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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Is it normal that a DAC outputs spikes?

I'm using a Raspberry Pi 2 and an AD/DA Board, with no extra circuit. Reading on the internet though, I found that DACs need a low pass filter in their output, and I don't understand if the behaviour ...
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Effect of window function on audio output

If I have a low pass filter, does the choice of window function I use have any effect on the quality of the audio input? What would be the effective difference between using a rectangular window and a ...
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$Q$ factor refers to what on a lowpass filter? (EE kind of definition)

For what I'm learning, the bandwidth (of a BPF, for example) refers to the $\Delta$ between $f_L$ and $f_H$ at (usually) $-3\textrm{ db}$ from the $f_0$ (from wiki). For a peaking filter instead, $f_L$...
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Why analog anti aliasing filter is used before analog to digital converter when there is already a digital filter after ADC?

Normal data acquisition consist of: Analog anti aliasing filter( Sampling frequency : $5\textrm{ kHz}$) ADC - Digital Filter - (Sampling : 200K samples /sec) Digital low pass filters Filters DAC ...
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3answers
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What is the intuition of “averaging is a low pass filter”?

As I know, the shape of a low pass filter in time and frequency are as follow: But how averaging work like a normal function in time domain? Is there any intuition in it?
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Velocity estimation from distance sensor

I know it is a generic question but I did not find a specific answer to it online. My question is: I have a range sensor based on the time of flight (Infrared Sensor. Transmitter-Reflector-Receiver)....
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Two way Audio crossover MATLAB code using Bessel

Could MATLAB's besself function be used in designing an Audio crossover (Tweeter and Woofer) with highpass and lowpass filters? Most likely the same with this ...
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1answer
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How to design and implement a low-pass AR1 filter in Matlab?

I have a data series y(n) that I want to filter using a low-pass AR1 filter. I found that this filter has a transfer function ...
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Detecting Zero Crossing in Image Filtered by Laplacian Operator

This is my first question on this site, complete noob. But here goes. I have a positive Laplacian operator [[0,1,0], [1,-4,1], [0,1,0]] Now this Laplacian ...
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1answer
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Cutoff frequency of a first order recursive filter

I've implemented a very simple first order recursive low pass filter in c using the algorithm: ...
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Resampling an audio signal

I am trying to resample an audio signal in my application. It will be between 48kHz and 96kHz audio (both upsampling and downsampling). Looking at the Wikipedia page for resampling I see there are ...
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800 views

FIR analog low-pass filter?

Is there a way to construct FIR analog low-pass filter? (By this, I believe I am asking for no-pole analog low-pass filter, in terms of Laplace transform.)
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Optimizing Two Parallel IIR Filter Sweeps

Got an audio related question here; So I have two IIR low pass filters that work in parallel to do a filter sweep. Basically: ...
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1answer
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Beginner's question on LPF, Need for Sinc and windowing [duplicate]

As a student who has never constructed any real world DSP application, I am inclined to ask the following. Convolution in time domain is multiplication in frequency domain. For low pass filtering, ...
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Cancel Drift after numerical integration

I am trying to integrate angular acceleration obtained from a set of accelerometers positioned specifically at opposite corners of a cube, based on the paper EcoIMU: A Dual Triaxial-Accelerometer ...
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1answer
514 views

Kalman filter after lowpass filter: bad idea?

I am new to Kalman filter and am enjoying playing with it. However, I generated some random velocities and acceleration and Kalman filter (with the covariance matrices I have chosen) is comparable ...
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Interpolation filter gain

If somebody could clarify why everywhere is written that the interpolation filter should have gain $L$ where $L$ is upsampling factor. What this gain actually means? Does it compensates for the ...
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Does lowpass filtering and decimating improve SNR?

I did not manage to find clear answers to this question: does decimation after an appropriate lowpass filter improve the signal to noise ratio? To investigate this question, I considered data ...
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1answer
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How to Get Rid of Ripples from a Gradient Image of a Smoothed Image?

I have a gray scale UINT8 image. When I smooth it with a Gaussian filter and then compute the gradient image with a Sobel filter in OpenCV, ripples would appear. ...
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486 views

Sampling the impulse response

Given two IIR filters (one Low-Pass and one High-Pass) designed using Butterworth, how can I know if they can be obtained using sampling the impulse response?
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Making a low pass filter for irregular samples

We are interested in reconstructing the red channel of a scene. The red channel is slowly varying. Unfortunately the camera does not have a Bayer sensor, and the red samples are irregularly spaced ...
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Output of lowpass filter with damped sine wave input

The random process $$Y(t)=\cos(\omega_0t)\cos(\omega_0t+\pi N(t))$$where $N(t)$ is a Poisson process of parameter $\lambda$ enters a lowpass filter with transfer function $$H(j\omega) = \left\{ \...
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819 views

Low-pass elliptic filter design: equation for calculating required minimum order

In this paper on page 4 (or in slide page, 23), it describes the equation for calculating the minimum required order for elliptic filter for design specification. However, it says it is only the ...
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Bandwidth of lowpass filter?

This article here on the Window method for FIR filter design speaks of the total normalized bandwidth of the lowpass filter in $\textrm{Hz}$ (counting both negative and positive frequencies) What ...
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Can you turn a square wave into a sine wave using a low pass filter?

And if it could, would it make the sound of the square wave thinner than before because of losing its harmonics?
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Fast Integer 8 Hz 2nd Order LP for Microcontroller

I need an 8 Hz 2nd order LP filter. It does not have to be terribly accurate but it should be 2nd order-ish. Performance is much more important. My sample rate at the moment is ~9.5 kHz. The filter ...
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1answer
6k views

How can I construct a Band-pass filter from a low and a high-pass filter?

