Underwater acoustic channels are recognized for being one of the most difficult propagation media due to considerable difficulties such as: multipath, ambient noise, time-frequency selective fading. The exploitation ...
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Underwater acoustic channels are recognized for being one of the most difficult propagation media due to considerable difficulties such as: multipath, ambient noise, time-frequency selective fading. The exploitation of sparsity contained in underwater acoustic channels provides a potential solution to improve the performance of underwater acoustic channel estimation. Compared with the classic 10 and 11 norm constraint lms algorithms, the p-norm-like (Ip) constraint lms algorithm proposed in our previous investigation exhibits better sparsity exploitation performance at the presence of channel variations, as it enables the adaptability to the sparseness by tuning of p parameter. However, the decimal exponential calculation associated with the p-norm-like constraint lms algorithm poses considerable limitations in practical application. In this paper, a simplified variant of the p-norm-like constraint lms was proposed with the employment of Newton iteration m to approximate the decimal exponential calculation. Num simulations and the experimental results obtained in physical shallow water channels demonstrate the effectiveness of the proposed method compared to traditional norm constraint lms algorithms.
The design of adaptive nonlinear filters has sparked a great interest in the machine learning community. The present paper aims to present some recent developments in nonlinear adaptive filtering. It provides an in-de...
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The design of adaptive nonlinear filters has sparked a great interest in the machine learning community. The present paper aims to present some recent developments in nonlinear adaptive filtering. It provides an in-depth analysis of the performance and complexity of a class of kernel filters based on the least-mean-squares algorithm. A key feature that underlies kernel algorithms is that they map the data in a high-dimensional feature space where linear filtering is performed. The arithmetic operations are carried out in the initial space via evaluation of inner products between pairs of input patterns called kernels. The SNR improvement and the convergence speed of kernel-based least-mean-squares filters are evaluated on two types of applications: time series prediction and cardiac artifacts extraction from magnetoencephalographic data.
In passive radar,the target's echo could be masked by the sidelobes of the direct path and multipath interference (DPI and MPI) received by the receiver *** the direct path and multipath interference cancellation ...
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ISBN:
(纸本)9781467321006
In passive radar,the target's echo could be masked by the sidelobes of the direct path and multipath interference (DPI and MPI) received by the receiver *** the direct path and multipath interference cancellation is a key factor in passive radar *** this paper,an improved lms algorithm is proposed,and its performance in passive radar direct path and multipath interference cancellation is *** proposed algorithm,which varies the step size,has better performance compared with the traditional lms *** computer simulation results show that the algorithm has a faster convergence rate and a smaller steady state error.
In this paper we introduce a novel adaptation algorithm for adaptive filtering of FIR and IIR digital filters within the context of system identification. The standard lms algorithm is hybridized with GA (Genetic Algo...
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In this paper we introduce a novel adaptation algorithm for adaptive filtering of FIR and IIR digital filters within the context of system identification. The standard lms algorithm is hybridized with GA (Genetic algorithm) to obtain a new integrated learning algorithm, namely, lms-GA. The main aim of the proposed learning tool is to evade local minima, a common problem in standard lms algorithm and its variants and approaching the global minimum by calculating the optimum parameters of the weights vector when just estimated data are accessible. In the proposed lms-GA technique, first, it works as the standard lms algorithm and calculates the optimum filter coefficients that minimize the mean square error, once the standard lms algorithm gets stuck in local minimum, the lms-GA switches to GA to update the filter coefficients and explore new region in the search space by applying the cross-over and mutation operators. The proposed lms-GA is tested under different conditions of the input signal like input signals with colored characteristics, i.e., correlated input signals and investigated on FIR adaptive filter using the power spectral density of the input signal and the Fourier-transform of the input's correlation matrix. Demonstrations via simulations on system identification of IIR and FIR adaptive digital filters revealed the effectiveness of the proposed lms-GA under input signals with different characteristics.
The coefficients of an echo canceller with a near-end section and a far-end section are usually updated with the same updating scheme, such as the lms algorithm. A novel scheme is proposed for echo cancellation that i...
