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.
0
votes
1answer
21 views
Hamming Window LUT function
I'm working on a project where I need to generate a LUT for Hamming Window to perform a FFT.
I'm using this function (doing the table with MATLAB):
...
1
vote
0answers
23 views
How to determine number of poles or zeros in prony's method?
I would like to use Prony's method for signal modelling. I want to design a filter such that its impulse response is equal to the message signal. I use the function ...
0
votes
1answer
21 views
Upsampled input to an Adaptive filter?
I will try to explain the issue I am having as clearly as possible without going into my coding or maths. I have my own and a MATLAB Central implementation pf standard LMS in MATLAB. Fixed step size. ...
1
vote
0answers
38 views
Locate first minima in signal
I have many datasets which all looks roughly like this:
All the datasets have this in common. First there is an artifact around 0 with an unknown (but likely very high) intensity. Then there is some ...
0
votes
2answers
42 views
Should i high pass audio for real time frequency detection?
Haven't been able to find a clear answer on this. I'm interested in 75-500Hz bandwidth. It's for a tuner(android app) so i want a pretty quick response time.
Is it worth the computational time to ...
0
votes
1answer
21 views
convolve to differentiate black and white colors
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 ...
0
votes
0answers
30 views
How do band pass filters work [closed]
What's the technical and mathematical explanation on how a band pass filter in the discrete time domain can allow one frequency to pass through and attenuate/suppress others?
I know how a nulling ...
0
votes
0answers
23 views
Data bias due to filtering [closed]
If I sample a signal that is purely random and filter it, for example using a high pass filter on projection images generated from Monte Carlo simulation to remove overlapping of the Fourier-...
1
vote
2answers
91 views
How to know the filter order
Could anyone explain to me how can I know the filter order based on algorithm?
For example:
$y[n] = \frac12 x[n] – x[n–1] + \frac12 x[n–2]$ has order filter 2.
$y[n] =2 x[n] – x[n–1] + y[n–1]$ has ...
3
votes
1answer
34 views
Why does the separable filter reduce the cost of computing the operator?
A separable filter in image processing can be written as product of two more simple filters. Typically a 2-dimensional convolution operation is separated into 2 onedimensional filters. This reduces ...
0
votes
2answers
87 views
A property of all-pass filters
Let $G(z)$ be a real coefficient stable all-pass transfer-function with
degree greater than zero. Then it can be shown that $|G(z)| < 1$ for
$|z| > 1$: Given that the poles occur in complex ...
0
votes
1answer
61 views
Analogue filter analysis (band-pass)
The transfer-function:
$$G(s) = \frac{\beta s}{s^2 + \beta s + \omega_0^2}$$
is to be used in an application that requires the magnitude of the frequency response to be of the band-pass form.
...
0
votes
2answers
93 views
IIR coefficients and difference equation implementation in C language
I can't find anything on this topic, so either I'm in the wrong direction or else there is just nothing about it on the internet.
So let's say I have 3 $b_i$ coefficients ($b_0,b_1,$ and $b_2$) and 2 ...
-1
votes
0answers
36 views
A great book for engineer? [duplicate]
I am hardware engineer, but yet the whole area of DSP is still not really clear to me in terms of practical application= writing actual code.
I am looking for some book, but not another "gray" book, ...
1
vote
1answer
45 views
Calculating cutoff frequency for Butterworth filter
I have a problem while calculating cutoff frequency, suppose we have these specs.
Firstly, I calculated the order of the filter and got $N=5.8858$ and round it up to get $N=6$.
Now I'm supposed to ...
0
votes
0answers
30 views
What does this fourier transform of 2D filter do?
I have fourier transform of discrete time 2d filter. Would like to see what does filter do when we apply this to any image. It is a good idea to impelement in matlab but i dont know how to get inverse ...
-1
votes
1answer
79 views
Filtering everything outside 20 - 20000 Hz
With an audio .wav file (sampled at 44.1 or 96 Khz), I would like to filter out everything outside the 20 - 20000 Hz range.
I tried a Butterworth bandpass filter and there were unstability issues, ...
