• 로봇학회논문지
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• 제9권4호
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• pp.206-215
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• 2014

Acoustic signal is crucial for the autonomous navigation of underwater vehicles. For this purpose, this paper presents a method of acoustic source localization. The proposed method is based on the probabilistic estimation of time delay of acoustic signals received by two hydrophones. Using Bayesian update process, the proposed method can provide reliable estimation of direction angle of the acoustic source. The acquired direction information is used to estimate the location of the acoustic source. By accumulating direction information from various vehicle locations, the acoustic source localization is achieved using extended Kalman filter. The proposed method can provide a reliable estimation of the direction and location of the acoustic source, even under for a noisy acoustic signal. Experimental results demonstrate the performance of the proposed acoustic source localization method in a real sea environment.

• Journal of Electrical Engineering and Technology
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• 제13권6호
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• pp.2212-2219
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• 2018

This paper describes extended fault location algorithm using estimated remote source impedance. The method uses data only at the local end and the sequence current distribution factors for more accurate estimation. The proposed algorithm can respond to variation of the local and remote source impedance. Therefore, this method is especially useful for transmission lines interconnected to a wind farm that the source impedance varies continuously. The proposed algorithm is very insensitive to the variation in fault distance and fault resistance. The simulation results have shown the accuracy and effectiveness of the proposed algorithm.

• 전기학회논문지
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• 제58권5호
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• pp.885-890
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• 2009

A fault location algorithm using estimated local source impedance after a fault is proposed in this paper. The method uses after fault data only at the local end. It uses the negative sequence current distribution factor for more accurate estimation. The proposed algorithm can keep up with the variation of the local source impedance. Therefore, the proposed algorithm especially is valid for a transmission line interconnected to a wind farm that the equivalent source impedance changes continuously. The performance of the proposed algorithm was verified under various fault conditions using the Simpowersystem of MATLAB Simulink. The proposed algorithm is largely insensitive to the variation in fault distance and fault resistance. The test results show a very high accurate performance.

• The Journal of the Acoustical Society of Korea
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• 제20권4E호
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• pp.52-60
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• 2001

This paper describes real time implementation of subband sound localization system on a floating-point DSP TI TMS320C31. The system determines two dimensional location of an active speaker in a closed room environment with real noise presents. The system consists of an two microphone array connected to TI DSP hosted by PC. The implemented sound localization algorithm is Subband CPSP which is an improved version of traditional CPSP (Cross-Power Spectrum Phase) method. The algorithm first split the input speech signal into arbitrary number of subband using subband filter banks and calculate the CPSP in each subband. It then averages out the CPSP results on each subband and compute a source location estimate. The proposed algorithm has an advantage over CPSP such that it minimize the overall estimation error in source location by limiting the specific band dominant noise to that subband. As a result, it makes possible to set up a robust real time sound localization system. For real time simulation, the input speech is captured using two microphone and digitized by the DSP at sampling rate 8192 hz, 16 bit/sample. The source location is then estimated at once per second to satisfy real-time computational constraints. The performance of the proposed system is confirmed by several real time simulation of the speech at a distance of 1m, 2m, 3m with various speech source locations and it shows over 5% accuracy improvement for the source location estimation.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2001년도 하계학술대회 논문집 A
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• pp.467-469
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• 2001

This paper presents an efficient location estimation methodology for a partial discharge(PD) source in the power transformer. The methodology previously proposed is not suitable for locating a PD source because the ultrasonic signal with medium attenuated through the inner structure. In general, the propagation characteristic of ultrasonic signal with medium is different. Therefore, using this propagation characteristic, we can detect the location of PD source. In this paper, the proposal algorithm finds the PD source using three sensors attached to one side of the transformer without attenuation of the ultrasonic signal and applying the neural network based geometrical method. The proposed methodology demonstrates the effectiveness and validity on an experimental transformer.

• 대한전자공학회논문지
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• 제26권9호
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• pp.1436-1443
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• 1989

In this paper, we propose a maximum-likelihood estimation(MLE) method to obtain the location and the amplitude of the pulses in MPE( multi-pulse excitation)-LPC speech synthesis using multi-pulses as excitation source. This MLE method computes the value maximizing the likelihood function with respect to unknown parameters(amplitude and position of the pulses) for the observed data sequence. Thus in the case of overlapped pulses, the method is equivalent to Ozawa's crosscorrelation method, resulting in equal amount of computation and sound quality with the cross-correlation method. We show by computer simulation: the multi-pulses obtained by MLE method are(1) pseudo-periodic in pitch in the case of voicde sound, (2) the pulses are random for unvoiced sound, (3) the pulses change from random to periodic in the interval where the original speech signal changes from unvoiced to voiced. Short time power specta of original speech and syunthesized speech obtained by using multi-pulses as excitation source are quite similar to each other at the formants.

