• Transactions of the Korean Society of Mechanical Engineers A
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• v.25 no.9
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• pp.1400-1409
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• 2001

This paper describes a fiber optic sensor suitable for non-contact detection of ultrasonic waves. This sensor is based on a fiber optic Sagnac interferometer. Quadrature phase bias between two interfering laser beams in Sagnac loop is introduced by a polarization controller. A stable quadrature phase bias can be confirmed by observing the interferometer output versus phase bias. This method eliminates a digital signal processing for detection of ultrasonic waves using Sagnac interferometer. Interference intensity is affected by the frequency of ultrasonic waves and the time delay of Sagnac loop. Collimator is attached to the end of the probing fiber to focus the light beam onto the specimen surface and to collect the reflected light back into the fiber probe. Ultrasonic waves produced by conventional ultrasonic transducers are detected. This fiber optic sensor based on Sagnac interferometer is very effective for detection of small displacement with high frequency such as ultrasonic waves used in conventional non-destructive testing.

• The KIPS Transactions:PartB
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• v.9B no.2
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• pp.199-204
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• 2002

Since motion estimation and motion compensation methods remove the redundant data to employ the temporal redundancy in images, it plays an important role in digital video compression. Because of its high computational complexity, however, it is difficult to apply to high-resolution applications in real time environments. If we have information about the motion of an image block before the motion estimation, the location of a better starting point for the search of an exact motion vector can be determined to expedite the searching process. In this paper, we present an adaptive motion estimation approach bated on temporal correlations of consecutive image frames that defines the search pattern and determines the location of the initial search point adaptively. Through experiments, compared with DS(Diamond Search) algorithm, the proposed algorithm is about 0.1∼0.5(dB) better than DS in terms of PSNR(Peak Signal to Noise Ratio) and improves as high as 50% compared with DS in terms of average number of search point per motion vector estimation.

• Journal of Biomedical Engineering Research
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• v.18 no.4
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• pp.389-396
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• 1997

Several types of electrical artifact seen on electroencephalogram( EEG) records are described. Those are the EOG and the PVC roller pump noise, and so on. An adaptive digital filtering of the electroencephalogram( EEG) is a successful way of suppressing mains interference, but it affects some of the frequency components of the signal, whore artifacts may not be acceptable in some cafes of automatic EEG processing. Thus we studied the method for cancelling these artifacts. This proposed method does not use the reference channel, and is realized by connecting the linear predictor and the fixed FIR filter for the EOG artifact, and by cascading the linear predictor and the noise canceller for the pump artifact. The simulation results illustrate the performances of the proposed method in terms of the capability of interferences suppression. In the results we obtained about 20 dB noise reduction.

• Journal of Biomedical Engineering Research
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• v.34 no.1
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• pp.24-33
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• 2013

To enhance the speech perception of hearing aid users in noisy environment, most hearing aid devices adopt various beamforming algorithms such as the first-order differential microphone (DM1) and the two-stage directional microphone (DM2) algorithms that maintain sounds from the direction of the interlocutor and reduce the ambient sounds from the other directions. However, these conventional algorithms represent poor directionality ability in low frequency area. Therefore, to enhance the speech perception of hearing aid uses in low frequency range, our group had suggested a fractional delay subtraction and integration (FDSI) algorithm and estimated its theoretical performance using computer simulation in previous article. In this study, we performed a KEMAR test in non-reverberant room that compares the performance of DM1, DM2, broadband beamforming (BBF), and proposed FDSI algorithms using several objective indices such as a signal-to-noise ratio (SNR) improvement, a segmental SNR (seg-SNR) improvement, a perceptual evaluation of speech quality (PESQ), and an Itakura-Saito measure (IS). Experimental results showed that the performance of the FDSI algorithm was -3.26-7.16 dB in SNR improvement, -1.94-5.41 dB in segSNR improvement, 1.49-2.79 in PESQ, and 0.79-3.59 in IS, which demonstrated that the FDSI algorithm showed the highest improvement of SNR and segSNR, and the lowest IS. We believe that the proposed FDSI algorithm has a potential as a beamformer for digital hearing aid devices.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.3
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• pp.15-24
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• 2015

