**1.724 filters questions.**

This is now a second time I am attempting to ask this very important but simple question here. What I want to know is can you do deconvolution by convolving a signal. It is often stated that, for ...

Can anyone explain why exactly an "Overshooting" phenomena is observed when the fundamental harmonic is removed as seen on the figures? Is it technically right to call this "overshooting" at all ? If ...

Thanks to previous question, I got some clues to resolve this one.
But, I still get some problems here.
...

I have studied convolutions and filters a long time ago. Today, I am trying to review the basics using some notes of mine, but I am finding difficult to solve easy problems. Since I don't have ...

I am trying to design an experiment to determine the peak amplitude of an EEG signal in response to a stimulus. Till now, our team has been using MATLAB and since we wish to go open source, we are ...

I have data thats not too noisy and I am trying to detect a pattern where it gradually increases then decreases in a short period of time (20 ticks? it should be roughly similar per session but can ...

for a Project I need to build an Audio Modem in GNU Radio.
I decided, that it would be the best to use DBPSK Modulation, because that is fast and relatively noise resistant.
But as it seems, the ...

Given the following equations for the achievable rate of the Minimum Mean-Squared Error Receiver [1]:
$$\mu = \frac{1}{K-1} \sum_{i=1, i \neq k}^{K}{\frac{1}{Mpd_{i}\left(1 - \frac{K-1}{M}+ \frac{K-1}...

I ran a finite-difference simulation and the behavior of an output signal, $s$, in time, $t$ (sampled with period $\Delta t$) behaves approximately as in the figure below. It is well-described by a ...

What happens if the noise has no zero mean? I mean, if the exercise is something like:
$$y(k) = A x + \eta(k)$$
When I have zero mean, I start from:
$$y = A x$$
$$\Rightarrow \hat{y} = A \hat{x}$$
...

This is my first study about signal analysing. I'm very confused about filter order. My problem is how can I know whether its 12-th order, or 2nd order like the book says so? I already knew that slope ...

a figure for instance of size 500*500 has half above part with black and below half white should result in a white line where the
white meets the black (something like a single line at line 250 with ...

I've been trying to design a bandpass filter using scipy but I keep getting a LinAlg Singular Matrix error. I read that a singular matrix is one that is not invertable, but I'm not sure how that error ...

I have a blurred and noisy image $X$, I want to apply the Wiener filter on it and get a deblurred and denoised image $Y$ (i.e apply inverse of blurring filter while at the same time reducing some ...

Or, if that's too broad, what is/are the most popular algorithms?
Background: I have no formal DSP training but much informal tinkering. I am trying to program a crossover for an audio effect. ...

For a given narrowband Gaussian filter with a specific central frequency and filter width, I need corners of a bandpass Butterworth filter whose amplitude response is close enough to the Gaussian ...

Good Morning!
I am new in signal processing and I am trying to do a work in noise control of an electronic steering lock device (ESL). My aim is to calculate the loudness (Zwicker Method- ISO 532 B) ...

I'm afraid this might be a bit of an elementary question, but I'm afraid I'm a little new to this. I've got a system that measures a frequency spectrum to find resonant frequencies. It does this by ...

Consider the LTI system given by: $H(z) = 1 - \frac{1}{2}z^{-1}+\frac{3}{4}z^{-2}$
Let $x[n] = (\frac{1}{2})^nu[n]$ be the input to the system. We want to find the output for $n = 0,1,...,N_a$, using ...

If a practical filter can't remove all unwanted frequency components like ideal filter, does that mean unwanted frequency component are still present after filtering?
how can we use such distorted ...

For the adaptive filter to work properly, a desired signal d(n) needs to be provided. The output from the equalizer y(n) is subtracted from d(n) to produce an error signal, which is used to adjust the ...

With scipy I can use signal.butter and signal.lfilter on a time signal.
Performing a Fourier ...

I understand to an extent various filter like low pass filter, high pass filer, Wiener filter Kalman filter etc. I also understand some of this filter will decorrelate/uncorrelate the signal. The ...

I have a discrete-time system which can be described as:
$$
Y_m = \sum_{r=-N_g}^{R-1+N_g} c_r x[R(m-1) + r]
$$
The unknowns are $c_k$ but I know that they have the following approximate behavior:
$$...

