• ETRI Journal
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• v.35 no.3
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• pp.542-545
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• 2013

State-of-the-art voice codecs have been developed to extend the input bandwidth to enhance quality while maintaining interoperability with a legacy codec. Most of them employ a modified discrete cosine transform (MDCT) for coding their extended band. We propose a source filter model-based coding algorithm of MDCT spectral coefficients, apply it to the ITU-T G.711.1 super wideband (SWB) extension codec, and subjectively test it to validate the model. A subjective test shows a better quality over the standardized SWB codec.

• Journal of Korean Society on Water Environment
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• v.30 no.2
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• pp.159-165
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• 2014

Vegetative Filter Strip (VFS) is one of effective Best Management Practices (BMPs) to prevent sediment-laden water problem, is installed at the edge of source area such agricultural area so that sediment occurred in source area is trapped by VFS before it flow into stream or river. Appropriate scale of it needs to be simulated before it is installed, considering various field conditions. In this study, a model using VFSMOD-w model and Genetic Algorithm to determine effective VFS length was developed, it is available to calibrate input parameter related to source area sediment yield through thousands of VFSMOD-w simulations. Useful DBs, moreover, are stored in the model so that very specific input parameters can be used with reasonable values. Compared simulated values to observed data values for calibration, R2 and Nash-Stucliffe model efficiency coefficient were 0.74 and 0.65 in flow comparison, and 0.89 and 0.79 in sediment comparison. The model determined 1.0 m of Filter Length, 0.18 of Filter Slope, and 0.2 cm of Filter Media Spacing to reduce 80% of sediment by VFS. The model has not only Auto-Calibration module also DBs for specific input parameters, thus, the model is expected to be used for effective VFS scale.

• Proceedings of the KIPE Conference
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• 2000.07a
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• pp.735-738
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• 2000

In this paper we propose a series active power filter and a simple calculation method acquiring the reference voltage. A series active power filter is suitable to suppress harmonics produced by voltage type harmonic source such as a diode rectifier with filter capacitor on the DC side The proposed series active power filter system is applied to 3-phase 3-wire power system including the voltage type harmonic source. Experimental result obtained from a laboratory model are shown to verify the viability and effectiveness of the proposed system.

• Proceedings of the KIPE Conference
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• 2001.07a
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• pp.319-322
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• 2001

In this paper, a 3phase-3wire series active power filter compensating current harmonics and unbalanced source voltages is presented. The system is composed of series active power filter and shunt passive filters that are tuned 5th and 7th harmonics. In this system, series active power filter improves harmonic compensation characteristics of the shunt passive filters and compensates the unbalanced source voltages. In the proposed algorithm, compensation voltage for harmonic reduction is calculated by a performance function, and compensation voltage for the unbalanced source voltage is calculated in based on the synchronous reference frame. Some results obtained from the experimental model using the proposed method are presented to demonstrate and confirm its validity.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.61 no.2
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• pp.276-283
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• 2012

In this paper, a practical quasi-optimal DOA(direction of arrival) estimator is proposed in order to develop a one-axis gimbaled ultrasonic source tracker for mobile robot applications. With help of the gimbal structure, the ultrasonic moving source tracking problem can be simply reduced to the DOA estimation. The DOA estimation is known as one of the representative long-pending nonlinear filtering problems, but the conventional nonlinear filters might be restrictive in many actual situations because it cannot guarantee the reliable performance due to the use of nonlinear signal model. This motivates us to reformulate the DOA estimation problem in the linear robust state estimation setting. Based on the assumption that the received ultrasonic signals are noisy sinusoids satisfying linear prediction property, a linear uncertain measurement model is newly derived. To avoid the DOA estimation performance degradation caused by the stochastic parameter uncertainty contained in the linear measurement model, the recently developed NCRKF (non-conservative robust Kalman filter) scheme [1] is utilized. The proposed linear DOA estimator provides excellent DOA estimation performance and it is suitable for real-time implementation for its linear recursive filter structure. The effectiveness of the suggested DOA estimation scheme is demonstrated through simulations and experiments.

• Journal of the Korean Geotechnical Society
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• v.30 no.6
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• pp.23-39
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• 2014

