text stringlengths 81 47k | source stringlengths 59 147 |
|---|---|
Question: <p>I'm using two microcontrollers to implement an adaptive LMS filter to filter out the noise from the signal. One is recording the noise and streaming the data to the other microcontroller which is recording the (noised) signal and using the noise reference, filtering it out.</p>
<p>How does the delay betwe... | https://dsp.stackexchange.com/questions/26664/lms-adaptive-filter-relatively-delayed-signal-and-reference-inputs |
Question: <p>The MATLAB code below is for equalizer using lms algorithm adaptive filter and then plotting MSE (Mean Square Error) Vs Iteration numbers</p>
<hr />
<pre><code>%% Channel Equalization using Least Mean Square (LMS) algorithm
% Author: SHUJAAT KHAN
clc;clear all;close all;
%% Channel and noise level
h = [0.9... | https://dsp.stackexchange.com/questions/72047/matlab-model-for-equalizer-using-lms-algorithm-adaptive-filter-and-unexpected-ou |
Question: <p>How would the LMS equalizer dimensions change for the MISO case?</p>
<p>LMS adaptive filters are typically described for equalizing a single input signal, <span class="math-container">$x(t)$</span>. Can the LMS algorithm be modified in the MISO case to perform diversity combining when the transmitter has <... | https://dsp.stackexchange.com/questions/84076/can-a-lms-adaptive-filter-be-adapted-for-miso |
Question: <p>The algorithms given for <a href="https://en.wikipedia.org/wiki/Recursive_least_squares_filter#Lattice_recursive_least_squares_filter_(LRLS)" rel="nofollow noreferrer">un-normalized LRLS</a> and <a href="https://en.wikipedia.org/wiki/Recursive_least_squares_filter#Normalized_lattice_recursive_least_squares... | https://dsp.stackexchange.com/questions/61544/updating-forgetting-factor-in-un-normalized-lattice-recursive-least-squares-adap |
Question: <p>One way of separating downgoing and upgoing wavefields in offshore seimic processing is to add signals from hydrophone and vertical component of the geophone (they are co-located). Hydrophone only registers a change in the pressure whereas geohphone as well as registering a change in seismic field also rea... | https://dsp.stackexchange.com/questions/49221/adaptive-filter-to-scale-and-phase-shift-two-sensors-output |
Question: <p>This wikipedia page <a href="https://en.wikipedia.org/wiki/Recursive_least_squares_filter" rel="nofollow noreferrer">https://en.wikipedia.org/wiki/Recursive_least_squares_filter</a> (and in fact other sources) do not explain the apparent paradox of the cost function that computes the MSE of the output and ... | https://dsp.stackexchange.com/questions/94203/rls-adaptive-filter-intuitive-explanation-of-the-so-called-desired-signal |
Question: <p>I am working with adaptive filters and similar adaptive models (mainly with gradient adaptation) for a few years. I and my colleagues always struggle to find out the correct size of regression vector.</p>
<p>So far, as I find out by the hard way (experience):</p>
<ol>
<li><p>for filtration mostly works w... | https://dsp.stackexchange.com/questions/37855/regression-vector-size-for-prediction-reconstruction-and-filtration-with-adapti |
Question: <p>I am currently reading a chapter about adaptive filters from the <a href="https://link.springer.com/book/10.1007/978-3-540-49127-9" rel="nofollow noreferrer">Springer Handbook of Speech Processing</a>. </p>
<p>In a formulation of the variable stepsize normalized least mean squares (VSS-NLMS)-algorithm, i ... | https://dsp.stackexchange.com/questions/59715/unknown-symbol-expression-in-text-about-adaptive-filters-cst |
Question: <p>I need to identify the coefficients of a linear, causal, time-invariant physical system that can be described by a classical state-space formulation.</p>
<p>For the sake of the example, suppose that the system has two states, only one of which is observed, and one input. Suppose that this state-space is d... | https://dsp.stackexchange.com/questions/19125/is-there-a-widely-available-implementation-of-an-adaptive-recursive-numericall |
Question: <p>The context is as follows: I want to measure (as in, digitize) some input signal <span class="math-container">$x$</span> (I have no detailed knowledge about <span class="math-container">$x$</span> or its statistical description).</p>
<p>A noise signal <span class="math-container">$x_N$</span> contaminate... | https://dsp.stackexchange.com/questions/54660/adaptive-filter-for-noise-cancellation-when-measuring-some-input |
Question: <p>I have the following diagram for the adaptive seperation of a narrowband and wideband signal using LMS algorithim,</p>
<p><a href="https://i.sstatic.net/IZCIN.png" rel="nofollow noreferrer"><img src="https://i.sstatic.net/IZCIN.png" alt="enter image description here"></a></p>
<p>The way it was explained ... | https://dsp.stackexchange.com/questions/54079/seperation-of-wideband-and-narrowband-adaptive-filter |
Question: <p>I am writing LMS filter to suppress noise in wav file (I know there are many modules to do this but I need to write LMS manually now as I will translate it into C later).</p>
