Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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Where can I find a table of $z$-domain coefficients for Butterworth filters?

The primary source lists Butterworth polynomials in s-domain and provides a link to bilinear transform for digital implementation. But, who needs analog specification in our digital world? Why should ...
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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 ...
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What does the uncertainty principle say about recursive filters?

The uncertainty principle is usually stated as a relationship between a continuous signal and that signal's Fourier transform, and says that $$ \int_{-\infty}^{\infty} \! x^2 f(x) \ \mathrm{dx} \int_{-...
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1answer
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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 ...
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How were windows originally conceived?

I am aware of the common types of windows, (Hamming, Hanning, Kaiser, Tukey, etc etc). However while many books describe them - almost none tell me just how exactly they were derived. What is so ...
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Multiresolution discrete wavelet transforms: First steps

I have been looking at trying to use wavelets for multichannel audio dynamic range processor. I am in the very early stages of trying to get decomposition and reconstruction working. In this thesis, ...
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0answers
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Filtering voice signal from randomly added frequencies

I have this assignment that I can't figure how to solve. I have a sound.wav human-voice file, and use matlab add-random-noise (monotonic) Goal: to remove the noise from the sound (and the sound from ...
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2answers
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Multiple filters or not?

I'm processing audio signal sampled at 44.1 kHz. After some other processing, I need to filter the signal between ~5-50 Hz. Achieving this for a signal with such a high sampling frequency appears to ...
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1answer
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DC blocking IIR filter clipping?

I'm processing audio data for voice input from a mic. The data arrives in 32 bit floats [-1 ~ +1]. My first filter is to remove DC: ...
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1answer
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What is the most efficient type of software bandpass filter?

I'm writing a C++ application to extract four modulated carrier signals at four separate frequencies, from a single input signal. I've designed four FIR filters at t-filter.appspot.com but they run ...
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1answer
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How do I convert a filter from Spatial to Fourier domain?

I have a small 2D Gaussian image filter that I am currently applying as a window by convolution. I would like to do this in the Fourier domain as a single multiplication. I would like to convert my ...
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2answers
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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 ...
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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 ...
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9answers
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Why do we use window in time domain rather than do FFT modify the spectrum and than inverse FFT

I thought that DSP would be done by using FFT of portions of a signal, modify the samples that result from FFT (since they represent the spectrum of our signal + noise) and remove any unwanted signals ...
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1answer
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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 ...
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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).
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3answers
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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 ...
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2answers
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Oversampling then filtering and downsampling

Currently I'm working on a project to record ELF (Extremely Low Frequency) signals from a receiver. The highest frequency of interest is about 30 Hz and the system should have a flat frequency ...
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2answers
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Filter order vs number of taps vs number of coefficients

I'm learning DSP slowly and trying to wrap my head around some terminology: Question 1: Suppose I have the following filter difference equation: $$y[n] = 2 x[n] + 4 x[n-2] + 6 x[n-3] + 8 x[n-4]$$ ...
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1answer
861 views

How to filter pulse like jumps in a signal?

I have a noisy signal with jumps to high value and back to normal for limited time bins (40 for example) which I want to filter to have a smooth signal with no jumps. Here is the graph, the blue one. ...
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0answers
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How to get translation and roation data using 6DOF MPU6150

I am trying to get the position data from an IMU 6 Degrees of Freedom (MPU-6150) board using an Arduino Uno. I am trying Jeff Rowberg’s example MPU6050_DMP6.ino that may be found here. This is the ...
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2answers
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Types of rounding in coefficients quantization

Suppose we have create an IIR filter with matlab function "ellip", and then we want to quantize the coefficients using: \begin{align*} bq=Quantize('round',b,2^8); \cr aq=Quantize('round',a,2^8); \...
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Fill matlab “ellip” function through transfer function magnitude response

The matlab "ellip" function can be used to design the unquantised coefficient set. From matlab website: $ [b,a]=ellip(n,Rp,Rs,Wp) , $ n: order of filter Wp: normalized passband edge ...
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1answer
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Determine the type of filter from the transfer function

i got that system $x_n \to x_n-x_{n-1}$, so $h_n=[.....,0,1,-1,0,...]$, with $h_0=1$ and $h_1=-1$, so the transfer function given by: $$\sum_{i=-\infty}^{\infty} h_ne^{-jwn} = h_0e^{-jw(0)}+h_1e^{-jw(...
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1answer
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Dynamically changing cut-off and sampling frequency of a digital filter

I have designed a low pass filter that smooths the output coming from an accelerometer attached to a vibrating machine. I designed this assuming cut-off frequency $f_{c1}$, and sampling frequency $f_s$...
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1answer
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Relative performance of RLS and LMS filters

It's known that the RLS filter converges faster than the LMS filter in general, but that if you're tracking time varying parameters the LMS algorithm can perform better. My question is under what ...
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583 views

Choice of cost function in adaptive noise cancellation

Why is the least squares cost function used in adaptive noise cancellation with the recursive least squares (RLS) algorithm, but the mean squared error used in signal estimation? For an ergodic ...
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2answers
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Realtime filtering on a microcontroller, with dynamic cut off frequencies

