• Convergence Security Journal
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• v.11 no.6
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• pp.73-80
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• 2011

In this study, software products developed in the course of testing, software managers in the process of testing software and tools for effective learning effects perspective has been studied using the NHPP software. The delayed software S-shaped reliability model applied to distribution was based on finite failure NHPP. Software error detection techniques known in advance, but influencing factors for considering the errors found automatically and learning factors, by prior experience, to find precisely the error factor setting up the testing manager are presented comparing the problem. As a result, the learning factor is greater than automatic error that is generally efficient model could be confirmed. This paper, numerical example of applying using time between failures and parameter estimation using maximum likelihood estimation method, after the efficiency of the data through trend analysis model selection were efficient using the mean square error and $R^2$(coefficient of determination).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.1B
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• pp.89-98
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• 2000

This paper presents an active noise cancelling model using wavelet transform and subband filter banks based on adaptive filter. The analysis filter banks decompose input and error signals into QMF filter banks of lowpass and highpass bands. Each filter bank uses wavelet filter with dyadic tree structure. The decomposed input and error signals are iterated by adaptive filter coefficients of each subband using filtered-X LMS algorithm. The synthesis filter banks make output signal of wideband with perfect reconstruction to prepare adaptive filter output signals of each subband. The analysis and synthesis niter hants use conjugate quadrature filters for Pefect reconstruction. Also, The delayed LMS algorithm model for on-line identification of error path transfer characteristics is used gain and acoustic time delay factors. The proposed adaptive active noise cancelling modelis suggested by system retaining the computational and convergence speed advantage using wavelet subband filter banks.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.928-938
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• 2008

Though services requiring Quality-of-Service (QoS) guarantees such as Voice over Internet Protocol (VoIP) have been widely deployed on the internet, most of internet backbone networks, unfortunately, do not distinguish them from the best-effort services. Thus estimating the effective capacity meaning the traffic volume that the backbone networks maximally accommodate with keeping QoS guarantees for the services is very important for Internet Service Providers. This paper proposes a test-bed based on ns-2 to evaluate the effective capacity of backbone networks and then estimates the effective capacity of an experimental backbone network using the test-bed in terms of the service standard of the VoIP service. The result showed that the effective capacity of the network is estimated as between 12% and 55% of its physical capacity, which is depending on the maximum delay guarantee probability, and strongly affected by not only the type of offered workload but also the quality standard. Especially, it demonstrated that in order to improve the effective capacity the maximum end-to-end delay requirement of the VoIP service needs to be loosened in terms of the probability to guarantee.

• The Journal of the Acoustical Society of Korea
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• v.27 no.7
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• pp.333-341
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• 2008

In this paper, a new interpolation model for the head related transfer function (HRTF) was proposed. In the method herein, we assume that the impulse response of the HRTF for each azimuth angle is given by linear interpolation of the time-delayed neighboring impulse responses of HRTFs. The time delay of the HRTF for each azimuth angle is given by sum of the sound wave propagation time from the ears to the sound source, which can be estimated by using azimuth angle, the physical shape of the underlying head and the distance between the head and sound source, and the refinement time yielding the minimum mean square error. Moreover, in the proposed model, the interpolation intervals were not fixed but varied, which were determined by minimizing the total number of HRTFs while the synthesized signals have no perceptual difference from the original signals in terms of sound location. To validate the usefulness of the proposed interpolation model, the proposed model was applied to the several HRTFs that were obtained from one dummy-head and three human heads. We used the HRTFs that have 5 degree azimuth angle resolution at 0 degree elevation (horizontal plane). The experimental results showed that using only $30\sim40%$ of the original HRTFs were sufficient for producing the signals that have no audible differences from the original ones in terms of sound location.

• Journal of Food Hygiene and Safety
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• v.17 no.1
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• pp.20-25
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• 2002

This study was aimed to determine the urinary metabolite of profenofos, one of the organophos-phorus pesticides, as the biomarkers of exposure. Urine samples were collected fort 24 hours in metabolic cages after oral administration and dermal application of profenofos to rats. Identification of the derivatized urinary metabolite was determined by GC/MS and excretion time courses of the urinary metabolite was analyzed by GC/MS. Urinary metabolite of profenofos, 4-bromo-2-chlorophenol, was detected in rats urine both after oral administration and dermal application of profenofos. Parent compound was not detected in the experiment. In GC/MS, the mass spectral confirmation for 4-bromo-2-chlorophenol ion was identified at m/z 208.4-bromo-2-chlorophenol was excreted within 48 hours and 72 hours after oral administration and dermal application of profenofos, respectively. In this study, the same urinary metabolite of profenofos was detected both in oral and dermal exposure. Generally, excretion of the urinary metabolite after oral administration was detected faster than after dermal application. It is suggested that urinary 4-bromo-2-chlorophenol could be used as the biomarkers of exposure to profenofos.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.22 no.6
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• pp.566-572
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• 2021

