Questions tagged [filter-design]

Fiter design is the process of designing a filter.

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1answer
2k views

Single-pole IIR low-pass filter - which is the correct formula for the decay coefficient?

A single-pole IIR low-pass filter can be defined in discrete time as y += a * (x - y), where y is the output sample, ...
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1answer
97 views

Orders of complimentary butterworth filters for IMU

I'm currently trying to design a complimentary filter for deployment on a drone, as part of an assessed semester project. Even though processing power is not an issue (we have GHz, lots of pipelining ...
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8answers
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Is sound the best way to transmit a signal under water?

Could sound be used in transmitting sound waves under water digitally? Could the sound then be converted by a repeater to transit by radio waves? What other ways can a signal be transmitted other than ...
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1answer
48 views

EEG Sampling, Which Order should cover my result?

I have EEG data sampling rate is 250 Hz and I want to band pass this data in ( 0.5-3 Hz & 3-8 Hz & 8-13 Hz & 13-30 Hz & 30-40 Hz). Currently, the data is in the frequency range of ...
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1answer
97 views

Doubts about Notch filter design

I am following the process that is described in this question: Transfer function of second order notch filter , I want to create a notch filter with the band suppressed equal to $f_c = 4000$ Hz, so ...
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1answer
39 views

How do I add more resolution / taps to an analog -> digital filter?

(I am trying to create an IIR audio filter that adds reverb to an initial sample) Say I designed an analog filter to model acoustic attenuation based on the following mathematical model: $$ I = I_0 ...
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2answers
121 views

Practical Least Squares FIR filter design: How to find a realistic desired response

I'm using Matt. L.'s lslevin.m function to design long ($N\approx 400$) filters to an arbitrary phase and magnitude specification. The specification is the result of a numeric optimization and does ...
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1answer
110 views

Denoising a signal using eigendecomposition

I have a complex observable series $Y(t)$ which is the result of summing two complex r.v $X(t)$ (unobservable) and a $\epsilon(t)$ (unobservable). $$Y(t)=X(t)+\epsilon(t)$$ Assume that $X$ and $\...
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1answer
99 views

“Side Lobes” When Using a Butterworth filter

So I am attempting to filter a signal containing frequencies at 7Hz, 11Hz, and 74Hz. The signal lasts for 30s and looks like this between -1s and 1s: Using a butterworth filter of order 2 with lowcut=...
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1answer
253 views

FIR design for arbitrary magnitude and phase: Why can I separate real and imaginary parts like this?

In this question about the design of a FIR filter with arbitrary magnitude and phase specifications user robert bristow-johnson suggested to split the desired complex frequency response $H$ into its ...
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0answers
49 views

Using a Passive 1st Order RC filter to Extract DC Component

I am trying to extract the DC component out of a PWM signal produced by the MCU using a LPF. I understand that 1st order RC filter is looked upon as being amateurish because the gain in the passband ...
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2answers
55 views

Digital Filter Design using difference equations

I am reading a chapter on digital filter design from analog filter design using difference equations. What they do first of all is that they map $s$ (Laplace variable) to $z$ ($z$-transform) by the ...
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1answer
68 views

Signal reconstruction from sampled signal? [closed]

Can we construct exact analog signal at receiver as same as the source from the sampled signal?
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0answers
101 views

Generation of Noise-shape filter from Power Spectrum Density

This is my very first time in dealing with signal processing, so I am sorry if I will not use a rigorous terminology. I am dealing with some issues about noise modeling in matlab. I'm trying to ...
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1answer
141 views

Find filter coefficients to model a device using its measurement

I am trying to realize a digital filter that has the same freq. response of an existing speaker. I have fed an audio sine sweep to the speaker and measured the speaker output, both at 48kHz. Then I ...
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1answer
152 views

Why must the Region Of Convergence (ROC) contain infinity and the system function be a right-sided sequence for it to be causal?

