• The Journal of the Acoustical Society of Korea
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• v.30 no.3
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• pp.115-122
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• 2011

In this paper, we present a method to estimate an accurate real-time sound source direction based on time delay of arrival by using generalized cross correlation with four cross-type microphones. In general, existing systems have two disadvantages such as system embedding limitation due to the necessity of data acquisition for signal processing from microphone input, and real-time processing difficulty because of the increased number of channels for sound direction estimation using DSP processors. To cope with these disadvantages, the system considered in this paper proposes hardware design for enhanced real-time processing using microphone array signal processing. An accurate direction estimation and its design time reduction is achieved by means of an efficient hardware design using spatial segmentation methods and verification techniques. Finally we develop a system which can be used for embedded systems using a sound codec and an FPGA chip. According to experimental results, the system gives much faster real-time processing time compared with either PC-based systems or the case with DSP processors.

• Journal of Digital Convergence
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• v.13 no.12
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• pp.143-149
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• 2015

This study aims to analyze the predict capability of some of the popular software NHPP reliability models(Goel-Okumo model, delayed S-shaped reliability model and Rayleigh distribution model). The predict capability analysis will be on two key factors, one pertaining to the degree of fitment on available failure data and the other for its prediction capability. Estimation of parameters for each model was used maximum likelihood estimation using first 80% of the failure data. Comparison of predict capability of models selected by validating against the last 20% of the available failure data. Through this study, findings can be used as priori information for the administrator to analyze the failure of software.

• 제어로봇시스템학회:학술대회논문집
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• 1986.10a
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• pp.128-133
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• 1986

설비진단에 응용되는 신호처리의 기법으로는 파워스펙트럼, 바이스펙트럼, 켑스트럼 등이 사용되었다. 파워스펙트럼은 이론적인 면과 계산과정 그리고 신호처리에서의 적용방법등이 잘 알려져서 성공적으로 사용되어져 왔다. 특히 음향분야에서는 여러가지 응용기술이 개발되어 실제계에 적용되고 있으며 계측장비도 파워스펙트럼 해석법에 알맞게 개발되어져 왔다. 파워스펙트럼해석법을 사용하여 진동계를 구성하는 각 요소들의 고유진동수와 진동계 전체를 나타내는 진동파들의 주파수성분 간의 관계에 의하여 진동의 원인 및 소음원 등을 추정하는 것이 가능하다. 그러나 파워스펙트럼은 일반적으로 정상적인 신호를 갖는 진동계에 대한 해석 일 때는 그 이론과 실제가 잘 일치하지만, 진동계 자체가 항시 임의의 주파수를 갖고서 움직일 때 그 해석에는 다음과 같은 문제점이 생긴다. 첫째, 불규칙한 진동계에서는 규칙적인 진동계보다 잡음의 영향을 많이 받기 때문에 실제로 잡음이 진동계의 고유주파수 부근에 있을 경우에는 파워스펙트럼해석으로는 불가능한 경우가 있다. 둘째, 진동파 중에 포함되어 있는 위상이라는 중요한 정보가 없다. 셋째, 시간지연에 따른 진동계의 정확한 정보를 얻을 수 없다. 이상에서 볼 때 파워스펙트럼해석법은 한계가 있음을 알 수 있다. 따라서 본 논문은 바이스펙트럼이라는 해석법을 사용하여 정상과정에서 비정상과정으로 시간지연에따라 변하는 진동계 또는 정상적인 진동계의 저주파에서의 상호간섭 정도 및 위상관계를 관찰함으로써 파워스펙트럼과 비교하여 바이스펙트럼해석법의 타당성을 검토한다. 바이스펙트럼의 실제적인 계산방법은 P. J. Huber가 세가지 접근 방법을 제안했는데 시간영역에서의 평균화를 행하여 계산하는 법, 연속된 기록들을 평균화하는 것, 주파수 영역에서의 평균화를 행하는 것 등이 있다. 본 논문에서는 FFT를 먼저 행하고 파워스펙트럼과 바이스펙트럼 및 바이코히어런스를 구하였다. 그러나 바이스펙트럼해석법은 수치해석적인 면에서 볼 때 파워스펙트럼해석법에 비하여 미약한 점이 많고 통계학적인 그 의미가 확실하게 알려져 있지 않기 때문에 본 논문에서는 시뮬레이션을 통하여 그 물리적 의미를 규명하고져 한다.

