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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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“Side Lobes” When Using a Butterworth filter

So I am attempting to filter a signal containing frequencies at 7Hz, 11Hz, and 74Hz. The signal lasts for 30s and looks like this between -1s and 1s: Using a butterworth filter of order 2 with lowcut=...
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19 views

Using a Passive 1st Order RC filter to Extract DC Component

I am trying to extract the DC component out of a PWM signal produced by the MCU using a LPF. I understand that 1st order RC filter is looked upon as being amateurish because the gain in the passband ...
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1answer
30 views

Inverse DTFT low pass filter

can someone explain why the ideal frequency response of a low pass filter cant be implemented using the inverse discrete time fourier transform. i understand the concept of why it cant be implemented ...
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25 views

Noise Shape Digital Filter

My objective is to build a noise shape filter from a given transfer function (in one case) and from a given PSD (for another case). Checking my precedent questions you can see that this argument is ...
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0answers
18 views

Specifying the frequencies when designing a filter

I understand the general method of designing high pass and low pass filters with a specific number of poles and zeros. However, if given a specific sampling frequency, let's say 2.4 kHz, and asked to ...
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2answers
38 views

Digital Filter Design using difference equations

I am reading a chapter on digital filter design from analog filter design using difference equations. What they do first of all is that they map $s$ (Laplace variable) to $z$ ($z$-transform) by the ...
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1answer
69 views

Recursive filter gain

I implemented a filter similar to the one below, within a FPGA. Two substractors and an accumulator. The recursive equation is given by: My inputs are exponentially decaying pulses of the form ...
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1answer
23 views

Why the frequency response plots (of causal filters) only show positive frequency?

Take an example of the below plot for an LPF (Source : WikiPedia) The plot starts from $0$. We know that the fourier transform of any signal brings in negative frequencies due to complex exponentials ...
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23 views

Linear Predictive Coding for general signals

I have a signal that's monotonic and roughly linear and have been looking at using Linear Predictive Coding to encode information and compress my signal. I guess my first general question is if this ...
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10 views

Filtering out uncorrelated known signal from channel

I have 2 channels. One of them contains the noise signal and the other one contains the signal of interest and noise. The relationship between the signal and noise is not known (not a weighted sum). ...
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1answer
49 views

Why do we put poles close to the zeros in notch filter?

In the book I'm studying from "Applied digital signal processing - Manolakis" it says: The problem with FIR notch filters is that the bandwidth of the notches is large. A simple way to create ...
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2answers
262 views

Estimating effect of filter on headroom

I am working with beamforming algorithms for speaker arrays. One of the problems we face is that we need to estimate the maximum amplitude of our test signals after we applied FIR/IIR filters to them. ...
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1answer
40 views

Recursive form representation of FIR filter

Do we have methods to represent FIR filter in recursive form? One of the methods that I know is frequency sampling method. Can someone provide me any method?
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53 views

How to implement the RLS for matrices

I need to implement the RLS algorithm but it's for matrices instead for vectors, I have made the below code, but still something wrong is not working well, EDIT: The code should be done as below, ...
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1answer
34 views

How can I implement a triangular filter in MatLab, given it's Z-transform representation?

I have to implement an anti-aliasing filter for a certain processing step in MatLab. While searching in literature for inspiration, I came accross a paper in which the authors wrote the following: ...
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0answers
51 views

Digital or analogue filtering?

I'm trying to self learn the art of signal processing whilst moving through my third year pure maths degree. Sorry if my terminology is incorrect however I hope I am understandable! I am looking at ...
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1answer
29 views

edge detection of 1-D signals

Would you please introduce me to some methods (or references) to do edge detection on some pulses (rectangular pulses embedded in noise). At first I apply some filters to smooth the signals, then I ...
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1answer
32 views

Why delete high frequence in fft return the area of the most changes?

