• Journal of Digital Convergence
• /
• v.10 no.10
• /
• pp.277-282
• /
• 2012

The speech recognition system can not quickly adapt to varied environmental noise factors that degrade the performance of recognition. In this paper, the echo noise robust HMM learning model using average estimator LMS algorithm is proposed. To be able to adapt to the changing echo noise HMM learning model consists of the recognition performance is evaluated. As a results, SNR of speech obtained by removing Changing environment noise is improved as average 3.1dB, recognition rate improved as 3.9%.

• Journal of KSNVE
• /
• v.8 no.3
• /
• pp.485-493
• /
• 1998

Many of the adaptive noise control systems utilize a form of the least mean square (LMS) algorithms. In the active control of noise, it is common practice to locate an error microphone far from the control source to avoid the near-field effects by evanescent waves. Such a distance between the control source and the error microphone makes a certain level of time-delay inevitable and, hence, may yield undesirable effects on the convergence properties of control algorithms such as filtered-x LMS. This paper discusses the dependence of the convergence rate on the acoustic error path in these popularalgorithms and introduces new algorithms which increase the convergence region regardless of the time-delay in the acoustic error path. Performances of the new LMS algorithms are presented in comparison with those by the conventional algorithms based on computer simulations and experiments.

• Journal of the Korean Institute of Telematics and Electronics
• /
• v.21 no.2
• /
• pp.29-35
• /
• 1984

LMS algorithm is used widely in adaptive filtering because of its simplicity. In this paper it is shown that the one dimensional LMS adaptive filter can be extended in the two dimensional adaptive filter and the methods for improving the convergence rate and the several problems inherent in the two dimensional adaptive filter are discussed.

• 제어로봇시스템학회:학술대회논문집
• /
• 2003.10a
• /
• pp.57-60
• /
• 2003

We have proposed the adaptive noise reduction algorithm using the MDFT. The algorithm proposed use the linear prediction coefficients of the AR method based on Sign algorithm that is the modified LMS instead of the least mean square(LMS). The signals with a random noise tracking performance are examined through computer simulations and confirmed that the high speed adaptive noise reduction processing system is realized with rapid convergence.

• Journal of Korean Society for Atmospheric Environment
• /
• v.2 no.3
• /
• pp.52-56
• /
• 1986

In this paper a adaptive LMS(least mean-square) lattice predictor, which is composed of the adaptive lattice algorithm and LMS algorithm by Widrow-Hopf, is used to predict the future air pollution of the extraordinary levels in the environmental system. This prediction algorithm is applied to the one-step forward prediction of atmospheric CO concentration by using real observed data. Computer simulation proves that the power in the forward error sequences decreases as the number of stages in the lattice is increased.

• Proceedings of the Korean Institute of Intelligent Systems Conference
• /
• 2000.05a
• /
• pp.102-105
• /
• 2000

본 논문에서는 잡음환경에서의 음성신호복원(Speech Enhancement) 시스템 구현을 목적으로 한다 이를 위한 적응필터로서 LMS(Least Mean Square)알고리즘 FIR필터를 제안한다. 또 정밀 필터로서 신경망 필터를 제안한다. 잡음환경에서의 음성신호 복원 시스템은 잡음에 의해 왜곡된 음성신호에서 잡음성분만을 제거함으로써 음성신호를 복원하는 시스템이다. 일반적으로 잡음은 시변특성과, 비선형적인 전달특성을 갖는다. 그러므로 파라미터가 고정된 필터로는 제어하기가 힘들다. 이러한 이유로 본 논문에서는 LMS알고리즘 적응필터를 적용한다. 신경망 필터는 오차 역전파 학습 알고리즘에 의해 오차를 최소화하는 방향으로 필터의 파라미터를 수정한다. 제안한 필터로 잡음환경에서의 음성신호복원 시스템을 구성하고, 실험을 통해 필터의 성능을 확인한다.

• Proceedings of the KIEE Conference
• /
• 2007.10a
• /
• pp.283-284
• /
• 2007

In this paper we introduce a compensation of nonlinearity Heterodyne laser interferometer. The Laser Interferometer is used for length measurement in various industries. However, it has nonlinearity error caused by the imperfect optical equipment. This acts as an obstacle in the measurement improvement. We propose an adaptive error compensation using least mean square(LMS) to improve precision.

• Journal of the Institute of Convergence Signal Processing
• /
• v.4 no.1
• /
• pp.15-26
• /
• 2003

Ultrasonic inspection methods are widely used for detecting flaws in materials. The signal analysis step plays a crucial part in the data interpretation process. A number of signal processing methods have been proposed to classify ultrasonic flaw signals. One of the more popular methods involves the extraction of an appropriate set of features followed by the use of a neural network for the classification of the signals in the feature space. This paper describes an alternative approach which uses the least mean square (LMS) method and expectation maximization (EM) algorithm with the model based deconvolution which is employed for classifying nondestructive evaluation (NDE) signals from steam generator tubes in a nuclear power plant. The signals due to cracks and deposits are not significantly different. These signals must be discriminated to prevent from happening a huge disaster such as contamination of water or explosion. A model based deconvolution has been described to facilitate comparison of classification results. The method uses the space alternating generalized expectation maximization (SAGE) algorithm In conjunction with the Newton-Raphson method which uses the Hessian parameter resulting in fast convergence to estimate the time of flight and the distance between the tube wall and the ultrasonic sensor Results using these schemes for the classification of ultrasonic signals from cracks and deposits within steam generator tubes are presented and showed a reasonable performances.

• Proceedings of the Korean Society of Precision Engineering Conference
• /
• 2008.11a
• /
• pp.97-98
• /
• 2008

• Journal of the Korean Institute of Telematics and Electronics S
• /
• v.36S no.7
• /
• pp.81-89
• /
• 1999

Image obtained by incomplete communication always include noise, blur and distortion, etc. In this paper, we propose and apply the new spatial filter algorithm, called an optimal adaptive generalized stack filter(AGSF), which optimizes adaptive generalized stack filter(AGSF) using neural network weight learning algorithm of back-propagation learning algorithm for improving noise removal and edge preservation rate. AGSF divides into two parts: generalized stack filter(GSF) and adaptive multistage median filter(AMMF), GSF improves the ability of stack filter algorithm and AMMF proposes the improved algorithm for reserving the sharp edge. Applied to neural network theory, the proposed algorithm improves the performance of the AGSF using two weight learning algorithms, such as the least mean absolute(LAM) and least mean square (LMS) algorithms. Simulation results of the proposed filter algorithm are presented and discussed.