An artificial rainwater reservoir installed in urban areas for recycling rainwater is an eco-friendly facility for reducing storm water effluence. However, in order to recycle the rainwater directly, the artificial rainwater reservoir requires an auxiliary system that can remove non-point source pollutants included in the initial rainfall of urban area. Therefore, the conventional soil filtration technology is adopted to capture non-point source pollutants in an economical and efficient way in the purification system of artificial rainwater reservoirs. In order to satisfy such a demand, clogging characteristics of the sand filter layers with different grain-size distributions were studied with real non-point source pollutants. For this, a series of lab-scale chamber tests were conducted to make a prediction model for removal of non-point source pollutants, based on the clogging theory. The laboratory chamber experiments were carried out by permeating two types of artificially contaminated water through five different types of sand filter layers with different grain-size distributions. The two artificial contaminated waters were made by fine marine-clay particles and real non-point source pollutants collected from motorcar roads of Seoul, Korea. In the laboratory chamber experiments, the concentrations of the artificial contaminated water were measured in terms of TSS (Total Suspended Solids) and COD (Chemical Oxygen Demand) and compared with each other to evaluate the performance of sand filter layers. In addition, the accumulated weight of pollutant particles clogged in the sand filter layers was estimated. This paper suggests a prediction model for removal of non-point source pollutants with theoretical consideration of the physical characteristics such as the grain-size distribution and composition, and change in the hydraulic conductivity and porosity of sand filter layers. The lumped parameter ${\theta}$ related with the clogging property was estimated by comparing the accumulated weight of pollutant particles obtained from the laboratory chamber experiments and calculated from the prediction model based on the clogging theory. It is found that the lumped parameter ${\theta}$ has a significant influence on the amount of the pollutant particles clogged in the pores of sand filter layers. In conclusion, according to the clogging prediction model, a double-sand-filter layer consisting of two separate layers: the upper sand-filter layer with the effective particle size of 1.49 mm and the lower sand-filter layer with the effective particle size of 0.93 mm, is proposed as the optimum system for removing non-point source pollutants in the field-sized artificial rainwater reservoir.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.59 no.6
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• pp.1126-1130
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• 2010

Speech production can be viewed as a filtering operation in which a sound source excites a vocal tract filter. The vocal tract is modeled as a chain of cylinders of varying cross-sectional area in linear prediction acoustic tube modeling. In this modeling the most common implementation assumes equal length of tube sections. Therefore, to model complex vocal tract shapes, a large number of tube sections are needed. This paper proposes a new vocal tract model with unfixed sectionlengths, which uses the reduced lattice filter for modeling the vocal tract. This model transforms the lattice filter to reduced structure and the Burg algorithm to modified version. When the conventional and the proposed models are implemented with the same order of linear prediction analysis, the proposed model can produce more accurate results than the conventional one. To implement a system within similar accuracy level, it may be possible to reduce the stages of the lattice filter structure. The proposed model produces the more similar vocal tract shape than the conventional one.

• Proceedings of the KIEE Conference
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• 1989.11a
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• pp.180-185
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• 1989

In the current source type active power filter, the DC current source is implemented using an inductor with large inductance by maintaining the inductor current constantly. In this case, to compensate the loss of the switching devices of the power converter and the inductor, some real power should be supplied to the filter from the source. This process is analyzed through the equivalent circuit which expresses the loss of the switching devices and the inductor with the equivalent resistor R. This work discusses the validation of this DC side equivalent circuit and points out the problems, through the experiments using the experimental active power filter with 220V, 10KVA ratings, and suggests a more accurate equivalent circuit which puts the saturation voltage of the power transistors and the threshold voltage of the diodes into consideration.

• Journal of Institute of Control, Robotics and Systems
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• v.17 no.3
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• pp.211-217
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• 2011

This paper proposes a quasi-optimal linear DOA (Direction-of-Arrival) estimator which is necessary for the development of a real-time robot auditory system tracking moving acoustic source. It is well known that the use of conventional nonlinear filtering schemes may result in the severe performance degradation of DOA estimation and not be preferable for real-time implementation. These are mainly due to the inherent nonlinearity of the acoustic signal model used for DOA estimation. This motivates us to consider a new uncertain linear acoustic signal model based on the linear prediction relation of a noisy sinusoid. Using the suggested measurement model, it is shown that the resultant DOA estimation problem is cast into the NCRKF (Non-Conservative Robust Kalman Filtering) problem [12]. NCRKF-based DOA estimator provides reliable DOA estimates of a fast moving acoustic source in spite of using the noise-corrupted measurement matrix in the filter recursion and, as well, it is suitable for real-time implementation because of its linear recursive filter structure. The computational efficiency and DOA estimation performance of the proposed method are evaluated through the computer simulations.

• The Transactions of the Korean Institute of Power Electronics
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• v.6 no.5
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• pp.386-393
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• 2001

A series active power filter compensating current harmonics and unbalanced source voltages in a 3phase-3wire power system is presented. The system is composed of series active power filter and shunt passive filters that are tuned at 5th and 7th harmonics. The proposed series active power filter improves harmonic compensation characteristics of the shunt passive filters, reduces source side harmonic currents and compensates the unbalanced source voltages. In the proposed algorithm, compensation voltage for harmonic reduction is calculated by a performance function, and compensation voltage for the unbalanced source voltage is calculated based on the synchronous reference frame. Some results obtained from the experimental model using the proposed method are Presented to demonstrate and confirm its validity.