• Journal of the Korean Geophysical Society
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• v.2 no.2
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• pp.100-110
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• 1999

In order to find a good seismic source for high-resolution reflection surveys on a tidal flat, characteristic features of several sources were examined through test recording by the walkaway configuration. The sources comprise portable vibrator, 1.2 kg monkey spanner, 4.7 kg sledge hammer, and weight drops with a 30 kg and a 100 kg iron ball, respectively. We analyzed raw and filtered seismic data for the resolution of individual events, then compared various seismic sources to define suitable one for the high-resolution survey in terms of energy level and frequency contents. The study reveals that the portable vibrator is the most suitable source for the detection of and imaging geologic structures less than 20-30 m deep in a tidal flat. We ascribe this to the wide frequency band and high-frequency contents of the portable vibrator. In contrast, the hammer may be an alternative where the target depth increases up to 100 m.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.11
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• pp.130-137
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• 2013

This paper presents analog front-end(AFE) circuits for bio-potential measurement. The proposed AFE is composed of IA(instrument amplifier), BPF(band-pass filter), VGA(variable gain amplifier) and SAR(successive approximation register) type ADC. The low gm(LGM) circuits with current division technique and Miller capacitance with high gain amplifier enable IA to implement on-chip AC-coupling without external passive components. Spilt capacitor array with capacitor division technique and asynchronous control make the 12-b ADC with low power consumption and small die area. The total current consumption of proposed AFE is 6.3uA at 1.8V.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.44 no.7 s.361
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• pp.37-44
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• 2007

A low-voltage high-linear bipolar OTA and its application to IF bandpass filter for GSM cellular telephone are presented. The OTA consists of a low-voltage linear transconductor, a translinear current gain cell, and three current mirrors. The bandpass filter is composed of two cascaded identical second-order bandpass filters, which consist of a resistor, a capacitor, and a grounded simulated inductor realized with two OTA's and a grounded capacitor. SPICE simulations using an 8 GHz bipolar transistor-array parameter show that the OTA with a transconductance of 1 mS exhibits a linearity error of less than ${\pm}2%$ over an input voltage range of ${\pm}0.65\;V$ at supply voltages of ${\pm}2.0\;V$. Temperature coefficient of the transconductance is less than $-90ppm/^{\circ}C$. The bandpass filter has a center frequency of 85 MHz and Q-factor of 80. Temperature coefficient of the center frequency is less than $-182ppm/^{\circ}C$. The power dissipation of the filter is 128 mW.

• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.367-374
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• 2003

In this paper, we propose an audio watermarking method using the attribute of tonal masker. First, the attribute of tonal masker as an audio watermarking attribute is analyzed. According to existing researches, it is possible to be imperceptible modulation for the energies of the frequencies that compose a tonal masker. And when the relation between the tone energy and the left or right frequency energy after various signal processing is compared with the one before the processing, very few changes are showed. We propose an audio watermarking method using these attributes of tonal masker. A watermark bit is embedded by the modulation of the difference between the two neighboring frequency energies of a tone. In the detection, the modulated the tonal masker is searched using the key wed in the embedding without original audio and the embedded watermark bit is detected. After each attack of noise insertion, band-pass filtering, re-sampling, compression, echo transform and equalization, the detection error ratios of the proposed method were average 0.11%, 1.26% for Classics and Pops. And the SDG(Subjective Diff-Grades) scale evaluation of the sound quality of the watermarked audio result in the average SDG -0.31.

• The Journal of the Acoustical Society of Korea
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• v.15 no.4
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• pp.83-92
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• 1996

This paper describes Lombard effect compensation and noise suppression so as to reduce speech recognition error in noisy environments. Lombard effect is represented by the variation of spectral envelope of energy normalized word and the variation of overall vocal intensity. The variation of spectral envelope can be compensated by linear transformation in cepstral domain. The variation of vocal intensity is canceled by spectral magnitude normalization. Spectral subtraction is use to suppress noise contamination, and band-pass filtering is used to emphasize dynamic features. To understand Lombard effect and verify the effectiveness of the proposed method, speech data are collected in simulated noisy environments. Recognition experiments were conducted with contamination by noise from automobile cabins, an exhibition hall, telephone booths in down town, crowded streets, and computer rooms. From the experiments, the effectiveness of the proposed method has been confirmed.

