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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EEG filtration via EMG
i got little problem. I'm trying to detect HFOs in single channel EEG (the recording is multichannel, but there is not enough channels to use ICA) for my diploma thesis experiment and i want to first ...
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Initial conditions of backward filter for forward-backward filtering on chunked real data in python
I have to apply a downsampling filter on a realtime data stream (signal) and I want to do it in python. The data is sampled at 16 kHz and loaded in a loop in chunks of 1 second. I have to downsample ...
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Poor Attenuation With High Pass Filter
I am having issues with my implementation of a FIR high pass filter. The low pass filter has an acceptable level of attenuation in the stopband, but the high pass filter has poor attenuation in the ...
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Notch Digital Filter
I am given a notch digital filter with the $z$-transform being:
$$W(z)=MF(z)F(z^{*})^{*}=M\frac{z-q}{z-p}\frac{z-q^{*}}{z-p^{*}}$$
where $M$ is the normalisation factor, $q=e^{-i2\pi\frac{f_0}{f_s}}$, ...
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Basic audio denoising in the frequency domain using minimum statistics?
I'm trying to do some example elementary denoising of the audio signal. Let's say input is speech with constant traffic background noise.
First I calculated block-based overlap-add Fourier transform (...
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Deriving the difference equation of the filter
Here I have a diagram (at the bottom) and I am trying to create a difference equation for the filter.
I've been using practice examples to try figure this one out however I am unable to produce a ...
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Relation between filter and convolution?
https://www.quora.com/How-is-signal-filtering-done-with-convolution
What is the relation between convolution and a filter?
Is it mainly that convolution process is used to determine the output of an ...
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Scipy filter analog vs. digital
It would be greatly appreciated if the usage of the python package scipy's filter (e.g. butter) analog=True argument could be explained. I don't understand what is meant by this (any signal being ...
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Bandpass filter with a changing passband
The problem:
I have a signal that contains multiple, relatively stable frequency components and I want to extract only one of them. I attached a plot that illustrates the situation with a simple ...
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IIR bandpass filter attenuates frequencies within the pass band
I applied Butterworth filters (order=5) that have different cutoff frequencies to a simple signal composed of four sinusoidal oscillations and computed its PSD.
<...
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Minimum signal length for notch filter?
I'm back again with another probably very basic question, but I searched a lot about the minimum length of signal for a filter to work, and mine seems to satisfy this (from filtfilt requirement in ...
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Frequency components in Bin averaging
I am back with another question.
Context for the question: I am trying to smooth out the angular velocity data from an encoder. The encoder has 720 ppr and the rough angular speed of the wheel is ...
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Output Precision for Filter
I am designing a low pass filter which takes a 19 bit input.
\begin{align}
\log_2(\max(\mbox{LPF_COEFF})) &= 16.64\\
\log_2(\mbox{sum}(\mbox{LPF_COEFF})) &= 17.87
\end{align}
Now should I take ...
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What are the step by step actions taken in order to convert PDM data to PCM data?
I would like to convert PDM (Pulse Density Modulation) microphone data into PCM (Pulse Code Modulation) data. I understand that ...
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Can Synchrosqueezing be use to Derive IIR Filter From Impulse Response
I am looking for alternate solutions to determine the IIR filter coefficients from the impulse response, or more specifically determine the closest IIR filter for a given FIR filter. Much of this is ...
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Linear phase impulse response causes non linear phase frequency response in GNU Octave/ Matlab
Assume linear phase FIR symteric impulse response like [1 2 3 4 5 5 4 3 2 1],shoud resault in a frequency response with linear phase negative sloped phase. but it ...
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Calculating the filter coefficients for 2nd order bandpass and bandstop Butterworth filter
I want to calculate the filter coefficients for both bandpass and bandstop filter for 2nd order Butterworth filter, like
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What parameter to use to decide if accelerometer data are noisy?
I would like to do regression thus I need to know if my input data are noisy.
In this case, I generate accelerometer data [x, y, z] from the sensor.
I have a look ...
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Matlab "resample" function
Say I have a signal x(n) and I want to resample from rate 30Hz to 20Hz. In Matlab, I can make a filter with:
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Notch filter with recursive math [duplicate]
Bit over my head in math here
I want to do a notch filter in real-time programmation with codesys (the structure text here look alot like C)
so I have this: Linear phase Notch filter / Band reject ...
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50/60Hz suppression Filter Implementation
is anyone could help understand this 50/60Hz rejection filter?
The basic information:
Fs=1MHz, DOR(data output rate)=25SPS, with both 50Hz & 60Hz rejeciton, settling within 1 cycle, 1/DOR.
The ...
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IIR Filter order vs group delay
From what I understand FIR filter order is same as it's group delay because of the linear phase and every frequency being delayed by same amount.
For IIR filter, filter order only tells that filter ...
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"There's no ideal lowpass filter" - really?
Sinc is $\propto 1/t$. If $x(t)$ is bounded, then there exists $t: |x(t)/t| < \epsilon_M$, where $\epsilon_M$ is machine epsilon. If $x(t)$ is also time-limited, it also means there's $\tau$ such ...
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How do I match a filter to an AWGN channel?
I have the task to find a transition that fulfills the first Nyquist Criterion. After that I am supposed to match that filter to an AWGN channel. For the first part I thought a raised cosine ...
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Real coefficient linear phase FIR filter design based on least square
I want to build up linear phase filter by least square approach.
I have formulated a least square problem similar to Matt's blog. To approximate the desired complex response $F \times 1 $ vector $d$ ...
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How to remove PWM cycles present in a signal?
I have applied PWM across a DC motor. The voltage measured across the motor is shown above.
