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Question: <p>Given I have coefficients a0, a1, a2, b1, and b2, defining the difference equation for a digital filter as:</p>
<p><code>y[n] = a0 * x[n] + a1 * x[n - 1] + a2 * x[n - 2] - b1 * y[n - 1] - b2 * y[n - 2]</code></p>
<p>Which defines a low-pass filter with particular cutoff frequency, how can I obtain the coef... | https://dsp.stackexchange.com/questions/69493/digital-filter-coefficients-from-low-pass-to-high-pass |
Question: <p>A digital low pass Butterworth filter that has been designed using Bi-linear transformation has been a pole at $z=0.6$. It is also known that the filter's attenuate (at digital frequency) $\omega = 1.2$ is about $44$ dB. Find the filter order. Give at least one other pole of the digital filter (in Z domain... | https://dsp.stackexchange.com/questions/23137/digital-low-pass-butterworth-filter |
Question: <p>I want to design a digital filter with the following phase response in MATLAB. i.e. at 1kHz, the phase response should be 9 degrees, at 2khz phase response should be 18 degrees ,at 3kHz phase response should be 27 degrees, at 4kHz phase response should be 36 degrees and so on upto 8kHz.How to design such f... | https://dsp.stackexchange.com/questions/76933/design-of-digital-filter-with-desired-phase-response |
Question: <p>I recently designed the LPF of the IQ demodulator using the Butterworth LPF refering to <a href="https://dspillustrations.com/pages/posts/misc/baseband-up-and-downconversion-and-iq-modulation.html" rel="nofollow noreferrer">https://dspillustrations.com/pages/posts/misc/baseband-up-and-downconversion-and-iq... | https://dsp.stackexchange.com/questions/74501/lpf-in-the-stage-of-iq-demodulator-is-it-a-analgor-filter-or-digital-filter |
Question: <p>I designed a digital filter using fdatool of matlab and obtained the filter coefficients from the tool.</p>
<p>The problem is that i designed a 4th order filter. This gave me 5 filter values </p>
<pre><code>h[] = {0.1930,0.2035,0.2071,0.2035,0.1930}
x[k] = Discrete time input signal
</code></pre>
<p>Now... | https://dsp.stackexchange.com/questions/1243/what-do-the-filter-coefficients-in-a-digital-filter-represent |
Question: <p>My objective is to build a noise shape filter from a given transfer function (in one case) and from a given PSD (for another case). Checking my precedent questions you can see that this argument is keeping me busy by long time.
You can check <a href="https://dsp.stackexchange.com/questions/52657/noise-shap... | https://dsp.stackexchange.com/questions/53226/noise-shape-digital-filter |
Question: <p>I am currently working on digital filters that can predict my input signal(assume that input signal is bandlimited). In other words, I want my filter to have a flat magnitude response in bandwidth of interest (let's say, <span class="math-container">$0$</span> to <span class="math-container">$\pi/4$</span>... | https://dsp.stackexchange.com/questions/90236/design-of-digital-filters-with-negative-group-delay |
Question: <p>I have been researching Wave Digital Filters and looking for one of the foundational papers by Alfred Fettweis (1971)</p>
<p>A. Fettweis "Digital filter structures related to classical filter networks", Archiv für Elektronik und Übertragungstechnik, 25, 79-89 (1971)</p>
<p>This is self-reference... | https://dsp.stackexchange.com/questions/95993/where-to-get-fettweis-1971-digital-filter-structures-related-to-classical-fil |
Question: <p>First the question(s):</p>
<blockquote>
<p>How should I write unit tests for a digital filter (band-pass/band-stop) in software? What should I be testing? Is there any sort of <em>canonical test suite</em> for filtering?</p>
<p>How to select test inputs, generate expected outputs, and define "conforma... | https://dsp.stackexchange.com/questions/24819/how-to-test-digital-filters |
Question: <p>Normal data acquisition consist of:</p>
<ol>
<li>Analog anti aliasing filter( Sampling frequency : $5\textrm{ kHz}$) </li>
<li>ADC - Digital Filter - (Sampling : 200K samples /sec)</li>
<li>Digital low pass filters Filters</li>
<li>DAC</li>
</ol>
<p>Questions:</p>
<ol>
<li><p>My question is why analog ... | https://dsp.stackexchange.com/questions/35562/why-analog-anti-aliasing-filter-is-used-before-analog-to-digital-converter-when |
Question: <p>I have experience with the design of FIR, IIR digital filters. I also know about the Kalman filter, but I am not skilled at using them. Consider the case of a low frequency signal from discrete samples and the signal is corrupted by high frequency noise. It seems a digital low pass filter and a Kalman filt... | https://dsp.stackexchange.com/questions/25518/digital-low-pass-filter-vs-kalman-filter |
Question: <p>In one system there is a maximum sampling limit of 500 Hz. And in the analog signal, there are waves with a frequency in the range up to 1600 Hz. With 500 Hz sampling, is it possible to remove frequencies higher than 200 Hz using a digital filter so that the aliasing does not occur?
