Questions tagged [filtering]

Filtering is a way to process an input signal so that the output signal has interesting features enhanced or unwanted features reduced.

Filter by
Sorted by
Tagged with
0 votes
2 answers
29 views

Sample rate of digital modems--how do they do digital filtering if sampling below Nyquist rate?

I can't find this answer anywhere. I have a couple satellite modem manuals and they refer to digital filtering functions that they do, but they say almost nothing about their sample rate. I always ...
Blue42's user avatar
  • 1
1 vote
1 answer
459 views

Understanding the Farrow Filter

I found the following diagram of a Farrow Filter online: As far as I understand the purpose of a Farrow filter is to interpolate samples at a fractional ratio. For example: we have an ADC sampling at ...
shaiko's user avatar
  • 147
0 votes
0 answers
34 views

How to apply a delta-filter

I'm trying to apply delta filter to remove pre-fault conditions from my signal. It is formed by a vector (7680x1), the signal was generated at 60Hz and has 32 samples per cycle lasting 4 seconds. The ...
Dragnovith's user avatar
1 vote
2 answers
135 views

What is the best method to filter a signal where baseline and the signal of interest have overlapping frequency range?

I am reading out the movement of a motor arm using a Hall sensor and a magnet pair. The hall sensor measures the distance between the sensor and the magnet. The motor arm is being moved with a band-...
chinmayeelm's user avatar
0 votes
1 answer
37 views

How does the sampling rate of a digital filter limit the sampling rate that can be used for replay of filtered signal?

I need a band-limited white noise to characterize a biological sensor that senses mechanical movement. For that, I am generating a Gaussian white noise (MATLAB Communications toolbox) of length 10 ...
chinmayeelm's user avatar
0 votes
2 answers
85 views

How do I isolate a part of this signal in time-domain in Matlab?

I want to isolate a reflection part of a signal which I've transformed from Frequency to Time-Domain (see Picture). So far, I've tried to create a Hamming-Window with the specific window-width for the ...
Anne-Mary's user avatar
3 votes
1 answer
39 views

Specific de-trending method for damped oscillation?

Say I have a set signals consisting of damped harmonic oscillations, that I want to extract the characteristics of with the prony method. In the datasets I'm working with, it is possible for these ...
George's user avatar
  • 133
0 votes
1 answer
80 views

Sample Use vs Sampling Frequency

I need to acquire an analog signal via an ADC and send it via a digital communication line (CAN bus) with a given frequency $f_{send}=100Hz$. If I understand correctly this means that this constraint ...
Jhonathan Asimov's user avatar
5 votes
1 answer
184 views

"Tone Control" IIR filter coefficients

The documentation for Analog Device's sigmastudio shows some of the formulas behind their biquad IIR filter coefficients. The usual suspects (HP, LP, peaking, ...) as described in RBJ's cookbook are ...
playduck's user avatar
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 ...
red15530's user avatar
1 vote
1 answer
44 views

Filter a 1D time series taking advantage of accuracy values for every data point

I have an assignment on a time series that contains the x positions of a point in space. This series comes with another one of the same length, where every element goes from 0.0 to 1.0 and represents ...
ThePirate42's user avatar
0 votes
1 answer
83 views

How to demodulate a bit sequence that is offset-modulated with a complex exponential?

I have a discrete-time complex exponential $a[n]*(\cos(\omega*n/N) + j \sin(\omega*n/N)) + b[n]$ where $a$ is constant and $b$: $\{0,1\}$ So it is basically an offset-modulated sequence of bits and I ...
neolith's user avatar
  • 147
0 votes
0 answers
88 views

Using python to remove Gaussian noise and a high frequency signal from a audio file

I have an audio file where people are talking, and Gaussian noise and high-frequency signals have been added to the file. My goal is to remove the noise from this file using Python. For this ...
Joe's user avatar
  • 1
0 votes
2 answers
67 views

Transmitting data with QPSK using an SDR

My question assumes that I want to transmit some data using QPSK using and SDR. The SDR will be fed with IQ data and this will transmitted at a sample rate Fs and be up converted to a carrier ...
Koisto's user avatar
  • 1
0 votes
1 answer
45 views

Characteristics of linear systems in the time and frequency domain

Why is the output signal obtained through frequency response different from the other two. ...
Alice Khomich's user avatar
0 votes
1 answer
103 views

Filtering in time domain vs filtering in frequency domain

The signal is an impulse repsonse. I am filtering the signal using a windowed sinc filter in both frequency and time domain. I am interested in knowing the differences between the two methods and ...
Tanmayee Pathre's user avatar
3 votes
1 answer
268 views

What's the point of differential beamforming?

