In signal processing, a filter is a device or process that transforms a signal by selectively choosing specific frequencies and leaving others with appropriate gain or as it is.

learn more… | top users | synonyms (1)

-1
votes
0answers
18 views

Calculating FIR filter coefficients with the windowing method and Convolution algorithm

Please advice how to calculate FIR filter coefficients with the windowing method using the Rectangular Window function and convolution method? Is there some utility or FIR design application to ...
0
votes
0answers
19 views

Calculating SNR for ECG signals

I'm reading through the literature for finding the best way to calculate SNR for a ECG signal. While there are many algorithms for improving SNR, I couldn't find any that details any standard ...
0
votes
1answer
33 views

Understanding my Butterworth filter

My input is x and my filtered output is y. The sampling rate of x is ...
0
votes
2answers
32 views

Decimation of motion

I have a motion signal that you can think of as a sequence of pairs (x,y); what would be the best approach to decimate this sequence at real time? I haven't found anything similar on the web and I'm ...
0
votes
0answers
10 views

How complementary filter will remove gyro drift?

I just discovered the complementary filter and want to try it on a quadcopter. I can read that it should remove the gyro drift, but I don't really understand why... I created this simple Excel array: ...
0
votes
2answers
56 views

How to remove heart beats interference from pectoralis major electromyogram?

I am using electromyography to detect activity on pectoralis major, however I found an interference from heart beats that is affecting my posterior analysis. The sampling rate is 2000 Hz, and the time ...
0
votes
1answer
29 views

Problem to implement a sinusoid oscillator in matlab

I'm pretty new in dsp and I'm having tough times trying to get the output for a simple digital oscillator. I'm starting from the recursive formula $y[n]=x[n]+a \times y[n-1]-b \times y[n-2]$, and need ...
0
votes
1answer
21 views

MATLAB IIRs — extracting and scaling the coefficients of second order sections

suppose I generate the following filter: ...
1
vote
1answer
31 views

The two types of stability and “Why exponential”

When I was learning about LTI systems, I noticed that LTI system is said to be BIBO stable if and only if its square sum of impulse response is finite. This expression is found on many textbook of ...
0
votes
1answer
30 views

Is it possible to know the order of the filter, just looking on the pole zero plot?

is it possible to know the order of the filter, just looking on the pole zero plot. I know how to get the order of the filter using calculations(highest order), but I want to know is it possible to ...
2
votes
1answer
75 views

Common Filter Types for Audio Applications

I'm beginning to write a basic "virtual analog" synthesizer and want to implement a lowpass with variable cutoff frequency and resonance. Does anyone have insight into filter types commonly used to ...
0
votes
1answer
39 views

Frequency response of FIR (MTI delay line canceller) with non constant period

Short question How to plot the magnitude of frequency response of the delay line canceller with non constant period using GNU Octave? Or more directly: how to plot magnitude of frequency response of ...
1
vote
0answers
35 views

How to process the .wav file to get meaningful information like the number of inhales/exhales

The plot of the data from the wav file. Link to the sound file - https://drive.google.com/open?id=0BxAPlY66uX7qWWtDZUw3WkpjSGs&authuser=0 Also the FFT plot is like this I am new to the field ...
0
votes
0answers
10 views

Transposing pitch profile of one audio on another

I have one audio file with rap music and one audio with sound record of my friend's reading some text. I want to transpose pitch profile of rap music on sound recording, is that possible? I can ...
-1
votes
1answer
36 views

Recursive Implementation of the Gaussian Filter

I'm trying to implement an IIR form to approximate the Gaussian Blur Filter. I'm working with the article "Recursive Implementation of the Gaussian Filter" by Ian T. Young and Lucas J. van Vliet. ...
0
votes
2answers
37 views

Playing back an audio file at an arbitrary rate with upsampling and downsampling

Background: My overall goal is to create an audio file player which can playback an audio file at an arbitrary rate. The rate can change continuously as the file plays. This means upsampling to ...
0
votes
1answer
18 views

How to equalize frequency contents using filter?