Or is an analog low-pas... | https://dsp.stackexchange.com/questions/93973/digital-filter-response-at-frequencies-higher-than-the-nyquist-frequency |
Question: <p><strong>Preambule:</strong><br>
I'm designing a sound model for my small submarine game. Model is running on the server, and I want to present the client with a mono-channel wav-stream from his hydrophone (20kHz discretization should suffice, I target 20Hz-10kHz band). I want that signal to be relatively-r... | https://dsp.stackexchange.com/questions/47454/digital-filter-simulating-hydroacoustic-signal-distortion |
Question: <p>I'm using digital filters to apply spectral mangling-type special effects to audio.</p>
<p>When using a digital filter (vsts/standalone DSP programs/outboard digital filter, etc.), especially when using narrow transition bands/brickwall filters, are there any effective ways to remove ringing artifacts int... | https://dsp.stackexchange.com/questions/2182/what-methods-can-be-used-to-remove-ringing-artifacts-in-the-output-of-a-digital |
Question: <p>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. </p>
<p>What is the direct equation or algorithm to implement such a filter?</p>
Answer: <p>okay, this is, or can be, stuff straight outa a textbook. by "SOS", you mean "2... | https://dsp.stackexchange.com/questions/13204/algorithm-for-implementing-an-iir-digital-filter-chebyshev-type-i-low-pass |
Question: <p>Can oversampling decrease the delay of digital IIR filter? Imagine there is some digital signal going into processor that applies low pass filter.Lets say its 1 KHz sample rate and the filter is second order gaussian lowpass with -3db point at 100 Hz.</p>
<p>The putput of this digital filter will be delay... | https://dsp.stackexchange.com/questions/51498/digital-iir-filter-delay-and-oversampling |
Question: <p>I'm trying to implement a digital filter, which is given by the following transfer function:</p>
<p><span class="math-container">$$
1+2V K \frac{K+c_m+2Kz^{-1}+(K-c_m)z^{-2}}{1+2Kc_m+K^2+(2K^2-2)z^{-1}+(1-2Kc_m+K^2)z^{-2}}
$$</span>
<span class="math-container">$$
+V^2K^2\frac{1+2z^{-1}+z^{-2}}{1+2Kc_m+K^... | https://dsp.stackexchange.com/questions/70912/help-implementing-digital-filter-from-tansfer-function |
Question: <p>I'm currently attempting to study up on adaptive digital filters. My book presents the diagram I've included below and I'm having trouble understanding conceptually what it's indicating. The problem deals with noise cancelation. The idea is that someone is driving and makes a phone call. The <em>x(k)</em> ... | https://dsp.stackexchange.com/questions/22325/adaptive-digital-filter-block-diagram-question |
Question: <p><strong>Short question</strong><br>
What are main stages (steps) of calculation <a href="http://en.wikipedia.org/wiki/Frequency_response" rel="nofollow noreferrer">frequency response</a> of digital filter by their structure?</p>
<p><strong>Detailed question</strong><br>
Let suppose that there is discrete... | https://dsp.stackexchange.com/questions/21806/calculation-frequency-response-of-digital-filter-with-known-structure |
Question: <p>I am having trouble wrapping my head around digital filters with different orders of numerator and denominator. Let me know if any of these points is wrong:</p>
<ol>
<li>All (digital or analog) transfer functions have the same number of poles and zeros, <em>if</em> you include the ones at infinity. So $... | https://dsp.stackexchange.com/questions/14739/digital-filters-with-more-zeros-than-poles |
Question: <p>Hi i'm a beginner in signal processing i want to know what'sthe pass band ripple and stop band attenuation of a digital filter ?
Thanks.</p>
Answer: <p>I hope the plot below helps answer your question. Typically I have seen the "passband ripple" and "stopband attenuation" expressed in... | https://dsp.stackexchange.com/questions/38564/whats-the-pass-band-ripple-and-stop-band-attenuation-of-a-digital-filter |
Question: <p>Suppose I have a signal <span class="math-container">$\mathbf{x}\in \mathbb{C}^{N}$</span> and a digital filter with impulse response <span class="math-container">$\mathbf{h}\in\mathbb{C}^L$</span>, where <span class="math-container">$L<N$</span>. If we pass the signal through the filter, the output wil... | https://dsp.stackexchange.com/questions/89904/truncating-the-output-of-a-digital-filter-which-part-to-discard |
Question: <p>Although I have a solid experience in designing audio engines and such, I am fairly new to the realm of Digital Filter Design, particularly IIR and FIR filters. In other words, I'm trying to learn as much as I can on how to design filters and derive their difference equations. I'm starting from the basics,... | https://dsp.stackexchange.com/questions/9541/digital-filter-design-basic-principles-iir-fir |
Question: <p>I have a second order analogue high pass transfer function (unity gain at infinity). It's magnitude response hits the -20 decibel line at a frequency of 5706 Hz (the corner frequency is 18000 and sample rate is 44100). When I convert this analogue filter to a digital IIR via the BLT method, the digital fil... | https://dsp.stackexchange.com/questions/22071/how-to-predict-the-cramped-frequency-of-a-digital-filter-based-on-an-analogue-fr |
