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Questions tagged [digital-filters]

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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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15 views

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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19 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
23 views

Overshoot when using Bessel filter

Here is simple code to illustrate overshoot using bessel filter: ...
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23 views

Accepted delay in digital receiver?

At the receiver end, we will have a delay because of DSP such as delay of the digital filter, equalizer, decoder..., My question is what is a reliable delay so that we can have synchronization with ...
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14 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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39 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
10 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
27 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
31 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
29 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
64 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
43 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
29 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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36 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
291 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
33 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
58 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
60 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
24 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
54 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
21 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
36 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
113 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
99 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
32 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
39 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
47 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
59 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
47 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
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 ...
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1answer
319 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
126 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
66 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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1answer
117 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
111 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
235 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
139 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
126 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
49 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
44 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 $T$. ...
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1answer
68 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 ...
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43 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 ...
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1answer
67 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 : ...
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1answer
45 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.
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2answers
71 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 ...
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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: ...
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1answer
267 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:...
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95 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 ...
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1answer
412 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 ...