I want to count the ripples present in the data. The ripples exist at a different frequency than the PWM ...
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LTI system: can I infer the system is causal based only on the transfer function without the ROC?
Suppose we have an linear time-invariant (LTI) system which acts on discrete signals. Suppose someone tells us the transfer function is: $$H(z) = \frac{1}{z-2},$$ but doesn't specify the ROC. Now the ...
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FIR filter with no ripple (only one zero at Nyquist frequency)
I need to design a short, symmetric low-pass FIR filter with a monotonic frequency response (no ripple). In other words, the differential of its amplitude response will be zero or less for all ...
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Matlab: How to implement a bandpass filter to isolate harmonic with 0 gain at this frequency? [closed]
I want to isolate the harmonic at 440Hz and I need to design a filter with zero gain at 440Hz.
I tried this with matlab:
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Using Zero-Phase Anti-Causal Filters in Real-Time Embedded Systems
Wanted to know the feasibility and usefulness of implementing Zero-Phase Anti-Causal filters such as those mentioned at this link in modern embedded signal processing applications given the ...
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Reconstruction of original signal after minimum phase filter
The question is pretty self explanatory. I want to know if it is possible to recover a signal originally injected into noisy data (coloured gaussian noise) after whitening the data with an AR minimum ...
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What type of filter is that?
I have a transfer function in z-plane with two poles and two zeros. I plotted the function with matlab
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How do I use the Lazzarini-Timoney DSVF as a sine generator?
I've tested out the Chamberlin digital state variable filter (DSVF) as a sine generator as per this well-known approach. At $f_0=686\text{Hz}$ tone and $f_s=48000\text{Hz}$ it sounds fine and the FFT ...
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How we determine type of filter with pole(s), zero(s)? [duplicate]
Let's say we have this Laplace transform:
$$H_{1}(s)=\frac{1}{(s+1)(s+3)}\;, \; \Re{e} (s)>-1 $$
So, we know that there is a poles at $s=-1$ and $s=-3$.
With these informations, we found that to be ...
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Creating a polyphase filter
I need to design a 3/2 resampling filter using polyphasic filters. On my original filter I have coefficients $$h[n] = [a_0, a_1, a_2, a_3, a_4, a_5, a_6, a_7, a_8, a_9, a_{10}, a_{11}]$$ making it an ...
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Resampling in frequency domain
I have a problem to solve, I need to get the frequency spectrum from the first plot, and then transform it into the second plot using resampling and filtering.
My thought process was, I can upsample ...
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how to implement 0.05Hz high pass filter?
How is the high-pass filter with a cutoff frequency of 0.05 Hz(or 0.2 Hz or 0.5 Hz) implemented in Real time for ECG devices?
What kind of filter or algorithm is used?
The order of FIR filter is high. ...
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How can I correlate two noisy voices in order to enhance their result?
I have an array of two microphones, spaced out, which continuously transmits audio input to a central processing unit. There is a slight delay between them, which is sometimes variable because they ...
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FFT convolution question
Designing a signal filter - I'm trying to understand how it is possible to take the FFT of a filter kernel and multiply it by the FFT of the signal to provide signal filtering as surely the FFT of the ...
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Algorithm for Hue correction behind HSL sliders in image processing software
I have a project where i need to recreate hue specific hue adjustment sliders like in Lightroom/Capture One/DxO photolab.
Initial image.
Image after hue adjustment (Orange hue range) in DxO Photolab....
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Pole Quantization Patterns in 2nd Order IIR Resonators
Background
My typical approach to fixed point design for digital filters is to iteratively scale and increase quantization while comparing the fixed point simulation to the floating point design ...
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Why does the filter function in matlab get different results from the IIR filters using direct form?
I have 6 first order filters, each one has 2 coefficients (B(x1,x1), B(x2,x2) for the numerator and A(y1,y1),A(y2,y2) for the denominator).
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What does frequency filtering actually practically mean?
I've got an electronics circuit which was filtering between a Passive HPF stage (1uF and 330K) and an Active LPF stage (4.7uF and 10K). As you can see it was filtering between 0.48Hz and 3.38Hz, and ...
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How to Find Elliptic IIR Filter Order?
Can anyone explain to me how to use this equation to find the Order?
Let's say
$$\Omega_p/\Omega_s =0.509$$ and $$e/\delta = 0.00349$$
How can I find the Order? I'm so lost i don't know how to use ...
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Trying to implement a digital A frequency filter (continuation)
Following this post some user advised that the filter developed there should be broken down into first and second-order sections and cascaded. In a paper i found the same thing is told:
So i made ...
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Distributed Arithmetic FIR Vs basic FIR digital implementation
I make a comparison between the basic FIR filter Vs Distributed Arithmetic FIR
First: basic FIR
Second: DA FIR
Then, I implemented Distributed Arithmetic FIR Filters by 2 methods:
1st: Serial
If the ...
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How does the state estimate selection work?
I have been solving following problem. I have two open loop state estimators used for estimation of the unmeasurable states of a given linear dynamic system. The first estimator provides estimate $\...
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Are allpass filters maximum-phase systems?
There are few notes online stating that an all-pass filter is a maximum phase filter (e.g., Link). The core of the claim is that an all-pass filter is a maximum phase filter since its zeros are ...
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FMCW chirp data acquisition
I am trying to understand the full mathematics behind FMCW. Using linear-frequency chirp $x(t)$ that has center frequency $f_c$, bandwidth $\Delta f$, duration $T_r$, and amplitude $A$, the ...
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Can a filter be fit to explain a response?
I'm trying to classify a stimulus-response relationship and would like to fit a filter that best explains the response. I don't really work in DSP so there may be some vocabulary I'm missing, but ...
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