Questions tagged [filter-design]
Fiter design is the process of designing a filter.
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When to choose ZoH over Tusin method to transform continuous to discrete
I have a continuous transfer function here: a 2nd Order Butter Worth filter
$$\frac{1}{6.416\cdot10^{-11}s^2+1.33\cdot10^{-5}s+1}$$
How would one choose which discrete method to use to transform ...
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17 views
Tool or code to design non-uniform filter bank (nufb)
Hell all, I am a research scholar. For my work, I have to use a non-uniform filter bank (NUFB). Please provide link for tool/code to design an NUFB. It would be preferable to have such tool developed ...
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1answer
21 views
Why does my signal magnitude increase after adaptive filtering?
I am using a series-cascade of multiple NLMS adaptive filters each with step size 0.0040, leakage factor 1.0, and 100 filter taps. My signal gains magnitude at each step of the filtering, say the peak ...
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1answer
51 views
Analytic solution for non-flat filter design
I'm trying to write an accelerometer calibration script that uses filters to convert from volts into $m/s^2$. As accelerometers tend to have non-flat response curves, this means I have to design a ...
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2answers
112 views
Removing Noise from time across signal data
I am new to the field of signal processing. I have sensor data (Impedance across time), that output is like this.
On the right side, I have a noisy signal portion in my data, marked by the red block....
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20 views
Capture the ambient sound
I sit in a cafe and start recording my voice, reading a poem. Then I sit at my room, at complete isolation, and I read the same exact poem once again and record it.
Now, Considering I have these two ...
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28 views
Removal of EEG noise using cascading adaptive filtering technique
I am required to filter out noise from EEG data using preferably Python or MATLAB. My dataset contains values for 64-electrode EEG along with their time-corresponding HEOG (horizontal eye movement), ...
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2answers
31 views
Can some one explain what is the meaning ( in real life) of anti causal (precursor) part of the channel taps?
The channel taps are classified into postcursor (causal) and the precursor (anti-causal) parts.
\begin{equation}y_{k}=\sum_{l=-L_{2}}^{l=L_{1}} h_{l} x_{k-l}+n_{k}\end{equation}
here $h_{l}$ is the $...
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29 views
Hydrophone signal calibration by frequencies
First of all excuse me if anything I post is completely or partially wrong (or poorly explained), I am new in signal processing and still learning a lot from this community, I will do my best to ...
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1answer
38 views
FIR Filter - Low Pass Filter coefficients to High Pass filter coefficients
I need to convert this 12 Tap Low Pass Filter to a High Pass Filter.
The coefficients of the Low Pass Filter are as follows:
...
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1answer
44 views
Finding poles of an abstract transfer function
When finding the poles of something like the following transfer function, would I be able to write $z=\sqrt[L]{\mu}$ since square roots aren't technically defined on the complex plane?
$$Y(z) = \frac{...
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HF ionospheric Channel Model Example in matlab : Watterson model
In the example file in matlab HFIonosphericChannelModelExample.m, the watterson model is simulated. In this model the channel is represented as one tap (for the sake of simplification) delay channel.
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81 views
why does DFE feedback filter matters?
I recently came across an article on the inventor of adaptative equalizer Robert Lucky, In this article : An Oral History by Robert Lucky (http://www.ieeeghn.org/wiki/index.php/Oral-History:...
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63 views
How to make low pass filter using frequency sampling method?
https://www.allaboutcircuits.com/technical-articles/design-of-fir-filters-using-frequency-sampling-method/
So there is two main equation:
I wish to filter out frequency $\le 10000Hz$, for example. ...
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25 views
removing noise using Scipy.signal.butter:
I am going to remove the noise from a brain recorded signal. It was a continuous recording and with sample rate=30kHz, it was digitized. So now it is a digital signal.
I have written the code here ...
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1answer
35 views
phase response of an analog filter
I am trying to understand the relation between phase response of an analog filter and its influence on the quality of a signal with PSK modulation.
For example I want to design a BPF for HF ...
