• Journal of Advanced Marine Engineering and Technology
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• 제31권7호
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• pp.863-871
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• 2007

This paper proposes a methodology to estimate the system coefficients for the infinite impulse response(IIR) digital filters using real code GA. In the traditional real coded GA, it adapts the general genetic operations, whereas in this paper the proposed real coded GA applies improved genetic operations in order to search the optimal solution in given problems. Each of unknown IIR digital coefficients collected as forms of a chromosome. Two illustrative examples including the band pass and band stop IIR digital filters are demonstrated to verify the proposed method.

• Structural Engineering and Mechanics
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• 제56권4호
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• pp.535-548
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• 2015

A series of experimental results on thin mild steel plates clamped at the boundary subjected to gas detonation shock loading are presented. Detonation occurred by mixing Acetylene (C2H2)-Oxygen (O2) in various volume ratio and different initial pressure. The applied impulse is varied to give deformation in the range from 6 mm to 35 mm. Analytical modeling using energy method was also performed. Dependent material properties, as well as strain rate sensitivity, are included in the theoretical modeling. Prediction values for midpoint deflections are compared with experimental data. The analytical predictions have good agreement with experimental values. Moreover, it has been shown that the obtained model has much less error compared with those previously proposed in the literature.

• The Journal of the Acoustical Society of Korea
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• 제16권2E호
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• pp.14-20
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• 1997

The future teleconferencing systems will need an appropriate system which controls properly the acoustic echo for the convenient communication. The conventional acoustic echo cancellation algorithms involve large adaptive filters identifying the impulse response of the echo path. The use of adaptive IIR filters appears to be a reasonable way to reduce computational complexity. Effective cancellation of acoustic echo presented in teleconferencing system requires that adaptive filters have a rapid convergence speed. One of the main problems of acoustic echo cancellation techniques is that the convergence properties degrade for an highly correlated signal input such as speech signals. By the way, the introduction of linear prediction filers onto the structure of the acoustic echo cancellation represents one approach to decorrelate the speech signal. And variable step-size LMS algorithm improves the convergence speed through a little increasing of computational complexity. In this paper, we applied these two methods to the acoustic echo canceller(AEC) and showed that these methods have better performances than the conventional AEC.

• ETRI Journal
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• 제25권5호
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• pp.345-355
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• 2003

In this paper, we propose a method for designing a class of M-channel, causal, stable, perfect reconstruction, infinite impulse response (IIR), and parallel uniform discrete Fourier transform (DFT) filter banks. It is based on a previously proposed structure by Martinez et al. [1] for IIR digital filter design for sampling rate reduction. The proposed filter bank has a modular structure and is therefore very well suited for VLSI implementation. Moreover, the current structure is more efficient in terms of computational complexity than the most general IIR DFT filter bank, and this results in a reduced computational complexity by more than 50% in both the critically sampled and oversampled cases. In the polyphase oversampled DFT filter bank case, we get flexible stop-band attenuation, which is also taken care of in the proposed algorithm.

• The Journal of the Acoustical Society of Korea
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• 제25권3E호
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• pp.123-130
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• 2006

The adaptive echo canceller (AEC) has become an important component in speech communication systems, including mobile station. In these applications, the acoustic echo path has a long impulse response. We propose a running-average least mean square (RALMS) algorithm with a detection method for acoustic echo cancellation. Using colored input models, the result clearly shows that the RALMS detection algorithm has a convergence performance superior to the least mean square (LMS) detection algorithm alone. The computational complexity of the new RALMS algorithm is only slightly greater than that of the standard LMS detection algorithm but confers a major improvement in stability.

• 한국음향학회지
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• 제17권2호
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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.

• 한국정밀공학회지
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• 제19권10호
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• pp.130-138
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• 2002

This paper presents the modeling and analysis of dynamic characteristics of hydraulic snubber in electric power plant. The nonlinear state-space model of 14th order to describe the dynamics of the snubber was established by Simulink. The simulation results show that the hydraulic snubber reacts as like the conventional shock absorbers against the high pulse shock load. The snubber also shows the peculiar characteristics to the small step load, which temporarily lock the control valves up, however maintain same steady-state pressures of all internal chambers in the long run. Two case studies for the analysis of the snubber were addressed. Practical pulse testing method was also proposed to identify the frequency response characteristics of the snubber.

• 대한전자공학회논문지
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• 제24권4호
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• pp.719-726
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• 1987

In this paper, an adaptive IIR (infinite impulse response) notch filter of lattice type is constructed and its adaptation algorithm is proposed for the detection and retrieval of a sine wave signal embedded in noise. A modified method which adapts only one coefficient of the filter is also suggested. All these methods adapt the coefficients while keepting the poles of the filter inside the unit circle on z-plane, and thus they satisfy the condition on the stability of the IIR filter after it has converged. To investigate the convergence characteristics of these methods such as convergence speed and output S/N ratio, intensive computer simulation has been performed by varying the frequency of the sine wave and the input S/N ratio. And the results of the simulation have been compared to those of Rao and Kung's which shows relatively fast convergence speed. The methods proposed here, especially the second one. shows faster convergence speed and higher output S/N ratio than the Rao and Kung's.

• 한국생산제조학회지
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• 제24권1호
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• pp.87-93
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• 2015

The dynamic responses of a telescope loaded on an STSAT-3 satellite were analyzed, and environmental tests were conducted to verify the reliability of the design. The space use telescope COMIS (compact imaging spectrometer) is a major payload of the STSAT-3 launched on November 21, 2013. Vibration responses such as the acceleration, displacement, and velocity with respect to random vibration and shock impulse inputs were obtained based on theoretical fundamentals in conjunction with finite element analysis. The main focus of this study was on developing technology for accurate and/or favorable modeling and analysis of the structure and fitting the results to that of experiments. Cutting-edge technology for manipulating the vibrations of a space use telescope is suggested.

• 한국통신학회논문지
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• 제19권1호
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• pp.1-9
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• 1994

In general, when IIR digital filter are designed from analog filters, the bilinera transformation and the impluse invariant tramsformation are commonly used. It is known, however, that high frequency response of digital filters designed by these transformations can not be well approximated to the sampled analog signals. In this paper, the symbol pulse invariant transformation is analyzed theoretically so that the symbol pulse invariant transformation which was originally application to a rectangular pulse is newly applied to double rate pulse signals and generic shape pulse signals. Also, the relation of spectra between a transfer function of digital filter and one of analog filter is considered. Further, we apply to design the digital high pass filters using the symbol pulse invariant transformation method.