Suppose there is a bounded continuous signal $f(t)$ that can be fourier-transformed into $F(\omega)$ of frequency contents. $|F(\omega)|$ is non-zero for only finite number of frequencies: ...
0
votes
0answers
11 views

Plotting Basis of 3D Steerable Filter

I am plotting isosurface plots of Basis of 3D Steerable filters which should look like Spherical Harmonics of first Zero, first and second order plots. Correct me ...
-1
votes
0answers
14 views

Generate human skin pattern with Reaction-Diffusion equaton or other model

I have been looking for a Reaction-Diffusion equation (or whatever model) which accurately generates this kind of pattern on human skin. I have tried multiple models, inventing own equations, ...
0
votes
1answer
20 views

How to manipulate attenuation in Parks MClellan algorithm?

I am using GNU Radio C++ source code in my app (pm_remez()) for generating FIR filters coefficients. Input parameters here are: ...
0
votes
1answer
40 views

How is the lowpass to bandpass transformation derived?

I'm using a lowpass to bandpass filter transformation: $$ s \leftarrow Q \left( \frac{s}{\omega_0} + \frac{\omega_0}{s} \right) $$ Based on the wiki article: ...
0
votes
1answer
44 views

just getting started in Signal Processing - easy question

I am reading Cycle Analytics for Traders by John Ehlers and need help. in the first section "Transfer Response" he refers to: so if I had $4$ prices say $8+8+8+8$ then average would be $8$ and ...
4
votes
3answers
135 views

Processing Accelerometer Vibration Data With FFT

For a project I am attempting to use an accelerometer to measure vibration in an RC aircraft and determine frequency from the result. To do this I am using an MPU-6000 accelerometer sampling @ 1000Hz. ...
2
votes
1answer
174 views

Don't understand coefficients in MATLAB “butter” low pass result

I don't unstand the nature of the output in Matlab's "butter" command. Calling [b,a] = butter(4, 0.5, 'low'); gives me two vectors, which the documentation says ...
0
votes
0answers
18 views

Evaluate frequency response of a FIR filter in Matlab without using freqz [duplicate]

I'm looking to evaluate a FIR filter (a row vector with five elements) in the frequency domain using only the exponential or trigonometric functions; I can't make use of the freqz function in Matlab, ...
1
vote
3answers
64 views

Is there a way to separate 2 signals recorded on the same channel?

I have 2 low frequency signals on the same line that look like this: I would like to separate them so that I obtain the two signals that can be seen here: The sampling is not perfectly ...
0
votes
0answers
24 views

Method for determining Farrow coefficients

I've been reading the paper Polynomial-based interpolation filters for DSP Applications and I'm confused about how the coefficients for the farrow structure should be determined. My first approach ...
2
votes
2answers
82 views

Removal of noise from vibration recordings

Problem I am analyzing accelerometer recordings that are contaminated by 'spikes' caused by issues in the digital circuitry. The spikes are problematic, because I wish to determine the peak-amplitude ...
0
votes
2answers
102 views

IIR filter implementation in Direct Form 2

I have implemented Direct Form 2 IIR filter. The input is the Kronecker delta function. I have written code for the response of the filter for Kronecker delta input. The code is: ...
1
vote
1answer
90 views

Plot the frequency response of MTI delay line canceller using Octave

I try to plot the frequency response of the delay line canceller (FIR filter which used for MTI). The delay line canceller has the following structure: ...
0
votes
0answers
35 views

Correct the phase of several damped sines

I have a signal $s(t)$ which consists of 3 damped sines it can be written as: $$ s(t) = A_1\sin(\omega_1t + \phi_1)\exp(-t/t_1) + A_2\sin(\omega_2 t + \phi_2)\exp(-t/t_2) + \\A_3\sin(\omega_3 t + ...
0
votes
0answers
31 views

How to use coefficients generated in c header file of fdatool in matlab?

I'm using the fdatool in matlab to design a high pass filter.I then generated the coefficients for direct form 1 structure.I exported it as a C header.The trouble is I have a vague idea as to how to ...
1
vote
1answer
41 views

Manual Extraction of Filter Phase Response with MATLAB using sweep sine

I'm currently at a loss for why my code seems to not be working. I'm doing a sine sweep on a filter to calculate its freq and phase response. The frequency response checks out, however the phase ...
2
votes
0answers
45 views

Matlab: Help in learning EKF

I have AR(1) model with data samples $N=500$ that is driven by a nonlinear input $x$. THe observation $y$ is corrupted with measurement noise $v$ of zero mean. The model is $y(t) = ay(t-1) + x(t) + ...
0
votes
0answers
20 views

Relation between Gaussian and Savitsky-Golay kernels

I would like to make a mathematical comparison between two methods in my field. One smooths with a Gaussian filter, and one smooths using a second-order Savitsky-Golay filter. Let's say the window ...
2
votes
1answer
177 views

How to test the FIR filter in C++?

