Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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65 views

implement bandpass filter on complex valued signal? (C++)

I want to filter out complex valued samples - Bandpass filter. I have 100 complex samples and I wish to filter out a range of frequencies from it, as you can see in the result of the FFT, there are 3 ...
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67 views

Frequency Sampling Filters (FSFs) NOT the Frequency Sampling Method

A class of digital filters called Frequency Sampling Filters (FSF) has been mentioned in a side note by Richard Lyons in another related question comparing IIR digital filter design using mapping ...
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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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1answer
23 views

how would gradient descent work in optimizing a practical frequency response of a sinc function?

The gradient descent algorithm should work by minimizing the ripples in the stop and pass band, then increase the slope of the transition band. So basically the parameters of interest will be passband,...
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1answer
41 views

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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1answer
46 views

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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1answer
61 views

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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1answer
81 views

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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1answer
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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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Automatic Gain Control (AGC) by using Least Mean Square (LMS) Method

I am try to do Automatic Gain Controller by using Normalized LMS. Can somebody help me? I could not find how to start it. I am try to do it on MATLAB.
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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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1answer
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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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1answer
47 views

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
53 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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31 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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78 views

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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1answer
100 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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111 views

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
46 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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2answers
73 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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35 views

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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1answer
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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1answer
39 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
171 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
38 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
94 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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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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1answer
67 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
56 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
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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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3answers
189 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
61 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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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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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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1answer
58 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
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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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2answers
59 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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