<p>According to this answer[1], the inputs will be the noisy voice and a shifted version of it
here is my python code:</p>
<pre><c... | https://dsp.stackexchange.com/questions/55723/lms-adaptive-filter-noise-suppression-question-about-my-implementation |
Question: <p>Suppose I have a IIR filter represented by <span class="math-container">$$G_0\left(z\right)=\frac{1}{1-0.2z^{-1}-0.1z^{-2}}$$</span></p>
<p>I would like to use the LMS algorithm to model an FIR filter <span class="math-container">$G\left(z\right)$</span> of order <span class="math-container">$N = 15$</span... | https://dsp.stackexchange.com/questions/71770/iir-adaptive-filter-in-matlab |
Question: <p>I am attempting to make ANC headphones, so far I have constructed the headphones and validated that every component works. I have also constructed an FIR adaptive filter using leaky NLMS for its algorithm. I have validated that this filter works by using the signal from the headphones reference microphone ... | https://dsp.stackexchange.com/questions/96618/problem-with-feedback-in-lms-adaptive-filter-for-anc |
Question: <p>I will try to explain the issue I am having as clearly as possible without going into my coding or maths. I have my own and a MATLAB Central implementation pf standard LMS in MATLAB. Fixed step size. No normalization or other stuff.</p>
<p>I am trying to use it in a system identification setup. I generate... | https://dsp.stackexchange.com/questions/37074/upsampled-input-to-an-adaptive-filter |
Question: <p>My understanding of interpolation specific to resampling applications is limited to the concept of inserting zeros, then designing a filter to minimize distortion in the passband and reject the images the zero-insert creates (to desired performance levels), such as what is depicted in a simple interpolate ... | https://dsp.stackexchange.com/questions/38568/spline-based-adaptive-interpolation-filters |
Question: <p>General questions:</p>
<ul>
<li>Is the Kalman filter (they have used Unscented Kalman Filter) adaptive or not? Is the Unscented Kalman Filter used in the paper an adaptive algorithm? </li>
<li>Adaptive algorithms such as Constant Modulus and Least Squares are adaptive. Why? What is being adapted ? Based o... | https://dsp.stackexchange.com/questions/42710/explain-the-adaptive-part-of-adaptive-algorithms-kalman-filter-and-least-mean |
Question: <p>I want to find the resonant frequency of specific <a href="https://en.wikipedia.org/wiki/End-blown_flute" rel="nofollow noreferrer">end-blown flute</a> called <a href="http://persianney.com/technique.html" rel="nofollow noreferrer">Persian ney</a>, Using LMS in arrangement of system identification. Two sig... | https://dsp.stackexchange.com/questions/88123/modeling-end-blown-flute-instrument-using-adaptive-filter |
Question: <p>i am currently attempting system identification using the LMS algorithm. The input and the output data are available and are very noisy and consists of multiple frequencies. The input and the output data are shown below.
<a href="https://i.sstatic.net/rNeKT.png" rel="nofollow noreferrer"><img src="https://... | https://dsp.stackexchange.com/questions/85113/lms-adaptive-filter-for-system-identification |
Question: <p>For the adaptive filter to work properly, a desired signal d(n) needs to be provided. The output from the equalizer y(n) is subtracted from d(n) to produce an error signal, which is used to adjust the filter weights.</p>
<ol>
<li><p>The adaptive filter is located on the receiver side, so how to obtain the... | https://dsp.stackexchange.com/questions/51200/recursive-least-square-adaptive-linear-equalizer |
Question: <p>I'm trying to filter out line noise(60Hz and 120Hz) from a EEG signals received over a bluetooth link. I'm proposing to use a IIR notch filter to filter out the line-noise which varies w.r.t to distance between the bluetooth transmitter and receiver. I'm not able to establish a clear relationship between t... | https://dsp.stackexchange.com/questions/19612/adaptive-line-noise-removal-using-iir-notch-filters |
Question: <p>I am trying to understand the entire signal chain and all the algorithms associated with adaptive filtering as mentioned in the case above. From my understanding:</p>
<ul>
<li><p><strong>Adaptive noise filtering (ANF)</strong>- can be performed with the help of Weiner filters, wavelet packet based auditor... | https://dsp.stackexchange.com/questions/43222/signal-chain-for-voice-calls-including-adaptive-noise-cancellation-adaptive-ech |
Question: <ol>
<li><p>Considering non-linear filtering technique like Extended Kalman filtering with Expectation Maximization (EM). EM is an iterative technique but what is Kalman filtering? Is Kalman Filtering called iterative approach? </p></li>
<li><p>Adaptive signal processing algorithms like Least Mean Square and ... | https://dsp.stackexchange.com/questions/26860/terminologies-adaptive-recursive-and-iterative |
Question: <p>I am looking for methods to enhance noisy images, where:</p>
<ul>
<li>some pixels in the image are very noise,</li>
<li>some other pixels do not contain so much noise. </li>
</ul>
<p>My first thought is to build an adaptive Gaussian filter.