Suppose, I need to construct a Band-pass filter in OpenCV. But, I know, there are no functions in OpenCV for Band-pass filters. Now, what I need to do is to have a low-pass filter and a high-pass ...
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Upsampling with FIR filter causing discontinuity at start of new buffer [closed]

I have implemented an oversampling algorithm for an audio signal in C++ which is producing a distortion of the signal at the start of a buffer. The algorithm is wrapped in a class with methods for up ...
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Transfer function of second order notch filter

Can anyone mention the transfer function of second order notch filter to remove the line frequency of 50 Hz, in terms of frequency and sampling rate. Just like for Low pass Butterworth filter as, $$ ...
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Filtering/Analyzing Complex Data stream

I receive complex I and Q data from an USRP. I want to do a Hamming window LP filter on the data. However, I don't get the grasp of how I use the complex data. It is unusual because most of the time ...
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217 views

2D Digital Signal Processing - Applying High Pass on and Image

I have been doing my research on 2D digital signal processing. What I am doing is about noise removal of digital image: Type of noise that I want to remove from an images is like the Gaussian noise ...
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Translating Bessel filter from python to C - odd behavior past the cutoff frequency

I recently translated the code from scipy.signal.bessel to C in order to implement a bessel filter in a piece of analysis ...
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2answers
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Formula for Bessel low-pass filter coefficients

When I am filtering a signal in python, it has a built in function to generate bessel filter coefficients given a cutoff ratio and a filter order (number of poles). I am trying to translate this to C ...
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Continuous low-pass filter (LPF) on a periodic signal

I have a continuous-time signal $x(t)$ that has the following properties: It is periodic, with a periodicity of 1 (sec). It has no discontinuities or infinite values at any time. I want to come up ...
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854 views

Slope of 2nd order low pass vs band pass Butterworth filters

I have noticed that some manufacturers of audio processing products label a 2nd order Butterworth low pass as having a slope of 12dB per octave, but then also label a 2nd order Butterworth band pass ...
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When input data into a low pass filter, are the first several data unusable?

For example, I want to feed a DC into a low pass filter. The the filter's first several output will not have the same DC value I want. Does it mean that I need to dispose first several results? How ...
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extending windowing function to higher dimensions

I was looking at this particular thread that discusses extending windowing functions for singling function to higher dimension. There is one bit in the accepted answer that I could not get my head ...
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220 views

graph of lowpass filter

Check the equation 4.8-1 $H(u,v)=1$ if $D(u,v)\le D_0$ my question is the high frequency is at center. $D(u,v)$ is smaller if it is close to center. why $H(u,v)=1$ in that case. Thanks
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Which method to choose to get linear phase response zero group delay low-pass filter?

Background: I do time-dependent simulations of physical systems. Like this: http://journals.aps.org/prb/abstract/10.1103/PhysRevB.91.115431 During the simulation, observables are recorded and later ...
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1answer
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Short lowpass filter that also extrapolates lines

I'm looking for a short lowpass filter that also extrapolates lines, [0 1 2 3] --> 4, so I need (correct me) $H([\ 0\ \ 0.25\ \ 0.5\ ]) \approx [ 1\ \ 2\ \ 0 ]$ . How close can one get to this with 2 ...
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MATLAB fvtool: Setting the passband ripple below 0dB for a FIR Filter

I'm new to using filter tools in MATLAB and I'm having a bit of difficulty. I'm designing a lowpass FIR filter and I'm trying to keep the passband ripple below 0dB in the magnitude response. So far, ...
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How do I find the phase response of a filter from the filter equation?

I am new to this forum. I have been teaching myself DSP for the last two years, so my experience and knowledge has a long way to go. I am trying to find the phase response of a specific filter that I ...
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1answer
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Is there a set formula for calculating FIR coefficients (for a LPF ir 96 taps)?

I need to calculate a series of coefficients for several FIR filters having 96 taps is there a standard or set formula for calculating these, it is a low pass filter.
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how to design LPH and HPF from APF

I came across a concept that says we can design LPH and HPF from APF when the transfer function of the two APFs given as $$\begin{align}A_0(z)&=1\\ A_1(z)&=\frac{-a+z^{-1}}{1-az^{-1}}\\\\ H_{...
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Low Pass Filter - pixel correlation

I am trying to understand low pass filters for image processing, I came across an article that stated that low pass filters effect noise more than the real data due to that the image Noise always ...
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1answer
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Why Butterworth filter always starts my signal from from zero mark (amplitude)?

I am trying to filter my force signal using Butterworth low pass filter. so this is the signal, I have to filter (noise frequency is around 33 Hz). So when I use Butterworth filter with following ...
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How do I smooth this noisy signal?

I have a numerical solution to a problem in mechanics. I am computing the force applied to an object as a function of its deformation and in the problem, there is an instability, due to which the ...
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Choosing minimum sampling frequency of a filtered signal

I have a time series of acceleration sampled at 1000Hz to capture random impacts on a structure. The ringing of the structure (not of interest) produces noise which a 50Hz filter seems to eliminate ...
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Transform one-pole cascaded system into a biquadratic section

I have two cascaded one-pole sections with the same coefficient b for both sections. $$ y_A[n]=x_A[n]+b\cdot(y_A[n−1]−x_A[n])\\ y_B[n]=y_A[n]+b\cdot(y_B[n−1]−y_A[n]) $$ The final output of the system ...
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Cascaded one-pole sections and relationship to time constant

I have implemented a typical one-pole lowpass filter to smooth out some control signals. The form of the filter is: $$ y1 = x0 + b1\centerdot(y1 - x0) $$ x0 is the current input sample. y1 is the past ...

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