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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Why is time domain low pass filter the "sinc" shape?
I'm looking at low pass filters, and I see that the time domain representation of an "ideal" filter resembles the shape above whereas the frequency domain is a box. I also get the as you ...
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Why is time domain low pass filter the "sinc" shape?
I'm looking at low pass filters, and I see that the time domain representation of an "ideal" filter resembles the shape above whereas the frequency domain is a box. I also get the as you ...
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Confused About Relationship Between Number of Points and Solving Wiener Filter Coefficients in Time Domain vs Fourier Domain
I'm interested in computing M, FIR filter taps / coefficients using a Wiener filter approach with known signal and noise timestreams. I know of two ways to compute the coefficients: one in the Fourier ...
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Is there any way of calculating the portion of signal lost as we keep on reducing the passband edge frequency of a band-pass filter?
Any kind of references would be appreciable.
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Which filter is suitable for reducing noise in feature detection?
I have a feature detection algorithm that is called FAST - Feature Accelerated Segmented Test.
It's a very fast algorithm for finding feature points, e.g "corners" inside an image. Here is ...
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RLS Adaptive Filter for estimating signal
I am currently working on a project where i am to estimate a signal x_T using x_1 and x_2 with an RLS filter.
I have a problem where i don't quite get the results i am looking for. I think there is a ...
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Understand basic Impulse Response from Impulse, can you help me visualizing it?
I was looking at this graph from The Scientist and Engineer's Guide to Digital Signal Processing, copyright ©1997-1998 by Steven W. Smith:
where basically it shows how differnt representations can ...
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How to interpret integer coefficients for a filter generated by Matlab
I have always used high-performance 32-bit microcontrollers with hardware FPU support. To calculate the coefficients of digital filters, I used the fdatool package in Matlab. I used single-precision ...
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Why lag compensator is preferred over PI for sinusoidal reference?
In this post, LJSilver mentioned that a PI compensator is not appropriate for a constantly changing reference, such as a sinusoidal waveform. In this scenario, a lag compensator is considered the ...
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Filter response weighted formula representation
I am trying to represent a filter response numerically using the formula shown in the end. I know its electromagnetic response.
Could you please help me understand how to make in Matlab such filter ...
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Compress filter coeficients?
if we have a FIR with 100 coefficients, is there a method to "compress" the length? so for example, have a similar response with 50 coefficients?
I am ok with the response degrading within ...
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FIR Filter Design in python
This is mine code for fir_filter_design in python. But I'm getting wrong output for low-pass, band-pass, and high-pass magnitude response.
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I'm struggling to understand the FIR Filter and its application
I'm new to GNU Radio, and I was watching this video However, I didn't quite understand the purpose of the FIR Filter. I know what decimation or downsampling is because I have the book "Digital ...
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Allpass Filter with Sign Switch
I want to design a digital allpass IIR filter with the following transfer function.
$$
H(\omega) =
1 \textrm{ for } \omega < \omega_1 \textrm{ or } \omega > \omega_2
$$
and
$$
H(\omega) =
-1 \...
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Calculating the settling time of lowpass and bandpass SOS filters constructured using SciPy
I have constructed a high pass filter using scipy:
...
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How does function c2d in MATLAB manage fractional delay?
the function c2d allows to convert a continuous laplace transfer function to a discrete z-transform transfer function. The base method is the Zero Order Holder. In example:
...
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2x oversampling channelizers and perfect reconstruction
For perfect reconstruction, it seems that $2\times$ oversampling allows a prototype filter that ensure the spectrums of the subchannels overlap correctly.
In the gnuradio reference on this it seems ...
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ISI and RRC basic concepts
I've been studying a kind of "introduction" for a few days now to get to talking about equalizers and ISI, and I would like to know if I have understood the things correctly or if I got ...
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Applying Lowpass filter on a signal in time domain gives ringing artifacts - how to get rid of them
EDITED AND UPDATED:
My signal is an impulse response. In the following figure:
The frequency spectrum of the impulse response (original signal) is shown by blue line.
The original signal is ...
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Trying to understand scipy.signal.kaiserord
I am trying to understand scipy.signal.kaiserord example. The example says
We will use the Kaiser window method to design a lowpass FIR filter
for a signal that is sampled at 1000 Hz.
We want at ...
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Filter coefficients conversion from Matlab for FPGA, follow up
My question is a follow up of original question, specifically the answer provided by @Dan Boschen Dans answer.
The question is about converting floating point coefficients to fixed point signed ...
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Why does an asymetric transfer function cause a non linear group delay?
I know that an IIR filter exhibits non linear group delay (so different frequencies can be delayed more) and that this is related to the fact that it is asymetric (due to considering previous outputs),...
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Phase correction of a received signal
How is it possible to correct the possible phase distortion of a typical signal which has been received by an antenna and passed through different analog elements including an amplifier and a band ...
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Intuitive explanation of magnitude-phase-relationship for minimum phase filters
I know that, given the magnitude response $|H(e^{j \omega})|$ of a filter $H(z)$, it's minimum-phase response is given by
$$
\phi(\omega) = -\mathscr{H}\Big\{ \log(|H(e^{j \omega})|) \Big\} \ .
$$
I ...
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FIR filter design with nonlinear phase from measured amplitude and phase responses
I am having trouble when design FIR filter fitting to the complex data (i.e., amplitude and phase responses from measurements). I did try to use Matt. L's lslevin method here since this method is to ...