Is it possible to dynamically filter a signal in realtime on a microcontroller? What I mean by dynamically is to change the cut off frequency of the filter on the fly. I have some DSP knowledge ...
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1answer
479 views

IIR filter design in digital domain using the magnitude squared

Does anyone have any good references for deriving parameters of an IIR Low pass/High Pass filter directly in the digital domain using the magnitude squared at the corner frequency? I have been able ...
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1answer
882 views

IIR filter parallelization

Is there any current ideas about IIR filter algorithm parallelization with thread count greater than filter order? Ideally, parallelization method should load GPU efficiently.
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Bessel filter second-order sections Q and Fc multiplier derivation

A practical guide for digital IIR audio filters has cookbook-style values for creating higher-order Bessel filters out of biquads, but the values listed aren't very precise: ...
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2answers
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how to extract telephone audio only from an mp3

I have mp3s of radio talk shows in which the host interviews people on the phone. The host uses a microphone, and the interviewee uses either a landline phone or a cell phone. I want to save just the ...
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Chebfun versus Parks-McClellan

There is a Matlab-based package called Chebfun that (among other things) computes polynomial approximations of functions. See here: http://www2.maths.ox.ac.uk/chebfun/. I don't know anything about ...
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1answer
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IIR filter design in matlab

I have to design 2 IIR bandpass filter with the following specifications: Sampling frequency 1000 Hz Pass-band 50 Hz to 200 Hz Order 6 using bilinear transform and impulse invariance with Butterworth ...
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1answer
252 views

Implementing IIR filter using FSM

How do I implement an Infinite Impulse Response (IIR) filter using Finite State Machine(FSM)? The problem is FSM has finite memory, but an IIR filter has infinite outputs. The closest match is this ...
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1answer
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Online filter for EEG

My goal is to build a real-time filter for EEG data. The data are being recorded in 512Hz and I wish to filter it (in a real-time, from 0.5Hz to 30Hz in the most optimal way). Unfortunately I'm not ...
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1answer
156 views

A filter for for a realtime system?

Apologies if this is too basic. I need to implement a band-pass filter, for a realtime system. Not sure what to use. FIR? IIR? Something else ? Any tips would be highly appreciated.
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What is the ideal FIR length, given a specific pulse length?

I am trying to ascertain what the 'ideal' FIR filter length should be, given the pulse length $T_p$ of a windowed sinusoid in noise that I seek to filter. As parameters into an FIR filter that I ...
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2answers
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A Simple Lowpass Filter Behaving so Strangely

I don't know much about filter design. After reading a few articles on wikipedia I am playing with filter design and I see something weird and would like some help in understanding what exactly is ...
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1answer
113 views

Vestigial Filter problem?

I have been stuck on this question for a while now. It has to do with vestigial sideband. I wasn't sure if I should be dividing $H(\omega)$ graph values by 2 because only the positive side of the ...
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2answers
465 views

Reverse Engineering Bessel LPF (sample frequency needed)

I have the following sourcecode. For all I know, it should be a Bessel LPF at 10Hz. Is there a way to get the sampling frequence out of this? ...
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1answer
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Choice of Gaussian kernel parameters when lowpass filtering before image resampling?

I need to decimate a signal by a factor of q. More specifically my signal is a 3D "image": $\ I(x_i,y_j,z_k)$, which I need to downsample by a factor of two in the z direction. I want to do lowpass ...
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1answer
234 views

Applying a filter kernel defined by a continuous equation in the frequency domain

I have a filter kernel $K(\omega)$ completely defined in the frequency domain as a continuous function of angular frequency $\omega$. I know $K(\omega)$ defined as a continuous mathematical equation. ...
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1answer
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Pre-emphasis / De-emphasis filter design

I've designed a simple FM communication system and found that high-frequency noise is a real problem. It appears that this is a well known issue, which has been solved through pre-emphasis of high ...
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1answer
11k views

FIR filter gain

I'm writing a C function to generate bandpass FIR filter coefficients using a simple window method (more details here FIR filters by windowing) The important code snippet is the following (f1 and f2 ...
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1answer
135 views

Characterizing real-time sampled signal

I have a signal which is sampled in real-time. At every time interval I receive a value. I need to characterize this signal as active or inactive (to take priority in the control actions). By active ...
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1answer
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Oversampling and decimation: What filter to use?

I have a sensor producing (more or less) bandlimited data with a cut-off of about 45Hz, with a roll-off and AWGN. I have an ADC that samples said signal at 800Hz, with a single-pole anti-aliasing ...
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2answers
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Matlab filter design,through group delay

I have the measurements of a filter’s Group delay and S-parameters. The S-parameters are of the following form, presented in a touchstone file. !Agilent Technologies,N5242A,MY49421489,A.09.33.09 !...
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1answer
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Abs(poles) < 1 by what margin for a stable filter?

I checked the literature for recent algorithms used to design a digital filter that is a minimax approximation of a desired frequency response. All the articles I found work out examples where all the ...
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2answers
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Downsample Equiripple FIR filter

I am designing some Downsampling filters using the remez exchange algorithm (Matlab FDAtool). I see that some times i get a spike at the first and last coefficients. does anybody knows why is that? my ...