I'm new to ||image-processing|| and || fft filters || I have a relative smooth $\mathtt{2D}$ (circle) gaussian surface (height $10$) with several "holes" in it, and the surrounding area [has] some ...
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2answers
35 views

Matlab frequency response

I'm trying to make a frequency response of that function: in matlab a made this: ...
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0answers
51 views

Why I don't get the right PSD

I need to model a noise with a given PSD. To do this, I am starting from a white gaussian noise (WGN) and feed with the WGN a transfer function, which will act like a filter. In fact,it's easy to ...
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25 views

Notch filter cutoff

I have what I think would be a simple question for a signal processing expert. I have used a notch filter with a cutoff of 100 and 200 Hz using a 4th order butterworth. I am needing to find the ...
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0answers
48 views

Explicit formulation of the role of eigenvectors of the matrix of a linear time-invariant filter

At this point in a lecture by MIT Professor Strang the idea of gaining insight into a filter by studying its eigenvectors is introduced. The idea is clear with the example chosen in the lecture: an ...
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1answer
29 views

how to create a harmonic mask from fundamental frequency?

Using an algorithm that gives the fundamental frequency across time, example https://www.mathworks.com/help/audio/ref/pitch.html (or some other one?) how would I then use that, to create a harmonic ...
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0answers
53 views

Noise Shape Filter to obtain a given PSD

I have experimental PSD data of disturbances in terms of acceleration on a mass. My first problem is that these data are presented as $ (m/s^2)/\sqrt(Hz)$ which is not the PSD unit. Those unit should ...
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1answer
70 views

Why Convolution between signal and channel is not done directly in SIMO channel

In OFDM system, I need to transmit a signal $x$ in SIMO channel $H$ with one $Tx$ antenna and 4 $Rx$ receiver's antennas. The initial equation for that convolution is $r=Hx+n$ where $n$ is the noise. $...
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21 views

Estimating velocity using MEMS sensor (accelerometer part)

I wanted to know how I can estimate velocity of a particle (contained with a sensor) from the acceleration output of a MEMS sensor device in order to find the total kinetic energy of the particle? ...
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0answers
36 views

Filter to remove high frequencies in R

I'm having a problem with a signal, I need to leave only the low frequencies so smooth as possible, because then this signal will be the input of a neural network, so I need to reduce complexity. The ...
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1answer
23 views

Why does filtfilt function require a negative sign?

I am trying to extract features from a signal using filtfilt function in MATLAB. I implement the filtfilt function on my original signal using a high pass filter(second derivative) and use the peak of ...
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1answer
29 views

Prove that convolution with a separable filter is equivalent to convolution on each 1D filter

I would like to prove that convolution of an image $I \in \mathcal{M}_{m_1 \times n_1}$ with respect to a separable 2D filter $F$, (i.e., $F = F_1 F_2$, where $F \in \mathcal{M}_{m_2\times n_2}(\...
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0answers
25 views

Infinite-wideness limit of Weierstrass transform (Gaussian filtering)

Weierstrass transform (Gaussian filtering), with infinitely wide Gaussian, is expected to give zero and if we start from the definition it is consistent with this expectation as $\lim_{\sigma \to \...
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1answer
64 views

Is ideal band pass filter (brick wall filter) linear phase? [duplicate]

I'm very new in digital signal processing. I have multiple sensors and the way I filters the signals in post processing is: take FFT of the signals. put zero on out range of interesting frequency (...
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2answers
190 views

Moving Average for Notch filtering

I have a periodic signal(ECG) with period of ~1 seconds. It does not have features that are shorter than 0.04 seconds. For removal of 60Hz, I thought instead of implementing a notch filter, doing a ...
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1answer
47 views

Moving average algorithm that conserves integral

Suppose I have point array $y_i$ of size $N$. How to implement moving average algorithm that conserves quantity $$ I = \sum_{i=1}^{N}y_i $$ NOTE: I don't want time shift so I would prefer to use ...
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0answers
14 views

Relation between variance of gaussian noise and intensity variance in Bilateral filter

