• 한국음향학회지
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• 제43권1호
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• pp.89-94
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• 2024

Ultra-Short BaseLine(USBL) 은 센서 간격이 좁은 배열을 사용하기 때문에 위치 추정 성능 향상을 위해서는 정밀한 동기화가 필요하다. 그러나 수중 환경은 비교적 강한 잡음과 다중 경로 및 도플러 등의 수중 음향 채널로 인해 동기화 오류가 발생하여 위치 추정 성능이 저하된다. 본 논문에서는 수중 USBL 시스템의 위치 추정 성능을 향상시키기 위한 공분산 기반 동기 보상 기법을 제안한다. 제안 방법은 상호상관을 통해 신호를 정렬한 후, 정렬된 신호의 공분산을 계산한다. 공분산에서 동기 오차는 위상차와 선형적으로 관련되어 있으므로 위상차를 공분산으로부터 추정하여 동기 오차를 보상한다. 전산 모의실험을 통해 제안 방법이 기존 상호상관 방법보다 우수한 위치 추정 성능을 가지는 것을 보였다.

• 정보과학회 논문지
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• 제43권1호
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• pp.13-26
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• 2016

본 연구에서는 임의 공간에서 정사면체 형태의 마이크로폰 어레이(microphone array)를 이용하여 음원(sound source)추적 성능을 개선할 수 있는 알고리즘을 제안하였다. 음원추적 시스템은 마이크로폰 어레이의 각 마이크로폰에 도착하는 음원신호의 도착 지연시간(TDOA, Time Delay Of Arrival) 정보를 이용하여 음원의 방향성 정보를 추정한다. 임의 3차원 공간에서 음원추적을 위해서는 최소 3개 이상의 마이크로폰이 필요하다. 3개 마이크로폰으로 구성된 음원추적 시스템의 경우 만약 1개의 마이크로폰이라도 신호 오차가 발생한다면 정확한 음원 방향성 추정이 불가능하다. 본 연구에서는 이러한 문제점을 개선하기 위하여 1개의 마이크로폰을 추가한 정사면체 형태(tetrahedron shaper)의 마이크로폰 어레이를 구성하고 좌표변환 기법을 이용하여 주변 잡음이나 오류에 강인한 새로운 음원추적 알고리즘을 제안하였다. 제안 알고리즘의 성능을 입증하기 위하여 3개의 마이크로폰을 이용한 삼각형 기반 음원추적 시스템과 본 연구에서 제안한 정사면체 기반 음원추적 시스템에 대하여 실시간 비교 실험을 수행하였으며, 실험 결과 제안된 정사면체 기반의 시스템이 최대 약 16% 이상의 향상된 검출율을 보였다.

• 로봇학회논문지
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• 제12권4호
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• pp.402-410
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• 2017

Acoustic based localization is essential to operate autonomous robotic systems in underwater environment where the use of sensorial data is limited. This paper proposes a localization method using artificial underwater acoustic sources. The proposed method acquires directional angles of acoustic sources using time difference of arrivals of two hydrophones. For this purpose, a probabilistic approach is used for accurate estimation of the time delay. Then, Gaussian sum filter based SLAM technique is used to localize both acoustic sources and underwater vehicle. It is performed by using bearing of acoustic sources as measurement and inertial sensors as prediction model. The proposed method can handle directional ambiguity of time difference based source localization by generating Gaussian models corresponding to possible locations of both front and back sides. Through these processes, the proposed method can provide reliable localization method for underwater vehicles without any prior information of source locations. The performance of the proposed method is verified by experimental results conducted in a real sea environment.

• The Journal of the Acoustical Society of Korea
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• 제20권1E호
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• pp.31-37
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• 2001

Matched Field Processing (MFP) is a successive process of correcting mismatches between true and assumed parameters by matching the measured acoustic field data with numerically simulated data which we call replica. The MFP is widely used both in geo-acoustic parameter inversions and in source localizations. Whether a certain parameter can be inverted effectively or whether a source can be localized correctly depends on the amount of the influence that a parameter has on the acoustic field during the matching process. Sensitive parameters can be better estimated than the less sensitive ones in MFP. On the contrary, the sensitive parameters affect adversely on the source localization results when they have uncertainties. In this paper, a sensitivity index is defined based upon the field variation resulting from the perturbed parameters. Numerical test results show that the index behaves in accordance with the results of source localization under a mismatched environment and also with the inversion solutions.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2007년도 춘계학술대회논문집
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• pp.1080-1084
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• 2007

Many different methods for sound source localization have been developed. Most of them mainly depend on time delay of arrival (TDOA) or on empirical or analytic head related transfer functions (HRTFs). In real implementation, since the direct path between a source and a sensor is interrupted by obstacles as like a head or body of robot, it has to be considered the number of sensors as well as their positions. Therefore, in this paper, we present the methods, which are included sensor position problem, to localize the sound source with 4 microphones to cover the 3D space. Those are modified two-step TDOA methods. Our conclusion is that the different method has to be applied in case to be different microphone position on real robot platform.