• 한국음향학회지
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• 제35권1호
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• pp.35-41
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• 2016

본 논문에서는 상관함수와 최소자승 기법에 기반하여 단일 음원의 위치가 원거리 혹은 근거리 모두 가능한 경우, 이에 상관없이 닫힌 형태로 위치를 추정하는 기법을 제안한다. 최근 균일 환영 배열에서 상호상관함수를 이용하여 원거리 단일 음원의 위치를 2차원으로 추정하는 기법이 제안되었으며, 이를 확장하여 근거리 단일 음원의 위치를 3차원으로 추정하는 기법이 제안되었다. 그러나 기존 기법은 음원의 위치가 원거리, 혹은 근거리로 제한된 상황만 다루고 있다. 반면 제안 기법은 먼저 원거리 음원으로 가정하여 거리 독립적으로 방위와 고각을 구하고 이후 거리 추정에서 원거리와 근거리 음원을 구분하여 혼합 환경에 적용가능하다. 시뮬레이션에서는 두 가지 경우 모두에 대해 실험하여 제안 기법의 타당성을 검증하였다.

• 융합정보논문지
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• 제10권9호
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• pp.15-21
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• 2020

본 논문은 다수의 전자전 센서에서 추출된 시간지연 차이정보와 도플러주파수 차이정보를 이용하는 Two-step weighted least-squares 기반의 이동 신호원 위치 및 속도 추정 기법에서, 수집 정보의 이상치를 검출하는 알고리즘을 제안하고자 한다. 다수의 전자전 센서에서 추출되는 정보는 다양한 요인에 의해 정보에 이상치가 발생할 수 있으며, 이를 효과적으로 검출하고 데이터 융합과정에서 이상치를 배제하여 이동 신호원의 위치와 속도 추정의 정확도를 높이고자 한다. 본 논문에서는 이상치를 제외한 최소의 정상치 정보 집합을 추출하고, 이를 기반으로 나머지 정보의 이상치 여부를 확률적으로 판단하는 알고리즘을 제안하였으며, 이를 모의실험을 통해, 정보의 이상치가 효과적으로 제거되어 위치 및 속도 추정의 정확도를 향상시킬 수 있음을 확인하였다. 정상치 거리정보 잡음이 20dB 이하인 경우, 이상치 정보를 효과적으로 제거하여, Cramér-Rao lower bound에 근접한 위치 및 속도 추정 정확도를 얻음을 확인하였다.

• 융합정보논문지
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• 제10권8호
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• pp.53-59
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• 2020

본 논문은 음원 탐지 및 음원 위치를 추정하는 IoT Device 시스템에 관한 것으로, 보다 구체적으로는 복수의 마이크로폰 센서로부터 수집된 음원 신호의 도달 시간차를 분석하여 음원의 방향을 정확히 검출하고, IoT 센서를 이용하여 음원의 발생방향을 추적할 수 있는 음원 방향 탐지 Device를 이용한 시스템이다. 음파를 이용하여 위치를 추정하는 기술은 예전부터 군사적 목적으로 개발되어 왔지만 현재는 이를 응용하여 방범·방재 분야 등에 많이 쓰이고 있다. 이에 따라 본 시스템의 제작을 통해 옥외에 설치한 후 여러 방향에서 음원 발생시켜 성능 시험을 실시하였다. 그 결과 음향 탐지 영역 140dB, 반응시간 1초 이내, 방향 각도 분해능 1° 이내로 매우 정확하게 동작함을 확인할 수 있었다. 향후에는 본 설계안을 바탕으로 빅데이터 분석을 통한 인공지능 알고리즘을 반영하여 보다 신뢰성을 향상시켜 상용화할 계획에 있다.

• 한국음향학회지
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• 제30권3호
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• pp.129-135
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• 2011

An algorithm is presented for estimating the 3-D location (i.e., azimuth angle, elevation angle, and range) of multiple sources with a uniform circular array (UCA) consisting of an even number of sensors. Recently the rank reduction (RARE) algorithm for partly-calibrated sensor arrays was developed. This algorithm is applicable to sensor arrays consisting of several identically oriented and calibrated linear subarrays. Assuming that a UCA consists of M sensors, it can be divided into M/2 identical linear subarrays composed of two facing sensors. Based on the structure of the subarrays, the steering vectors are decomposed into two parts: range-independent 2-D direction-of-arrival (DOA) parameters, and range-relevant 3-D location parameters. Using this property we can estimate range-independent 2-D DOAs by using the RARE algorithm. Once the 2-D DOAs are available, range estimation can be obtained for each source by defining the 1-D MUSIC spectrum. Despite its low computational complexity, the proposed algorithm can provide an estimation performance almost comparable to that of the 3-D MUSIC benchmark estimator.