I recently got into differential beamforming, in particular into the work of Benesty and Cohen. It seems to me that differential beamforming is something that just "takes place" when ...
northgeist's user avatar
0 votes
2 answers
151 views

Upsampling of a signal

A low-passed signal, bandwidth limited to 4KHz is originally sampled 10 KHz. If I want to resample it at 20 KHz, I take these steps. Are these correct? Am I missing an step? First we need to filter ...
Jacob's user avatar
  • 121
1 vote
0 answers
134 views

How to get an impulse response of a MIMO system?

So I have data vectors from a MIMO system and I wish to see the impulse response Any idea how I can proceed with this?
user avatar
0 votes
1 answer
125 views

What does a Gabor filter "filter" out?

Almost all of the searches online have returned results related to image-processing, but I am looking for a more rudimentary example with some physical intuition (eg; a high-pass filter only lets the ...
SNIreaPER's user avatar
1 vote
2 answers
318 views

Why does convolution give the output of a passing a signal through a filter?

I have a rudimentary understanding of Convolution, the Convolution Theorem and why the output z(t) of an LTI system can be found using the convolution of input signal x(t) and the impulse response h(t)...
SNIreaPER's user avatar
0 votes
0 answers
58 views

single tone -> multi bandpass filter -> incorrect output

background: Trying to filter out all other signal frequencies except -> signal whose frequency can be in multiple's of 16Hz {16,32,48.......160]. so started experimenting with a single tone [no ...
particlereddy's user avatar
1 vote
0 answers
303 views

How to recover the LTI system step response by the known output and input signals?

Having the input signal as a step-like pulse and the output as its distorted version after passing through the system: is it possible to somehow recover the step response of the system? In the Figure ...
Curious's user avatar
  • 355
0 votes
1 answer
154 views

ICWT filtering signal with a cut-off frequency

In ssqueezepy, by finding a certain scale is it possible to do an icwt only for the scales above (or bellow) ? For exemple if I have a double sin (with $f_1$ and $f_2$), by taking the frequency $f_c$ ...
user68570's user avatar
1 vote
0 answers
48 views

Does filtering by wavelet decomposition and reconstruction introduce a phase shift in the filtered signal?

I am removing low frequency noise from a signal using Matlab using the DWT and then reconstruction at a specific level: I am removing the approximation signal at level 10 which approximates the low ...
HP W's user avatar
  • 11
0 votes
1 answer
85 views

How to find system output by its step response?

Inspired by this post, I tried to reproduce the procedure described in the answer in Python considering rectangular pulse: ...
Curious's user avatar
  • 355
1 vote
1 answer
119 views

What is the necessity of shifting the impulse response of a filter to the front of the time domain buffer and windowing?

I was reading the answer to this question provided by Phil Karn. In the answer, it has been said: Ensure that the impulse response of your lowpass filter is shifted to the front of your time domain ...
alireza's user avatar
  • 117
1 vote
3 answers
240 views

How to estimate the local trend in a signal?

I need to remove trend from my time-series which looks like the following images. However, I want to estimate the trend before removing it. Hence directly removing it won't do it. The simple ...
Mark's user avatar
  • 357
2 votes
1 answer
70 views

Filtering after a phase domain ADC

While researching receiver architectures for the demodulating constant envelope signals like a GFSK, I stumbled over the concept of phase domain ADCs, where after $$\boxed{\text{LNA}}{\longrightarrow}\...
njg's user avatar
  • 21
1 vote
0 answers
128 views

The correct way to perform signal processing on S-parameter data

I have a question to ask you which has been bugging me as of late. I have s-parameter data with 5001 samples acquired from a vector network analyzer. I am trying to 'gate' this s parameter data, i.e. ...
Vatatia's user avatar
  • 41
3 votes
1 answer
369 views

Autocorrelation function of a filtered periodic signal

I'm having trouble with this exercise. The signal $x(t)= \cos(2\pi f_0 t)$ is sent in input to a non-linear system with input-output relationship equal to: $g(x)= 0$ if $x<0$ and $g(x)=3x$ if $ x &...
Maghreb_1911's user avatar
0 votes
0 answers
128 views

Is this how to show decimation in the frequency domain?

Let's say I have a discrete-time signal $y[n]$ that is passed through an Ideal Lowpass Filter (impulse response $h[n]$) to get $y_1[n]$. The filter has cutoff frequencies at $\pm \pi/N$ rad. Now this $...
user avatar
1 vote
2 answers
177 views

Correct baseline for slow sinusoidal noise signal

I acquired some noise signal (5GHz samplerate, 520 samples) data from a device and recognized that it is not gaussian distributed. There seems to be some slow sinusoidal signal underlying. I tried to ...
Tim Buktu's user avatar
0 votes
1 answer
126 views

Bandpass filtering EEG and getting weird distortions, parameter issue?

I am trying to bandpass filter an EEG signal, nothing fancy but it's coming out pretty distorted. EEG data is taken from forehead. Sampling rate is 250 Hz. Cutoff is 2.5 Hz & 120 Hz. Tried in both ...
fillifilterbutterfly's user avatar
0 votes
1 answer
139 views

What highpass filter is implemented in Audacity?