• The Transactions of the Korean Institute of Electrical Engineers C
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• v.48 no.5
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• pp.326-332
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• 1999

In this paper, we designed and fabricated High-Tc Superconducting (HTS) microstrip bandpass filters using Stepped Impedance Resonators(SIR) and studied on their characteristics. The $high-T_c$ superconducting $Y_1Ba_2Cu_3O_{7-x}$ epitaxial films were deposited by Pulse Laser Deposition (PLD) system on MgO. The fabricated filters were designed so as to operate in Ku band with central frequency 17.25 GHz, bandwidth 2.896% and ripple 0.01 dB. These filters were composed of parallel coupled microstrip SIR of which impedance ratio (K) are 0.5, 1.5. In the measured response, HTS filters had showed insertion loss below -0.5 dB. For comparison with normal conducting filter, we fabricated the Au counterpart that consists of the resonators as K=1.5 in the same dimension and measured performance of the Au filter. In comparison, HTS filter designed optimally got superior response to gold conterpart.

• Transactions of the Korean Society of Automotive Engineers
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• v.13 no.4
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• pp.152-159
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• 2005

The method of induction noise reduction can be classified by using passive control or active control method. However, the passive control method has a demerit to reduce the effect of noise reduction to low frequency (below) 500Hz) range and to be limited in a space of the engine room. Whereas, the active control method can overcome the demerit of passive control method. The algorithm of active control is mostly used in LMS (Least-Mean-Square) algorithm because it can obtain the complex transfer function easily in real-time. Especially, Filtered-X LMS (FXLMS) algorithm is applied to an ANC system. However, the convergence performance of LMS algorithm could not match if the FXLMS algorithm is applied to an active control of the induction noise under rapidly accelerated driving conditions. So, in order to solve the problem in this study, the Moving Bandpass Filter(MBPF) was proposed and implemented. The ANC using MBPF for the reduction of the induction noise shows that more noise reduction as 4dB than without MBPF.

• Journal of IKEEE
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• v.26 no.2
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• pp.271-279
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• 2022

This paper proposes an advanced dead-time compensation method for dual inverter with a floating capacitor. The dual inverter with floating capacitor is composed of double two-level inverters and a bulk electrolytic capacitor. The output voltage of the dual inverter is dropped by the conduction voltage of the power semiconductors. The voltage drop and dead-time cause the fundamental and harmonic distortions of output currents. When supplied power for OEW-load is low, the dual inverter operates as single inverter for effective operation. The dead-time compensation method for the dual inverter operated as single inverter is needed for reliability. The proposed method using band pass filter in this paper compensates dead-time, dead-time error and changed voltage drop error of power semiconductors for the dual inverter and dual inverter operated as single inverter. The effectiveness of the proposed method is verified by simulation results.

• The Journal of the Acoustical Society of Korea
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• v.29 no.4
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• pp.243-250
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• 2010

This paper proposes a spectral modeling of Korean traditional instrument, Haegeum, using cepstral analysis to naturally describe Haegeum sounds varying with time. To get a precise result of cepstral analysis, we set the frame size to 3 periods of input signal and more cepstral coefficients are used to extract formants. The performance is enhanced by flexibly controlling the cutoff frequency of bandpass filter depending on the resonances in the synthesis process of sinusoidal components and the deleting peaks remained in the residual signal. To detect the change of pitch, we divide the input frames into silence, attack, and sustain region and determine which region the current frame is involved in. Then, the proposed method readjusts the frame size according to the fundamental frequency in the case of the current frame is in attack region and corrects the extraction errors of the fundamental frequency for the frames in sustain region. With these processes, the synthesized sounds are much more similar to the originals. The evaluation result through the listening test by a Haegeum player says that the synthesized sounds are almost similar to originals (96~100 % similar to the original sounds).

• The Journal of the Acoustical Society of Korea
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• v.28 no.6
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• pp.526-533
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• 2009

This paper proposes a formant synthesis method of Haegeum sounds using cepstral envelope for spectral modeling. Spectral modeling synthesis (SMS) is a technique that models time-varying spectra as a combination of sinusoids (the "deterministic" part), and a time-varying filtered noise component (the "stochastic" part). SMS is appropriate for synthesizing sounds of string and wind instruments whose harmonics are evenly distributed over whole frequency band. Formants extracted from cepstral envelope are parameterized for synthesis of sinusoids. A resonator by Impulse Invariant Transform (IIT) is applied to synthesize sinusoids and the results are bandpass filtered to adjust magnitude. The noise is calculated by first generating the sinusoids with formant synthesis, subtracting them from the original sound, and then removing some harmonics remained. Linear interpolation is used to model noise. The synthesized sounds are made by summing sinusoids, which are shown to be similar to the original Haegeum sounds.