• Journal of the Computational Structural Engineering Institute of Korea
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• v.14 no.2
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• pp.173-179
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• 2001

성능지수는 제어 대상이 되는 구조물의 응답과 제어기의 성능에 관한 가중함수로 구성되어 있다. 따라서 가중함수의 설계에 따라 성능지수가 변화되며 제어 효율이 달라진다. 본 논문에서는 최적 능동제어 알고리듬의 일종인 시간 영역에서의 성능지수를 고려한 LQR기법과 LQG기법 및 주파수 영역에서의 성능지수를 고려한 H₂기법에 대하여 동일한 가중함수를 적용하여 제어 성능인 제어율과 제어력을 비교하는데 목적을 두고 있다. 그러나, LQG기법은 모든 상태 변수를 알아야 하는 LQR기법의 한계를 극복할 수 있으며 LQR기법과 동일 수준의 제어율과 제어력을 보이고 있고 출력 제어라는 장점을 고려하면 현실적인 기법이라고 말할 수 있다. 마지막으로 구조물 응답과 제어기의 주파수 특성을 고려하여 주파수 필터의 가중함수를 설계하는 H₂기법을 분석하였다. H₂기법은 제어력을 저주파수 영역에 집중시킬 수 있기 때문에 구조물 응답을 효과적으로 제어할 수 있는 방법으로 분석되었다.

• Proceedings of the KIEE Conference
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• 2001.07d
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• pp.2274-2276
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• 2001

본 논문에서는 미세 구조물 가공 기술(MEMS)로 제작된 실리콘 자이로스코프를 위한 폐루프 제어기를 주파수 가중 $H_{\infty}$ 제어 기법과 주파수 가중 모델 축소 기법을 이용하여 설계하였다. 실리콘 자이로스코프는 수 kHz 대의 공진 특성을 이용하여 각속도를 측정하는 센서로서, 공진 주파수 영역의 특성이 매우 중요하므로 주파수 영역을 고려하고 공정 오차를 감안한 주파수 가중 $H_{\infty}$ 제어기가 필요하다. 본 논문에서는 고차 강인 제어기의 회로적 구현과 ASIC화가 가능하도록 하기 위해, 주파수 가중 모델 축소 기법을 이용하여 공진 주파수 영역에서 성능을 유지하는 저차 제어기를 설계하였으며, 시뮬레이션을 통해 기존의 제어기 및 고차 제어기와 성능을 비교하였다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.6
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• pp.124-131
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• 2003

In this paper, we propose a Progressive scalable video coding algorithm using frequency weighting in the DCT domain. Since the human visual system (HVS) can be modeled as a nonlinear point transformation, called the modulation transfer function (MTF), we tan use the frequency weighting matrix to enhance the video image quality. We change this frequency weighting matrix into the frequency shift matrix to apply to the bit-plane coding method for the fine granular scalable (FGS) video coding We also define a new error metric JNDE (just noticeable difference) to measure the perceptual image quality in terms of human vision.

• Geophysics and Geophysical Exploration
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• v.5 no.3
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• pp.169-177
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• 2002

We develop the weighted-averaging finite-element method which uses four kinds of element sets. By constructing global stiffness and mass matrices for four kinds of element sets and then averaging them with weighting coefficients, we obtain a new global stiffness and mass matrix. With the optimal weighting coefficients minimizing grid dispersion and grid anisotropy, we can reduce the number of grid points required per wavelength to 4 for a $1\%$ upper limit of error. We confirm the accuracy of our weighted-averaging finite-element method through accuracy analyses for a homogeneous and a horizontal-layer model. By synthetic data example, we reconfirm that our method is more efficient for simulating a geological model than previous finite-element methods.

