Questions tagged [filters]

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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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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3answers
6k views

1/3 Octave Analysis

Good Morning! I am new in signal processing and I am trying to do a work in noise control of an electronic steering lock device (ESL). My aim is to calculate the loudness (Zwicker Method- ISO 532 B) ...
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1answer
447 views

Wavelet filter coefficients from the Morlet wavelet

How can I obtain the wavelet filter coefficients ( i.e. lowpass decomposition filter, highpass decomposition filter, lowpass reconstruction filter, highpass reconstruction ...
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78 views

Efficient implementation of spike removal

I have implemented the moving median absolute deviation (moving MAD) and it seems like bit-exact to Matlab's implementation. Nevertheless, I am sure that it is not efficient. The usual median filter ...
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27 views

Inconsistant Variation in BLE Beacon RSSI Values for Distance Measurement

I am developing an application to estimate the distance to a BLE Beacon using its RSSi values measured from a Mobile Phone. But when I started to collect data I could see that they varied so much that ...
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1answer
474 views

How to reduce synthesis artifacts produced by phase vocoder?

I implemented the phase vocoder algorithm in Python to time-stretch speech signals by following this paper and referring to this MATLAB tutorial. I can distinguish words in the original signal from ...
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135 views

How to implement the RLS for matrices

I need to implement the RLS algorithm but it's for matrices instead for vectors, I have made the below code, but still something wrong is not working well, EDIT: The code should be done as below, ...
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0answers
197 views

What is the meaning of a inner system/transfer function?

In an academic paper I'm currently reading [1], the notion of an inner system is central. It is defined as follows A (linear) system $G$ (with $m$ inputs and $p$ outputs) is called inner if it ...
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Pre-emphasizing in speech recognition

Usually, in speech recognition, the techniques that are used are based on the linear prediction model (Fant, 1960). Parallel to this, the human speech production mechanism causes energy to drop ...
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206 views

Given an output from a filter how do you find the input?

I have the output from a narrow bandpass filter. I know the details of the filter completely. The filter removes noise but also adds some contamination to the output. How can I find the input ...
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69 views

Can you use Fourier transformations (or other) to read multiple superimposed barcodes?

If you printed bar codes on tracing paper/acetate etc. and then positioned several in front of one another, could you extract the individual codes from the aggregate overlaid image? I feel intuitively ...
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488 views

Box filter approximations for oriented gabor filters

I'm currently working on a library to generate synthetic fingerprints using the SFinGe method (by Maltoni, Maio and Cappelli) link :http://biolab.csr.unibo.it/research.asp?organize=Activities&...
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5k views

1/3 octave spectra from fft

I have got a signal in frequency domain. This is a frequency response function from software, so I can do nothing about it and have to leave it in frequency domain. I want to transfer the data to 1/3 ...
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714 views

Matlab Simulink Sampling

First, I apologize for the vague title, to be honest, I'm not even sure how to describe my question in a title. If someone would like to edit it so its more reflective of what I'm asking, then please ...
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234 views

Deconvolution of non-stationary, 1-D signal?

I have a time series that has been measured after convolution with a moving average filter. Knowing the parameters of the moving average filter, is it possible to reconstruct/constrain the values of ...
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Convolution of spectra to sensor response function?

I am really sorry if it is wrong place to ask this question but i couldn't not find any stackexchange website appropriate. i have some reflectance values of a material which are measured in 10.5 and ...
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604 views

Why do Savitzky-Golay filters of degrees 4 and 6 have roughly opposite ripples?

