Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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64 views

Minimal group delay filter to remove harmonics but preserve fundamental

I have a signal that contains energy at a fundamental frequency (60 Hz) plus significant energy at harmonics (2nd, 3rd etc up to 15th). The signal is sampled by an ADC at 1920 Hz. I want to construct ...
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286 views

High Dynamic Range FIR Filters

Related to this question on using the Windowing method for FIR filter design versus the optimized algorithms such as least squares (firls in MATLAB, Octave and ...
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Estimating change in RMS of noise given FIR filter coefficients

Filtering noise leads to a change in the overall RMS of the noise as demonstrated by the following code-block. which uses scipy's firwin2 to calculate the taps. The ...
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1answer
38 views

Implementing HPF using frequency sampling method

I'm on it for a few hours trying to tweak it in all sort of ways but the output comes out scrambled and I can't understand why. I am trying to implement an HPF with a stopband frequency of 500Hz and ...
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3answers
65 views

Create filter without regard to phase response (amplitude only)

Why I only care about amplitude response I'm working on equalizing a DAC where I only care about the flatness of an AWGN spectrum. Since AWGN has random phase for any frequency bin, I do not care ...
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1answer
40 views

FIR filter | how can I change the axis to unnormalized?

I don't understand why my output graphs and not showing the real frequencies, tried to change it in a number of ways with no luck. so far I'm stuck. This is what I have done so far while implementing ...
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30 views

How to derive a 2d jinc highpass filter from a lowpass filter?

The following code define a jinc lowpass filter, how to derive a corresponding highpass filter? Is it possible? ...
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1answer
19 views

Find the output signal in a Low Pass Filter for a given cutoff frequency range

I have a scenario where 𝑥(𝑡)=2⋅sin150𝜋t +sin250𝜋t and g(t)=𝑥(𝑡)sin250𝜋 . The signal g(t) is passed as input through an ideal lowpass filter with cutoff frequency(fc)= 300𝜋 and passband gain= 3....
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Signal filtering with physical limitations and estimated STD

I have a prediction mechanism that predicts some signal value. Generally the prediction look like this: The are some ranges of systematic error and I cannot do anything with this. But in many points ...
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1answer
30 views

Filter for ISI Channel

I know that OFDM has many advantages in removing ISI. But I think there is one more way of removing ISI(equalizing or inverting technique). Syppose there is a channel $H(\omega)$ and we have our ...
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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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2answers
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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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1answer
96 views

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
35 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
49 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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49 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
69 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
92 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
59 views

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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30 views

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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56 views

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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32 views

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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1answer
59 views

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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3answers
60 views

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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58 views

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
70 views

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
49 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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19 views

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
56 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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1answer
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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79 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
139 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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49 views

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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113 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
49 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
89 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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37 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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138 views

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
27 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
44 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
172 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
39 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
108 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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1answer
71 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
72 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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