Questions tagged [filter-design]
Fiter design is the process of designing a filter.
1,223
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Resolution after Oversampling, Filtering-Decimation
The ADC I am working on has a resolution of 24 bits and a Signal-to-Noise and Distortion (SINAD) of 108.4 dB. Assuming only quantization noise is present, the Signal-to-Noise Ratio (SNR) of the ADC ...
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Separating 'heart sound' from 'lung sound'
I have audio files recorded from electronic stethoscope and in those files I want to filter out heart sounds and retain just the breathing sounds. How can I do this using just the signal processing ...
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low shelving filter frequency response diagnosis
I am building a second order modified biquad filter based on an implementation by Will Pirkle in "designing audio effect plugins in c++". However when I look at my implementation's ...
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group delay of Bessel filters
Wikipedia states that Bessel filters have a group delay within passband of $1/\omega_0$. While the inverse relationship to knee frequency makes sense, it ignores the effect the filter order has on the ...
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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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Problem by designing and filtering a Bessel-Thomson filter in Python
I have an issue when I want to filter a signal with a 4th order Bessel-Thomson filter with a cutoff frequency at 18 GHz.
1. Generate data at 50 Gbps (T-spaced)
...
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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
$$
H(\omega) =
-1 \...
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Calculating the settling time of lowpass and bandpass SOS filters constructured using SciPy
I have constructed a high pass filter using scipy:
...
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Trying to understand scipy.signal.kaiserord
I am trying to understand scipy.signal.kaiserord example. The example says
We will use the Kaiser window method to design a lowpass FIR filter
for a signal that is sampled at 1000 Hz.
We want at ...
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Understanding the parameters of a notch filter
Adobe Audition has a very interesting notch filter that have been very useful to me. It has the following parameters
According to their documentation, the chosen frequency is the center frequency and ...
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Filter coefficients conversion from Matlab for FPGA, follow up
My question is a follow up of original question, specifically the answer provided by @Dan Boschen Dans answer.
The question is about converting floating point coefficients to fixed point signed ...
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Intuitive explanation of magnitude-phase-relationship for minimum phase filters
I know that, given the magnitude response $|H(e^{j \omega})|$ of a filter $H(z)$, it's minimum-phase response is given by
$$
\phi(\omega) = -\mathscr{H}\Big\{ \log(|H(e^{j \omega})|) \Big\} \ .
$$
I ...
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1
answer
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FIR filter design with nonlinear phase from measured amplitude and phase responses
I am having trouble when design FIR filter fitting to the complex data (i.e., amplitude and phase responses from measurements). I did try to use Matt. L's lslevin method here since this method is to ...
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Raised cosine filter and root raised cosine filter IQ for modulated signal QAM/QPSK
If I am passing a QAM signal through Root raised cosine Tx filter and measure the constellation at output, it seems very bad. If this bad output again pass with other another root raised cosine Rx ...
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Theory behind transfer function transforms of the type $H(s) \rightarrow H(f(s))$
My brother wanted me to derive a high-pass version of a Butterworth low-pass filter. I found that the transform $H \left( s \right) \rightarrow H \left( j - j s \right)$ does the thing, but I can't ...
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Given two low-pass digital IIR filters, find bandpass coefficients
In the case of FIR filters it is easy to get a band-pass filter by subtracting the coefficients of two low-pass filter filters or by convolving a high-pass and a low-pass filter:
...
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single tone -> multi bandpass filter -> incorrect output
background:
Trying to filter out all other signal frequencies except -> signal whose frequency can be in multiple's of 16Hz {16,32,48.......160].
so started experimenting with a single tone [no ...
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Scaling for windowed differentiator FIR filters
I recently took a dive into DSP and I was wondering what the appropriate scaling factor for a wide-band differentiator is.
Let's derive the impulse response of such a differentiator.
$$\begin{aligned}
...
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Type 1 Chebyshev 3rd Order Lowpass Filter
I'm trying to implement a 3rd order Type 1 Chebyshev digital lowpass filter in C++. I've been having a lot of problems and I'm not sure why. I can plot the transfer function in MATLAB and see that it ...
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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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2
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How to normalize FIR filter gain from the amplitude response in frequency domain?
I've got a set of measured data (amplitude and phase responses) of a filter. And I want to normalize its gain to be unit (equal to 1) for further processing. I know that we can convert to time-domain ...
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Numerical issues in scipy's Savitzky Golay filter coefficients for large polynomial order
Consider the design of a Savitzky-Golay filter of window length 101 and (high) polynomial order 20. Using scipy version 1.10.1, the filter coefficients can be obtained in python as:
...
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Least squares filter with non-linear phase and independent weights for phase and magnitude
Intro
My question is related to a previous one linked here.
I am interested in non-linear phase FIR filters with a specific desired phase response.
After I tried the options in the linked question I ...
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Matching two different type of equalizer responses
AutoEq (by Jaakko Pasanen) is an utility for headphone equalization. I'm trying to make a EQ converter for SuperEQ (source code for implementation found in DeadBeef), which can be found on few ...