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53 views
C-Weighting filter
I'm trying to implement "accurate" C-weighting filter for range 20Hz-20kHz by using other than bi-linear transformation method.
What I've achieved so far is 10th order filter by using Magnitude ...
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27 views
Why is the length of smoothed data from filtfilt always double of the input?
I am trying to filter the data I have using python with the code :
...
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3answers
68 views
best technique to remove noise from bio signal data
I'm still learning about signal processing so sorry if this is an intro problem. I want to remove noise from the following data so I can separate each wave and run analysis on it.
The data is sampled ...
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1answer
47 views
How is the formula of the cutoff frequency of a 2nd order Butterworth low-pass filter derived?
According to articles and notes, the cutoff frequency of a 2nd order Butterworth lowpass filter:
is expressed by the formula:
$\omega _c = {1\over\sqrt{R_1R_2C_1C_2}}$
and since $\omega _c = 2 \pi ...
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1answer
27 views
Low pass filter with different non linear phases [closed]
I want to design low pass filters with different non linear phases. Means the magnitude response of all low pass filters will be same but having different non linear phases.
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2answers
41 views
High Order Buttersworth Filter
Let's suppose i want to implement a Buttersworth filter of order 12 (N=12).
First of all ,I can decompose the Buttersworth polynomial in elemental polynomials:
Now, if i know how to physically ...
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2answers
42 views
Transfer Function vs Frequency Response
Talking about Butterworth filters my prof said that a filter is described by its frequency response $H(\omega )$.
What I think is that filter is really described by its transfer function, since there ...
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5answers
4k views
Is a notch filter for 50 Hz without delay possible?
I am processing an EEG signal (2 kHz sampling frequency) from hardware and the built-in filter doesn't work as intended. FIR gives a delay, and IIR is unstable.
The signal of interest is from 4&...
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1answer
39 views
Confused about filter's cutoff frequency in rational sampling rate converter
A interpolator which increase sampling rate by L times is followed by an anti-imaging filter, the filter is said to have a cutoff frequency of Fs/2L. Is Fs here is the frequency before or after ...
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37 views
What is effect of fractional sample delay effect on FIR Filters?
In the DSP lecture slides we have this slide:
And
So why this part of FIR Type 2 and Type 4 filter impulse response equation bolded in this slide?
Thanks for your attention.
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What is importance of the conjugate reciprocal and none reciprocal zeros in FIR Type 2 Filters?
In DSP class we have this slide:
So i like to know why this classification is so imoortant?
I have seen this post :
Conjugate reciprocal pairs of zeros and poles in ...
Assuming the impulse ...
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2answers
154 views
Justification of bilinear transform
I would like to understand the "justification" for the bilinear transform. The basic idea as I understand it is that by integration rule of Laplace transform we have for continuous $y(t)$:
$$\mathcal{...
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17 views
Where can I find more information about how to create filters on frequency domain?
I'm newbie and I'm very interested in filtering images on frequency domain.
I haven't found a lot filters on frequency domain and I wondering how can I create new ones.
How can I create new ones? ...
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1answer
21 views
Constraints for Type 3 Linear Phase Filter
This might be a strange question - but most of the Type 3 LPF I'm seeing are having this in common for $h[n]$, assuming $h[n]$ is real:
We cannot have outermost element of h[n] lower than middle ...
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0answers
22 views
Moving average filter using Blockram
I implemented ADC on FPGA.
Sampling frequency of ADC = 1.563MHz.
Reconstructed signal frequencies range =0-3khz
ENOB achieved for 100hz signal is 7.0844 and SINAD is 44.81db.
Now, I designed a ...
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1answer
26 views
Removing vocal but retaining all background ambient noises
I have one audio file containing human speeches, a lot of ambient noises like audience laughter, birds chirping, sounds of natures etc..
Now I want to separate it into two audio files where one ...
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14 views
Phase concentration for a broad range of frequency over time
I have local-field potential signals from two animals and I am trying to do phase analysis on these signals. What I am interested in is how phase concentration (or interatrial phase coherence) changes ...