0
votes
1answer
55 views
Sine chorus effect
I want to create a sine chorus effect function in MATLAB where the inputs and the outputs will be: y=chorus(x, f_sine, delay, depth, mix, fs). What I'm trying to do ...
0
votes
1answer
19 views
python rewrite lfilter (iir) with for
it's a beginner question, but usefull to users from python - signal.lfilter , i was using lfilter from
Find reverse one pole lowpass filter
the doubt now is, how to rewrite the iir filter to 'revert'...
1
vote
1answer
25 views
Is there any difference between using convolution and correlation for finding edges with Sobel?
I know that Sobel is a filter for edge detection and we should use convolution to find edges, but is there any difference if we use correlation instead of convolution? I think Sobel tries to find a ...
1
vote
1answer
69 views
Nulling filter coefficients
I'm wondering if I have calculated these frequencies of a nulling filter correctly:
$\ 1, -2cos(0.44\pi), 1$
I have to make a nulling filter that filters out the frequencies $\theta = 0.44\pi$. I'...
1
vote
2answers
89 views
Find reverse one pole lowpass filter
I need to find a filter that revert the one pole filter of the current signal, a function using Python (or MATLAB) scipy.signal.filtfilt or ...
-1
votes
1answer
25 views
matched filter>>Why is the maximum output (s/N) does not depend on the particular shape of the waveform? [closed]
Why is the maximum output (s/N) does not depend on the particular shape of the waveform?
0
votes
1answer
50 views
Removing an unwanted sinusoidal signal using an FIR filter
I am learning about the $\mathcal Z$-transform and FIR filters and I do have a problem with the following exercise:
There is given a signal $$x[n] = s[n]+\sin\left(2\pi f_n n\right)$$ where $s[n]$ ...
4
votes
1answer
72 views
Is there such a thing as a moving bandpass filter algorithm?
I am looking for an algorithm, preferably available in a Python library, which passes only frequencies of interest, with the center frequency continuously changing as it moves through an array of ...
1
vote
1answer
34 views
How to calculate the output image with the following kernel?
What is the advantage of using this equation? I guess we may use Taylor series, but I tried my best and could not get the equation.
0
votes
0answers
29 views
How to implement a time-varying filter?
I'm working on a 10-second sound, sampled at 44.1 khz.
I want to do filtering, and have a desired EQ (equalization) curve that varies over time, as suggested here (here $f0=250\ Hz$)
How to ...
1
vote
0answers
31 views
Zero STFT bins (and not FFT bins)
Filtering by taking FFT, zeroing bins, and inverse FFT is a bad idea, as discussed here.
But what about:
take a STFT, (i.e. multiply the input signal by moving window function, and take FFT)
zero ...
0
votes
0answers
20 views
How to decide about length of vectors for estimating correlation matrices
I am optimizing a filter $f$ using mean squared error criterion.
$$MSE = f^T*(R_{xx}-R_{yx}^T*R_{yy}^{-1}*R_{yx})*f \tag{1}$$
where
$x\in \mathbb{R}^n$ input column vector
$y\in \mathbb{R}^m$ ...
0
votes
0answers
18 views
Sobel Operator in a single Kernel
The Sobel operator for the X axis ( Gx ) of a 2D signal is:
-1 0 1
-2 0 2
-1 0 1
The Sobel operator for the Y axis ( Gy ) of a 2D signal is:
1 2 1
0 0 0
-1 -2 -1
The resulting Gradient is ...
0
votes
1answer
71 views
linear phase notch filter matlab
I need to filter 9 Hz from a signal sampled by 256 Hz using a linear phase filter.
If someone can bring an explanation or a code example I would greatly appreciate it.
I have tested his code
...
2
votes
1answer
44 views
Why do Dyadic filterbanks downsample the high pass signal portions?
I'm currently programming a dyadic filter bank and have a question. I notice in all of the visual representations:
(from here (Dyadic Analysis Filter Bank)), the high pass filtered output for each ...
-1
votes
1answer
57 views
Why many DACs uses series of half band filters for interpolation instead of single one?
