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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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2answers
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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2answers
503 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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1answer
225 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 ...
3
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1answer
271 views

Echo cancellation supporting long delays and without frequency domain processing

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 ...
2
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1answer
445 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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1answer
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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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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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
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 ...
0
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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 ...
0
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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
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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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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0answers
492 views

Phase noise estimation and compensation schemes

Brief background There is a known problem that high order modulation schemes suffers from phase noise of reference generator. It's very important issue for e.g. line-of-sight (LOS) modems exploiting ...
2
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0answers
743 views

Speex echo cancellation takes a long time (~20s) to converge

I have been experimenting with Speex's echo canceller during the past few days. I got the echo canceller to work, however, it takes a long time until it actually begins filtering out the echo. It ...
2
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0answers
261 views

Kalman filter without model versus RLS

As I found out in my previous question Where to get transtion matrix for Kalman filter?, I need a model for correct usage of Kalman filter. In paper On the Intrinsic Relationship Between the Least ...
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0answers
285 views

Motion artifact cancellation from ECG signal using 3-axis accelerometer data

I have a ECG signal measured as a test subject is performing 3 different activities - sitting, walking(2mph) and jogging(5mph) and also the corresponding 3-axis accelerometer(X,Y and Z) signals. I can ...
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0answers
22 views

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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0answers
37 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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0answers
60 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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0answers
534 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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0answers
153 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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0answers
116 views

Instability problem in NLMS adaptive algorithm

I have been used NLMS algorithm to equalize 4x4 MIMO signals, but the bit-error-rate (BER) after equalization is unstable with iterations. I don't know if it is the normal behavior of the adaptive ...
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0answers
166 views

Removing low frequency vibrations from measured signal

Suppose I have two sensors $s_1$ and $s_2$. $s_1$ measures the desired heart signal (with most of the frequency content below 100 Hz), and the other is a reference sensor picking mainly background ...
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0answers
149 views

RLS Algorithm Convergence

I am looking for some help to understand the concept how RLS converges? If possible to present it graphically that would be best. It is very easy to understand the understanding of convergence in case ...
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0answers
1k views

AEC Speex . How does it work?

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 ...
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0answers
206 views

RLS adaptive filter

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 ...
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0answers
814 views

adaptive filter does not converge for all inputs

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 ...
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0answers
39 views

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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0answers
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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0answers
83 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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0answers
334 views

Wiener filter yielding negative MSE (and MMSE)

I implement a Wiener filter using the following code ...
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0answers
108 views

Transient and steady-state analysis for adaptive filter

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. I have done common (in my ...
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0answers
566 views

“Desired signal” in LMS adaptive filters?

I'm trying to understand how to specify the "desired signal" in adaptive LMS filters such as the following: This one, or this one page 7, or this one. It seems like the example there uses it for ...
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0answers
362 views

Audio block-processing discontinuities

I have an algorithm that looks at an audio frame, calculates a threshold in db-rms, and decides to filter the frame whether or not the threshold is crossed. However, filtering each block is producing ...
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78 views
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183 views

Need a fast algorithm of adaptive convolution

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 (not the point of observation $\textbf{x}$) ...
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0answers
505 views

Maximum step size for adaptive filter convergence

I’m trying to understand the conception of function maxstep The foundation of this function is function firwiener with input parameters: length of adaptive ...