Question: <p>I am looking into designing a Bandpass Butterworth filter in python, but, I was not sure I am designing my filter correctly. What I have are the following:</p>
<ul>
<li>High cutoff frequency = 200Hz</li>
<li>Low cutoff frequency = 10Hz</li>
<li>Sampling frequency = 1000Hz</li>
<li>for my data, I used Filte... | https://dsp.stackexchange.com/questions/79394/how-to-design-a-digital-butterworth-bandpass-filter |
Question: <p>I'm trying to understand how diode circuits are implemented in wave digital filters, particularly for clippers. The research papers and other sources I've looked at use the equation</p>
<p><span class="math-container">$$I(V) = 2I_s \sinh\left(\frac{V}{V_t}\right)$$</span></p>
<p>for two reverse-polarity di... | https://dsp.stackexchange.com/questions/73135/wave-digital-filter-diode-equation |
Question: <p>Doing some work at the minute on digital filters in matlab, I have a file with artifical noise added (sine wave added at specific frequency). The goal is to filter the signal and get it as close as possible to the clean signal provided.</p>
<p>I've done an FFT and plotted the results and found a very larg... | https://dsp.stackexchange.com/questions/36041/digital-filter-not-removing-noise-at-specific-frequency-matlab |
Question: <p>In a <a href="https://dsp.stackexchange.com/questions/70960/given-a-3db-octave-filter-that-makes-pink-noise-how-can-i-make-a-3db-octave">related question</a> a probable solution was given to build a first-order digital filter and then cascade three of them in order to turn white noise into pink. I have ap... | https://dsp.stackexchange.com/questions/70969/cascading-first-order-digital-filters-in-c |
Question: <p>I am reading Introduction to Digital Filters by J.O Smith III, which is an amazing book. The part for which I have a question is quoted below.</p>
<blockquote>
<p>By virtue of Euler's relation and the linearity of the
filter, setting the input to <span class="math-container">$ x(n) = e^{j\omega nT}$</span>... | https://dsp.stackexchange.com/questions/80740/issue-understanding-implementing-digital-filters-in-practice |
Question: <p>Suppose I have a digital filter implemented in Direct Form II. How do I initialize the state of the filter as if the input <span class="math-container">$x[n]$</span> had a fixed value <span class="math-container">$x_0$</span> for all <span class="math-container">$n<0$</span>?</p>
<p><a href="https://i.... | https://dsp.stackexchange.com/questions/56476/how-do-i-initialize-the-state-of-a-digital-filter-in-direct-form-ii |
Question: <p>I can't find this answer anywhere. I have a couple satellite modem manuals and they refer to digital filtering functions that they do, but they say almost nothing about their sample rate. I always thought, without considering it too much, that all the modems I've worked with were only sampling at a rate ... | https://dsp.stackexchange.com/questions/93664/sample-rate-of-digital-modems-how-do-they-do-digital-filtering-if-sampling-belo |
Question: <p>I am trying to implement a digital filter over a uC (it doesn't really matter which filter and which micro controller because I'm looking forward to learn how to do it in the future with different filters and different microcontrollers).
I've been told that you can design, implement and debug a digital fil... | https://dsp.stackexchange.com/questions/59688/how-to-design-a-digital-filter-in-python-that-will-run-over-an-uc |
Question: <p>I am currently working with vibration measurements in structures. In the netherlands there is a guideline for verifying vibration measurements for damage to machinery. This is the so-called "SBR Trillingsrichtlijn". In this guideline a frequency weighting function is specified to modify the time series fro... | https://dsp.stackexchange.com/questions/55303/digital-filter-design-of-time-series-for-specified-frequency-response-function |
Question: <p>In 'Digital Filters' by Hamming there is a cryptic section where he describes how the Gibbs phenomenon can be viewed as the displacement between the centers of two functions as they are convolved together. This is on pages 112 - 113 of the 3rd edition.</p>
<p>In the process of this he shows that truncatin... | https://dsp.stackexchange.com/questions/7605/gibbs-phenomenon-in-hammings-digital-filters |
Question: <p>(I am trying to create an IIR audio filter that adds reverb to an initial sample)</p>
<p>Say I designed an analog filter to model acoustic attenuation based on the following mathematical model:</p>
<p><span class="math-container">$$
I = I_0 e^{pt},
$$</span></p>
<p>Where <span class="math-container">$p$... | https://dsp.stackexchange.com/questions/53669/how-do-i-add-more-resolution-taps-to-an-analog-digital-filter |
Question: <p>Please elaborate on why this mathematical transform can help analyzing as well as designing any type of digital filter.</p>
Answer: <p>The Z Transform is to discrete-time (digital) signals precisely the same role that the Laplace Transform is to continuous-time (analog) signals.</p>
<p>Linear Time-Invari... | https://dsp.stackexchange.com/questions/55043/why-is-the-z-transform-so-important-in-digital-filters-analysis-and-design |
Question: <p>In Digital Filter Design by Parks and Burrus, p. 19.</p>
<hr>
<p>The transfer function of an FIR filter is given by the $\mathcal Z$-transform of $h(n)$ as:</p>
