• Proceedings of the KIEE Conference
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• 2009.04b
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• pp.228-230
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• 2009

Nowadays, the LCL-filter type becomes an attractive grid interfacing for grid-connected Voltage Source Inverter (VSI). LCL-filter can render the current harmonics attenuation around the switching frequency by using smaller inductance than L-filter. This paper presents a study about the LCL-filter design for single-phase grid-connected inverter in Photovoltaic (PV) system. According to the expected current ripple, the inductances of the filter can be determined. Based on the absorbed reactive power on capacitor, the capacitance can be calculated. Due to the theoretical analysis, a LCL-filter based single phase grid connected inverter control system are simulated. The studied simulation results are given to validate the theoretical analysis.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.127-130
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• 2000

In this paper, we explore the possibility to use wavelet decomposition based on modified octave structured 5-level filter banks as a set of features for speech recognition. The HMM (Hidden Markov Model) is used as a recognizer 〔l〕. We compared the performance of the wavelet decomposition with the mel-cepstrum and LPC cepstrum. Experimental results show favorable results.

• Journal of the Korean Institute of Telematics and Electronics
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• v.17 no.4
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• pp.11-17
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• 1980

This paper describes realization of digital filter using $\mu$p controler and discusses measured characteristics of this filter, The idea of P. L. implementation[1] is used in realization, and in this system we utilize a DMA and arithmatic control program of $\mu$p. In this way, we can get more flexible capability than the basic PL method, and higher speed than a filter using general purpose $\mu$p in hardware, Furthermore, we get a 15 KHz sampling frequency(fs) as speed limit in real time processing, and know that this limitation is restricted by execution time (58$\mu$ sec) of DMA control statement. As for filter charateristics, maximum stop band frequencies (fsp) are 1665 Hz, 1445 Hz in Butterworth and Tchebichef approximation, respectively, when fs = 14 KHz. Measured errors between the design specification and the actual result are 0.2dB, 0.1 dB in pass band (when cufoff frequency is 500Hz),-1.1dB (when fsp is 1000Hz), 0.4dB(when fsp is 750 Hz) in stop band frequency of Butterworth and Tchebichef, respectively.

• The KIPS Transactions:PartB
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• v.13B no.4 s.107
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• pp.489-498
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• 2006

Clustering results differ according to the datasets and the performance worsens even while using web documents which are manually processed by an indexer, because although representative clusters for a feature can be obtained by statistical feature selection methods, irrelevant features(i.e., non-obvious features and those appearing in general documents) are not eliminated. Those irrelevant features should be eliminated for improving clustering performance. Therefore, this paper proposes three feature-filtering algorithms which consider feature values per document set, together with distribution, frequency, and weights of features per document set: (l) features filtering algorithm in a document (FFID), (2) features filtering algorithm in a document matrix (FFIM), and (3) a hybrid method combining both FFID and FFIM (HFF). We have tested the clustering performance by feature selection using term frequency and expand co link information, and by feature filtering using the above methods FFID, FFIM, HFF methods. According to the results of our experiments, HFF had the best performance, whereas FFIM performed better than FFID.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.5
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• pp.963-969
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• 2000

In this paper, we define two input-output gains of linear periodic time-varying systems. One is the ratio of output with worst-case l2-norm over all inputs with unit 12-norm. It denotes G($\iota_2,\iota_2$.The other is the ratio of output with worst-case RMS value over all inputs with unit RMS value. It denotes G(RMS, RMS) .It is fact that these two gains are equivalent for linear time-invariant system. In this paper, we prove these two gains are also equivalent for linear periodic time-varying system. In addition, the relationship between two method of obtaining the generalized frequency responses for linear periodic time-varying system is derived. Finally, we apply the defined input-output gains to M-channel filter-bank which is multi-rate signal Processing system, used to speech coding. In the filter-bank, generally, aliasing distortion, magnitude distortion, and phase distortion are present. It is shown that these are kept small if the filter-bank is designed by a method that optimizes the gain G($\iota_2,\iota_2$ of an error system.