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1answer
30 views
Parallel form filter design
I have a transfer function coefficients as following:
[b,a] = ellip(4,.2,40,[.41 .47]);
I'd like to factor this system into its PFE using ...
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0answers
43 views
pressure decay filtering
So, I've been dealing with a series of pressure decay curves from a pressure transducer. Each one has a waterhammer effect as a consequence of shutting down pumps. Here are examples of 3 curves:
I'm ...
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1answer
32 views
Cut-off frequency in Fourier-Spectrum of a noise message
I have a problem involving cut-off frequency.
I have a voice message in an audio file .mat which have an eletromagnetic noise. I am trying to recover the original message by applying digital ...
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36 views
Bandwidth in the filter-Hilbert method
I understand that the Hilbert transform on broadband data produces uninterpretable results. And so we first bandpass the data, and then run the Hilbert transform.
My question is, how wide can the ...
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69 views
Convolution of a non-symmetrical window function by a cosine signal in the frequency domain
A have a time signal:
The associated DFT spectrum of this signal:
The time signal can be considered as a non-symmetrical rectangular window function multiplied by a cosine signal with a frequency $...
2
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1answer
90 views
What are the limitations of FIR filters?
I know that for type II FIR filter it adds a zero at $z=-1$ so it cannot design a high-pass filter.
My question:
what type of limitations the other types have (type I, III, IV)?
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1answer
68 views
Real Fast Fourier Transformation (FFT) in 2D changes with changing the axes order
I am using Python to calculate the real FFT for a 2D array. I found that the real FFT function does not return an array with the same size as the input array, rather, it returns an array with the same ...
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1answer
100 views
Use recursive least square as an “actual” filter?
I am currently studying recursive least square (RLS) and as far as I understand, the setup is (for example) that we are given a process with uncertain parameters/noise and we want to adapt parameters ...
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1answer
34 views
To find the filter Co-efficients of the following equation
Given $y[n]=h_1[n]x[n]+h_2[n]x[n-1]+h_3[n]x[n-2]$ is an LTI system with unity gain at $\omega =0$ and zero gain at $\omega =\pi$. $h[n]\neq0$ .Also given that the system has a linear phase. Compute
$...
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1answer
48 views
How to make signal smooth using filter?
I have measured the output of a mechanical system and obtained the signal shown at the attached image. I want to make it smoother by removing the chattering (I think it is called chattering). In order ...
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1answer
59 views
Is gradient vector of pole zero carry usful information?
Consider a transfer function (TF) with $n$ number of poles $(p_1,..p_n)$ and $m$ number of zeros $(z_1,..z_n)$. One can write the magnitude of the frequency responce of the TF in terms of poles and ...
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22 views
Bandpass Experiment: Weird graph! amplitude mismatch?
Why is this generic BP filter seem to magnifying amplitudes when I run it at lower sample rates? The bass range seems to 32 times louder at 1275 Hz (44.1 kHz / 32) than at 44.1kHz?
Here is a graph of ...
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2answers
144 views
Where should I set my anti-aliasing filter corner frequency for this signal?
I have a Fourier transform of an arbitrary digital signal. The transform has For 1024 frequency bins. The magnitude plot of the first 100 is below, with no surprises in the unplotted 924 bins.
Now, ...
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1answer
46 views
How do I interpret this block diagram correctly?
I am reading notes ahead of the class and have encountered this particular slide:
While I completely agree with the first block diagram, I am at a loss trying to understand how the second block ...
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36 views
Wiener Filtering to perform minimum mean square estimation
As the Title suggests, I am trying to design a Weiner Filter for getting the lower bound on my LTE channel estimation algorithm.
To give you a high level on the chain, First after generating the ...
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1answer
85 views
Are MATLAB function zp2tf() and tf2zp() are complementary or not?
I was under impression that given, pole, zero and gain the transfer function (filter coefficients b and a) is fixed. Therefore, ...
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1answer
56 views
How to smooth filter response?
I have a filter designed in matlab with the function cheby2( N, Rs, Ws, 'stop'). The filter would give nice frequency response with for a given parameter set when the filter order is 2 or 4 (N=4). But ...
