Skip to main content

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.

Filter by
Sorted by
Tagged with
0 votes
0 answers
17 views

BER curve versus SNR for using matched filter receiver

I want to simulate matched filter receiver with rectangular pulse shape in an AWGN channel. My goal is to plot the BER curve versus SNR for BPSK modulation. I have written the code and my curve is as ...
Omid Abasi's user avatar
1 vote
1 answer
51 views

Why is it said to be alias free in such a multirate system?

Page 246 in multirate systems and filter banks by P. P. Vaidyanathanm. The sampling in the first stage should have brought the aliases already, why does the author claim that it is an aliaa-free ...
user71925's user avatar
0 votes
1 answer
39 views

How can I calculate how much of my signal at the beginning and end will be affected by the filtering process?

I'm using a Butterworth bandpass filter to process my signal (low-pass + high-pass, I am also applying a notch filter). I would like to calculate how much of my signal at the edges will be affected by ...
moray_'s user avatar
  • 3
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 ...
Al Grant's user avatar
  • 121
0 votes
0 answers
36 views

Butterworth Bandpass Filter Design

Determine the transfer function for the Butterworth bandpass filter? (At the Transfer Function Factoring and Sp Substitution stage) The Parameter $$a_{pass} = - 1dB$$ $$a_{stop} = - 20dB$$ $$f_{pass1} ...
SKevinAR18's user avatar
0 votes
1 answer
52 views

Cross-correlation of two processes generated from the same signal through different LTI systems

A problem in Statistical and Adaptive Signal Processing (problem 10.15) presents two WSS signals both generated from zero-mean white Gaussian noise with $\sigma_w = 1$. They are described by $v_1(n) = ...
Matthew Faigan's user avatar
0 votes
2 answers
55 views

Linear Phase in equiripple filters

How come equiripple filters are symmetric/anti-symmetric in time, but do not posses linear phase, but only piecewise-linearity? Do not all FIR+symmetric/anti-symmetric have Generalized linear phase?
Processor48's user avatar
0 votes
1 answer
39 views

How to make the FIR convolution sum more efficient for continuously filtered signal?

So I want to continuously filter a signal with a causal FIR filter $h$ such that at every sample timestep, I add a new sample to my current set of samples and remove the oldest sample in that set and ...
Quéston Questioneer Esquerie's user avatar
3 votes
2 answers
284 views

First order filters in Direct Form I (RBJ Cookbook)

I've implemented second order biquad filters using the famous RBJ Cookbook. I also want to implement first order lowpass and highpass filters. I was hoping to use the same code, so I need to define ...
MysteryPancake's user avatar
2 votes
2 answers
143 views

Deconvolution with unknown impulse response

This is not a duplicate of Filter secondary bounces of a pulse signal because in that question, a convolution model is finally not appropriate, whereas here, the question is about convolution. Given ...
g6kxjv1ozn's user avatar
0 votes
0 answers
8 views

Difference transformation and Stationarization of Moving Average

I have a temperature sensor data, I want to denoise it. The first thing that came to my mind was to take the moving average, it was very smooth but it is still not stationary. If I take the log ...
Clankk's user avatar
  • 1
1 vote
2 answers
117 views

Filter secondary bounces of a pulse signal

I have a single "pulse" signal (looks like a gaussian but it is not exactly), followed by several bounces, with a constant period, that probably come from the electronics used (I don't have ...
g6kxjv1ozn's user avatar
1 vote
1 answer
79 views

Filtering of sampling effects

I again need your help. You were already nicely able to help with predicting the signal I'm dealing with. Thanks a lot. (Forecasting of a periodic movement based on sensor data) Now I need to remove ...
Alexander Jasper's user avatar
0 votes
0 answers
61 views

Correcting Drift and Stepped Pattern in Pressure Sensor Data

I’m observing drift and a stepped pattern in data from a piezoelectric pressure sensor measuring constant atmospheric pressure. the results are the following: The data should ideally be a flat line ...
Optical_flow_lover's user avatar
3 votes
0 answers
304 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 ...
Creator's user avatar
  • 88
1 vote
0 answers
57 views

How can it be possible that I hear people voice from my computer speaker?

I searched on web and apparently can be due to truck drivers communicating at CB band which is around 20MHz. And apparently this communication may interference with the frequencies used by computer's ...
allexj's user avatar
  • 13
0 votes
0 answers
21 views

Filter coefficients hamming window for low-complexity Welch PSD computation

I am trying to implement the proposed method from this paper 1, to eventually have a more efficient PSD computation for feature extraction on a microcontroller. The method is suitable since I am using ...
michael's user avatar
0 votes
1 answer
49 views

How to estimate temperature based on resistance thermal dependency?