<p>$$H(z)=\sum_{n=0}^{N-1}h(n)z^{-n}$$</p>
<p>(where $h$ is the filter)</p>
<p>The frequency response of a filter is defined as</p>
<p>$$H(\... | https://dsp.stackexchange.com/questions/31155/periodicity-of-transfer-function-of-fir-filter-proof-parks-and-burrus-digital |
Question: <p>Practical <a href="https://en.wikipedia.org/wiki/Infinite_impulse_response" rel="nofollow noreferrer">infinite impulse response</a> (IIR) filters are usually based upon analogue equivalents (Butterworth, Chebyshev, etc.) using a transformation known as the <a href="https://en.wikipedia.org/wiki/Bilinear_tr... | https://dsp.stackexchange.com/questions/72729/how-to-design-iir-digital-filters |
Question: <p>As far as I have seen, almost all theoretical filter design occurs in Laplace or Z-space. Also, there is a pervasive connection to real life analog filters in the design. If one is just thinking in a mathematical theoretical thing (or something that could be implemented digitally), why wouldn't one filter ... | https://dsp.stackexchange.com/questions/70754/why-is-fourier-space-not-adequate-for-theoretical-or-digital-filters |
Question: <p>I have the numerator and denominator of a lowpass digital elliptic filter. I know how to create a minimum-phase filter with the same magnitude response using cepstrum technique. But I came across <a href="https://www.dsprelated.com/freebooks/filters/Linear_Phase_Really_Ideal.html" rel="nofollow noreferrer"... | https://dsp.stackexchange.com/questions/94048/create-a-minimum-phase-filter-from-an-elliptic-digital-filter |
Question: <p>I want to A-weight a time series with arbitrary sample rate. </p>
<p>An analog A-weighting filter is defined exactly by IEC 61672-1. But there's no definition for a digital filter. One method is to use the bilinear transform (BLT) to convert the analog filter to the digital filter (as done here <a href="h... | https://dsp.stackexchange.com/questions/36077/design-of-a-digital-a-weighting-filter-with-arbitrary-sample-rate |
Question: <p>I have read some articles on Allan deviation and understand that the slope of the <span class="math-container">$\sigma(\tau)$</span> diagram corresponds to the exponent of power-law noise:</p>
<p><span class="math-container">$$S_y(f)\sim f^\alpha \implies \sigma(\tau) \sim \tau ^{-\frac{\alpha + 1}{2}}$$</... | https://dsp.stackexchange.com/questions/97781/designing-digital-filters-on-basis-of-sigma-tau-diagrams |
Question: <p>I'm trying to self learn the art of signal processing whilst moving through my third year pure maths degree. </p>
<p>Sorry if my terminology is incorrect however I hope I am understandable!</p>
<p>I am looking at data which is coming from an accelerometer, distance data from a separate sensor and time da... | https://dsp.stackexchange.com/questions/52920/digital-or-analogue-filtering |
Question: <p>I'm trying to implement a digital filter that has the frequency response shape equal to the image below:</p>
<p><a href="https://i.sstatic.net/rYXJ2.png" rel="nofollow noreferrer"><img src="https://i.sstatic.net/rYXJ2.png" alt="enter image description here" /></a></p>
<p>Where i will use equation (11) to i... | https://dsp.stackexchange.com/questions/85733/trying-to-implement-a-digital-a-frequency-filter |
Question: <p>I have posted this question "Electrical Engineering", but this seems a more appropiate place. I am trying to model a bireciprocal Cauer filter in LTspice but I don't get the expected results. More precisely, using this formula for the coefficients</p>
<p><span class="math-container">$\gamma=\frac... | https://dsp.stackexchange.com/questions/15112/bireciprocal-lattice-wave-digital-filter |
Question: <p>Consider a set of dense, but generally irregularly-spaced frequency response measurements of some real low-pass analog filter. Denote the maximum frequency for which a frequency response measurement is available as F_a_max.</p>
<p>I would like to create a digital filter model for this analog filter. The... | https://dsp.stackexchange.com/questions/64207/obtaining-a-high-sample-rate-digital-approximation-to-an-analog-filter-from-lowe |
Question: <p>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 $f_s=1000\text{Hz}$.</p>
<p>Applying the bilinear transform in Matlab however appears to yield a filter with the a different 3... | https://dsp.stackexchange.com/questions/11345/digital-implementation-of-first-order-analog-filter-using-bilinear-transformatio |
Question: <p>I have an analog filter with its frequency response curve in dB described by the following expression:
$$
N_{dB}=20log_{10}\omega t_1 \sqrt{\frac{1+(\omega t_2)^2}{1+(\omega t_1)^2}}
$$
This expression is derived from the series connection of two lowpass filters each associated with the following RC circui... | https://dsp.stackexchange.com/questions/15134/approximate-the-magnitude-response-of-an-analog-filter-with-a-digital-filter-st |
Question: <p>I am trying to design a digital ButterWorth filter for the given specifications.</p>
<pre><code>rp=3;
rs=15;
FS=1;
wp=0.5*pi;
ws=0.75*pi;
pwp=2*FS*tan(wp/2);
pws=2*FS*tan(ws/2);
[n,wn]=buttord(pwp,pws,rp,rs,'s')
[b,a]=butter(n,wn,'s');
[bn,an]=bilinear(b,a,FS);%error
freqz(bn,an,512,FS);
</code></pre>
<p>I... | https://dsp.stackexchange.com/questions/19279/digital-butterworth-filter-design-error |