And why, according to some sources, these filters have different number of taps (the last is the shortest one)? Does it really reduce a cost of computation?
0
votes
1answer
44 views
Estimating Q/Damping Factor from noisy measurements
I have a damped, tuned circuit and want to measure its Q factor. The hardware sends an impulse 'ping' and samples the output as it rings down.
Is there an efficient way to fit an equation of the form ...
4
votes
3answers
119 views
Design of a digital A-weighting filter with arbitrary sample rate
I want to A-weight a time series with arbitrary sample rate.
An analog A-weighting filter is defined exactly by IEC 61672-1. But there's no definition for a digital filter. One method is to use the ...
0
votes
1answer
40 views
Frequency Response FIR Filter Bank
I want to determine the frequency response of a filter bank
from the input x[n] to the four output channels.
$H_0$ is an lowpass and $H_1$ is a highpass filter.
Both are FIR filters and I have the ...
0
votes
0answers
23 views
Extended Kalman filter to remove noise from Gyro signal (sensor fuson) then integrate it - Unexpected behavior
Good morning,
I'm trying to use an Extended Kalman Filter in order to perform data fusion between accelerometers and gyroscopes to remove noise from gyro signals, in order to get better angles when ...
0
votes
0answers
36 views
Filtering time series to forecast with specific error distribution
I've asked this question in Cross Validate already, where it was suggested I have a look here:
I have a time series of weather data (wind) $x[n]$. I would like to create a continuous-time signal $s(t)...
0
votes
2answers
62 views
Digital filter not removing noise at specific frequency Matlab
Doing some work at the minute on digital filters in matlab, I have a file with artifical noise added (sine wave added at specific frequency). The goal is to filter the signal and get it as close as ...
0
votes
1answer
45 views
MATLAB's $\tt filter()$ function for complex valued data [closed]
I want to use the filter() function to implement an FIR filter having complex data values as the input to it. For real, I used this function as ...
0
votes
2answers
94 views
Output signal as convolution of impulse response and step signal
Hi All: This question is kind of an add on question to the gorgeous answer by Juancho provided at this link:
How does this "simple filter" work?
The answer explains how the limit of ...
0
votes
0answers
29 views
How do we use signal::filter and signal::butter in R for EEG data?
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 ...
0
votes
1answer
55 views
How to interpret these filter coefficients?
I am trying to make sense of the filter coefficients.
The author claims that they are running a 3rd order lowpass Butterworth filter and were willing to give their design. They are not the most ...
0
votes
1answer
33 views
Why is being jointly WSS important in signal estimation with LMMSE estimator?
Assume that we have $y_k = s_k + n_k $.
We have observed $y_k$ and want to estimate $s_k$. The goal is to use LMMSE (Linear MMSE) estimator to find $y_k$ and on the other hand, we know that our ...
0
votes
1answer
29 views
Are circular buffers relevant to transposed direct form IIR filters?
I'm currently working on a project where I'm programming IIR digital filters. I have already finished the direct form 1 and & 2 implementations.
When doing direct form 1 and 2, I was able to ...
0
votes
1answer
90 views
DFT in IIR filters
To further summarize, I want to create a function in matlab that finds the time domain signal $y(n)$ and its $n$ time components ($n=0,1,2,...$) given the numerator and denominator of a transfer ...
1
vote
0answers
32 views
variance of filtered polynomial
Consider the following system:
What is the variance of $y$, $\mathbb{E}(y^2)$ ?
(EDIT: I know input signal has infinite power but will be made bandlimited by $H$. Both $H$ and $G$ are simple ...
0
votes
1answer
50 views
0
votes
1answer
46 views
Separating signal from noise in similar frequency range [duplicate]
As shown in the figures, both of my signal and noise have the same frequency domain, how can I separate them?
3
votes
1answer
73 views
How to find a 'square root filter' such that $d*\overline{d}=f$ given $f$?
I have a desired frequency response $F(\omega)$. I know how to create a FIR filter (say, with linear phase) $f(n)$ with this frequency response.
But how can I get a filter $d(n)$ such that $D(\omega)...
