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2nd IEEE International Conference on Wireless Communications, Signal Processing and Networking (WiSPNET)

作者： Honade, Shrikant J. Ingole, Prashant V. Dudul, Sanjay V. Shelke, Nitin A. GH Raisoni Coll Engn Dept Elect & Telecomm Engn AMIETE SGBAU AmravatiLMISTE Amravati Maharashtra India Prof Ram Meghe Inst Technol & Res Dept Elect & Telecomm Engn Amravati Maharashtra India St Gadge Baba Amravati Univ Amravati Maharashtra India GH Raisoni Coll Engn Dept Comp Sci & Engn Amravati Maharashtra India

ISBN: (纸本)9781509044429

adaptive signal processing is extensively used as an area of research from few decades. There are various kinds of adaptive filtering algorithms are available in the literature which are used for different purposes. The work presented in this paper focused mainly on the performance analysis of various versions of most popular LMS (Least Mean Square) adaptive filtering algorithm when it is operated in unknown environment and deciding which algorithm has better performance in terms of minimum mean square error (MSE). From, the experimental results, it is observed that, when a same random input, random noise and desired response are considered for the different algorithms, with variable step size parameter and variable no. of filter taps, it very difficult to form a concrete decision on it, but in broader sense we can say that the performance of LMS sign sign algorithm for M = 2 and NLMS/Complex NLMS algorithm for higher value of M is found to be better in terms of mean square value of error when operated in an unknown situation.

关键词： adaptive filter Mean Square Error Filter weights desired response adaptive filtering algorithm LMS

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adaptive filtering Method of MFL Signal on Rail Top Surface Defect Detection

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IEEE ACCESS 2021年 9卷 87351-87359页

作者： Ji, Kailun Wang, Ping Jia, Yinliang Ye, Yunfei Ding, Shunyi Nanjing Univ Aeronaut & Astronaut Coll Automat Engn Nanjing 211000 Peoples R China Nanjing Inst Railway Technol Sch Intelligent Engn Nanjing 210031 Peoples R China

Magnetic flux leakage (MFL) detection technology provides an effective method to conduct high-speed detection of the damage suffered by rail surface. With regard to high-speed detection, there is frequently a complex noise contained in the magnetic signal of railway leakage, which is similar to the amplitude of defect and the overlaps of frequency spectrum. In this paper, an improved adaptive filtering method is proposed to solve the problem caused by filtering the MFL signal on the rail top surface. Through the characteristics of distribution shown by defects on the top surface of the railway and those of the data collected by the rail top array sensor, this method is applied to construct a virtual channel containing almost only interference signals but no defects. Then, in combination with the adaptive filtering algorithm, the virtual channel signal is taken as the reference input of the adaptive canceller, each single channel MFL signal is taken as the original input of the adaptive canceller, and the filtered MFL signal is taken as the output. Then, the MFL signal of rail top is collected by the train at the speed of 30km / h on the manual calibration line. According to the experimental results, the noise intensity of MFL signal is reduced by up to 81.44%. In addition, the filtering method is adopted to process MFL signals with different directions and varying detection speed. As indicated by the results, the noise intensity of MFL signal is reduced by more than 74%.

关键词： Rails filtering Magnetic separation Interference Inspection adaptive filters Vibrations High speed rail detection magnetic flux leakage signal adaptive filtering algorithm reference signal adaptive noise canceller

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Active feedback control using adaptive filtering

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JOURNAL OF VIBRATION AND CONTROL 2004年第1期10卷 25-38页

作者： Wang, FL Mechefske, CK Queens Univ Dept Engn Mech Kingston ON K7L 3N6 Canada

In this paper we apply a filtered-X algorithm to an active feedback control structure and derive the transfer function of a closed-loop control system. Simulation studies are then carried out on the closed-loop property while varying the parameters (input frequency, delays in plant, amplitude and phase of modeling filter). Several properties of adaptive feedback control are revealed. Experimental studies on feedback active noise control of noise in a finite duct and a small enclosure are described, and outstanding active noise control effects are achieved. Experimental results of closed-loop frequency response are also provided.

关键词： active noise control feedback control algorithm closed-loop frequency response adaptive filtering algorithm

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Selective Body Biasing for Post-Silicon Tuning of Sub-Threshold Designs: An adaptive filtering Approach

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IEEE TRANSACTIONS ON COMPUTER-AIDED DESIGN OF INTEGRATED CIRCUITS AND SYSTEMS 2015年第5期34卷 713-725页

作者： Geng, Hui Liu, Jianming Luo, Pei-Wen Cheng, Liang-Chia Grant, Steven L. Shi, Yiyu Missouri Univ Sci & Technol Dept Elect & Comp Engn Rolla MO 65409 USA Ind Technol Res Inst Hsinchu 31040 Taiwan

