Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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Karplus-Strong: filter section identification

Out of curiosity I was digging into the Amid Fish Karplus-Strong implementation to see what had been tweaked to improve on a stock implementation. Most of the code is straightforward - but it includes ...
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How can I add poles to this filter to reduce the higher frequencies leaking through?

I have built a resonance filter using the code from [here]https://wiki.multimedia.cx/index.php/Impulse_Tracker#IT214_sample_compression Would it be possible to changed this from a two pole filter to a ...
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Ideal Allpass filter

Is it possible to implement an ideal allpass filter in Matlab? The filter response should be like this: Note: the photo is imaginary (I did it using paint).
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LTE: How to distinguish three frequencies within one band?

I'm new to learning LTE. Had some basic queries, I ran over stack exchange couldn't find them because these may be very simple queries but still I'm not able to understand them. For a particular band ...
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Filter Ranking based on Noise

I spent a good amount of my time searching but I can't seem to find any straightforward answer. I want to know the comparison/ranking of the filters (Butterworth, Chebyshev I, Chebyshev II, Elliptic) ...
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How can the location of the resonant peak in RBJ's shelving filters be calculated?

I'm attempting to use the low- and high-shelf filters from robert bristow-johnson's Audio EQ Cookbook1 for a filter that will have a pronounced peak - high $Q$, with a range of up to +24dB. The goal ...
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Problem wih (diverging?) digital IIR filter for an online acquisition system

I am trying to plot windows of acquired data from an LSL-compatible system. For the example below, let's consider 1-second window of a 64 channel EEG system. In practice, I use shorter 0.2-window, but ...
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How to correctly set the zi parameter (initial condition) for a scipy IIR filter

Let's take as an example an IIR filter using the 'sos' output, as this is what I use the most. To apply a bandpass filter, you can do: ...
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inverse z transform performed on 6th order IIR filter

we are told to find coefficients and impulse response of IIR filter of order of 6. There are 6 zeros and 6 poles in the design. Pole and zero pairs are conjugate and poles are within the unit circle ...
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Problems discretizing a notch filter - Chebyshev Type I

I'm trying to implement and discretize a notch filter (Chebyshev type I) these specifications: Ws = 2pi[2000 2500]; Wp = 2pi[1000 7000]; Rp = 0.5; Rs = 40; But when I try to use discretization methods,...
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ANSI S1.11.2004 Octave-Band Filter implementation

I am trying to implement some code to firstly generate an Octave-Band and Fractional-Octave Band filter. I have used MATLAB's octaveFilter method to generate compliant Class-1 and Class-0, 1/1 and 1/...
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Implementing FIR Hilbert Transformer in Python

I want to implement FIR Hilbert Transformer in Python using window method. I'm new to digital signal processing so please point out to me all mistakes you notice. Ideal Hilbert transformer is ...
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Doubling of the scipy.signal's bandpass filter order during the design [duplicate]

I am trying to extract filter coefficients for a bandpass, using the "scipy.signal.iirfilter" function: ...
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Phase in FIR filters [closed]

Does anyone have any idea what is Clear Phase technology ? How is it implemented ? And which audio channels whether left ,right center , sub woofer should have this technology implemented and why ?
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Confusion understanding phase shift/delay?

I am reading proakis, as shown highlighted in attached snap shot,there is '-'sign along with j,but still it is written (underlined red) implies shift in positive n direction,why and how positive?when ...
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On what it means for a system to be minimum phase

From my undestanding I get that when a system's poles and zeros are all inside the unit circle then it means that it's minimum phase. But I don't get how the location of zeros and poles are related to ...
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Why do FIR systems need to be antisymmetric about M/2 to be linear phase?

Why do FIR systems need to be antisymmetric about M/2 to be linear phase? where M=delay.Also im talking about discrete time.And why are hilbert transformers antisymmetric on n=τ?
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Implementation of CIC filter using floating point

I am trying to implement a CIC filter and was reading through the literature and found that CIC filter cannot be implemented using floating point. I am unable to understand why that would happen. The ...
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Reconstructing input from the output of a known discrete-time IIR filter

I have known discrete-time IIR filter. I pass a signal x(n) through the filter and obtain the output y(n). How would I design a filter to obtain x(n) from y(n)? In general, is it possible to design ...
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Recommendation for courses / studies on digital signal processing

I hold a master's degree in mechanical engineering. However at my job I am more and more diving into topics of signal processing and data science. I find it great to discover about new topics and to ...
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FIR Filter design, using different filter length than signal length

Given some neurophysiological application I am filtering in real time data of limited length (e.g. in epochs of 100 ms) using a sampling rate of e.g. 100 Hz. Due to this short time segment I am trying ...
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Why must the order of a Band-Pass and Notch filter always be even?

My professor mentioned that the order of a band-pass and a notch filter must always be even, when showing an example of designing a digital filter using the bilinear transformation. Then he also ...
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What is difference between “Direct form” and “Normal Direct form”?

What is difference between "Direct form" and "Normal Direct form"? I was reading about filters implementations(structures) and found out both words "direct form"and "...
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Confusion regarding terminology?

While studying the topic of filter design,i came across following terms: 1)Factored form 2)SOS(second order sections)form 3)Rational form 4)Transfer function form What is the difference between all ...
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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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115 views

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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231 views

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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How to optimally normalize filter coefficients?

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