• 한국통신학회:학술대회논문집
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• 한국통신학회 1984년도 추계학술발표회논문집
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• pp.92-95
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• 1984

This paper describes the output P.D.F. of various Compandors, optimum, -law and A-law for the Gaussian and Laplacian Density as an input analog signal. Also we consider the truncated densities compensated by weighted impulse or density coefficient.

• 한국경영과학회지
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• 제24권2호
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• pp.69-80
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• 1999

A genetic algorithm to determine the optimal signal sequence and double cycle pattern is described. The signal sequence and double cycle pattern are used as the input for TRANSYT to find optimal signal timing at each junction in the area traffic networks, In the genetic process, the partially matched crossover and simple crossover operators are used for evolution of signal sequence and double cycle pattern respectively. A special conversion algorithm is devised to convert the signal sequence into the link-stage assignment for TRANSYT. Results from tests using data from an area traffic network in Leicester region R are given.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2001년도 하계종합학술대회 논문집(1)
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• pp.77-80
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• 2001

In this paper, it is analyzed the statistical characteristics on the transmission performance of the optical SCM transmission in the CDMA RF signal relay system. Using Mach-Zehnder modulator, optical signal is regenerated into CDMA RF signal. Transfer function model of mach-Zehnder modulator is approximated to the third power of the input signal using Taylor series. BER results are analyzed under the multiple access interference and multi-path fading environment.

• 한국안전학회지
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• 제12권4호
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• pp.3-8
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• 1997

This paper shows a performance estimation of windowing a single tone with added Gaussian noise and uniform noise. Signal-to-noise ratio can be determined by the ratio of the output noisy signal variance to the input noisy signal variance of a window. Standard deviation of noise is reduced by windowing Signal-to-noise ratio of the noisy signal is reduced by the windowing operation. Thus, performance of window function can be determined by this filtering operation that improved the signal-to-noise ratio.

• 전자공학회논문지D
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• 제35D권8호
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• pp.69-75
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• 1998

We demonstrate wavelength conversion of 2.5Gb/s optical signals by four-wave mixing (FWM) in a semiconductor optical amplifier (SOA). We investigate the effect of input pump and signal powers on the coversion efficiency, optical signal-to-noise ratio (OSNR) and extinction ratio to be a measure of performance in a wavelength converter. As a result, we show that the maximum bit error rate (BER) performance can be obtained by co promising among high-vonversion efficiency (minimum Pprobe), high-OSNR (maximum Pprobe) and low-cross-gain saturation effects (Pprobe kept at least 6dB weaker than Ppump). In our experiment, we obtain optimum performance at +3 dBm pump power and -6dBm signal power. The power penalty incurred in the wavelength conversion can be minimized by careful selection of the input pump and signal powers. We show that about 0.5dB power penalty for 3.2nm wavelength coversion at 10-10 BER is achievable.

• 한국정보전자통신기술학회논문지
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• 제4권3호
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• pp.189-196
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• 2011

본 논문에서는 다중입출력 (MIMO : multiple-input multiple-output) 시스템에서 순차적 간섭 제거 기반 (SIC : successive interference cancellation) 신호 검출 기법의 성능을 분석한다. 고려되는 신호검출 기법들은 SIC 기법와 LR-SIC 기법이며, 이러한 신호 검출 기법들의 블록오류확률 (BLER; block error ratio) 성능을 나타내는 식을 유도 하고, 모의실험 결과를 통해 유도된 식과 성공적으로 일치함을 확인한다.

• 전자공학회논문지S
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• 제34S권10호
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• pp.80-88
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• 1997

This paper describes a design of a Wavelet Adaptive Filter(WAF) for the removal of the baseline wandering and the minimization of the signal distortion using by wavelet transform and adaptive filter in the ECG signal. WAF consists of two parts. The first part is wavelet transform that decomposes the ECG signal into seven frequency bands using Vaidyanathan and Hoang wavelet. The second part is adaptive filter that uses the signal of seventh low frequency band among the wavelet transformed signals as primary input and a unit impulse sequence as reference input. For the evaluation of the performance of WAF, we used several baseline wandering elimination filters such as commerical standard filter with cutoff frequency of 0.5Hz and general adaptive filter. We made use of MIT/BIH database and real patient data for the evaluation. In conclusion, WAF showed a lower ST segement distortion than standard filter and adaptive filter and has a higher eliminated noise power than standard filter and adaptive filter.

• 비파괴검사학회지
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• 제28권1호
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• pp.54-58
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• 2008

Time reversal (TR) of body waves in fluids and isotropic solids has been used in many applications including ultrasonic NDE. However, the study of the TR method for anisotropic materials is not well established. In this paper, the full reconstruction of the input signal is investigated for anisotropic media using an analytical formulation, called a modular Gaussian beam (MGB) model. The time reversal operation of this model in the frequency domain is done by taking the complex conjugate of the Gaussian amplitude and phase received at the TR mirror position. A narrowband reference signal having a particular frequency and number of cycles is then multiplied and the whole signal is inverse Fourier transformed. The original input signal is seen to be fully restored by the TR process of MGB model and this model can be more generalized to simulate the spatial and temporal focusing effects due to TR process in anisotropic materials.

• 대한전기학회논문지:시스템및제어부문D
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• 제52권10호
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• pp.598-605
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• 2003

In this paper, we propose the method that separates PPG signal and motion artifact signal from two input signals using new independent component analysis algorithm in time domain. In order to eliminate the large level artifact efficiently, block interleaving. lowpass time filtering and innovation processing technique were applied in ICA preprocessing, and FastICA algorithm were applicable. Experiments are made with the numerical simulation and the real PPG signal including four kinds of motion artifact pattern. Our results show that ICA can effectively detect, separate and remove motion artifact in input signals. Then from the separated signals we restore the original PPG signal and propose a new method which computes SpO$_2$ using ICA mixing matrix.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 1995년도 학술대회
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• pp.165-170
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• 1995

In this paper, it is attempted to apply the adaptive digital signal processing to compensate for a linear distortion of a loudspeaker and implement a real time hardware for that purpose. The real time system is implemented by using the DSP56001, a general purpose signal processor, as a host processor and the DSP56200, a cascadable adaptive FIR filter peripheral chip, as an adaptive digital filter. The system has 1000 taps at a 44.1kHz. After inverse modeling of under_compensation_speaker, the system reduces loudspeaker's linear distortions by pre-processing an input audio signal to loudspeaker. The experiment shows satisfactory results; after adaption with white noise as input signal for 60sec, the flat amplitude and linear phase frequency characteristics is found to lie over a wide frequency range of 100Hz to 20kHz.