Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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Peaking filter (loudness) cutoff frequency in an equalizer

An equalizer we have consists of Loudness ,Bass ,Mid and Treble .The loudness is nothing but a Peaking filter with cutoff frequency 20Hz .I read this in a book that outlines the EQ design. I was ...
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Developing Lead Compensator to Decrease Settle Time

This is a follow up to this question. I've approximated the transfer function for a system to be H = zpk([0.012 -1.05 18],[-0.22 -0.22 -45 -1000],10000);. My goal ...
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How to choose median filter width?

In my signal processing pipeline, I normally expect data sampled at 100Hz, on which I then apply a median filter of width 10. My question is: How should I change the width for data sampled at ...
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Tuning a PID Controller

I have the following transfer function $$10000 \cdot\frac{(s-0.012)(s+1.05)(s-18)}{(s+0.22)^2(s+45)(s+1000)}$$ For which I am trying to tune a PID controller for. I'm using the pidTuner in MATLAB to ...
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Recursive, non-recursive systems; FIR, IIR systems

I am confused with the classifications of an LTI system as recursive or non-recursive systems and FIR or IIR systems. I understood what the FIR and an IIR systems are, but is it correct to say that ...
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Practical IIR filter implementation

I'm trying to implement a digital IIR filter on an FPGA and would be happy for some inputs regarding the actual digital implementation. I don't have a lot of experience with FPGAs and digital filters ...
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Chebyshev Filter Transfer Function

I'm trying to derive the transfer function for Chebyshev filter. $$|H(\Omega)|^2=\frac{1}{\sqrt{(1+\epsilon^2T_n^2(\frac{\Omega}{\Omega_c})}}$$ where $$T_n(x)=\cos(N\cos^{-1}(x)) \forall x \le 1$$ $$...
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Where do RBJ's analog shelving filters come from?

In RBJ's Audio EQ Cookbook, the analog prototype for a low shelf filter is given as such: $$H(s)=A\cdot\frac{s^2+\left(\frac{\sqrt{A}}{Q}\right)\cdot s + A}{A\cdot s^2 + \left(\frac{\sqrt{A}}{Q}\right)...
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Formula for designing Lynn's low pass filter

For the second order Lynn's low pass filter, the general form of the transfer function is: $$ H(z)=\frac{(1−z^{-m})^2}{ (1−z^{-1})^2 } $$ where $m$ is a positive integer. The gain for this is $m^2$, ...
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Designing digital low pass filter with low pass-band group delay

I'm interested in specifying a digital low pass filter with low (and near constant) pass band group delay. I tried python's spicy.signal.minimum_phase to transform ...
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How to produce uniformly distributed low-frequency noise?

I want to design noise for the sake of modulation and it should have a frequency of under 1 Hz. This seems impossible to do for a frequency that low. Here is my Matlab script. The lowest frequency ...
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Butterworth-like filter with no zero at Nyquist?

I have an old digital synthesizer that has a Butterworth-like 2nd order filter with resonance. However, with high frequency tones it appears there is no steep cutoff near Nyquist (I am sure it is the ...
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Highpass filter design that is better than the Moving Difference filter

The question is actually from the problem set 1.3 of the "Wavelets and Filter Banks" book (authors Nguyen/Strang). Problem 4: Invent a Highpass filter K with 3 or 4 taps (coefficients) that ...
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How to design a good IIR smoothing filter?

I need to design a smooting filter for computing the envelope of a signal. My specs in Matlab are ...
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Is Kaiser-Bessel filter good for notch filtering with time-sensitive data?

I'm working with ECG signal processing and want to implement digital notch filter. Would Kaiser-Bessel filter be optimal for signals sensitive to time domain disruptions or should I use different ...
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Calculate ScaleVlues of IIR Filter manually from sosMatrix of MATLAB

while using "design" in matlab to design filters, at the output we can see a ScaleValues matrix that is used to compensate for overflow problem. my question is that how this matrix is ...
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Explain the relationship between Tikhonov regularization, SVD , least squres, and the Wiener filter

I found in the Wikipedia site for Tikhonov regularization that the SVD for a Tikhonov regularized problem take us to the least squares regularized solution: \begin{eqnarray} \hat{x}= V D U^T b, \end{...
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How to find Cutoff frequencies (f1 &f2) for a narrow band pass filter for a particular central frequency?

