We’re rewarding the question askers & reputations are being recalculated! Read more.

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.

Filter by
Sorted by
Tagged with
1
vote
1answer
114 views

Multiply FS Coefficients by 0 or 1 to get Low Pass Filter?

I have written a bit of code to upsample and interpolate a sample waveform in MATLAB. It's at the point where I have taken a signal, upsampled it 2x and filled in the gaps with 0's, and then I found ...
2
votes
2answers
790 views

A Better High-Order Low-Pass Filter?

Does there exist a discrete low-pass filter algorithm that is greatly more computationally efficient for high-order (e.g. greater than 10) low-pass filters than straightforward methods like series of ...
2
votes
1answer
134 views

Perform lowpass filter by hand

I've done research and could not find any resources on this topic. I have message signal composed of more sinusoids. Then I preform amplitude modulation, that is multiply that signal with higher ...
0
votes
1answer
1k views

Calculate cut-off frequency of lowpass IIR-filter

Given a an IIR lowpass-filter in z-space: $ H(z) = \frac{\sum_{i=0}^P b_{i} z^{-i}}{1+\sum_{j=1}^Q a_{j} z^{-j}} $ How to calculate it's 3dB cut-off frequency? I about evaluating it's fourier ...
0
votes
1answer
464 views

Why we need distances in Ideal LPF?

Here I have a MATLAB code for Ideal Low pass filtering an Image. In the below code they have used D=sqrt(U.^2+V.^2); which finds some distances. Then this distance ...
4
votes
4answers
3k views

Removing values from FFT result same as filtering?

I don't quite understand why the textbooks say it is impossible to implement an ideal low pass filter. If I was to take the FFT of a discrete signal x[n], with Matlab's fft function I'd be returned ...
1
vote
0answers
120 views

Recovery of Bandlimited Signals after Companding and Low Pass Filter

Let $\Omega > 0$ and denote by $\mathcal{B}_\Omega$ the subspace of $L^2(\Bbb R)$ consisting of signals that are bandlimited to $(-\Omega, \Omega)$. Denote $\mathcal{L}_{\Omega} : L^2(\Bbb R) \...
4
votes
1answer
888 views

Envelope of a signal via Convex hull / Edge detection

Is there a method in order to compute some sort of edge detection / convex hull of a signal x[n] (in orange here). Here of course the orange signal is not convex ...
2
votes
1answer
136 views

What are filter banks

What are filter banks and what's their application? What's the meaning of $H(z)$ in descriptions of filter banks?
2
votes
3answers
8k views

Digital implementation of first order analog filter using bilinear transformation

I'm trying to create a digital filter from a first order analog filter with transfer function $$H(s)=\frac{1}{1+\tau s}$$ with time constant $\tau=.1\text{s}$, and sampling rate $f_s=1000\text{Hz}$. ...
0
votes
1answer
1k views

Envelope follower not smooth enough

With the following code, I get the "envelope" of a sound signal. However in the "envelope" there are small oscillations (high frequency) : it is not smooth enough... I imagine that it's possible to ...
1
vote
3answers
2k views

Draw a frequency response graph from a simple equation

How is it possible to draw an audio frequency response graph (x-axis : freq from 20 to 20 000hz, y-axis : dB) from a simple equation like y[n] = x[n] - x[n-486] ...
0
votes
0answers
365 views

Sine-weighted moving average on non-periodic, stationary signal

I am looking for -or trying to create- a filter with a piece-wise monotonic second derivative in such a way that when placed on a non-periodic input signal, changes in sign of the second derivative ...
2
votes
1answer
140 views

How to append two bandlimited signals and make the result bandlimited without modifying the first signal

I need to append (not add) a bandlimited signal F(b) to a bandlimited signal F(a) and keep the result bandlimited without modifying the part that corresponds to F(a). Both are bandlimited to the same ...
0
votes
1answer
204 views

Turn acausal Butterworth output into output of a causal Butterworth without input signal

I have a dataset which contains a large number of time-series which have been filtered with acausal Butterworth filters. For a real-time application, I can only use causally filtered data. Is it ...
2
votes
2answers
4k views

Mathematical Proof of Complementary Filter?

