Signal Processing

Questions tagged [filter-design]

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Fiter design is the process of designing a filter.

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Approximating inverse of unstable difference of Gaussians filter

I am trying to invert a difference of Gaussians (DoG) filter. The inverse is not stable and so I am trying to find an approximation applied to a specific input. The DoG filter increases contrast at ...
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When to Use Composite Filters and When to Use Separable Filters?

I’m a beginner in image processing, and was wondering since seperated (decomposed) filters help give faster and more efficient results, when do we even need to use composite filters? All I heard is ...
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How to remove heartbeat signal from blood pressure signal?

I am working on a medical engineering project that involves processing signals received from a human body. We use a sensor to record the blood pressure, $B(t)$. However, if you put your finger on many ...
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Prove the following property of All-Pass Filters

If A(z) is an all-pass filter given by $$A(z)=\frac{z^{-1}-d_1}{1-d_1 z^{-1}}$$ where $d_1$ is a real coefficient Then prove that $$ |A(z)|\left\{ \begin{aligned} &<1 &|z|>1 \\ &=1 &...
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Getting different values for order of filter using Chebyshev method-:

Here are the 2 solutions. I am getting 2 different values of order of filter for the given question. Which is wrong and which is right? Or are both right? Please guide $|H(j\Omega)|=-20\log_{10}\...
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1answer
44 views

Attenuation of reconstruction filter for Digital-to-Analogue (DAC) converter

I am using a 12-bit DAC to produce signals between 10 kHz and 400 kHz, produced at a rate of 3.33 MSamples/s (Nyquist frequency of 1.665 MHz). I would like to create a "reconstruction filter"...
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26 views

How to convert passband and stopband frequencies to poles and zeros

I have a NI 9229 digitizer with the following datasheet. The datasheet mentions: a passband frequency of $0.453f_s$ with a flatness of $\le0.1\,\text{dB}$ a stopband frequency of $0.547f_s$ with ...
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How to identify low pass, bandpass, band reject, high pass filter etc from given $|H(e^{jw}|$?

1) How is this a low pass filter? $0.99 \leq |H(e^{jw}\;)|\leq1.01 for \; 0\leq|w|\leq0.19\pi$ $|H(e^{jw}\;)|\leq0.01\;for\; 0.21\pi\leq|w|\leq\pi$ 2) What kind of filter is this? Low pass, high pass, ...
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31 views

What is Passband Ripple?How is this expressed in dB scale?

If my passband ripple is given as 0.2 dB $$\implies 20\log_{10}(1-\delta_1)=0.2 \\\implies 1-\delta_1=10^{\frac{0.2}{20}} ...
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Writing the specifications of digital filter, am I correct?

Design the FIR filter using suitable window for the specifications-: $0.899\leq |H (e^{j\omega}\;)|\leq1\;for\; |\omega| \leq0.2\pi$ $\; |H(e^{j\omega}\;)| \leq 0.01, \text{ for } \; 0.4\pi\leq \...
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Design an FIR linear phase lowpass filter using windows to meet the following specifications. What is given in the question? How to start solving it?

What is given in this question? I have not seen a single solution of these types of filter design problems in the internet. So I am confused, how do I proceed solving these problems. Can you guide me ...
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1answer
37 views

Having trouble working understanding a noise shaping filter

As context, I'm trying to understand and implement this Von Karman turbulence model in a simulation I'm working on. In order to generate turbulence velocities on the fly using the model, a ...
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60 views

Design of all pass filter with desired phase response

I want to design an all pass filter with a desired phase response in MATLAB. But in MATLAB, all pass filters are designed either by specifying the all pass filter coefficients or from a transfer ...
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How to find lattice coefficients when k2=1?

$H(z)=1+2z^{-1}+z^{-2}$ We know the formula $a_{m-1}\;(k)=\frac{a_m\;(k)-k_m\;a_m\;(m-k)}{1-k_m\;^2}$ Here $k_2=1$ That makes $a_{m-1}\;(k)=infinity$ How to find lattice coefficients of all pole IIR ...
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How to read the filter IIR design numericals specifications?

First of all I want to let you know about notations that I am using-: $\alpha_p$=attenuation factor of passband(in dB) $\alpha_s$=attenuation factor of stopband (in dB) $A_p$=Gain at passband edge ...
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26 views

Design of digital filter with desired phase response

I want to design a digital filter with the following phase response in MATLAB. i.e. at 1kHz, the phase response should be 9 degrees, at 2khz phase response should be 18 degrees ,at 3kHz phase response ...
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36 views

Filter model of passive attenuation of headphones

It is possible to model passive attenuation of headphones as a Matlab/Python filter ? For example I would like to find digital filter which will be simulating the way of the passive attentuation of ...
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59 views

How to find Lattice structure when k3=1?

