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.
518
questions with no upvoted or accepted answers
6
votes
1
answer
2k
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 ...
- 61
5
votes
0
answers
105
views
Find moments of a discrete process using $\mathcal{Z}$-Transform method
For a linear process relating a variable $Y_i$ to random, independent variables $X_i$ using the equation:
\begin{equation}
Y_i = aY_{i-1} + (1-a)X_i
\end{equation}
which has the solution:
\begin{...
- 213
4
votes
0
answers
229
views
Down-sampling a signal after convolution
I was trying to understand more about image compression using the Wavelet transform and I got quite confused about the down-sampling step. I took a picture from here for reference:
At each step a ...
user62908
4
votes
0
answers
197
views
Origin of the name "Olympic filter" in signal processing
Olympic filter is the name of a digital filter for removing spikes in the data. In the "Image Processing Handbook" 7th Ed., pg 192, this name is rationalized as follows:
A modification to ...
- 569
4
votes
0
answers
7k
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 ...
- 41
4
votes
1
answer
109
views
log(conv) vs conv(log)
Suppose an arbitrary, strictly positive $x[n]$, transformed as
$$
x_l[n] = \log(1 + C x[n]) \tag{0}
$$
where $C$ is freely chosen. Given the following, where $h[n]$ is a Gaussian lowpass filter (or ...
- 8,913
4
votes
1
answer
262
views
New to DSP and having trouble filtering data
I am trying to make program in matlab that counts the number of times the accelerometer was lifted up and down (a rep counter for exercise). Basically my phone is getting accelerometer data on all ...
- 41
4
votes
1
answer
1k
views
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 ...
user5162
4
votes
1
answer
657
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 ...
- 41
3
votes
0
answers
276
views
+100
Is this better detection and Gaussian noise cancellation technique than Matched Filter
The code below generates one sinusoid and considered as known signal. One needs to find the location of the signals under additive Gaussian noise.
The known signals is comprise of the sinusoids with ...
- 88
3
votes
1
answer
85
views
Why are Some Filter implementations Preferable for Adaptive IIRS?
I am aware that some filter implementations such as lattice/ladder and SoS sections are advantageous over high order transversal filter structures in terms of coefficient update convergence in ...
- 111
3
votes
0
answers
379
views
Edge artifacts in frequency-domain Moving Average filter
I am currently trying to implement in python an algorithm to identify and filter out one specific periodic EEG artifact -- the grad fMRI artifact -- as described in this paper, and I am incurring in ...
- 31
3
votes
0
answers
253
views
CIC filter internal register truncation
I try to implement CIC filter with truncation in hardware, but there is some misunderstanding. For some reason the output signal is broken.
My filter is N = 6, R = 20, M = 1, Bin = 16, Bout = 16.
(...
- 31
3
votes
0
answers
279
views
Kalman Filter for position : introducing acceleration estimates
I want to estimate the position on a 3D environment by introducing only acceleration estimates. Is that possible? If I use the extended Kalman Filter and introduce these estimates will I have the ...
- 31
3
votes
0
answers
44
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 ...
- 31
3
votes
0
answers
709
views
Avoiding transients during filtering: Padding or Gustafsson's method?
I'm pretty uneducated when it comes to filtering, more of a stats guy, but I have to bandpass some signals and after a bunch of reading today I understand that the ...
- 219
3
votes
0
answers
175
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,
...
- 598
3
votes
0
answers
265
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 ...
- 191
3
votes
0
answers
414
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. ...
- 143
3
votes
0
answers
2k
views
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 ...
- 35
3
votes
0
answers
269
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 ...
- 754
3
votes
0
answers
365
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 ...
- 233
3
votes
0
answers
76
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 ...
- 131
3
votes
0
answers
523
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&...
- 131
3
votes
0
answers
775
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 ...
- 131
3
votes
0
answers
705
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 ...
- 608
3
votes
0
answers
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 ...
- 227
3
votes
0
answers
216
views
Reducing the Blurriness of Anisotropic Diffusion
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 ...
- 341
2
votes
0
answers
59
views
Smoothing Filter on Signal
With python scipy I am processing (in real time) chunks of 250,000 each a CW pulsed beep radio signal in python from a RTLSDR. The signal after some basic processing and decimation is 9995 samples ...
- 121
2
votes
0
answers
39
views
What is the cleanest, most transparent way to rapidly excite a resonant bandpass to a given amplitude (for musical synthesis)?
Background
Music audio synthesis can be performed using underdamped resonant bandpasses as harmonic oscillators (set to harmonic frequencies of a given note). These can then be excited when input ...
- 523
2
votes
0
answers
94
views
Implementation of an IIR filter with integer coefficients
I'm trying to figure out the implementation of IIR filters with integer coefficients, but I've run into a problem that I can't figure out.