When I selected highpass filter effect in Audacity (an open-source audio editor), it lets me choose the cutoff frequency and the roll-off but it doesn't specify which filter type is used (IIR ...
Triceratops's user avatar
0 votes
1 answer
135 views

FIR filter for flipping the phase in negative phase

I have built up a FIR filter based on least square approach. I deploy the IDFT matrix $$F_{inv}$$ in above document page 7 and the desired response for compensated is H_F, the objective is $$ min_{...
simon Liao's user avatar
0 votes
2 answers
94 views

How to achieve a DC portable power source with low noise in 400 MHz frequencies

I am working on an SDR composed system that is plugged into a Raspberry Pi board powered by a 5V DC Power bank. The SDR is configured to measure a signal around 400 MHz. The noise floor measured by ...
M.FAKHFAKH's user avatar
0 votes
1 answer
112 views

What does frequency filtering actually practically mean?

I've got an electronics circuit which was filtering between a Passive HPF stage (1uF and 330K) and an Active LPF stage (4.7uF and 10K). As you can see it was filtering between 0.48Hz and 3.38Hz, and ...
DribbleNibble's user avatar
0 votes
2 answers
79 views

Method for splitting time-sampled signal into two signals

Update: the yellow area in the graph below can be ignored, it shows power produced. I'm only interested in the blue line, and how to separate power consumed by the heating system from the rest. I have ...
sebastian's user avatar
  • 121
2 votes
1 answer
147 views

Selective Discrete Time Derivative Filter

I have hit a problem which could be probably solved via discrete time derivative filter able to calculate the derivative of two specific harmonics of the input signal (namely frequencies: $f_1 = 100\,\...
Steve's user avatar
  • 395
1 vote
1 answer
76 views

What's the best way of modeling 3d target motion with only 2d angle observations?

A maneuvering target is flying in 3d cartesian space, but a sensor (passive infrared or mic array, etc.) can only observe it in polar coordinate with 2d orientations (azimuth, elevation). For ...
Steven Ding's user avatar
2 votes
1 answer
151 views

Approaches to work around differing signal lengths when using Kalman filter

I have a set of vector valued signals $\boldsymbol{y}_{1:T}$ where each $\boldsymbol{y}_k \in \mathbb{R}^{v_k}$. Each signal is potentially of different dimensionality. I'd like to apply filtering and ...
FakeBrain's user avatar
  • 123
2 votes
1 answer
1k views

Why does sequential update of Kalman Filter work when you have multiple sensors?

If you are using a kalman filter with multiple sensing sensors there are two ways to fuse them. One way is doing a single observation step where you include all the sensors in a single vector and a ...
FourierFlux's user avatar
5 votes
2 answers
327 views

Amplifier Placement in a Wireless Receiver

In the context of wireless receiver system design, what is the optimum choice of amplifier placement? Should the received signal be amplified first then filtered or should it be the other way round? ...
malik12's user avatar
  • 488
1 vote
0 answers
28 views

How to design a phonographic sound filter in python?

I'm making a phonographic filter or simulator to make one of my songs to sound as if it was recorded on a phonographic cylinder. Two important things about phonographic recoding is that sound gets ...
user avatar
0 votes
1 answer
175 views

Real time processing in matlab

I have to extract a particular frequency range from an audio signal. For that I am using an FIR filter of order 50 and it is written in MATLAB as below: ...
Deepa's user avatar
  • 315
4 votes
0 answers
100 views

Combination of IIR filter with Hilbert transform

I am familiar with the combination of the FIR filter with Hilbert transform, leading to the analytic signal: $$y_a(t)=y(t)+jH(y(t))=x(t)*h(t)+jH(x(t)*h(t))=x(t)*\left(h(t)+jH\left(h(t)\right)\right)$$ ...
Gideon Genadi Kogan's user avatar
2 votes
1 answer
188 views

How to implement the recursive FIR filtering of a signal with zero boundary condition?

Now I have a signal $f(k)$ with $k=0,1,\ldots, N-1$, and $f(k)=0$ if $k<0$ or $k>N-1$, that is, the signal satisfies the zero boundary condition. Then by convolving this signal with a symmetric ...
Wang Yun's user avatar
  • 124
0 votes
1 answer
521 views

Demodulate qpsk with gnu radio

I'm trying to demodulate a qpsk signal with gnu radio. The signal is from the meteor satellite on 137MHz https://www.sigidwiki.com/wiki/Low_Rate_Picture_Transmission_(LRPT). I recorded the signal with ...
fjsdlkfj's user avatar
0 votes
1 answer
256 views

Signal filtering with rrcos filter doesn't seem to work

I'm trying to learn some signal processing and recorded some iq data from the meteor satellite which transmits data with qpsk. I read the file with python and tried applying a rrcos filter on the ...
jklsdjf's user avatar

1
2 3 4 5
8