Question: <p>A reviewer has asked me to re-filter my data to remove baseline drift.</p>
<p>Each sweep is 170ms sampled at 1000Hz and the reviewer wants it high-pass filtered at 0.5Hz. The original bandpass settings on the hardware were 0.15 to 1000Hz</p>
<p>Although this is easy to code in matlab, the epoch is very ... | https://dsp.stackexchange.com/questions/17246/digital-filter-and-epoch-length |
Question: <p>I found this digital filter in code I am working on. It is a low pass filter. In the code it is called an "alpha filter", but it is not the same as the <a href="https://en.wikipedia.org/wiki/Alpha_beta_filter#Alpha_filter" rel="nofollow noreferrer">alpha filter mentioned here</a>.</p>
<p>I post the releva... | https://dsp.stackexchange.com/questions/46278/what-is-the-name-of-this-digital-low-pass-filter |
Question: <p>I have an <a href="https://github.com/bcrowell/kcals" rel="nofollow noreferrer">open-source software project</a> whose purpose is to analyze a GPS track, or a similar track made by an application such as google maps, and estimate the physical exertion required to hike or run that route. Traditionally, peop... | https://dsp.stackexchange.com/questions/37197/appropriate-digital-filter-for-gps-tracks |
Question: <p>I am working on the demodulation of digital signals, I am following <a href="https://pysdr.org/_images/sync-diagram.svg" rel="nofollow noreferrer">this</a> block diagram.</p>
<p><img src="https://pysdr.org/_images/sync-diagram.svg" alt="block diagram"><br>
<sup>Source: <a href="https://pysdr.org/content/sy... | https://dsp.stackexchange.com/questions/96513/matched-filter-in-digital-demod |
Question: <p>I'm studying the IIR filter design that is described in the book: <a href="http://www.nt.tuwien.ac.at/fileadmin/users/gerhard/diss_Lang.pdf" rel="nofollow noreferrer">Algorithms for the constrained design of digital filters with arbitrary phase and magnitude responses</a>. </p>
<p>You can get the code at ... | https://dsp.stackexchange.com/questions/10455/least-squares-digital-iir-filter-design-with-arbitrary-responses |
Question: <p>In the process of applying a lowpass Bessel filter to my signal, I realized that besself function does not support the design of digital Bessel filters and the bilinear function can be used to convert an analog filter into a digital form, except for Bessel filters. The digital equivalent for Bessel filters... | https://dsp.stackexchange.com/questions/82411/matlab-how-to-design-digital-equivalent-for-a-lowpass-bessel-filter-thiran-fil |
Question: <p>I'm pretty well versed in statistics, but not really digital signal filtering. I have a data scenario where I expected to be able to pretty easily filter out some noise (human pulse) that's at a known frequency band, but I'm having a lot of trouble using the standard tools in the scipy.signal library and t... | https://dsp.stackexchange.com/questions/69643/advice-on-designing-a-digital-filter-that-doesnt-have-phase-sensitive-edge-arti |
Question: <p>I'm interested in composing filters for realtime audio processing on an microcontroller (MCU). Ideal frequency response is unity as a default, with deviations up and down at specific freq-domain pointers according scalers, and some type of smooth transition between these points. This is conceptually simila... | https://dsp.stackexchange.com/questions/79091/composing-digital-filters |
Question: <p>I'm trying to create a digital filter in code(C) but any language is fine. Now I've got an analogue filter that I have represented by an equation in the Laplace domain and I want to try and implement it digitally. </p>
<p>So my filter has this form in the Laplace domain:
$$\frac{as+b}{cs^2+ds}$$</p>
<p>I... | https://dsp.stackexchange.com/questions/18329/creating-a-digital-filter-from-laplace-to-mathcal-z-transform-zero-order-ho |
Question: <p>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 Hz and 40-45 Hz transition bands. Usually, I design the filter with the selection "least-squares", which, i... | https://dsp.stackexchange.com/questions/10056/about-designing-digital-filters |
Question: <p>@Jazzmaniac has a good answer to the question of how to design an alias-free digital nonlinear time-invariant filter here: <a href="https://dsp.stackexchange.com/a/28787/18276">https://dsp.stackexchange.com/a/28787/18276</a></p>
<p>Basically, according to that answer, a digital nonlinear time-invariant fi... | https://dsp.stackexchange.com/questions/51533/alias-free-digital-nonlinear-filter-design |
Question: <p>I am working in something were I should use a upsampling filter. I have decided to use a Nyquist filter(Lth filter). I know that there are two constraints. The first The frequency vector values must mirror each other in pairs around $\pi/2$. The second is the amplitude vector values must mirror each other ... | https://dsp.stackexchange.com/questions/29386/nyquist-nth-digital-filters |
Question: <p>I want a highly damped highpass filter (damping of at least $2$), with a cutoff somewhere around $1\textrm{ Hz}$ ($51.2\textrm{ Hz}\quad f_s$)</p>
<ul>
<li>How do i go about designing the filter with adequate control over the damping?</li>
</ul>
<p>My best guess was to use a standard second order respons... | https://dsp.stackexchange.com/questions/37089/highly-damped-iir-fir-digital-filter |
