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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Savitzky-Golay filter parameters

I am trying to smooth a series of data in order to obtain a continuous function that could represent that given data set. It came out that the Savitzky-Golay method could be a good way. Now, I don't ...
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Lowpass then Inverse filtering in python

I wrote a simple lowpass filter in python to run against lena. Now I'd like to add Gaussian noise to the lowpass filtered data and then run an inverse filter against the lowpass and try to get the ...
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Integer upsampling - leave original samples unaltered

I'm looking at the problem of integer upsampling (i.e. interpolation). This can be achieved by first increasing the sample rate by factor $L$ (i.e. inserting $L-1$ zeros between the original samples) ...
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4answers
109 views

Is there an algorithm for second order filtering of data where the filter frequency is not known a priori?

It is well known that a moving average algorithm done in the time domain is equivalent to a filter with frequency response $\mathrm{sinc}(\omega\tau)$ where $\tau$ is the averaging time. (see this ...
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98 views

Malaysia Flight 370 Image Processing

I wish that all those people on that flight were with us, and maybe they still are, as I like to think they are on an island somewhere waiting to us to find them ... however, it does not look good... ...
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Why does the inverse fourier transform of a lowpass filter have complex components in matlab?

I am quite confused whether the following numerical differences i find are just severe round off errors made by matlab, or something i am doing wrong. The following happened when trying to see what a ...
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84 views

Intro Question to Signal Processing (Low-Pass Filter)

I have a noisy signal file in Matlab and I have to denoise the polluted signal using a discrete Fourier transform. I'm asked to perform the fourier transform, then take its absolute value. Then ...
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5answers
404 views

How to separate a low amplitude signal of same frequency but different phase

I have a signal consisting of two sine waves of same frequency but have different phase (and also a lot of hf noise). Their amplitudes differ by about 120dB. How to I go about separating them? SNR is ...
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72 views

Oversampling in OFDM

Why do we need an oversampling in OFDM signal? and how to do it? If around half of the Sub-carriers are only used to send data, then do we need to perform oversampling to the signal? Is it necessary ...
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123 views

How to use low pass filter in Matlab for discrete data? [closed]

As see in the picture, there are high frequency noise in the signal. How to design a low-pass filter to eliminate the noise? How to apply this filter in Matlab?
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49 views

Low pass butterworth filter is distorting [duplicate]

my low pass filter keeps distorting and I'm at a loss to explain why. It calculates the correct coefficients, but distorts everything. The code is as follows. It's driving me mad. What's weird is my ...
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17 views

Effect of an AA filter on sampling time

An AA filter is used in control schemes to filter out the non band limited behaviour that is mostly due to the sensors noise. This avoids aliasing by eliminating these unnecessary frequencies. What I ...
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56 views

A question on filters

I am fairly new to DSP have been reading up on digital filters and have a few questions. I have a signal with very high frequencies whose DC component i want to estimate. The magnitudes of these ...
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69 views

Analytical Calculation of the Impulse Response of an Arbitrary Filter

This is a rather general question. Suppose I have an arbitrary filter, $F$, defined in the Laplace domain; $$ F=k\frac{\prod_i (s+z_i)}{\prod_j(s+p_j)}. $$ How do I calculate the impulse response of ...
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78 views

What are the values on the x-axis when I plot the impulse response of a lowpass filter?

I am trying to develop a filtering routine, using Java, and I am using the windowing method. From my studies, I was told that if I make a plot of the impulse response of an ideal lowpass filter, I ...
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74 views

Low analog lowpass + ADC vs High analog lowpass + ADC + low digital lowpass

I'm doing EEG data collection and my amp forces use of an analog lowpass filter before the ADC. I'm fairly sure that I won't be interested in phenomena associated with frequencies of 30Hz or higher, ...
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101 views

Doing resampling, how can we choose lower cutoff frequency for LPF filter?

I would like to resample an audio signal. Sampling frequency of an original audio signal is 50k Hz. What I want to down sampling frequency is 30k Hz. In this case, I followed the procedures as ...
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46 views

Signal Processing and adaptive ECG filter based on wavelet transforms

I am reading Zhu et al's paper about A Novel Automatic Detection system for ECG Arrhythmias Using Maximum Margin Clustering with Immune Evolutionary Algorithm. I am thinking how the adaptive ECG ...
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217 views

Algorithm for implementing an IIR digital filter, Chebyshev type I low pass

I am trying to implement a Chebyshev type I low-pass IIR digital filter in C. I have got the SOS Matrix and scale values from Matlab. What is the direct equation or algorithm to implement such a ...
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158 views

Realisation of a passive RC low pass filter

I have an unusual low-pass filter problem that I would like to realise with a passive RC network (Note 1). It's easy enough to find prototype filters for passive RLC networks, or for the various ...
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60 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 ...
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2answers
112 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 ...
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65 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 ...
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228 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 ...
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112 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 ...
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234 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 ...
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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) ...
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88 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 ...
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69 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?
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472 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 $f_s=1000\text{Hz}$. ...
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107 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 ...
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308 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] ...
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126 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 ...
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131 views

Polyphase IIR halfband decimator using biquad stages

I am trying to use matlab to generate coefficients for polyphase IIR halfband decimator. Which can be achieved using : ...
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1answer
116 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 ...
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64 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 ...
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392 views

Mathematical Proof of Complementary Filter´╝č

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 ...
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176 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 ...
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497 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 ...
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124 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 ...
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73 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 ...
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1answer
633 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 ...
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2answers
418 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 ...
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1answer
172 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 ...
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216 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 ...
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2answers
149 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 ...
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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 ...
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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 ...
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358 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, ...
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710 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: ...