Questions tagged [filter-design]
Fiter design is the process of designing a filter.
1
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1answer
34 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, ...
0
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0answers
29 views
How to filter a discrete waveform by having continuous domain poles and zeros
Let's consider we want to model a continuous time filter that has one zero at 30GH and two poles at 35GHz.
I also have a sampled waveform with sampling frequency of $f_s$, so the time interval ...
1
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1answer
71 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 ...
0
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0answers
26 views
How does works the transfer functions in the “cookbook formulae for audio EQ”?
To try to understand the audio filtering, I'm working with those two documents :
Introdution to digital filters with audio applications
Cookbook formulae for audio equalizer biquad filter ...
5
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7answers
2k views
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 ...
0
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1answer
27 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 ...
-1
votes
1answer
39 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 fc = 4000Hz, so using ...
1
vote
1answer
31 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:
$$
I = I_0 ...
0
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0answers
18 views
Octave's decimate — adequate filter order as a function of downsampling factor?
The documentation for decimate in Octave (I believe also in Matlab) indicates fixed default filter lengths (Chebyshev order 8 if IIR, and order 30 if FIR). The Python documentation for the equivalent ...
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0answers
16 views
How to design a special digital filter, which could automatically adjust the cut-frequency in real-time according to the input parameters?
My audio processing object is a 7-second noisy piano signal.
[x_n,fs]=audioread('piano_test.wav');
X_n=x_n(:,1); % single channel
In the first step, I used a ...
5
votes
2answers
58 views
Practical Least Squares FIR filter design: How to find a realistic desired response
I'm using Matt. L.'s lslevin.m function to design long ($N\approx 400$) filters to an arbitrary phase and magnitude specification. The specification is the result of a numeric optimization and does ...
2
votes
1answer
60 views
Denoising a signal using eigendecomposition
I have a complex observable series $Y(t)$ which is the result of summing two complex r.v $X(t)$ (unobservable) and a $\epsilon(t)$ (unobservable).
$$Y(t)=X(t)+\epsilon(t)$$
Assume that $X$ and $\...
-1
votes
1answer
26 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=...
4
votes
1answer
101 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 ...
1
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0answers
20 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 ...
-1
votes
2answers
42 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
45 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
16 views
Fitting filter coefficients within a narrow band of frequencies, and using them as multi-rate
Intro:
I have a vibrating system that I am trying to model. I've input a sine sweep up to 1200Hz and measured the device's input and output into MATLAB. I am interested in finding the inverse filter ...
1
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0answers
59 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 ...
1
vote
1answer
59 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 ...
0
votes
1answer
40 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.
1
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1answer
38 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 ...
8
votes
3answers
263 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:
...
2
votes
2answers
103 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 ...
0
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0answers
47 views
Filtering Audio Signals in MATLAB
I'm trying to apply a filter to an audio signal in MATLAB and having some trouble processing it.
So far, I have a transfer function that describes a K-weighted filter, and I am able to create a bode ...
1
vote
2answers
87 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 ...
1
vote
1answer
61 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}}| \...
1
vote
0answers
31 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 ...
0
votes
0answers
30 views
How to select Input parameters of FIR band pass filter (narrow band e.g Alpha (8-12 Hz)). A basic question?
I have to filter eeg data (500 samples) offline using a FIR bandpass filter for alpha rhythms (8-12 Hz) providing the fixed order of filter (i-e., 128).
Can some one explain what should be the ...
2
votes
1answer
94 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 "...
5
votes
4answers
463 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 ...
0
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0answers
30 views
Phase considerations in image filter design for precise noise removal (image in post)
I have images with systematic noise caused by the instrument in different subdomains of the image, and I'd like to design a filter to - on average - improve the quality of the images.
Since the noise ...
2
votes
2answers
221 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 ...
1
vote
1answer
51 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 ...
1
vote
0answers
92 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 ...
4
votes
1answer
148 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 (...
0
votes
1answer
66 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 ...
0
votes
1answer
81 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^{...
2
votes
1answer
42 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.
2
votes
2answers
110 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 ...
0
votes
1answer
125 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 ...
5
votes
6answers
202 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 ...
0
votes
0answers
16 views
Seam in between output segment in linear convolution and filter design
I am attempting to design a matching filter from long sequences. I want to do this in the time domain. I have 3 inputs. Let's called it p,s, and r. The matching filter is to be designed between p and ...
0
votes
2answers
33 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 ...
2
votes
1answer
219 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 ...
0
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1answer
45 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 :
...
-1
votes
1answer
45 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-...
1
vote
2answers
64 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 ...
0
votes
1answer
39 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 ...
0
votes
1answer
52 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 ...
