• The Journal of the Acoustical Society of Korea
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• v.17 no.2
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• pp.25-31
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• 1998

본 연구에서는 자동차 유리판에 초음파 대역 진동을 효율적으로 발생시킬 수 있는 압전 액츄에이터를 개발하고자 하였다. 자동차 유리판을 단순지지 평판으로 가정하였으며, 액츄에이터로는 PZT 진동자를 사용하였다. 유한 요소법(Finite Element Method)을 통하여 유리판의 최적 공진 주파수와 유리판에 부착할 압전 진동자의 구조 조건인 최적 위치와 개 수를 결정하였다. 설계 결과에 따라 자동차 유리판에 압전 진동자를 부착하여 임펄스 응답 해석(impulse response analysis)과 임피던스 해석(impedance analysis)을 수행하여 구동 주 파수를 확인하였다. 그 결과, 이 주파수로 가진하여 유리판 전면에 걸쳐 고른 초음파 진동이 분포함을 확인하였으며, 나아가 전기적 임피던스 매칭(impedance matching)을 통해 더 큰 효율의 진동을 발생시킬 수 있었다.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.1 s.106
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• pp.34-39
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• 2006

For the evaluation of the normalized impact sound pressure level, the reverberation time of the receiving room should be measured. This paper deals with the effect of the time constant of FFT analyzer and the measuring points on reverberation time. It is found that the time constant should be in the range between 10 ms and 35 ms. While the effect of measuring points on the reverberation time is significant when the bandwidth is narrow it is negligible in the evaluation of the normalized impact sound pressure level.

• Journal of the Korean Institute of Telematics and Electronics T
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• v.35T no.1
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• pp.33-39
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• 1998

This paper discusses approximation modelling of discrete-time linear time-varying system(LTVS). The wavelet transform is considered as a tool for representing and approximating a LTVS. The joint time-frequency properties of wave analysis are appropriate for describing the LTVS. Simulation results is included to illustrate the potential application of the technique.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.265-268
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• 2000

트랜스듀서의 삽입손실과 임펄스 응답특성 등을 정확히 해석하기 위해서는 대상 주파수 대역에 대한 특성을 알아야한다. 방사임피던스는 주파수의 함수이기 때문에 이러한 특성을 결정하는 중요한 요소이다. 따라서 본 연구에서는 정방형 방사면을 가지는 트랜스듀서 이용하여, 무한배플에서의 방사임피던스를 고려한 삽입손실을 이론적으로 계산하고 무한배플로 근사할 수 있는 모델을 제작하고, 이를 실험으로 확인한 결과 좋은 일치를 보이고 있어 본 연구에서 제안한 해석법의 타당성을 확인하였다.

• Proceedings of the KIEE Conference
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• 2007.07a
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• pp.1307-1308
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• 2007

본 연구에서는 비선형 저항 특성을 지진 산화 몰리브덴 (MoO3) 소자를 이용한 새로운 피뢰 소자 기술을 제안한다. MoO3 소자는 가열법과 RF 스퍼터링법에 의해 각각 제작하였으며, 이들을 서로 겹쳐서 제작된 양면박막의 갭형 소자에 대해, 절연내력, 응답특성 등의 전기적 특성과 방전 후의 표면 특성 변화 등에 대하여, 실험적으로 고찰하였다. 그 결과, MoO3 소자는 총 10회로 연속적으로 인가한 임펄스 전압실험에서 양호한 동작을 보였으며, 2회 인가 시, 저항 값이 다소 감소하기는 하였으나, 약 $800k{\Omega}$의 저항 값을 일정하게 유지하여 피뢰 소자로서의 성능을 계속 유지할 수 있음을 확인할 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.208-212
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• 2009

Ground wave is an acoustic wave propagating at a sediment sound speed in the case that sediment sound speed is constant with depth, which is explained by modal dispersion effects. In this paper, the ground wave in time domain is simulated using the ray-based approach, which is possible because the modal dispersion can be explained by the guiding of energy caused by reflection and refraction in the waveguide geometry. For a Pekeris waveguide, the ground wave can be interpreted as a sequence of head waves, called a head wave sequence [Choi and Dahl, J. Acoust. Soc. Am. 119, 3660-3668 (2006)]. The ground wave is simulated by convolution of the source signal with a channel impulse response of the head wave sequence, which is compared with simulated signals obtained via a Fourier synthesis of a complex parabolic equation (PE) field.