This means that the Gaussian kernel will depend on the (estimat... | https://dsp.stackexchange.com/questions/16326/adaptive-gaussian-filter-for-image-denoising |
Question: <p>Adaptive IIR filters is not straightforward, and may be unstable. Many people say that adaptive IIR filters <em>use less coefficients</em> than FIR filters. What I'm curious about is how many coefficients can IIR save?</p>
<p>I tried to use adaptive IIR filters to estimate transfer function of a 32-order ... | https://dsp.stackexchange.com/questions/32129/whats-the-advantage-of-adaptive-iir-filter-against-fir |
Question: <p>I am currently working on a project where i am to estimate a signal x_T using x_1 and x_2 with an RLS filter.</p>
<p>I have a problem where i don't quite get the results i am looking for. I think there is a problem with the filter coefficients. I don't know which ones i am to use.</p>
<p>This is what i hav... | https://dsp.stackexchange.com/questions/90089/rls-adaptive-filter-for-estimating-signal |
Question: <p><strong>Problem:</strong> I am looking at an adaptive filtering application where the eigenvaluespread of the autocorrelation matrix <span class="math-container">$R$</span> is important for the convergence of the algorithm. For a <strong>single</strong> channel system the autocorrelation matrix <span class... | https://dsp.stackexchange.com/questions/73686/autocorrelation-of-multiple-signals |
Question: <p>In my design and implementation of a SIR particle Filter, I don't have the state process equation of the actual system, which would have given a very good estimation of the real signal. I was wondering, if there are any mathematical methods out there to extrapolate the next state (i.e construction of proce... | https://dsp.stackexchange.com/questions/81914/adaptive-particle-filter-unknown-process-equation |
Question: <p>I am using gradient descent on an adaptive IIR filter for the below system <a href="https://i.sstatic.net/Q6D6S.png" rel="nofollow noreferrer">1</a>. At the moment I am just assuming the known system is not there and it works fine. However, occasionally when the known system has slower dynamics it does not... | https://dsp.stackexchange.com/questions/84700/approximate-a-known-system-with-adaptive-filter-and-an-unknown-system-in-a-serie |
Question: <p>Here you can see building blocks of DFE from <strong>"Adaptive Filters: Theory and Applications”</strong> a book by Behrouz Farhang-Boroujeny in chapter 17.</p>
<p>Figure 17.9 shows the overall building blocks of DFE which in the input we have noisy signal x that passed through channel and feeds to eq... | https://dsp.stackexchange.com/questions/91920/decision-feedback-equalizer-building-blocks-from-adaptive-filters-theory-and-a |
Question: <p><a href="https://i.sstatic.net/bZHOLDIU.jpg" rel="nofollow noreferrer"><img src="https://i.sstatic.net/bZHOLDIU.jpg" alt="LMS algorithm" /></a></p>
<p>When applying an algorithm like LMS, for example, in order to updates the weight coefficients I need the error signal at time <span class="math-container">$... | https://dsp.stackexchange.com/questions/94512/adaptive-filtering-isnt-the-desired-signal-dn-already-known |
Question: <p>It seems both names are used for the same algorithm:</p>
<p><strong>least mean square</strong> - mainly literature before 1990, for example: <em>Widrow, Bernard, and Samuel D. Stearns. "Adaptive signal processing prentice-hall." Englewood Cliffs, NJ (1985).</em></p>
<p><strong>least mean squares<... | https://dsp.stackexchange.com/questions/83988/lms-adaptive-filter-is-it-least-mean-square-or-least-mean-squares |
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>I am aware that some filter implementations such as <a href="http://www.signal.uu.se/Staff/pd/DSP/Doc/applicat/chap6.pdf" rel="nofollow noreferrer">lattice/ladder</a> and <a href="https://www.dsprelated.com/freebooks/filters/Series_Parallel_Filter_Sections.html" rel="nofollow noreferrer">SoS sections</a> a... | https://dsp.stackexchange.com/questions/85359/why-are-some-filter-implementations-preferable-for-adaptive-iirs |
Question: <p>I am recording data from an accelerometer attached to the chest (1000Hz). I need to extract the respiratory waveform. I tried an adaptive bandpass filter based on a dominant frequency in my signal based on <a href="https://cutt.ly/j24zdj" rel="nofollow noreferrer">https://cutt.ly/j24zdj</a>.</p>
<p>Steps ... | https://dsp.stackexchange.com/questions/59831/adaptive-band-pass-filter-for-extracting-respiratory-waveform-from-accelerometer |
Question: <p>Now I am trying to design FIR based adaptive filter for rejection of Jamming in GPS device. (just self-learning purpose, Jamming is simple tone)</p>
<p>I've designed NOT-BAD performance adaptive filter for low sampling frequency. (~44100 Hz)</p>
<p>But when I use this adaptive filter for high sampling fr... | https://dsp.stackexchange.com/questions/25111/sampling-frequency-of-gps |
Question: <p>This <em>might</em> be a terminology question but I am not sure. </p>
<p>Basically, what is the difference between <em>conventional</em> beamformers, and <em>adaptive</em> beamformers? I thought that all beamformers were inherently adaptive to some criteria, like minimization of distortion or variance, or... | https://dsp.stackexchange.com/questions/7825/difference-between-conventional-and-adaptive-beamformers |
