In signal processing, a filter is a device or process that transforms a signal by selectively choosing specific frequencies and leaving others with appropriate gain or as it is.

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This is result after bilateral filter then Harris corner detector then watershed segmentation

Now I have gotten two segments as shown.Now i want to detect white markers that have Harris in segments given by watershed. How should I proceed?
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Designing bandpass FIR filter MATLAB

I am trying to design a FIR bandpass filter for a STM32F407 microcontroller with a passband of 1kHz between 59500 Hz and 60500 Hz or something similar (since I don't really know how tight I can ...
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Does the S-transform have phase linearity?

S-transform is said to be a time-frequency distribution and also it is linear with frequency. Is it linear with phase also? If not, how can I find the phase relation between two signals?
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Derivation of Bessel filters

I was reading Winder's Analog and Digital Filter Design and the section on Bessel filter. I was hoping to see a complete derivation of the Bessel filter theory, but Winder's book gives only \begin{...
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4answers
105 views

Why is a linear phase important?

If symmetry conditions are met, FIR filters have a linear phase. This is not true for IIR filters. However, for what applications is it bad to apply filters that do not have this property and what ...
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29 views

How to construct a preemptive digital filter to neutralize an unwanted filter

I'm trying to think of how to construct a filter based on the following scenario. I have some time trace with a certain known power spectral density (I can verify this with a periodogram of the time ...
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54 views

Bandpass general equation to difference equation

I know this is a very basic question and I am coming out from a quarter of DSP. I want to create a function in Java which can taken in two parameters, either ...
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Analytically Deriving butterworth Filter [closed]

I am pretty new to Digital Signal processing, I did my back end research, but I was overwhelmed by the variety of the solutions I got. I want to Analytically derive a Butterworth filter with only "...
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2answers
68 views

Remove noise from linear signal

This signal represents the distance in an experimental device. As the speed is constant, the signal should be linear (untill it drops). However, for some reason the device showed interferences as ...
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1answer
63 views

Decomposition of $H(z)$ as maximum-phase, minimum-phase

The frequency response is: $$H(z) = 2-7z^{-1}+7z^{-2}-2z^{-3}$$ I see that it has $3$ zeros: $z_{01} = \frac 12$, $z_{02} = 2$, and $z_{03} = 1$; and $3$ poles in: $$z_x = 0$$ Now, I have to write ...
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1answer
44 views

Confusion about adjoint of convolution

I have the following problem with convolution operators. I know that if I have a filter $g$, then the adjoint filter would be given by $h(x,y)=g(-x,-y)$, so that $\langle g * u,v \rangle = \langle u,...
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38 views

Maximum phase with the same magnitude response

I'm trying to solve an exercise in a general way and I can't find if my answer is correct. Given this frequency response: $$H(z) = 4 + 2\sqrt2 z^{-1} + z^{-2}$$ I need to find a frequency response ...
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18 views

Difference between Envelope filters and a unit with variable envelope response

Would a unit with variable envelope response shape a square sound wave into a sine wave? I know that an envelope filter kind of does that through subtractive synthesis. I'm trying to find a way to ...
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2answers
86 views

Inter-filter bit width

I'm designing a $2$-stage filter and was simulating the effect of quantization to a certain number of bits at the filter output. The result I got was far from expected - and would appreciate any ...
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822 views

Can you turn a square wave into a sine wave using a low pass filter?

And if it could, would it make the sound of the square wave thinner than before because of losing its harmonics?
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1answer
28 views

Smoothing complex data by convolution

I need to smooth noisy complex data with a Gaussian filter. Right now, I apply the filter to real and imaginary part of the data separately, which needs two convolutions. The intended results are the ...
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25 views

4-band FIR filter equivalency with 2-band filter compositions?

On pg. 18 of this paper: On the Dual-Tree Complex Wavelet Packet and M-Band Transforms there is a discussion about creating a single 4-band filter bank from a 2-level, 2-band set. There are two ...
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2answers
59 views

What is the difference between convolution filter, low-pass filter, and median filter?