• The Journal of the Acoustical Society of Korea
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• v.9 no.2
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• pp.42-52
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• 1990

A method of measuring impulse response of acoustic system, two kinds of cross-correlation methods (the direct correlation method and the M-sequence modulation correlation method) were applied. According to the direct correlation method, by using stationary random noise source and by calculating the cross-correlation function between the sourece and the output signal, equivalent impulse response can be obtained not being influenced by the back ground noises. By applying this method, the measurement of echo-time patterns in rooms and oblique incident sound absorption characteristics of sound absorbing materials was carried out. In the case of the M-sequence modulation correlation method which was contrived by Aoshima, an intermittent random nosie modulated by M-sequence signal is used as the source signal, and the cross-correlation function between the M-sequence signal and the squared output signal is calculated. According to this method, equivalent energy impulse response (squared impulse response) of te propagation system can be obtained without being influenced by the back ground noises and the air fluctuation caused by wind. As the applcaition of this measuring method to the architectural acoustics, the meaurements of echo-time patterns, reverberation decays and sound pressure lev디 distributions in rooms and sound insulation efficiencies in buildings were carried out. From these experimental studies, it has been found that this M-sequence modulation correlation method is markedly useful especially for the field masurement of sound insulation under high back ground noise condition.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.38 no.2
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• pp.188-194
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• 2010

In this paper, the typical Synthetic Aperture Radar (SAR) image quality parameters and analysis method are defined, and the SAR image analysis tool is presented for SAR image evaluation. The structure of the developed SAR image analysis tool consists of four key modules; point target analysis (PTA) module, distributed target analysis (DTA) module, ambiguity analysis (AMA) module, and NESZ analysis (NESZA) module. The developed tool is able to extract the various SAR system parameters from standard SAR product format files. Based on these extracted system parameters, typical SAR image quality parameters are derived from SAR image data.

• The Journal of the Acoustical Society of Korea
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• v.23 no.1
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• pp.61-74
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• 2004

In this paper, a contort- and speaker-dependent cepstrum extraction method and a channel normalization method for minimizing the loss of speaker characteristics in the cepstrum were proposed for a robust speaker recognition system over the channel. The proposed extraction method creates a cepstrum based on the pitch synchronous analysis using the inherent pitch of the speaker. Therefore, the cepstrum called the 〃pitch synchronous cepstrum〃 (PSC) represents the impulse response of the vocal tract more accurately in voiced speech. And the PSC can compensate for channel distortion because the pitch is more robust in a channel environment than the spectrum of speech. And the proposed channel normalization method, the 〃formant-broadened pitch synchronous CMS〃 (FBPSCMS), applies the Formant-Broadened CMS to the PSC and improves the accuracy of the intraframe processing. We compared the text-independent closed-set speaker identification on 56 females and 112 males using TIMIT and NTIMIT database, respectively. The results show that pitch synchronous km improves the error reduction rate by up to 7.7% in comparison with conventional short-time cepstrum and the error rates of the FBPSCMS are more stable and lower than those of pole-filtered CMS.

• The Journal of the Acoustical Society of Korea
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• v.36 no.4
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• pp.285-291
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• 2017

Many methods have been studied to estimate the time delay between incoming signals to two receivers. In the case of the method based on the channel estimation technique, the relative delay between the input signals of the two receivers is estimated as an impulse response of the channel between the two signals. In this case, the characteristic of the channel has sparsity. Most of the existing methods do not take advantage of the channel sparseness. In this paper, we propose a time delay estimation method using BPD (Basis Pursuit Denoising) optimization technique, which is one of the sparse signal optimization methods, in order to utilize the channel sparseness. Compared with the existing GCC (Generalized Cross Correlation) method, adaptive eigen decomposition method and RZA-LMS (Reweighted Zero-Attracting Least Mean Square), the proposed method shows that it can mitigate the threshold phenomenon even under a white Gaussian source, a colored signal source and oceanic mammal sound source.