• Journal of the Institute of Electronics Engineers of Korea TE
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• v.42 no.3
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• pp.39-44
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• 2005

In this paper, we use the 5-ary frequency shift keying method for robust powerline communication. To transmit the data by the 3-ary carrires among 5-ary carrirers and three carriers are 115kHz, 120kHz and 125kHz. Each time of three carrires is 0.33ms and total transmitting time duration from zero crossing point has 1ms. We use the dummy carriers which are 50kHz and 350kHz for monitoring the powerline channel state, dummy signal duration is received during 1ms. Experimentation result is shown that the implemented PLC transceiver has better than chirp transceiver at the capacitive load testing in spite of low transmission power.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.22 no.9
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• pp.847-851
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• 2011

In this paper, a novel IQ modulator that precisely controls the magnitude and phase of input signals is proposed. The proposed IQ modulator consists of low phase deviation attenuators, a splitter, and a combiner. In order to overcome the phase deviation characteristics found in conventional attenuators, a novel phase compensation technique has been adopted and mathematically analyzed. Linear vector arrays along the center point with large magnitude output signal variations in a full $360^{\circ}$ phase control are achieved on a polar plane by the proposed IQ modulator.

• Proceedings of the KIEE Conference
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• 1996.07c
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• pp.1867-1869
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• 1996

In this paper, in order to obtain the statistic information for parameters of electric and magnetic field waveforms associated with lightning discharges, the electric and magnetic fields produced by lightning discharges in the summer of 1995 were measured by a hemisphere-type electric field sensor and a loop-type magnetic field sensor, which were installed at the campus of Inha University in Inchon. The signals of the electric and magnetic fields were continuously recorded by a transient digitizer having a resolution of 12 bit and a memory capacity of 5000 point. Negative lightning discharges are produced very more than positive lightning discharges in the summer of Korea. The 10 to 90 % rise time of electric and magnetic fields was 2 $2\;{\sim}\;10\;{\mu}s$. And the zero-crossing time of electric field was $15\;{\sim}\;25\;{\mu}s$, whereas that of magnetic field was $20\;{\sim}\;30\;{\mu}s$.

• Proceedings of the KIPE Conference
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• 2002.07a
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• pp.727-729
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• 2002

In this paper, we proposed quad variable rates ADPCM coding method and its implementation on C6000 DSP, which is modified from the standard ADPCM of ITU G.726 for speech quality improvement considering the environmental noise Four coding rates, 16Kbps, 24Kbps, 32Kbps and 40Kbps are used for speech window samples and the rate decision threshold is decided by the environmental noise level. The object of the proposed method is to reduce the coding rate while retaining the speech quality and the speech quality is considerably close to 40Kbps single rate coder with the coding rate close to 16Kbps single rate coder under the environmental noise. The environmental noise level affects the coding rate and the noise level is calculated per every speech window samples. At high noise level, more samples are coded at higher rates to enhance the quality, but at low noise level, only the big speech signals are coded at higher rates and more speech samples are coded at lower coding rates to reduce the coding rates. The influence of the noise on tile speech signal is considerably high for small signals and the small signal has the higher ZCR (zero crossing rate). The method is simulated in PC and to be implemented on C6000 floating point DSP board in real time operations.

• Proceedings of the IEEK Conference
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• summer
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• pp.237-243
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• 2004

A multichannel smart sound sensor capable to detect and identify sound events in noisy conditions is presented in this paper. Sound information extraction is a complex task and the main difficulty consists is the extraction of high­level information from an one-dimensional signal. The input of smart sound sensor is composed of data collected by 5 microphones and its output data is sent through a network. For a real time working purpose, the sound analysis is divided in three steps: sound event detection for each sound channel, fusion between simultaneously events and sound identification. The event detection module find impulsive signals in the noise and extracts them from the signal flow. Our smart sensor must be capable to identify impulsive signals but also speech presence too, in a noisy environment. The classification module is launched in a parallel task on the channel chosen by data fusion process. It looks to identify the event sound between seven predefined sound classes and uses a Gaussian Mixture Model (GMM) method. Mel Frequency Cepstral Coefficients are used in combination with new ones like zero crossing rate, centroid and roll-off point. This smart sound sensor is a part of a medical telemonitoring project with the aim of detecting serious accidents.

