• Journal of Broadcast Engineering
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• v.27 no.1
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• pp.91-103
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• 2022

Determination of sound source characteristics such as: sound volume, direction and distance to the source is one of the important techniques for unmanned systems like autonomous vehicles, robot systems and AI speakers. There are multiple methods of determining the direction and distance to the sound source, e.g., using a radar, a rider, an ultrasonic wave and a RF signal with a sound. These methods require the transmission of signals and cannot accurately identify sound sources generated in the obstructed region due to obstacles. In this paper, we have implemented and evaluated a method of detecting and identifying the sound in the audible frequency band by a method of recognizing the volume, direction, and distance to the sound source that is generated in the periphery including the invisible region. A cross-shaped based sound source recognition algorithm, which is mainly used for identifying a sound source, can measure the volume and locate the direction of the sound source, but the method has a problem with "blind spots". In addition, a serious limitation for this type of algorithm is lack of capability to determine the distance to the sound source. In order to overcome the limitations of this existing method, we propose a QRAS-based algorithm that uses rectangular-shaped technology. This method can determine the volume, direction, and distance to the sound source, which is an improvement over the cross-shaped based algorithm. The QRAS-based algorithm for the OSSD uses 6 AITDs derived from four microphones which are deployed in a rectangular-shaped configuration. The QRAS-based algorithm can solve existing problems of the cross-shaped based algorithms like blind spots, and it can determine the distance to the sound source. Experiments have demonstrated that the proposed QRAS-based algorithm for OSSD can reliably determine sound volume along with direction and distance to the sound source, which avoiding blind spots.

• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.5
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• pp.807-812
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• 2020

This paper proposes a blind speech source separation algorithm using a microphone to separate only the target speech source signal in an environment in which various speech source signals are mixed. The proposed algorithm is a model of frequency domain representation based on independent component analysis method. Accordingly, for the purpose of verifying the validity of independent component analysis in the frequency domain for two speech sources, the proposed algorithm is executed by changing the type of speech sources to perform speech sources separation to verify the improvement effect. It was clarified from the experimental results by the waveform of this experiment that the two-channel speech source signals can be clearly separated compared to the original waveform. In addition, in this experiments, the proposed algorithm improves the speech source separation performance compared to the existing algorithms, from the experimental results using the target signal to interference energy ratio.

• The Journal of the Acoustical Society of Korea
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• v.31 no.5
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• pp.332-339
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• 2012

Beamspace transform algorithm transforms spatial-domain data - such as x, y, z dimension - into incidence-angle-domain data, which is called beamspace-domain data. The beamspace transform method is generally used in source localization and tracking, and adaptive beamforming problem. When the beamspace transform method is used in multichannel audio source separation, the inverse beamspace transform is also important because the source image have to be reconstructed. This paper studies the beamspace transform and inverse transform algorithms for multichannel audio source separation system, especially for the beamspace-domain multichannel NMF algorithm.

• Proceedings of the Korea Information Processing Society Conference
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• 2018.10a
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• pp.863-865
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• 2018

본 논문에서는 피부영상을 이용한 호흡 반영 음원 조율 방법을 제안한다. 얼굴 영상으로부터 호흡 신호를 추정하기 위해 ROI(Region of Interest)를 지정하고 지정된 영역의 색상 체계를 RGB에서 YCgCo로 변환한다. 피부 관심 영역으로부터 계산된 Cg색상 데이터 평균값에 필터링을 적용하여 호흡 신호를 검출한다. 검출된 호흡 신호를 통하여 사용자의 호흡 상태를 반영한 음원 조율방법을 제안하고, 이를 구현한 응용 프로그램을 소개한다. 구현한 응용프로그램의 성능평가를 위해 피험자 15명을 대상으로 블라인드 테스트와 MOS 평가방법을 사용하였으며, 실험 결과 9명의 피실험자가 호흡을 반영한 음원과 반영하지 않은 음원에 대한 차이를 느꼈다. 또한, MOS 평가방법으로 두 음원의 선호도를 조사한 결과 총 5점 만점 중 호흡을 반영한 음원이 4점, 원음이 3.6점을 얻었으며 이를 통해 피실험자들이 호흡이 반영된 음원을 선호한다는 결과를 확인하였다.

• Journal of Korea Multimedia Society
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• v.8 no.3
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• pp.328-335
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• 2005

This paper proposes a frequency domain independent component analysis (ICA) algorithm to separate the mixed speech signals using a multiple microphone array By estimating the delay timings using a input cross-correlation, even in the delayed mixture case, we propose a good initial value setting method which leads to optimal convergence. To reduce the calculation, separation process is performed at frequency domain. The results of simulations confirms the better performances of the proposed algorithm.

