• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.12a
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• pp.229-232
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• 2000

각종 범용 컴퓨터 및 디지탈 신호처리에서 중요한 역할을 하는 16비트 정수형, 2의 보수 형태의 곱셈연산을 수행하기 위한 고속 승산기구조를 설계하고 시뮬레이션 하였다. 부분곱을 합하는 부분은 일반적으로 전체 곱셈기 처리 지연시간의 절반정도를 차지하므로 이 부분의 설계방법이 곱셈기의 궁극적인 속도향상에 직접적인 영향을 미친다. 부분곱의 개수를 줄이기 위하여 Booth encoder를 사용하였고, partial product(부분곱)의 덧셈시간을 줄이기 위하여 4:2 CSA(can save adder)와 3:2 CSA로 CSA tree를 구성 하였으며, 최종결과는 carry look- ahead tree로 얻어진다. Hyundai CMOS 0.35$\mu\textrm{m}$ 1-poly 4-metal 공정으로 layout하여 설계하였으며, 곱셈시간은 2.7ns(tipical case)이하로 측정되었다.

• Proceedings of the KIEE Conference
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• 2005.10b
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• pp.306-308
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• 2005

The vehicle detection method using pulse radar has the advantage of maintenance in comparison with loop detection method. We have the information about the vehicle being and position by dividing the signals into sectors in accordance with SSC method, and by applying the discriminant function based on stochastical data. We also reduce the signal processing time.

• Journal of the Korean Society for Nondestructive Testing
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• v.20 no.6
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• pp.511-520
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• 2000

Ultrasonic pulse echo method comes to be difficult to apply to the multi-layered structure with very thin layer, because the echoes from the top and the bottom of the layer are superimposed. We can easily meet this problem when the silicon chip layer in the semiconductor is inspected by a SAM equipment using fairly low frequency lower than 20MHz by which severe attenuation in the epoxy mold compound of packaging material can be overcome. Conventionally, deconvolution technique has been used for the decomposition of superimposed UT signals, however it has disabilities when the waveform of the transmitted signal is distorted according to the propagation. In this paper, the wavelet transform based deconvolution(WTBD) technique is proposed as a new signal processing method that can decompose the superimposed echo signals with superior performances compared to the conventional deconvolution technique. WTBD method uses the wavelet transform in the pre-stage of deconvolution to extract out the common waveform from the transmitted and received signal with distortion. Performances of the proposed method we shown by through computer simulations using model signal with noise and we demonstrated by through experiments for the fabricated semiconductor sample with partial delamination at the top of silicon chip layer.

• Information and Communications Magazine
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• v.13 no.11
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• pp.181-191
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• 1996

In this paper we discuss narrow-band interference rejection schemes for spectrum overlay communications where direct sequence spread spectrum (DS/SS) communication system and narrow-band communication systems share the same frequency band. In this case, it is essential to reject the narrow-band signals to guarantee the performance of DS/SS system, while it is not necessarily required to reject the DS/SS signal to recover the narrow-band signal. We discuss several schemes such as time domain processing, frequency domain processing and exploiting the device characteristics to reduce the effect of narrow-band signal on DS/SS communication systems. Furthermore, we suggest a structure of tunable notch filter using gyrator that can easily be customized to ASIC.

• Journal of the Korean Society for Railway
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• v.14 no.2
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• pp.130-136
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• 2011

This paper suggests realization method for real-time speech recognition using VoiceXML in telephony environment based on SIP for Railroad Reservation Service. In this method, voice signal incoming through PSTN or Internet is treated as dialog using VoiceXML and the transferred voice signal is processed by Speech Recognition System, and the output is returned to dialog of VoiceXML which is transferred to users. VASR system is constituted of dialog server which processes dialog, APP server for processing voice signal, and Speech Recognition System to process speech recognition. This realizes transfer method to Speech Recognition System in which voice signal is recorded using Record Tag function of VoiceXML to process voice signal in telephony environment and it is played in real time.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.3 s.303
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• pp.121-128
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• 2005

