4,845 research outputs found
Maximum-likelihood estimation of delta-domain model parameters from noisy output signals
Fast sampling is desirable to describe signal transmission
through wide-bandwidth systems. The delta-operator provides an ideal discrete-time modeling description for such fast-sampled systems. However, the estimation of delta-domain model parameters is usually biased by directly applying the delta-transformations to a sampled signal corrupted by additive measurement noise. This problem is solved here by expectation-maximization, where the delta-transformations of the true signal are estimated and then used to obtain the model parameters. The method is
demonstrated on a numerical example to improve on the accuracy of using a shift operator approach when the sample rate is fast
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Review of Unbiased FIR Filters, Smoothers, and Predictors for Polynomial Signals
Extracting an estimate of a slowly varying signal corrupted by noise is a common task. Examples can be found in industrial, scientific and biomedical instrumentation. Depending on the nature of the application the signal estimate is allowed to be a delayed estimate of the original signal or, in the other extreme, no delay is tolerated. These cases are commonly referred to as filtering, prediction, and smoothing depending on the amount of advance or lag between the input data set and the output data set. In this review paper we provide a comprehensive set of design and analysis tools for designing unbiased FIR filters, predictors, and smoothers for slowly varying signals, i.e. signals that can be modeled by low order polynomials. Explicit expressions of parameters needed in practical implementations are given. Real life examples are provided including cases where the method is extended to signals that are piecewise slowly varying. A critical view on recursive implementations of the algorithms is provided
Acoustic Echo and Noise Cancellation System for Hand-Free Telecommunication using Variable Step Size Algorithms
In this paper, acoustic echo cancellation with doubletalk detection system is implemented for a hand-free telecommunication system using Matlab. Here adaptive noise canceller with blind source separation (ANC-BSS) system is proposed to remove both background noise and far-end speaker echo signal in presence of double-talk. During the absence of double-talk, far-end speaker echo signal is cancelled by adaptive echo canceller. Both adaptive noise canceller and adaptive echo canceller are implemented using LMS, NLMS, VSLMS and VSNLMS algorithms. The normalized cross-correlation method is used for double-talk detection. VSNLMS has shown its superiority over all other algorithms both for double-talk and in absence of double-talk. During the absence of double-talk it shows its superiority in terms of increment in ERLE and decrement in misalignment. In presence of double-talk, it shows improvement in SNR of near-end speaker signal
A Stochastic Total Least Squares Solution of Adaptive Filtering Problem
An efficient and computationally linear algorithm is derived for total least squares solution of adaptive filtering problem, when
both input and output signals are contaminated by noise. The proposed total least mean squares (TLMS) algorithm is designed by
recursively computing an optimal solution of adaptive TLS problem by minimizing instantaneous value of weighted cost function.
Convergence analysis of the algorithm is given to show the global convergence of the proposed algorithm, provided that the
stepsize parameter is appropriately chosen. The TLMS algorithm is computationally simpler than the other TLS algorithms and
demonstrates a better performance as compared with the least mean square (LMS) and normalized least mean square (NLMS)
algorithms. It provides minimum mean square deviation by exhibiting better convergence in misalignment for unknown system
identification under noisy inputs
A kepstrum approach to filtering, smoothing and prediction
The kepstrum (or complex cepstrum) method is revisited and applied to the problem of spectral factorization
where the spectrum is directly estimated from observations. The solution to this problem in turn leads to a new
approach to optimal filtering, smoothing and prediction using the Wiener theory. Unlike previous approaches to
adaptive and self-tuning filtering, the technique, when implemented, does not require a priori information on the
type or order of the signal generating model. And unlike other approaches - with the exception of spectral
subtraction - no state-space or polynomial model is necessary. In this first paper results are restricted to
stationary signal and additive white noise
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