• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.46 no.12
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• pp.975-985
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• 2018

The drones have been developed for military purposes and are now used in many fields such as logistics, communications, agriculture, disaster, defense and media. As the range of use of drones increases, cases of abuse of drones are increasing. It is necessary to develop anti-drone technology to detect the position of unwanted drones using the physical phenomena that occur when the drones fly. In this paper, we estimate the DOA(direction of arrival) of the drone by using the acoustic signal generated when the drone is flying. In addition, the dynamics model of the drones was applied to the unscented kalman filter to improve the microphone array detection performance and reduce the error of the position estimation. Through simulation, the drone detection performance was predicted and verified through experiments.

• The Journal of the Korea institute of electronic communication sciences
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• v.14 no.6
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• pp.1249-1256
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• 2019

In this paper, we propose a new algorithm to remove acoustic feedback and noise in hearing aids. Instead of using the conventional FIR structure, this algorithm is a deep learning algorithm using neural network adaptive prediction filter to improve the feedback and noise reduction performance. The feedback canceller first removes the feedback signal from the microphone signal and then removes the noise using the Wiener filter technique. Noise elimination is to estimate the speech from the speech signal containing noise using the linear prediction model according to the periodicity of the speech signal. In order to ensure stable convergence of two adaptive systems in a loop, coefficient updates of the feedback canceller and noise canceller are separated and converged using the residual error signal generated after the cancellation. In order to verify the performance of the feedback and noise canceller proposed in this study, a simulation program was written and simulated. Experimental results show that the proposed deep learning algorithm improves the signal to feedback ratio(: SFR) of about 10 dB in the feedback canceller and the signal to noise ratio enhancement(: SNRE) of about 3 dB in the noise canceller than the conventional FIR structure.

• Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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• 2000.05a
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• pp.5-12
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• 2000

본 연구에서, Daubechies'Mother Wavelet를 이용한 이산 웨이블렛 변환(Discrete Wavelet Transform)에 기초한 새롭고 효과적인 특정추출방법을 제안하였다. 특정추출을 이용하여 응용방향을 설명하고 또는 통계적 파라메터의 평가를 행하였다. 본 연구에서는 다음과 같은 몇 가지 사실을 알 수 있었다. 1. 시스템에서 발생된 (인가전압이 0[V]) 노이즈라 볼 수 가있는 렌덤노이즈(Random Noise)를 디지털필터인 FIR(Finite Impulse Response)필터를 통하여 상당한 노이즈를 억제할 수가 있었다. 2. 이산 웨이블렛 변환 시 레벨 1~4까지 변환한 결과 최적의 변환상태 Level-3을 기준으로 하였다. 3. 특정추출 파라메터는 음향방출신호의 최대값, 평균값, 분산, 왜도, 첨쇄도를 특정추출파라메터로 이용하였다. 4. 특정추출 결과를 이용하여 전체 열화시간 중 대표적 음향방출신호 중 초기열화신호, 중기열화신호, 말기열화신호를 얻을 수 있었다. 이런 특정추출을 통하여 변압기열화상태를 진단할 수 있는 가능성을 확인 할 수가 있었다.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.39 no.4
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• pp.41-54
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• 2002

In this study, we have analyzed the effects of the membrane layer and the bragg reflector layers on the resonance characteristics through comparing the characteristics of the membrane type FBAR (Film Bulk Acoustic Wave Resonator) and the one type bragg reflector layers with those of the ideal FBAR with top and bottom electrode contacting air by using equivalent circuit technique. It is assumed that ZnO is used for piezoelectric film, $SiO_2$ are used for membrane layer and low acoustic impedance layer, W are used for the high acoustic reflector layer and Al is used for the electrode. Each layer is considered to have a acoustic propagation loss. ABCD parameters are picked out and input impedance is calculated by converting 1-port equivalent circuit to simplified equivalent circuit that ABCD parameters are picked out possible. From the variation of resonance frequency due to the change of thickness of electrode layers, reflector layers and membrane layer it is confirmed that membrane layer and the reflector layer just under the electrode have the greatest effect on the variation of resonance frequency. From the variation of resonance properties, K and electrical Q with the number of layers, K is not much affected by the number of layers but electrical Q increases with the number of layers when the number of layers is less than seven. The electrical Q is saturated when the number of layers is large than six. The electrical Q is dependent of mechanical Q of reflector layers and membrane layer. Both ladder filter and SCF (Stacked Crystal Filters) show higher insertion loss and out-of-band rejection with the increase of the number of resonators. The insertion loss decreases with the increase of the number of reflector layers but the bandwidth is not much affected by the number of reflector layers. Ladder Filter and SCF with membrane layer show the spurious response due to spurious resonance properties. Ladder filter shows better skirt-selectivity characteristics in bandwidth but SCF shows better characteristics in insertion loss.