Question: <p>I'm having some trouble implementing my LMS Adaptive Filter in MATLAB to separate wideband and narrowband signals from a voice signal.</p>
<p>I'm using a delayed version of my input as a reference as well as the error term.</p>
<pre><code>step = 0.01;
w = zeros(1, N);
xDelayed = [zeros(1, 100) x']'; % d... | https://dsp.stackexchange.com/questions/53461/adaptive-lms-algorithm-matlab |
Question: <p><a href="https://dsp.stackexchange.com/questions/146/how-can-one-improve-the-robustness-of-adaptive-beamformers-to-signal-mismatches">It has been shown</a> that 'diagonal loading' a covariance matrix derived for an adaptive beamformer can improve robustness of the beamformer when the antenna array is pertu... | https://dsp.stackexchange.com/questions/303/why-does-diagonal-loading-of-a-covariance-matrix-make-an-adaptive-beamformer-mor |
Question: <p>I am quite new to the idea of equalization. I have a few queries regarding the same.</p>
<ul>
<li>In my application I require to equalize a channel whose impulse response is an IIR response. Is it possible to design an adaptive equalizer based on LMS algorithm to equalize it? I tried some MATLAB simulatio... | https://dsp.stackexchange.com/questions/17838/adaptive-equalizer-for-iir-channel |
Question: <p>I read a paper about adaptive filtering specifying the following procedure:
<span class="math-container">$$ q_{FIR}(k) \overset{\textrm{FIR}}{\longleftarrow} q(k)$$</span></p>
<p>The framework processes the input signal (sampled in <span class="math-container">$f_s$</span> sampling rate) using STFT and wor... | https://dsp.stackexchange.com/questions/95783/filter-processing-frequency-to-time-truncate-taps-and-back-to-frequency-doma |
Question: <p>I have the following equalization problem as shown in the figure below:</p>
<p><a href="https://i.sstatic.net/O9Ws2.png" rel="nofollow noreferrer"><img src="https://i.sstatic.net/O9Ws2.png" alt=""></a></p>
<p>Now I can compute the coefficients for my adaptive FIR filter c (dim(c) = N) the following:<br>
... | https://dsp.stackexchange.com/questions/63914/adaptive-equalization-vs-inverse-of-transfer-function |
Question: <p>I'm just a bit confused about the least mean squares algorithm to separate wideband and narrowband in an adaptive filter for voice conversation. I'm interested in the narrowband part and I'm confused about the LMS equation as follows:</p>
<p><a href="https://i.sstatic.net/ghBem.png" rel="nofollow noreferr... | https://dsp.stackexchange.com/questions/53444/least-mean-squares-algorithm-confusion-with-adaptive-line-enhancement |
Question: <p>I want to extrapolate a signal <strong>X</strong> of length 11, using the weiner filter coefficients <strong><em>W</em></strong> of length 7. The procedure I am using is as follows:</p>
<ol>
<li>Compute the autocorrelation matrix upto lag 8 .</li>
<li>Using Levinson recursion, invert the autocorrelation m... | https://dsp.stackexchange.com/questions/11422/adaptive-wiener-filter-coefficients-calculation |
Question: <p>AEC algorithms mostly rely on LMS adaptive filtering, i.e. you update FIR filter coefficients then perform the filtering. Theoretically, the FIR must be as long as the maximum echo length you want to cancel. For instance to cancel delays up to 500ms on a 48kHz signal, you'll need a 24000 point FIR.
When yo... | https://dsp.stackexchange.com/questions/26689/echo-cancellation-supporting-long-delays-and-without-frequency-domain-processing |
Question: <p>What is the best filter for removing Gaussian noise without destroying the edges? I am using the standard Lena images with additive Gaussian noise and I want to denoise before applying anisotropic diffusion. I don't want to median filter because edges become blurred. I tried adaptive filtering but results ... | https://dsp.stackexchange.com/questions/1365/how-to-remove-gaussian-noise-from-an-image-without-destroying-the-edges |
Question: <p>I have to apply some kind of adaptive filter to my function $f(x).$ I present each point of my signal as a Gaussian, whose bandwidth depends on its location <strong>(not the point of observation $\textbf{x}$)</strong> as $h(t),$ which is a known pre-calculated function. The final output function $s(x)$ is ... | https://dsp.stackexchange.com/questions/10561/need-a-fast-algorithm-of-adaptive-convolution |
Question: <p>I see mention that normalized LMS "usually converges faster than LMS", in Diniz "Adaptive Filtering" p.152, can this be made more precise? IE, for which signal distributions does it hold?</p>
<p>I'm in particular interested in overparameterized regime where misadjustment is zero for LMS... | https://dsp.stackexchange.com/questions/83564/when-is-normalized-lms-better-than-lms |
Question: <p>In adaptive filters, the development of LMS algorithm typically starts from the Weiner-Hopf equation, while the development of RLS algorithm starts from the normal equation. As I understand, these two equations are the same, and both their solutions is the optimal coefficients that the adaptive filter has... | https://dsp.stackexchange.com/questions/17632/what-is-the-difference-between-the-weiner-hopf-equation-and-the-normal-equation |
Question: <p>Is it when the cost function such as the error signal or Mean Square Error (MSE) signal is minimised?</p>
Answer: <p>The adaptive filter is converged when the error is what they call Wide Sense Stationary, meaning the mean and variance of the error are unchanging over long time intervals.</p>