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The coefficients of an echo canceller with a near-end section and a far-end section are usually updated with the same updating scheme, such as the lms algorithm. A novel scheme is proposed for echo cancellation that is based on the minimisation of two different cost functions, i.e. one for the near-end section and a different one for the far-end section. The approach considered leads to a substantial improvement in performance over the lms algorithm when it is applied to both sections of the echo canceller. The convergence properties of the algorithm are derived. The proposed scheme is also shown to be robust to noise variations. Simulation results confirm the superior performance of the new algorithm.
In this paper,an adaptive semi-active SSDV(Synchronized Switch Damping on Voltage) method based on the lms algorithm is proposed and applied to the vibration control of a composite *** the SSDV method,the value of vol...
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In this paper,an adaptive semi-active SSDV(Synchronized Switch Damping on Voltage) method based on the lms algorithm is proposed and applied to the vibration control of a composite *** the SSDV method,the value of voltage source in the switching circuit is critical to its control *** the adaptive approach proposed in this study,the voltage coefficient is adjusted adaptively using the lms *** new adaptive approach is compared with the derivative-based adaptive SSDV proposed in the former study in the control of the first mode of a composite *** control results show that adaptive adjustment of voltage coefficient is effective in the vibration control of the composite beam and that lms-based approach is slightly better than the derivative-based approach.
A new generalized sidelobe canceller (GSC) based minimum variance distortionless response (MVDR) beamformer is proposed in this paper, which applies constrained stability least mean squares (CS-lms) algorithm for ...
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A new generalized sidelobe canceller (GSC) based minimum variance distortionless response (MVDR) beamformer is proposed in this paper, which applies constrained stability least mean squares (CS-lms) algorithm for adaptive weights optimization. With a different recursive formula, CS-lms algorithm is more flexible than normal lms algorithm. Then, this algorithm is introduced into GSC based MVDR beamformer. As a result, a good signal of interest (SOI) extraction and interferences nulling performance is obtained. Simulation experiments show the merits of the proposed beamformer against the other two, GSC based MVDR beamformer with lmsalgorithm and fully adaptive MVDR beamformer.
Echo path estimation in echo canceling for teleconference system is a problem in double-talk *** correlation function based algorithms were defined by the authors to solve this *** this paper,in order to improve the c...
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Echo path estimation in echo canceling for teleconference system is a problem in double-talk *** correlation function based algorithms were defined by the authors to solve this *** this paper,in order to improve the convergence speed of correlation function based algorithm,we propose a new modified proportionate step-size adaptation method,and then implement it into frequency domain extended correlation lms algorithm (FEClms).The new algorithm is called proportionate frequency domain extended correlation lms algorithm (PFEClms).The computer simulation results support the theoretical findings and verify the robustness of the proposed algorithm in the double-talk situation.
Starting from the characteristic of voice signal itself, a new variable step-size lms algorithm based on DCT is proposed in order to improve the comprehensive performance in adaptive noise cancellation, which combines...
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Starting from the characteristic of voice signal itself, a new variable step-size lms algorithm based on DCT is proposed in order to improve the comprehensive performance in adaptive noise cancellation, which combines the merits of normalized DCT-lms algorithm and variable step-size lms algorithm, and give full play to the decorrelation capability of DCT and rapid convergence effect of variable step-size algorithm. The c simulation results show that the new algorithm has faster convergence rate and smaller steady state error compared with traditional lms and Nlms algorithm, however, the computational complexity is comparable to Nlms algorithm at the same time.
Echo path estimation in echo canceling for teleconference system is a problem in double-talk *** correlation function based algorithms were defined by the authors to solve this *** this paper,in order to improve the c...
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Echo path estimation in echo canceling for teleconference system is a problem in double-talk *** correlation function based algorithms were defined by the authors to solve this *** this paper,in order to improve the convergence speed of correlation function based algorithm,we propose a new modified proportionate step-size adaptation method,and then implement it into frequency domain extended correlation lms algorithm(FEClms).The new algorithm is called proportionate frequency domain extended correlation lms algorithm(PFEClms).The computer simulation results support the theoretical findings and verify the robustness of the proposed algorithm in the double-talk situation.
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