I am almost done with design and implementation of my first FIR filter using C++. Its a Sinc-Windowed FIR filter designed for a cut off frequency of 100 Hz with sampling rate of 500, with a filter ...
1
vote
0answers
28 views

non-standard error function for adaptive filter

I want to create an adaptive filter. Its coefficients have this general shape: When the input signal for the filter is a sine wave, the filter behaves the desired way if the look-back window is set ...
1
vote
1answer
93 views

IIR low pass filter: floating point and fixed point

I have implemented an IIR low pass filter in direct form 2, the input to the filter is a unit impulse $\delta[n]$. I have calculated the output of the filter for the mentioned input using the ...
0
votes
1answer
56 views

Why should the cut off frequency and the transition bandwidth (BW) be expressed as a fraction of sampling rate?

I came across this confusion in Chapter 16,"Book - The Scientist and Engineers guide to DSP" and Topic is "Windowed Sinc-Filters". I am doing some calculations for designing the sinc filter and came ...
3
votes
1answer
73 views

sequence to perform filtering? which filter first?

Hi all Is there a generally accepted rule as to which order to perform filtering in? for example if I have three iir filters, a low pass filter a notch and a high pass filter, which sequence would ...
5
votes
2answers
631 views

What kind of filter is that? Is it IIR?

I am trying to answer the following question: Is the system described by $y[n]=0.5y[n-1]+x[n]-0.5x[n-1]$ an IIR filter? My answer is yes. Thank you
2
votes
3answers
106 views

when is a kalman filter different from a moving average?

this thread asks when a discrete-time Kalman filter is better/different from a simple moving average of the observations: ...
1
vote
1answer
41 views

kalman filter with time-varying noise?

in regular discrete-time (1 dimensional) kalman filter, it is assumed that we have white gaussian noise affecting the transitions and the observations: $x(t+1) = Ax + w$ $y(t) = Cx(t) + v$ ...
5
votes
3answers
158 views

How can I automatically identify and remove a fundamental frequency and all of its harmonics?

I have some microscopy data that is contaminated by a heartbeat artifact that I'd like to remove. The data consists of a large time series of images captured at ~60Hz. Here's a small example clip in ...
0
votes
0answers
9 views

Is a complementary filter usable with irregular readings?

I'm experimenting with a complementary filter for sensor fusion (the usual acc/gyro/mag). I've read whatever I could google, but two things are not very clear yet: (1) can a complementary filter be ...
0
votes
0answers
63 views

Time frequency domain analyis of signals

In the time frequency domain I have used the following denoising methods Thresholding or schwirnking - Soft,Hard,Median and with methods (minimax, adaptative and sure). However if recomposing the ...
1
vote
1answer
28 views

How to concatenate IIR filter (Butterworth) with a Matlab function and evaluate the integral

I have a function of $j\Omega$ and $v$ (velocity), i.e.: $$A(j\Omega,v)=\frac{(j\Omega)^2}{p + \frac{j\Omega}{v}}$$ where $p$ is the pole. I want know how $A(j\Omega,v)$ modify its behavior, when I ...
0
votes
2answers
38 views

Help me understand the phase of an FIR filter

I'm going over an exercise of designing an simple FIR filter, I have the solution in front of me, but I struggle to understand some parts of it. I'm asked to have an FIR filter with only 3 ...
0
votes
1answer
37 views

Obtain transfer function from signal flow graph

I have a pretty complex signal flow graph with all the blocks and the branches completely describing an analog filter, in the Laplace complex variable s. The signal $X(s)$ is the input and $Y(s)$ is ...
0
votes
2answers
62 views

Can someone please explain or describe me the behaviour of equation 2 in my question, relevant to filter response (Sinc Interpolation)

At present, I am learning the theory of operation of resampling for bandlimited periodic discrete signals using sinc interpolation.I am developing a design flow and having difficulty in understanding ...