My question is tightly related to my different question. Let's say I have an inverter fed three phase induction motor drive where in the braking phase (when the motor operates in a generator mode) the ...
Steve's user avatar
  • 395
1 vote
1 answer
77 views

How does Pre-emphasis mitigate finite precision effects?

"The digitised speech signal $s(n)$ is put through a low order digital system (usually first order FIR filter) to spectrally flatten the signal and make it less susceptible to finite precision ...
Anantha Krishnan's user avatar
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
1 answer
57 views

The discrete-time impulse response of an IIR filter

Suppose you begin with the following model of a discrete impulse response: $$h[n] = \begin{cases} \displaystyle\sum_{k=1}^K \alpha_k p_k^n & n \ge 0 \\ 0 & n < 0 \end{cases}$$ i.e. a linear ...
DangerousTim's user avatar
0 votes
0 answers
18 views

Fiber Photometry (Fluorescence) Signal Preprocessing through Source Separation and Baseline Correction

I am working on preprocessing fiber photometry data. The data collection involves inserting fibers into an animal, where the reflection of specific wavelengths of light is captured. We collect two ...
Fazla Rabbi Mashrur's user avatar
1 vote
1 answer
47 views

Initial conditions for 2 IIR causal filters applied to the same time-series consecutively

I have a buffer of shape (n_samples, n_channels) in which new samples are coming in at regular intervals and old samples are discarded. In this example, the buffer array is called ...
Mathieu's user avatar
  • 213
1 vote
0 answers
57 views

Wiener Filter for Noise Reduction in GPS Sensor Measurement

I am getting position and velocity measurements out of a GPS sensor and I want to filter these data, so I can have a better, less noisy, estimation of the true measurements. I thought of doing this ...
useeeeer132's user avatar
0 votes
0 answers
36 views

Formants extraction from LPC

I would like to do formants extraction using LPC. I made the necessary pre-emphasis before feeding the speech samples to the LPC function. ...
Anantha Krishnan's user avatar
1 vote
0 answers
26 views

Removing noise from data series of road surface data

I have a data series containing underlying noise, the plot of which is : The issue is to remove the noise leaving the pattern which is raised above the lower level. The lower level is road surface ...
Roy Jaques's user avatar
0 votes
0 answers
51 views

Does the high-pass filter cause the accelerometer to miscalculate displacement?

I have an accelerometer and I've written a program for it in the ESP32. The program uses a 750sps ADC sampling rate and an accurate 100Hz interrupt for calculations. P.S. The high-pass filter is at 0....
leo_0916's user avatar
3 votes
2 answers
117 views

FIR filter insufficient stopband attenuation

I used PyFDA to design a low-pass FIR filter for the final stage of decimation of a complex (I/Q) stream, from 2048 MHz to 16kHz. The last stage, 32kHz to 16kHz is the only non half-band filter, as ...
Deepstop's user avatar
  • 153
1 vote
1 answer
68 views

How to understand haw many matched filters to use?

I have a question that i can not find a concrete answer. Haw many matched filters should be used for 8-PAM/16-QAM/64-QAM/BPSK? Can somebody shortly explain the logic we should follow? Thank you.
Viktorija Popova's user avatar
1 vote
1 answer
61 views

measuring delay introduced by filter

I am trying to measure the delay introduced by smoothening filters to my data, including a double-exponential smoothening filter. I run my series data through this function: ...
dnclem107's user avatar
0 votes
1 answer
78 views

MMSE is approximately zero-forcing at high SNR

In https://stanford.edu/~dntse/Chapters_PDF/Fundamentals_Wireless_Communication_chapter8.pdf page 360, it is stated that for a flat fading MIMO channel with receiver CSI only: at high SNR, the $\...
RTS's user avatar
  • 1
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 ...
mike's user avatar
  • 523
0 votes
1 answer
130 views

How to derive matched filters for PPM modulation?

I am trying to demodulate a PPM signal in MATLAB. My signal is constructed using 2 base functions : ...
Aloys Reant's user avatar
0 votes
1 answer
58 views

Remove noise from lung sound wav

I am working in this dataset data.mendeley.com/datasets/jwyy9np4gv/3? aiming to develop a machine learning model for classifying various diseases based on lung sounds during both expiration and ...
Zayo's user avatar
  • 1
1 vote
2 answers
88 views

A filter that decays proportionately to the wavelength of each freq? So for each wavelength at each freq, that freq's amplitude is attenuated by (*c)?

