( \omega \...
6
votes
Why do digital filters work?
You have probably used filtering a lot already. A moving average is a filter!
Think of general filtering as performing a fancy moving average where instead of averaging every component in a window ...
6
votes
Make a signal that fits another the best possible with a limitation in the 2nd derivative
Hmmmmmmmmm, interesting question.
Since you want to use the second derivative as your criteria, it would seem that you would want to have the maximum second derivative absolutie value for as short of ...
6
votes
Accepted
What's Logic Behind the Construction of Sobel's Filter in Image Processing?
A first rationale is to be very short, as there was a time when computing on images was expensive. Then, a contour or an edge often present a fast variation in image intensities, that can be enhanced ...
6
votes
Numerically Stable IIR filter
First of all, what is the order of your IIR filter? The highest order I have ever used was an order-10 IIR filter for a control loop application. I feel like it is unlikely that you need more that ...
6
votes
Filtering Angular Measurements
If a first-order IIR will do, modify that slightly, and you're done.
So the usual first-order low-pass filter can be defined as $y_n = h(\theta_n)$ such that $y_n = y_{n-1} + a(\theta_n - y_{n-1})$. ...
6
votes
Accepted
Take FFT of audio represented in PDM
In a PDM microphone, the sigma-delta convertor pushes the noise to frequency regions above the 0-20kHz spectrum. So if you'd take the FFT of the signal that comes directly from the PDM microphone, the ...
6
votes
Accepted
Designing digital low pass filter with low pass-band group delay
Minimum phase filters will not give you a near constant group delay. You can design a non-linear phase FIR filter with a linear desired passband phase with a specified group delay that is smaller than ...
6
votes
Why must the order of a Band-Pass and Notch filter always be even?
In principle there is no reason why the filter order of a general bandpass or bandstop filter must be even. Such a restriction is a consequence of a specific design procedure. In classic IIR filter ...
6
votes
High Dynamic Range FIR Filters
The problem lies in the formulation of the desired response, and especially in the "don't care" region, which is extremely wide for the chosen filter length. Even though I can't give any ...
5
votes
Why do digital filters work?
I think you're looking for intuition as to why you get a certain frequency domain behavior when computing a weighted sum of input samples. As you know, the output signal of a causal length $N$ FIR ...
5
votes
Why do digital filters work?
To the useful answers that have been added so far, I would like to add, on the point of intuition, that filtering works because it is based on Wave theory and specifically, the interaction of waves. ...
5
votes
Accepted
Problems with implementation of a band-stop filter on an MCU (dsPIC) using fixed-point arithmetic
The denominator (recursive coefficients Ai) look OK: the poles of your system are at 45 degree angles ($\pi/4$), with magnitude 0.68 (which is not very aggressive for a notch filter; in my opinion ...
5
votes
Accepted
determining type of filter given its pole zero plot
You'd have to figure out the frequency response of the filter. Here are two methods. I prefer Method 2 because it's quick and dirty, and you don't really care about the exact gain values in the ...
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