Question: <p>In context of transition width vs. stopband attenuation, different windows (Blackman, Hamming, etc) are compared in terms of <em>tradeoffs</em> between the two, always noting that one cannot perfect both.</p>
<p>Why not? Make it long enough - problem solved. We're working with <em>finite</em> frequency and... | https://dsp.stackexchange.com/questions/70462/is-there-really-no-perfect-digital-filter |
Question: <p>It would be greatly appreciated if the usage of the python package scipy's filter (e.g. butter) analog=True argument could be explained. I don't understand what is meant by this (any signal being processed by scipy in python on a computer is discrete and will always be digital?). I an pretty familiar with ... | https://dsp.stackexchange.com/questions/87005/scipy-filter-analog-vs-digital |
Question: <p>I am working on a board that has no antialisaing filter at the input of the ADC. I have option to I implement my own filter using RC + Opamp circuit. But is it also possible to implement Anti Aliasing filter after sampling by ADC and processing in Digital domain: a digital Anti aliasing filter? </p>
A... | https://dsp.stackexchange.com/questions/9205/can-we-have-a-digital-anti-aliasing-filter |
Question: <p>I came across this paper entitled "Design of Efficient Digital Interpolation Filters and Sigma-Delta Modulator for Audio DAC" where the author oversamples an input frequency, fsig = 1kHz with ratio L = 128 and update frequency, fsi = 64kHz. The interpolation filter specification is given by:</p>
<ul>
<li>... | https://dsp.stackexchange.com/questions/59929/design-of-efficient-digital-interpolation-filter |
Question: <p>I am using an adaptive RLS adaptive filter for noise cancellation. My sampling freq. is 500 Hz, but I am interested in only frequencies of up to 60 Hz. I filter the input and the reference signal to the desired frequency range and then apply the adaptive filter. The adaptive filter does a good job at remov... | https://dsp.stackexchange.com/questions/10017/rls-adaptive-filter |
Question: <p>How often do problems arise that let you use adaptive filters? Unless I am understanding something incorrectly it seems the requirement that the input signal be stationary(or even WSS) is too strong for most places I would want to use adaptive filters.</p>
<p>Am I wrong? How often do adaptive filters come... | https://dsp.stackexchange.com/questions/52380/how-general-are-adaptive-filtering-techniques |
Question: <p>I'm trying to code the algorithm described in <a href="https://www.microsoft.com/en-us/research/wp-content/uploads/2016/02/ICASSP01makurt.pdf" rel="nofollow noreferrer">Speech dereverbaration via maximum-kurtosis subband adaptive filtering</a> by <em>Gillespie, Malvar and Florencio</em>, and the signal loo... | https://dsp.stackexchange.com/questions/45344/speech-dereverbaration-via-maximum-kurtosis-adaptive-filtering |
Question: <p>I am using a series-cascade of multiple NLMS adaptive filters each with step size 0.0040, leakage factor 1.0, and 100 filter taps. My signal gains magnitude at each step of the filtering, say the peak magnitude increases from 0.2 originally to 2.5 after using the first adaptive filter to 12.5 after the usi... | https://dsp.stackexchange.com/questions/65901/why-does-my-signal-magnitude-increase-after-adaptive-filtering |
Question: <p>I have designed an adaptive filter for noise cancellation. Is there any standard way of testing adaptive filters?</p>
Answer: <p>It is usually evaluated using the Mean Square Error:</p>
<p>$$ e(n) = \frac{\displaystyle\sum_{i=1}^{N}(d_{i}(n) - y_{i}(n))^2}{N} $$</p>
<p>Where $ d(n) $ are the values of t... | https://dsp.stackexchange.com/questions/28054/performance-of-adaptive-filter |
Question: <p>When studying neural networks from Neural Networks and Learning Machines, by Simon Haykin, the author highlights the close similarity between of adaptive filtering and neural networks.</p>
<p>From a scalar-valued signal, if we put a tapped delay input along with an ADALINE (Adaptive Linear Neuron), do we h... | https://dsp.stackexchange.com/questions/87750/tapped-delay-line-adaline-adaptive-filter |
Question: <p>In the audio domain, I am currently attempting to use MATLAB to distil:</p>
<ol>
<li><p>$\textrm{signal}$ from $\textrm{noise + signal}$</p></li>
<li><p>$\textrm{noise}$ from $\textrm{noise + signal}$
using two adaptive filters $\rightarrow$ two results. </p></li>
</ol>
<p>I get the first answer quite ... | https://dsp.stackexchange.com/questions/28672/matlab-adaptive-filters |
Question: <p>I'm trying to find the optimum filter length for an Adaptive Filtering, using RLS Algorithm.</p>
<p>I'm using this design:
<a href="https://i.sstatic.net/UG9w9.png" rel="noreferrer"><img src="https://i.sstatic.net/UG9w9.png" alt=""></a></p>
<p>So the "error" signal is the signal without noise (and that's... | https://dsp.stackexchange.com/questions/37902/adaptive-filtering-optimum-filter-length-and-delay |
Question: <p>Does anyone know about different adaptive filtering implementations (LMS, RLS ...) in 2D or even 3D ? I have sequences of 2D images and 3D volumes with repeating patterns but small differences. I was thinking of using one as my reference input and extract differences between the pair (A simple subtraction ... | https://dsp.stackexchange.com/questions/10482/2d-adaptive-filters |