I need the simple logic on the condition a system becomes causal. We know that causal contains only past values. I can't relate this with the Region Of Convergence (ROC) concept.
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1answer
73 views

4-phase FIR LPF in MATLAB

I'm trying to create a low pass FIR filter using the coefficients I found in ITU 1770-3 on page 18. This is the last piece of the puzzle for an audio loudness metering algorithm I'm working on. I'm ...
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3answers
3k views

Savitzky–Golay filter vs. IIR or FIR linear filter

A traditional IIR / FIR filter (lowpass to remove the high freq oscillations), e.g. moving average, or a Savitzky-Golay filter can all be useful to smoothen a signal, such as an envelope signal: ...
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2answers
164 views

Bandpass filter that automatically adapts its bandwidth when a transient is detected (to avoid to smoothen the transient)

Let's say we want to isolate a band 1000 hz +/- 50 hz. Obviously, limiting the bandwidth by applying a passband filter will always destroy a bit the sharp transients (a Dirac or a rectangular ...
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2answers
252 views

How to compare many bandpass filters (for their signal propogation delay)?

Let's say we have an audio signal sampled at 96 kHz, and we want to compare several bandpass filters to find the one with the lowest rectangular-envelope-propagation-delay (see graph below). The band ...
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1answer
187 views

Design of a complex FIR filter

For apply least-squares linear-phase FIR filter design,with frequency domain specification is not symmetrical. The pass-band error function, $$E(\mathbf{h})_p=\int_{\omega_{p_1}}^{\omega_{p_2}}| \...
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0answers
46 views

How far can numerical differentiation be accurate for sinusoids?

Suppose you have samples of uniform interval of a sinusoidal signal that is some sum of sinusoids. Forget about aliasing, as the signal is bandlimited. If we have infinite number of samples at ...
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1answer
159 views

How does a digital filter work?

To introduce my situation: I'm developing a digital synthesizer in a form of a C++ library, working with low level APIs like WASAPI, ASIO, ALSA etc. It's probably not very practical and I'm mostly "...
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4answers
2k views

Why is the first value in digital first-order IIR filter impulse response not the largest?

When I create a digital first-order IIR low-pass filter with scipy (code below), I get the following coefficients: B: [ 0.1367 0.1367] A: [ 1. -0.7265] The ...
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2answers
272 views

Alias-free digital nonlinear filter design

@Jazzmaniac has a good answer to the question of how to design an alias-free digital nonlinear time-invariant filter here: https://dsp.stackexchange.com/a/28787/18276 Basically, according to that ...
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1answer
62 views

Problem with 1st order Massberg LPF [duplicate]

I'm trying to implement this filter in title by following the book "Designing Audio Effect Plug-Ins in C++" By Will Pirkle Problem I'm facing is the magnitude response error when comparing against ...
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0answers
140 views

Issue with the Massberg Analog-Matched Low Pass Filter

Implemented the filter mentioned in title using Octave and couple sources Vicanek and Zeitlin (LPF.cpp). Got filter working otherwise but, ... when tested the responses against 'analog model' of the ...
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1answer
177 views

Advancing a causal IIR filter's impulse response

Given an infinite impulse response (IIR) filter as coefficients in a topology of your choice, what is a straightforward way to modify the filter so that the impulse response is shifted in time (...
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1answer
101 views

What does it take to design a 4th order band-pass filter with -24dB/Octave?

I would like to implement in software the same isolators that the Pioneer EFX-500 features: The manual says the following about the isolators: High-performance 3-band Isolator Function [Fourth-order ...
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1answer
130 views

FIR filter design: evaluating the error function

Question 1 $$ H(e^{j\omega})=\sum_{n=0}^{N-1}h[n]e^{-jn\omega} =\mathbf{c}^H(\omega)\cdot \mathbf{h} \tag{1} $$ $$ =\mathbf{h}^H\cdot\mathbf{c}(\omega) \tag{2} $$ $$H(\mathbf{h})=\sum_{k=1}^Kh[k]e^{...
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1answer
100 views

Where to learn about “analog prototype filters”?

Where to learn about "analog prototype filters"? I've heard about them, but I'm unsure about what they really are and how they're constructed.
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3answers
177 views

Detect valleys of a signal

I have raw values that are coming directly from a sensor (on the fly). These values are coming from a wearable sensor installed on a cows neck. When the cow ruminates (meaning that it chews food that ...
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1answer
319 views

Design of FIR Filters with Arbitrary Magnitude and Phase Responses

I found many information in this thesis "Algorithms for the Constrained Design of Digital Filters with Arbitrary Magnitude and Phase Responses",i want to understand it. This question is a ...
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6answers
355 views

Optimal $ n $ -th Order IIR /AR Approximation of a Moving Average Filter

I would like to approximate a moving average filter with an IIR filter of much lower order than the tap-length of the moving average filter. Optimality shall refer to the $L_2$ norm of the impulse ...
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2answers
70 views

What is meant by “arbitrary” in the context of digital filter design?

