• Journal of the Korean Society for Railway
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• v.12 no.2
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• pp.254-259
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• 2009

DC transit system has been adopted in the metropolitan area, Korea since 1974. Electric multiple (EMU) in this system always reiterates that acceleration and retardation. When EMU decelerates using electric breaking, regenerative power occurs. Regenerative power can be consumed in vicinity EMU on the same line or in resistor. If DC transit system has inverter for reusing regenerative power, Energy efficiency in DC transit system and the replacement cycle of brake shoe in EMU will be increased and dust due to mechanical braking decreased. This paper present the developed inverter for regenerative power and its test equipment. Test for developed inverter is performed at test equipment and is divided into three items, which are regeneration mode, active filter mode, and system link test.

• Journal of the Korea Society of Computer and Information
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• v.16 no.7
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• pp.157-163
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• 2011

In this paper, configuration control for the Horizontal Maintenance and driving of the mobile inverted pendulum robot has been studied using ARS(Attitude Refrence System). The inverted pendulum technique is getting attention and there have been many researches on the seg-way since the US. Using its 2 freedom, a mobile inverted pendulum robot can move in various modes and Our robot performs goal reaching ARS. Mobile inverted pendulum robot fall down to the forward or reverse direction to converge to the stable point. Kalman Filter is normally used for the algorithm and numerous research is progressing at the moment. To calculate the attitude in ARS using 2 axis gyro(roll, pitch) and 3 axis accelerometers (x, y, z). In this paper we present a two wheel robot system for an autonomous mobile robot. This paper realized the robot control method which is much simpler but able to get desired performance by using the IMU and PID control.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.2
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• pp.44-50
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• 2002

Conventional acoustic echo cancelers using ES(Exponentially weighted Stepsize) algorithm have simple operational configuration and fast convergence speed batter then NLMS algorithm, but they are very weak in external noise because ES algorithm updates filter taps using an average energy reduction rate of room impulse response in specific acoustical condition. So, a new configuration of acoustic echo canceler with stepsize generator and selector is proposed in this thesis. The proposed stepsize generator and selector improve conventional acoustic echo canceler's weakness in external noise and improve the system robustness. The stepsize generator generates additional stepsize value using moving averager, which is the residual noise energy of error signal multiplied by constant ${\gamma}$. The stepsize selector selects the stepsize value that has better performance in an acoustic echo canceler using a coefficient decision factor ${\Delta}_{differ}$ The simulation results show that the proposed algorithm reduces residual error by 5[dB] to 10[dB], improves misadjustment regardless of external noise's SNR. 

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.2
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• pp.84-89
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• 2002

In this paper, we present the design method of noise reduction circuit in telephonic system. The circuit consists of compressor, expander and a filter. The basic idea of a proposed method compresses the audible signal in order to mask the channel noise during transmission and then expand at the reverse rate the transmitted signal to naturally recover the original signal. Of course, there should be no distortion or other degradation of the audio itself in passing through companding(compress/expand) cycle. In the compressing process, the gain of compressor is automatically controlled by the envelope level of input signal in order to increase the effective dynamic range of input signal and to improve the signal to noise ratio. The compressed rate is the root time of a audible signal. The compressed signal should be expanded at the square time of the signal to recover a original signal. Simulation shows the proposed method improves the performance of the noise reduction of a channel noise as well as stability. 

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.45 no.3
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• pp.44-53
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• 2008

In this paper, we present the design of delta-sigma modulation-based class-D amplifier for driving headphones in portable audio applications. The presented class-D amplifier generates PWM(pulse width modulation) signals using a single-bit fourth-order high-performance delta-sigma modulator. To achieve a high SNR(signal-to-noise ratio) and ensure system stability, the locations of the modulator loop filter poles and zeros are optimized and thoroughly simulated. The test chip is fabricated using a standard $0.18{\mu}m$ CMOS process. The active area of the chip is $1.6mm^2$. It operates for the signal bandwidth from 20Hz to 20kHz. The measured THD+N(total harmonic distortion plus noise) at the $32{\Omega}$ load terminal is less than 0.03% from a 3V power supply.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.37 no.3
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• pp.9-17
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• 2000