Question: <p>I am confused as to the difference between neural networks and adaptive filters: As far as I understand it, "neural networks" are largely used for solving inverse problems, where an unknown system is to be identified by the neural network in order to, for example, predict some output. The same is... | https://dsp.stackexchange.com/questions/78687/is-a-neural-network-an-adaptive-filter |
Question: <p>Can somebody please provide an intuitive answer or reference for the following questions?</p>
<p><strong>Q1: Dependence of estimation performance on number of data points</strong> -- I could not find any information whether the estimation performance of Adaptive filters such as Least Mean Square (LMS), Co... | https://dsp.stackexchange.com/questions/43255/performance-of-adaptive-filters |
Question: <p>I have general theoretical questions: </p>
<ul>
<li>Is it true that an adaptive filter with two inputs (one normal and one delayed by the single time increment) can completely get rid of any <strong>single frequency</strong> noise? </li>
<li>Is it then true that a three-input adaptive filter can (complete... | https://dsp.stackexchange.com/questions/31608/adaptive-filter-with-two-inputs |
Question: <p>I just have a question about using an least-mean-squares algorithim adaptive filter for system identification. Consider the following
<a href="https://i.sstatic.net/homVX.png" rel="nofollow noreferrer"><img src="https://i.sstatic.net/homVX.png" alt="enter image description here"></a></p>
<p>I am told that... | https://dsp.stackexchange.com/questions/53924/system-identification-using-lms-adaptive-filter |
Question: <p>The quadratic performance surface of an adaptive filter is a paraboloid. Its minimum can be found wherever the gradient is zero. However, since there are two types of paraboloids (elliptical and hyperbolic), is there a way to tell if the minimum detected is a global minimum or just a saddle point?</p>
Ans... | https://dsp.stackexchange.com/questions/23143/adaptive-filter-gradient-descent |
Question: <p>I have 3 sensor inputs: $a(t)$, $b(t)$ and $c(t)$. I want to design a filter such that the weighted linear combination of the three is always a constant. Kind of like:</p>
<p>$$w_1(t)a(t) + w_2(t)b(t) + w_3(t)c(t) = k$$</p>
<p>So from my undergrad modules I think I need a adaptive filters. I can perform ... | https://dsp.stackexchange.com/questions/17617/adaptive-filter-weight-adjustment |
Question: <p>I studied a bit about adaptive filter on internet and found that its a special filter which keep on updating its filter value as soon as it proceeds. It finds out the difference between input and output and using the error function and previous coefficients finds out the new filter coefficients.</p>
<p>Bu... | https://dsp.stackexchange.com/questions/1572/what-does-an-adaptive-filter-do |
Question: <p>I want to mention upfront that I'm not very experienced in this field.</p>
<p>I have a signal <span class="math-container">$u(k)$</span> that I get from a black box simulation (sampled irregularly). The signal looks like this:</p>
<p><a href="https://i.sstatic.net/miL0y.png" rel="nofollow noreferrer"><img ... | https://dsp.stackexchange.com/questions/88750/adaptive-filtering |
Question: <p>I'm having some confusion learning about the LMS Adaptive Filter. I understand that the whole model of adaptive filters relies on the fact that we give it a reference signal to which it keeps comparing the input * filter with and the filter coefficients keep changing until the error between input and refer... | https://dsp.stackexchange.com/questions/53427/modelling-unwanted-signal-in-a-lms-adaptive-filter |
Question: <p>I got a problem when I was trying to denoise a signal. Actually, it is a simple simulation. The signal is the addition of a step signal (The info I wish to get) and a sinusoidal one (the noise I wish to remove). See below<img src="https://i.sstatic.net/NcyNr.jpg" alt="(a) The noise (b) The signal and (c) S... | https://dsp.stackexchange.com/questions/23229/why-adaptive-filter-does-not-work-in-my-application |
Question: <p>I’m trying to understand the conception of function <a href="http://www.mathworks.com/help/dsp/ref/maxstep.html" rel="nofollow noreferrer">maxstep</a> </p>
<p>The foundation of this function is function <code>firwiener</code> with input parameters: length of adaptive filter, samples of input signal, whi... | https://dsp.stackexchange.com/questions/3554/maximum-step-size-for-adaptive-filter-convergence |
Question: <p>I'm trying to understand how to specify the "desired signal" in adaptive LMS filters such as the following: <a href="http://www.mrtc.mdh.se/projects/wcet/wcet_bench/lms/lms.c" rel="nofollow noreferrer">This one</a>, or <a href="http://read.pudn.com/downloads158/ebook/707037/10%20DSP%20applications%20using%... | https://dsp.stackexchange.com/questions/31892/desired-signal-in-lms-adaptive-filters |
Question: <p>What is the advantage of Variable step size LMS over Leaky-LMS adaptive filter algorithm? Which one has a better performance?</p>
Answer: <p>Variable step size LMS is generally used to improve the speed of convergence or decrease steady-state error.