• The Journal of the Acoustical Society of Korea
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• v.28 no.8
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• pp.822-831
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• 2009

The porcelain insulators are important devices, that are used to isolate electrically and hold mechanically in the high-voltage power transmission systems. The faults of the insulators induce very serious problems to the power transmission line. In this paper, we introduce techniques for fault detections of insulators by acoustic radiation noises from them. We measured radiation noises from normal state insulators and fault state insulators. The used insulators were two different type porcelain insulators, a cut out switch, two different type line posters, and a lightning arrester. Each results was compared each other in time domain, frequency domain and filter banks' outputs. We found the possibility of detection of insulators' faults and also suggested techniques for fault detections.

• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.35-43
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• 1999

In this paper, a new robust adaptive algorithm and a post-processing method are proposed to improve the performance of AEC without computational burden. Its step-size is normalized by the sum of the powers of the reference input signal and the desired signal. When the near-end speaker's speech and noise are applied into the microphone, the step-size becomes small and the misalignment of coefficients are reduced. To reduce the residual echoes, a new post-processing method, which is co-operated with the proposed noise-robust adaptive algorithm, is proposed in this paper. The method is based on the correlation of the desired signal and the estimation error signal. The residual echoes are attenuated as proportional to the correlation normalized with the power of desired signals. The normalized correlation plays a role as Wiener filter for residual echoes. In the double-talk situation, the estimation error signals, that are residual echoes, dominantly include the near-end speaker's speech and the normalized correlation closes to 1. Therefore, the near-end speaker's speech can be transmitted without being attenuated. When the desired signals consists of only the acoustic echoes, the residual echoes are mostly attenuated and canceled by the proposed post-processor. The computation of AEC using the proposed post-processor is comparable to NLMS algorithm.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.24 no.7
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• pp.525-536
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• 2014

In this paper, GUI program for microspeaker system simulation program was developed and verified through closed box, vent box and 6th order bandpass enclosure system. By using the pseudo loudspeaker model concept, TS parameters and rear volume of microspeaker were identified. Their suitabilities were proved by comparing test results with simulations of electrical impedance and sound pressure response curves for the three box types; closed box, vent box and 6th order bandpass box. Also, MSSP was found to be effective regardless of the microspeaker's shape, either circular or rectangular shape. MSSP can be used for the microspeaker system simulation, and can give a general prediction of such as; sound pressure level curve, electrical impedance, diaphragm velocity and displacement curve according to multiple design parameters; diaphragm mass, compliance, force factor, front and rear volume, front and rear port's diameter and length.

• Journal of Korea Multimedia Society
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• v.4 no.3
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• pp.230-237
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• 2001

Adaptive filters are widely used for acoustic echo canceler, adaptive equalizer and adaptive noise canceler. Coefficients of adaptive filters are updated by NLMS algorithm. However, Coefficients are misaligned by ambient noises when they are adapted by NLMS algorithm. In this Paper, a method determined the adaptation constant by low-pass filtered instantaneous gradient vector of LMS algorithm using orthognality principles of optimal filter is proposed. At initial states, instantaneous gradient vector, that is the cross-correlation of input signals and estimation error signals, has large value because input signals are remained in estimation error signals. When an adaptive filter is conversed, the cross-correlation will be close to zero. It isn's affected by ambient noises because ambient noises are uncorrelated with input signals. Determining adaptation constant with the cross-correlation, adaptive filters can be robust to ambient noises and the convergence rate doesn't slower As results of computer simulations, it is shown that the performance of proposed algorithm is betted than that of conventional algorithms.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.11
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• pp.2090-2094
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• 2007

Film bulk acoustic wave resonator(FBAR) technology has attracted a great attention as a promising technology to fabricate the next-generation RF filters mainly because the FBAR technology can be integrated with current Si processing. The RF filters are basically composed of several FBAR devices connected in parallel and in series, and their characteristics depend highly on the FBAR device characteristics. Thus, it is important to design high quality FBAR devices by device or process optimization. This kind of effort may enhance the FBAR device characteristics, eventually leading to FBAR filters of high performance. In this paper, we describe the methods to more effectively improve the resonance characteristics of the FBAR devices.

• Journal of Advanced Marine Engineering and Technology
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• v.36 no.2
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• pp.258-266
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• 2012

Recently, Acoustic Emission (AE) technique is widely applied to develop the early fault detection system, and the problem about a signal processing method for AE signal is mainly focused on. In the signal processing method, envelope analysis is a useful method to evaluate the rolling element bearing problems and Wavelet transform is a powerful method to detect faults occurred on gearboxes. However, exact method for AE signal is not developed yet. Therefore, in this paper, two methods, which is Hilbert transforms (HT) and Hilbert-Huang transforms (HHT), will be compared for development a signal processing method for early fault detection system by using AE. AE signals were measured through a fatigue test. HHT has better advantages than HT because HHT can show the time-frequency domain result. But, HHT needs long time to process a signal, which has a lot of data, and has a disadvantage in de-noising filter.