• The Transactions of The Korean Institute of Electrical Engineers
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• v.56 no.2
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• pp.424-427
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• 2007

A fully differential transconductor using the series composite transistor is proposed. Simulation results show that THD is less than 1.2% for the differential input signal of up to $1.5V_{p-p}$ when the input signal frequency is 10MHz. i he proposed transconductor is used to design a third-order elliptic Gm-C lowpass filter with 138kHz cutoff frequency for ADSL Tx filter. The design procedure is based on signal flow graph(SFG) of a doubly-terminated LC ladder filter by means of fully differential transconductors and capacitors. The filter is fabricated and measured with a $0.35{\mu}m$ CMOS process.

• The Journal of the Acoustical Society of Korea
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• v.28 no.1E
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• pp.29-34
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• 2009

MPEG4 High Efficiency Advanced Audio Coding (HE-AAC) and MPEG Surround are one of the most efficient combinations for low bit rate multi-channel audio coding. Based on the fact that these two codecs have identical quadrature mirror filter (QMF) analysis and synthesis structures, we propose a dual-domain connection scheme for the codecs. Specifically two time-domain connection methods are analyzed and compared to the QMF subband-domain connection method. Experimental results show that both the time-domain connection methods cause no subjective sound quality degradation compared to the QMF subband-domain connection method, which verifies that one can select either of them depending on application scenarios.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1917-1919
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• 2002

An adaptive algorithm for the acoustic echo canceller is presented. This paper proposes a modified LMS algorithm for the adaptive filter and applys the algorithm to he acoustic echo canceller, An objective of the proposed algorithm is to reduce the hardware complexity. In order to est the performances, a model of the echo path is established, and a program is described. The impulse reponses of the echo path have the length of 125msec or ore, and then the FIR filter with 1000 taps is required. he results from simulations show that the acoustic echo canceller adopting the proposed algorithm achieves the ERLE of 25dB or more within 1sec. If an echo canceller is implemented with this algorithm, its computation quantity s reduced to two times less than the one that is implemented with the normal LMS algorithm, without the degradation of performances.

• Journal of The Korean Astronomical Society
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• v.36 no.spc1
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• pp.45-47
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• 2003

A white light flare was observed at the limb on 16 August 1989 in He 10830 ${\AA}$ spectra, H$\alpha$ slit jaw photo-grams, and white light filter-grams of ${\lambda}=5600{\AA}{\pm}800{\AA}$. The kernels of the white light flare are not spatially related with Ha brightenings, suggesting that the flare energy would be released at the photosphere.

• Journal of the Korean Operations Research and Management Science Society
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• v.16 no.1
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• pp.113-119
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• 1991

The author is currently assistant professor of Management Science at Korea Advanced Institute of Science and Technology, following a few years as assistant professor of Industrial Engineering at Kyung Hee University, Korea. He received his doctorate from the department of Industrial Engineering and Operations Research, University of California, Berkeley. His research interests are time series and forecasting modelling, Bayesian forecasting and the related software development. He is now teaching time series analysis and econometrics at the graduate level.

• Journal of Biomedical Engineering Research
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• v.32 no.3
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• pp.218-229
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• 2011

Companding algorithms have been used to enhance speech recognition in noise for cochlea implant users. The efficiency of using companding for digital hearing aid users is not yet validated. The purpose of this study is to evaluate the performance of the companding for digital hearing aid users in the various hearing loss cases. Using HeLPS, a hearing loss simulator, two different sensorinerual hearing loss conditions were simulated; mild gently sloping hearing loss(HL1) and moderate to steeply sloping hearing loss(HL2). In addition, a non-linear compression was simulated to compensate for hearing loss using national acoustic laboratories-non-linear version 1(NAL-NL1) in HeLPS. In companding, the following four different companding strategies were used changing Q values(q1, q2) of pre-filter(F filter) and post filter(G filter). Firstly, five IEEE sentences which were presented with speech-shaped noise at different SNRs(0, 5, 10, 15 dB) were processed by the companding. Secondly, the processed signals were applied to HeLPS. For comparison, signals which were not processed by companding were also applied to HeLPS. For the processed signals, log-likelihood ratio(LLR) and cepstral distance(CEP) were measured for evaluation of speech quality. Also, fourteen normal hearing listeners performed speech reception threshold(SRT) test for evaluation of speech intelligibility. As a result of this study, the processed signals with the companding and NAL-NL1 have performed better than that with only NAL-NL1 in the sensorineural hearing loss conditions. Moreover, the higher ratio of Q values showed better scores in LLR and CEP. In the SRT test, the processed signals with companding(SRT = -13.33 dB SPL) showed significantly better speech perception in noise than those processed using only NAL-NL1(SRT = -11.56 dB SPL).

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.6
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• pp.1089-1093
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• 2001

A H$\infty$ control theory is applied to the design problem of synthesis filters in a mutirate filter bank. We select a desired pure time-delay system as reference model, and then consider the error system between the mutirate filter bank and the reference model. 1'he synthesis filters minimize the ι$_2$-induced norm of the error system.

• The Transactions of the Korea Information Processing Society
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• v.7 no.7
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• pp.2076-2085
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• 2000

Almost all existing software reliability models are based on the assumptions of he software usage and software failure process. There, therefore, is no universally applicable software reliability model. To develop a universal software reliability model this paper suggests the predictive filter as a general software reliability prediction model for time domain failure data. Its usefulness is empirically verified by analyzing the failure datasets obtained from 14 different software projects. Based on the average relative prediction error, the suggested predictive filter is compared with other well-known neural network models and statistical software reliability growth models. Experimental results show that the predictive filter generally results in a simple model and adapts well across different software projects.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.376-379
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• 1999

The wavelet transform chip is implemented with Daubechies' 4 tap filter. It works at 20MHz in Field Programmable Gate array (FPGA) implementation of Quadrature Mirror Filter(QMF) Lattice Structure. In this paper, the structure contains taro-channel quadrature mirror filter, data format converter(DFC), delay control unit(DCU), and three 20$\times$8 bits real multiplier. The structures for the DFC and DCU need to he regular and scalable, require minimum number of regular, and thereby lead to an efficient and scalable architecture for the Discrete Wavelet Transform(DWT). These results present the possibility that it can be used in Digital Signal Processing(DSP) application faster than Fourier transform at small area with lour cost.

• International journal of advanced smart convergence
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• v.1 no.1
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• pp.48-51
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• 2012

The conventional method for filtering force plate data, low-pass filtering, does not always give accurate results when applied to force data from a custom-made, instrumented treadmill. Therefore, this study compares low-pass filtered data to the same data passed through a wavelet filter. We collected data with the treadmill running. However these include motor noise with ground reaction force at two force plates. We found that he proposed wavelet method eliminated motor noise to result in more accurate force plate data than the conventional low-pass filter, particularly at high speed motor operation. In this study we suggested the convolution wavelet (CNW) which was compared to that of a low-pass filter. The CNW showed better performance as compared to band-pass filtering particularly for low signal-to-noise ratios, and a lower computational load.