This is a basic filter question I think. I want to define a narrow bandpass filter's cutoff frequencies (f1 & f2) for a particular central frequency (fc) in iir filter I think in this case ...
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What is the name of the filter or technique to reduce depth aliasing around object edges when working with camera depth?

I'm working with a point cloud, created by sampling a 2D camera texture and a depth map. The color is combined with the depth and is projected onto a 3D scene to create a "point cloud". The ...
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Filter H(z) manipulation frequency response changes

I'm trying to understand how altering the frequency response of a H(z) low pass filter, will visually alter it's frequency response plot. For example: by doing H(z^2), would the frequency response ...
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How to create/find a dereverberation filter?

I want to find the coefficients of dereverberation filter in MATLAB. How to do that? I have found the coefficients of reverberation filter: ...
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Linkwitz-Riley Crossover Sum as Allpass Filter

I was curious about crossover filter design, so I did some reading on Linkwitz-Riley filters. Seems to me that the general idea is that if you add HP and LP filters and they are properly designed, ...
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Does my filter desing for unusual EEG experiment looks good?

I have made some progress since my last question, and again I am asking you for advice. Hi, I'm still designing an unusual (in terms of signal analysis) EEG experiment, which forces me to design my ...
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49 views

How to optimally normalize filter coefficients?

I have designed an IIR filter in Matlab that comprises the following biquads ...
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55 views

Butterworth filter's gain formula does not agree with R's `signal` package

I'm trying to calculate the Butterworth filter gain. If I use the formula mentioned on Wikipedia: $$ G^2(\omega) = \frac{G_0^2} {1+\left(\frac{j\omega}{j\omega_c}\right)^{2n}} $$ I don't get a ...
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What is the relationship between group delay and propagation delay for a filter? [duplicate]

I was trying to design a filter for a system and I was able to obtain the group delay plot in matlab using grpdelay. The group delay output is in samples. However, I was not able to relate that data ...
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Aliasing Issues in Sub Band Coding

The general structure of Sub-Band Coding (SBC) is shown below: The diagram shows that after each of the analysis filters, decimation is performed by a factor M which corresponds to the number of ...
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CIC filter internal register truncation

I try to implement CIC filter with truncation in hardware, but there is some misunderstanding. For some reason the output signal is broken. My filter is N = 6, R = 20, M = 1, Bin = 16, Bout = 16. (...
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Setting up RRC filter and useful equations

I have looked at previous answers but cant seem to get everything I need in one place to help me and my textbook doesnt explain this in practical terms. Signal Parameters: $F_s = 44100$ $R_s = 300$ $...
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Filter design for an unusual EEG experiment

I'm designing a quite unusual (in terms of signal analysis) EEG experiment, which forces me to design my own filter. I have never done this before, so kindly please check my work and suggest ...
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The optimal estimator using FIR Wiener filter

The received signal $y(n)$ is defined as $$y(n) = h(n)*x(n) +v(n)$$ where $h(n)$ is unknown impulse response and $v(n)$ is a white Gaussian noise. The value of $x(n)$ for $n=1, ..., 32$ are given. The ...
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Implementation of IIR filter

Suppose we have a discrete input waveform (with sampling frequency Fs = 32*56e9). We want to filter this waveform with a filter that has two complex conjugate poles (at 22 GHz) and one zero (at 17GHz) ...
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Training of adaptive filters

I am trying to understand adaptive filters as I want to implement a denoising algorithm based on them. Do we need to train all thee adaptive filters before they start their actual process? For the ...
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Transposed Direct Form II

In the book digital Signal Processing by Proakis 4th Edition, at page 593 a table is drawn about some second order modules for discrete time systems. In that table there Transposed Form II drawn But ...
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Analogue Filters (Picking the right filter!)

