Questions tagged [lowpass-filter]

A low-pass filter is an electronic filter that passes low-frequency signals but attenuates signals with frequencies higher than the cutoff frequency.

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Amplitude reduction of Simple Moving Average on sinusoid

When a -causal/non-centered- simple moving average filter (arithmetic mean) with length $n$, is applied to a sinusoidal curve with period $p$, and the resulting curve is compared to the original input ...
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How do Anti-Aliasing Filters filter the signal?

I have programmed a anti-aliasing filter that uses the fourier-transformation and makes a perfect cutoff on my desired cutoff frequency for downsampling without aliasing. I have read about other ...
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Lowpass Butterworth filter equation

I am using a 5th order lowpass Butterworth filter with a varying cutoff frequency, $f_c$, to smoothen some data coming from a spectrometer. An example of a spectrum, plotted with the filtered data, ...
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Adjusting coefficients of a LP and HP filter for a new sampling rate

I am pretty new to DSP and try to undertand some concepts and the relations between them. I currently learning about the Pan-Tompkins algorithm and trying to recreate it in Matlab (Pan–Tompkins ...
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123 views

What happens to the standard deviation of a white Gaussian noise after low pass filter?

If I apply an ideal box-car low pass filter on a random signal with white Gaussian noise, what happens to the Gaussian distribution's $\sigma$? I know that the auto-correlation function becomes a sinc(...
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Correct signal analysis flow (Wind Turbine vibration)?

I have vibration data from WT (gearbox) sampled at 25.6 kHz. From a previous post I received a lot of good recommendation. What I am trying to do: Find early stage bearing faults by using an ...
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FIR Filter - Low Pass Filter coefficients to High Pass filter coefficients

I need to convert this 12 Tap Low Pass Filter to a High Pass Filter. The coefficients of the Low Pass Filter are as follows: ...
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Envelope analyis basic question

I have a vibration signal sampled at 25.6 kHz, and I want to perform Envelope Analysis on it. "Vibration-Based Condition Monitoring of Wind Turbines" by T. Barszcz has the following figure: I ...
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How can I cut exactly the first harmonic (H1) from a FFT of an image

I have 4D( 2D + slices along z axis + time frames) gray-scale image for the heart beating on different moments. I do like to take Fourier Transform along time axis(for each slice separately), and ...
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why does my integrator perform noise shaping?

I am designing (in matlab) a simple system with a DC signal at 0 Hz and some added noise. I am performing a differentiation of the signal (by simply subtracting the last values from each other), next ...
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How to derive low pass filter $\frac{N}{N+S}$

I watched the video for PID control system where it mentioned the Laplace domain function and low pass filter to be $\frac{N}{N+S}$. I used asymptotic analysis to see that it made sense. However, I'm ...
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Is a “Spectral Tilt Filter” the best way to get an arbitrary slope and cutoff LPF?

I need an arbitrary slope audio LPF for which both the cutoff frequency and slope in dB/oct can be smoothly modulated in real time. Is the "Spectral Tilt Filter" described by Julius Smith the correct ...
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Why is the length of smoothed data from filtfilt always double of the input?

I am trying to filter the data I have using python with the code : ...
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Calculate input impedance from S-parameters

I wrote this simple matlab code to calculate input impedance of a low pass filter but not sure whether all the steps are correct. * First attempt ...
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y(t) a of an integrator circuit

I have a signal $$ x(t)= \frac{1}{T} e^{-\frac{t}{T}} u(t) - \frac{1}{T} e^{\frac{t}{T}} u(-t) $$ and I know that it transits in a integrator circuit and I have to find y(t) in time and frequency ...
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What is the correct input format to perform IFFT on a spectrum?

Say I have response of a band pass $\left(f_1-f_2\right)$ or a lowpass filter $\left(0-f\right)$ and if I want to estimate filter length then I may take the inverse transform of the frequency response ...
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Pandas resample.mean()

I have a question to all you filter experts out there. I have some engine data sampling irregularly at about 50Hz. Recently I've been downsampling via the Pandas resample().mean() method. Basically ...
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How is the formula of the cutoff frequency of a 2nd order Butterworth low-pass filter derived?

According to articles and notes, the cutoff frequency of a 2nd order Butterworth lowpass filter: is expressed by the formula: $\omega _c = {1\over\sqrt{R_1R_2C_1C_2}}$ and since $\omega _c = 2 \pi ...
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Low pass filter with different non linear phases [closed]

I want to design low pass filters with different non linear phases. Means the magnitude response of all low pass filters will be same but having different non linear phases.
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Choosing Right Type of Filter for time series data

I am working on a temperature time-series data which is very noisy. I am trying to measure true low and ...
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Why does the solution for the voltage of this RLC circuit result in a maximum capacitor value?

I have a circuit with an inductor (L), capacitor (C), and two resistors (R1=R2): This represents an equivalent electrical circuit to a plectrum or hammer (cap=spring, inductor=mass) striking a string ...
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What did I do wrong with this simple filter build?

I tried to put everything I have learned from people here together to code my first filter from scratch. Unfortunately, it didn't go well and I'm not getting the expected output. The math/code became ...
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187 views

Does mixing brown noise and white noise create pink noise?

What happens if you take a white noise generator, run it through a one-pole LPF set to a low cutoff frequency, and then mix the two together? Do you get pink noise? Say you have this code: ...
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Can you predict how long a discontinuity will take to reach its target if it is low pass filtered at a certain frequency?

