• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.8
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• pp.759-766
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• 2016

This paper is on a system design of recognizing sound sources and tracing locations from detecting a section of sound sources which is strong in surrounding environmental sounds about sound sources occurring in an abnormal situation by using signals within the section. In detection of the section with strong sound sources, weighted average delta energy of a short section is calculated from audio signals received. After inputting it into a low-pass filter, through comparison of values of the output result, a section strong in background sound is defined. In recognition of sound sources, from data of the detected section, using an HMM(: Hidden Markov Model) as a traditional recognition method, learning and recognition are realized from creating information to recognize sound sources. About signals of sound sources that surrounding background sounds are included, by using energy of existing signals, after detecting the section, compared with the recognition through the HMM, a recognition rate of 3.94% increase is shown. Also, based on the recognition result, location grasping by using TDOA(: Time Delay of Arrival) between signals in the section accords with 97.44% of angles of a real occurrence location.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.05a
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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.

• Proceedings of the KIEE Conference
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• 2007.10a
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• pp.143-144
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• 2007

최근들어 유비쿼터스 환경에서 인간 친화적 인터페이스를 갖는 서비스 로봇에 대한 연구가 매우 활발하다. 인간과 자연스러운 상호작용을 위하여, 화자를 바라보고, 깨끗한 음성신호를 얻는 과정에서 음원의 위치 측정은 필연적이다. 본 논문은 마이크로폰에 도달되는 동일 신호의 도착시간지연(Time Delay of Arrival: TDOA) 특성 행렬을 정의하고, 이를 이용하여, 음원의 위치 측정 시스템을 구현한다. 휴머노이드 로봇에 구현된 실제 시스템을 통하여 방향 검지 및 높이 구분이 잘 이루어짐을 확인한다.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.19 no.4
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• pp.355-362
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• 2009

Sound source localization method based on the time delay of arrival(TDOA) is applied to many research fields such as a robot auditory system, teleconferencing and so on. When multi-microphones are utilized to localize the source in 3 dimensional space, the conventional localization methods based on TDOA decide the actual source position using the TDOAs from all microphone arrays and the detection measure, which represents the errors between the actual source position and the estimated ones. Performance of these methods usually depends on the number of microphones because it determines the resolution of an estimated position. In this paper, we proposed the localization method using spatially mapped GCC functions. The proposed method does not use just TDOA for localization such as previous ones but it uses spatially mapped GCC functions which is the cross correlation function mapped by an appropriate mapping function over the spatial coordinate. A number of the spatially mapped GCC functions are summed to a single function over the global coordinate and then the actual source position is determined based on the summed GCC function. Performance of the proposed method for the noise effect and estimation resolution is verified with the real environmental experiment. The mean value of estimation error of the proposed method is much smaller than the one based on the conventional ones and the percentage of correct estimation is improved by 30% when the error bound is ${\pm}20^{\circ}$.

• Earthquakes and Structures
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• v.11 no.6
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• pp.1143-1164
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• 2016

Time Delay of Arrival (TDOA) estimation methods based on correlation function analysis play an important role in the micro-seismic event monitoring. It makes full use of the similarity in the recorded signals that are from the same source. However, those methods are subjected to the noise effect, particularly when the global similarity of the signals is low. This paper proposes a new approach for micro-seismic monitoring based on cross wavelet transform. The cross wavelet transform is utilized to analyse the measured signals under micro-seismic events, and the cross wavelet power spectrum is used to measure the similarity of two signals in a multi-scale dimension and subsequently identify TDOA. The offset time instant associated with the maximum cross wavelet transform spectrum power is identified as TDOA, and then the location of micro-seismic event can be identified. Individual and statistical identification tests are performed with measurement data from an in-field mine. Experimental studies demonstrate that the proposed approach significantly improves the robustness and accuracy of micro-seismic source locating in mines compared to several existing methods, such as the cross-correlation, multi-correlation, STA/LTA and Kurtosis methods.