Having found some unofficial sources on Complementary Filter (Thousand Thoughts Sensor Fusion and The Balance Filter by Shane Colton), I wish to work out its rigorous mathematical proof. The ...
4
votes
1answer
237 views

New to DSP and having trouble filtering data

I am trying to make program in matlab that counts the number of times the accelerometer was lifted up and down (a rep counter for exercise). Basically my phone is getting accelerometer data on all ...
1
vote
1answer
2k views

Why use a gaussian low pass filter, when we can convolve with gaussian kernel

Sorry if this question is too simplistic, but I am new to image processing and after successfully writing two programs, a convolution program and a low pass filtering program, I noticed that for ...
1
vote
2answers
358 views

Filter design for frequencies in range 0-60Mhz

First of all excuse me for asking a seemingly vague question. But i am going nowhere in the problem i am trying to solve. I am working on a powerline communication system where i need to upconvert ...
0
votes
1answer
114 views

How does one “count” filter order?

This is primarily a question about how to name my filter. Suppose that I have a photon detector, and that I want to measure the rate at which photons arrive, with a simple gated counter (because this ...
2
votes
1answer
2k views

low pass filter after fft in matlab

I'm new to signal processing and I'm learning on my own. Therefore I might ask questions not worthwhile your time. At the moment I'm trying to figure out the concept of a lowpass filter. Using matlab ...
3
votes
2answers
4k views

Low pass filter with DFT

This is my first question at this site, please excuse me if I'm not asking the question properly. I have an acoustic signal of a test and I want to remove the "thin" peaks, because they don't really ...
1
vote
1answer
900 views

How to determine the parameter of a Complementary Filter?

I know that the Complementary Filter has the functions of both LPF and HPF. But I think my understanding on the principal behind it is still unclear. I am quite new on digital signal processing, and ...
1
vote
1answer
2k views

with SciPy, how to generate waveform with common pulse shapes?

With Python/Scipy, if I have a simply BPSK signal sequence [1, -1, -1, 1, ...], how can I generate a real-valued baseband waveforms with oversampling rate ...
3
votes
2answers
1k views

About designing digital filters [duplicate]

I'm currently using MATLAB's fdatool for filter design. Using that tool, I can easily design different kind of filters. For example, let's take a band-pass FIR filter with 10-40 Hz passband, and 5-10 ...
3
votes
0answers
562 views

Why do Savitzky-Golay filters of degrees 4 and 6 have roughly opposite ripples?

Savitzky-Golay smoothing filters of polynomial degrees 4 and 6 have frequency response with ripples of roughly opposite signs. Thus averaging the coefficients of two filters of degrees d and d+2 is an ...
1
vote
2answers
9k views

Does low-pass filtering a square wave cause a 'ringing' to develop in the signal?

Using the figure below as an example, I wonder if it is possible that the small oscillations in the signal at the time of each step shift could be caused solely by the application of a low-pass filter ...
0
votes
1answer
4k views

How to determine the window size and weights in Weighted Moving Average (WMA), given desired cut-off frequency?

I am trying to smooth my discrete-time data points using the method of WMA. Currently, I am using n as the window size and the weight array, ...
1
vote
1answer
2k views

Low pass signal filtering using FFT or simple infinite impulse response filter

I am currently in need for filtering accelerometer data for an Android application. First of all, I used a simple low-pass filter (simple infinite impulse response filter) as follows: ...
0
votes
1answer
726 views

Calculating filter parameters from system response

In trying to answer another question, I got stuck applying what I thought would be a valid way to answer it: using the system function. Here's what I did: First, I put the question ...
3
votes
6answers
5k views

Why do we need to have unity gain at DC for a lowpass filter?

I have seen many times, including in a post on this forum, that it is recommended to have unity gain at DC for a lowpass filter. I would like to now the mathematical explanation.
0
votes
1answer
6k views

How to determine ALPHA, smoothing constant, of a LPF?

I am implementing a simple LPF in Java. But I have problems in choosing the value of ALPHA. ...
3
votes
5answers
16k views

matlab How to design lpf\bpf\hpf without builtin functions

related to Filtering voice signal from randomly added frequencies I'm trying to create a simple low pass filter but i'm not sure what is wrong. my code: ...
1
vote
1answer
251 views

How to choose one among different types of LPF?