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Does block-based FIR filtering necessarily imply using the DFT?
Assume I want to filter a stream of data with a rather small FIR filter (in the order of 16 to 64 coefficients). In any case I have to split the input stream into chunks (blocks) of data.
Searching ...
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Time-varying shot noise generation
I'd like to model the behavior of photodiodes, the input is like data bits (0 & 1) where each bit is represented maybe with 100 samples / bit and the bit rate is $B$ bits/sec with period $T_{bit}=\...
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Why does causality imply that the system function is analytic?
It is cited in multiple places that the fact that a filter is causal (i.e. the impulse response is zero for t < 0) implies that the system function is analytical.
I couldn't find any proof of this, ...
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How to design low pass filter for this case
From control systems perspective, I have two subsystems that are connected in a cascade structure. An outer loop represents a joystick that provides position that runs at 1kHz and an inner loop ...
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Weiner Filter - why does this computation explain that the necessary filter is a weiner filter?
$X_1(t), X_2(t)$ are random WSS processes with expectation 0, and correlation functions $R_{X_1}(\tau), R_{X_2}(\tau), R_{X_1,X_2}(\tau)$ $n(t)$ is a white noise with SPD $S_n(f) = \frac{N_0}{2}$ ...
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Total delay of cascade FIR filters
I have a doubt regarding the total delay in terms of taps for cascade filters.
Imagine that I have an SOS IIR Filter (2 delays) and an FIR filter of 10 taps (10 delays) and another of 20 taps (20 ...
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Does bandwidth include negative frequencies?
I've seen countless definitions of bandwidth that disregard negative frequencies.
Is that done due to the symmetrical nature of a certain signal eg. a cosine or due to the inherent lack of meaning ...
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Matlab - dfilt filter produces different results for SOS and normalized SOS
I think the title says it all. Anyone could provide a solid answer to why it happens? Here is a sample code to replicate the issue.
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What is the name of a low-pass filter that tracks rate of change?
When applying a low-pass filter to a constantly changing signal, there will be a lag between the actual value and the filtered value. For example, if f(x) = x, the ...
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Can the standard deviation of the Gaussian window in a Gabor filter be made infinitesimally small?
My understanding is that the standard deviation of the Gaussian window in a Gabor filter dictates the temporal resolution.
Wouldn't it always be better then to make the window smaller, thus achieving ...
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What is the difference between STFT and Gabor filter?
My understanding is that a Gabor filter gives us information about when in the input signal does the frequency of interest (equal to the frequency of the sinusoid in the Gabor Filter) occur, thus ...
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Unexpected frequency components after applying bandpass filter in python
When I apply a bandpass filter in python to a pure sinusoidal signal, it seems that after filtering I get different frequency components across the spectrum (see orange line below). It looks like ...
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Digital Low Pass Filter and PWM Signal
I have a PWM signal going into an ADC.
After the ADC the signal is being filtered using an IIR single pole filter.
Performance is exactly what I want with one exception. The purpose of the filter is ...
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What does a Gabor filter "filter" out?
Almost all of the searches online have returned results related to image-processing, but I am looking for a more rudimentary example with some physical intuition (eg; a high-pass filter only lets the ...
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Why does convolution give the output of a passing a signal through a filter?
I have a rudimentary understanding of Convolution, the Convolution Theorem and why the output z(t) of an LTI system can be found using the convolution of input signal x(t) and the impulse response h(t)...
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removing spikes from spectrum
I am working on 50 samples obtained from time domain signal by FFT. It appears at certain frequencies which are -- according to our experimentalist -- due to some issues with their oscilloscope. What ...
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Modeling an Acoustic Reflection from a Wall - a Paradox?
I am trying to simulate the reflection of a sound ray, that goes from a sound source, bounces off a wall, and is received by a microphone.
The wall has a an absorption coefficient, and a specular ...
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Change filter frequency sample by its coefficients
I have a 3 stage biquad filter, so I have 15 coefficients, that filter was created for a 48kHz signal,
AFAIK, filter coefficients are calculated in a normalized frequency matter, can they be scaled of ...
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Lowpass filter to downsample data
I have some data that is sampled at a certain rate (16384Hz). I am only interested in the 10-30Hz region, so I want to downsample the data. My idea to do this goes as follows:
I apply some kind of ...
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CWT: selection of scales and filtering
I am using CWT to obtain scalograms of ECG signals as the one below:
In my research, most articles contain a pre-processing step before performing CWT (that usually includes signal normalization and ...
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Low-pass filtering a clipped signal
In order to downsample a signal sampled at 48KHz, I implemented an anti-aliasing filter. An Elliptic LPF with a cutoff at 16KHz and order of 10. Everything looks OK until the input to this filter is a ...
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Unique Root of a DoG filter
I have a mathematics problem that can be related to Signal Processing. Let $f$ be a real function, striclty increasing, striclty convex on $(-\infty,0)$, strictly concave on $(0,\infty)$. We ...
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Why the difference of a vectorized VS non-vectorized FIR convolution differ?
I have an FIR filter that is a low pass at 11025Hz for 44100Hz, 461 taps. The reference implementation, i.e. naive convolution, works as expected. Now I've written a vectorized implementation to take ...
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IIR/FIR equations for custom frequency response
Given a frequency response plot where gain is 0dB up to Fc and after Fc gain increases linearly by factor K (example chart below):
I am seeking how to arrive at its IIR/FIR equations so it can be ...
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