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2
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Low-pass Filter with constant phase response
I would like to design (e.g. in GNU-radio) a low-pass filter where up to the cut-off frequency the phase-response is constant (I do not mean a "linear-phase response"). The rationale is to ...
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Design a Real FIR with arbitrary Phase Response
Intro
My question is related to this one: Correcting phase response of a signal.
I need to design an FIR low-pass filter with real coefficients for data received from a sensor. An additional ...
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1
answer
64
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How to trade poles of All pole filter with zeroes?
I have filter of order 2000 IIR all-pole. I want to implement it via cascade IIR filter. Is there any way to reduces poles order and increase zeroes as its cost. To balance zero and poles. Since all ...
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1
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Converting transfer function to cascaded IIR filter [closed]
I have two function, But I don't know which one is correct to convert my Transfer Function(tf) to secound order sections(sos).
I'm using GNU octave, and it have intrinsic ...
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1
answer
222
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High pass filter with low cutoff frequency
I'm trying to design the highpass filter in Python for my accelerometer data in order to integrate them. I need to preserve low frequencies. For example, data with a sampling frequency of 2 kHz need ...
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Create large amounts of motion blur using small filters
As part of an image preprocessing step, I need to generate vertical and horizontal motion blur with a filter that's 30 pixels long:
...
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1
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What are the tradeoffs between shifting a lowpass FIR filter vs using a bandpass FIR filter?
Is there a difference between shifting a lowpass FIR filter to an arbitrary frequency offset versus creating an equivalent bandpass filter, such that the passband and cutoff frequencies are identical? ...
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How to explain the ripple in this graph?
Hi everyone I need your help to understand this phase response would you please share your ideas?
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Filter choice for the signal with the possible time jitter in sampling and frequency aliasing
I'm trying to filter data from the force sensor mounted on the end effector of the industrial robot arm. Mechanical vibration caused by the robot's motion (possibly by actuators, reductors, etc.) ...
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How to produce a causal, linear phase filter kernel with an arbitrary magnitude frequency response
I have been reading The Scientist and Engineer's Guide to DSP to learn how to create filter kernels with an arbitrary frequency response (I design the magnitude response by hand). The method proposed ...
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What is the proper way to compute a real-valued time series given a continuous $1/\sqrt{\omega}$ spectrum?
I have never fully been able to wrap my head around Fourier transforms, so I apologize if what I am trying to do is trivial or violates basic theory in some way.
What I have is a "made up" ...
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Brrrrrrrrr clanky clank VHS noise
I have audio of VHS with speech and music, and with noise of frequency that's higher and lower than most of human speech, with apparently least but still significant energy in-between, that cannot be ...
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How to export raw audio from audacity?
Request to Experts:
I am a DSP novice. I would like to experiment with some methods for low pass filtering in the time domain.
To that purpose, I would like to generate a tone in audacity and export ...
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Allpass Filter Gain Issue
Background
I am having issues implementing an allpass filter to model wave dispersion in a stiff string. In order to simulate wave propagation in a string, I am using a digital waveguide. I ...
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Phase compensation using IIR allpass filters
As an exercise with IIR allpass filters, I am trying to compensate the phase of a lowpass Linkwitz-Riley filter using the Matlab integrated function iirgrpdelay.
The code I am using is the following:
<...
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Pre- or post-filtering for image doubling
I am doubling the resolution of images by bilinear resampling (which amounts to averaging). I seem to understand that this should be combined to a sharpening filter for better quality of the ...
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How to initialize states of second order direct form 2 transposed filter
Background
This question is an extension to this post. I need to calculate states ($S_1$ and $S_2$) of a second order direct form 2 transposed (DF2T) filter that result in minimum 'transient' when ...
2
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1
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Filtering after a phase domain ADC
While researching receiver architectures for the demodulating constant envelope signals like a GFSK, I stumbled over the concept of phase domain ADCs, where after
$$\boxed{\text{LNA}}{\longrightarrow}\...
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Some questions about realtime filter in control system
Can I use filtfilt function in realtime filter. filtfilt first filters a piece of data forward and then filters it backward. I ...
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Linear phase with a non-integer delay?
Let's say you are given a frequency domain specification for a filter design.
As a simple example, let's use an ideal (brickwall) LPF. Now the impulse response of an ideal LPF is a $\operatorname{sinc}...
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Integral Calculation in Matlab for Filter Optimization
I am trying to calculate the area under my FIR filter kernel using trapz but I'm getting weird results. The reason I want to know the area is eventually I would ...
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How can an impulse generate an output in the past time frame?
I am studying signal processing and currently doing signals & systems. While going through convolution and especially the impulse response , there are problems where LTI systems wherein the input ...
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Get the inverse transfer function from the measured response
What is a numerically stable way to obtain the inverse transfer function from a measured response?
I have a system that shows a low-pass behavior. I would like to increase the bandwidth by some form ...
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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 ...