So I implemented the code for bilateral filter from here and ran it for various values of spatial sigma and intensity sigma. I noticed that when I add gaussian noise to an image with variance say 0.02,...
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23 views

blind channel estimation using EM implementation

I would like to implement EM for blind channel estimation. The algorithm is briefly described in this paper, under the section "EM for blind channel identification". I'd like to derive the algorithm ...
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0answers
19 views

Variance of a weakly stationary stochastic signal after filtering

I have a weakly stationary signal $X(t), t\in R.$ The covariance function of $X$ is $r_x(\tau) = \frac{1}{1 + \tau^2}$ where $\tau$ is the time difference. I have a filter with "filter-function" $H(f) ...
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1answer
33 views

4-phase FIR LPF in MATLAB

I'm trying to create a low pass FIR filter using the coefficients I found in ITU 1770-3 on page 18. This is the last piece of the puzzle for an audio loudness metering algorithm I'm working on. I'm ...
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16 views

Integrating values in a matrix in MATLAB- Audio Loudness Meter

I'm trying to implement an algorithm in MATLAB as specified by ITU-R BS.1770-3. The goal is to measure loudness of an audio file in 3 different ways: un-gated LKFS, gated LKFS, and true peak. So far I ...
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3answers
204 views

Savitzky–Golay filter vs. IIR or FIR linear filter

A traditional IIR / FIR filter (lowpass to remove the high freq oscillations), e.g. moving average, or a Savitzky-Golay filter can all be useful to smoothen a signal, such as an envelope signal: ...
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2answers
100 views

Bandpass filter that automatically adapts its bandwidth when a transient is detected (to avoid to smoothen the transient)

Let's say we want to isolate a band 1000 hz +/- 50 hz. Obviously, limiting the bandwidth by applying a passband filter will always destroy a bit the sharp transients (a Dirac or a rectangular ...
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NLMS Echo Cancellation: How do we estimate when the time-step at which the Far-End Echo will be generated?

I'm new to this domain. From my understanding, we do the following in our algo: ...
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0answers
35 views

Filtering Audio Signals in MATLAB

I'm trying to apply a filter to an audio signal in MATLAB and having some trouble processing it. So far, I have a transfer function that describes a K-weighted filter, and I am able to create a bode ...
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2answers
69 views

Conversion from stationarity to non-stationarity

Is there any way to convert a non-stationary signal to a stationary one, perform operations on it meant for a stationary signal and then convert it back to the non-stationary one?
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1answer
35 views

Sobel and Prewitt operators decomposition

I was reading in the wikipedia page of Sobel operator and Prewitt operator that is possible to decompose these two operators (quote form the Formulation paragraph): "as the products of an ...
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0answers
38 views

Filtering Jerk from data

I am having force data from an experimental setup. The experimental data is having jerk force being added due to inefficient mechanical design of the setup. Is there any way to remove the 'jerk force' ...
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1answer
71 views

What is an “innovation filter”?

I'm a math postgrad student working through a paper on eigenvalue decompositions of matrices of FIR filters (used for stuff like total decorrelation, convolutive mixing, MIMO). Towards the beginning, ...
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1answer
51 views

Manual filter or digital filter first for EEG?

I am getting conflicting advise regarding how to clean my EEG data: 1) Manually remove artefacts first and then apply digital filters OR 2) Apply digital filters first and then manually remove ...
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0answers
27 views

Make an envelope smoother but keep the sharp attack

I have a signal x for which I would like the envelope. I have taken the Hilbert transform (like in this question): ...
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1answer
59 views

In What Way Is the Difference of Gaussian (DoG) More Tunable than the Laplacian of Gaussian (LoG)?

I was reading through this post "What is the difference between Difference of Gaussian, Laplace of Gaussian, and Mexican Hat wavelet?" and noted in Chris Luengo's answer that he mentioned that the ...
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2answers
66 views

Realization of a filter based on its transfer function

How can we check whether the filter is realizable given its transfer function and What are the parameters the realization depends on? Here is an example: Show that a filter with transfer function ...