I could not type this so I got a snip shot. Pardon me for that. Here k3=-1. That means the formula Here we will get x/0=∞. How to solve this question?
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1answer
161 views

Does it make sense for my filter to depend on the discrete sample variable n?

I feel a little nervous asking this question as I am not sure if know what exactly I am trying to ask. But please see if you can help. Ok, I have a filter whose output grows as long as the input is ...
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1answer
37 views

analog filter magnitude flatness problem

I have designed an analog filter that includes a high pass filter with a cut-off frequency of 5 Hz and a notch filter with a cut-off frequency of 50 Hz. I want to sample the output signal of the ...
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1answer
77 views

Can you explain remez exchange algorithm?

What I don't understand about this algorithm-: What is the diffeence between w and w cap? What is the meaning of extremals? What does it means to reject p-r superfluous potential...
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27 views

Where can I find a solved numerical example of FIR digital filter design using Kaiser window?

I am studying for digital signal processing exam. And I need to learn about Kaiser window numerical for it. It is pretty important. I searched the whole internet but could not find 1 example of Kaiser ...
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70 views

How is this given impulse response of infinite duration? Isn't it just from -π to +π?

How is hd(n) infinite duration when it is from -π to +π. Book says as it is infinite duration, we in the next step take-: h[n]=hd[n] from n=-(N-1)/2 to (N-1)/2 and 0 otherwise. I can't see how this ...
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62 views

Varying phase shift between two signals

I have two microphones separated by a distance d placed in end-fire configuration. Audio input data was played from the front side (i.e corresponding to 0 degrees.) The data picked by microphone1 and ...
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1answer
32 views

GNU Radio filter normalization

I have noticed that all filter taps from the firdes class are normalized by dividing them over the sum of all taps magnitude, as shown in the example below for Gaussian Filter. Is this normalization ...
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1answer
31 views

How would I approach implementing a windowed sinc filter with changing cutoff in real time?

I moved this out of another question, where I'd put it to give more context, and answered it here for the next reader. ELI5 how to implement a windowed sinc FIR filter in real time for an audio ...
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134 views

What is a simple method, if any, to calculate a variable IIR filter for audio effects in real time?

Question edited based on first answer which raised a lot of good points about what problem I'm trying to solve. Also reordered some things to stick context of how I did this with FIR so far at the ...
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1answer
56 views

Notch Filter via Bernstein Polynomial

Context: In the review article [1], the authors explain that we can model the frequency response of an FIR notch filter using Bernstein polynomials. The paper shows the details, but at the end of it, ...
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28 views

Frequency response of a moving average filter/box car

I have designed a simple filter for an embedded system handling magnetic field data. The incoming sample rate is 128Hz from the sensor and we have output data rates of 1Hz, 16Hz, 32Hz & 128Hz So ...
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35 views

How to calculate analog cutoff frequency to design a digital filter?

$\Omega_c$=cutoff analog frequency In this example, what is going to be value of $\Omega_c$? Some teachers in youtube are teaching this method-: Is this method correct? If I use this formula, then ...
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46 views

Different formulations of Chebyshev II resulting in different poles?

I am currently trying to implement a C++ program to calculate digital filter coefficients using a Chebyshev Type II filter design technique. The text I am using to understand the theory of the ...
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1answer
41 views

Analog Signal Treatment for FFT with Padding

My education and career has been focused in mechanical and materials science; sorry if the following is well known, but I haven't been able to find the appropriate SEO terms. Wanting input to further ...
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1answer
27 views

Accuracy of time-domain filters and frequency domain filters

I am trying to calculate the Hilbert transform using 3 different method: scipy.signal.hilbert function, time domain filter, and frequency domain filter. The test ...
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51 views

IIR Filter Stability

I have designed an inverse IIR high pass filter by inverting its transfer function B(z)/A(z) to A(z)/B(z). However, the inverse filter has a pole at z=-1. Is this filter considered stable (marginally ...
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44 views

Mitigating high-frequencies after sampling

I have a dummy signal that I am sampling at $T = 100 \ \mu s$. According to the Nyquist frequency, I can identify frequencies up to $\pi/T = 31416 \ rad/s$, as we can see in the FFT of my signal (blue ...
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1answer
43 views

Is the Schroeder Allpass implementation in Freeverb incorrect?

I've been looking into the venerable Freeverb and I see that CCRMA state that it uses 4 Shroeder Allpass filters in series, but when I look at the source code I see: ...
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1answer
29 views

how to calculate $H_{lp} = z(1- H_{hp}) $ , given coefficients for $H_{hp}$?

Given a high pass transfer fn of the form $H_{hp}=a_{1}*z^0 +a_{2}*z^{-1} + ... a_{n}*z^{-n}$ Is it possible to calculate a causal low pass filter using $H_{lp} = z*(1-H_{hp})$ ? attempting $H_{...
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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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1answer
21 views

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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3answers
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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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1answer
85 views

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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1answer
36 views

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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2answers
176 views

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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35 views

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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