• Title/Summary/Keyword: Least mean square

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Experimental Study on Bi-directional Filtered-x Least Mean Square Algorithm (양방향 Filtered-x 최소 평균 제곱 알고리듬에 대한 실험적인 연구)

  • Kwon, Oh Sang
    • Journal of Korea Society of Digital Industry and Information Management
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    • v.10 no.3
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    • pp.197-205
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    • 2014
  • In applications of adaptive noise control or active noise control, the presence of a transfer function in the secondary path following the adaptive controller and the error path, been shown to generally degrade the performance of the Least Mean Square (LMS) algorithm. Thus, the convergence rate is lowered, the residual power is increased, and the algorithm can become unstable. In general, in order to solve these problems, the filtered-x LMS (FX-LMS) type algorithms can be used. But these algorithms have slow convergence speed and weakness in the environment that the secondary path and error path are varied. Therefore, I present the new algorithm called the "Bi-directional Filtered-x (BFX) LMS" algorithm with nearly equal computation complexity. Through experimental study, the proposed BFX-LMS algorithm has better convergence speed and better performance than the conventional FX-LMS algorithm, especially when the secondary path or error path is varied and the impulsive disturbance is flow in.

Active Noise Control Using Wavelet Transform Domain Least Mean Square (웨이블릿 변환역 최소평균자승법을 이용한 능동 소음 제어)

  • Kim, Doh-Hyoung;Park, Young-Jin
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2000.06a
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    • pp.269-273
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    • 2000
  • This paper describes Active Noise Control (ANC) using Discrete Wavelet Transform (DWT) Domain Least Mean Square (LMS) Method. DWT-LMS is one of the transform domain input decorrelation LMS and improves the convergence speed of adaptive filter especially when the input signal is highly correlated. Conventional transform domain LMS's use Discrete Cosine Transform (DCT) because it offers linear band signal decomposition and fast transform algorithm. Wavelet transform can project the input signal into the several octave band subspace and offers more efficient sliding fast transform algorithm. In this paper, we propose Wavelet transform domain LMS algorithm and shows its performance is similar to DCT LMS in some cases using ANC simulation.

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Inertia and Coefficient of Friction Estimation of Electric Motor using Recursive Least-Mean-Square Method (순환 최소자승법을 이용한 전동기 관성과 마찰계수 추정)

  • Kim, Ji-Hye;Choi, Jong-Woo
    • The Transactions of The Korean Institute of Electrical Engineers
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    • v.56 no.2
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    • pp.311-316
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    • 2007
  • This paper proposes the algorithm which estimates moment of the inertia and friction coefficient of friction for high performance speed control of electric motor. The proposed algorithm finds the moment of inertia and friction coefficient of friction by observing the speed error signal generated by the speed observer and using Recursive Least-Mean-Square method(RLS). By feedbacking the estimated inertia and estimated coefficient of friction to speed controller and full order speed observer, then the errors of the inertia and coefficient of friction and speed due to the inaccurate initial value are decreased. Inertia and coefficient of friction converge to the actual value within several times of speed changing. Simulation and actual experiment results are given to demonstrate the effectiveness of the proposed parameter estimator.

Hearing aid application of feedback cancellation algorithm in frequency domain (주파수 대역에서의 피드백 제거 알고리즘의 보청기 응용)

  • Jarng, Soon-Suck
    • The Journal of the Acoustical Society of Korea
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    • v.35 no.4
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    • pp.272-279
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    • 2016
  • In this paper, the realization of a hearing aid adaptively cancelling feedback noise was considered. Conventional least mean square method in time domain was transformed into frequency domain in order to minimize computational burden. The adaptive filter algorithm was evaluated by Matlab (Matrix laboratory), and it was confirmed by CSR 8675 Bluetooth DSP IC (Digital Signal Processor Integrated Circuit) chip firmware realization. Some remote control features by a smart phone was added to the smart hearing aid for user interface easiness.

Power Spectral Estimation of Background EEG with LMS PHD (LMS PHD에 의한 배경단파 파워 스펙트럼 추정)

  • 정명진;최갑석
    • Journal of Biomedical Engineering Research
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    • v.9 no.1
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    • pp.101-108
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    • 1988
  • In this paper the power spectrum of background EEG is estimated by the LMS PHD based on least mean square. At the power spectrum estimatiom, the stocastic process of background EEG is assumed to consist of the nonharmonic sinusoid and the white noise. In the LMS PHD the model parameters are obtained by the least mean square at optimal order which is obtained from the fact that the eigenvalue's fluctuation of autocorrelation matrix of the normal back-ground EEG is smaller at some order than at other order when the power spectrum of background EEG is esitmated by PHD. The optimal order of this model is the 6-th order when the eigenvalue's fluctuation of autocorrelation matrix of background EEG is considered. The estimation results are with compared the results from the Maximum Entropy Spectral Estimation and Pisarenko Harmonic Decomposition. From the comparison results. The LMS PHD is possible to estimate the power spectrum of background EEG.

