• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.64-73
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• 1995

We developed a surface acoustic wave (SAW) computer aided design (CAD) for mobile communication using Kaier window function. The systems are composed of modules for designing apodization weighted IDT-uniform IDT, withdrawal weighted IDT-withdrawal weighted IDT, and resonator type. The design of SAW bandpass with center frequencies from 222MHz to 343MHz were simulated by the developed CAD system. Although the method proposed in this paper is formulated primarily for SAW filters, it is equally applicable to finite impulse response (FIR) digital filter design.

• Smart Structures and Systems
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• v.18 no.6
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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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• v.12 no.1
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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.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1996.06a
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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.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.2
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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.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2008.11a
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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.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.25 no.3
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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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.5
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• pp.549-560
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• 1989

In this paper, the method of determining the initial weighting vlaues fuctioning as a filter under the Directional Constrained Minimization of Power(DCMP) algorithm is presented. By analyzing the sideband beamformer with the Finite Impulse Response (FIR) filter concepts, the constraints of any desired directions are obtained and the initial weighing values with fast adaptation time are formulated from those constraints. By applying this proposed initial weighting values to the DCMP and the spatial averaging processor, the interference of a desired direction and the coherent noises are eliminated at the same time. The improvement of this method compared with the existing algorithm is confirmed by computer simulation.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.5 no.1
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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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.8
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• pp.1599-1605
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• 2012

We introduced an approach of modeling of a sound absorbing medium that had different absorbing coefficient according to frequency. To obtain the time domain result of the frequency characteristics of a sound absorbing medium, transmission line matrix modeling was used. To input sound absorbing effect in TLM modeling, we added a FIR filter at a node instead of absorbing component using resistance component. There were simulated the characteristics of time-shift, low pass filter, high pass filter using the FIR filter with 7-tap coefficients, then compared with theoretical results. From various simulation results, we could find that added FIR filter coefficient in TLM modeling was an useful way to model a sound absorbing medium.