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Questions tagged [adaptive-filters]

An adaptive filter is a filter that self-adjusts its transfer function according to an optimization algorithm driven by an error signal.

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How to Subtract One Signal from Another?

I am capturing some body internal signals. I am using 2 identical mics. One captures the in body signal and one open air. I want to remove/delete/subtract the open air signal (noise for me) from ...
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1answer
224 views

Why does block LMS have the same performance as LMS?

The block LMS and conventional LMS have the same convergence rate and the same misadjustment. I am having trouble wrapping my head around this. The block LMS uses a more accurate estimate of the ...
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222 views

How can a RLS algorithm utilise Wiener filter as FIR (M-tap) block?

I'm currently working with a dataset of $5000$ pulses of $N=15000$ samples each. I managed to implement the RLS algorithms with a FIR M-Tap filter such that $M\leq 15000$ ($150$ seems to achieve the ...
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444 views

What Is the Difference between RLS, LMS and Wiener Filter? When Is One Preferred Over Another?

I'm dealing with a channel equalization problem where the channel is modeled as a WSS process. I understand LMS utilises a Wiener-like approach, ie it converges to the optimal (wiener) solution. I ...
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502 views

IIR Lattice Filter in C programming

I would like to implement Lattice IIR filter in c, i can't defined the value of gN, i know that x(n) = fN(n), but how to initialize gN ?. you can find the algorithm here : n
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Linear Combiner Based LMS Transfer Function

I am currently working on narrowband beamforming and looking to compute the transfer function for a linear combiner based LMS. While doing a quick search I have found the following article which ...
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39 views

BER performance of fractional equalizers for colored noise

I am trying to compare the BER performance of fractional and symbol spaced equalizers for channel equalization in colored noise scenario. I have found from simulation that for positively correlated ...
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181 views

Regression vector size for prediction, reconstruction and filtration with adaptive filters

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. ...
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1answer
34 views

Modelling unwanted signal in a LMS Adaptive Filter

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 ...
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1answer
74 views

Equalizers with low data rate systems

In case of using equalizers with ZigBee systems, what type of fading does the channel show, flat or frequency selective, fast or slow? Is the inter-symbol interference (ISI) problem encountered in ...
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1answer
84 views

Speech dereverbaration via maximum kurtosis adaptive filtering

I'm trying to code the algorithm described in Speech dereverbaration via maximum-kurtosis subband adaptive filtering by Gillespie, Malvar and Florencio, and the signal looks cleaner in when I plot it. ...
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1answer
59 views

Active Noise cancellation for non periodic signals

I observed that the coded algorithm for active noise cancellation is not able to cancel some of the signals like a human voice. Is there any solution for this? Can we really cancel non-periodic ...
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For complex values, why use complex conjugate in convolution?

Taken from Adaptive Filter Theory (2014) written by Haykin page 110 : $$y(n) = \sum_{k=0}^{\infty} w_k^*u(n-k), \quad n=0,1,2,...$$ where $u$ and $w$ are complex values. My question is why use ...
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1answer
60 views

Adaptive filter for noise cancellation when measuring some input

The context is as follows: I want to measure (as in, digitize) some input signal $x$ (I have no detailed knowledge about $x$ or its statistical description). A noise signal $x_N$ contaminates $x$ ...
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1answer
55 views

Implementation of Block LMS

In the implementation of block LMS, i need one clarification. In the 3rd step as shown in the figure attached, the summation over a product of input, $\mathbf{u}$ and error, $e$, associated with each ...
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1answer
69 views

Highest rate achievable by current DSP and A/D chips?

I'm trying to determine whether an application I have in mind is feasible with currently available DSP technology (DSP chips and ADCs). I want to eliminate an interfering signal using an LMS adaptive ...
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1answer
97 views

Can Temperature data be predicted using adaptive filter using LMS algorithm?

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. ...
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1answer
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Seperation of wideband and narrowband - Adaptive Filter

I have the following diagram for the adaptive seperation of a narrowband and wideband signal using LMS algorithim, The way it was explained to us was that "The narrowband signal is correlatied over ...
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1answer
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Acoustic Echo Cancellation

I have the following diagram for acoustic echo cancellation I'm having a hard time figuring out what's present on the top and bottom lines in terms of reflected echos, original signal etc. Here's ...
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1answer
32 views

Echo Cancellation over Voice Telephone Network

I'm a bit confused about the way echo canellation over a voice telephone network works like in the following figure My issue is with the fact that the point of the adaptive filter and LMS algorithim ...
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1answer
65 views

System Identification using LMS Adaptive Filter

I just have a question about using an least-mean-squares algorithim adaptive filter for system identification. Consider the following I am told that as the error converges to a small value, the ...
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1answer
56 views

Adaptive LMS Algorithim MATLAB

I'm having some trouble implementing my LMS Adaptive Filter in MATLAB to seperate wideband and narrowband signals from a voice signal. I'm using a delayed version of my input as a reference as well ...
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2answers
50 views

Least Mean Squares Algorithim confusion with Adaptive Line Enhancement

I'm just a bit confused about the least means squares algorithim to seperate wideband and narrowband in an adaptive filter for voice conversation I'm interested in the narrowband part and I'm confused ...
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2answers
126 views

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

Let's say we want to isolate a band 1000 hz +/- 50 hz. Obviously, limiting the bandwidth by applying a passband filter will always destroy a bit the sharp transients (a Dirac or a rectangular ...
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3answers
68 views

