• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2014.11a
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• pp.111-114
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• 2014

본 논문에서는 다양한 스테레오 환경에서도 정확한 음원 위치 추정이 가능한 방법을 제안한다. 기존의 음원 위치 추정 방법은 방향성을 가지고 있는 주성분 신호와 방향성이 없는 주변 성분으로 구성된 스테레오 환경에서만 음원의 위치 추정이 가능했다. 그러나 현재 제공되고 있는 스테레오 신호는 방향성을 가지는 다수의 음원으로 구성되어있고, 기존의 음원 위치 추정 방법으로는 정확한 음원 위치 추정이 어렵다. 이와 같은 문제 때문에 다수의 음원을 분리한 뒤, 음원의 위치를 추정하는 방법이 제안되었다. 그러나 음원의 분리 과정에서 생기는 분리 오차가 커서 음원 위치 추정이 정확하지 않다. 이에 본 논문에서는 정확한 음원 위치 추정을 위하여 음원 분리와 음원 위치 추정이 통합된 새로운 알고리즘을 제안한다. 제안한 알고리즘은 음원 위치를 기존의 방법보다 정확하게 추정하는 것을 확인할 수 있었다.

• Proceedings of the Korea Information Processing Society Conference
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• 2012.04a
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• pp.950-952
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• 2012

음원 위치추정 시스템은 일반적으로 여러 개의 마이크에서 수집된 음원의 시간 간격을 이용해 음원의 위치를 추정하는 방식을 적용한다. 본 논문에서는 감시카메라에 적합한 4개의 마이크로폰을 이용한 음원 위치추정 시스템에서 마이크로폰에 수신된 음원의 도착순서를 이용해 음원의 위치를 추정하는 알고리즘을 제안하였다. 제안한 알고리즘을 시뮬레이션 프로그램을 통해 검증한 결과, 음원 추정각도의 오차는 $2^{\circ}{\sim}11.25^{\circ}$로 확인되었으며, 이는 실제각도의 오차범위인 $0^{\circ}{\sim}22.5^{\circ}$ 내에 해당하기 때문에 추정각도의 오차가 최대로 발생하더라도 음원이 발생한 위치를 파악 할 수 있음을 의미한다.

• The Journal of the Acoustical Society of Korea
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• v.35 no.1
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• pp.35-41
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• 2016

Based on correlation and least square method, a closed-form algorithm for estimating the location of mixed far-field and near-field source is presented using the Uniform Circular Array (UCA). Recently, for a homogeneous circular arrangement case, a correlation based closed-form algorithm is proposed to estimate 2-D angle (azimuth, elevation) and the extended algorithm is proposed to 3-D location (azimuth, elevation, range). These algorithms assume the far-field source or near-field source only. Therefore, for mixed source localization, the proposed algorithm estimates source location with the assumption of far-field source, and then estimates the range to distinguish the far-field from the near-field source. For both cases, numerical experiments have been performed, which confirmed the validity of the proposed algorithm.

• The Journal of the Acoustical Society of Korea
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• v.26 no.8
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• pp.415-425
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• 2007

This paper proposes a method of underwater source localization using the wideband interference patterns matching. By matching two interference patterns in the spectrogram, it is estimated a ratio of the range from source to sensor5, and then this ratio is applied to the Apollonius circle. The Apollonius circle is defined as the locus of all points whose distances from two fixed points are in a constant value so that it is possible to represent the locus of potential source location. The Apollonius circle alone, however still keeps the ambiguity against the correct source location. Therefore another equation is necessary to estimate the unique locus of the source location. By estimating time differences of signal arrivals between source and sensors, the hyperbola equation is used to get the cross point of the two equations, where the point being assumed to be the source position. Simulations are performed to get performances of the proposed algorithm. Also, comparisons with real sea experiment data are made to prove applicability of the algorithm in real environment. The results show that the proposed algorithm successfully estimates the source position within an error bound of 10%.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.185-188
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• 1998

본 연구에서는 256(16$\times$16)개의 마이크로폰 정방형 배열에 의한 음향 홀로그래피 시스템을 제작하고, FFT에 의한 음향 홀로그래피법 알고리즘을 이용한 음원 위치 추정에 관하여 기술한다. 본 연구에서 설계한 측정 시스템은 방사된 음들을 동시 수음함으로서 실시간 데이터 처리가 가능하다. 또한 환경 잡음이 존재하는 실음장에서도 계측시간을 단축함과 동시에 고분해능으로 안정하게 음원의 위치를 추정할 수 있다. 본 연구의 타당성을 검증하기 위해 SYSNOISE에 의한 음장해석과 음향 홀로그래피 알고리즘을 이용하여 마이크로폰 간격 및 측정면 크기, 측정거리의 최적 조건을 구한 후 실음장 측정 실험에 적용하였다. 수치 시뮬레이션과 무향실에서 실험 데이터에 의해 음원 위치를 추정한 결과 유사한 결과를 얻었다.

• 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.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 %.

• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.437-442
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• 2018

In this paper, we propose a method for estimating the position of a source in a closed form when a single source has coherently distributed property against a circular array. When a sound source reaches a sensor through multipath environments, it is seen as a distributed source and can be represented by four variables: the nominal azimuth, nominal elevation, azimuth angular spread, elevation angular spread. Therefore, it requires a lot of computation by a search method such as DSPE (Distributed Source Parameter Estimator). In this paper, we propose a method of estimating the nominal azimuth and elevation angle in a closed form using correlation function and least squares method for fast position estimation. In particular, if the source is assumed as Gaussian distribution model, the standard deviation is also estimated in a closed form. In the simulation, the validity of the proposed method is confirmed by comparing with the DSPE.

• The Journal of the Acoustical Society of Korea
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• v.24 no.3
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• pp.166-170
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• 2005

This paper describes the experimental study for the position estimation method of underwater sound source using the Nearfield Acoustic Holography. The result confirms that it can be used in the identification of underwater noise sources. The sound sources in the experimental work consists of 2 spherical projectors and the near-Held sound pressure is measured in the hologram plane. From the cross-power spectra of the measured data, the complex sound pressures on the hologram plane is derived and its spatial transformation gives sound fields in a source region. The obtained sound fields in a source region showed that the position of each sound source and their relative source strength are exactly estimated. In conclusion, this technique can be applied for estimation of each source position and its relative strength contribution for the underwater multiple sound sources.

• The Journal of the Acoustical Society of Korea
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• v.43 no.1
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• pp.72-77
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• 2024

Recently, the necessity for sound source localization has grown significantly across various industrial sectors. Among the sound source localization methods, sound intensimetry has the advantage of having high accuracy even with a small microphone array. However, the increase in localization error at high Helmholtz numbers have been pointed out as a limitation of this method. The study proposes a method to compensate for the bias error of the measured sound intensity vector according to the Helmholtz numbers by applying deep learning. The method makes it possible to estimate the accurate direction of arrival of the source by applying a dense layer-based deep learning model that derives compensated sound intensity vectors when inputting the sound intensity vectors measured by a tetrahedral microphone array for the Helmholtz numbers. The model is verified based on simulation data for all sound source directions with 0.1 < kd < 3.0. One can find that the deep learning-based approach expands the measurement frequency range when implementing the sound intensimetry-based sound source localization method, also one can make it applicable to various microphone array sizes.