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Questions tagged [filters]

In signal processing, a filter is a device or process that transforms a signal by selectively choosing specific frequencies and leaving others with appropriate gain or as it is.

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Online backward-looking derivative filter for robot sensor data

I have a laser rangefinder on my robot and I need to compute the speed of the detected object. I have seen many different filters for that, but all of them apply only to offline data, or introduce ...
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Plane Settings of the Matched $z$-transform Method

I've come across that the matched $z$-transform maps poles of the $s$-plane design to locations in the $z$-plane. My question is, what is the $s$-plane and what does this mean? I'm aware that the ...
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I need help with this problem

I can't figure out c,d,e,f. Can I get some help?
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Comparison Between Mahoney Filter and Kalman Filter in Euler Angles (Yaw, Pitch, Roll) Estimation

What are the advantages of Mahoney Filter over Kalman Filter for Euler Angle (Yaw, Pitch, Roll) estimation? Could anyone compare them for Angular State Estimation? Resources Mahoney Filter ...
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What is the name of this filter?

I am having this simple function which acts on two state variable $s_t$ (states being $\{0,1\}$, boolean in sense), and I am evaluating it in discrete time steps. The function is $f_t=1$ if $s_t$ has ...
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Odd-order Butterworth filter gain at cutoff frequency

Wikipedia gives the gain of an n-order Butterworth filter as $$G^2(\omega)=\left |H(j\omega)\right|^2 = \frac {{G_0}^2}{1+\left(\frac{j\omega}{\omega_c}\right)^{2n}}$$ here. If you look at the gain ...
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How to achieve a smoothed frequency phase response with no outliers

I have two sampled signals, which I then perform wavelet analysis on, by synthesising the Filterbank, perform the dft on each kernel, and the dft on the sampled signals, and multiply together. ...
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How Could I extract the signal in the below Image

This is my first post so sorry if I make any mistake. The entire signal length in the screen shot below is 10 second. I ran a first order low pass filter (digital) at 10Hz cut-off frequency, and I ...
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Automatically crop out certain region around red objects

I have the image seen below and I would like be able to automatically crop out everything around each red section - not enough to overlap but enough that you can see how connected to the green parts ...
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Overshoot when using Bessel filter

Here is simple code to illustrate overshoot using bessel filter: ...
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finding harmonics from fundamental

https://s3.amazonaws.com/izotopedownloads/docs/rx700/en/interactive-tools/index.html#selection-tools From the above link, under the "Magic Wand" tool description, how does that algorithm work? In RX,...
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Reading a Wav File

I need to read two .wav files. One of them is a song, and the other one is impulse response. Then, I have to convolve them. In fact, I have read them by using ...
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How is the Gaussian kernel related to the Euclidean distance of the neighbourhoods in the non-local means algorithm?

I am reading the paper A non-local algorithm for image denoising that describes the original non-local means algorithm. They define (p. 3 of the pdf) the distance between two square fixed ...
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Remove Powerline interference from an Electrocardiogram(ECG)

how do i go about removing 50Hz powerline interference from an ECG using filter design techniques such as pole zero placement and Butterworth filter design?
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2answers
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Why can convolution only be applied to compute the output of a linear filter?

We apparently cannot compute the output of a bilateral filter (BF) using convolution (with the image) because the BF is a non-linear filter. In general, why can convolution only be applied to compute ...
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How does the power spectrum of the amplitude envelope depend on its underlying carrier frequency?

when looking at the amplitude envelope of an oscillation (see red trace in Figure below), it seems to me as if the frequency content of the envelope depends on the the frequency of the underlying ...
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Delay in Savitzky-Golay filtering

I am applying a Savitzky-Golay filter to a signal, using the scipy function. I need to calculate the lag of the filtered signal, and how much is it behind the original signal. Could someone shed ...
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Is the WTST-NST filter available as a module in some python library?

I need to implement the WTST-NST filter (wavelet transform based stationary non-stationary filter) to denoise some sound files (preferably in Python). I am from a Mechanical Engineering background ...
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Why isn't bilateral filtering shift-invariant?

I have a source (that I can't share) that states that bilateral filtering isn't shift-invariant (that is, it depends on the input signal), hence we cannot use convolution and the Fourier convolution ...
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Convolution of Two Impulse Signals

I have encountered convolution of two different impulse signals. ...
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Frequency warping while creating filter coefficients

If I have an audio signal at 16kHz with frequencies band-limited to 8kHz. I want to low-pass filter this signal such that fc=4000Hz. Meaning when I reconstruct the signal I do not want any content ...
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How to implement separable bilateral filter?

I’m trying to optimize my naively implemented O(n^2) bilateral filter for 2D images. I googled and found a lot of theoretical explanations about the separate one. However, it seems there are things ...
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1answer
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Biquad Filter Non-Linearity

I'm using a biquad filter algorithm in my project. I got 3, 16 size variable arrays summing into one and then gets filtered. Now i want to filter each array seperately and sum after that, which if ...
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How to characterize this whitening filter?

A little bit of background: I am interested in whitening in the least-square estimation. In this sense, consider a univariate signal $y$. Assuming the covariance of $y$ is $\mathbf{A}\sigma^2$. ...
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Why apply filterbank in power sepstrum not log power spectrum for cepstral coefficients feature(MFCC, IMFCC, LFCC) [duplicate]

In the original method for MFCC and any other method based on cepstral coefficients The original step is like this: |FFT|^2 -> Filterbank -> Log -> DCT but why not like that: |FFT|^2 -> Log -> ...
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low pass filter

I am new to DSP, I'm trying to implement a low pass filter in a microcontroller and I'm looking for examples on the internet. One of the simplest low pass filter (as its title says) is this. But I ...
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How to calculate Walsh hadamard transform for image patches of odd sizes?

