Questions tagged [filtering]

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14
votes
3answers
31k views

What is the difference between linear and non-linear filters?

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 ...
11
votes
3answers
9k 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 ...
10
votes
1answer
2k 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 ...
8
votes
3answers
15k 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 ...
7
votes
1answer
701 views

Adaptive filtering: Optimum filter length and delay

I'm trying to find the optimum filter length for an Adaptive Filtering, using RLS Algorithm. I'm using this design: So the "error" signal is the signal without noise (and that's the signal that I ...
6
votes
2answers
6k 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 (...
5
votes
1answer
11k views

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

Suppose that I have two signals $x[n] = \left\{2,4,1\right\}$ and $p[n] = \left\{5,1,8\right\}$ and I want to multiply them. How do you do that? How different is it from convolving two signals? I ...
5
votes
1answer
3k views

Removing transients in highpass filtering with MATLAB

I've made a simple first order IIR highpass filter with a zero at z = 1 and a pole at z = 0.9. Its frequency response looks like this: Now, I filter a DC signal using this filter. Here's the MATLAB ...
5
votes
1answer
310 views

What type of low-pass filter to use for gyroscope filtering in control application?

I am using a gyroscope to generate a feedback signal that drives a motor to resist rotation, just as in a camera gimbal. The gyroscope is sampling at 32 KHz with a bandwidth of 8 KHz, and my control ...
5
votes
1answer
270 views

Is there a $z$-transform like for variable sampling rate signals?

I'm working with signals with variable sampling rate (the time space between samples is not constant). I know the delay between samples but I don't wont to interpolate the signal. Is there a way to ...
5
votes
2answers
624 views

Filtering Overlapping Frequencies in sound file

I have been given a sound file of a plane passing over a rain forest filled with birds. I am supposed to filter out the sound of the plane as it flies over. I've accomplished this with various types ...
4
votes
2answers
101 views

Filtering without using any transforms

A composite signal is defined as $$x(t) = \sin(2\pi 10t) + \sin(2\pi 40t) + \sin(2\pi 60t) + \sin(2\pi 50t)$$ Separate the four frequencies without using any of the transform techniques.
4
votes
1answer
5k views

Simple software low pass filter

I am measuring voltages and currents of a 3-phase electrical machine and I need to calculate the power. Every interrupt (frequency between 30-70kHz) I get values (voltages, currents) from analogue-...
4
votes
1answer
556 views

Polyphase Filter decomposition. It is not working

So, I want to do a polyphase implementation of a filter bank (I had problems with it since the beginning and I already asked for help to the community in the past, indeed: Norm MPEG-1 Layer III (Mp3) ...
4
votes
3answers
1k views

Convolution of Signal with a 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. ...
4
votes
2answers
613 views

How to design a filter with a certain magnitude response

I am trying to design a filter whose magnitude is the same as that of a given signal. The given signal is wind turbine noise, so it has significant low-frequency content. After designing the filter, I ...
4
votes
1answer
1k 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% ...
3
votes
2answers
3k 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 ...
3
votes
4answers
956 views

Does a filter add oscillations to a signal?

this is a more general version of a question which i asked previously. From what I understand, the purpose of a filter is to change the amplitude of a specific frequency band. As I went through the ...
3
votes
2answers
3k 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 ...
3
votes
1answer
1k 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 ...
3
votes
4answers
653 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 ...
3
votes
1answer
6k 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? ...
3
votes
1answer
515 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 ...
3
votes
1answer
2k 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 ...
3
votes
1answer
5k views

Apply butterworth filter (lowpass) to a signal

I'm working on a DSP lab and I can't figure out how to apply the filter I've created to a sound. The only thing I found online that did something to my sound was ...
3
votes
4answers
799 views

Equalizing $1/f$ noise

I am recently acquiring data through ADC on signals ranging from $0.1\textrm{ Hz}$ to $50\textrm{ kHz}$ ($F_s = 128\textrm{ kHz}$). I noticed that apart from the white noise, I have something similar ...
3
votes
1answer
788 views

Optimal windowing strategy for filtering live signal

I have a stream of 250 samples per second arriving at a pipeline stage which looks something like this simplified example. ...
3
votes
1answer
218 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 ...
3
votes
1answer
423 views

Wiener filter to estimate constant signal

I want to compare the performance of a Wiener Filter and the Kalman filer to estimate the value of a constant $d$ using mesurements corrupted by a white noise. That is, my measurements are of the form ...
3
votes
4answers
540 views

Removing Bounded Noise from an Audio Signal

I'm currently trying to remove some noise from a "contaminated" audio file. From what's given, I know that the noise is isolated to a specific frequency band - that is, from ~1400 Hz - ~1800 Hz. The ...
3
votes
4answers
7k views

difference between mean filter and order statistic filter

I am working on my paper about comparing between mean filter and ordering statistics filter, The mean filter is contraharmonic mean filter and the ordering statistic filter is alphatrimmed mean filter....
3
votes
1answer
232 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 ...
3
votes
1answer
108 views

Use samples of Fourier transform as DFT?

