• Title/Summary/Keyword: Least mean squares algorithm

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A Design of New Digital Adaptive Predistortion Linearizer Algorithm Based on DFP(Davidon-Fletcher-Powell) Method (DFP Method 기반의 새로운 적응형 디지털 전치 왜곡 선형화기 알고리즘 개발)

  • Jang, Jeong-Seok;Choi, Yong-Gyu;Suh, Kyoung-Whoan;Hong, Ui-Seok
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.22 no.3
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    • pp.312-319
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    • 2011
  • In this paper, a new linearization algorithm for DPD(Digital PreDistorter) is suggested. This new algorithm uses DFP(Davidon-Fletcher-Powell) method. This algorithm is more accurate than that of the existing algorithms, and this method renew the best-fit value in every routine with out setting the initial value of step-size. In modeling power amplifier, the memory polynomial model which can model the memory effect of the power amplifier is used. And the overall structure of linearizer is based on an indirect learning architecture. In order to verify for performance of proposed algorithm, we compared with LMS(Least Mean-Squares), RLS(Recursive Least squares) algorithm.

Interference Cancellation Based on Adaptive Signal Processing for MIMO RF Repeaters (MIMO RF 중계기를 위한 적응 신호처리 기반의 간섭 제거)

  • Lee, Kyu-Bum;Choi, Ji-Hoon
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.35 no.9C
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    • pp.735-742
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    • 2010
  • In this paper, we propose adaptive algorithms for interference cancellation in RF repeaters with multiple transmit and receive antennas. When multiple antennas are used in a repeater, the imperfect isolation between transmit and receive antennas causes the feedback interference which is modeled as multi-input multi-output (MIMO) channel. To remove the feedback interference, we derive the least mean square (LMS) algorithm and the recursive least squares (RLS) algorithm for interference cancellation based on adaptive signal processing techniques. Through computer simulations for the proposed algorithms, we analyze the convergence characteristics and compare the steady-state performance for interference cancellation.

Frequency-Domain Adaptive Noise Canceller and Its Algorithm with Adaptive Compensator (적응보상기를 채용한 주파수영역 적응 잡음제거 시스템 및 알고리즘)

  • 손경식;김수중
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.27 no.9
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    • pp.1456-1467
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    • 1990
  • The time domain adaptive noise canceller (time domain ANC) with the adaptive compensator and its algorithm, so called compensated least mean squares(CLMS) algorithm, had been introduced to improve the performance of ANC[1]. In this paper the time domain ANC with the adaptive compensator is transformed into the frequency domain ANC with the adaptive ocmpensator. An compensated frequency-domain least mean squares(CFLMS) algorithm that can adapt the proposed frequency domain ANC is presented.

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Error in Variable FIR Typed System Identification Using Combining Total Least Mean Squares Estimation with Least Mean Squares Estimation (입출력 변수에 부가 잡음이 있는 FIR형 시스템 인식을 위한 견실한 추정법에 관한 연구)

  • Lim, Jun-Seok
    • The Journal of the Acoustical Society of Korea
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    • v.29 no.2
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    • pp.97-101
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    • 2010
  • FIR type system identification with noisy input and output data can be solved by a total least squares (TLS) estimation. However, the performance of the TLS estimation is very sensitive to the ratio between the variances of the input and output noises. In this paper, we propose an iterative convex combination algorithm between TLS and least squares (LS). This combined algorithm shows robustness against the noise variance ratio. Consequently, the practical workability of the TLS method with noisy data has been significantly broadened.

Mixture Filtering Approaches to Blind Equalization Based on Estimation of Time-Varying and Multi-Path Channels

  • Lim, Jaechan
    • Journal of Communications and Networks
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    • v.18 no.1
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    • pp.8-18
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    • 2016
  • In this paper, we propose a number of blind equalization approaches for time-varying andmulti-path channels. The approaches employ cost reference particle filter (CRPF) as the symbol estimator, and additionally employ either least mean squares algorithm, recursive least squares algorithm, or $H{\infty}$ filter (HF) as a channel estimator such that they are jointly employed for the strategy of "Rao-Blackwellization," or equally called "mixture filtering." The novel feature of the proposed approaches is that the blind equalization is performed based on direct channel estimation with unknown noise statistics of the received signals and channel state system while the channel is not directly estimated in the conventional method, and the noise information if known in similar Kalman mixture filtering approach. Simulation results show that the proposed approaches estimate the transmitted symbols and time-varying channel very effectively, and outperform the previously proposed approach which requires the noise information in its application.