| https://dsp.stackexchange.com/questions/28366/how-is-it-determined-that-an-adaptive-filter-has-converged |
Question: <p><a href="https://i.sstatic.net/huadl.jpg" rel="nofollow noreferrer"><img src="https://i.sstatic.net/huadl.jpg" alt="enter image description here"></a><a href="https://i.sstatic.net/p7Tm1.jpg" rel="nofollow noreferrer"><img src="https://i.sstatic.net/p7Tm1.jpg" alt="Image shows the matlab code of lms algori... | https://dsp.stackexchange.com/questions/64515/why-is-the-error-between-the-desired-signal-and-estimated-signal-in-the-case-of |
Question: <p>I am trying to create an adaptive noise canceller using the RLS algorithm. The dsp toolbox from matlab offers <a href="https://de.mathworks.com/help/dsp/ref/dsp.rlsfilter-class.html" rel="nofollow noreferrer">the RLS adaptive filter</a> already implemented, so this saved me some trouble.</p>
<p>My goal is... | https://dsp.stackexchange.com/questions/43075/noise-cancellation-using-rls-filter |
Question: <p>I understand how echo cancellation works with a single speaker and no reverberation (using adaptive filtering and freezing the coefficients during double-talk). However, in cases with more than one speaker source or reverberation causing different propagation of the same signal, would this not completely c... | https://dsp.stackexchange.com/questions/50679/how-do-state-of-the-art-echo-cancellation-algorithms-deal-with-variable-propagat |
Question: <p>In "adaptive filter theory" and "advance signal processing and Noise reduction" they have directly come up with the term gain without stating how they got it.</p>
<p>In adaptive filter theory they have jumped from equation 13.16 to 13.18.</p>
<p><a href="https://i.sstatic.net/Ene7I.png" rel="nofollow nor... | https://dsp.stackexchange.com/questions/35337/how-the-gain-term-k-left-n-right-is-derived-why-is-it-called-gain |
Question: <p>I am using a Recursive least square adaptive filter to process electromyography signals and it is working decently so far. I decided to implement an LMS adaptive filter as a noise cancellation, so that I can compare the results, however, going through the matlab documentation for the <a href="https://de.ma... | https://dsp.stackexchange.com/questions/42938/difference-between-leaking-factor-and-forgetting-factor |
Question: <p>While reading a book on adaptive filter, I have came across a term 'nonlinear' associated with signal.</p>
<p>But I have learned about the linear and non linear system, which is defined by the principle of homogeneity and additive. I am curious to know the definitions of a non-linear signal. I have search ... | https://dsp.stackexchange.com/questions/97847/non-linear-signal-mean |
Question: <p>I do understand that MPC is a control method and requires known model in the feedback path.
LMS, on the other hand, is more like an adaptive filtering, which estimates the tap coefficients yielding the minimum mean squared error. Plus, LMS does not require known model.</p>
<p>Besides the differences I ment... | https://dsp.stackexchange.com/questions/73701/what-is-the-difference-between-lms-and-mpc |
Question: <p>I'm tinkering in Matlab with a problem that's very similar to active noise cancellation. In the literature, the secondary path is described as the transfer function from the output of the adaptive filter to the error input sensor. The algorithm needs to model this path to obtain good results.</p>
<p>How... | https://dsp.stackexchange.com/questions/28058/modeling-adc-in-active-noise-cancellation |
Question: <p>I meet a problem with designing a filter. I have two different instruments that could measure the same AC signal (usually ~200hz, always <1kHz), <strong><em>A</em></strong> and <strong><em>B</em></strong>. <strong>A</strong> can carry out signal measurement during the normal operation of the instrument... | https://dsp.stackexchange.com/questions/67728/how-to-design-a-filter-that-can-filter-out-noise-accurately-after-setting-the-p |
Question: <p>I am trying to remove low frequencies from a signal and intuitively I chose the high-pass filter, more specifically - a Butterworth filter, Order 4 (because I am not sure how to choose properly the order and 4 seemed as a good choice) and cutoff frequency of 50 Hz. The problem is, that the filter removes t... | https://dsp.stackexchange.com/questions/43374/removing-low-frequencies-from-a-signal |
Question: <p>Suppose,I have an input of length of lenght 100. data=[x0...x99];
I take a window of lenght 11 from x0 to x10.
windowed data=[x0 x1 x2 x3 x4 x5 x6 x7 x8 x9 x10],
now, I compute 7 wiener coefficients which are
[w0 w1 w2 w3 w4 w5 w6];
next I will move by window one sample forward,
new windowed data= [x1 x... | https://dsp.stackexchange.com/questions/11475/adaptive-calculations-of-wiener-filter-coefficients |
Question: <p>$$\hat f(x, y) = g(x, y)-\frac{\sigma_n^2}{\sigma_L^2}\left[g(x, y)-m_L\right]$$</p>
<p>What are the meanings of the following terms:</p>
<ul>
<li>$m_L$</li>
<li>${\sigma_\eta}^2$</li>
<li>${\sigma_L}^2$</li>
</ul>
<p>Here we see that $m_L$ is subtracted from the Image and then the whole term is multipl... | https://dsp.stackexchange.com/questions/29513/what-is-an-adaptive-mean-filter |
Question: <p>I'm working with AEC of Speex. The algorithm is based on the MDF adaptive filter + an adaptive learning rate. I'm using it like a ANC and it works very well.
