• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 하계종합학술대회 논문집(5)
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• pp.81-84
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• 2000

This paper proposes a new sigma delta modulator structure based on a LMS(Least Mean Square) algorithm that minimizes the quantization noise. The proposed architecture provides 40dB SNR improvement and 35dB wider dynamic range over conventional sigma delta modulation. The proposed architecture provides superior performance especially when the input signal is small.

• 한국경영과학회:학술대회논문집
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• 대한산업공학회/한국경영과학회 1990년도 춘계공동학술대회논문집; 한국과학기술원; 28 Apr. 1990
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• pp.254-262
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• 1990

A predictive functional relationship model is presented for the calibration problem in which the standard as well as the nonstandard measurements are subject to error. For the estimation of the relationship between the two measurements, the ordinary least squares and maximum likelihood estimation methods are considered, while for the prediction of unknown standard measurementswe consider direct and inverse approaches. Relative performances of those calibration procedures are compared in terms of the asymptotic mean square error of prediction.

• 한국통신학회논문지
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• 제31권8C호
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• pp.801-805
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• 2006

A normalization approach to coefficient adaptation in the escalator(ESC) filter structure that conventionally employs least mean square(LMS) algorithm is introduced. Using Taylor's expansion of the local error signal, a normalized form of the ESC-LMS algorithm is derived. Compared with the computational complexity of the conventional ESC-LMS algorithm employs input power estimation for time-varying convergence coefficient using a single-pole low-pass filter, the computational complexity of the proposed method can be reduced by 50% without performance degradation.

• 한국지능시스템학회논문지
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• 제10권4호
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• pp.324-329
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• 2000

본 논문에서는 잡음환경에서의 음질향상(Speed Ehnacement) 시스템 구현을 목적으로 한다. 이를 위한 적응필터로서 LSM(Least Mean square)알고리즘 FIR필터를 적용한다. 또 정밀 필터로서 다충신경망(MLP, Multi-Layer Perceptorn) 필터를 적용한다. 잡음환경에서의 음성신호 복원 및 음질향상 시스템은 잡음에 의해 왜곡된 음성신호에서 잡음성분만을 제거함으로써 음성신호를 복원하는 시스템이다. 신경망 필터는 오차 역전과 학습 알고리즘에 의해 오차를 최소화 하는 방향으로 필터의 피라미터를 수정한다. 제안한 필터로 잡음환경에서의 음성신호복원 시스템을 구서오하고, 실험을 필터의 성능을 확인한다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(4)
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• pp.413-416
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• 2002

In this paper, a speech enhancement system using microphone array with adaptive Post-Processing is implemented in real-lime with TMS320C6201 DSP. It consists of delay-and-sum beamformer and adaptive post-processing filters with NLMS (Normalized Least Mean Square) algorithm. THS1206 ADC is used for collection of 4-channel microphone signals. Sizes of program memory, data ROM and data RAM of the implemented system are 15,744, 748 and 47,540 bytes, respectively. Finally 21.839${\times}$106 clocks per second is required for real-time operation.

• 한국지능시스템학회:학술대회논문집
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• 한국퍼지및지능시스템학회 2000년도 춘계학술대회 학술발표 논문집
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• pp.102-105
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• 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.

• Journal of electromagnetic engineering and science
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• 제8권4호
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• pp.139-143
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• 2008

In this paper, a new scheme to calculate the weighting factor of the 2-D isotropic-dispersion finite difference time domain(ID-FDTD) is proposed. The weighting factor in [1] was formulated in free space, so that it may not be optimal in dielectric media. Therefore, the weighting factor was reformulated by considering the material properties and using the least mean square method. As a result, a minimum numerical dispersion error for any dielectric media is guaranteed.

• Journal of information and communication convergence engineering
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• 제4권1호
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• pp.10-12
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• 2006

In this paper, we present a method to evaluate the BER performance of a coherent QPSK system using an adaptive array to eliminate CCI and demonstrate closed-form expressions for obtaining exact BER of the desired user for the case in which the time delays of all users are equal. The theoretical results are verified by computer simulation under the assumption that Least Mean Square beamforming algorithm is employed.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2007년도 심포지엄 논문집 정보 및 제어부문
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• pp.283-284
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• 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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(1)
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• pp.153-156
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• 2002

An adaptive predistorr is proposed to compensate for the nonlinear distortion of a high power amplifier (HPA) in 16 QAM system. It fumed out that the proposed predistorter using a variable step-size least mean square (VSSLMS) algorithm is stable and can reduce the Total Distortion (TD) to 0. 1dB at the HPA output backoff=0.0 dB.