In this paper we implement computer simulation system of 4FSK signal MODEM using Quadrature detector and analyze overall tranceiver system. We follow the FLEX wireless paging system standards and construct premodulation filter and data frame. We propose an efficient open loop symbol timing recovery algorithm which takes advantage of 128 bit length preamble pattern and also propose a 32 bit UW pattern which Is based on the optimal UW detection method, and excellent aperiodic autocorrelation characteristic. The BER simulation in the fading channel as well as AWGN is performed with BCH coding and Interleaving to the Quadrature detector system and it is shown that a high coding fain occurs in the fading channel rather than AWGN channel.

• The Journal of the Acoustical Society of Korea
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• v.24 no.5
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• pp.294-300
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• 2005

Generally, traditional approaches to track the target position are to estimate ranges and bearings by 2-D MUSIC (MUltiple 519na1 Classification) method. and to associate estimates of 2-D MUSIC made at different time points with the right targets by JPDA (Joint Probabilistic Data Association) filter in the near field. However, the disadvantages of these approaches are that these have the data association Problem in tracking multiple targets. and that these require the heavy computational load in estimating a 2-D range/bearing spectrum. In case multiple targets are adjacent. the tracking performance degrades seriously because the estimate of each target's Position has a large error. In this paper, we proposed a new tracking algorithm using Position innovations extracted from the senor output covariance matrix in the near field. The proposed algorithm is demonstrated by the computer simulations dealing with the tracking of multiple closing and crossing targets.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.36-42
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• 2001

This paper propose new sound localization algorithm that detects the sound source bearing in a closed office environment using two microphone array. The proposed Subband CPSP (Cross Power Spectrum Phase) algorithm is a development of previously Down CPSP method using subband approach. It first split the received microphone signals into subbands and then calculates subband CPSP which result in possible source bearings. This type of algorithm, Subband CPSP, can provide more robust and reliable sound localization system because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, a real time simulation was conducted and it was compared with previous CPSP method. From the simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than 5％ average accuracy for sound source detection.

• The Journal of the Acoustical Society of Korea
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• v.35 no.2
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• pp.134-142
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• 2016

In this paper, a method synthesizing speech signal using the 40 kHz ultrasonic signals reflected from the articulatory muscles was introduced and performance was evaluated. When the ultrasound signals are radiated to articulating face, the Doppler effects caused by movements of lips, jaw, and chin observed. The signals that have different frequencies from that of the transmitted signals are found in the received signals. These ADS (Acoustic-Doppler Signals) were used for estimating of the speech parameters in this study. Prior to synthesizing speech signal, a quantitative correlation analysis between ADS and speech signals was carried out on each frequency bin. According to the results, the feasibility of the ADS-based speech synthesis was validated. ADS-to-speech transformation was achieved by the joint Gaussian mixture model-based conversion rules. The experimental results from the 5 subjects showed that filter bank energy and LPC (Linear Predictive Coefficient) cepstrum coefficients are the optimal features for ADS, and speech, respectively. In the subjective evaluation where synthesized speech signals were obtained using the excitation sources extracted from original speech signals, it was confirmed that the ADS-to-speech conversion method yielded 72.2 % average recognition rates.

• Journal of the Korean Society of Marine Environment & Safety
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• v.20 no.2
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• pp.202-209
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• 2014

In this paper, a developed device for detecting target's location and avoiding collision is proposed. Velocity and acceleration model of target are derived to estimate target's information, i.e. position, velocity and acceleration considering process and measurement noise. Kalman filtering method applied to the estimation process and its results was confirmed by simulation. The distance measurements system using laser sensor for moving target system is also developed to confirm the effectiveness of the proposed scheme. Experiments to get information of moving target with velocity and acceleration model was executed. The data with filtering and without filtering was compared by experiments. Discontinuous measured data was changed to smooth and continuous data by Kalman filtering. It is confirmed that desired data was obtained by applying proposed scheme. UI for measuring and monitoring the target data is developed and visual and auditory alarm function is attached on the system Finally, position estimation system of moving target with good performance is achieved by low price equipments.