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Content Based Image Retrieval Using Combined Features of Shape, Color and Relevance Feedback

  • Mussarat, Yasmin;Muhammad, Sharif;Sajjad, Mohsin;Isma, Irum
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.7 no.12
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    • pp.3149-3165
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    • 2013
  • Content based image retrieval is increasingly gaining popularity among image repository systems as images are a big source of digital communication and information sharing. Identification of image content is done through feature extraction which is the key operation for a successful content based image retrieval system. In this paper content based image retrieval system has been developed by adopting a strategy of combining multiple features of shape, color and relevance feedback. Shape is served as a primary operation to identify images whereas color and relevance feedback have been used as supporting features to make the system more efficient and accurate. Shape features are estimated through second derivative, least square polynomial and shapes coding methods. Color is estimated through max-min mean of neighborhood intensities. A new technique has been introduced for relevance feedback without bothering the user.

Improvement Noise Attenuation Performance of the Active Noise Control System Using RCMAC (RCMAC를 이용한 능동소음 제어시스템의 소음저감 성능개선)

  • Han, S.I.;Yeo, D.Y.;Kim, S.H.;Lee, K.S.
    • Journal of Power System Engineering
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    • v.14 no.5
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    • pp.56-62
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    • 2010
  • In this paper, a recurrent cerebellar modulation articulation control (RCMAC) has been developed for improvement of noise attenuation performance in active noise control system. For the narrow band noise, a filter-x least mean square (FXLMS) method has bee frequently employed as an algorithm for active noise control (ANC) and has a partial satisfactory noise attenuation performance. However, noise attenuation performance of an ANC system with FXLMS method is poor for broad band noise and nonlinear path since it has linear filtering structure. Thus, an ANC system using RCMAC is proposed to improve this problem. Some simulations in duct system using harmonic motor noise and KTX cabin noise as a noise source were executed. It is shown that satisfactory noise attenuation performance can be obtained.

Reverse Filtering of Sound Field by Adaptive Filter and Neural Network (적응필터 및 신경회로망에 의한 음장의 역 필터링)

  • Choi, Jae-Seung
    • The Journal of the Korea institute of electronic communication sciences
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    • v.5 no.2
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    • pp.145-151
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    • 2010
  • This paper proposes a reverse filtering system of sound field obtaining a state of sound field transmitted from two sounds, using an adaptive filter and neural network. The proposed system uses the reverse filtering method with calculating and renewing a coefficient of a filter, using least mean square. Based on training the neural network, experiments confirm that the proposed system is effective for a simple waveform with non-linear distortion, by using neural network and adaptive filtering method.

Active Noise Control by Using Wavelet Packet and Comparison Experiments (웨이브렛 패킷을 이용한 능동 소음제어 및 비교실험)

  • Jang, Jae-Dong;Kim, Young-Joong;Lim, Myo-Taeg
    • Journal of Institute of Control, Robotics and Systems
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    • v.13 no.6
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    • pp.547-554
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    • 2007
  • This thesis presents a kind of active noise control(ANC) algorithm for reducing noise due to engine inside a car. The proposed control algorithm is, by using WP(Wavelet Packet), a one improving the instability due to delay of noise transmission and the lack of response ability for the rapid change of noise, which are defects of the existing FXLMS(Filtered-X Least Mean Square) algorithm. The chief character of this system is a thing that faster operation than the FXLMS is implemented by inserting WP in the secondary path. In other words, WP implements parallel operation. Then, the weights of filter in the adaptive algorithm will be updated faster. In addition, because WP have so excellent a resolution, they can process very minute noise. The efficiency of this control algorithm will be demonstrated in the matlab simulation and in the actual experiments by using a Labview program and a car.

Wireless Repeating Interference Canceller Using Delay Estimation Least Mean Square Adaptive Algorithm (지연 추정 LMS 적응 알고리즘을 이용한 무선 중계 간섭 제거기)

  • Kang, Yong-Jin;Song, Joo-Tae;Jeon, Ig-Tae;Kim, Joo-Wan;Ha, Sung-Hee;Van, Ji-Hun;Lee, Jong-Hyun
    • Proceedings of the IEEK Conference
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    • 2007.07a
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    • pp.119-120
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    • 2007
  • The operation of Interference cancellation algorithm for wireless repeater cancellation depends on either existing correlation properties between desired signal and reference signal or not At the time, due to the correlation properties at the ICS system, adaptive algorithms without considering system delay do not function properly. Thus, this system should be oscillated. In this paper, to solve these problems, we use the delayed least mean square algorithm. For the best performance of ICS, the system delays must be estimated. To efficiently estimate the delay of ICS, we use relations between bandwidth and correlation properties of the received signal.

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