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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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How do I generate invertible causal filters with stable deconvolution?

I'm writing a machine learning algorithm that needs to search in the space of all possible causal and invertible convolutions/filters of size $k$ represented with floating point precision for one that ...
2 False's user avatar
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Why Does This Sliding Window Goertzel Filter Trick Work?

I'm reading an analog signal at a constant interval on an embedded device, and I want to extract a specific frequency component. Since the signal is continuously being read I want to constantly scan ...
misprit7's user avatar
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Oversampling and decimation leads to slightly different result in frequency domain

I am trying to explore the effect of oversampling and decimation with the following Matlab code ...
Vic's user avatar
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RLS adaptive filter intuitive explanation of the so-called desired signal

This wikipedia page https://en.wikipedia.org/wiki/Recursive_least_squares_filter (and in fact other sources) do not explain the apparent paradox of the cost function that computes the MSE of the ...
AlanTuring's user avatar
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Is there a computable operation that minimizes entropy?

I have recently experimentally learned that decorrelation does not necessarily imply good compression, but good compressions have the effect of data decorrelation. Since I cannot use autocorrelation ...
2 False's user avatar
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Choice of filter for accelerometer data

I had downloaded the android app Phyphox and collected acceleration data. It provides the acceleration at time t, in a CSV file. I was tasked with finding the position of the object. I would like it ...
Schrodinger_Cat's user avatar
2 votes
3 answers
164 views

Optimized Filters for Delta-Sigma Converters with Finite Time Limit

Quick read of my question The optimum estimator of the mean for N samples of a sampled DC level in the presence of white noise is a block average (an average with all samples given equal weight). What ...
Dan Boschen's user avatar
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Debugging a Haar wavelet transform in the Fourier domain

I'm trying to get a minimum-viable prototype going for a project on sensor fault detection using a scale-zero Haar wavelet (frequency-domain locality is not important). After some Googling, I managed ...
user169291's user avatar
1 vote
1 answer
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Removing DC bias

I have a signal that I have recorded from a AirSpyHF and want to remove a unwanted narrow signal with a notch filter (see figure). The spike in the image below appears to be a center spike, however it ...
Al Grant's user avatar
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What is the best way to emulate a tube EQ in DSP?

What is the best way to do it? Quite a noob question but could you design a linear filter then add a mask with the non-linearities of the tube (harmonic distortion, saturation, noise)? I'm sure there ...
thc's user avatar
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Butterworth filter vs Elliptic filter - artefacts and huge transient

I'm working on a project involving digital filters and DFT. I want to use sampled ECG data to determine the heart rate using power spectral density. While filtering the ECG samples (samples are from ...
minghierid's user avatar
1 vote
2 answers
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Baseband downconversion without imaging

I am a high school intern at a company employing heavy DSP and trying to learn the basics from the ground up. One feature of a test application I'm creating is the ability to zoom an FFT graph and ...
moltarze's user avatar
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Digital filter response at frequencies higher than the Nyquist frequency

In one system there is a maximum sampling limit of 500 Hz. And in the analog signal, there are waves with a frequency in the range up to 1600 Hz. With 500 Hz sampling, is it possible to remove ...
Arsalan Ahmadi's user avatar
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How to do RRC filtering in multiple stages?

The goal of the approach is to speed up RRC filtering with (very) large oversampling ratio. To this end I calculate the prime factors of the oversampling ratio and RRC filter the signal with each ...
notEnoughScrews's user avatar
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Interpolation Control for doing Timing Recovery in FPGAs

I intend to make a timing recovery circuit on an FPGA and for that I have to make synthesizable models of Xilinx Blocks for Interpolation Control (Module-1 Counter based) in Simulink. I have made ...
Iron Bat's user avatar
1 vote
2 answers
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Farrow filter - Effect of Delta Delay

Looking at the Farrow structure we see that the closer the specific FIR block to the output, the less effect the Delta has on it. When the output of H4 isn't effected at all by the value of the Delta ...
shaiko's user avatar
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IFFT return complex values in Matlab

I've been experimenting with the frequency sampling method for designing FIR filters. I created a low-pass filter that has a linear transition band. However, when I performed an IFFT on the frequency ...
minghierid's user avatar
2 votes
0 answers
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How to handle non-causality when decomposing a 4th order IIR filter into a parallel bank of second order filters?

What I am trying to do I am trying to code a Gaussian smoothing filter using the 4th order IIR filter described in Van Vliet's paper "Recursive Gaussian derivative filters". My code works ...
Omar Emara's user avatar
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Software based energy meter

I have been implementing a software energy meter. In other words my software has to integrate immediate electrical power over time. The immediate electrical power is calculated based on following ...
Steve's user avatar
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2 votes
1 answer
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Derive the filter mask that corresponds to the double filtering process

There are two mask filters, ...
Ryan's user avatar
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1 answer
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16-QAM with RRC filter shows a point in the middle (bad EVM)

I build a PCB with the DAC3174 from TI and I'm using it initially to test it as a channel, like shown in the diagram from the datasheet: The analog signal is mixed with the PLL carrier frequency (at ...
johnny_1010's user avatar
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1 answer
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What's the advantage of the multirate filter banks?

Slower clocks mean lower power consumption, any other benefits? Could the multirate filter be more sharper than the single-rate filter? Previously, someone asked the similar question, but the answer ...
user71925's user avatar
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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
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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
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1 answer
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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
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2 votes
1 answer
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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
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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
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1 answer
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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
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2 answers
57 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
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1 answer
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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
370 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
159 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
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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
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1 vote
2 answers
121 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
80 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
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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
324 views

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
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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
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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
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1 vote
1 answer
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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
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0 answers
37 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
58 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
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0 answers
38 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
49 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
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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
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0 answers
47 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
27 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
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0 answers
69 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
137 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
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