Questions tagged [digital-filters]
The digital-filters tag has no usage guidance.
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0answers
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Why do additional harmonics arise after the filter bank?
I'm trying to implement direct implementation of MDCT filter bank.
So, some additional artefacts have appeared after all of the routines.
The spectrum of the initial signal looks like:
...
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1answer
40 views
Can CV or DS be seriously considered DSP roles? [closed]
This is not a technical question but can have a yes/no answer, and a why statement. My understanding about digital signal processing (DSP) is that it allows to take real-world analog signals (1D, 2D, ...
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0answers
24 views
Correctly interpret Butterworth filter frequencies
I am using the butter function from the signal package in R to filter my data.
The data is Fourier-transformed (complex numbers) - and I want to filter out specific frequencies (I have a 256Hz ...
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2answers
119 views
Floating point to fixed point in C
I've been trying to make a digital filter.
I've used the following transfer function to get my digital coefficients
https://gyazo.com/00195ef119f0b2ff93fa5855715abdc9
Which i've then transfered to ...
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1answer
35 views
How to design a a filter and how to determine the poles and zeros on the basis of following informations?
A sinusoidal signal of 300Hz frequency,3V amplitude is contaminated with line frequency (50Hz,1V). Design a digital filter including poles and zeros to remove the interference component. Consider the ...
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1answer
35 views
FIR filter design with time and frequency domain constraints
I deal with the problem of the synthesis of the filter with the requirements of both the frequency and
time characteristics.
Now there is a filter that meets the requirements.
Roughly speaking the ...
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1answer
54 views
Not able to reach minimum phase using Hilbert transform
My problem is pretty simple, I've designed a magnitude response and I would like to find the corresponding minimum phase filter. I'm using the code below and unless there is a bug my eyes don't want ...
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1answer
63 views
How to prove a filter has linear phase response?
I have designed an FIR filter to have linear phase response using odd-symmetry design. The coefficients of this filter are {2,1,3,1,0,-1,-3,-1,-2}. I am now being asked to prove it has linear phase ...
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2answers
85 views
Interpolation and harmonics
A real valued causal sequence $x1[n]$ exists with length of the sequence being $N$.
Valid indices of x conform to $0 \le n \le N-1 $
The DFT of x[n] is:
$$
X1[k] = \sum_{n=0}^{N-1} x1[n].e^{-j.2.\...
2
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1answer
64 views
Using Matlab's filter() function for solving difference equation with initial conditions
I am trying to solve a difference equation with initial conditions using Matlab's filter function:
$$y[n]=\frac12y[n-1]+x[n],\qquad y[-1]=1\tag{1}$$
For $x[n]=6u[...
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1answer
37 views
What is the type number of a discrete time system given $H(z)$?
Given a continuous time impulse response $h(t)$, if I take the Laplace transform and count the no. of poles at origin, that gives the type number of the system. For e.g., $$H(s) = \frac{2}{s(s+2)}$$ ...
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0answers
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How to find the filter coefficient of matched filter to detect the pulse of a signal?
I would to understand how to find the coefficient of a matched filter.
Let's consider we have a signal and the signal is buried in the following noise. we want to find the coefficient of a matched ...
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0answers
21 views
IFFT, negative frequencies and noncausal impulse
When using the IFFT to compute an approximation to the IDTFT, frequency range is 0 to 2$\pi$ versus $-\pi$ to $\pi$. This produces a different impulse sequence(causal I think).
How do you calculate ...
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1answer
27 views
Overshoot when using Bessel filter
Here is simple code to illustrate overshoot using bessel filter:
...
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0answers
25 views
What is a simple smoothing filter for signals sampled at non-uniform times?
An answer:
A simple smoothing filter for signals that has a DC gain of one, is a simple RC low pass filter. This filter has a exponentially rising step response,
$$O(t) = 1 - e^{\frac{-t}{\tau}},$$
...
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0answers
44 views
persist state of digitalFilter
Suppose I created a digitalFilter using designfilt (in matlab).
I am interested in using it to filter windows (I dont have the whole data set offline).
When I call filter() with this filter and a ...
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0answers
12 views
Frequency warping while creating filter coefficients
If I have an audio signal at 16kHz with frequencies band-limited to 8kHz. I want to low-pass filter this signal such that fc=4000Hz. Meaning when I reconstruct the signal I do not want any content ...
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1answer
28 views
confused in practical example of shift invariance?
I am reading book of w smith ,"The scientist and engineer guide to digital signal processing" 2 edition
I am trying to understand topic of shift invariance from theory given on pg 92
I have attached ...
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1answer
50 views
How do I initialize the state of a digital filter in Direct From II?
Suppose I have a digital filter implemented in Direct Form II. How do I initialize the state of the filter as if the input $x[n]$ had a fixed value $x_0$ for all $n<0$?
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1answer
45 views
How to characterize this whitening filter?
A little bit of background: I am interested in whitening in the least-square estimation.
In this sense, consider a univariate signal $y$. Assuming the covariance of $y$ is $\mathbf{A}\sigma^2$. ...
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0answers
85 views
signal interpolation (upsampling) by factor 2
The task is to interpolate the signal by increasing the sampling rate by factor 2.
Matlab has a dsp.FIRHalfbandInterpolator function.
Let's try using it.
...
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1answer
93 views
How to find lattice coefficients of all pole IIR filter with filter coefficient $k_m$ is 1?
Given the following system function
$$ H(z) =\frac{1}{1+\frac{2}{3}z^{-1}+\frac{5}{8}z^{-2}+\frac{2}{3}z^{-3}+z^{-4}} \tag{1} $$
We have to draw lattice structure for above system function
The ...
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1answer
37 views
Generating FIR filter in Web Audio API
I'm creating an HTML that applies a low-pass Windowed Sinc filter with a window size of 8192 to white, pink, or Brownian noise. There are nine radio buttons, eight of which set a cutoff frequency and ...
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40 views
Please explain Multiplication property in Z transform?
I am having problem visualizing contour any example will be great help.
As far as as where I need it I was trying to find Z transform of a contracted signal by first multiplication by impulse train.
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3answers
305 views
Protect an IIR filter from being reverse-engineered
I created a somewhat unique IIR filter and I want to protect the filter from being reverse-engineered
I think you all know it is quite easy to get all the different weights of an IIR by using impulse ...
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0answers
38 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 ...
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1answer
61 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.
...
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1answer
73 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.
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2answers
25 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 ...
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1answer
60 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
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1answer
27 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 ...
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1answer
63 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 ...
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1answer
116 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 ...
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1answer
123 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 $(A^c = [a^c_m, ..., a^c_1, a^c_0])$. ...
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1answer
78 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 ...
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1answer
56 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 ...
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1answer
60 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?
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1answer
80 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 ...
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2answers
53 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 ...
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0answers
55 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 ...
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1answer
391 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 ...
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1answer
136 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 ...
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1answer
86 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
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1answer
137 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 "...
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3answers
153 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 ...
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2answers
243 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 ...
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1answer
159 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-...
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2answers
132 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 ...
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1answer
54 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 ...
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1answer
52 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 $T$. ...
