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1answer
55 views

Fir filter confusion

I am learning about fir filter and i'm confused. I tried googling about it and it becamse worse. I am trying to find out different types of fir filter. 1. Is direct form and n-tap fir filter the same? ...
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4answers
109 views

Is there an algorithm for second order filtering of data where the filter frequency is not known a priori?

It is well known that a moving average algorithm done in the time domain is equivalent to a filter with frequency response $\mathrm{sinc}(\omega\tau)$ where $\tau$ is the averaging time. (see this ...
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2answers
60 views

filter and resample or resample and smooth?

Currently doing some signal analysis in python for a major project in my physics degree which is due really soon. I need some help! Say I have two signals, f(t) and g(t) which are recorded over the ...
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1answer
27 views

diffusion filter

I need to know what is iteration and divergence in anisotropic diffusion filter technique. here 't' refers to iteration what is this iteration.how it is related with filtering. I know that iteration ...
1
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1answer
66 views

convolution of signal with wavelet

I know that the complex wavelet transform is given by $$\int_{-\infty}^{\infty}f(x)\psi^*_{s,u}(x)\,dx$$ where $f$ is the signal, $\psi_{s,u}(x)=\frac{1}{\sqrt{s}}\psi(\frac{x-u}{s})$ is the wavelet. ...
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3answers
286 views

Difference between linear and non linear filter

How a mean filter is called as linear filter and a median filter is called as non linear filter? I understand how a mean and median filter operates, but I was not able to relate with the term linear ...
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2answers
79 views

Apply Singular Value Decomposition on Gabor filters

According to my question in this link, I applied (with a big help from lennon310) the separability of Gabor filters of any orientation. The only disadvantage of this method is the convolution in the ...
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0answers
47 views

Relationship between fractional delay coefficients and frequency?

Some background: I am putting a sig-gen signal through an ADC, into an FPGA where I can modify it, out a DAC and up to a SpecAn. All that works fine. I am experimenting with fractional delay ...
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1answer
92 views

Filtering an image with two gabor filters

I have an image of size 160x160 of pixels. I am trying to convolve this image with two Gabor filters of different scales (7 and 9 respectively) but with the same orientation. I red in an article the ...
2
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1answer
166 views

Design a comb notch filter when sampling rate is not an integer multiplicative of the cutoff frequency

Let's say I have a signal sampled at 976 Hz and I want to remove network interference noise, i.e. 50 Hz and all the harmonics. Is it possible theoretically to design a comb filter to do that? I have ...
2
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2answers
156 views

Detecting & Removing steps in signal

Given the signal shown below, what is the best way to remove the steps and local maximas it contains. The signal contains some steps which can last up to 100 Samples before they return to about the ...
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2answers
264 views

interpret wavelet scalogram

My knowledge of wavelets is less than epsilon. Bear with me. If I have a signal of two well separated sinusoids (15 and 48 Hz) plus some random noise, I can clearly make out the two in a spectrogram ...
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1answer
411 views

What exactly does ordfilt2 do?

Can you please explain in simple terms what do the input parameters indicate in the ordfilt2 function in matlab? ...
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1answer
69 views

What are filter banks

What are filter banks and what's their application? What's the meaning of $H(z)$ in descriptions of filter banks?
0
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1answer
79 views

Effect of bandwidth reduction in bit growth

I just want to ask, whenever you reduce a bandwidth of signal by filtering, I wonder why do you need to improve the number of bits used to represent that signal. The number of bit growth is ...
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0answers
68 views

Can somebody help me solve this signal accumulation problem?

Investigation of the method of suppression of random noise by coherent signal accumulation Purpose - to identify opportunities for the coherent accumulation for cases of stationary and ...
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0answers
37 views

Extract data that contributes to strong correlation

I need to discover the effect of hidden variables on on a predictor I'm working on. I have two random variables, which have around 50% correlation to each other, if all the data is counted for. The ...
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2answers
340 views

Filtering in frequency and time domain

I am trying to filter a sinusoid ($f_{sample}=50 Hz$) ,, i need to remove all frequencies above 10 Hz with accepted transition region 2 Hz (stop band freq=12 Hz) the question is why after filtering ...
2
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1answer
83 views

Where does homomorphic filtering stand in regards to DSP applications?

I am studying my Oppenheim and Shaffer book, (New Edition), and the last chapter deals with something called homomorphic filtering. I have read the wiki and some other websites about it, but they do ...
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1answer
89 views

Constraint on a filter

First of all thanks for your patience: it is the first time for me posting a question in this forum. I am not a DSP expert, but I should get by if you give me in depth explanation. This is an example ...
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2answers
1k views

How are two signals multiplied? And how is it different from convolving two signals?

suppose that i have two signals x[n] = {2,4,1} and p[n] = {5,1,8} and i want to multiply them. How do you do that? How different it is from convolving two signals? I understand that multiplication in ...
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0answers
116 views

FFT of irregular shaped data

I have an irregular shaped 3D image that I want to do frequency domain filtering on. Currently I simply zero-pad the data to make it a cube, then do an fft. I was wondering if this was a valid ...
3
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2answers
616 views

Is interpolation (interp1) better than FIR filtering when rational integers are close to 1?

Question I've been attempting to resample a GPS signal in MATLAB. I've built a few FIR filters using fvatool and from handmade transfer functions (punched out with ...
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3answers
1k views

Can we have a Digital Anti Aliasing filter?

I am working on a board that has no antialisaing filter at the input of the ADC. I have option to I implement my own filter using RC + Opamp circuit. But is it also possible to implement Anti ...
2
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1answer
214 views

Is it not possible to perform non-linear filtering in frequency domain?

This link contains the following statement: In contrast to the frequency domain, it is possible to implement non-linear filters in the spatial domain. In this case, the summations in the ...
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3answers
202 views

Other techniques like overlap save/overlap add

I know overlap save and overlap add are used for long data sequence filtering. Are there any other similar or better techniques like these?
1
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1answer
110 views

How to apply convolution on a buffer

I'm trying to do some simple filtering for an audio signal using a window-sinc low pass filter. Supposing that my buffer has M values and the filter kernel size is N, after convolving these two arrays ...
3
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1answer
206 views

Frequency response of Low pass filter and High pass filter

The transfer function of a Low pass filter is H(w). From this I want to develop a high Pass filter. I read in DSP by Proakis, Sec 4.5, that the high pass filter can be obtained by translating H(w) by ...
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2answers
190 views

Strange noises when filtering audio signal

I am using Naudio open source library and I am trying to do some simple filtering. The problem is that I hear some "clicks", not too loud. The library offers me the possibility to use at least two ...
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2answers
681 views

How do convolutions become matrix multiplications?

I know this question may not be very relevant to programming, but if I don't understand the theory behind image processing I'll never be able to implement something in practice. If I got it right ...
4
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1answer
282 views

Approximating the frequency response of a median filter

A median filter is a non-linear and lossy process, so it doesn't have a closed form frequency response as would a FIR filter (say a box filter of the same length) in an LTI system. But how closely ...
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0answers
40 views

STFT overlap add method - What am i doing wrong?

i am trying to implement a short time fourier transform of an audio signal to do some filtering (subtract another stft signal using spectral subtraction). I am using a hamming window function with 50% ...