• Journal of Sensor Science and Technology
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• v.1 no.2
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• pp.155-163
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• 1992

The phased array antenna has the ability to perform adaptive sampling by directing the radar beam without inertia in any direction. The adaptive sampling capability of the phased array antenna allows each sampling time interval to be varied for each target, depending on the acceleration of each target at any time. In this paper we design a three-dimensional adaptive tracking algorithm for the phased array radar system with a given set of measurement parameters. The tracking algorithm avoids taking unnecessarily frequent samples, while keeping the angular prediction error within a fraction of antenna beamwidth so that the probability of detection will not be degraded during a track update illuminations. In our algorithm, the target model and the sampling rate are selected depending on the target range and the target maneuver status which is determined by a maneuver detector. A detailed simulation is conducted to test the validity of our tracking algorithm for encounter geometries under various conditions of maneuver.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.6
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• pp.163-174
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• 2009

Applying the maximally polyphase decomposition and noble identity to the adaptive filter in subband structure, the conventional fullband affine projection algorithm is translated to the subband affine projection (SAP) algorithm. The Maximally polyphase decomposed SAP (MPDSAP) algorithm is a special version of the SAP algorithm, and its adaptive sub-filters have unity projection dimension. The weight updating formular of the MPDSAP is similar to that of the NLMS algorithm, so it may be more proper algorithm than other AP-type algorithms for many practical applications. This paper presents a new statistical analysis of the MPDSAP algorithm. The analytical model is derived for autoregressive (AR) inputs and the nonunity adaptive gain in the subband structure with the orthonormal analysis filters (OAF), The pre-whitening by the OAF allows the derivation of a simple-analytical model for the MPDSAP with the AR inputs and the nonunity adaptive gain.

• Journal of the Earthquake Engineering Society of Korea
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• v.13 no.5
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• pp.23-29
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• 2009

On December 13, 1996, an earthquake with local magnitude (M$_L$) 4.5 occurred in the Yeongwol area of South Korea. The epicenter was 37.2545$^{\circ}$N and 128.7277$^{\circ}$E, which is located inside the Okcheon Fold Belt. The waveform inversion analysis was carried out to estimate source parameters of the event according to the filtering bandwidth of seismic data. Using 0.02$\sim$0.2 Hz filtering bandwidth, focal depth and seismic moment were estimated to be 6 km and 1.3$\times$10$^{16}$ N$\cdot$m, respectively. This seismic moment corresponds to the moment magnitude (M$_W$) 4.7. The focal mechanism by the waveform inversion and P wave first motion polarity analysis is a strike slip faulting including a small thrust component, and the direction of P-axis is ENE-WSW. The moment magnitude estimated by spectral analysis was 4.8, which is similar to that estimated by waveform inversion. Average stress drop was estimated to be 14.3 MPa.

• Journal of Advanced Navigation Technology
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• v.16 no.4
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• pp.602-610
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• 2012

Ionospheric delay error, one of main error sources in GPS signal, varies with signal frequency. Dual-frequency user uses L1, L2 frequency pseudorange to estimate the ionospheric delay, and there are errors caused by pseudorange measurement noise. So, filter is usually used to smooth the measurement. Weighted hatch filter can estimate optimal smoothed pseudorange measurement. But measurement noise model is needed to use this filter. In this paper, measurement noise modeling is conducted for NDGPS reference station. Using noise modeling result, weighted hatch filter estimate smoothed pseudorange measurement and ionospheric delay. Standard deviation of ionospheric dealy error drops to one-twenty fifth of non-filtered result.

• Transactions of the Korean Society of Mechanical Engineers A
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• v.41 no.5
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• pp.401-407
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• 2017

This paper describes wavenumber filtering for damage detection using single-frequency standing wave excitation and laser scanning sensing. An embedded piezoelectric sensor generates ultrasonic standing waves, and the responses are measured using a laser Doppler vibrometer and mirror tilting device. After scanning, newly developed damage detection techniques based on wavenumber filtering are applied to the full standing wave field. To demonstrate the performance of the proposed techniques, several experiments were performed on composite plates with delamination and aluminum plates with corrosion damage. The results demonstrated that the developed techniques could be applied to various structures to localize the damage, with the potential to improve the damage detection capability at a high interrogation speed.