My signal graph is as follows I have < 10 reputation, thus cannot post the photo... The graph are simply roughly the series of -5, 0, +5, 0, -5, 0, +5, 0.... but with sawtooth-like small peaks ...
1
vote
0answers
537 views

Building a Low-Pass FFT filter for a noisy current pulse signal

I have some (well lots) of data signals which are measurements of current pulses. The current pulses are the result of spark breakdowns (20 kV + 700 pF capacitor so quite energetic) and my problem is ...
3
votes
2answers
676 views

Spectrum analyser and resolution bandwidth

For the preliminary part of a lab I had to do, I was asked the following question: I get how to do 1.1, but with 1.2, I really don't get what's going on; I really would've thought that the resolution ...
4
votes
1answer
2k views

Implementation of basic 2-pole low-pass filter

I know this question has been asked to hell and back from many people, and that it will probably get downvoted until it's deleted, but please bear with me. I am designing a sound engine for one of my ...
2
votes
2answers
8k views

Even after applying a low pass filter, why is there a DC offset?

I need to smooth some data segments and one of the techniques I am experimenting is a low-pass filter. So I designed a low-pass filter in matlab and applied it to my signals. Here is what happened ...
1
vote
0answers
2k views

How do decide which order my butterworth filter should have?

I have a signal that was recorded with around 300 samples in a second. Now I want to run a low pass and a high pass butterworth filter over it. The frequency for the high pass filter is around 0.05 Hz ...
3
votes
1answer
1k views

Interpolated FIR filter

I am confused about this Q&A : Interpolation by factor of 2 If my input signal $$ x[n]= x0,x1,x2,x3 $$ then according to the threads explaination my $v[n]$ will be $$ v[n]=x0,0,x1,0,x2,0,x3,0 ...
5
votes
3answers
886 views

Low pass fir filter with non-unity gain

In my filter specs, the passband gain is given as 6.02db. How do I achieve this? Usually all texts give the passband gain as 0db (as well as passband ripples if any).
4
votes
3answers
2k views

Interpolation by factor of 2

I want to up-sample my input signal by a factor of $2$. I saw zero padding followed by low pass filter method being used in few cases. But still I need some help in this. Say I have $10$ input ...
8
votes
3answers
27k views

How does a low-pass filter programmatically work?

I have been working on a simple low pass filter for < 100 Hz metering in my application. But so far, I am struggling with the theory behind it all. It's cool that I got it working, but I'd really ...
4
votes
2answers
1k views

Low Pass Filter Artifacts

I have created a Low Pass filter (I am trying to understand the topic of DSP, and it's hard.. But moving along with samples, books and a nice post on Stack Overflow). I am saving the 'output' to a ...
1
vote
1answer
179 views

Approximation using a Fourier transform with low pass filter

I need to approximate a function f, but I cannot do so with frequencies that exceed 1kHz What is the best approximation I can get? Is taking the Fourier transform then zeroing any term above 1kHz the ...
2
votes
2answers
797 views

Strange noises when filtering audio signal

I am using Naudio open source library and I am trying to do some simple filtering. The problem is that I hear some "clicks", not too loud. The library offers me the possibility to use at least two ...
0
votes
2answers
202 views

Memory of a filter used on real time digital signal processing

My question is regarding the memory of a filter when we want to implement it on a real time dsp environment. We have certain memories like scratch memory , intermediate memory and internal memory. ...
1
vote
2answers
251 views

Filter for Reducing Background Noise in Image

I have images where there is a lot of "black" background (few shades of black). (In many images at least half the pixels are background). I need to get interest points from the image, but because the ...
0
votes
1answer
448 views

Downsampling and Gradient filtering

Are the following sequence of operations the same - Down-sampling filtering followed by gradient filtering with kernel [-1 0 1] Gradient filtering with kernel [-1 0 1] followed by down-sampling ...
1
vote
1answer
2k views

Is a DC signal 0 frequency or extremely high?

If I pass a signal with DC component through a Low Pass Filter will the output be without the DC component?