• Economic and Environmental Geology
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• v.43 no.4
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• pp.349-358
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• 2010

For imaging complex subsurface structures such as salt dome, faults, thrust belt, and folds, seismic prestack reverse-time migration in depth domain is widely used, which is performed by the cross-correlation of shot-domain wavefield extrapolation with receiver-domain wavefield extrapolation. We apply the prestack reverse-time migration, which had been developed at KIGAM, to the seismic field data set of Block 5 in Jeju basin of Korea continental shelf in order to improve subsurface syncline stratigraphy image of the deep structures under the shot point 8km at the surface. We performed basic data processing for improving S/N ratio in the shot gathers, and constructed a velocity model from stack velocity which was calculated by the iterative velocity spectrum. The syncline structure of the stack image appears as disconnected interfaces due to the diffractions, but the result of the prestack migration shows that the syncline image is improved as seismic energy is concentrated on the geological interfaces.

• Proceedings of the Korean Society of Computer Information Conference
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• 2008.06a
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• pp.337-341
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• 2008

음원은 개인의 감정변화에 많은 영향을 주는 것으로 알려져 있다. 하지만 객관적인 근거가 없어 어떤 자극원이 영향을 주는지 알 수가 없다. 사람은 감정은 실시간적으로 변한다. 그렇기 때문에 이것을 측정 할 수 있는 심전도 센서를 가지고 심박변화율을 측정하여 변화되는 자극을 측정 할 수 있다. 음악은 주파수로 이루어져 있으며 음악을 들을 때 동시에 여러 대역에서 음악이 나온다. 이러한 주파수의 변화와 심박의 변화를 분석하면 감정변화의 기반하는 특징을 알 수 있을 것이다. 이것의 기초 단계로 안정감을 주는 음악은 주파수 영역이 저음 영역으로 규칙적이고 반복적이며 파형의 변화가 없다. 저음영역에서 고음영역으로 변화 되는 음원을 사용하여 심박변화율을 살펴봄으로써 자극음원이 사람에게 영향을 끼치는지를 분석하였다.

• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.358-367
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• 2016

SSL (Sound Source Localization) has been applied to several applications such as man-machine interface, video conference system, smart car and so on. But in the process of sound source localization, angle estimation error is occurred mainly due to the non-linear characteristics of the sine inverse function. So an approach was proposed to decrease the effect of this non-linear characteristics, which divides the microphone's covering space into narrow regions. In this paper, we proposed an optimal space dividing way according to the pattern of microphone array. In addition, sound source's 2-dimensional position is estimated in order to evaluate the performance of this dividing method. In the experiment, GCC-PHAT (Generalized Cross Correlation PHAse Transform) method that is known to be robust with noisy environments is adopted and triangular pattern of 3 microphones and rectangular pattern of 4 microphones are tested with 100 speech data respectively. The experimental results show that triangular pattern can't estimate the correct position due to the lower space area resolution, but performance of rectangular pattern is dramatically improved with correct estimation rate of 67 %.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.2
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• pp.114-119
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• 2009

Sound source localization systems in service robot applications estimate the direction of a human voice. Time delay information obtained from a few separate microphones is widely used for the estimation of the sound direction. Correlation is computed in order to calculate the time delay between two signals. Inverse cosine is used when the position of the maximum correlation value is converted to an angle. Because of nonlinear characteristic of inverse cosine, the accuracy of the computed angle is varied depending on the position of the specific sound source. In this paper, we propose an efficient sound source localization system using angle division. By the proposed approach, the region from $0^{\circ}$ to $180^{\circ}$ is divided into three regions and we consider only one of the three regions. Thus considerable amount of computation time is saved. Also, the accuracy of the computed angle is improved since the selected region corresponds to the linear part of the inverse cosine function. By simulations, it is shown that the error of the proposed algorithm is only 31% of that of the conventional a roach.

• The Journal of the Acoustical Society of Korea
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• v.31 no.5
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• pp.317-331
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• 2012

In this paper, we develop a multichannel blind source separation algorithm based on a beamspace transform and the multichannel non-negative matrix factorization (NMF) method. The NMF algorithm is a famous algorithm which is used to solve the source separation problems. In this paper, we consider a beamspace-time-frequency domain data model for multichannel NMF method, and enhance the conventional method using a beamspace transform. Our decomposition algorithm is applied to audio source separation, using a dataset from the international Signal Separation Evaluation Campaign 2010 (SiSEC 2010) for evaluation.