Isn't blur filters, like median filter, a type of convolution filter? How does a low-pass filter relate to them? Why does this guy differentiate convolution filters here from blur filters here?
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81 views

Image processing - Is mean filter shift-invariant?

Update: So, according to this article. For a 2D mean filter, $$ H[f(x,y)]=\frac{1}{NM}\sum_{k=0}^{N-1}\sum_{p=0}^{M-1}f(x-k,y-p)=g(x,y)\tag{1} $$ $$ g(x+x_0,y+y_0)=\frac{1}{NM}\sum_{k=0}^{N-1}\sum_{p=...
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1answer
40 views

System identification/ Filter estimation to mimic frequency equalizer of audio with Scipy

At the current problem I'm working on, I have two signals: One "original" signal that contains audio (voice). The second signal is the same audio file but edited with a frequency equalizer, for ...
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1answer
70 views

Sampling signal after root raised cosine

I am generating series of 1 and 0. then pulse shaped them with root raised cosine and after than deciding to do match filtering and recovering the bits however it turns out that I am not getting the ...
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1answer
40 views

Multiple biquads vs higher order filtering

I've heard that you can stack biquads on top of each other to get the same (or similar) effect as higher order filters. For instance a biquads is order 2, so putting something through 3 biquads is ...
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42 views

Synthetic sounds to describe motion in 3d or 100d

Watch a butterfly flitting about, or an optimizer chugging along in 3d or 100d -- a sequence of points $X_1 \ X_2 \ X_3\ \ldots$ How could one generate synthetic sounds that convey moving fast or ...
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What does a Sobel Norm of an image represent?

Why would we want to find the Sobel Norm of an image?
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3answers
53 views

LMS filter weight update

I have a general question regarding Least mean squares adaptive filters. Using the example of noise cancellation, I understand that if you have a set of reference signals (S) and corrupted signals (S+...
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70 views

detect what is noise in audio signal

I'm trying to filter noise from an audio signal. The signal was and is going to be recorded in a city environment which always contains some white/pink noise. At this moment I'm using two options to ...
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1answer
51 views

Butterworth band pass filter

I am trying to build a Butterworth bandpass filter. My intent is to have a pass band between $250\textrm{ Hz}$ and $1000\textrm{ Hz}$. However the plot doesn't look right, as can be seen from the ...
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46 views

The little problem with Kalman filter

I am trying to simulate the Kalman Filter. I have the covariance matrix P_{0|0}. Tell me please, how can I get the predicted (a priori) estimate covariance matrix ...
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48 views

Difference equation with complex zero

Let's assume I have the following transfer function: $$ H(z)=\frac{z-\left(\frac{1}{\sqrt{2}}+i \cdot \frac{1}{\sqrt{2}}\right)}{z} $$ It looks like a first order highpass-filter with a complex zero ...
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1answer
74 views

Compensating Loudspeaker frequency response in an audio signal

I have been working on a project in which I was required to work on the audio signals recorded from the loudspeaker kept in front of a filter. So, to simply explain it: $$\boxed{\rm LoudSpeaker} \...
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2answers
34 views

Why do we do oversampling in raised cosine?

I would like to do know why do we need to do oversampling in a raise cosine filter. There is already a question like this but the answer is very confusing and not sufficient. I need to know what ...
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1answer
22 views

How to filter a signal knowing the frequency response of a system in MATLAB?

I have a signal of period $T_0=8$, let's say $x(t)$, and it has the following Fourier coefficients: $$ a_k=\frac{1}{4} \mathrm{sinc}^2\left(\frac{3k}{8}\right) e^{ik\frac{\pi}{2}} $$ for $1\le|k|\le6 ...
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Remove background noise from recording

I have a sample speech recording with background noise. I managed to extract part of the noise signal, and put the noise and the contaminated signal in the same plot. I have done fft-analysis on ...
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61 views

Why is $\pi$ in the domain of filters?

