• The Journal of the Acoustical Society of Korea
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• v.11 no.2
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• pp.59-67
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• 1992

Recently, the problem which actively control the unwanted noise propagated from the technical structure by the generated secondary sound has become considerable topic from the environmental preservation point of view. In most of these studies, active noise control deals with a plane wave propagation at low frequency using adaptive filtering techniques. On the other hand, in real acoustic systems are mostly short due to the limitation of geometric configuration. In this case, the acoustic properties such as reflections and resonances inside the acoustic system should be considered. In this paper, the acoustic modeling method for short length duct was introduced using the transfer matrix method, and the active noise control problem was investigated with \implementation of FIR filter for the transfer function of control system derived from this modeling method. The identification methods for the acoustic model of actual control system was proposed by numerical computation technique based on the estimation of optimal FIR filter coefficients. The acceptable attenuation on the real acoustic system and stability of the controller are predicted in this computational simulation.

• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.395-402
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• 2016

In this paper, a wideband adaptive beamforming approach for an acoustic vector sensor linear array is presented. It is a very important issue to estimate the stable covariance matrix for adaptive beamforming. In the conventional wideband adaptive beamforming based on coherent signal-subspace (CSS) processing, the error of bearing estimates is resulted from the focusing matrix estimation and the large number of data snapshot is necessary. To alleviate the estimation error and snapshot deficiency in estimating covariance matrix, the steered covariance matrix method in the pressure sensor is extended to the vector sensor array, and the subarray technique is incorporated. By this technique, more accurate azimuth estimates and a stable covariance matrix can be obtained with a small number of data snapshot. Through simulation, the azimuth estimation performance of the proposed beamforming method and a wideband adaptive beamforming based on CSS processing are assessed.

• The Journal of the Acoustical Society of Korea
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• v.33 no.2
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• pp.118-125
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• 2014

An approximation of speed of sound in the measurement plane is essential for the inverse estimation of temperature. To this end, an inverse problem relating the measured retarded time data in between set of sensors and actuators array located on the wall is formulated. The involved transfer matrix and its coefficient vectors approximate speed of sound of the measurement plane by using the radial basis function with finite number of interpolation points deployed inside the target field. Then, the temperature field can be reconstructed by using spatial interpolation technique, which can achieve high spatial resolution with proper number of interpolation points. A large number of retarded time data of acoustic paths in between sensors and arrays are needed to obtain accurate reconstruction result. However, the shortage of interpolation points due to practical limitations can cause the decrease of spatial resolution and deterioration of the reconstruction result. In this works, a regeneration for obtaining the additional retarded time data for an arbitrary acoustic path is suggested to overcome the shortage of interpolation points. By applying the regeneration technique, many interpolation points can be deployed inside the field by increasing the number of retarded time data. As a simulation example, two rectangular duct sections having arbitrary temperature distribution are reconstructed by two different data set: measured data only, combination of measured and regenerated data. The result shows a decrease in reconstruction error by 15 % by combining the original and regenerated retarded time data.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.56-66
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• 1999

Thermoacoustic oscillation is a significant problem in cylindrical-type combustors such as common internal combustion engines, industrial furnaces, gas turbine, etc. This kind of low frequency oscillation can lead to serious consequences such as destruction of the combustor and production of strong noise. The accurate numerical simulation of thermoacoustic phenomena is a complex and challenging problem, especially when considering the chemical reaction of mixtures. As with other simulations of aerodynamics and aeroacoustics, the direct computation of thermoacoustic phenomena requires that Navier-Stokes equations be solved using accurate numerical differentiation and time-marching schemes, with non-reflecting boundary conditions. The numerical approach used here aims at qualitative analysis and efficient prediction of those problems, not at the development of an accurate scheme. The numerical prediction developed in this work is shown to be reasonably matched with experimental result.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.48 no.9
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• pp.725-730
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• 2020

This study was conducted to reduce the computation time required for the computational acoustic analysis of the supersonic rocket jet plume. In order to reduce the computation time, computational acoustic analysis was performed assuming that the supersonic jet plume is a two-dimensional axis-symmetric problem. The results of computational acoustic analysis showed similar results to the acoustic load measurement results. Through this study, it was confirmed that the acoustic load prediction of the supersonic rocket jet plume can be predicted using a two-dimensional axis-symmetric computational analysis.

