Questions tagged [digital-filters]
The digital-filters tag has no usage guidance.
1
vote
0answers
28 views
Band Edge Filters
I am interested in using Band Edge Filters to do carrier frequency recovery but this is kind of secondary to my question which is:
How do you build an FIR filter which isn't symmetric about the ...
0
votes
1answer
52 views
Extremely low frequency filters
I have a data acquisition board(A/D+Digital Signal Processor) and I want to check if a digital high-pass filter(implemented in DSP) at an extremely low cut-off frequency(0.05Hz) is actually working.
...
0
votes
1answer
44 views
Why is the Z-transform so important in digital filters analysis and design? [closed]
Please elaborate on why this mathematical transform can help analyzing as well as designing any type of digital filter.
0
votes
2answers
22 views
What is it called when you have a single-pole recursive digital lowpass filter where the coefficients don't add up to unity?
Is there a name for a single-pole recursive digital lowpass filter where the coefficients don't quite add up to 1?
For example: with the filter ...
3
votes
1answer
48 views
Is it possible to design a band-pass filter with fewer taps than a series HPF + LPF?
I am digitally filtering a signal that is streaming live. The goal is to reduce the filter delay as much as possible while maintaining the quality of some measurements based on the filtered signal. ...
0
votes
1answer
16 views
Why do cutoff frequencies of a high-pass digital filter differ from the ones i used to transform it from a low-pass analog filter in MATLAB?
I need to transform an elliptic analog low-pass filter to a digital high-pass filter but i'm having a problem because the resulting filter has lower cutoff frequencies than the ones i set. Here is the ...
0
votes
1answer
34 views
How do I know what is causing the noise in my signal
I have a noisy audio file, available here (It is safe to download if you wish to)
https://ufile.io/pstrt
Now, I inspect it in matlab:
And then i plot it on normalised frequency:
Now I need to ...
1
vote
1answer
106 views
Computation of parameter filter to match a given frequency response
I'm looking for the practical way to compute the parameters ($a_1$, $a_2$) of a digital filter to match a certain frequency response. I'm studying the 12-poles filter of the vintage component SP0256 ...
0
votes
1answer
65 views
How can I get the continuous-time transfer function coefficients (or poles and zeros) from the corresponding discrete-time TF and vice versa?
Let's say I have a continuous time transfer function which I know its numerator coefficients $(B^c = [b^c_m, ..., b^c_1, b^c_0])$ and denominator coefficients $(A^c = [a^c_m, ..., a^c_1, a^c_0])$. ...
1
vote
1answer
18 views
How does cascading low pass modules affect pass band ripples?
This question 10.21 is taken from GATE IN paper 2015.
I don't understand the answer given,
I mean why is the frequency response at the pass band frequency considered to be $(1-\delta)$, isn't the ...
0
votes
1answer
31 views
Low pass to low pass transformation coefficient?
I am unable to solve this question, 10.10 from GATE IN 2004 (a previous year question paper for an exam targeted at engineering graduates in India.)
So I tried to solve the 10.10 by considering the ...
1
vote
1answer
38 views
Does the output signal from the FIR filter have transient response?
In matlab, I have shown the signal passed through the FIR filter in the time domain. Then the beginning of the signal is like the transient response. Is the result appropriate?
2
votes
1answer
36 views
Why is interpolation a time varying system
I was reading about interpolation (Interpolation and Decimation of Digital Signals - A tutorial Review, Ronald E. Crochiere) and found that Interpolation filter is a time varying system. Can someone ...
-1
votes
2answers
44 views
Digital Filter Design using difference equations
I am reading a chapter on digital filter design from analog filter design using difference equations. What they do first of all is that they map $s$ (Laplace variable) to $z$ ($z$-transform) by the ...
1
vote
0answers
54 views
Digital or analogue filtering?
I'm trying to self learn the art of signal processing whilst moving through my third year pure maths degree.
Sorry if my terminology is incorrect however I hope I am understandable!
I am looking at ...
2
votes
1answer
177 views
Sobel and Prewitt operators decomposition
I was reading in the wikipedia page of Sobel operator and Prewitt operator that is possible to decompose these two operators (quote form the Formulation paragraph):
"as the products of an ...
3
votes
1answer
98 views
What's logic behind the constuction of Sobel's filter in image processing?