• Proceedings of the Acoustical Society of Korea Conference
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• 1996.06a
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• pp.29-32
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• 1996

본 논문에서는 시변 주파수를 추정하기 위한 방법으로 기존의 시간 가중 칼만 추정기법에 변형된 Huber함수를 적용하여 충격성 잡음환경 하에서도 강인한 칼만추정기법을 제안하였다. 기존의 시간 가중 칼만 추정기법은 오차가 정규분포를 가진다고 가정된 상태에서는 적합한 파라메타 추정을 할 수 있지만 충격성 잡음이 존재하는 경우에는 수렴속도나 시변적응능력에서의 성능저하가 나타난다. 제안된 알고리듬은 영향함수 측면에서 충격성 잡음에 의해 생기는 오차의 크기를 제한함으로써 기포나 인위적인 충격성 잡음환경 하에서도 시변 주파수 추정을 할 수 있으며 알고리듬의 타당성은 모의실험을 통해 보였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2C
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• pp.213-221
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• 2004

A new method for design of two-channel finite-impulse response(FIR) quadrature mirror-image filter(QMF) banks with low reconstruction delay using weighting function is proposed. The weighting function used in this paper is calculated from the previous updated filter coefficients vector which is adjusted from iteration to iteration in the design of QMF banks. In this paper, passband and stopband edge frequency are used in design of QMF banks with low delay characteristic in time domain instead of specific frequency interval where the artifacts occur in conventional design method. The investigation of specific frequency interval where artifacts occur can not be required by using passband and stopband edge frequency. Some comparisons of performance are made with other existing design method to demonstrate the proposed method for QMF bank design. and it was observed that the proposed method using the weighted function and passband and stopband edge frequency improves the peak reconstruction error by 0.001 [dB], the peak-to-peak passband ripple by 0.003[dB], SNR with a white noise by 7[dB] and SNR with a step input by 32[dB], but with a reduction of the computational efficiency because of updating the weighting function over the conventional method in Ref [11].

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.62-68
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• 2002

It is well known that the MDVR beamforming outperforms the conventional delay-sum beamformer in the sense of noise rejection and bearing resolution. However, the MDVR method requires long observation time to achieve high frequency resolution. The STMV method uses the steered covariance matrix of sensor data, so it has an ability to form an adaptive weight vector from a single time-series snapshot. But it uses the same weight vector across all frequencies. In this paper, we propose an SSMV method. The basic idea of the SSMV method is to decompose a full frequency band into several subbands to acquire a weight vector for each subband, individually. Also the wrap may be divided into several subarrays in order to reduce a computational load and the bandwidth of each subband. Simulations using real sea trial data show that the proposed SSMV method has good performance with short observation time.

• The Journal of the Acoustical Society of Korea
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• v.15 no.5
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• pp.61-64
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• 1996

To improve the performance of a recognition system, namely the recognition rate, we propose a hew implementation of perceptual distance using LPC cepstrum(perceptual cepstrum, PLC). The PLC is caculated by convolution of a usual LPC cepstrum and a perceptual lifter(PL). To caculate PL, we define a new weighting function in the linear frequency domain considering the frequency scale(Bark-scale) characteristics. The PL is the inverse Fourier transform of the exponents of the weighting function. We verified our method through the speech recognition experiments. The performance of PLC was compared with that of the rasied sine liftering method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.10
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• pp.1867-1872
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• 2016

In the area of digital signal processing, it is necessary to improve the quality of the speech signal after removing the background noise which exists in a various real environments. The important thing to consider when removing the background noise acoustically is that to solve the problem, depending on the information of the human auditory mechanism is mainly the amplitude spectrum of the speech signal. This paper introduces the characteristics of a frequency-weighted spectrum shaping filter for the extraction of the amplitude spectrum of the speech signal with the primary purpose. Therefore, this paper proposes an algorithm using the methods of a Wiener filter and the frequency-weighted spectrum shaping filter according to the acoustic model, after extracted the amplitude spectral information in the noisy speech signal. The spectral distortion (SD) output of the proposed algorithm is experimentally improved more than 5.28 dB compared to a conventional method.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.37 no.2
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• pp.8-15
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• 2000

This study presents an H$_{\infty}$ optimal controller synthesis by the frequency domain analysis of weighting function. The main purpose of our study is to visualize weighting function domains and open loop transfer function domains that satisfy robustness and performance. Also we aim to simplify the iterative algorithm for H$\infty$ optimal controller synthesis. We report that the designed regulator by the proposed method in this paper satisfies the desired specifications and performance in spite of the plant uncertainty variation at any operating point.