Does anybody have some material, as block scheme, data flow diagram of the AEC of Speex.
I have read the documentation but it is not very useful for ... | https://dsp.stackexchange.com/questions/15075/aec-speex-how-does-it-work |
Question: <p>I'm trying to implement a simulation of an ANC system with python, using this model <a href="https://www.mathworks.com/help/audio/ug/active-noise-control-with-simulink.html#d122e12289" rel="nofollow noreferrer">here</a>. <a href="https://i.sstatic.net/5sCIt.png" rel="nofollow noreferrer"><img src="https://... | https://dsp.stackexchange.com/questions/73668/what-is-causing-my-anc-lms-update-to-diverge |
Question: <p>I found algorithms that seems the same to me, but they are described with different names (in field of adaptive filtering).</p>
<p>For example:</p>
<ol>
<li><p><strong>LMS - least-mean-squares</strong> seems to be <strong>GD - stochastic gradient descent</strong> </p></li>
<li><p>Often the <strong>stocha... | https://dsp.stackexchange.com/questions/30605/whats-the-difference-between-lms-and-gradient-descent-adaptation |
Question: <p>I am trying to implement an NLMS algorithm for a multi-input single-output(MISO) structure. </p>
<p>We take a reference signal x, then we made a new set of P input signals from it as follows: x_op (k) = x(k)^p. k denotes the k-th sample of our reference signal x.</p>
<p>For the case where P = 1, our adap... | https://dsp.stackexchange.com/questions/55874/nlms-algorithm-for-a-miso-structure |
Question: <p>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 poles have magnitude less than 0.92, or less that 0.89, etc. I havn't seen a published example with a pole... | https://dsp.stackexchange.com/questions/3057/abspoles-1-by-what-margin-for-a-stable-filter |
Question: <p><a href="http://freeverb3.sourceforge.net/iir_filter.shtml" rel="nofollow">A practical guide for digital IIR audio filters</a> has cookbook-style values for creating higher-order Bessel filters out of biquads, but the values listed aren't very precise:</p>
<pre><code>You should multiply the Fc for each st... | https://dsp.stackexchange.com/questions/7830/bessel-filter-second-order-sections-q-and-fc-multiplier-derivation |
Question: <p>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? </p>
<p>For an ergodic source (where time and ensemble averages should be the same) I would have thought the choice was arbitra... | https://dsp.stackexchange.com/questions/8165/choice-of-cost-function-in-adaptive-noise-cancellation |
Question: <p>I am multiplying two sine waves with the same frequency (f), but might have a phase difference 0 <-> 90 degrees. The product is a two frequency sinusoidal wave with f1 = f-f = DC and f2 = f+f = 2f. I now want to filter out the 2f component. I am currently sampling at Fs = 32 x f. I get a 40th order equi... | https://dsp.stackexchange.com/questions/10460/how-does-a-fir-equiripple-filter-behave-close-to-dc |
Question: <p>I'm trying to get my dsp sea legs a bit, and am trying to complete a problem that asks for the mean delay and expected SNR boost for a given difference equation: y[n] = (x[n] + ... + x[n-N+1])/N</p>
<p>I'd love some general guidance on how to learn more about how to address this. Specifically,</p>
<ul>
<... | https://dsp.stackexchange.com/questions/15048/fir-filter-mean-delay-and-snr-from-difference-equation |
Question: <p>As I understand it, FIR filtering is a linear process. That mean for me that the whole filtering process will have a fully predictable behavior.