There is an excellent post on this topic, and I took it as ...
- 95
2
votes
1
answer
55
views
2x oversampling channelizers and perfect reconstruction
For perfect reconstruction, it seems that $2\times$ oversampling allows a prototype filter that ensure the spectrums of the subchannels overlap correctly.
In the gnuradio reference on this it seems ...
- 81
2
votes
0
answers
47
views
Trying to understand scipy.signal.kaiserord
I am trying to understand scipy.signal.kaiserord example. The example says
We will use the Kaiser window method to design a lowpass FIR filter
for a signal that is sampled at 1000 Hz.
We want at ...
- 131
2
votes
0
answers
100
views
Bandpass filter with a changing passband
The problem:
I have a signal that contains multiple, relatively stable frequency components and I want to extract only one of them. I attached a plot that illustrates the situation with a simple ...
- 21
2
votes
0
answers
65
views
Can Synchrosqueezing be use to Derive IIR Filter From Impulse Response
I am looking for alternate solutions to determine the IIR filter coefficients from the impulse response, or more specifically determine the closest IIR filter for a given FIR filter. Much of this is ...
- 51.3k
2
votes
0
answers
54
views
Deconvolving while suppressing other known signals
Given a known impulse response $h(t)$ and an observed signal $y(t)=(h*x)(t)+n(t)$, my aim is to recover an estimate of $x(t)$ in the presence of unknown additive noise $n(t)$.
If the mean power ...
- 193
2
votes
0
answers
67
views
Numerical optimization method for optimal edge detector
John Canny, in his paper "A Computational Approach to Edge Detection" (PDF), finds the expression of an optimal edge detector by first using an analytical approach (Euler-Lagrange equation). ...
- 51
2
votes
0
answers
87
views
Ideal low pass filter with cut-off B does not affect a band-limited baseband signal with maximum frequency B
Let $x(t)$ be a band-limited signal with spectrum (fourier transform) lying between -B and +B. Such a signal is not warped by an ideal low-pass filter with cut-off frequency equal or higher than +B.
...
- 324
2
votes
0
answers
118
views
Equivalence between the linear phase of FIR filters and the (anti)symmetry of the coefficients
According to the Matt L.'s answer to this question, real-valued and causal FIR filters have a (generalized) linear phase response if and only if their coefficients satisfy (anti)symmetry condition. ...
- 726
2
votes
0
answers
418
views
How to read the filter IIR design numericals specifications?
First of all I want to let you know about notations that I am using-:
$\alpha_p$=attenuation factor of passband(in dB)
$\alpha_s$=attenuation factor of stopband (in dB)
$A_p$=Gain at passband edge ...
- 157
2
votes
0
answers
154
views
Multiple 1D gaussian filter
Given
$$
\begin{cases}
&f_0(x)=1 \\
&f_{n+1}(x)= (\varphi*(f\mathbb{I}_{[a_n,b_n]}))(x)=\int_{-\infty}^{+\infty}\varphi(x-t)f_n(t)\mathbb{I}_{[a_n,b_n]}(t)dt = \int_{a_n}^{b_n}\varphi(x-t)f_n(...
- 143
2
votes
0
answers
51
views
Showing that filtering a signal with bandwidth B with a brickwall filter of bandwidth W>B has no effect in time domain
The time-domain representation of $G(f) H(f)$, where $H(f)$ is an ideal brickwall filter of bandwidth $1/(2T)$ is:
$$
\int g(\tau) \operatorname{sinc}\left(\frac{t-\tau}{T}\right) d\tau
$$
I want to ...
- 473
2
votes
0
answers
515
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 ...
- 213
2
votes
0
answers
256
views
The size of an FIR filter for PDM-PCM conversion
I have been trying to understand how the conversion from PDM to PCM happens so that I could interface with a microphone. I have been doing some research online and I found that the most common way to ...
- 173
2
votes
0
answers
70
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 ...
- 523
2
votes
0
answers
227
views
How Do Particle Filters Get Velocity When Tracking with Pos Measurements
I am new to particle filtering.
I can see when particle filters are used with Ensemble Kalman's, the velocity of the states are taken care of by the Kalman.
When tracking using particle filters ...
- 21
2
votes
0
answers
70
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. ...
- 21
2
votes
0
answers
273
views
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 ...
2
votes
0
answers
209
views
Extracting signal from data using Otsu thresholding in MATLAB?
I am very new to signal processing. I am trying to extract a signal from my data using otsu threholding but it's not fully accurate. In this case I know that there is going to be a single signal in ...
- 183
2
votes
0
answers
249
views
Octave filtfilt converted to R code
R has a function called filtfilt that performs zero-phase digital filtering by processing the input data in both the forward and reverse directions. But it suffers from having steep transition bands....
- 21