Questions tagged [digital-filters]

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Determining window Size for Savitsky-Golay filter (Savgol_filter)

First, I am aware that an ideal optimal window task is really signal/task-specific, i.e. nature of signal/sampling rate/characteristics of interest etc. What I am interested in: "Is there an optimal ...
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2answers
81 views

Downsides of higher order filtfilt's

I understand that with a standard filter, higher order will yield sharper frequency selection at the cost of increased phase delay. However, with so-called zero-phase-distortion filters like that ...
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1answer
141 views

MATLAB fir2 - npt and lap

This might not be the right place to ask this, but I'm hoping someone can explain two of the arguments in the MATLAB fir2 function. It is a function for designing ...
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3answers
320 views

Reverse engineering a digital filter

I have 2 simultaneous signals that both are designed to measure eye movements. They are sampled at 250 Hz. We have 12 subject recordings. For 3 stable periods in each subject, we choose 256 points and ...
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1answer
102 views

What is the threshold to prefer polyphase channelizer over FIR filtering?

A wideband signal need to be processed to extract several narrowband channels from it. There are two straightforward approaches to do so: For each narrowband channel shift frequency and filter it ...
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2answers
479 views

Numerically Stable IIR filter

I am making an IIR Filter and I am running into problems with stability with higher orders. The values go off to infinity as soon as it starts. So far, I've been using MATLAB to generate my filters ...
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1answer
181 views

Adaptive Band pass filter for extracting respiratory waveform from accelerometer data

I am recording data from an accelerometer attached to the chest (1000Hz). I need to extract the respiratory waveform. I tried an adaptive bandpass filter based on a dominant frequency in my signal ...
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1answer
3k views

How to design a digital filter in python that will run over an uC?

I am trying to implement a digital filter over a uC (it doesn't really matter which filter and which micro controller because I'm looking forward to learn how to do it in the future with different ...
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0answers
24 views

Converting the equalizer coefficients from VSA at 2 sample/Symbol to 1.5 sample/Symbol

I was looking for some feedback on conversion of equalizer/filter coefficients at different sampler rate. Problem Statement: We have built a transmitter system whose output is at 1.5 sample/symbol (...
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2answers
77 views

Where can I find information about how to implement IIR filters on micro-controllers?

I am trying to implement an IIR filter on a LPC1769 (I don't know if it's relevant). I already have the H(z) so all the design part is prety much solved. What I am looking for is information about: ...
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1answer
79 views

bibliography about implementation of IIR filters

I need to implement a 6th order IIR filter. The filter I need to implement is the A-weighting. I've simply used the bilinear function of Matlab with an fs of 20Khz. I need to implement this over the ...
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3answers
2k views

How to determine if an IIR filter is causal or non-causal?

It may sound impractical, as only causal system are passively realizable, but what I am facing is a general statement made regarding recursive digital filters: for them to be stable, the poles should ...
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0answers
75 views

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
100 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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2answers
227 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
66 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
171 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
139 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
1k 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
116 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.\...
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1answer
796 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
187 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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1answer
112 views

Overshoot when using Bessel filter

Here is simple code to illustrate overshoot using bessel filter: ...
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0answers
45 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
64 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
176 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
248 views

How do I initialize the state of a digital filter in Direct Form 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
141 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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2answers
729 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
299 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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0answers
574 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
427 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
161 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
77 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
352 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
48 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
89 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. ...
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1answer
83 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
114 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
189 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
433 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
227 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
223 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
205 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
501 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
61 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
90 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
694 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
471 views

What's Logic Behind the Construction 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
342 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?

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