I am basically very new to this image processing field. I am presently working on edge detection on colour images. While learning the basics of edges and edge detection in images, I encountered image ...
0
votes
1answer
51 views
Backward finite difference differentiation filter frequency response
As title says what would be frequency response of backward finite difference differential filter, or what would be error of this differential filter, analyzed upon frequency of a signal?
2
votes
1answer
109 views
How does a digital filter work?
To introduce my situation: I'm developing a digital synthesizer in a form of a C++ library, working with low level APIs like WASAPI, ASIO, ALSA etc. It's probably not very practical and I'm mostly "...
1
vote
3answers
75 views
Using white noise to test filter freq. response?
If I want to test a digital filter, is using white noise a good way? I mean input white noise to the filter, and visualize the frequency out from the output.
Is that procedure the same as testing a ...
2
votes
2answers
230 views
Alias-free digital nonlinear filter design
@Jazzmaniac has a good answer to the question of how to design an alias-free digital nonlinear time-invariant filter here: https://dsp.stackexchange.com/a/28787/18276
Basically, according to that ...
1
vote
1answer
103 views
Design 2nd order sections IIR filter for high orders (more than 500)
I am currently building a Reverb that needs to fit on a DSP. I am using the model proposed in this link https://ccrma.stanford.edu/~jos/pasp/Freeverb.html.
It involved to create some Feedback Comb-...
2
votes
2answers
120 views
Detect valleys of a signal
I have raw values that are coming directly from a sensor (on the fly). These values are coming from a wearable sensor installed on a cows neck. When the cow ruminates (meaning that it chews food that ...
-1
votes
1answer
38 views
Analog equivalent for transition width in Digital radios
I have started off with GNUradio and I am not sure if I understand the transition width parameter available in GNU radio companion filter blocks .
As per online documentation transition width of a ...
1
vote
1answer
39 views
Best digital noise reduction in optical spectra measurement
I am using an optical spectrometer to measure the reflectance of some surfaces; the output is essentially a set of photodiodes' signals: for each photodiode I have a discrete time signal every $T$. ...
0
votes
1answer
62 views
Band pass filter (digital resonance)
Q1: Consider a discrete-time band-pass filter (BPF) that uses only two poles as
$$ H(z) = G \frac{1}{\big(1-r \, e^{j \omega_0} z^{-1}\big)\big(1-r \, e^{-j \omega_0} z^{-1}\big)} $$
where this ...
0
votes
0answers
42 views
Can we sample the signal with any period as long as the Nyquist rate is respected
I have a question regarding the sampling rate. Suppose that we transmitted a signal $m(t)$ then at receiver side, we received the continuous signal $x(t)$, which is time varying and sampled with a ...
2
votes
1answer
62 views
How can I get the polyphase components of a signal
My question, What is the polyhase components of the signal? What does mean a polyphase component for a signal?
Is it true that example :
...
0
votes
1answer
38 views
designing FIR narrowband pass filter using windowing technique
I m designing FIR narrowband pass filter using windowing technique. is bandpass filter is same as narrowband pass filter? I m not finding any good article on this.
0
votes
2answers
70 views
High/Low Pass Filters from Zero
I everyone,
I'm a computer scientist but not from the signal processing community. I want to implement a real time digital low pass filter (to begin with) for a sound processing application. I've ...
0
votes
0answers
35 views
I get strange BER when I build a code of OFDM in MATLAB with multipath
I have built the code for OFDM using Multi-path channel, but when I plot the BER, I get it abnormal results . (please, any help) .. It's urgent please
here is the code I made:
...
1
vote
1answer
230 views
How to calculate the delay generated by a 2nd order all-pass filter
I'm trying to simulate the dispersion effect of a stiff string in a digital waveguide system, by using second-order all-pass filters.
I use a second order all-pass filter with this transfer function:...
2
votes
0answers
70 views
How can I oversample the received signal by double of sampling time and have two signals
I have a question, how can I make the sampling in Matlab for the received the signal by half of time duration of symbol (Ts), and have two copies of signal (that's the process of fractional sampling ...
1
vote
1answer
313 views
How to plot the poles and zeros of this Bandpass Filter in MATLAB?
So I'm trying to design a band pass filter in MATLAB (with a center frequency of 10kHz and a sampling frequency of 44kHz).