• Journal of Magnetics
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• v.20 no.2
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• pp.166-175
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• 2015

In this study, we describe the design method of a Brush-less DC (BLDC) motor with delta winding connection. After designing delta winding connection model with the $60^{\circ}$ flat-top region of the Back Electro-Motive Force (BEMF), an ideal current source analysis and a voltage source analysis, with a 6-step control, were conducted primarily employing Finite Element Method. In addition, as a current controller, we considered the Current Regulator with PI controller using Simulink for the comparison of torque characteristics. When the input current is controlled, the switching regions and reference signals are determined by means of the phase BEMF zero-crossing point. In reality, the input current variation depends on the inductance as well as input voltage, and it causes a torque ripple after all. Therefore, each control method considered in this research showed different torque ripple results. Based on the comparison, the causes of the torque ripple have been verified in detail.

• Journal of Korean Society of Coastal and Ocean Engineers
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• v.4 no.2
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• pp.72-82
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• 1992

To accurately estimate nearshore current in shallow water regions. it is necessary to investigate the irregular wave transformation characteristics and radiation stress produced by random sea waves. This research is to investigate the application or the individual wave Analysis Method. the Component Wave Analysis Method and Representative Wave Analysis Method in the shallow water region. These methods were estimated by wave shallowing transformation when the waves propagate from deep water to shallow water region b)r generating regular waves, two component waves and irregular waves (Bretschneider-Mitsuyasu type). That is, the Indivisual Wave Analysis Method is to investigate from the viewpoint of shallow water transformation of wave statistical characteristics and their zero-down-crossing waves (wave height period and wave celerity). And the component Wave Analysis Method is to investigate from the view point of shallow water transformation of basic frequency component wave and their interference frequency component wave. In addition, this research is to compare the measured mean water level elevation with the calculated one from radiation stress of irreguar waves that is assumed in the three methods above.

• The Transactions of the Korean Institute of Power Electronics
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• v.12 no.1
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• pp.9-18
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• 2007

This paper proposes the modeling of sensorless drive system using 120 degree conduction method for IPM (Interior Permanent Magnet) BLDC motors and analyzes characteristics of the terminal voltage that is used to detect the rotor position. This paper shows that the ZCP (Zero-Crossing Point) of the measured terminal voltage used In sensorless control is ahead of that of the back EMF of IPM motors because they have a saliency. This research also analyzes that the amount of position detection error is related to saliency, rotor speed, and load condition. In addition, this paper shows that motors have bigger advance angles than we have expected because the ZCP of terminal voltage precedes the actual ZCP, and under operation conditions such as heavy load and high speed it may generate abnormal currents that flow toward opposite direction after phase current becomes zero.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.2000-2011
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• 2001

In this Paper, we convert conventional low speed(1x, 6x) DVD systems designed by analog PLL(Phase Locked Loop) into digital PLL to operate at high speed systems flexibly, and present optimal DPLL model in high speed(20x) DVD systems. Especially, we focused on the design of DPLL that can overcome channel effects such as bulk delay, sampling clock frequency offset and asymmetry phenomenon in high speed DVD systems. First, the modified Early-Late timing error detector as digital timing recovery scheme is proposed. And the four-sampled compensation algorithm using zero crossing point as asymmetry compensator is designed to achieve high speed operation and strong reliability. We show that the proposed timing recovery algorithm provides enhanced performances in jitter valiance and SNR margin by 4 times and 3dB respectively. Also, the new four-sampled zero crossing asymmetry compensation algorithm provides 34% improvement of jitter performance, 50% reduction of compensation time and 2.0dB gain of SNR compared with other algorithms. Finally, the proposed systems combined with asymmetry compensator and DPLL are shown to provide improved performance of about 0.4dB, 2dB over the existing schemes by BER evaluation.

• Speech Sciences
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• v.10 no.4
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• pp.117-124
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• 2003

Speaker recognition is the technology that confirms the identification of speaker by using the characteristic of speech. Such technique is classified into speaker identification and speaker verification: The first method discriminates the speaker from the preregistered group and recognize the word, the second verifies the speaker who claims the identification. This method that extracts the information of speaker from the speech and confirms the individual identification becomes one of the most efficient technology as the service via telephone network is popularized. Some problems, however, must be solved for the real application as follows; The first thing is concerning that the safe method is necessary to reject the imposter because the recognition is not performed for the only preregistered customer. The second thing is about the fact that the characteristic of speech is changed as time goes by, So this fact causes the severe degradation of recognition rate and the inconvenience of users as the number of times to utter the text increases. The last thing is relating to the fact that the common characteristic among speakers causes the wrong recognition result. The silence parts being included the center of speech cause that identification rate is decreased. In this paper, to make improvement, We proposed identification rate can be improved by removing silence part before processing identification algorithm. The methods detecting speech area are zero crossing rate, energy of signal detect end point and starting point of the speech and process DTW algorithm by using two methods in this paper. As a result, the proposed method is obtained about 3% of improved recognition rate compare with the conventional methods.