Questions tagged [filter-design]
Fiter design is the process of designing a filter.
887
questions
1
vote
1answer
22 views
Finding number of alternation in parks mcclellan
I'm given this question and asked to find number of alternation for filter 1 and 2, using parks mcclellan algorithm.
I search for solution and find this answer :
My question is why the number of ...
0
votes
0answers
10 views
window function considerations as a bandpass in discrete sine transfrom
Having to sine-transfrom a truncated signal, I apply a window function in the truncated region to reduce the effect of side lobes after transforming, in a sense it acts as a bandpass fitler. However, ...
0
votes
0answers
27 views
Python equivalent code for Matlab bandpass function
I have used a bandpass filter function in Matlab as follows
...
1
vote
1answer
37 views
Impulse Invariant method for digital filter design
One of the known methods for discretizing analog filters is impulse response invariant. We get the impulse response in time domain, discretize it and then get the Z transform.
What I am trying to ...
0
votes
0answers
24 views
Remove noise from distance calculated from accelerometer data
I got distance calculated from accelerometer raw datas using double integral accelerometer.I attached the image below.But it shows noise with it.Please help me to remove this noise.
0
votes
0answers
71 views
Reconstruct $x(t)$ from $y(t)$ and $z(t)$
Let $x(t)$ be band-limited signal with $X(j\omega) = 0$ for $|\omega|\gt \omega_M$. We use $$s(t) = \sum_{k =-\infty}^{+\infty}(-1)^k\delta(t - \frac{kT}{2})$$ for sampling. So we have $z(t) = x(t)s(t)...
0
votes
0answers
41 views
what rule has coefficients to follow for an IIR filter to be stable? [duplicate]
I'd like to play with IIR filters, but not by designing them with algorithm but rather play with the coefficients directly.
i got a stability problem though, because of course IIR filters are ...
1
vote
1answer
43 views
The Stability of the RLS Algorithm for Equalization
I am reading in the litterature that LMS is more stable than RLS.
But RLS is far more faster in convergence.
So my concern is how to be sure that my RLS algorithm will be stable when I am doing ...
0
votes
0answers
23 views
How to deal with different audio formats for audio classification?
I am working on an audio classification problem statement to classify between two audio classes. I have collected samples from jotform, they are providing audio widget to collect .wav audio but it ...
1
vote
1answer
40 views
Deriving of magnitude, phase response and impulse response of transfer function
I have a transfer function as
$$H(z) = \frac{-0.0625z^4 + 0.25z^3 + 0.625z^2 + 0.25z - 0.0625}{z^4}$$
I want to derive magnitude and phase response of this equation.
can someone help me from here I ...
0
votes
1answer
22 views
Direct form 1 filter equation for a biquad filter
In general, we all know that the direct form 1 equation for a biquad filter is given by
$$y(n) = b_0x(n) + b_1x(n-1) + b_2x(n-2) - a_1y(n-1) - a_2y(n-2)$$
But when it comes to practical applications ...
0
votes
2answers
30 views
How to plot poles, zeros and impulse response of a transfer function
I want to plot poles and zeros as well as impulse response of a transfer function that is given below:
$$H(z) = -0.0625z^4 + 0.25z^3 + 0.625z^2 + 0.25z -\frac{0.0625}{z^4}$$
For poles and zeros, I ...
1
vote
2answers
67 views
3 dB cut-off frequency of first-order IIR high-pass filter
As descipbed in this great answer (thank you Matt L.), the suggested exact formula was very precise.
I need a first order high-pass with similar precision.
EDIT: (clarification)
The low-pass filter ...
1
vote
2answers
38 views
FIR filter with relaxed transition band
Is there a way to design a lowpass FIR filter with a more relaxed transition band than what MATLAB FilterDesigner tool generates?
What I intend to do is to reduce number of taps needed to implement ...
0
votes
1answer
31 views
changing filtering “speed” (cutoff frequency) of an IIR filter knowing only its coefficients
i got an IIR filter bu I have only the coefficients.
now i'd like to be able to change the filter characteristics of the low pass filter (the "cutoff" frequency) but all i got are a and b ...
0
votes
1answer
38 views
Converting Analog Filter into Digital Filter, why Bilinear Transform?
What's the advantage of using the Bilinear Transform?
$$H_d(z) = H_c(s)\bigg|_{s=\frac{2}{T_s}\frac{z-1}{z+1}}$$
When you can just use this equation?
$$H_d(\omega) = H_c(\Omega)\bigg|_{\Omega=\omega/...
