• 한국음향학회지
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• 제14권4호
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• pp.64-73
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• 1995

본 연구에서는 카이저 창함수를 이용하여 이동통신용 표면탄성파 필터를 설계하기 위한 CAD를 제작하였다. 제작된 CAD는 아포다이제이션 형태로서 하중함수가 주어진 빗살변환기와 정규형형태의 빗살변환기, 위드드롤로 형태로서 하중함수가 주어진 빗살 변환기 한쌍으로된 변환기와 아포다이제이션 형태로서 하중함수가 주어진 빗살변화기와 정규형 변환기, 반사기를 설치한 공진기 형태의 필터로서 구성되어 있다. 중심 주파수가 222MHz에서 343MHz 영역내의 표면탄성파 필터를 제작된 CAD로 모의 실험 하였다. 본 연구에서 제작된 CAD는 표면탄성파 필터로써 응용될 뿐만 아니라 디지틀 FIR 필터에도 응용될 수 있다.

• Smart Structures and Systems
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• 제18권6호
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• pp.1125-1143
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• 2016

Forced vibration analysis of a simple supported viscoelastic nanobeam is studied based on modified couple stress theory (MCST). The nanobeam is excited by a transverse triangular force impulse modulated by a harmonic motion. The elastic medium is considered as Winkler-Pasternak elastic foundation.The damping effect is considered by using the Kelvin-Voigt viscoelastic model. The inclusion of an additional material parameter enables the new beam model to capture the size effect. The new non-classical beam model reduces to the classical beam model when the length scale parameter is set to zero. The considered problem is investigated within the Timoshenko beam theory by using finite element method. The effects of the transverse shear deformation and rotary inertia are included according to the Timoshenko beam theory. The obtained system of differential equations is reduced to a linear algebraic equation system and solved in the time domain by using Newmark average acceleration method. Numerical results are presented to investigate the influences the material length scale parameter, the parameter of the elastic medium and aspect ratio on the dynamic response of the nanobeam. Also, the difference between the classical beam theory (CBT) and modified couple stress theory is investigated for forced vibration responses of nanobeams.

• Journal of Electrical Engineering and Technology
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• 제12권1호
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• pp.378-384
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• 2017

In this paper, an efficient short-time Fourier analysis method for the quasi-periodic signals is proposed via an optimal fixed-lag finite impulse response (FIR) smoother approach using a receding horizon scheme. In order to deal with time-varying Fourier coefficients (FCs) of quasi-periodic signals, a state space model including FCs as state variables is augmented with the variants of FCs. Through an optimal fixed-lag FIR smoother, FCs and their increments are estimated simultaneously and combined to produce final estimates. A lag size of the optimal fixed-lag FIR smoother is chosen to minimize the estimation error. Since the proposed estimation scheme carries out the correction process with the estimated variants of FCs, it is highly probable that the smaller estimation error is achieved compared with existing approaches not making use of such a process. It is shown through numerical simulation that the proposed scheme has better tracking ability for estimating time-varying FCs compared with existing ones.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 1996년도 학술대회
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• pp.223-227
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• 1996

In this paper we propose an adaptive fast image restoration filter, which is suitable for reducing the blocking artifacts in the compressed image in real-time. The proposed restoration filter is based on the observation that quantization operation in a series of coding process is a nonlinear and many-to-one mapping operator. And then we propose an approximated version of constrained optimization technique as a restoration process for removing the nonlinear and space varying degradation operator. We also propose a novel block classification method for adaptively choosing the direction of a highpass filter, which serves as a constraint in the optimization process. The proposed classification method adopts the bias-corrected maximized likelihood, which is used to determine the number of regions in the image for the unsupervised segmentation. The proposed restoration filter can be realized either in the discrete Fourier transform domain or in the spatial domain in the form of a truncated finite impulse response (FIR) filter structure for real-time processing. In order to demonstrate the validity of the proposed restoration filter experimental results will be shown.

• 전자공학회논문지S
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• 제34S권2호
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• pp.79-93
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• 1997

This paper presents a linear shift-invariant system euqivalent to morphological dilation for a memoryless uniform source in the sense of the power spectral density function, and comares it with dialtion. This equivalent LSI system is found through spectral decomposition and, for dilation and with windwo size L, it is shown to be a finite impulse response filter composed of L-1 delays, L multipliers and three adders. Th ecoefficients of the equivalent systems are tabulated. The comparisons of dilation and its equivalent LSI system show that probability density functions of the output sequences of the two systems are quite different. In particular, the probability density functon from dilation of an independent and identically distributed uniform source over the unit interval (0, 1) shows heavy probability in around 1, while that from the equivalent LSI system shows probability concentration around themean vlaue and symmetricity about it. This difference is due to the fact that dilation is a non-linear process while the equivalent system is linear and shift-ivariant. In the case that dikation is fabored over LSI filters in subjective perforance tests, one of the factors can be traced to this difference in the probability distribution.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2008년도 추계학술대회논문집
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• pp.497-502
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• 2008