I try to understand the 'arbitrary' what does it meant? I had read many references ,such a one it is 'begin the process with a transfer function of your choice' ,in other reference :related by the ...
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1answer
2k views

Pulse shaping with RRC : Number of taps

I am running a BPSK flowgraph in GNU Radio which is based on RRC pulse shape. I noticed that the BER is quite poor when the number of taps is low. However, using the number of taps around $20\times ...
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1answer
81 views

Arbitrary-Response Filter Design

I want to design a FIR filter by using 'cfirpm' function , which can i use the group delay and grid frequency, my code is : ...
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1answer
79 views

Design digital filter with model order reduction (MOR) and other methods

IIR filters can be designed using different methods,such as: Analog Prototyping Direct Design Generalized Butterworth Design Parametric Modeling https://www.mathworks.com/help/signal/ug/iir-filter-...
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2answers
83 views

A question about Gibbs phenomenon example in Richard Lyons book

If anyone has a copy of this book could you please shed some light here: I was trying to reproduce the example shown in Chapter 5, section 5.3.1 Window Design Method's Figure 5-19 which illustrates ...
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1answer
83 views

CCDE processing vs. Scilab function

I have used Scilab functions to produce a low-pass filter for an audio signal and the coefficients for the associated constant coefficient difference equation (CCDE). I then produced filtered signals ...
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1answer
86 views

Band pass filter (digital resonance)

Q1:  Consider a discrete-time band-pass filter (BPF) that uses only two poles as $$ H(z) = G \frac{1}{\big(1-r \, e^{j \omega_0} z^{-1}\big)\big(1-r \, e^{-j \omega_0} z^{-1}\big)} $$ where this ...
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3answers
166 views

Complex Least Squares Approximation

In the case of frequency domain FIR filter design, the error function given by : $$E(\omega)=H(e^{j\omega})-D(e^{j\omega}) \tag{1}$$ is a linear function with respect to the unknown filter ...
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1answer
88 views

Damped Harmonic Oscillation as an LTI

The goal is to create an LTI filter which is exactly, or approximates, damping of harmonic modes. The equation of course is: $$\frac{d^2 x}{dt^2} + 2 \xi \omega \frac{dx}{dt}+\omega^2x=0$$ This can ...
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1answer
112 views

The desired frequency-response specifications

The complex function $ D (e^{-jw})$ is defined on the domain of approximation $\Omega$ .In most cases the domain $\Omega$ is the union of several disjoint frequency bands which are separated by ...
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3answers
74 views

Usefulness of Practical filter ouptput

If a practical filter can't remove all unwanted frequency components like ideal filter, does that mean unwanted frequency component are still present after filtering? how can we use such distorted ...
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1answer
351 views

IIR filter design fitting step response

I have access to the step response of a system and I want to find its poles and zeros without knowing the order of the system. Consider an example of a step response of the system shown in the ...
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0answers
51 views

The grid frequencies In the design of a digital filter

I want to find many difference 'h' with same group delay and length filter using this code : https://gist.github.com/mattdsp/17f508d2b368db61f47689e4622d8841/c1518f22a5aa451951601b56c5d9f72743868bfe ...
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1answer
121 views

Do you recognize this filter?

I'm trying to decipher the uncommented source code of SoundBox which is a little music tracker written in javascript. There is an included synthesizer, which applies a filter on the output samples. I'...
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1answer
525 views

Shelving filter (2nd order)

I'm trying to find the equation of a shelving filter of the second order. I easily found the equation for a first order filter on wikipedia with a transition region which have a 6db per octave slope. ...
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0answers
173 views

IIR (non linear phase) filters

As you know, a nonlinear phase filter (in passband), distorts the frequency contents of passband region of signal. In general, Why do we use such filters in the case they cause distortion in the ...