0
votes
1answer
39 views
ARMA Filter Output Stationary and set up?
I have questions regarding ARMA Filters.
Is the output of a ARMA Filter stationary or just wide sense stationary?
I do know that you can obtain an ARMA filter by connecting an MA filter with an AR ...
0
votes
1answer
29 views
Match an Equalizer Curve
I’ve passed a sine sweep wave file into an equalizer plugin I have, now I want to model it in matlab so that I can ‘copy’ the equalizer setting sound. I’ve both in.wav and out.wav audio files and ...
1
vote
0answers
21 views
Why filtic function does not work for a specific use?
And how can I solve this issue?
I designed a filter via bilinear transformation in order to obtein a high-pass filter of Butterworth. When I want to set initial conditions with filtic function, it ...
0
votes
2answers
80 views
Distortion of OFDM signal when using RRC filter
I am trying to transmit OFDM signal in pass-band based on RRC filter, but what I noticed that the signal before the RRC filter is completely different compared with signal after the RRC filter (It ...
0
votes
1answer
28 views
Butterworth band-pass filter transfer function
I have some complex data in the frequency domain and want to multiply by an appropriate function to get complex data in the frequency domain that is appopriately filtered.
I think such a function ...
3
votes
1answer
60 views
How FIR filters provide a linear phase?
I know that we can design a linear phase FIR filter by choosing filter coefficients symmetric or antisymmetric, what is the intuitive idea behind that?
0
votes
2answers
40 views
Why when you use any filter, it results first a steepness at the beginning to a certain level?
Is there any tool to avoid that? For instance, I wish after filterin my signal, the filtered signal start from a certain level, in other words, at the offset level.
In this case, I've use Butterworth,...
0
votes
0answers
6 views
For single tone transmission impaired by IES, how Doppler Spread is translated in Channel impulse response computation?
If we are to transmit in single tone transmission in a Single Input Single Output configuration. And the channel is impaired by awgn and inter symbol interference with the following characteristics:
...
0
votes
1answer
32 views
invfreqz() and frequency domain filter specification
I am trying to go from a time-domain description of a filter, via frequency domain, generating a filter based on that frequency domain description, then seeing how far from the original I end up.
<...
0
votes
1answer
36 views
Remove AC component from the mixed signal using an elliptic filter
I have a signal like this:
$s(t) = C + a \cdot sin(2 \cdot \pi \cdot f \cdot t)$
where $C$ - arbitrary constant, $f$ - frequency of AC component (Hz) and $a$ - amplitude of AC component.
Parameters:...
1
vote
1answer
109 views
Why a pair of complex-conjugate zeros provide a nulling filter? (FIR filter case)
FIR filter having a pair of complex-conjugate zeros that lie on the unit circle, with zeros of the form:
$$
z_1 = e^{j\omega_i}\qquad\text{and}\qquad z_2 = e^{-j\omega_i}
$$
And transfer function:
$$
...
0
votes
1answer
29 views
Do we need Channel estimation prior to (time domain DFE or linear) adaptative equalization in single tone transmission?
If we are to transmit a single tone signal over a channel impaired by inter symbol interference and awgn, and if the equalization is adaptative (LMS/RLS track) and is done in time domain by LE (linear ...
1
vote
1answer
37 views
How to remove the bump from the signal using filters?
I want to have only the slope of the signal. How can I do? I have tried to apply some filters but so far I have not been able to hit the nail on the head. I use Matlab.
0
votes
1answer
40 views
Order of filters
on the 24th page of this pdf file for Slaney's Gammatone filter bank this sentence is written:
The result is returned as an array of filter coefficients. Each row of
the filter arrays contains ...
0
votes
0answers
31 views
Droop compensation - How to do it?
Lets say i have a LPF response that has some kind of droop (some attenaution in the passband, usually could be due to cic filter how to make CIC compensation filter,,,, or can also be due to large ...
0
votes
0answers
44 views
Why does the frequency in fft plot of the baseband modulated signal isn't changing at the receiver
I have designed a transceiver system for cellular frequencies(800MHz). On the receiver side, I have converted the band signal to baseband signal through digital mixing. I have added the Inphase and Q ...
1
vote
2answers
108 views
Digital low pass filter vs continuous low pass filter,?
What is difference between both implementations of low pass filter? Discrete/digital implementation and continuous implementation?
As far i am able to understand
In case of discrete implementation,...
0
votes
0answers
24 views
Filter design with arbitrary response [duplicate]
I would like to design a feedforward filter for a current controller. I could of course run MATLAB to brute force an arbitrary response for me, but I have some ranges "don't cares" in the response ...