On the research field for speech signal, neural network mainly uses for the category classification in speech recognition and applies to signal processing. Accordingly, this paper proposes a noise reduction system using a time-delay neural network, which implements the mapping from the space of speech signal degraded by noise to the space of clean speech signal. It is confirmed that this method is effective for speech degraded not only by white noise but also by colored noise using the noise reduction system, which restores the amplitude component of fast Fourier transform.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.5C
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• pp.393-402
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• 2012

In this study, we propose a signal processing algorithm to measure the arousal level of a human subject using a PPG(Photoplethysmography) sensor. From the measured PPG signals, the arousal level is determined by PPI(Pulse to Pulse Interval) and discrete-time signal processing. We ran psychophysical experiments displaying visual stimuli on TV display while measuring PPG signal from a finger, where the nature landscape scenes were used for restorative effect, and the urban environments were used to stimulate the stress. However, the measured PPG signals may include noise due to subject movement and measurement error, which results in incorrect detections. In this paper, to mitigate the noise impact on stimulus detection, we propose a detecting algorithm using digital signal processing methods and statistics of measured signals. A filter is adopted to remove a high frequency noise and adaptively designed taking into account the statistics of the measured PPG signals. Moreover we employ a hysteresis method to reduce the distortion of PPI in decision of emotional. Via experiment, we show that the proposed scheme reduces signal noise and improves stimulus detection.

• Journal of the Korea Convergence Society
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• v.7 no.3
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• pp.77-85
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• 2016

We propose a digital signal processing technic that can compensate the non-linearity inherent in audio amplifiers, and present the result of the simulation. The inherent non-linearity of the audio power amplifier arising from analog devices is compensated via a digital signal processing technic consisting of indirect learning architecture and an adaptive filter. The simulation results show that the compensator can be realized using a third-order polynomial and compensates odd-order non-linearity efficiently. The even-oder non-linearity is mainly due to the dc offset at the output, which is difficult to eliminate with the proposed method. Care must be taken in designing the bias circuit to avoid the DC offset at the output. The proposed technic has significance in that digital signal processing technic can compensate for the impairment that is an inherent characteristic of an analog system.

• Korean Journal of Optics and Photonics
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• v.18 no.5
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• pp.309-316
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• 2007

We demonstrate the wavelength swept source and signal processing for the frequency domain optical coherence tomography. The laser output performance is improved by using a semiconductor optical amplifier with a booster amplifier. The laser generates 14 mW of average power of which wavelength shift in the lasing spectral shape is compensated. Adopting a Fabry-Perot etalon and digital signal processing, the broadening of the beat frequency due to the variance of wavelength-sweep-velocity is calibrated. The optical coherence tomography system shows 154.4 kHz of axial scanning speed, 0.95mm of depth range, and $12{\pm}0.37{\mu}m$ of axial resolution.

• Journal of Sensor Science and Technology
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• v.9 no.4
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• pp.316-326
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• 2000

A new signal processing algorithm for white light interferometry has been proposed and investigated theoretically. The goal of the algorithm is to determine the absolute optical path length of an interferometer with very high precision (<< one optical wavelength). The algorithm features cross-correlation of interferometer fringe scans and hypothesis testing. The hypothesis test looks for a zero order fringe peak candidate about which the cross-correlation is symmetric minimizing the uncertainty of misidentification. The shot noise limited performance of the proposed signal processing algorithm has been analyzed using computer simulations. Simulation results were extrapolated to predict the misidentification rate at Signal to-Shot noise ratio (SNR) higher than 31 dB. Root-mean-square phase error between the computer-generated zero order fringe peak and the estimated zero order fringe peak has been calculated for the changes of three different parameters (SNR, fringe scan sampling rate, coherence length of light source). Results of computer simulations showed the ability of the proposed signal processing algorithm to identify the zero order fringe peak correctly. The proposed signal processing algorithm uses a software approach, which is potentially inexpensive, simple and fast.