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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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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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NLMS Echo Cancellation: How do we estimate when the time-step at which the Far-End Echo will be generated?

I'm new to this domain. From my understanding, we do the following in our algo: ...
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2answers
40 views

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
52 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
27 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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1answer
59 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
42 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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3answers
49 views

Usefulness of Practical filter ouptput

If a practical filter can't remove all unwanted frequency components like ideal filter, does that mean unwanted frequency component are still present after filtering? how can we use such distorted ...
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1answer
68 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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23 views

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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24 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
50 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
28 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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40 views

Autocorrelation Matrix of a AR process

i'm beggining to study signal processing, and i'm having trouble on writting it ( specially defining the signal x(n) ) in matlab and finding the solution for this problem, if anyone could help I would ...
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35 views

audio filter designing help

buzzer frequency range recording of audio signals(environmental sounds with speech) with buzzer soundI have audio signals(environmental sounds with speech) with buzzer sound, I have to attenuate only ...
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1answer
53 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
66 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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49 views

How to implement a LMS filter bank using multi-threading in python?

Since it involves parallelism and concurrency I planned on using multi-threading in python. Is it possible to do it this way if so how?
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1answer
275 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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39 views

numerical instability while solving a physical model

I am trying to solve an estimation problem and my physical model look smth like this: $A'/A^2 = B'/B^2 + C'/C^2$, from which I have measured $A$ and a reference for $B'$. I want to try adaptive ...
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1answer
123 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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1answer
94 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
118 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
215 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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25 views

Need some help in optimization problem in blind channel equalization

Blind Channel Equalization is used to inverse model the channel effect on recieved signal so that the unnecessary noise and ISI is removed from recieved signal constellaion.The most famous algorithm ...
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141 views

Why linear equalizer (LMS,RLS) cannot equalize every (FIR) channel?

MATLAB have an example that explain how equalizers work, that is nicely work on impulse response chan=[.986; .845; .237; .123+.31i], but I altered it to ...
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1answer
56 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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31 views

Calculate Autocorrelation-Sequence In A System Identification Scenario With Sensor Noise

The system below is given. ${\bf c}$ are the coefficients of the adaptive filter (filter order 3), which shall approximate the system $$ H(z) = 3 + z^{-1} $$ $S_1(z)$ and $S_2(z)$ are transfer ...
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59 views

Adaptive (Complex) Noise Cancellation using LMS

I have been strugglling with getting a hold of LMS algorithm for Noise Cancellation and following is one of the problem I am trying to solve:Given a Complex valued input with an additive complex ...
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1answer
74 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
46 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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25 views

An implementation of an VSLMS/VSSLMS filter?

I've been reading into VSLMS/VSSLMS lately and finally got my hand on a few papers that explain some things alright. However, so far it's just been mathematical formulas that did not help me ...
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66 views

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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19 views

Fast RLS Array Algorithm and Hyperbolic Givens

My question is specific to array algorithm version of RLS implemented by applying alternating Givens and Hyperbolic Givens transformations as described in Chapter 37 of A. H. Sayed, Adaptive Filters, ...
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21 views

Reverberation rejection algorithm

I am working on colored noise (reverberation) rejection algorithms. One approach present in literature is through prewhitening filter. In this approach as per my understanding, first we estimate ...
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1answer
725 views

Removing low frequencies from a signal

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 ...
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1answer
70 views

Performance of adaptive filters

Can somebody please provide an intuitive answer or reference for the following questions? Q1: Dependence of estimation performance on number of data points -- I could not find any information whether ...
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44 views

Signal chain for voice calls including adaptive noise cancellation, adaptive echo cancellation, and automatic gain control and their algorithms

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: Adaptive noise filtering (ANF)- can ...
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341 views

Noise cancellation using RLS filter

I am trying to create an adaptive noise canceller using the RLS algorithm. The dsp toolbox from matlab offers the RLS adaptive filter already implemented, so this saved me some trouble. My goal is to ...
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1answer
145 views

Difference between Leaking factor and Forgetting factor

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 ...
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1answer
216 views

Frequency tracking with huge noise

I'm working on a frequency tracking problem with noise, where the amplitude of the noise is orders of magnitude higher than the amplitude of my signal (~1000x). Some details: The signal is roughly ...
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850 views

Noise variance in MATLAB wiener filter

I applied a white gaussian noise g=imnoise(f,'gaussian',0,0.01) to my grayscale image. Then I used a Wiener filter with ($3\times 3$) mask for denoising. In Wiener ...
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2answers
591 views

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. ...
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2answers
70 views

Are all least square filters adaptive?

Are least square filters, or filters that minimize error energy, the same as least mean square adaptive filters?
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85 views

SNR for channel after equalization

Suppose I have a channel described by $h[n]$ or $H(z)$ over which I send a simple PAM2 signal. After the channel AWGN is added. What is the best possible SNR with equalization? As suggested in the ...
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2answers
350 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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2answers
43 views

Stochastic approximation algorithm

The goal is to find the FIR filter coefficients $\mathbf{h} = [5;3]$ with the help of the adaptive FIR filter $\mathbf{w}$ of order $p = 2$. I have implemented the Stochastic approximation algorithm ...
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1answer
189 views

LMS and delta impulse response- Equalization concepts

I have been reading about equalization and channel estimation in order to apply to my application of estimating trajectory path for a two-wheeled robot. The application is quite similar to ...
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2answers
196 views

Levinson Method

i have just one question about Levinson, my question is: can we apply this method to determine the coefficients for FIR filter? if yes, please help me ! this is an example: ...
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1answer
408 views

Using recursive least square filter

I am fairly new to the signal processing world and that being said I have little to no experience. The problem that I am having is that I am not quite sure how to use ...