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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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1answer
84 views

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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16 views

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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27 views

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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23 views

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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2answers
46 views

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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34 views
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57 views

How to filter out noise in high frequency signal?

I am trying to filter this signal (download-zip): ...
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20 views

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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1answer
50 views

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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21 views

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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1answer
107 views

windowed sinc filter in matlab

Hello i'm designing a low pass filter windowed sinc in matlab ...
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1answer
55 views

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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3answers
184 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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43 views

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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2answers
68 views

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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23 views

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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34 views

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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1answer
39 views

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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2answers
68 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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1answer
38 views

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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61 views

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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2answers
96 views

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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3answers
108 views

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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1answer
65 views

Manually creating a lowpass filter from the frequency domain: what phase?

In an attempt to gain a better understanding of DSP, I want to create a very simple (1D) lowpass, or I think more correctly "stop-band" or "band-reject" filter, to filter out a single frequency, e.g., ...
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1answer
86 views

Low-Pass Filtering of not evenly sampled signal

I have a signal sampled unevenly over 1 million ns. the signal is sampled over 1GHZ clock and the samples are as the following: 0-100 ns - sample every 1 ns. 100-1000 ns - sample every 10 ns. 1000-...
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1answer
59 views

analog-to-discrete-to-analog system sampling problem

This is not a homework problem, I am solving practice problems for my exam. Consider the analog-to-discrete-to-analog system shown in figure 1. The CT signal $x_a(t)$ is sampled at a frequency of $...
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1answer
82 views

Side Lobes in the Magnitude Response of a Low Pass FIR Filter

I have the impulse response (from the filter coefficients) of an FIR filter obtained from MATLAB's "interp" function using the command: [y, b] = interp(x, 5); % where b contains the interpolation ...
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65 views

low-pass filter and the sharp cut-off filter

are the low-pass filter and the sharp cut-off filter the same thing?? Here there is the definition of sharp cut-off filter https://en.wikipedia.org/wiki/Filter_(large_eddy_simulation) . I tried to ...
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1answer
53 views

Low-Pass FIR that keeps all values positive

Let's say, that I want to low-pass the norm (hence positive values) of an analytic signal (related to Hilbert transform). This allows me to perform experiments with the envelope of audio. For low-...
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1answer
29 views

Filtered signal does not follow my cut off frequency

I filtered my signal with low pass filter butterworth forth order with cutoff frequency of 50 hz. However when I compare the filtered signal with original one in frequency domain, it seems that the ...
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0answers
78 views

Quadrature demodulator. Relationship of sampling rate and error

Task: to estimate the frequency of the power grid signal. I use a quadrature demodulator. I get the I / Q components. I calculate the angle (phi = atan(Q / I)). I differentiate the angle and get the ...
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14 views

Differentiating high frequencies if a signal

I have a power signal (in dBm) which contains lots of peaks (around -80 dBm) and most of the time is around -110 dBm which we can call it noise level (obviously with fluctuations). As you know, both ...
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1answer
39 views

What impact have different types of border-handling on high pass or low pass filtering?

When high or low pass filters are being applied to an (grayscale) image, the borders need to be handled separately. They could be filled with 0's or 255's (b/w), repeat the nearest pixel value, or ...
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1answer
103 views

Amplitude demodulation of sinusidal signal in matlab

AM-demodulation task and I was asked to demodulate the signal in MATLAB and the result should contain three pictures. plotting the $fft$ the (frequency spectrum)of the signal i was able to determine ...
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1answer
230 views

Low-pass filter with varying cut-off frequency over time

Is there a novel way to create a low-pass filter (Finite Impulse Response Filter) with varying cut off frequency over time? For example, I have a white noise that is 10 seconds long sampled at 44.1 ...
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32 views

How do I filter out the specific components at 20MHz, in this figure?

I used the filter builder (lowpass) , to cut off the frequency at 0.5 MHz, and thus used a stopband of 0.1pi, and a passband of 0.0125pi. (Sampling Frequency= 80MHz). However, the output was still ...
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1answer
136 views

Fixed-point Exponential Moving Average filter with given cutoff frequency

I want to implement in fixed point arithmetic an exponential moving average filter, with a specific cutoff frequency for a given sampling rate. The formula for the filter is $$ y_n = \alpha x_n + (1 -...
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1answer
31 views

How to draw the magnitude plot of a variable slope low/high pass shelving filter? (eg. as a function of cascaded shelving filters)

I am interested in making a magnitude graph of a variable slope low/high pass filter. I would like to set slope in dB/oct and a cutoff pitch. The y-axis should be amplitude from 0 to 1, and the x-axis ...
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1answer
69 views

Why is Cheby 2 filter uncommon?

I've been reading the Chebyshev filters chapter in The Scientist and Engineer's Guide to Digital Signal Processing and it says "The Chebyshev filters discussed in this chapter are called type 1 ...
4
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1answer
161 views

2-d circularly symmetric low-pass filter

For a square pixel grid, the ideal 2-d low-pass filter with a horizontal and a vertical cut-off angular frequency $\omega_c$ in radians has an impulse response (kernel) $h_{\small\square}(x, y)$ that ...
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0answers
80 views

Why do I have to scale filter coefficients during bandlimited sample rate conversion?

I am writing an example for how to perform sample rate conversion using windowed sinc and polyphase filters. Here is the C++ code as a github gist. So far, this works, and I think I have understood ...
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0answers
72 views

High quality removing gravity from accelerometer data without Sensor Fusion

My smartphone have only accelerometer sensor. I need to get data close to (0,0,0) in hold in my own application. I did use some app from Google Play and data was very close to it (some like 0.1 and ...
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0answers
29 views

Frequency warping while creating filter coefficients

If I have an audio signal at 16kHz with frequencies band-limited to 8kHz. I want to low-pass filter this signal such that fc=4000Hz. Meaning when I reconstruct the signal I do not want any content ...
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1answer
190 views

Filtering on time or frequency domain?

I'm applying an ideal filter on the frequency domain. My code below: ...
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0answers
29 views

What is the difference between a range filter and a low-pass filter?

A low-pass filter attenuates the high frequencies, preserving only smooth variations in the provided image. The result of an image which is low-pass filtered is an image with blurred edges (hence ...
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1answer
33 views

Why exactly is the Guassian filter a low-pass filter?

In image processing, a Guassian filter is used to blur an image. A low-pass filter is a filter that attenuates the high frequencies, preserving only smooth variations in the provided image. Is a ...
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98 views

“Improved” MZT/IIM type One pole LPF

Some time ago, while playing with all kind of approximation of common math functions, I came up to this idea to calculate a1 coefficient for a 1st order LPF filter: \begin{array}{l}a1=-(1+x+\frac12x^...
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1answer
103 views

Understading sampling theorem and aliasing

A real-valued analog signal with a flat spectrum between f = 0 Hz, and fmax is sampled at a sampling frequency fs = 24 kHz. The sampled signal is then processed with an ideal lowpass filter with ...
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0answers
163 views

How to implement one dimensional low pass Gaussian filter with zero-phase shift?

I am trying to apply a low pass Gaussian filter on a signal and find the exact location of maximum/minimum in the resulting smoothed signal. I have to do it for a few values of sigma (Gaussian ...
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0answers
44 views

How to low pass very few samples

I'm asking myself what is the best way to low pass a signal which only has a few hundred samples? Or is there a proper way at all? It must not be a sound signal but I used Audacity to display the ...