• Journal of Institute of Control, Robotics and Systems
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• v.6 no.11
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• pp.963-967
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• 2000

This paper deals with a sliding mode control with uncertainty adaptation for the robust stabilization of input-delay systems with unknown uncertainties. A sliding surface including a state predictor is employed to compensate for the effect of the input delay. The proposed method does not need a priori knowledge of upper bounds on the norm of uncertainties, but estimates those upper bounds by adaptation laws based on the sliding surface. Then, a robust control law with the uncertainty adaptation is derived to ensure the existence of the sliding mode. A numerical example is given to illustrate the design procedure.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.8
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• pp.1619-1624
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• 2012

If multiple Tags in the RFID System transmit their IDs to the Reader at the same time, tag identification time is delayed due to collisions. Therefore, to reduce the reader's identification time, an efficient anti-collision technology is needed. In this paper, we propose a hybrid anti-collision method based on the QT and DFSA. Then, the performances of proposed method are compared with the existing method through extensive simulations.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.113-116
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• 1999

신호 대 잡음비가 낮은 환경에서 센서 어레이에 입사된 입력 신호 사이의 지연시간 추정(Time Delay Estimation, 이하 TDE)은 높은 분해능이 요구된다. 본 연구에서는 높은 분해능의 TDE를 구하기 위해 상호상관함수(Cross-Correlation)에 평균 차 함수(Average Magnitude Difference function, AMDF)의 역수를 가중한 MCC (Modified Cross-Correlation)알고리즘을 제안한다. 모의신호를 사용한 수치 시뮬레이션 실험으로 종래의 AMDF, Cross-Correlation 알고리즘과 본 연구에서 제안한 MCC 알고리즘의 분해능을 비교 분석하였다 각 알고리즘의 TDE 결과에 대해 STFT(Short Time fourier Transform)에 의한 시간 주파수 해석을 한 결과 MCC알고리즘을 사용하여 TDE 분해능이 향상되었음을 보고한다.

• Journal of Institute of Control, Robotics and Systems
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• v.4 no.5
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• pp.630-635
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• 1998

The objective of this paper is to propose a new efficient technique for the estimation of time-delay parameters using wavelet theory and third-order cumulants, yielding good performance even in the case of low SNR. In particular, band-limited non-Gaussian signals with non-zero skewness and spatially correlated Gaussian noises are considered here. The approach is based on the fact that the effects of spatially correlated Gaussian noises on time-delay estimation can be reduced by using the projection sequences (based on the redundant wavelet decomposition) of given measurements in the higher-order cumulant domain. Finally, the performance of the proposed approach is demonstrated using simulations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.1978-1982
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• 2001

In a Multi-carrier modulation system, CP (Cyclic prefix) is inserted in the transmit tame in order to eliminate the ISI (Intersymbol Interference) and ICI (Interchannel Interference) caused by delay spread of a received signal, which in rum degrades the throughput of the system. TEQ (Time-domain equalizer) improves the system throughput by shortening the CIR (Channel Impulse Response) time and maintaining the CP length to the minimum regardless of the channel condition. In this paper, a new MMSE (Minimum Mean Square Error) TEQ algorithm is proposed and its performance is analyzed in order to speed up computing the optimum tap coefficients of the equalizer by employing Newton method.

• Journal of Advanced Navigation Technology
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• v.14 no.5
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• pp.625-631
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• 2010

In this paper, robustness performance of SNUR message is compared with those of existing ones, RTCM(Radio Technical Commission for Maritime Services) v2 MT(Message Type) 18/19 and MT 20/21 under a poor broadcast condition such as temporary data loss or disconnection We defined the temp oral data loss as 2 second delay and reconnection after disconnection as 7 second latency, and then evaluated its robustness for each latency case by double differentiating the observables. Our result shows that SNUR protocol method can reduce the latency error of the existing RTCM messages by 30~60%. Moreover, a rover using SNUR message, whose latency error is bounded within 1/4 L1 wave length, can figure out its own fixed position continuously in spite of 7 second disconnection, while the other using RTCM message, whose error is larger than half wave length, cannot keep its previous fixed solution.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.2C
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• pp.246-253
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• 2010

In this paper, a method of removing the space skew and cross-talk cancellation is proposed where the self-generated signals from the subject are used to obtain the subject's location. In the proposed method, the good spatial sound image is maintained even when the listener moves from the sweet spot. Two major parts of the proposed method are as follows: listener position tracking using the stimuli from the subject and removal of the space skew and cross-talk signals. Listener position tracking is achieved by estimation of the time difference of arrival (TDoA). The position of the listener is then computed using the Talyer-series estimation method. The head-related transfer functions (HRTF) are used to remove the space skew and cross-talk signals, where the direction of the HRTF is given by the one estimated from the listener position tracking. The performance evaluation is carried out on the signals from the 100 subjects that are composed of the 50 female and 50 male subjects. The positioning accuracy is achieved by 70%~90%, under the condition that the mean squared positioning error is less than $0.07m^2$. The subjective listening test is also conducted where the 27 out of the 30 subjects are participated. According to the results, 70% of the subjects indicates that the overall quality of the reproduced sound from the proposed method are improved, regardless of the subject's position.