So, could someone explain why a universal deterministic and optimal filter design method to obtain the desired response doesn't exist and why recursive approach... | https://dsp.stackexchange.com/questions/17198/why-are-recursives-methods-useful-for-fir-filter-design |
Question: <p>I have a RLC circuit with transformation $$H(s) = \frac{R}{Ls+\frac{1}{Cs}+R}$$ and I know that by setting $s=j\omega$ I can obtain the frequency response of this system. What I don't understand is how I can use the frequency response graph to decide which values I should set on R and f $(L=2\pi f, C=1\mu ... | https://dsp.stackexchange.com/questions/18829/which-values-do-i-use-to-filter-a-specific-frequency-with-an-rlc-circuit |
Question: <p>Why a filter of the form </p>
<p>$$
1+a_0 z^{-1} +a_1 z^{-2}+ a_2 z^{-3}
$$
has the same amplitude response when in the reversed form
$$
a_2+a_1 z^{-1} +a_0 z^{-2}+z^{-3}
$$
but the phase response is different. I don't get it, and can't understand what the differences in terms of the relative group delays... | https://dsp.stackexchange.com/questions/22728/why-this-two-fir-filters-have-the-same-amplitude-but-different-phase-response |
Question: <p>I would like to be able to implement an All-Pass filter with a frequency-dependent group delay. I need a maximum group delay at low frequency of about 20 samples (for F<em>s</em> = 44.1kHz), and it needs to fall to zero(-ish) at the Nyquist frequency. Ideally, I would like to specify a corner frequency a... | https://dsp.stackexchange.com/questions/24031/any-simple-digital-all-pass-filter-design-tools |
Question: <p>Have been trying to design IIR and FIR 60Hz notch filters but all have step response way above 10ms. </p>
Answer: <p>You can trade off steepness of the notch against time domain ringing. That's a fundamental trade off and there is no way around it. When using an IIR notch you can simply adjust the "Q" (qu... | https://dsp.stackexchange.com/questions/26161/is-it-possible-to-design-a-digital-filter-that-rejects-60hz-noise-while-keeping |
Question: <p>FIR filter coefficients are known. Then what is the matlab code or function that is used to determine the corresponding farrow structure coefficients? </p>
Answer: <p>There is no standard way to determine the coefficients of the Farrow structure for a given FIR filter. The Farrow structure is an implement... | https://dsp.stackexchange.com/questions/27094/how-to-find-out-the-farrow-coefficients-if-fir-coefficients-are-given |
Question: <p>I have a 10 bit ADC sample stream in which I would like to apply a digital band pass filter. Is there a theoretical filter tap count that would define the threshold of "usefulness"? </p>
<p>I guess my thought process is: There are only 10 bits, so at some point (via increasing the tap count) the attenuati... | https://dsp.stackexchange.com/questions/37314/max-useful-filter-tap-count-for-given-fixed-point-bitwidth |
Question: <p>hi i'm going to design a low pass FIR filter for an EEG signal for the detection of eppileptic siezure and i want to know what is the suitable design Method to the filter? i need this FIR filter just to smooth the signal to use it for discret wavelet transform.
thanks</p>
Answer: <p>For a quick filter des... | https://dsp.stackexchange.com/questions/38620/what-is-the-suitable-design-method-to-the-filter |
Question: <p>Please tell me, why low-pass FIR Equiripple filters are designed with 3 dB of ripple in the pass-band and at least 60 dB of attenuation in the stop-band
and not other values ?
I want to know are these values simply a convention?</p>
Answer: <p>Stop band attenuation is a good thing right? After all it's t... | https://dsp.stackexchange.com/questions/38759/why-low-pass-fir-equiripple-filter-is-designed-with-3-db-of-ripple-in-the-pass-b |
Question: <p>I was studying about realization structures of digital filters. Is it mandatory to have the order of numerator must be less than that of denominator of transfer function for realization of filter using direct form I and II?</p>
Answer: <p>no. but, for a <strong>causal</strong> LTI digital filter with a ... | https://dsp.stackexchange.com/questions/46046/direct-form-i-and-direct-form-ii |
Question: <p>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 starting point,</p>
<p>In other hand i read about Chebyshev approximation with arbitrary magnitude,
how can i... | https://dsp.stackexchange.com/questions/51111/what-is-meant-by-arbitrary-in-the-context-of-digital-filter-design |
Question: <p>In the window method for filter design it is explained that we do an ideal filter and then we pass it to time domain, but it will be time limited and then...</p>
<p>Why don't we filter directly in the frequency domain?</p>
Answer: | https://dsp.stackexchange.com/questions/56586/fir-filter-window-method |
Question: <p>What are the uses of ripples in the filter design. We try to design a filter with least ripples but sometimes we design a filter to have 2 or 3 ripples. Why is it so? </p>
Answer: <p>Ripples are usually an undesired side effect. E.g., when designing a frequency selective filter you normally want a piecewi... | https://dsp.stackexchange.com/questions/60113/ripples-in-filter |
Question: <p>I'm trying to write an accelerometer calibration script that uses filters to convert from volts into <span class="math-container">$m/s^2$</span>. As accelerometers tend to have non-flat response curves, this means I have to design a rather complex filter. I'm not worried about phase, as I can just apply ... | https://dsp.stackexchange.com/questions/64858/analytic-solution-for-non-flat-filter-design |
Question: <p>I am trying to go from a time-domain description of a filter, via frequency domain, generating a filter based on that frequency domain description, then seeing how far from the original I end up.</p>
<pre><code>L = 10-1;
h = zeros(L+1,1);
h(2) = 1;
L2 = 1024-1;
H = fft(h, L2+1);
H1 = H(1:(L2/2+2));
W = 2*... | https://dsp.stackexchange.com/questions/68166/invfreqz-and-frequency-domain-filter-specification |
