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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Guess stability based on filter coefficients

is it possible to guess the stability of a filter with given coefficients a and b? I know that the right way would be to calculate the pole positions and check if they are in the unit circle but I ...
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17 views

Structuring Kalman filter for tracking problem where only position is known

I'm new to Kalman filters and my extensive web search about them has helped me understand the majority of it (or so I think). However I still need some light shed on my problem formulation. I have a ...
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1answer
14 views

How can I duplicate this type of image?

I'm very new to the idea of signal processing, could someone point me in the direction of of a program to toy with images, below is a link to 2 taken a split second apart that showed up on my Amazon ...
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1answer
28 views

IDFT of H(z) sampled in N values

If a have a causal IIR filter described by $H(z)$ and I sample it in $N$ equispaced values around the unit circle, I get a DFT of $N$ points. That DFT corresponds to $h[n]$ truncated in $n=N-1$ or to ...
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31 views

why do low pass and high pass filters generate a phase shift?

I just learned following notation for a sine wave: $A e^{j\phi t}$ A low passfilter and a highpass filter respectively generate a phase shift in the complex plane of the sine wave as follows: $A ...
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Digital filtering - Just set values to 0? [duplicate]

Let's say we want to digitally filter a signal to preserve only relevant frequencies. We calculate the frequency response $F(\omega)$ of the time series data $x(t)$ using the FFT. Imagine that we ...
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2answers
59 views

Filtering a digital signal online in real-time using python

I am currently trying to apply a bandpass filter to a signal in real-time. There are samples coming in with a constant sampling rate and i would like to calculate the corresponding bandpass filtered ...
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1answer
15 views

What is the difference between photon shot noise and salt and pepper noise?

I am currently working on an application which deals with the restoration of fluorescent beads examined using widefield microscopy. There are two major sources of noises that are being encountered in ...
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1answer
11 views

Filtering data on a known signal

I have data that is affected by temperature. How can I filter out the temperature signal considering I have temperature data collected?
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25 views

Applying Filter to Complex Data

I have data being streamed in the time domain, I would like to use a transfer function for a low pass filter in order to filter the data. I use DFT to get the data to the frequency domain, in the form ...
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1answer
49 views

How can resonance be added to a fractional order Butterworth low pass

I am interested in how to add resonance (Q) to the magnitude response of a Butterworth low pass when it is expressed in the form: $$ G^2(\omega)=\frac {1}{1+\left(\frac{\omega}{\omega_c}\right)^{2n}} ...
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Fractional Interpolation Polyphase

I explain my problem. I have to implement an oversampled filter bank. The first stage of the general architecture is serial to parallel converter with a decimation factor M < K channels. In my case ...
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18 views

Filter FM and sideband overload

I'm programming a synthesizer for musical purposes. DSP is not my specialty, so I'm hitting a kind of road block here. I'm using a typical 2 pole biquad digital filter. I'm modulating the cutoff at ...
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15 views

Recover original image values in unfiltered backprojection

When performing unfiltered back-projection, is there a way to have the reconstructed image values match the values from the original image? For example, if one executes the following example code from ...
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3answers
71 views

3dB and 6dB Per Octave Rolloff Digital Filters / Generating Pink and Brown Noise

So I would like to have a design method for filters with 3dB and 6dB per octave roll off -- for generating pink and brown noise respectively. I know the following 'pinking' filters exist: Filter to ...
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27 views
+50

Gaussian filter of 2d image with location dependant sigma in Python

I am trying to implement in Python a Gaussian filter in which the kernel is $$ G(x,y) = \frac{1}{2 \pi \sigma_0^2} e^{-\frac{(x-x')^2 + (y-y')^2}{2 (\sigma_0 f(x',y'))^2}},$$ where $f(x,y)$ is some ...
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1answer
30 views

Channel Estimation / Equalization - Estimate Channel Inverse Using White Noise Statistics Only

Given the system defined in the following figure: We have a system $ G \left( f \right) $ which is unknown yet can be defined by $ {N}_{p} $ poles and $ {N}_{z} $ zeros. The signal $ x \left( t ...
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1answer
80 views

Are there other filter models besides IIR and FIR for processing digital signals?

I am very familiar with IIR and FIR filters for processing digital signals (audio is my main interest). I am keen to know if there are alternative filters based on different mathematical models to IIR ...
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2answers
53 views

How do I find the phase response of a filter from the filter equation?

I am new to this forum. I have been teaching myself DSP for the last two years, so my experience and knowledge has a long way to go. I am trying to find the phase response of a specific filter that I ...
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28 views

Help DFT Filter Bank

I'm implementing maximally decimated FB on matlab and it's happening a strange thing. If i implement parallel filters, the output is near reconstructed, without aliasing in other channels, while if i ...
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1answer
31 views

Playing samples in a loop, without clicks or pops

I'm trying to build an audio player that has this specific feature: when you scroll to x sample time, [x to x + 4096] samples are played in a loop until you move the mouse to another place. The ...
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16 views

What are the figures of merit/ performance measurements I can use for deconvolution?

I'm currently performing deconvolution on a bunch of 2D slices of fluorescent beads using MATLAB. Now I want to measure the performance of different deconvolution methods on the filtering of the ...
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18 views

Problem with DFT Filter Banks and MATLAB

I'm posting the results of dft FB on matlab, and I would discuss about them. To generate the analysis and synthesis prototype i used a simple method for pseudo-QMF, published by Mitra in his ...
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1answer
38 views

Is there a set formula for calculating FIR coefficients (for a LPF ir 96 taps)?