Let's say you have a signal that relatively slowly rises from 0 to 1, then suddenly from one sample to the next drops back to 0, creating a sharp discontinuity. Let's say you run it through a one pole ...
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Transfer function given as z-Domain fraction: gain calculation

I am stuck on a question regarding transfer functions. I have the answer and have attempted the question but am having trouble solving it. Q: Given the transfer function of a digital filter is (5z-5)/...
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Extra frequencies after applying low pass filter

I used Python/scipy to create an input data (sinewave with frequencies {600,1200,1800,2400,...,9000}) and sampling at 10000Hz. A lowpass filter is created using butter(10, 4000/(10000/2), 10000, btype=...
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Is there a way to get output in second order sections (SOS) format from Scipy's signal.firwin

I am using Scipy's signal.firwin to implement a lowpass filter. Using the rule of thumb formula(s) to estimate the order of the filter, I am getting a very high number. I looked into this question, ...
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Use recursive least square as an “actual” filter?

I am currently studying recursive least square (RLS) and as far as I understand, the setup is (for example) that we are given a process with uncertain parameters/noise and we want to adapt parameters ...
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Slew Rate Filter With Different Smoothing Factors in Each Direction

I'm smoothing the output of an ADC with a simple first order low pass filter. Of course the smoothing rate is the same whether the signal is rising or falling. Preferably the smoothing rate needs to ...
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2-D low pass filter

I have an ultrasound signal that is 1536x128. I first downsample the signal by a factor of 2x2 in x and t, then I upsample the missing data with zeros, then I do an fft, fftshift, then another fft to ...
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Shading Correction Using Lowpass Filtering

I was taking the course on DIP. The teacher uses Digital Image Processing by Rafael C. Gonzalez and Richard E. Woods as textbook. In the part about lowpass filter, there's this example. We found that ...
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Where should I set my anti-aliasing filter corner frequency for this signal?

I have a Fourier transform of an arbitrary digital signal. The transform has For 1024 frequency bins. The magnitude plot of the first 100 is below, with no surprises in the unplotted 924 bins. Now, ...
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How to filter out noise in high frequency signal?

I am trying to filter this signal (download-zip): ...
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Does separation of variables of a transfer function followed by a z-domain transform work?

I tried taking a low pass filter transfer function in the form $A/(s^2+Bs+C)$ and separating it into multiple fractions over each root $(D/(s+r1))+(E/(s+r2))$. Then I substituted $s =((2/T)*((z-1)/(z+...
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Low pass filter output showing instability

I am fairly new to DSP, and I'm trying to implement the Pan Tompkins algorithm for QRS detection of ECG signals in MATLAB. The first stage of the algorithm consists of a second-order low-pass filter. ...
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How to design a filter with smoothness regularisation?

I am rather new to the DSP topic. However, I would like to filter a signal with lowest error and highest smoothness. Until now, I was using tikhonov regularisation inside batch regression with some ...
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261 views

windowed sinc filter in matlab

Hello i'm designing a low pass filter windowed sinc in matlab ...
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Stability of the Low pass filter after Spectral Transformation

Using a spectral transformation a filter $G(z^{cap})$ is designed to be a low pass filter from an existing lowpass filter $G_{lp}(z)$ via $G(z^{cap}) = G(z)z=H(z^{cap})$ . Which property should H(z) ...
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268 views

Low-Pass Filtering on Small Chunks of Audio

I've got a script (Python) that chops 44.1 kHz realtime audio into 10 ms chunks, processes each chunk, and stitches the chunks back together with no added artifacts. I'm trying to create a low-pass ...
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Can Kaiser's formula be used to estimate the filter length of an FIR filter for all windowing methods?

I came across this problem while designing a FIR filter for given $\omega_p = 0.4\pi$ and $\omega_s = 0.6\pi$ along with the minimum stop-band attenuation $\delta_s$. I have worked out some examples ...
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How Butterworth low-pass filter can be applied on a digital signal (i.e. discrete representation of a signal)?

I would like to know how to apply a low pass filter (Butterworth) to a digital signal. So, I have some values of the signal, let's say $S(t)$. Those values are equally spaced in time. I have read the ...
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Closed loop detection system

I am trying to design a detector of the DC component of a signal (low pass filter). I found the following topology (called Closed Loop Detection System) in an article (https://ieeexplore.ieee.org/...
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Design of low pass filter

I'm designing a low pass filter to identify the DC content of a signal. The most important aspects of my application are the accuracy and the speed of the identification. I would like to know if the ...
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Why does sampling frequency affect the filter results?

I have a big issue, basically my goal is to create a low-pass butterworth filter around 0-35Hz with a sampling frequency of 5000Hz. My Matlab code is ...
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128 views

IIR Direct FormII Filter second order in VHDL, false output

I am trying to implement an IIR second order in Direct FormII in VHDL. To make it easier to understand for me and to avoid former timing issues, I implemented the operations sequentially. In my ...
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Unable to low pass filter the signal

I am trying to use a basic example of a low pass filter as shown in the example: https://de.mathworks.com/help/signal/ref/lowpass.html However, I change the frequency: ...
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Polyphase resampling delay

I want to implement polyphase resampling. For this, I wrote a script in matlab. The script performs the following sequence of actions: ...
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Old style signal processing?

I am wondering if "an array of overlapping band-pass filters" it is 'mostly' equivalent to a Short-Time Fourier transform. See this (figure down), where $s(t)$ is the signal; LPF a second-order low-...
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Reverse engineering a digital filter

I have 2 simultaneous signals that both are designed to measure eye movements. They are sampled at 250 Hz. We have 12 subject recordings. For 3 stable periods in each subject, we choose 256 points and ...

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