A methodology for the identification and coordinates estimation of air cavities under urban ground or sandy soil using its natural poles and natural resonant frequencies is presented. The potential of this methodology was analyzed. Simulation models of PEC (Perfect Electric Conductor)s with various shapes and dimensions were developed using an EM (Electromagnetic) simulator. The Cauchy method was applied to the obtained EM scattering response of various objects from EM simulation models. The natural poles of objects corresponding to its instinct characterization were then extracted. Thus, a library of poles can be generated using their natural poles. The generated library of poles provided the possibility of identifying a target by comparing them with the computed natural poles from a target. The simulation models were made assuming that there is an air cavity under urban ground or sandy soil. The response of the desired target was extracted from the electromagnetic wave scattering data from its simulation model. The coordinates of the target were estimated using the time delay of the impulse response (peak of the impulse response) in the time domain. The MP (Matrix Pencil) method was applied to extract the natural poles of a target. Finally, a 0.2-m-diameter spherical air cavity underground could be estimated by comparing both the pole library of the objects and the calculated natural poles and the natural resonant frequency of the target. The computed location (depth) of a target showed an accuracy of approximately 84 to 93%.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.185-188
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• 2000

본 연구에서는 화자의 음성신호를 이용하여 추출된 공간정보를 통해 화자의 위치를 실시간으로 추적하는 시스템을 제안하고 실시간 구현하였다. 기존의 대표적인 화자 위치 추출 알고리듬인 CPSP(Cross Power Spectrum Phase)는 실내환경에서 심각하게 일어나는 반향신호에 취약한 단점을 갖고 있으므로 구현된 시스템에서는 위치 추적 성능 개선을 위하여 반향신호에 강인한 hybrid window 알고리듬을 제안하여 적용하였다. Hybrid window 알고리듬은 실내 환경에 적합한 hybrid window를 설계하여 수신된 음성신호에 적용함으로써 반향신호에 의한 상호 상관관계를 줄이고 직접 경로에 의한 신호들의 상관관계를 높임으로써 보다 정확한 시간 지연 추정을 가능하게 한다. 제안된 시스템의 성능분석을 위해 DSP를 이용해 실시간 구현된 하드웨어를 이용해 기존의 CPSP 알고리듬과 제안된 hybrid window를 적용한 시스템을 실제 환경에서의 실험하였고 제안한 알고리듬을 적용한 시스템이 $22\%$ 이상 성공적으로 화자의 위치를 추적하였다.

• Journal of the Korean Society of Groundwater Environment
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• v.6 no.1
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• pp.33-41
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• 1999

Time-lag and amplitudes of the fluctuation of groundwater level are used to determine hydraulic parameters around Han River. The groundwater level at observation wells fluctuates between highest and lowest responding to the Han River level. The factors affecting the groundwater fluctuation are examined. The transmisivity of the Han River alluvium calculated from the measured water table fluctuations measured ranges from 9.39${\times}$10$^1$to 4.02${\times}$10$^3$㎡/day. Based on the hydrogeological data along the river sides, the flow characteristics of groundwater for water level fluctuations were simulated using a ground water flow model MODFLOW.

• Journal of the Korea Institute of Military Science and Technology
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• v.12 no.6
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• pp.768-775
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• 2009

In this paper, the sound source localization system in real time which uses the time delay of arrival signal is proposed. This system uses minimum microphones and surveillance camera for estimation of the sound source localization and sound direction. To apply this system to the military field, four models(model1~model4) are derived. Model 1 can be used to evaluate the sound source localization at the long distance. Model2 and model3 can be applied to estimate the sound direction. Model4 is useful for the special purpose and potable device. It is possible for this system to be used for the military guard and surveillance. As a result of experiments, It is shown that this system can estimate the sound source localization and the sound direction using minimum microphones.

• Journal of KIISE
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• v.43 no.3
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• pp.327-335
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• 2016

In this paper, we propose a buffer management scheme suitable for interactive multimedia services. We consider a typical delay optimization environment so that receiver buffer lengths vary according to the round trip time estimation. In this environment, we propose an optimization technique for minimizing the loss of information that may occur when a reduced buffer length forces I/P/B frames in the buffer to drop. We modeled our problem as a Knapsack Problem for which we used dynamic programing in order to find an approximate solution. The proposed technique is compared with the existing buffer management techniques. Through simulation studies, we found that our approach could increase PSNR, which is important to video quality.