• Proceedings of the KSPS conference
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• 2006.05a
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• pp.49-52
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• 2006

The performance of a sound source localization system degrades severely in reverberant and noisy environments. In addition, restriction on the distance between microphones, which is required by portable devices, also lower the system performance. This paper compares the sound source localization algorithms based on time delay of arrival, which are robust to reverberation and noises considering microphone sensor distance. As well, post filter which outputs maximum count time delay is adopted to increase the accuracy.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2015.11a
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• pp.84-86
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• 2015

본 논문에서는 2차원으로 배열된 4개의 스피커를 이용한 음향 재생환경에서 청취자의 위치에 따라 최적의 sweet spot을 제공하도록 스피커를 구동하는 방법을 제안한다. 먼저 청취자의 위치를 추적하여 청취자의 귀와 스피커 쌍 후보 사이의 경로의 비율을 구한다. 다음, 청취자의 위치를 고려하여 스피커 쌍 후보들의 스피커 간 거리 비율을 구한다. 마지막으로, 앞서 구한 두 개 거리 비율의 비율이 1에 가까운 스피커 쌍을 구동 스피커로 결정하고 cross-talk cancellation을 음원에 적용하여 청취자에게 제공한다. 모의실험을 통해, 청취자의 위치가 변할 때 제안하는 스피커 구동 방법을 사용하여 기존의 방법보다 청취자에게 넓은 sweet spot을 제공할 수 있음을 확인하였다.

• Proceedings of the KSPS conference
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• 2005.11a
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• pp.117-120
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• 2005

This paper suggests an algorithm that can estimate the direction of the sound source in realtime. Our intelligent service robot, WEVER, is used to implement the proposed method at the home environment. The algorithm uses the time difference and sound intensity information among the recorded sound source by four microphones. Also, to deal with noise of robot itself, the kalman filter is implemented. The proposed method takes shorter execution time than that of an existing algorithm to fit the real-time service robot. The result shows relatively small error within the range of ${\pm}$ 7 degree.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.38 no.3
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• pp.258-263
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• 2010

Sound localization algorithm usually adapts three step process: sampling sound signals, estimating time difference of arrival between microphones, estimate location of sound source. To apply this process in indoor environment, sound localization algorithm must be strong enough against reverberant and noisy condition. Additionally, calculation efficiency must be considered in implementing real-time sound localization system. To implement real-time robust sound localization system we adapt four low cost condenser microphones which reduce the cost and total calculation load. And to get TDOA(Time Differences of Arrival) of microphones we adapt GCC-PHAT(Generalized Cross Correlation-Phase Transform) which is robust algorithm to the reverberant and noise environment. The position of sound source was calculated by using iterative least square algorithm which produce highly accurate position data.

• Journal of the Korea Society for Simulation
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• v.25 no.3
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• pp.53-63
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• 2016

We consider simulation study combining static and mobile decoys for survivability of a surface warship against torpedo attack. It is assumed that an enemy torpedo is a passive acoustic homing torpedo and detects a target within its maximum target detection range and search beam angle by computing signal excess via passive sonar equation, and a warship conducts an evasive maneuvering with deploying static and mobile decoys simultaneously to counteract a torpedo attack. Suggesting the four different decoy deployment plans to achieve the best plan, we analyze an effectiveness for a warship's survival probability through Monte Carlo simulation, given a certain experimental environment. Furthermore, changing the speed and the source level of decoys, the maximum torpedo detection range of warship, and the maximum target detection range of torpedo, we observe the corresponding survival probabilities, which can provide the operational capabilities of an underwater defense system.