Savitzky-Golay smoothing filters of polynomial degrees 4 and 6 have frequency response with ripples of roughly opposite signs. Thus averaging the coefficients of two filters of degrees d and d+2 is an ...
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7k views

Simulink continuous to discrete question

I'm new to Matlab/Simulink and would like to convert the continuous system below to discrete for eventual conversion to C code. Matlab has a c2d function, but I don't know how to apply it to a ...
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206 views

Blurring in anisotropic outputs

I am trying to produce ideal anisotropic diffusion outputs using the standard lena image. Using the first coefficient , the output is coming somewhat close to ideal ouput but when I increase the kappa ...
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28 views

Showing that filtering a signal with bandwidth B with a brickwall filter of bandwidth W>B has no effect in time domain

The time-domain representation of $G(f) H(f)$, where $H(f)$ is an ideal brickwall filter of bandwidth $1/(2T)$ is: $$ \int g(\tau) \operatorname{sinc}\left(\frac{t-\tau}{T}\right) d\tau $$ I want to ...
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55 views

Filtering a square signal with a median filtering to preserve the edges

If needed, you can find my first post for this problem here. I am trying to clean the following signal: As proposed in the comment, I tried to use this post proposing 2 methods: median filtering and ...
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2answers
116 views

How do I convert this simple Laplace equation to Z-domain?

A basic model of coupled strings (eg. piano) is provided here as: The principle is that it has two identical string simulations each formed by a delay line and LPF. The outputs of these are summed at ...
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54 views

Coupling two resonant bandpass filters? (To simulate guitar/piano string effects)

A major part of the sound of a guitar or piano string comes from the coupling of the horizontal and vertical string vibrations. This concept is described in terms of transfer functions here. This ...
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1answer
206 views

Kalman Filter EM Estimation of Covariances

The question might be very simple, but I get a strange result from Kalman Filter. Let us consider the simplest state-space model, the random walk plus noise: $$ y_{t} = x_{t} + \varepsilon_{t}\\ x_{t} ...
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41 views

Is it possible to create a causal filter with a positively-sloped phase response?

I am currently prototyping a set of noise-cancelling headphones that utilize active EQ to modify the noise-cancelling signal, allowing the user to modify the frequencies that they hear or don't hear. ...
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How to calculate Walsh hadamard transform for image patches of odd sizes?

I was reading this paper on optical flow, which as apart of the algorithm makes use Walsh-Hadamard transform on image patches of size 2r+1. So far, where ever I have looked, I could only find that ...
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85 views

Using a Passive 1st Order RC filter to Extract DC Component

I am trying to extract the DC component out of a PWM signal produced by the MCU using a LPF. I understand that 1st order RC filter is looked upon as being amateurish because the gain in the passband ...
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1answer
240 views

Why I don't get the right PSD

I need to model a noise with a given PSD. To do this, I am starting from a white gaussian noise (WGN) and feed with the WGN a transfer function, which will act like a filter. In fact,it's easy to ...
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57 views

Explicit formulation of the role of eigenvectors of the matrix of a linear time-invariant filter

At this point in a lecture by MIT Professor Strang the idea of gaining insight into a filter by studying its eigenvectors is introduced. The idea is clear with the example chosen in the lecture: an ...
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1answer
98 views

Introductory resources to mathematical properties of “dynamically recalculable” filters in audio/musical equalizers?

Introductory resources to mathematical properties of "dynamically recalculable" filters in audio/musical equalizers? Particularly, I want to understand What mathematical features (e.g. monotonicity ...
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166 views

Gaussian filter convolution giving unexpected results

I am trying to smooth a time series signal with a Gaussian filter and then differentiate the signal (this is for an application for edge detection). A nice property of convolution is: $$ \frac{d}{dx} \...
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138 views

Perfect reconstruction of 2 channel filter bank

I am studying Perfect reconstruction of 2 channel filter bank for using Biorthogonal Spline wavelet for decomposing ECG signal. I am using Wavelets and Filter Banks By Gilbert Strang, Truong Nguyen to ...
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308 views

Applying Jake's remez exchange algorithm for filter design

I have found C implementation for remez exchange algorithm(https://github.com/janovetz/remez-exchange/blob/master/remez.c) and I want to use this to implement a generic prototype lowpass filter. ...
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84 views

Gapped-signals.. How to stitch?

