Questions tagged [filter-design]
Fiter design is the process of designing a filter.
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Mapping of Classic Filters for Digital Filter Design
Of the four classic analog filter types: Butterworth, Chebyshev, Elliptic and Bessel- are any of these relegated to obsolescence for purposes of digital filter design in comparison to optimized ...
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How to design a digital Butterworth bandpass filter?
I am looking into designing a Bandpass Butterworth filter in python, but, I was not sure I am designing my filter correctly. What I have are the following:
High cutoff frequency = 200Hz
Low cutoff ...
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Converting a Matlab filter-design to Matlab code and obtaining filter coefficients
Central problem (Abstract):
Using the Matlab filter-designer results in direct-form II transposed coefficients, which i need. However, generating Matlab code from the filter-designer app itself does ...
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How to get the difference equation from impulse response?
I'm trying to solve a problem about filter design.
We want to design a equalizer filter which have the following frequency response :
How can i get the difference equation from the impulse responce ...
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Why use a notch filter to implement a bandpass
I have seen in a few papers, for instance in this one, the following method :
Find an initial "frequency of interest" (FOI)
Filter the input signal with a Notch filter around the FOI
...
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Find some signal x[n] whose auto-correlation function is R[k] [closed]
I want to write a MATLAB code for generating a random message signal 𝑥[n] where its autocorrelation function
is:
R[k] = sinc^2 [1/4 k] where k is an integer. (Hint: Generate a process where ...
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Confusion about analog filters
I'm mixed about the concepts of analog filters, so I hope someone can explain me, or point to information, to unentangle the ideas.
I know there are filters (analog) as Chevyshev, Butterworth, ...
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Why is voice clipping happening after Dynamic Range Compressor (MATLAB) when applied in real time on chunks of data?
I have been working on real-time audio signals on smartphones (iOS and Android) and my goal is to make voice amplitude stable in a certain range even if a person is talking closer to the mic of the ...
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Filter Synthesis Formula to implement ZOH
$$h_n = \frac{1}{2\pi}\int_{-\pi}^\pi H_d(\omega) e^{jn\omega}\,\mathrm d\omega$$
I am attempting to design a FIR filter using the formula above for coefficients. Where
$$H_d(\omega) = \frac{\sin \...
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Find the equivalent of this python remez specs in C++ remez or Matlab firpm
I want to start with that I am quite new in this community, so if this question does not belong here, please point me to the right place. Thank you!
Background
I am learning and writing a half-band ...
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RT 60 from Impulse response
I have already the impulse response of an acoustic environment in the form of a series of signals over 300 samples. I have tried the procedure given in this platform - i.e. low pass filtering, Hilbert ...
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What is the transfer function of the filtfilt?
I designed a Parks McClellan filter to design a lowpass filter. First, I created the order of the filter with firpmord(). Then I designed the filter with firpm() and created the impulse response. I ...
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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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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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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 $|H(e^{j\omega})|$?
1) How is this a low pass filter?
$0.99 \leq |H(e^{j\omega})|\leq1.01 {\rm\ for\ } 0\leq|\omega|\leq0.19\pi$
$|H(e^{j\omega}\;)|\leq0.01{\rm\ for\ } 0.21\pi\leq|\omega|\leq\pi$
2) What kind of filter ...
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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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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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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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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 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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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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What is the construction of this filter
I have built a resonance filter using the code from here I would like to know what the construction of this filter is. I believe it is an IIR and that it is probably a second order/two pole filter but ...
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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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