I am trying to design a filter that decays amplitude of signal passed through it proportional to the number of wavelengths that pass at that frequency in a given time. In other words, where for every ...
mike's user avatar
  • 523
1 vote
1 answer
39 views

2D Fourier downsampling to even number of points gives incorrect modes in the last column

I would like to downsample an image in the frequency domain to an even shape, similar to what is described in this post. I have been able to do this successfully with a 1D signal to both an odd and ...
Jacob Helwig's user avatar
0 votes
0 answers
24 views

Cutoff Frequency Calculation for Basic Multimode / State-Variable Filter

I am working with a basic multimode / state-variable filter topology which combines highpass, lowpass, and bandpass outputs. I am wondering if there is a formula I can use to calculate cutoff ...
Emmett Palaima's user avatar
4 votes
1 answer
83 views

Connection between LTI filters and damped + driven harmonic oscillator?

In physics, a damped harmonic oscillator that is driven by a sinusoidal force has a steady state solution: $$ x(t) = \frac{F_0}{m Z_m \omega} \sin(\omega t + \varphi) $$ where $$ Z_m = \sqrt{\left(2\...
bluenote10's user avatar
2 votes
0 answers
95 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
0 votes
1 answer
65 views

Moving average of a signal

Problem is to find 1 second moving average of a signal having sample time as Ts. To solve this, it is suggested to pass the signal through a Butterworth low pass filter having the following ...
ShiS's user avatar
  • 103
1 vote
0 answers
34 views

Implementation of a time-domain solution of a transfer function

I have the the following transfer function (TF): $$ G(s) = \frac{b_2 s^2 + b_1 s + b_0}{a_3 s^3 + a_2 s^2 + a_1 s + a_0} $$ I want to program the numerical solution of the underlying system of ODEs ...
chris's user avatar
  • 11
0 votes
1 answer
56 views

Does upsampling require a low-pass filter?

I am currently wondering about the frequency spectrum when performing upsampling using linear interpolation. As far as I know, linear interpolation corresponds to a $\text{sinc}^2$ function. If I ...
hajo's user avatar
  • 101
3 votes
1 answer
252 views

How do amateur (ham) radio noise reduction filters work?

Amateur radio receivers often have several types of audio filters, and a common one is simply called "Noise Reduction." Compared to notch filters, impulse filters, and adjustable bandpass ...
gschro's user avatar
  • 103
0 votes
1 answer
73 views

Finding the impulse response between two microphones

I have made recordings of a single source (musical instrument) with multiple microphones and in the interest of limiting the size of the final project, I would like to find a way to get the impulse ...
arno sica's user avatar
1 vote
1 answer
46 views

output of CIC filter is scaled by what frequency?

Suppose I have a CIC with a down sampling rate $R$, differential delay $M$, and number of sections $N$. My sample rate is $F_s$. If I pass a signal through my CIC I will down sample by $R$. If I plot ...
CakeMaster's user avatar
13 votes
4 answers
2k views

Why is the time domain low-pass filter the "sinc" shape?

Consider: I'm looking at low-pass filters, and I see that the time domain representation of an "ideal" filter resembles the shape above whereas the frequency domain is a box. I also get the ...
thepman's user avatar
  • 133
1 vote
0 answers
22 views

Confused About Relationship Between Number of Points and Solving Wiener Filter Coefficients in Time Domain vs Fourier Domain

I'm interested in computing M, FIR filter taps / coefficients using a Wiener filter approach with known signal and noise timestreams. I know of two ways to compute the coefficients: one in the Fourier ...
Jenny Smith's user avatar
0 votes
0 answers
31 views

Is there any way of calculating the portion of signal lost as we keep on reducing the passband edge frequency of a band-pass filter?

Any kind of references would be appreciable.
Hardik Dey's user avatar
1 vote
0 answers
66 views

Which filter is suitable for reducing noise in feature detection?

I have a feature detection algorithm that is called FAST - Feature Accelerated Segmented Test. It's a very fast algorithm for finding feature points, e.g "corners" inside an image. Here is ...
euraad's user avatar
  • 405
1 vote
0 answers
35 views

RLS Adaptive Filter for estimating signal

I am currently working on a project where i am to estimate a signal x_T using x_1 and x_2 with an RLS filter. I have a problem where i don't quite get the results i am looking for. I think there is a ...
Ramin's user avatar
  • 11

1
2 3 4 5
59