• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.368-374
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• 2016

This paper presents constructing transmitter and receiver by using a direct sequence spread spectrum techniques to DPSK (Differential Phase-Shift Keying) scheme in underwater acoustic communication. Since DPSK signal can be demodulated if the receiver knows only the phase difference between the adjacent bits, DPSK receiver structure has the advantage of being simplified. In the conventional receiver, two adjacent symbols of transmitted signal before despread are passed to the transition correlator that detects data by comparing maximum correlation outputs. At this time, the error for maximum value of the correlator output may increase because of low SNR (Signal to Noise Ratio) or high Doppler shift frequency according to the underwater channel. In this paper, we propose a method for accurate detection result using the width as well as the magnitude among outputs produced by the correlator. The performances of the proposed method was evaluated by simulation and lake trial data.

• The Journal of the Acoustical Society of Korea
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• v.38 no.4
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• pp.371-377
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• 2019

In this paper, we present the sea trial results of long distance underwater acoustic communication in the East Sea, October 2018. One transmitter and sixteen vertical array receivers were used to collect underwater acoustic communication signals, and the maximum distance between the transmitter and the receiver was 90 km. Information was transmitted by BFSK (Binary Frequency Shift Keying) and BCSK (Binary Chirp Shift Keying) method, which are typical digital frequency modulation techniques. Experimental results show that there is no error in all cases at the transmission distance of 60 km, and BFSK and BCSK have average uncoded bit error rate of 0.0197 and 0.0007 respectively without channel coding at 90 km transmission distance.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.20 no.1
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• pp.101-106
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• 2020

This paper relates with the VS-QE-MMA (Varying Stepsize-Quantized Error-MMA) based on the varying stepsize for improving the equalization performance in the QE-MMA adaptive equalization algorithm that is possible to reducing the intersymbol interference occurred at channel. The SE-MMA use the high-order statistics of transmitted signal and sign of error signal. The QE-MMA was appeared for the H/W implementation easiness substitutes the multiplication and substraction into the shift and substraction in the updating the tap coefficient based on the power-of-two operation of error signal magnitude. The QE-MMA gives degradation of equalization performance due to the such simplification of arithmetic operation. For improving this problem, the propose algorithm, namely VS-QE-MMA, applies the varying stepsize of the nonlinear transformation of error signal. It was confirmed by simulation that the VS-QE-MMA gives better performance than current QE-MMA in the same channel and signal to noise ratio. As a result of simulation, the VS-QE-MMA has more better performance in the every performance index, and it was also confirmed that the varying stepsize effect can be obtained in the greater than 10dB of signal to noise ratio.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.12 no.2
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• pp.308-315
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• 2001

In this paper, a new power amplifier is proposed for reduction of amplified RX band noise signals as well as intermodulation distortion signals using feedforward technique. This power amplifier is implemented for IMT-2000 basestation TX frequency band. Both TX band intermodulation distortion signals and RX band noise signals are reduced by controlling variable attenuator, phase shifter and error amplifier. The proposed power amplifier, which contains two loops-intermodulation distortion signals cancellation loop and RX band noise signals cancellation loop, can provide duplexer with low TX path insertion loss for various wireless communication systems due to choice of loose RX attenuation characteristic. The principle of the proposed amplifier is described graphically based on the conceptual schematic diagram. A two-tone test for power amplifier is done at 2.14GHz with frequency spacing of 5MHz, and RX band rejection test is done over RX full band of 60MHz with 1.95GHz center frequency. Experimental results represent that the cancellation performance of intermodulation distortion signals and RX band noise signals are more than 3 1dB and 21dB, respectively.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3
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• pp.18-25
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• 1997

This study describes a modified preemphasis formula, what we call energy-dependent preemphasis(EDP). This uses the normalized short-term energy of speech signal, with the assumption that the source characteristics of the glottal pulses and the radiation characteristics of the lips are approximately proportional to the energy of speech signal. Using this method, speech analyses, such as AR spectrum estimation, formant detection, are performed for nonstationary starting parts of 5 Korean single vowels. The results are compared with the conventional two preemphasis methods. We found that the proposed preemphasis gave enhanced spectral shapes and more accurate formant frequencies and avoided overlapping phenomenon of adjacent two formants.

• 전자공학회논문지 IE
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• v.44 no.4
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• pp.41-47
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• 2007

In this paper, the performance of simulation model in mobile to mobile Rayleigh and Rician fading channel is analyzed and simulation results are compared. The auto-correlation and cross-correlation functions of Rayleigh fading channels not only based on received signals of Clarke's fading model and but also based on received signals of phase shifted Clarke's fading model are derived and their simulation results are found. The two models are also applied to mobile to mobile Rician fading channel models and their numerical results of auto-correlation functions of complex envelop received signals shows the statistical properties match the theoretical values very well and rapidly converge with the small number such as N=8.

• Proceedings of the KIEE Conference
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• 2000.11c
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• pp.543-546
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• 2000

본 논문에서는, 모의 진공차단기 내에서의 구리 증기입자의 공간적 및 시간적 밀도분포를 레이저유기형광법 (LIF)에 의해 계측하였다. 레이저 트리거 방전에 의해 생성된 어블레이션(ablation) 플라즈마에서의 구리입자를 324.75nm의 Nd:YAG 레이저를 이용하여 여기시키고, 이때 여기준위에서 기저준위로 천이할 때의 형광신호(510.55nm)를 분광기 및 광전자증배관을 이용하여 계측하였다. 음극점으로부터 넓은 각도를 가지고 전자가 튀어나오며, 넓은 범위에 걸쳐 전자나 이온이 존재하는 것을 알 수 있었다. 또한 음극으로부터 8mm ( x=0, y=8mm 진 위치에서 가장 강한 신호와, 2mm ( x=0, y=2mm 가장 약한 신호를 얻을 수 있었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.1
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• pp.1-11
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• 2016

We investigate compressive sensing techniques to estimate sparse channel in Orthogonal Frequency Division Multiplexing(OFDM) systems. In the case of large channel delay spread, compressive sensing may not be applicable because it is affected by length of measurement vectors. In this paper, we increase length of measurement vector adding pilot information to OFDM data block. The increased measurement vector improves probability of finding path delay set and Mean Squared Error(MSE) performance. Simulation results show that signal recovery performance of a proposed scheme is better than conventional schemes.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.763-767
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• 2008

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. So, We proposed TS(Transition Segment) including unvoiced consonant searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This research present a new method of TS approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high quality approximation-synthesis waveforms within TS by using frequency information of 0.547kHz below and 2.813kHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TS. This method has the capability of being applied to a new speech coding of Voiced/Silence/TS, speech analysis and speech synthesis.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39C no.10
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• pp.887-895
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• 2014

Drowsy driving is a large proportion of the total car accidents. For this reason, drowsiness detection and warning system for drivers has recently become a very important issue. Monitoring physiological signals provides the possibility of detecting features of drowsiness and fatigue of drivers. Many researches have been published that to measure electroencephalogram(EEG) signals is the effective way in order to be aware of fatigue and drowsiness of drivers. The aim of this study is to extract drowsiness-related features from a set of EEG signals and to classify the features into three states: alertness, transition, and drowsiness. This paper proposes a drowsiness detection system using errors-in-variables(EIV) for extraction of feature vectors and multilayer perceptron (MLP) for classification. The proposed method evaluates robustness for noise and compares to the previous one using linear predictive coding (LPC) combined with MLP. From evaluation results, we conclude that the proposed scheme outperforms the previous one in the low signal-to-noise ratio regime.