Dear signal processing enthusiasts: I am forced to cut off a part of a multi-frequency signal, creating a "gap" as shown in the figure. (only one frequency to simplify illustration). If I inject ...
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0answers
300 views

Software based synchronizing of multiple data streams/sensors with slightly differing sampling rates

Situation: I try to synchronize the data streams of multiple sensors (ADXL372) of the same type but with different data output rates. The data sheet states all sensors should have the same sampling ...
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0answers
162 views

Digital filter for sharp impulse in data

I am looking for an algorithm or a filter to remove sharp peaks from my data. I am collecting raw data after every 1 second from an sensor through an ADC. The ADC has an internal FIR filter. I use ...
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132 views

How to use FIR with Kaiser windows for 2D image

I am trying to use FIR filter (opensource codes from iowa hills software) as low pass filter, kaiser window, to filter 2D image. Does someone can guide me that how can I do that? I am able to obtain ...
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158 views

Gain of resonating filters

I'm trying to make a music instrument based on a resonator with a high Q, such as Ableton's Corpus and Ableton's Resonator (I think the former is based on a simple 2nd order resonant filter while the ...
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58 views

What Techniques Are Used for Signal / Image Denoising (Example Image Provided)?

I have an image that looks like this (it might appear low res in a browser because it is 16 MB). This image was taken with a scanning electron microscope, but because the equipment is outdated there ...
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86 views

Periodic signal filtering: transform from frequency domain doesn't meet expectation

The signal in question is: $$x(t)= \sum_{n=-\infty}^{+\infty} \text{rect}\left(\frac{t-4nT}{2T}\right)+\text{trian}\left(\frac{t-n4T}{T}\right)\quad\text{with}\quad T=1\,\textrm{ms}$$ Passes through ...
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0answers
346 views

NAN coming after Band Pass Filtering in MATLAB

I have applied a Band Pass Filter having cut-off frequencies [100 200] Hz and order 4 on a 10 sec signal (done in MATLAB). It is working fine. When I have increased the order to 6, though the filter ...
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0answers
2k views

Windowed Sinc filter in python

I designed a 1st order Delta Sigma converter in PSpice, except for the decimation filter. I want to implement said decimation filter in Python, rather than in PSpice. Being new to both Python and ...
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0answers
82 views

Is there a way to extract the difference between the 2 signals? Fitting?

I'm very new to signal processing and have been struggling this far. I have a device that measures the signal due to the magnetization of iron particles. In order to calculate the response of the ...
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173 views

How do FIR filters roll offs with no poles

FIR filters only have zeros, how does it work like a filter with rolloffs if there are no poles?
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136 views

Distribution of a Filtered Signal

For a given input signal, for each discretized sample, $dt=1$: $$ x(t) \sim N(0,1)$$ and a given LTI low order filter with transfer function: $$ H_{fil}(f)$$ which would be is the a priori ...
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77 views

Apply timeshift to undersampled signal after bandpass after sampling?

I have sampled my continuous signal $x(t)$ at $x[n]=x(n/f_{\rm samp})$. The signal is undersampled, and I know I will not be able to recover my original signal anymore. After sampling a bandpass is ...
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792 views

How to remove a common noise signal from a set of signals?

I have 2D measurement results which seem to have a constant non-trivial background as shown here: For every column in this image, there seems to be a similar oscillation. I'd like to distinguish this ...
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0answers
269 views

Amplitude/phase recovery on noisy signal

I have a noisy signal that has an ac component of fixed frequency but variable amplitude and phase. I'd like to recover the ac component. The signal (blue trace) is mostly smooth, but has a few ...
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0answers
86 views

maching convolution filters to results

I am practicing for an exam and I need help in the following question (it is just for me- no grade...) we get an input image, 6 convolution filters and 6 results. we need to find the matches between ...
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162 views

Open source GPS+IMU sensor fusion?

Are there any Open source implementations of GPS+IMU sensor fusion (loosely coupled; i.e. using GPS module output and 9 degree of freedom IMU sensors)? -- kalman filtering based or otherwise I see a ...

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