1
vote
1answer
25 views
Why is the frequency measured in radians in filter design?
This is my first time posting a question, so sorry if my articulation is not proper.
I'v just started out with FIR filter design and I see the domain being -$\pi$ to $\pi$. Now I understand that ...
0
votes
1answer
58 views
ouput of low pass filter
$$y[n]=(-1)^n w[n]+w[n]$$
$n$ ranges from $-\infty$ to $\infty$
where $w[n]$ is output of frequency response block with cutoff frequency = $\pi/2$, and input is unit impulse function.
I know ...
0
votes
1answer
33 views
How to design a filter that can filter out noise accurately, after setting the parameters of the filter using standard signal?
I meet a problem with designing a filter. I have two different instruments that could measure the same AC signal (usually ~200hz, always <1kHz), A and B. A can carry out signal measurement during ...
0
votes
2answers
30 views
IIR Cascaded Filter Response, Unexpected Results
I am attempting to understand how to manipulate portions of a cascaded series of biquad notch filters in a control system I am working with.
As part of determining the composite response of all the ...
0
votes
2answers
54 views
impulse response cascaded with time reversed of itself
Consider a filter with real-valued impulse response $h[n]$. The filter is cascaded with another filter whose impulse response is $h'[n] = h[-n]$, i.e. whose impulse response is the time-reversed ...
0
votes
0answers
23 views
The parameters of tolerance scheme of a FIR filter
in the first part of the problem they told us that an FIR filter with impulse response h[k] ( z-trafo H(z) l should be designed to the following parameters.
And then in the second part it says that ...
0
votes
2answers
32 views
Average image over $N$ frames with one frame buffer
Can you give some filter that is averaging an image over $N$ samples in a recursive way that only needs one frame buffer?
$\frac{1}{N} \sum_{k=1}^{N} y_{k}(m,n)$
I can only imagine filters that need ...
1
vote
1answer
26 views
How are these 2 frequencies aliases of each other?
I'm trying to understand this solved example from a book, given an analog signal $x(t) = 3\cos(100\pi t) $. This signal has a frequency of $50hz$. If sampled at $F_S=75Hz$ ,I obtain the discrete time ...
2
votes
1answer
52 views
Polynomial division to separate FIR and IIR in cascade
I have a FIR filter $L(z)$ and an IIR filter $H(z) = \frac{B(z)}{A(z)}$ which are cascaded together as $L(z)H(z) = L(z)\frac{B(z)}{A(z)}$.
For specific reasons I want to calculate a parallel topology ...
0
votes
2answers
95 views
Getting a single feedback peak from a basic 4 stages phaser
I try to set a very basic 4 stages phaser (two 2-pole all-pass filters), with a particular feedback path where only one peak is raised at center frequency when the feedback is above 0.
The following ...
4
votes
1answer
272 views
Why can't realisable IIR filters have linear phase?
I am studying IIR Filter Design and came across this arbitrary statement in my textbook which says that 'physically realisable and stable IIR filters can not have linear phase'. Would really ...
1
vote
2answers
61 views
How do you calculate the group delay in samples of a Biquad Filter at a given frequency from the coefficients?
If I have a BiQuad all-pass filter with a given set of coefficients such as:
g: 0.903228
a1: -1.88142
a2: 0.903228
B1: -2.08299
B2: 1.10714
And my operation per ...
0
votes
1answer
43 views
Building a FIR filter from an arbitrary frequency-magnitude response curve (eg. Least Squares fitting) [duplicate]
I'm trying to create a FIR filter from a magnitude equation, where my starting equation provides the magnitude (amplitude) between 0 and 1 for any given frequency in Hz.
I posted the magnitude ...
1
vote
2answers
44 views
what does the order of filter mean?
Recently, I read a paper. In it, it said "a $32^{nd}$ order 100MHz bandpass filter". So, how the filter was designed? Does it mean this filter have 32 taps?
0
votes
1answer
49 views
How to invert FFT these magnitude equations to get FIR filters?
I need filter designs that will reproduce or approximate each of the following:
1) ${\rm magnitude} = e^{-cf^{2}}$
2) ${\rm magnitude} = e^{-\frac{a+b\sqrt{gf}}{df}}$
Where: $a$, $b$, $c$, $d$ and $...
1
vote
0answers
20 views
Why are state-space filters good for time-varying systems?
"The state-space filter representation is especially powerful for time-varying linear systems."
Why is this the case?
What is the explanation behind the claim?