• The Journal of the Acoustical Society of Korea
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• v.39 no.3
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• pp.163-169
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• 2020

The asynchronous bistatic sonar needs to estimate direct blast arrival time at a receiver to localize targets, and therefore the direct blast arrival time estimation error could be added to target localization error in comparison with synchronous system. Direct blast especially appears as several peaks at the matched filter output by multipath, thus we compared the first peak detection technique and the maximum peak detection technique of those peaks for direct blast arrival time estimation through sea trial data. The test was performed in a shallow sea with bistatic sonar made up of spatially separated source and line array sensors. Line array sensors obtained the target signal which is generated from the echo repeater. As a result, the first peak detection technique is superior to maximum peak detection technique in direct blast arrival time estimation error. The result of this analysis will be used for further research of target tracking in the asynchronous bistatic sonar.

• Journal of Convergence for Information Technology
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• v.9 no.7
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• pp.106-113
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• 2019

The purpose of this paper is aimed to derive suggestions for convergence technology for effective management of distance education by analyzing a long-term case. The experiment was designed with notebook, smartphone or tablet based robot controlled by a remote instructor and a learner, who have experience of distance learning including robot assisted learning. The tablet based robot has the display system of feedback to speakers. During five months, three types of experiments were conducted randomly and a participant was interviewed thoroughly. The result, like the previous research, demonstrates that the task performance of the learner in telepresence robot-assisted learning was better than that in the notebook, and smartphone based. However, it is believed to be necessary to adjust the system for eye-contact and voice transmission for the remote instructor. The instructor required an additional sight by supplementing an extra camera and automatic direction control to source of sound.

• The Journal of Engineering Geology
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• v.25 no.4
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• pp.587-598
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• 2015

We examined various existing threshold methods for the determination of the first arrival time of acoustic emission (AE), and developed a new variable threshold method that could determine the first arrival time of AE more accurately and more quickly than existing methods. The new method, a modification of an existing threshold method, does not fix the threshold, but applies variable thresholds for the AE signals according to noise analysis. Two- and three-dimensional models were established to test the effectiveness of the new method. It could determine source locations of AE in a two-dimensional model 38.3% more accurately than the pre-existing threshold methods. Its accuracy improvement over the existing methods in a three-dimensional model was about 15.2%. A practical test involved measuring the source locations of AE during three-point bending tests of granite cores. The new method placed the sources closer to the fracture plane than did the pre-existing methods, indicating its superior (and quicker) ability to determine the source locations of AE.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.61 no.2
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• pp.276-283
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• 2012

In this paper, a practical quasi-optimal DOA(direction of arrival) estimator is proposed in order to develop a one-axis gimbaled ultrasonic source tracker for mobile robot applications. With help of the gimbal structure, the ultrasonic moving source tracking problem can be simply reduced to the DOA estimation. The DOA estimation is known as one of the representative long-pending nonlinear filtering problems, but the conventional nonlinear filters might be restrictive in many actual situations because it cannot guarantee the reliable performance due to the use of nonlinear signal model. This motivates us to reformulate the DOA estimation problem in the linear robust state estimation setting. Based on the assumption that the received ultrasonic signals are noisy sinusoids satisfying linear prediction property, a linear uncertain measurement model is newly derived. To avoid the DOA estimation performance degradation caused by the stochastic parameter uncertainty contained in the linear measurement model, the recently developed NCRKF (non-conservative robust Kalman filter) scheme [1] is utilized. The proposed linear DOA estimator provides excellent DOA estimation performance and it is suitable for real-time implementation for its linear recursive filter structure. The effectiveness of the suggested DOA estimation scheme is demonstrated through simulations and experiments.

• The Journal of the Acoustical Society of Korea
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• v.33 no.5
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• pp.326-334
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• 2014

This paper proposes a novel covariance matrix estimation scheme for minimum variance distortionless response (MVDR) beamforming. By accurately tracking direction-of-sound source arrival (DoA) information using audio-visual sensors, the covariance matrix is efficiently estimated by adopting a variable forgetting factor. The variable forgetting factor is determined by considering signal-to-interference ratio (SIR). Experimental results verify that the performance of the proposed method is superior to that of the conventional one in terms of interference/noise reduction and speech distortion.