I'm new to Analogue and I'm trying to find the Ideal filter for the following specifications: $0.1\ \rm dB$ ripple Cutoff Frequency $F_c = 80\ \rm{kHz}$ An attenuation of $40\ \rm dB$ at the stop-...
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Is it possible to specify number of taps and cutoff frequency independently?

When specifying the properties of a moving average filter, is it possible to specify the number of taps and the cutoff frequency? My guess is no since taps (the number of samples in a moving average ...
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Performance bounds for low-pass filters

Precisely how does the performance of an FIR filter depend on its design criteria? To illustrate, here is a specific question that probably does not have a convenient answer: The Parks-McClellan ...
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Correcting phase response of a signal

I have a sensor that applies a frequency dependant phase alteration to my signals. I'm trying to find a way to correct these phase responses using an FIR filter. Essentially, I'm looking for methods ...
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Interpolated FIR filter (from Oppenheim and Schafer's Discrete-Time Signal Processing, 3rd ed)

[from Discrete-time Signal Processing by Oppenheim and Schafer, 3rd ed., p.196] Two questions: In this context, the filter with system function represented by Eq. (103) is called an interpolated FIR ...
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Problem with applying elliptic filter to ecg signal

I'm currently working on an article for ECG classification which it says that it has used elliptic bandpass filter with 0.5Hz and 50Hz critical frequencies to eliminate base-line and interference etc. ...
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FIR filter design using Kaiser window - passband ripple and stopband attenuation

I have been given the following parameters : lower cutoff frequency = $\pi/3$ lower cutoff freq = $2\pi/3$ beta = 5.6533 M = 51 Now I have to get magnitude and phase response of the filter using ...
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Linkwitz-Riley Allpass filters of Nth order from cascaded SVF second order sections

I am in the process of implementing a Linkwitz-Riley filter of 16th order. (LR-16, 96dB/oct) I am using 2nd order SVF sections with three modal outputs on each section. (LP, BP, HP) I am looking to ...
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FIR amplitude response using Hr_type3 function in MATLAB

I am trying to design a type 3 FIR filter and generate its amplitude response from the function Hr_type3 (mentioned in Digital Signal Processing Using MATLAB by J. ...
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Converting a lowpass filter to a highpass filter. FIR filter-type 1

I was told that in the FIR filter type I, the impulse response of a highpass filter $h_H[n] $can be obtained as follows, $$h_H[n] = \delta[n − m] − h_L[n]$$ where $h_L[n]$ is the impulse response of a ...
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Derivation of the lowpass to bandpass transformation

I have a basic question. The "well known" lowpass to bandpass transformation is $$ s \longmapsto \frac{\bar{s}^2 + \omega_1\omega_2}{\bar{s}(\omega_1 - \omega_2)}, $$ which gives a bandpass ...
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FIR filter design with matlab firls() function

I have defined the following linear phase filter of order 4 (real and symmetric) in matlab: h_d = [0.8367 -1.537 1 -1.537 0.8367] I calculate the frequency response (for many points) and then I use ...
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What does nonnegative zero-phase response mean?

I am not exactly sure what nonnegative zero-phase response means. If a filter is zero-phase (i.e. symmetric and non-causal), then what does nonnegative imply? And what are the conditions to satisfy it?...
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When is the sum of two (parallel) minimum-phase filters also minimum-phase?

Say I have two minimum-phase filters: $$\frac{A(z)}{B(z)} \: \text{ and } \: \frac{C(z)}{D(z)}$$ That is, the roots of $A(z)$, $B(z)$, $C(z)$ and $D(z)$ are all in the stable region. If add them ...
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Is there a systematic method for converting an even length FIR filter to odd length?

I'm currently implementing a discrete wavelet transform (DWT) as a cascaded QMF filter bank (pictured below). I've put together a convolution function that attempts to filter an input signal in a non-...
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How can I automate the calculation for P and I coefficients of an FPGA-based digital PLL?

I am working on designing an FPGA-based PLL through an NI myRI0-1900 FPGA and LabVIEW software. I have been able to create a functioning PLL with a PI loop filter, and but I found functioning P and I ...

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