I was reading this paper on optical flow, which as apart of the algorithm makes use Walsh-Hadamard transform on image patches of size 2r+1. So far, where ever I have looked, I could only find that ...
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What is the difference between a range filter and a low-pass filter?

A low-pass filter attenuates the high frequencies, preserving only smooth variations in the provided image. The result of an image which is low-pass filtered is an image with blurred edges (hence ...
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Determining type of analog filter given its pole zero plot

How can i classify an analog filter given its pole zero map. For example: I've got my zero's located at $±2j$ and my poles located at $-1$ & $-2$ , then what is the nature of the filter? $ i.e, ...
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What is the minimum and maximum cutoff frequencies for the antialiasing filter for this ADC?

https://www.analog.com/en/products/ad9675.html https://www.analog.com/media/en/technical-documentation/data-sheets/AD9675.pdf It says in the data sheet that the Anti-Aliasing Filter has : "...
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Bandpass filtering with passband changing with time

Is it possible to implement some sort of filter which adapts as a function of time? Specifically, say I have a "noiseless" model of some signal which has the same frequency components at the same ...
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Continuous Wavelet Filter Bank Frequencies

I have followed instructions on how to create a single Morlet wavelet, and can change the centre frequency of the wavelet. My question now, is I would very much like to create a filter bank. How do ...
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Extracting signal from data using Otsu thresholding in MATLAB?

I am very new to signal processing. I am trying to extract a signal from my data using otsu threholding but it's not fully accurate. In this case I know that there is going to be a single signal in ...
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“Improved” MZT/IIM type One pole LPF

Some time ago, while playing with all kind of approximation of common math functions, I came up to this idea to calculate a1 coefficient for a 1st order LPF filter: \begin{array}{l}a1=-(1+x+\frac12x^...
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What is the difference between an impulse response and a kernel?

It seems to me they are two terms used to refer to the same concept, but, according to the Wikipedia page on Guassian filter a Gaussian filter is a filter whose impulse response is a Gaussian ...
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How to determine if a filter is bandpass/stopband from its pole-zero diagram in z-domain

How can we determine if a filter is bandpass or stopband, just by looking at its pole-zero diagram in z-domain? For exmaple, if we have a system with third-order pole at the origin and a zero on the ...
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1answer
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Moving average filter for sound in MATLAB

y[n] = b0x[n] +b1x[n-1] I have this MA filter in matlab as following ...
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What is used to initialise difference equations that require data from the output?

Given some difference equation for a filter like $y[k] = ax[k] + bx[k-1]+ cy[k-1]$, how would you initialize it? Since it needs an old output value (feedback) to calculate the new output value, it ...
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50 views

Discrete Time Fourier Analysis

Suppose we're given the following: $ x[n] = 2 + (-1)^n $, and are given the impulse response $ h[n] = u[n] a^n $, of an LTI system where $ |a| < 1$. We're asked to find the output $y[n]$, if $x[n]$...
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66 views

Hilbert Transformer processing block by block

I am processing audio and trying to obtain a 90º phase shift when processing block by block. I have implemented a Hilbert Transformer (from Lyon's 3rd. Ed. & MATLAB). MATLAB code below (I believe ...
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Plotting the Phase Response

I would appreciate it very much if someone would be able to provide some clarity on plotting phase responses. For instance, given that the frequency response of a filter can be written as H(exp(j*&...
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Determine phase response from pole-zero plot

I know how to determine the frequency response from the pole-zero diagram given the following formula: $\left | H(f) \right | = \frac{\prod \left | (e^{j2\pi f} - a_{i})\right |}{\prod \left | (e^{...
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1answer
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Generating FIR filter in Web Audio API

I'm creating an HTML that applies a low-pass Windowed Sinc filter with a window size of 8192 to white, pink, or Brownian noise. There are nine radio buttons, eight of which set a cutoff frequency and ...
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Accelerometers (again…) using appropriate transformation for drifting signal returning back to x-axis (0 m/s)

Quick question on processing velocity results from accelerometer data. I'm aware of the unreliability of the data, but I'm willing to tolerate a fair amount of error. Also willing to perform lots of ...
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1answer
79 views

Chebyshev I or Chebyshev II?

I am new to DSP, and I am trying to recreate some analysis of a signal, They use a chebyshev I filter to go from a 100Hz signal to a 50Hz signal. Whereas I am currently using a chebyshev II filter, ...
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HRIR and DT Convolution

You have an experimentally-measured HRIR and an audio signal. And you wish to implement filtering using scientific libraries in Python such as numpy, scipy and matplotlib(for plotting) Question A) ...
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What is meant by labels in 'gaussian shaped labels (reference attached)'?

I am trying to understand MATLAB code and it uses a line like: ...
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Basic question regarding third octave filters

I would like to use third octave filters on a time signal. Do I have to apply all third octave filters to a single frequency or do I only need to apply the one that has its center frequency the ...
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1answer
53 views

Filtering and peak finding

I have an observed signal which is the norm of the sum of two 3-vectors: $$ S(t) = || \mathbf{A}(t) + \mathbf{B}(t) || $$ Now, I have full knowledge of the vector $\mathbf{A}(t)$, $$ \mathbf{A}(t) = \...
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Third octave bandpass filter with python

I am very new to signal processing and coding which is why my questions might be really basic. I have a signal of the acoustic pressure p'(t) and I would like to use a third octave bandpass filter in ...