A sub-threshold design could provide a compelling approach to power critical applications. An exponential relationship exists, however, between the delay and the threshold voltage, that makes this design-time timing closure extremely difficult, if not impossible, to achieve. Several previous studies were focused on the technique of body biasing during post-silicon tuning for delay compensation. But they were mostly for super-threshold designs where spatially correlated L-eff variation dominates. They cannot be applied directly to sub-threshold designs in which purely random threshold voltage variations dominate. These works also assumed multiple body biasing voltage domains and multiple body biasing voltage levels, which involve significant design overhead. The problem of selective body biasing for post-silicon tuning of sub-threshold designs is examined in this paper. The possibility of using only one body bias voltage domain with a single body bias voltage is explored. The problem was formulated first as a linearly constrained statistical optimization model. The adaptive filtering concept from the signal processing community was then adopted so that an efficient, yet novel, solution could be developed. Using several 65 nm industrial designs, experimental results suggest that, compared with a seemingly more intuitive approach, the proposed approach can improve the pass rate by 57% on average with similar standby power and the same number of body biasing gates. This approach can reduce the standby power, on average, by 84%, with a 20% pass rate loss, more than the approach to bias all the gates.

关键词： adaptive filtering algorithm body biasing post-silicon tuning sub-threshold

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Normalized least mean square-based adaptive sparse filtering algorithms for estimating multiple-input multiple-output channels

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WIRELESS COMMUNICATIONS & MOBILE COMPUTING 2015年第6期15卷 1079-1088页

作者： Gui, Guan Xu, Li Adachi, Fumiyuki Akita Prefectural Univ Dept Elect & Informat Syst Akita Japan Tohoku Univ Grad Sch Engn Dept Commun Engn Commun Engn Sendai Miyagi 980 Japan Akita Prefectural Univ Dept Elect & Informat Syst Fac Syst Sci & Technol Akita Japan

This paper studies normalized least mean square-based adaptive sparse filtering algorithms for estimating multiple-input multiple-output (MIMO) channels. Although the MIMO channel is often modeled as sparse, traditional normalized least mean square-based filtering algorithm never takes the advantage of the inherent sparse structure information and thus causes some performance loss. Unlike the traditional method, the proposed two adaptive sparse channel estimation methods exploit the sparse structure information of MIMO channels. To validate the effectiveness of proposed MIMO channel estimates, theoretical analysis and simulation results are provided. We derive steady-state mean-square deviations of the proposed MIMO channel estimates and theoretically show that it is better than the traditional one. Moreover, their performance advantages are confirmed by computer simulations. Copyright (c) 2014 John Wiley & Sons, Ltd.

关键词： adaptive filtering algorithm normalized least mean square (NLMS) l(p)-norm NLMS l(0)-norm NLMS adaptive sparse channel estimation (ASCE) compressed sensing (CS)

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adaptive filtering for delivering multimedia services in W-CDMA cellular networks using genetic algorithms

Adaptive filtering for delivering multimedia services in W-C...

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International Conference on Wireless Networks, Communications and Mobile Computing

作者： De Angelis, F Habib, I CUNY City Coll New York NY 10031 USA

ISBN: (纸本)0780393058

This paper presents tin adaptive filtering algorithm for the delivery of multimedia content with different qualify of service (QoS) requirements in time-varying wideband code-division multiple access (W-CDMA) cellular networks. The algorithm objective is to guarantee levels of quality for two different user defined profiles (Premium, or Economy), while attempting to maximize the allocation of the radio channel bandwidth and maintaining the QoS requirements of downloaded traffic flows (audio, video, and data). Our proposed solution is based upon utilizing Genetic algorithms (GAs) to solve a multi-objective function that selects, during a control period, the bit rates and formats for the media codecs of downloaded traffic flows and assigns the air interface channel rate and power to each user, subject to different constraints. To show the performance of the algorithm, W-CDMA systems with and without filtering capabilities are compared.

关键词： adaptive filters bandwidth allocation broadband networks cellular radio code division multiple access filtering theory genetic algorithms multimedia communication quality of service QoS adaptive filtering algorithm code-division multiple access genetic a adaptive filters Code division multiple access Quality of service Genetic algorithm (GA) Multimedia services WCDMA filtering theory Broadband networks bandwidth allocation Multimedia Content Traffic flow audio video

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Variable kernel-based computing algorithms for estimating sparse multipath channels

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INTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS 2018年第7期31卷 e3393-e3393页

作者： Yang, Jie Zhao, Ruijie Song, Rongfang Xiong, Jian Liang, Yan Luo, Wang Xu, Li Gui, Guan Nanjing Univ Posts & Telecommun Coll Commun & Informat Engn Nanjing Jiangsu Peoples R China NARI Grp Corp State Grid Elect Power Res Inst Nanjing Jiangsu Peoples R China Akita Prefectural Univ Dept Elect & Informat Syst Yurihonjo Japan

Accurate channel estimation algorithms are required to mitigate the frequency-selective fading in the broadband wireless communication systems. Many physical experiments revealed that finite impulsive responses are distributed as sparse in delayed time domain. Hence, the inherent sparse information can be exploited by state-of-the-art computing algorithms, in order to improve channel accuracy. Existing computing algorithms, zero-attracting recursive least square (ZA-RLS) and reweighted ZA-RLS (RZA-RLS), have been developed to exploit channel sparsity. However, optimization theory implies that accuracy of the computing algorithms can be further improved to exploit more sparsity information. To further improve the estimation performance, this paper proposes a correntropy induced metric (CIM)-penalized RLS (CIM-RLS) based sparse channel estimation algorithm. Here, sparse constraint is performed by CIM function, while error constraint term is computed by RLS. In particular, Gaussian kernel is adopted for computing the CIM, and its variable kernel width (VKW) is computed for adaptively exploiting the channel sparsity. Monte Carlo simulation results are conducted to verify the proposed algorithm in different scenarios.