I have calculated the transfer function but I'm not sure how to enter this ...
0
votes
0answers
21 views
“Necessary condition for lpf to lpf transformation is that a magnitude response must keep its original values at $\omega=0$ and $\omega=\pi$”?
"Necessary condition for lpf to lpf transformation is that a magnitude response must keep its original values at $\omega=0$ and $\omega=\pi$"?
Why is it so?
But does that not essentially read that ...
3
votes
2answers
161 views
What happens when linear phase is critical? [duplicate]
I am studying the topic digital filters. And in the slides it said,"if linear phase is not critical, Infinite Impulse Response (IIR) filters yield a much smaller filter order for a given application". ...
0
votes
1answer
597 views
Z-domain transfer function to difference equation
So I have a transfer function $ H(Z) = \frac{Y(z)}{X(z)} = \frac{1 + z^{-1}}{2(1-z^{-1})}$. I need to write the difference equation of this transfer function so I can implement the filter in terms of ...
1
vote
2answers
180 views
what is the frequency selective channel and doubly-selective channel
Could anyone please help me to explain the frequency-selective channel and doubly-selective channel and the difference between them in brief and easy way. I have tried to check on internet, I couldn'...
0
votes
0answers
26 views
Why can Generalized eigenvector algorithm for blind equalization be used in SIMO?
I have a question as mentioned the title.
I don't know if it's possible to use the Generalized eigenvector algorithm for blind equalization in SISO, but in most of papers I've read, they are talking ...
0
votes
2answers
91 views
Analog vs digital
If there are means to process all signals after they have been captured and digitilized, why are we still designing ICs to do analog signal processing? For any giving case, can not we just digitilize ...
0
votes
0answers
119 views
Simple Pole-Zero placement bandpass filter doesn't work in MATLAB
I'm trying to make a simple bandpass filter using Pole-Zero Placement method, which has unity gain at frequency w=0.37pi and 3 dB cutoff frequency wc=0.42pi (has a gain of 1/sqrt(2) at that frequency)....
2
votes
3answers
501 views
What is the filter with the less phase shift?
I have to analyze a dynamic signal but there is too much noise so I applied low pass filter but then there is too much phase shift.So what is the most reactive filter I can apply to my signal ?
Best ...
6
votes
4answers
270 views
Make a signal that fits another the best possible with a limitation in the 2nd derivative
Consider this step function:
The signal that "fits" this should look like the following (in green):
The corners are now smooth because the maximum second derivative allowed is not infinite anymore.
...
1
vote
1answer
45 views
Show that $S[n]= \frac1M \sum e^{\frac{2\pi l n}M}$ for $l = 0$ to $M - 1$
I am learning about image processing. Currently, I am reading about upsampling and down sampling.
Here is some background what I am reading:
I was wondering if someone show me a proof of how ...
0
votes
1answer
53 views
is there a way to apply a median filter diagonally?
For example, some studies show how to smooth a spectrogram with a 2D median filter vertically and horizontally (along time or frequency). My question is how would I apply 2D median filter along the ...
0
votes
1answer
112 views
Understanding and Correcting phase shift in General Linear Phase FIR filters
I have been learning about General Linear Phase Filters and the four standard FIR implementations.
I was wondering in the case $\beta = \frac{\pi}{2}$ or $\beta = \frac{3\pi}{2}$ if there was a way ...
-1
votes
1answer
344 views
What do the filter coefficients in digital filter represent?
Could you help me? How can I understand the function of the filter coefficients practically? In the simple case, it is the impulse response of the LTI system. but how do they work?
0
votes
0answers
149 views
Online Frequency and Amplitude Estimation with an Extended Kalman Filter
I'm trying to reproduce some results from a research paper that uses an Extended Kalman filter to estimate the instantaneous frequency and amplitude from accelerometry data. However, I don't ...
0
votes
1answer
364 views
Why is the second-order-section cascade approach giving “worse” results?
It seems that as a general rule, for digital IIR filters of order more than 2, one is encouraged to use a cascade of second order sections, for reasons connected to the poor stability of high-order ...
5
votes
2answers
613 views
What is the largest “safe” order for the digital Butterworth filter of a given signal?
Say I want to apply a zero-phase Butterworth bandpass filter, implemented in MATLAB using butter and filtfilt. So, for an order-$...