• Journal of the Korea Institute of Military Science and Technology
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• v.14 no.4
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• pp.694-699
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• 2011

A closed-form technique is presented for estimating a single source location from a set of noisy time delay measurements between distributed sensors. The localization formula is derived from nonlinear least squares minimization over the unknowns of target range and bearing in polar coordinates. Computer simulation results are provided for the purpose of performance analysis. Constrained least squares minimization method with prior source location information is also discussed.

• JSTS:Journal of Semiconductor Technology and Science
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• v.14 no.2
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• pp.175-183
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• 2014

In this paper, we present the design and implementation of real-time sound localization based on $0.13{\mu}m$ CMOS process. Time delay of arrival (TDOA) estimation was used to obtain the direction of the sound signal. The sound localization chip consists of four modules: data buffering, short-term energy calculation, cross correlation, and azimuth calculation. Our chip achieved real-time processing speed with full range ($360^{\circ}$) using three microphones. Additionally, we developed a dedicated sound localization circuit (DSLC) system for measuring the accuracy of the sound localization chip. The DSLC system revealed that our chip gave reasonably accurate results in an experiment that was carried out in a noisy and reverberant environment. In addition, the performance of our chip was compared with those of other chip designs.

• Journal of KIISE
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• v.43 no.1
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• pp.13-26
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• 2016

This paper proposes a new sound localization algorithm that can improve localization performance based on a tetrahedron-shaped microphone array. Sound localization system estimates directional information of sound source based on the time delay of arrival(TDOA) information between the microphone pairs in a microphone array. In order to obtain directional information of the sound source in three dimensions, the system requires at least three microphones. If one of the microphones fails to detect proper signal level, the system cannot produce a reliable estimate. This paper proposes a tetrahedron- shaped sound localization system with a coordinate transform method by adding one microphone to the previously known triangular-shaped system providing more robust and reliable sound localization. To verify the performance of the proposed algorithm, a real time simulation was conducted, and the results were compared to the previously known triangular-shaped system. From the simulation results, the proposed tetrahedron-shaped sound localization system is superior to the triangular-shaped system by more than 46% for maximum sound source detection.

• Journal of the Korea Institute of Military Science and Technology
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• v.12 no.6
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• pp.768-775
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• 2009

In this paper, the sound source localization system in real time which uses the time delay of arrival signal is proposed. This system uses minimum microphones and surveillance camera for estimation of the sound source localization and sound direction. To apply this system to the military field, four models(model1~model4) are derived. Model 1 can be used to evaluate the sound source localization at the long distance. Model2 and model3 can be applied to estimate the sound direction. Model4 is useful for the special purpose and potable device. It is possible for this system to be used for the military guard and surveillance. As a result of experiments, It is shown that this system can estimate the sound source localization and the sound direction using minimum microphones.

• Proceedings of the KIEE Conference
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• 2003.11c
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• pp.878-881
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• 2003

To acquire an appointed speaker's clear voice signal from inspect-camera, picture-conference or hands free microphone eliminating interference noises needs to be preceded speaker's position automatically. Presumption of sound source position's basic algorithm is about measuring TDOA(Time Difference Of Arrival) from reaching same signals between two microphones. This main project uses ADF(Adaptive Delay Filter) [4] and CPS(Cross Power Spectrum) [5] which are one of the most important analysis of TDOA. From these analysis this project proposes presumption of real time sound source position and improved model NI-ADF which makes possible to presume both directions of sound source position. NI-ADF noticed that if auditory sense of humankind reaches above to some specified level in specified frequency, it will accept sound through activated nerve. NI-ADF also proposes practicable algorithm, the presumption of real time sound source position including both directions, that when microphone loads to some specified system, it will use sounds level difference from external system related to sounds of diffraction phenomenon. In accordance with the project, when existing both direction adaptation filter's algorithm measures sound source, it increases more than twice number by measuring one way. Preserving this weak point, this project proposes improved algorithm to presume real time in both directions.