• Journal of KSNVE
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• v.10 no.4
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• pp.679-685
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• 2000

A procedure is presented for calculating the magnitude and shape of impact pulses in a vibro-impact system when an arbitrary input force is applied to a point in the system. The procedure utilizes the condition that the displacements of two contacting point in the primary and secondary system are the same during a contacting period. The displacements of those points are calculated numerically through the convolution integral which involve the impulse response functions and applied forces. The validity of the calculation procedure is demonstrated by using it to calculated the impact forces of a simple system where a theoretical solution is known and also of systems for which other researchers have published results. The agreement between the results derived by the numerical method and the theoretical and also some published results is good.

• 제어로봇시스템학회:학술대회논문집
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• 1990.10b
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• pp.1222-1227
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• 1990

The problem of identification of continuous systems is considered when both the discrete input and output measurements are contaminated by white noises. Using a predesigned digital low-pass filter, a discrete-time estimation model is constructed easily without direct approximations of system signal derivatives from sampled data. If the pass-band of the filter is designed so that it includes the main frequencies of both the system input and output signals in some range, the noise effects are sufficiently reduced, accurate estimates can be obtained by least squares(LS) algorithm in the presence of low measurement noises. Two classes of filters(infinite impulse response(IIR) filter and finite impulse response(FIR) filter) are employed. The former requires less computational burden and memory than the latter while the latter is suitable for the bias compensated least squares(BCLS) method, which compensates the bias of the LS estimate by the estimates of the input-output noise variances and thus yields unbiased estimates in the presence of high noises.

• Journal of the Korean Society of Industry Convergence
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• v.6 no.4
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• pp.333-337
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• 2003

This paper proposed new techniques of rotor position detection for 8/6 pole Switched Reluctance Motor(SRM). This technique is very simple and easy to find out rotor position. The main idea uses the impulse responses which have different values between aligned and unaligned rotor position. In order to obtain the informations of the rotor position, the impulse applied to the unenergized phases and their responses are analyzed to control the speed of SRM without shaft sensor. Experimental results verify the feasibility of the proposed method.

• Journal of KSNVE
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• v.10 no.5
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• pp.800-805
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• 2000

To determine the modal matrix and modal frequency of engine mount system, we most solve so-called eigen-value problem. However eigen-value problem of engine mount system with hydraulic mount can not be solved by general eigne-analysis algorithm because the properties of hydraulic mount vary with frequency. so in this paper the method for modal analysis of rigid body motions of an engine supported by hydraulic mount is proposed. Natural frequencies and mode shapes of this nonlinear system are obtained by using complex exponential method and Laplace transformation method. In time domain, impulse response functions are calculated by (two-sided) discrete inverse Fourier Transformation of forced frequency response functions achieved by Laplace transformation of the differential equation of motion. Considering the fact that frequency response functions synthesized by modal parameters form proposed method are in good agreement with original FRFs, it is proved that the proposed method is very efficient and useful for the analysis of eigne-value problem of hydraulic engine mount system.

• ETRI Journal
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• v.46 no.3
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• pp.392-403
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• 2024

Even though generalized frequency division multiplexing is an alternative waveform method expected to replace the orthogonal frequency division multiplexing in the future, its implementation must alleviate channel effects. Least-squares (LS), a low-complexity channel estimation technique, could be improved by using the discrete Fourier transform (DFT) without increasing complexity. Unlike the usage of the LS method, the DFT-based method requires the receiver to know the channel impulse response (CIR) length, which is unknown. This study introduces a simple, yet effective, CIR length estimator by utilizing LS estimation. As the cyclic prefix (CP) length is commonly set to be longer than the CIR length, it is possible to search through the first samples if CP is larger than a threshold set using the remaining samples. An adaptive scale is also designed to lower the error probability of the estimation, and a simple signal-to-interference-noise ratio estimation is also proposed by utilizing a sparse preamble to support the use of the scale. A software simulation is used to show the ability of the proposed system to estimate the CIR length. Due to shorter CIR length of rural area, the performance is slightly poorer compared to urban environment. Nevertheless, satisfactory performance is shown for both environments.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.13-20
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• 2001

Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.

• Journal of IKEEE
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• v.26 no.1
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• pp.36-42
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• 2022

This study presents an analysis of the human body channel as an electric signal path using body impulse response (BIR). The human body communications (HBC) has recently emerged as an effective signal transmission method to create wireless body area networks (WBAN). We provide body channel characteristics based on measured BIR in a proper experimental environment for the HBC using capacitive coupling with a customized channel sounding device, which can be applied as a guideline for the HBC system design. The frequency response of the BIR, extracted by a customized signal processing for the measure signals, shows the channel path loss (CPS) between 0 MHz and 100 MHz with an average CPS of approximately 46.8 dB. In addition, the relative noise power distributions can provide estimations on the signal to noise ratio at the HBC receiver in terms of capacitor and resistor values in the measured frequency band and the frequency band lower than 3 MHz considering the baseband signal detection.

• Journal of Ocean Engineering and Technology
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• v.20 no.5 s.72
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• pp.36-41
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• 2006

The main object of this study is to develop an accurate and convenient method for the response analysis of offshore structures in real sea states. A numerical procedure is described for predicting the motion responses and tension variations of the ISSC TLP in multi-directional irregular waves. The developed numerical approach in the frequency domain is based on acombination of the three dimensional source distribution method, the dynamic response analysis method, and the spectral analysis method. Frequency domain analysis in the multi-directional irregular waves is expanded to a time domain analysis by using a convolution integral after obtaining the impulse response by Fourier transformation. The results of the comparison between responses in the frequency and time domain confirmed the validity of the proposed approach.

• Proceedings of the Computational Structural Engineering Institute Conference
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• 1997.10a
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• pp.42-49
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• 1997

In this paper, the method for soil-structure interaction analyses in the time domain is proposed. The far field soil region which is the outside of the artificial boundary is modeled by using explicit frequency-dependent two dimensional infinite elements which can include multiple wave components propagating into the unbounded medium. Since the dynamic stiffness matrix of the far field soil region using the proposed infinite elements is obtained explicitly in terms of exciting frequencies and constants in the frequency domain, the matrix can be easily transformed into the displacement unit-impulse response matrix, which corresponds to a convolution integral of it in the time domain. To verify the proposed method for soil-structure interaction analyses in the time domain, the displacement responses due to an impulse load on the surface of a soil layer with the rigid bed rock are compared with those obtained by the method in the frequency domain and those by models with extend finite element meshes. Good agreements have been found between them.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.3.2-6
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• 1998

본 논문에서는 실측된 실내 충격 응답을 모델링하여 실내 잔향 필터 설계를 시도하였다. 급강하법(steepest descent method)을 이용하여 측정된 실내 충격 응답을 4개의 콤 필터(comb filter)와 2개의 올패스 필터(allpass filter)로 이루어진 잔향 필터로 모델링하여, 잔향 필터의 계수를 결정하였다.