Blind Channel Estimator based on the RLS algorithm (RLS 알고리즘에 기반을 둔 블라인드 채널 추정)

  • 서우정;하판봉;윤태성
    • Proceedings of the IEEK Conference
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    • 1999.11a
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    • pp.655-658
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    • 1999
  • In this study, We derived Recursive Least Squares(RLS) algorithm with adaptive maximum -likelihood channel estimate for digital pulse amplitude modulated sequence in the presence of intersymbol interference and additive white Gaussian noise. RLS algorithms have better convergence characteristics than conventional algorithms, LMS Least Mean Squares) algorithms.

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Estimation of Acid Concentration Model of Cooling and Pickling Process Using Volterra Series Inputs (볼테라 시리즈 입력을 이용한 냉연 산세 라인 산농도 모델 추정)

  • Park, Chan Eun;Song, Ju-man;Park, Tae Su;Noh, Il-Hwan;Park, Hyoung-Kuk;Choi, Seung Gab;Park, PooGyeon
    • Journal of Institute of Control, Robotics and Systems
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    • v.21 no.12
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    • pp.1173-1177
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    • 2015
  • This paper deals with estimating the acid concentration of pickling process using the Volterra inputs. To estimate the acid concentration, the whole pickling process is represented by the grey box model consists of the white box dealing with known system and the black box dealing with unknown system. Because there is a possibility of nonlinear term in the unknown system, the Volterra series are used to estimate the acid concentration. For the white box modeling, the acid tank solution level and concentration equations are used, and for the black box modeling, the acid concentration is estimated using the Volterra Least Mean Squares (LMS) algorithm and Least Squares (LS) algorithm. The LMS algorithm has the advantage of the simple structure and the low computation, and the LS algorithm has the advantage of lowest error. The simulation results compared to the measured data are included.

A Method for Screening Product Design Variables for Building A Usability Model : Genetic Algorithm Approach (사용편의성 모델수립을 위한 제품 설계 변수의 선별방법 : 유전자 알고리즘 접근방법)

  • Yang, Hui-Cheol;Han, Seong-Ho
    • Journal of the Ergonomics Society of Korea
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    • v.20 no.1
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    • pp.45-62
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    • 2001
  • This study suggests a genetic algorithm-based partial least squares (GA-based PLS) method to select the design variables for building a usability model. The GA-based PLS uses a genetic algorithm to minimize the root-mean-squared error of a partial least square regression model. A multiple linear regression method is applied to build a usability model that contains the variables seleded by the GA-based PLS. The performance of the usability model turned out to be generally better than that of the previous usability models using other variable selection methods such as expert rating, principal component analysis, cluster analysis, and partial least squares. Furthermore, the model performance was drastically improved by supplementing the category type variables selected by the GA-based PLS in the usability model. It is recommended that the GA-based PLS be applied to the variable selection for developing a usability model.

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A new AR power spectral estimation technique using the Karhunen-Loeve Transform (KLT를 이용한 AR 스펙트럼 추정기법에 관한 연구)

  • 공성곤;양흥석
    • 제어로봇시스템학회:학술대회논문집
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    • 1986.10a
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    • pp.134-136
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    • 1986
  • In this paper, a new power spectral estimation technique is presented. At first, by transforming the original data with the Karhunen-Loeve Transform(KLT), we can reduce the amount of the redundant information. Next, by modeling the transformed data by means of the autoregressive(AR) model and then applying the least-squares parameter estimation algorithm to this model, even more accurate spectrum estimates can be obtained. The KLT is the optimum transform for signal representation with respect to the mean-square error criterion. And the least-squares method is used to overcome the inherent shortcomings of popular burg algorithm.

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Performance Comparison of Equalizers for HomePNA 2.0 Systems (HomePNA 2.0 시스템을 위한 등화기의 성능 비교)

  • 박기태;최효기;이원철;신요한
    • Proceedings of the IEEK Conference
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    • 2002.06a
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    • pp.61-64
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    • 2002
  • In this paper, various equalizers are considered to improve the performance of Home Phoneline Networking Alliance (HomePNA) 2.0 system under dispersive channel with intersymbol interference. We evaluate and compare the performances of Recursive Least Squares (RLS) and Least Mean Squares (LMS) adaptation algorithms. Computer simulations show that the equalizers utilizing tile RLS algorithm outperforms the LMS algorithm, especially for the system of high symbol rate and complex constellation.

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