• The Journal of the Acoustical Society of Korea
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• v.18 no.8
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• pp.3-8
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• 1999

Previous variable vocabulary speech recognition systems with context-independent acoustic modeling, could not represent the effect of neighboring phonemes. To solve this problem, we use allophone-based context-dependent acoustic model. This paper describes the method to improve acoustic model of the system effectively. Acoustic model is improved by using allophone clustering technique that uses entropy as a similarity measure and the optimal allophone model is generated by changing the number of allophones. We evaluate performance of the improved system by using Phonetically Optimized Words(POW) DB and PC commands(PC) DB. As a result, the allophone model composed of six hundreds allophones improved the recognition rate by 13% from the original context independent model m POW test DB.

• The Journal of the Acoustical Society of Korea
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• v.13 no.5
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• pp.15-23
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• 1994

We developed an electronic lens for acoustic imaging systems, which is linear array with 31 microphones equally spaced with distance 34mm. Resonant frequency fo receiver circuit coupled to microphone is 20 kHz. We arranged 16 microphones horizontally and 15 microphones vertically, so that the array allows us to obtain a 2 dimensional angle of source, and to track the motion of source in real time. Due to the problem of aliasing in discrete Fourier Transfrom, the maximum observable angle of the lens is limited to 15${\circ}$. We also employed quadrature phase detection scheme to adjust the focus. We have tested the acoustic lens with a personal computer in an anechoic room and obtained the results agreed with the acoustic imaging theory.

• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.499-505
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• 2018

Time delay estimation between two acoustic sensors is widely used in room acoustics and sonar for target position estimation, tracking and synchronization. A cross-correlation based method is representative for the time delay estimation. However, this method does not have enough consideration for the noise added to the receiving acoustic sensors. This paper proposes a new time delay estimation method considering the added noise on the receiver acoustic sensors. From comparing with the existing GCC (Generalized Cross Correlation) method, and adaptive eigen decomposition method, we show that the proposed method outperforms other methods for a colored signal source in the white Gaussian noise condition.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.383-386
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• 2001

최근 선진 자동차 회사에서는 변속기 개발 단계에서 기어 소음 문제를 해결하고 차량의 소음 품질을 향상시키기 위하여 많은 연구가 수행되고 있다. 그러나, 설계 단계에서 기어 소음을 고려한 기어 제원 결정이 불가능하여, 설계 완료 후 개발 단계에서 기어 소음 문제를 해결하는 것이 매우 어려운 일이었고, 설사 가능하여도 시간적, 성능적인 한계가 있었다. 기어 소음은 크게 Source 와 전달경로의 문제로 나누어지고, 본 연구는 주로 전자에 중점을 둘 것이며, 기어 화인 소음의 주 요인으로 알려져 있는 기어의 전달 오차를 최소화하는 것이다. 그 방법으로 기어 설계의 선진 핵심 기술인 기어 해석을 통한 기어 화인 소음을 최소화하는 최적 기어 설계 제원 결정이다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.147-150
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• 2001

본 논문은 비최소 위상을 가지는 시스템에 대한 역변환 문제를 실험적으로 고찰, 연구하였다. 일반적으로 선형적이고 인과적인 시스템의 입$cdot$ 출력관계는 행렬형태로 공식화할 수 있다. 최소위상(minimum phase) 시스템의 시스템행렬은 항상 역행렬이 존재하며 안정적이지만 비최소 위상(non-minimum phase)시스템의 시스템행렬은 근사특이(near-singular)행렬 또는 특이(singular) 행렬이므로 불량조건(ill-conditioning)이 발생하고 역변환이 존재할 수 없다. 비최소 위상 시스템의 역변환 문제는 다른 과정을 포함하지 않고서는 인과적이고 안정적인 역변환 필터를 가질 수 없다. 따라서 역변환 필터의 구현을 위해 SVD(singular value decomposition)를 이용하였다. 비최소 위상 시스템인 경우 시스템행렬은 하나이상의 매우 작은 특이 값을 가지며 이것은 시스템의 위상정보를 가진다. 이 성질을 이용하여 시스템의 근사적인 역변환 필터를 구현하고 비최소 위상을 갖는 외팔보에 대해 실험적으로 검증하였다.