Introductory resources to mathematical properties of "dynamically recalculable" filters in audio/musical equalizers?
Particularly, I want to understand
What mathematical features (e.g. monotonicity ...

I'm currently working with a dataset of $5000$ pulses of $N=15000$ samples each. I managed to implement the RLS algorithms with a FIR M-Tap filter such that $M\leq 15000$ ($150$ seems to achieve the ...

I have signal. I do Fast Fourier transform on it and now I have this output.
I want to delete the other except two that have high altitude. Can you help me ?
My f sample is 2500.
My complete time ...

I believe that there is no connection between the sampling frequency used for converting an analogue filter to digital filter and the one used to sample a signal that the filter will be used on. But I ...

I have to create a third-octave spectrum from a time signal on Octave GNU. I found some code on the net to help me but I don't have all the parts of the algorithm.
I have a .csv file which contains ...

I would like to know if it's possible to reconstruct the original time domain signal from it's time-stretched version? Is there any algorithm out there that can do this? Python, Matlab, etc?
I want ...

So it recently dawned on me that Bessel filters, despite being listed along with the other common types, are really an oddball that belongs in a different "class", and I'm trying to learn more about ...

I was reading Winder's Analog and Digital Filter Design and the section on Bessel filter.
I was hoping to see a complete derivation of the Bessel filter theory, but Winder's book gives only
\begin{...

I am working on a school project on converting a 6th order butterworth high pass filter to digital filter using bilinear transformation.
Just got a couple conceptual questions need to be clarified ...

I am new to DSP and filter design. I have developed a code in C++ to calculate FIR coefficients using Parks-McClellan algorithm. The inputs to calculations are:
Filter type (Low-Pass, High-Pass)
...

I am trying to design a Nth order continous time Butterworth filter in state space, with pole placement technique.
...

A digital low pass Butterworth filter that has been designed using Bi-linear transformation has been a pole at $z=0.6$. It is also known that the filter's attenuate (at digital frequency) $\omega = 1....

I am having trouble understanding the exact derivation of the butterworth filter and how it results in the output of the poles. I have researched multiple lecture series and textbooks and this is my ...

I have the strain signal of a lateral beam of a car measured at sampling rate 1,200Hz from data acquiring system. Even after using temperature compensation in strain gage, we are getting drift. So I ...

I learned that the time constant can be computed as $\frac{1}{2 \pi f_0}$, where $f_0$ is the half-power cutoff frequency of a high-pass filter.
However, I was wondering how the time constant and the ...

Actually I need to filter a raw ECG signal using low pass Butterworth filter at 70 hertz. And j need to implement the Butterworth filter first for that. Please any help since I'm very new to this ...

I work with depth time series data from electronically tagged fish. When the fish spend time on the bottom we get a prominent tidal signal (approx 1.96 cycles per day). This interferes with our ...

In steepest descent
methods of minimizing a function $f(x), x \in \mathbb{R}^d$,
it's common to approximate the gradient by finite differences:
$\qquad\qquad \nabla f(x) \approx gradest( x; h ) \...

To note, I am using Matlab R2017a for my signal processing.
I have filtered my EEG signal according to the frequencies I am interested in using butterworth (2Hz-30Hz). But there are some large ...

I'm trying to understand how to show that with real coefficients, the phase response of a filter is 0. Here is the impulse response
$h[n] = b_1d[n+1] + b_0d[n] + b_1d[n-1]$
How should I approach ...

I consider a signal of length $N = 2^n$ for some $n$. I want to derive two signal from it, one containing only the odd frequencies and one only the even frequencies. Each of these signals have length $...

What is the output of the convolution meaning which is used in the FIR filter?
how can conclude the output of the FIR filter in the output according to the input $x[n]$?

If anyone has a copy of this book could you please shed some light here: I was trying to reproduce the example shown in Chapter 5, section 5.3.1 Window Design Method's Figure 5-19 which illustrates ...

This is the simple code to find transfer function between sigout and sigin signals and then are the filter coefficients ...

I am reading a paper on dilated convolutions, and I have come across this piece of mathematics. It makes sense for the most part, except when the author says that the filter size is $(2r + 1)^2$. ...

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