Usefulness of Practical filter ouptput

If a practical filter can't remove all unwanted frequency components like ideal filter, does that mean unwanted frequency component are still present after filtering? how can we use such distorted ...
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0answers
51 views

How general are adaptive-filtering techniques? [closed]

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 ...
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2answers
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Finding the auto-correlation sequence $r_{xx}[k]$ for an AR(2) process

Consider the following recursive difference equation of a LTI system, where $v[n]$ is a white noise, zero-mean process with $\sigma_v^2 = 1$. $ x[n] = v[n] + 0.75x[n-1]-0.25x[n-2] $ I want to ...
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1answer
205 views

estimate the impulse response after equalization

I want to estimate the impulse response of the channel at the receiver. Assuming some arbitrary impulse response: h=[1 0.2 -0.4 0.0 0.6]. Once the equalizer is ...
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1answer
66 views

Modeling channel response as complex vs splitting I and Q

If I have a communications channel where a digital QAM signal is transmitted, I can either model the channel's impulse response as a complex $h(t)$, or as 2 separate impulse responses $p(t)$ and $q(t)$...
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1answer
92 views

Block LMS with overlapping blocks

In the Block LMS algorithm, the input is partitioned into nonoverlapping blocks of size $L$ and the filter coefficients are updated once every $L$ samples. Would convergence improve if we used more ...
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1answer
147 views

Recursive Least Square Adaptive Linear Equalizer

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 ...
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1answer
140 views

Hardware requirements for ambient active noise cancallation system using adaptive LMS

Background I'm working on a hobby project to reduce ambient low-frequency noise generated from HVAC equipment within a small area of a room. The noise is predictable and concentrated between 55Hz and ...
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0answers
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How do state of the art echo cancellation algorithms deal with variable propagation times of sound due to multiple speakers or reverberation?

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 ...
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1answer
107 views

Adaptive Filter Gradient Descent

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 ...
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What's the Difference Between LMS and Gradient Descent Adaptation?

I found algorithms that seems the same to me, but they are described with different names (in field of adaptive filtering). For example: LMS - least-mean-squares seems to be GD - stochastic gradient ...
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2answers
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Estimation of Input Signal to Obtain the Desired Output Signal for an Unknown Filter

Suppose $h(n)$ is a finite impulse response which is unknown. We can feed any input signal $x(n)$ into the system and observe the corresponding output signal $y(n)$. From this, is it possible to ...
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47 views

Adaptive algorithm for active noise control

I'm looking at building a cost-effective active noise control system. Basically, the system comprises two microphones - one reference and one error. A DSP unit samples and phase flips the reference ...
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1answer
74 views

Adaptive filter to scale and phase shift two sensors output

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 ...
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1answer
36 views

what is adaptive compressive sampling?

I have just started my work in compressive sensing. the measurement vectors are obtain by multiplying the sensing matrix with input signal. the thing i cant figure out adaptive compressive sampling. ...
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1answer
899 views

Are there any open source libraries for Adaptive noise cancellation?

I'm creating a chat application where people can input text via speech and I need to implement noise cancellation using Adaptive noise cancellation technique. I'm using dual microphones for this ...
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1answer
168 views

Doubts on LMS derivation

I have been trying to follow the Least Mean Square(LMS) algorithm derivation given by Wikipedia here and have the following questions. Here I expected $y(n)$ is to be computed by convolving $x(n)$ ...
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1answer
126 views

Can I have some feedback on this research paper?

I have submitted this paper "Effect of Transducer Positioning in Active Noise Control" to arXiv. I would like to get some feedback on the methodology of the same. Inputs of any sort are welcome. Can ...
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1answer
482 views

What do the filter coefficients in digital filter represent?

Could you help me? How can I understand the function of the filter coefficients practically? In the simple case, it is the impulse response of the LTI system. but how do they work?
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1answer
146 views

RLS Algorithm (Memoryless)

I have been studying the adaptive filters lately and now when I am at RLS (Recursive Least Squar) Algorithm I came across the term used in the weighting function of the RLS called forgetting factor ($\...
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1answer
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MATLAB : Proper estimation of weights and how to calculate MSE for QPSK signal for Constant Modulus Algorithm

I had asked conceptual Question about Constant Modulus Algorithm. I am implementing the simple steps of the algorithm, as I do not have the cma() built in module. ...
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1answer
98 views

Filter bank for complex spectrum

I'm using a complex-valued baseband signal as my input for subband adaptive filtering. Therefore i have a complex input spectrum. My problem is now, how to split my complex spectrum into $N$ equally ...
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1answer
355 views

MATLAB Adaptive Filters

In the audio domain, I am currently attempting to use MATLAB to distil: $\textrm{signal}$ from $\textrm{noise + signal}$ $\textrm{noise}$ from $\textrm{noise + signal}$ using two adaptive filters $\...
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Is there a filter to remove uncorrelated noise?

I am looking for a filter which can remove uncorrelated noise? Is there any such filter? I understand filters like Kalman filter, Wiener filter but they are not exactly what I am looking for. Yes, ...
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2answers
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Kalman Filtering with Unknown State Transition Matrix

I'm currently studying the use of Kalman filters for estimating linear systems. My current State Transition Matrix (STM) is the identity since so far I've been dealing with non time-varying systems. ...