Consider the LTI system given by: $H(z) = 1 - \frac{1}{2}z^{-1}+\frac{3}{4}z^{-2}$ Let $x[n] = (\frac{1}{2})^nu[n]$ be the input to the system. We want to find the output for $n = 0,1,...,N_a$, using ...
2
votes
2answers
2k views

Can the order of filtering and downsampling be exchanged?

Say for example, I have a signal with maximum frequency at 600Hz, and I originally oversampled it at 2000Hz. Afterwards I was only interested in frequency less than 500Hz. I guess I could first filter ...
2
votes
3answers
1k 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?
2
votes
1answer
134 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?
2
votes
3answers
90 views

Causal unstable system turn into stable anticausal?

I would appreciate it very much if someone would be able to provide some clarity, help or comment on this problem. I have been reading several papers on time series identification such as https://www....
2
votes
3answers
346 views

How to Classify Image Filter as Low Pass Filter (LPF) or High Pass Filter (HPF)?

I am currently reading on the topics of image filtering and convolution kernels. While the process of convolution is clear to me, I do not understand, how one arrives at the statement, that a certain ...
2
votes
2answers
1k views

Extended Kalman Filter (EKF) for Non Linear (Coordinate Conversion - Polar to Cartesian) Measurements and Linear Predictions

I'm new to Kalman filtering and state estimation and I'd like some guidance on EKFs. Currently, I'm trying to use a linear prediction model coupled with nonlinear measurements to estimate the state ...
2
votes
2answers
64 views

Recommendation for studying filters

I have a project to do about what happens to a periodic function when we pass it through a low-pass, high-pass and band-pass filter. I have no expressions for the filters or the function, I just have ...
2
votes
3answers
145 views

How to filter key clicks?

I am the author of an amateur radio application that produces a waterfall display across a range of frequencies. Time is on the $x$-axis, frequency is on the $y$-axis, and the relative strength of ...
2
votes
2answers
135 views

Equalizing frequency response in software

I'm playing audio through a speaker which I know to have a non-flat frequency response, with particularly poor response in the low ($<1\textrm{ kHz}$) ranges. My plan to get around this problem is ...
2
votes
2answers
940 views

Smoothing a staircase

I have some data from a position encoder, so naturally i want to estimate its speed. However, the data is very quantized, so it's difficult to smooth enough to differentiate easily: Each step level ...
2
votes
2answers
3k views

Root mean square bandwidth of a signal with flat spectrum

I am trying to derive a simple expression for root-mean-square bandwidth $B_{\rm rms}$ for a signal that has flat spectral density over its entire bandwidth $B$. The expression for the $B_{\rm rms}$ ...
2
votes
2answers
344 views

Filtering Frequency Bands Out of a Signal

Let's say I have a 2 second data set taken at 220Hz sample rate and I would like to filter out the frequency bands associated with the EEG Spectrum: $$\begin{align} \Delta:& [1,3]\text{ Hz}\\ \...
2
votes
1answer
2k views

Derive velocity and acceleration from position

I generate random spike data that represent the ouput from a rotary encoder. Here the output of my algorithm: The first plot is position, then velocity and then acceleration. Even though I downsample ...
2
votes
1answer
110 views

Standard deviation is varying with change in amplitude of DC signal

I am trying to make sense of the below code. I am taking a large sample so that I get good estimates for standard deviation. ...
2
votes
3answers
610 views

How do I eliminate line noise from USB microphone

I'm trying to record voice from a microphone into my laptop. Unfortunately, the laptop does not have a microphone jack, so I have to use the USB input. This doesn't present a problem when I record ...
2
votes
1answer
2k views

Interpolating/Decimating CIC Filter Group Delay

I have an Interpolating Cascaded Integrator Comb (CIC) filter designed as so: Rate Change = 100 Number of Stages = 4 Differential Delay = 1 I'd like to calculate ...