• 제어로봇시스템학회논문지
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• 제14권8호
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• pp.767-773
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• 2008

This paper presents a localization system based on the use of electric compass and tracking IR light source. Digital RGB(Red, Green, Blue)signal of digital CMOS Camera is sent to CPLD which converts the color image to binary image at 30 frames per second. CMOS camera has IR filter and UV filter in front of CMOS cell. The filters cut off above 720nm light source. Binary output data of CPLD is sent to DSP that rapidly tracks the IR light source by moving Camera tilt DC motor. At a robot toward north, electric compass signals and IR light source angles which are used for calculating the data of the location system. Because geomagnetic field is linear in local position, this location system is possible. Finally, it is shown that position error is within ${\pm}1.3cm$ in this system.

• 대한원격탐사학회:학술대회논문집
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• 대한원격탐사학회 1998년도 Proceedings of International Symposium on Remote Sensing
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• pp.138-148
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• 1998

The aim of the work described in this paper is to develop a complex underground acoustic system which detects and localizes the origin of an underground hammering sound using an array of hydrophones located about loom underground. Three different methods for the sound localization will be presented, a time-delay method, a power-attenuation method and a hybrid method. In the time-delay method, the cross correlation of the signals received from the way of sensors is used to calculate the time delays between those signals. In the power-attenuation method, the powers of the received signals provide a measure of the distances of the source from the sensors. A new hybrid method has been developed for estimating the origin of the underground acoustic source by coupling both methods. The Nelder-Meade simplex search algorithm is then used to numerically estimate the position of the source in those methods. For each method the sound localization is carried out in three dimensions underground. The distance between the true and estimated origins of the source is in some cases less than 6m for a search area of radius 250m.

• 대한전자공학회논문지SP
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• 제46권2호
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• pp.24-29
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• 2009

음원 추정 시스템은 로봇, 화상회의, CCTV(Closed-circuit television) 시스템에 쓰인다. 이러한 음원 추정 시스템은 사람을 대상으로 하며 사람이 말하는 동안 여러 개의 음성 데이터 프레임을 입력받을 수 있다. 본 논문에서는 입력된 음성 데이터 프레임으로부터 정확한 각도를 계산 할 수 있는 음성 데이터 프레임을 선별하여 각도 추정 오차를 줄이는 방법에 대해서 제안한다. 또한 선별된 데이터를 각도로 변환한 후 메디언 필터를 적용하여 음원 추정 시스템의 오차를 줄일 수 있다. 제안된 시스템을 이용하여 실험한 후 각도 추정 오차 평균이 31%감소함을 보인다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 2004년도 춘계학술발표대회 논문집 제23권 1호
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• pp.81-84
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• 2004

In this paper, we propose a virtual sound localization algorithm which improves the sound localization accuracy and sound color preservation for two channel and multi-channel surround speaker layouts. In conventional CPP laws, the sound direction is different from the panning angle and the sound color is different from real sound source especially when the speakers are spread out widely. To overcome this drawback, we design a virtual sound localization algorithm using directional psychoacoustic criteria (DPC) and sound color compensator (SCC). The analysis results show that in the case of the proposed system, the sound direction is the same as the panning angle in the audible frequency range and the sound color is less deviated from a real sound source than the conventional CPP law. In addition, its performance is verified by means of subjective tests using a real sound source.

• 한국정보처리학회:학술대회논문집
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• 한국정보처리학회 2012년도 춘계학술발표대회
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• pp.961-963
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• 2012

The sound source localization system is used to some area such as robotic system, object localization system, guarding system and medicine. So time delay estimation and angle estimation of sound direction are studied until now. These days time delay estimation is described in LabVIEW which is used to create innovative computer-based product and deploy measurement and control systems. In this paper, the development of signal monitoring platform is presented for sound source localization. This platform is designed in virtual instrument program and implemented in two stages. In first stage, data acquisition system is proposed and designed to analyze time delay estimation using cross correlation. In second stage, data obtaining system which is applied and designed to monitor analog signal processing is proposed.