关键词： adaptive filtering algorithm sparse channel estimation variable kernel computing zero-attracting constraint

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Effective and adaptive algorithm for pepper-and-salt noise removal

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IET IMAGE PROCESSING 2017年第9期11卷 709-716页

作者： Chen, Qing-Qiang Hung, Mao-Hsiung Zou, Fumin Fujian Univ Technol Sch Informat Sci & Engn 3Xueyuan Rd Fuzhou 350118 Fujian Peoples R China Fujian Univ Technol Key Lab Automot Elect & Elect Drive Fujian Prov Fuzhou 350108 Fujian Peoples R China

According to the characteristic of pepper-and-salt noise, the authors first classify pixels in a polluted image into two classes: suspected noise and noise-free pixels. For a suspected noisy pixel, by counting the number of closed grey-level and noise-free pixels in a neighbourhood, one can correctly determine a noise or a noise-free pixel. Noise filtering does not process noise-free pixels. For the noisy pixels, an adaptive filtering algorithm with weighting mean based on Euler distance achieves excellent noise removal and good detail preservation. The algorithm can handle different noise levels, and the authors do not need to manually adjust the parameters and thresholds. The experimental results indicate that the authors' proposed method effectively filters pepper-and-salt noise. The authors note that when noise-free and noisy pixels with the same grey level appear in the polluted images, the noise-removal performance by the proposed method is much more excellent than those of the other existing methods.

关键词： image denoising image classification image colour analysis image filtering image preservation weighting mean Euler distance adaptive filtering algorithm noise filtering closed grey-level pixel noise-free pixel class suspected noise pixel class polluted image pixel classification pepperand-salt noise removal

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Joint Real-Time Identification for Mutual Inductance and Load Charging Parameters of IPT System

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IEEE JOURNAL OF EMERGING AND SELECTED TOPICS IN POWER ELECTRONICS 2023年第4期11卷 4574-4590页

作者： Wang, Lei Sun, Pan Liang, Yan Sun, Jun Wu, Xusheng Shen, Haomin Deng, Qijun Naval Univ Engn Sch Elect Engn Wuhan 430030 Peoples R China Wuhan Univ Sch Elect Engn & Automat Wuhan 430072 Peoples R China

Accurate and fast identification for mutual inductance and load charging parameters of inductive wireless power transfer (IPT) system is essential for impedance matching, and the output control of a communication-free IPT system. A joint real-time identification method for mutual inductance and load charging parameters of the IPT system is proposed in this article. Only via measuring the rms value of fundamental voltage and current on the primary side, the double equivalent impedance modulus equations about mutual inductance and equivalent load resistance are established. To obtain real-time identification results of mutual inductance and equivalent load resistance from the high-order implicit function equations, an improved adaptive filtering algorithm based on least mean square (LMS) is proposed. Then, the identification values of load charging parameters are obtained according to the identified mutual inductance and equivalent load resistance. Finally, a 3.6-kW platform is built to verify the validity of the proposed method. And the experimental results show that, compared with other existing achievements, the proposed method can identify U-Bat, I-Bat, R-L, and M simultaneously in real-time without changing the system operation, and it has a faster rate and higher accuracy of identification.

关键词： adaptive filtering algorithm identification of load charging parameters inductive wireless power transfer (IPT) least mean square (LMS) mutual inductance identification parameter identification

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Two-channel forward NLMS algorithm combined with simple variable step-sizes for speech quality enhancement

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ANALOG INTEGRATED CIRCUITS AND SIGNAL PROCESSING 2019年第1期98卷 27-40页

作者： Bendoumia, Redha Univ Blida 1 Dept Elect Signal Proc & Image Lab LATSI Blida Algeria

This paper addresses the problem of speech quality enhancement by adaptive two-channel filtering algorithms. Recently, the forward blind source separation structure has been proposed and combined with normalized least-mean-square algorithm (FNLMS). The main drawback of two-channel FNLMS algorithm is its poor performance in steady state regime when the fixed step-sizes values are selected large. However, the slow convergence rate is observed with the small fixed step-size values. In this paper, we propose three new combinations of the basic FNLMS algorithm with simple variable step-sizes approaches, for improving both the steady state values and convergence rate (noted TVSF for Two-channel Variable Step-size Forward). In these modifications, we propose new configuration of two-channel forward structure by three simple and efficient variable step-sizes estimations. To confirm the good performance of three proposed TVSF algorithms compared with the classical fixed-step-size version, we have carried out several simulations in very noisy situations using several criteria.

关键词： Variable step-size Speech quality Output SNR adaptive filtering algorithm Two-channel forward structure

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