In this paper, we studied on an attenuation effect of automobile exhaust noise according to the direction of secondary sound source in duct ANC system. Automobile exhaust noise was recorded at 800rpm. 3500rpm and 5000rpm of a diesel engine. Directions of loudspeaker(second sound source) can be exchanged to $30^{\circ}$, $90^{\circ}$ and $150^{\circ}$ against the primary noise flow by acrylic ducts to be made for experimentation. DSP board with TMS320C6416 chip of Texas Instrument Co used to control adaptive ANC system. This ANC system is based on the single-channel FxLMS algorithm. In experiment result, when the loud speaker direction was $150^{\circ}$, the attenuation effect showed largely. In case of $90^{\circ}$ duct, the noise was a little increased. In case of $30^{\circ}$ duct, the noise was a little increased or decreased according to the frequency range and the sound pressure(dB) of exhaust noise to comply with engine rpm.

• 한국소음진동공학회논문집
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• 제25권3호
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• pp.191-198
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• 2015

Noise propagated through the ventilation openings of enclosures is actively controlled using an FIR filter. The enclosures considered in this paper are used for isolating noise due to machinery. This method is of limited use because of the ventilation openings through which most of noise is propagated. Feedforward control strategy is incorporated to minimize the acoustic power propagated through the openings. For the real-time implementation, although it is numerical study, the controller is implemented using an FIR filter. The acoustic transfer functions of the pressure on the openings of the enclosure to the primary source and to the secondary source are numerically calculated using the boundary element method. The performance analysis of the active control is conducted with the time-domain simulation using Matlab Simulink.

• 한국통신학회논문지
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• 제14권5호
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• pp.549-560
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• 1989

본 논문은 광대역 신호처리 DCMP 알고리즘에서 필터로서 동작하는 초기가중치의 설정 방법을 제안하였고 컴퓨터 시뮬레이션을 통해 최적의 Tap 수를 결정하는 방법을 기술하였다. 광대역 비임 형성기를 FiR 필터 개념으로 분석하여 원하는 신호의 입사 방향에 대한 제한 조건을 얻었고 이 제한 조건으로 부터 적응 속도가 빠른 초기 가중치를 설정하였다. 본 논문에서 제안된 초기 가중치를 기존의 DCMP프로세서나 공간 영역 평균법을 수행하는 프로세서에 적용할 때 원하는 신호의 입사 방향과 일치하는 잡음 그리고 Cohenrent Noise를 반복 적용하는 과정에서 동시에 제거할 수 있음을 컴퓨터 시뮬레이션으로 확인 하였다.

• International Journal of Fuzzy Logic and Intelligent Systems
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• 제5권1호
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• pp.13-20
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• 2005

This paper addresses a new blind channel equalization method using fourth-order cumulants of channel inputs and a three-layer neural network equalizer. The proposed algorithm is robust with respect to the existence of heavy Gaussian noise in a channel and does not require the minimum-phase characteristic of the channel. The transmitted signals at the receiver are over-sampled to ensure the channel described by a full-column rank matrix. It changes a single-input/single-output (SISO) finite-impulse response (FIR) channel to a single-input/multi-output (SIMO) channel. Based on the properties of the fourth-order cumulants of the over-sampled channel inputs, the iterative algorithm is derived to estimate the deconvolution matrix which makes the overall transfer matrix transparent, i.e., it can be reduced to the identity matrix by simple recordering and scaling. By using this estimated deconvolution matrix, which is the inverse of the over-sampled unknown channel, a three-layer neural network equalizer is implemented at the receiver. In simulation studies, the stochastic version of the proposed algorithm is tested with three-ray multi-path channels for on-line operation, and its performance is compared with a method based on conventional second-order statistics. Relatively good results, withe fast convergence speed, are achieved, even when the transmitted symbols are significantly corrupted with Gaussian noise.

• 한국정보통신학회논문지
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• 제16권8호
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• pp.1599-1605
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• 2012

본 연구에서는 해저의 반사면이나 흡음재와 같이 주파수에 따라 다른 특성을 가지는 매체를 시간영역에서 수치적으로 모델링하고자 시간 해석 기법의 하나인 전달선로행렬법을 이용하였다. 기존의 흡음재의 해석에 응용되던 감쇠를 이용하는 방법의 대안으로 유한임펄스응답 필터를 전달선로행렬법에 도입하였다. 7개의 탭을 가진 FIR 필터로 구현하였으며, 시간 지연, 저역통과필터, 고역통과필터의 시뮬레이션 결과를 이론치와 비교하였다. 여러 시뮬레이션 결과를 통해 흡음재의 주파수에 따른 흡음 특성을 유한임펄스응답 필터를 고려한 하나의 요소만으로 모델링하는 것이 가능하다는 것을 확인할 수 있었다.