I'm just beginning to learn (FIR) filter design and I was wondering, why is $\pi$ involved in the domain of such filters? I assume it's related to some trigonometric functions, but why do I see e.g. ...
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57 views

How to remove wind noise from video file

I've recorded video footage from a recent motorcycle trip using a helmet mounted Go Pro Hero 4. The video is good, however wind noise is drowning out the sound from the engine. Is it possible to ...
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1answer
26 views

Difference between demodulation and band-stop Filter

I was wondering what is the difference between demodulating a signal that is carried with a frequency $F_{\rm carrier}$ and applying a band-stop on a band around $F_{\rm carrier}$ ?
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156 views

Phase invariant detection of low frequency spikes

I am a physics graduate student studying a new physical device via simulations. Thanks to SE DSP, reading Lyon's Understanding DSP as suggested (not all the way through it yet though), and a bunch of ...
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26 views

How many types of Bandpass filters are there for image processing?

This article shows that: Ideal Bandpass filter Butterworth Bandpass filter Gaussian Bandpass filter Is that classification correct? Are there any other types?
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What are various types of Gabor Filters?

Gabor Filter has 4 types of kernels: Real Imaginary Complex Magnitude Squared Magnitude What do they actually mean? NB. Please, supply me any resource for further study.
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29 views

Artifacts in sub-sample delay

There are at least two ways of achieving sub-sample delay. One is to simply over-sample the signal and then delay it by the wanted sub-sample amount. The over-sampling is equivalent to creating new ...
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What is the Center Frequency of Gabor Filter?

If the standard form of a Gabor function is as follows, $$ g_{\lambda, \theta,\varphi, \sigma,\gamma}(x, y)=\exp\left(-\frac{x'^2+\gamma^2y'^2}{2\sigma^2}\right)\cos\left(2\pi\frac{x'}{\lambda}+\...
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What is the multi-resolution analysis form of Gabor filter?

The multi-resolution analysis form of Gabor filter is as follows: \begin{align} \psi\left(x,y;f_0,\theta\right)&=\frac{f_0^2}{\pi\gamma\eta}\exp\left[-\left(\frac{f_0^2}{\gamma^2}x'^2+\frac{f_0^2}...
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steady-state response

Why does the response of this system evaluated in $0.25$ such that $$H_p(0.25)=\frac{1}{1-0.5e^{-j2\pi /4}}=\frac{1}{1+j0.5}$$ equal $0.89943\angle-26.6º$?
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When should we use the full Gabor Function and when do we use only the real part?

In this YouTube video, we see that the guy is only using the real part of the Gabor Filter to implement his project. My questions are, When should we use the complex Gabor Function? When should we ...
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Deriving the multi-resolution analysis form of Gabor filter from the standard form

The standard form of a 2D Gabor filter is as follows: \begin{align} g_{\lambda, \theta,\varphi, \sigma,\gamma}(x, y)&=\exp\left(-\frac{x^2+\gamma^2y^2}{2\sigma^2}\right)\cos\left(2\pi\frac{x'}{\...
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Response of digital filter is represented given its difference equation

A digital filter is described by the difference equation: $$y[n]-0.1y[n-1]=x[n].$$ Determine the response $y[n]$ in steady state at the entrance $$x[n]=5 \cos⁡(0.2nπ+45°)+2 \cos⁡(0.8nπ+30°)$$ I ...
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What criteria should be used for smoothing

Actually I have some measurement which I want to get rid of noise I want to use different filter techniques but I am wondering what criteria I should use to check if I am removing noise or not or if ...
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53 views

O-QPSK constellation interpretation - GNU Radio

I am currently trying to recover the I&Q of a O-QPSK signal generate from the standard XBEE arduino module using a B200. I can get somehow get a "look-like" of a constellation. Here an screenshoot:...
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amplitude at exact frequency in wide band signal

Could anyone suggest the most computationaly efficient method for finding amplitude of exact frequency having a noisy wide band signal. To be more specific about a task. I have some physical ...
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1answer
28 views

Conditions for expressing a system as difference equation

I have read when studying this subject that a system is LTI and causal if and only if it can be expressed as a difference equation (if it is in continuos time, as a differential one). I don't know if ...