Leaky adaptation is used to combat problems like the pot... | https://dsp.stackexchange.com/questions/36664/variable-step-size-lms-vs-leaky-lms-adaptive-filter-algorithm |
Question: <p>I want to create an adaptive filter. Its coefficients have this general shape:</p>
<p><img src="https://i.sstatic.net/ZeXvs.jpg" alt="enter image description here"></p>
<p>When the input signal for the filter is a sine wave, the filter behaves the desired way if the look-back window is set to a length eq... | https://dsp.stackexchange.com/questions/21677/non-standard-error-function-for-adaptive-filter |
Question: <p>I am working on a project which requires me to implement adaptive filter as a predictor.
I have just started on adaptive filter and I intend to use least mean square algorithm for weight adjustment.</p>
<p>How can I predict future values from this system ?</p>
<p>Any help would be beneficial for me.
Than... | https://dsp.stackexchange.com/questions/54955/can-temperature-data-be-predicted-using-adaptive-filter-such-as-lms-algorithm |
Question: <p>I run many times in equations containing the trace of covariance matrix of an adaptive filter input. But it is not really clear what it is.</p>
<p>For example in <a href="http://ieeexplore.ieee.org/stamp/stamp.jsp?arnumber=902113" rel="nofollow noreferrer">this paper</a> the input covariance matrix is</p>... | https://dsp.stackexchange.com/questions/38161/covariance-matrix-of-an-adaptive-filter-input |
Question: <p>I am trying to make a frequency domain adaptive filter in matlab. It uses matlab adaptfilt.fdaf to create the filter parameters like step size and initializing initial filter weight values. Then I have tried to implement the overlap - save frequency domain adaaptive filter algorithm from the paper "<a href... | https://dsp.stackexchange.com/questions/9529/adaptive-filter-does-not-converge-for-all-inputs |
Question: <p>I have got review for my work saying, that my work (adaptive filter variant) should be analyzed in transient and steady-state before claiming it improves performance.</p>
<p>I have done common (in my opinion) analysis:</p>
<ol>
<li>prediction of linear system</li>
<li>prediction of non-linear system</li>... | https://dsp.stackexchange.com/questions/37100/transient-and-steady-state-analysis-for-adaptive-filter |
Question: <p>For adaptive filtering, both finite and infinite impulse response (FIR/IIR) filters can be utilized. As an advantage of FIR filters in this context, guaranteed stability is often mentioned, while IIR filters do not share this property (see <a href="https://dsp.stackexchange.com/questions/32129/whats-the-ad... | https://dsp.stackexchange.com/questions/61159/why-can-adaptive-iir-filters-result-in-unstable-solutions |
Question: <p>I'm trying to filter some motion noise from an ECG signal. To do that I'll try to implement an adaptive filter using the LMS algorithm.</p>
<p>I think I have to calculate the MSE of this:</p>
<p><code>E[e^2 ] = E[(s + n )^2 ]+ 2E[(s + n)X ]W^T + WE[X^T X ]W^T</code></p>
<p>in which <code>s+n</code> is t... | https://dsp.stackexchange.com/questions/9784/mse-in-adaptative-filter |
Question: <p>I'm looking to implement a feedback cancellation filter using Wiener Filtering, where an adaptive Wiener filter is used to cancel the feedback occurring in the path between a loudspeaker and a mic (assume PA system). The idea is essentially from this paper: </p>
<blockquote>
<p>Spriet, Ann, et al. "Adap... | https://dsp.stackexchange.com/questions/58956/why-is-white-noise-so-important-in-system-identification-or-adaptive-filters |
Question: <p>Are least square filters, or filters that minimize error energy, the same as least mean square adaptive filters?</p>
Answer: <p><strong>TL;DR:</strong> No, they are not necessarily the same.</p>
<hr>
<p><strong>Gory Details</strong></p>
<p>Least squares is just an optimization technique. It is used in ... | https://dsp.stackexchange.com/questions/42192/are-all-least-square-filters-adaptive |
Question: <p>In <a href="http://ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=1261952&tag=1" rel="nofollow">this paper</a> stands:</p>
<blockquote>
<p>The derivation and analysis of NLMS rest upon the usual independence assumptions.</p>
</blockquote>
<p>It has a footnote:</p>
<blockquote>
<p>The independence... | https://dsp.stackexchange.com/questions/33771/what-is-usual-independence-assumptions-on-adaptive-filters |
Question: <p>What does it mean by leakage in case of digital filters? My specific question is about the frequency domain adaptive filter function provided in the Matlab DSP toolkit, accessed as adaptfilt.fdaf. It has a parameter called LEAKAGE, but I am not sure what exactly does it represent or how it affects the filt... | https://dsp.stackexchange.com/questions/9441/what-is-leakage-in-frequency-domain-adaptive-filters |
Question: <p>Note: This post was made to aid with adaptive equalizer design.</p>
<p>Adapting an FIR filter using algorithms like LMS, RLS, etc... will generally result in updates that are non-symmetric and therefore non-linear phase.</p>
<p>When the adaptive FIR filter taps are linear phase, one can synchronize a "... | https://dsp.stackexchange.com/questions/95984/training-a-non-linear-phase-adaptive-filter |
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