New answers tagged filters
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How to Compensate for Frequency Offset in single carrier transmission Using Coarse and Fine Compensation?
What is the name of the FFT-based technique whose code is given in the second answer? Can I find information about this in any paper?
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Find smoothed first derivative from signal with noisy slope
If anyone here is curious in a computationally efficient solution (say you are working on a small microcontroller like Arduino), you can run two alpha filters (Exponential Smoothing) with slightly ...
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Help on audio filter with FFT on python
I've heard that some sounds in most languages use some very high and low pitched frequencies, so it would make sense it doesn't sound complete after those frequencies are removed. Getting rid of the ...
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Does bandwidth include negative frequencies?
Single-sided and double-sided bandwidth needs to be specified explicitly to be clear. The double-sided bandwidth of a modulated signal at baseband (including the negative frequencies) corresponds ...
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Matlab - dfilt filter produces different results for SOS and normalized SOS
First of all the difference is extremely small, it represents a numerical noise level of about -230dB, so it's indeed numerical noise. Keep in mind that you are using floating numbers, so all values ...
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Matlab - dfilt filter produces different results for SOS and normalized SOS
The result sum(abs(y1 - y2)) doesn't tell you much, because its value depends on the length of the input signal. You could look at the value of
...
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What is the name of a low-pass filter that tracks rate of change?
I do not think that there's a specific name for this type of lowpass filter. There are indeed similarities between the cascade of two lowpass filters as suggested in the OP's answer, and a combination ...
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What is the name of a low-pass filter that tracks rate of change?
I was able to remember how the filter works. The idea is very simple, a second low-pass filter tracks the steady-state error in the result of the first one, and it is then added to the output:
Based ...
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Can the standard deviation of the Gaussian window in a Gabor filter be made infinitesimally small?
If your gaussian window width goes to zero, you are, in essence, sampling the signal, $x(t)$, (that was multiplied by the complex exponential, $e^{j 2 \pi f_0 t}$, which always has a magnitude of 1) ...
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What is the difference between STFT and Gabor filter?
The Gabor transform is just a special case of the STFT, the main difference is, like you said, the Gaussian window.
One characteristics of window functions is that they are $0$ outside a specific ...
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Unexpected frequency components after applying bandpass filter in python
When you "turn on" filtering at time zero, the filtered output starts with a transient
plt.figure('signal filtered')
plt.plot(t, samples_filt)
To make ...
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Linear phase with a non-integer delay?
I don't think I had seen this question before today. If it's essentially how to find the coefficient set for a pure bandlimited delay of a non-integer number of samples, we first break the delay into ...
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Linear phase with a non-integer delay?
I'd like to add a response as I just computed some stuff, which hopefully is correct, but everything adds up, so I am hopeful!
I assume a system response $H(e^{j\omega}) = e^{-j\phi(\omega)}$, so a ...
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What does a Gabor filter "filter" out?
Multiplication in time-domain is convolution in Fourier domain. You have $\mathscr{F}\{\sin\}$ (assuming $\phi=0$) which is two Dirac-Delta impulses in the Fourier domain. The FT of a Gaussian (the ...
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Why does convolution give the output of a passing a signal through a filter?
You can describe the system with an operator acting over an input $x(t)$ transforming it into $z(t)$. If $L$ is the operator, $z(t)=L[x(t)]$.
Remember that the system and the operator are linear and ...
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Accepted
Why does convolution give the output of a passing a signal through a filter?
The convolution operation essentially computes a weighted sum of the input signal's values, with the weights determined by the filter. This process allows filters to capture patterns, features, or ...
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removing spikes from spectrum
These spikes look like harmonics of some sort of line noise at $250 \texttt{MHz}$. To get rid of these, what you need is a comb notch filter.
There are a few ways to build one.
If the fundamental (...
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Simulating analog filter using convolution or converting with fft
I need something similar in Python but I can't understand what's wrong...
creating simple 1th order low-pass Butterworth filter, Fc=10Hz
# creating I order, 10 Hz, low pass Butterworth filter
fc = ...
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Modeling an Acoustic Reflection from a Wall - a Paradox?
It seems that either the assumption of constant time delay across the frequency spectrum must be wrong then?
It's indeed a wrong assumption. Physical systems are causal and zero-phase doesn't exist ...
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Modeling an Acoustic Reflection from a Wall - a Paradox?
From the fact that your answer contains a matplotlib generated figure, I assume that you are doing this in Python, thus discrete time.
Discrete time means bandlimiting, which means low pass filtering ...
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