Jacket matrices which are defined to be $m{\times}m$ matrices $J^{\dagger}=[J_{ik}^{-1}]^T$ over a Galois field F with the property $JJ^{\dagger}=mI_m$, $J^{\dagger}$ is the transpose matrix of element-wise inverse of J, i.e., $J^{\dagger}=[J_{ik}^{-1}]^T$, were introduced by Lee in 1984 and are used for Digital Signal Processing and Coding theory. This paper presents some square matrices $A_2$ which can be eigenvalue decomposed by Jacket matrices. Specially, $A_2$ and its extension $A_3$ can be used for modifying the properties of hyperbola and hyperboloid, respectively. Specially, when the hyperbola has n times transformation, the final matrices $A_2^n$ can be easily calculated by employing the EVD[7] of matrices $A_2$. The ideas that we will develop here have applications in computer graphics and used in many important numerical algorithms.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.2
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• pp.151-155
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• 2016

In this paper, we present the LED dimming control system by using ADPCM (Adaptive Differential Pulse Code Modulation). This ADPCM apparatus accurately controls the LED current with high resolution reducing the RFI (radio frequency interference) due to the spreading out of the harmonics of current of pulses. Additionally, this makes it easier to increase the accuracy of control operation. This study introduces to make a digitally controlled circuit for controlling LED with high-energy efficient by adopting pulse current to LED. The LED current drive system we designed are two systems, the digitally-controlled unit and analog switching mode power supply unit, can be developed separately. The simulation shows the sigma delta modulation of digital to analog converter's output when the input level is 0.7. From this simulation, the output is approached to accurately 0.15% to target value with 510 pulses.

• Journal of the Korean Institute of Telematics and Electronics T
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• v.35T no.3
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• pp.120-127
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• 1998

In this paper, ultrasonic images of 25MHz bandwidth were acquired by applying peak value variation, time average and depth profile algorithm to acoustic microscopy and its performance was compared and analysed with each other. In the time average algorithm, total reflecting pulse wave from a spot on the coin was converted to digital data in time domain and average value of the converted 512 data was calculated in computer. Time average image was displayed by gray levels colour of acquired N x N matrix average data in the scanning area on the sample. This technique having smoothing effects in time domain make developed an ultrasonic image on a highly scattering area. In depth profile technique, time difference between the reference and the reflected signal was detected with minimum resolution performance of 2ns, thus we can acquired real 3 dimensional shape of the scanning area in accordance with relative magnitude. Through these experiments, peak value, time average and depth profile images were analysed and advantages of each algorithm were proposed.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.1
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• pp.49-57
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• 2002

Including RF Package inductance, substrate coupling through conductive silicon(Si)-substrate is modeled and quantitatively characterized. 2-port substrate coupling model is extended for the characterization of multi-port substrate coupling between digital circuit block and analog/RF circuit block. Furthermore, scalable parameter extraction model is developed. Multi-port substrate coupling can be investigated by linearly superposing a frequency-dependent 2-port substrate coupling model using scalable parameters. In addition, Substrate coupling including RF package inductance effect is quantitatively investigated. It is shown that package effect increases substrate coupling and shifts a characteristic frequencies(i.e., poles) to the higher frequency range. The proposed methodology can be efficiently used to the mixed-signal circuit performance verification.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.10
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• pp.135-143
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• 1998

Several equalizers are applied to the DVD-ROM(Digital Versatile Disc-Read Only Memory) system. Because imperfect writing process may cause nonlinear distortion in the reply signal, neural equalizers which have strong nonlinear mapping capability are applied to the system to compensate the nonlinear distortion. Experimental results to verify that the combination of decision-feedback type equalizers and modulation code is formidable are also given. The experimental results shwo that the SNR gain of the neural equalizers over the conventional equalizers becomes much as the nonlinearity in the channel increases. Furthermore, the decision-feedback type equalizers outperform the equalizers which have no decision-feedback in eliminating ISI(Intersymbol Interference) of random data sequence but there is no performance gain of the decision-feedback type equalizers over the equalizers without decision-feedback when these are applied to compensate the ISI of modulation-encoded data sequence.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.9
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• pp.103-112
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• 1999

In this paper, a speaker-independent isolated world recognition system for a car navigation system is implemented using a general digital signal processor. This paper presents a method combining SNR normalization with RAS as a noise processing method. The semi-continuous hidden markov model is adopted and TMS320C31 is used in implementing the real-time system. Recognition word set is composed of 69 command words for a car navigation system. Experimental results showed that the recognition performance has a maximum of 93.62% in case of a combination of SNR normalization and spectral subtraction, and the performance improvement rate of the system is 3.69%, Presented noise processing method showed good speech recognition performance in 5dB SNR in car environment.