I need to calculate a series of coefficients for several FIR filters having 96 taps is there a standard or set formula for calculating these, it is a low pass filter.
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1answer
55 views

why type 2 FIR cannot be used to design high pass filter

Kindly explain why type 2 FIR filter cannot be used to design high pass filter.
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1answer
49 views

How Matched filter is transformed due to frequency modulation?

Intro: Consider a signal source S which emits a PAM modulated signal, bits encoded as +g(t) and ...
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1answer
46 views

Determining type of filter given its equation

Given a general filter equation, how can one tell the type of filter that the same equation represents? Meaning how can I know if the filter is Low/High/Band Pass etc...? For example, the following ...
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26 views

Smoothing the discrete acceleration

In order to know if my signal is increasing or decreasing, I'm using the discrete derivative $y[n] = x[n] - x[n-1]$ or a smoothed version of it (for example Exponential Weight Moving Average of $y[n]$ ...
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55 views

How to apply a signal to FIR Filter in C/C++?

When I using code to compute FIR Coefficient and Window function, then I don't how to apply this generated filter to a audio data(PCM) signal. What is the procedure of creating a FIR filter and the ...
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34 views

Turn a decision problem into a linear filter problem

Let $x[n]$ be a time-series, and two filters: $A[n] = a_0 x[n] + a_1 x[n-1] + ... a_q x[n-q]$ $B[n] = b_0 x[n] + b_1 x[n-1] + ... b_r x[n-r]$ I think the answer is No, but is there a way to have ...
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2answers
81 views

Minimum Sampling Rate of Bandpass signal

Consider a DSBSC signal s(t) = m(t).cos(2.pi.20k.t) where say m(t) is multi-tone signal with frequencies less than 5 kHz. The ...
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3answers
88 views

Kalman filter : simple code example

I read lots of things about Kalman filtering, but in order to fully understand it, I would probably need to see it working on some data. Would you have a minimal example (Python code or any other ...
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1answer
62 views

Why Butterworth filter always starts my signal from from zero mark (amplitude)?

I am trying to filter my force signal using Butterworth low pass filter. so this is the signal, I have to filter (noise frequency is around 33 Hz). So when I use Butterworth filter with following ...
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14 views

SISO model? white noise excitation

I'm now trying to do this problem. I have data set including input and output for both 600 gal and 800 gal, and 3 white noise excitations. But i don't understand the model of this problem. 3 White ...
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27 views

Regarding maximizing signal to noise ratio

If I have 3 recordings, each of speech (recorded in a relatively silent room), speech+background noise and background noise then is there a way to utilise the data from all the three to extract clear ...
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1answer
22 views

How to merge transitions?

I need to represent my signal as a set of transitions from one level to another, and I am interested in transitions that have relatively big amplitude 2000+. I applied median filter to get rid of the ...
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2answers
47 views

Complex band pass filter

I'm trying to find some mathematical docs about the complex (analytic signal) band pass filtering in discrete time. I've read some text already but most of them describes the problem in real time ...
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0answers
25 views

Empirical Distribution White Noise Tests

All common tests for testing if a time series is white noise are based on common distribution assumptions from which inference is made that a process is white noise. However, in the case for most ...
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2answers
30 views

Regarding BIQUAD IIR implementation in MATLAB

I want to implement a high pass filter (order 4) in MATLAB using cascade form. ...
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1answer
39 views

Oversampled polyphase filter banks

i would a confirm of this: with polyphase structure is possible only design a filter bank with a INTEGER oversampling ratio? For non-integer i've seen the weighted overlap-add metod, is right? thanks ...
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1answer
42 views

Choosing low-pass filter parameters

I have position of objects from recording sampled at 30 fps. I wish to remove high frequencies from position with the help of low-pass filter. I am confused as to how to choose the filter parameters ...
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1answer
74 views

Is R suitable for digital signal processing

While asking a question about representing large time series in R I was discouraged from using R for digital signal processing. I understand that R is geared towards statistics. However, a signal is ...
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1answer
64 views

DFT Filter Banks

i have a doubt about DFT filter banks. For example I have $M=4$ channels and a filter with cut-off frequency of $0.25$ right. The prototype filter $H_0$ is a low-pass centered around $0$ ok. The ...
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1answer
45 views

Performance of adaptive filter

I have designed an adaptive filter for noise cancellation. Is there any standard way of testing adaptive filters?
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1answer
44 views

what is happening when i run this code line? [closed]

I have the following MATLAB code: x=[1 2 3; 4 5 6; 7 8 9;]; u=(0:255)/(255-1).^1.05; x2=u(x+1) when I run this code I get the following as a result in ...
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5answers
123 views

What kind of filter would have a sawtooth frequency response?

I want a complex FIR or IIR filter whose response is approximately a sawtooth in the frequency domain. How should this be formally described, and what formula or algorithm can I use to calculate taps ...
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1answer
64 views

Zero-Phase Algorithm in Matlab

https://ccrma.stanford.edu/~jos/fp/Forward_Backward_Filtering.html from this document it's says 1.Filter the Data 2.Flip that 3.Filter again 4.and Flip that again ...
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2answers
88 views

Are all IIR filters unstable in nature?

I know this question has been asked here numerous times. But I am still unclear here :( My book says, A linear time-invariant system is stable if its impulse response is absolutely summable. ...
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21 views

Finding Farrow structure coefficients [duplicate]

I want to design a high order FIR filter using farrow structure, where the poly phase components of the desired filter are expressed using Farrow structure. How to find the Farrow coefficients using ...
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2answers
32 views

signal sampling for 2nd time of sampled signal

A sinusoidal signal of 600hz is sampled with 1khz, If the sampled signal is applied to an ideal low pass filter of 500 hz, what is the output?