• Journal of Digital Convergence
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• v.11 no.5
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• pp.285-290
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• 2013

In a MPC coding using excitation source of voiced and unvoiced, it would be a distortion of speech waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration the multi-pulses of representation section. To solve this problem, this paper present a method of position compensation(PC-MPC) in a multi-pulses each pitch interval in order to reduce distortion of speech waveform. I was confirmed that the method can be synthesized close to the original speech waveform. And I evaluate the MPC and PC-MPC using multi-pulses position compensation method. As a result, $SNR_{seg}$ of PC-MPC was improved 0.4dB for female voice and 0.5dB for male voice respectively. Compared to the MPC, $SNR_{seg}$ of PC-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.1355-1359
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• 2004

The two way subscriber transmission systems have tendency to spread its carrier frequency bandwidth or information bit rate and average bit error rate according to popularization of high speed information through the digital communication system, transmission medium and the Internet. This fact is an important incentive to realize new systems. These two way subscriber transmission systems usually use same cable or same carrier frequency bandwidth for up stream channel and down stream channel. In the systems, the disturbances of noise, crosstalk or fading affect the characteristics. Specifically, these disturbances cause the decrease of information bit rate and degradation of transmission quality. This paper proposes the improved method of their degradations using the particular feature of two way subscriber transmission systems and it makes clear proposed method is effective by theoretically and some numerical examples.

• 한국초전도학회:학술대회논문집
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• v.10
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• pp.169-172
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• 2000

We have constructed a multi-channel SQUID magnetometer system. The system is designed to operate normally with 10 high temperature direct coupled SQUIDs. The main features of the system include a remote control by serial communication, low noise with wide bandwidth and high slew rate by several MHz modulation, signal conditioning and calibration by digital signal processing.

• Proceedings of the IEEK Conference
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• 2000.06a
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• pp.69-72
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• 2000

This paper shows the more improved design of MRI receiver compared to conventional one based on Elscint Spectrometer. At first, the low-cost ADC is 16 bits, 3MHz sampling A/D converter Comparing to conventional one with signal bits of 14 bits, this device with those of 16 bits helps getting Improved the image resolution improved. If frequency is designed centering around 7.6 MHz to be satisfied in 10 MHz of maximum input bandwidth of ADC. For 1st demodulation, fixed IF is used for the purpose of the implementing multi nuclei system. Control parts & partial digital parts are integrated on one chip(FPGA). In DDC(Digital Down Converter), we got required bandwidth of LPF by controlling its decimation rate. With above considerations, we designed optimal receiver for high resolution imaging to be implemented through PC interface & experimental test of receiver of MRI after receiver's fabrication.

• Journal of Digital Convergence
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• v.10 no.5
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• pp.233-238
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• 2012

In a speech coding system using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of exist a voiced and an unvoiced consonants in a frame. To solve this problem, this paper present a method of LMS-MPC uses individual pitch and LMS(Least Mean Square). I evaluate the MPC and LMS-MPC using LMS. As a result, SNRseg of LMS-MPC was improved 1.5dB for female voice and 1.3dB for male voice respectively. Compared to the MPC, SNRseg of LMS-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• Journal of the Society of Naval Architects of Korea
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• v.60 no.3
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• pp.165-174
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• 2023

Unmanned Surface Vehicle (USV)'s berthing and unberthing is the most difficult maneuvering tasks and have the highest risk of accidents. In this paper, we designed a berthing/unberthing control algorithm given human joystick command for an USV equipped with a waterjet and a bow thruster. The berthing and unberthing maneuvers are performed remotely by a joystick operator at the Ground Control Center (GCC) where the status of USV and environmental situation can be monitored. We interpret the human joystick commands into USV's desired speed, yaw rate, and heading angle commands. next, we developed a control algorithm for the desired target values of MIMO actuators (engine speed, bucket step, nozzle angle, and bow thruster state) to follow the interpreted commands. The validity of the control algorithm is confirmed through simulations and sea trials at Gwang Am port.

• KIEE International Transactions on Power Engineering
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• v.4A no.3
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• pp.129-133
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• 2004

This paper deals with the design and implementation of a TCSC (Thyristor Controlled Series Capacitor) simulator, which is a module for an analog type power system simulator. Principally, it presents configuration of controller hardware/software and its experimental results. An analog type power system simulator consists of numerous power system components, such as various types of generator models, scale-downed transmission line modules, transformer models, switches and FACTS (Flexible AC Transmission System) devices. It has been utilized for the verification of the control algorithm and the study of system characteristics analysis. This TCSC simulator is designed for 50% line compensation rate and considered for damping resister characteristic analysis. Its power rate is three phase 380V 20kVA. For hardware extendibility, its controller is designed with VMEBUS and its main CPU is TMS320C32 DSP (Digital Signal Processor). For real time control and communications, its controller is applied to the RTOS (Real Time Operation System) for multi-tasking. This RTOS is uC/OS-II. The experimental results of capacitive mode and inductive mode operations verify the fundamental operations of the TCSC.

• Proceedings of the Korean Society of Machine Tool Engineers Conference
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• 2004.04a
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• pp.219-224
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• 2004

In recent years, as the demands of VBNS(Very high speed Backbone Network Service) and VDSL(Very high-data rate Digital Subscriber Line) increase, the development of kernel parts of optical communication such as PLC(Planar Light Circuit), Coupler, WDM elements increase. The alignment and the attachment technology are very important to fabricate the optical elements for communication. In this paper, the mechanism of rotational stage, the contact sensing unit, and integrated control circuit for the optical alignment system are studied.

• Journal of Digital Convergence
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• v.11 no.1
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• pp.225-236
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• 2013

The voice recognition has been made continuously. Now, this technology could support even natural language beyond recognition of isolated words. Interests for the voice recognition was boosting after the Siri, I-phone based voice recognition software, was presented in 2010. There are some occasions implemented voice enabled services using Korean voice recognition softwares, but their accuracy isn't accurate enough, because of background noise and lack of control on voice related features. In this paper, we propose a sort of multi-purpose preprocessor to improve this situation. This supports Keyword spotting in the continuous speech in addition to noise filtering function. This should be independent of any voice recognition software and it can extend its functionality to support continuous speech by additionally identifying the pre-predicate and the post-predicate in relative to the spotted keyword. We get validation about noise filter effectiveness, keyword recognition rate, continuous speech recognition rate by experiments.

• International Journal of Naval Architecture and Ocean Engineering
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• v.13 no.1
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• pp.617-627
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• 2021

Underwater Acoustic Channels (UAC) have inherent sparse characteristics. The traditional adaptive equalization techniques do not utilize this feature to improve the performance. In this paper we consider the Variable Adaptive Subgradient Projection (V-ASPM) method to derive a new sparse equalization algorithm based on the Minimum Symbol Error Rate (MSER) criterion. Compared with the original MSER algorithm, our proposed scheme adds sparse matrix to the iterative formula, which can assign independent step-sizes to the equalizer taps. How to obtain such proper sparse matrix is also analyzed. On this basis, the selection scheme of the sparse matrix is obtained by combining the variable step-sizes and equalizer sparsity measure. We call the new algorithm Sparse-Control Proportional-MSER (SC-PMSER) equalizer. Finally, the proposed SC-PMSER equalizer is embedded into a turbo receiver, which perform turbo decoding, Digital Phase-Locked Loop (DPLL), time-reversal receiving and multi-reception diversity. Simulation and real-field experimental results show that the proposed algorithm has better performance in convergence speed and Bit Error Rate (BER).