Question: <p>For the second order Lynn's low pass filter, the general form of the transfer function is:
<span class="math-container">$$
H(z)=\frac{(1−z^{-m})^2}{ (1−z^{-1})^2 }
$$</span>
where <span class="math-container">$m$</span> is a positive integer. The gain for this is <span class="math-container">$m^2$</span>, ... | https://dsp.stackexchange.com/questions/73983/formula-for-designing-lynns-low-pass-filter |
Question: <p>Where to learn about "analog prototype filters"?</p>
<p>I've heard about them, but I'm unsure about what they really are and how they're constructed.</p>
Answer: <p>"Analog prototype" filters are well-known analog filters that have specific desirable properties. They include (but aren't limited to):</p>
... | https://dsp.stackexchange.com/questions/51277/where-to-learn-about-analog-prototype-filters |
Question: <p>I have built a resonance filter using the code from <a href="https://wiki.multimedia.cx/index.php/Impulse_Tracker#IT214_sample_compression" rel="nofollow noreferrer">here</a> I would like to know what the construction of this filter is. I believe it is an IIR and that it is probably a second order/two pole... | https://dsp.stackexchange.com/questions/75884/what-is-the-construction-of-this-filter |
Question: <p>I'm looking for an algorithm to detect the slew rate of a 50Hz discrete input signal sampled at 400Hz. I'm doing the obvious - so a low-pass filter on <em>(sample - prev_sample) / deltaT</em> but this results in an oscillating output. I think this is because often the input is zero because <em>sample == pr... | https://dsp.stackexchange.com/questions/83243/digital-slew-detection |
Question: <p>I'd like to implement freuqency sampling method for linear phase FIR filter design using IDFT transform.</p>
<p>My procedure goes like this :</p>
<ol>
<li>determine desired magnitude response value in frequency points</li>
<li>I add linear phase response function with group delay (N-1)/2 to get complex f... | https://dsp.stackexchange.com/questions/19625/fir-filter-design-with-frequency-sampling-method-setting-proper-phase-response |
Question: <p>How does the transition width of a FIR filter relate to phase delay at the output of the filter?</p>
<p>I have been trying to find information on the subject for days.</p>
Answer: <p>In <a href="https://dsp.stackexchange.com/a/34307/4298">this answer</a> I gave two well-known heuristic design formulas re... | https://dsp.stackexchange.com/questions/35607/relation-between-fir-filters-transition-width-and-phase-delay |
Question: <p>When decimating a narrow-band signal with a <a href="http://en.wikipedia.org/wiki/Cascaded_integrator-comb_filter">cascaded integrator-comb (CIC) filter</a>, which FIR filter is more suitable to compensate the CIC frecuency response?</p>
Answer: <p>There is no single answer to your question: as with any... | https://dsp.stackexchange.com/questions/160/fir-filter-compensator-when-using-a-cic-decimation-filter |
Question: <p>If a time-domain signal has sharp corners, its frequency spectrum will contain high-frequency components. Truncating the spectrum results in Gibbs' phenomenon. So if you're trying to design an FIR, you really want the target frequency response to be nice and smooth so that windowing the impulse response do... | https://dsp.stackexchange.com/questions/2934/phase-shift-filter-design |
Question: <p>Is it possible to dynamically filter a signal in realtime on a microcontroller? </p>
<p>What I mean by dynamically is to change the cut off frequency of the filter on the fly. </p>
<p>I have some DSP knowledge and realize that I probably do not have the computational power to do an FFT on the fly. Instea... | https://dsp.stackexchange.com/questions/8120/realtime-filtering-on-a-microcontroller-with-dynamic-cut-off-frequencies |
Question: <p>I have a filter design, and it filters over a 1-2 kHz range.<br>
What should I do if I want to apply it to data with a different sample rate than the one for which it was designed?</p>
<p>Let's say it consists of Bessel and Chebyshev filters. How do I find a function that defines each filter's coefficient... | https://dsp.stackexchange.com/questions/1104/applying-fir-filter-to-data-with-different-sample-rates |
Question: <p>The famous <a href="https://www.w3.org/TR/audio-eq-cookbook/" rel="nofollow noreferrer">https://www.w3.org/TR/audio-eq-cookbook/</a> offers a set of [real] biquad filter calculation formulas that generally work fine.</p>
<p>However, when filter's frequency approaches Nyquist frequency the Q (bandwidth) spe... | https://dsp.stackexchange.com/questions/19225/audio-eq-cookbook-without-frequency-warping |
Question: <p>I calculated the order of the FIR filter to be 31(N). so, the number of coefficients has to be 32 (N+1). so, I have to increase it to 33 to make it an odd number.</p>
<p>Why the number of filter coefficients is required to be an odd number?</p>
Answer: <p>I think it is about having a linear phase. Having... | https://dsp.stackexchange.com/questions/18413/why-the-number-of-filter-coefficients-in-fir-filter-has-to-be-an-odd-number |
Question: <p>I have a sensor producing (more or less) bandlimited data with a cut-off of about 45Hz, with a roll-off and <a href="http://en.wikipedia.org/wiki/Additive_white_Gaussian_noise" rel="noreferrer">AWGN</a>. I have an ADC that samples said signal at 800Hz, with a single-pole anti-aliasing filter at about 200Hz... | https://dsp.stackexchange.com/questions/3293/oversampling-and-decimation-what-filter-to-use |
Question: <p>Is there any current ideas about IIR filter algorithm parallelization with thread count greater than filter order?</p>
<p>Ideally, parallelization method should load GPU efficiently.</p>
Answer: <p>You might be interested in the following paper:</p>
<p>Nehab, D; Maximo, A; Lima, R; Hoppe, H: GPU-Efficie... | https://dsp.stackexchange.com/questions/8020/iir-filter-parallelization |
Subsets and Splits
No community queries yet
The top public SQL queries from the community will appear here once available.