Stack Exchange Network

Stack Exchange network consists of 175 Q&A communities including Stack Overflow, the largest, most trusted online community for developers to learn, share their knowledge, and build their careers.

Visit Stack Exchange

Questions tagged [digital-filters]

The tag has no usage guidance.

1
vote
0answers
27 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: ...
1
vote
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, ...
0
votes
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 ...
0
votes
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 ...
0
votes
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 ...
0
votes
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 ...
0
votes
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 ...
0
votes
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 ...
0
votes
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
votes
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[...
1
vote
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., $$H(s) = \frac{2}{s(s+2)}$$ ...
0
votes
0answers
17 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 ...
0
votes
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 ...
0
votes
1answer
27 views

Overshoot when using Bessel filter

Here is simple code to illustrate overshoot using bessel filter: ...
0
votes
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, $$O(t) = 1 - e^{\frac{-t}{\tau}},$$ ...
1
vote
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 ...
1
vote
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 ...
1
vote
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 ...
1
vote
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 $n<0$?
0
votes
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 $\mathbf{A}\sigma^2$. ...
1
vote
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. ...
0
votes
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 ...
0
votes
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 ...
0
votes
0answers
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.
6
votes
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 ...
1
vote
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 ...
0
votes
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. ...
0
votes
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.
0
votes
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 ...
3
votes
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
votes
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 ...
0
votes
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 ...
1
vote
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 ...
0
votes
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 $(A^c = [a^c_m, ..., a^c_1, a^c_0])$. ...
1
vote
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 ...
0
votes
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 ...
1
vote
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?
2
votes
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 ...
-1
votes
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 ...
1
vote
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 ...
2
votes
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 ...
3
votes
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 ...
0
votes
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
votes
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 "...
1
vote
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 ...
2
votes
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 ...
1
vote
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-...
2
votes
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 ...
-1
votes
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 ...
1
vote
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 $T$. ...