• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.1
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• pp.33-39
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• 2018

An additive noise reduction algorithm for a mass spectrum analyzer is proposed. From the measured ion signal, we first used an estimated threshold from the mode of the measured signal to eliminate background noises with the white Gaussian characteristics. Also, a signal block corresponding to each mass index is constructed to perform a second order curve fitting and a linear approximation to signal block. In this process, the effective signal block composed of only the ion signal can be reconstructed by removing the impulsive noises and the sample signals which are insufficient to be viewed as normal ion signals. By performing curve fitting on the effective signal block, the noise-free mass spectrum can be obtained. To evaluate the performance of the proposed method, a simulation was performed using the signals acquired from the development equipment. Simulation results show the validity of the threshold setting from the mode and the superiority of the proposed curve fitting and linear approximation based noise canceling algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.23 no.7
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• pp.816-821
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• 2019

A counter-type time-to-digital converter was designed using a dual edge T flip-flop. The time-to-digital converter was designed with a $0.18{\mu}m$ CMOS process at a supply voltage of 1.5 volts. In a typical time-to-digital converter, when the period of the clock is T, a conversion error corresponding to the period of the clock occurs due to the asynchronism between the input signal and the clock. However, the clock of the time-to-digital converter proposed in this paper is generated in synchronization with the start signal which is the input signal. As a result, conversion errors that may occur due to asynchronization of the start signal and the clock do not occur. The flip-flops constituting the counters are composed of dual-edge flip-flops operating at the positive and negative edges of the clock to improve the resolution.

• Journal of the Korean Institute of Intelligent Systems
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• v.15 no.2
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• pp.264-269
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• 2005

Advances in sensors, actuators, and computers and developments In information systems offer unprecedented opportunities to implement highly ambitious automation, control and decision strategies. There are also new challenges and demands for control and automation in modern industrial practices. There is a growing need for an active participation from the information systems in industrial, manufacturing and process industry environments because currently there are many control problems. This paper provides pattern recognition to the monitoring system for solvent manufacturing process and shows performance in real-time response with multiple input signals. Data is teamed by a multilayer feedforward network trained by error-backpropagation. The two kinds of test results show that the trained network has the ability to show the current system status with different input data sets.

• 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 IKEEE
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• v.22 no.4
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• pp.1109-1114
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• 2018

Electroencephalogram (EEG) is a tool used to study brain activity caused by external stimuli. In this process, artifacts are mixed and it is easy to distort the signal, so post-processing is necessary to remove it. Independent Component Analysis (ICA) is a widely used method for removing artifact. This method has a disadvantage in that it has excellent performance but some loss of brain wave information. In this paper, we propose a method to reduce EEG information loss by restricting the filter coverage using eye blink information obtained from Eyetracker. We then compared the results of the proposed method with the conventional method using quantization methods such as Signal to Noise Ratio (SNR) and Spectral Coherence (SC).

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.12
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• pp.1-11
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• 1993

This paper concentrates on models useful for analyzing the error performance of ML(Maximum Likelihood) estimators of a single unknown signal parameter: that is the error intensity model. We first develop the point process representation for the estimation error and the conditional distribution of the estimator as well as the distribution of error candidate point process. Then the error intensity function is defined as the probability dessity of the estimate and the general form of the error intensity function is derived. We then develop several intensity models depending on the way we choose the candidate error locations. For each case, we compute the explicit form of the intensity function and discuss the trade-off among models as well as the extendability to the case of multiple parameter estimation.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1227-1230
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• 1998

Phase control methods are used to expand the sound image in general AV system. However, these methods are effective only to the signal under 1kHz, and the listener must be located in front center of the speaker system. In this paper, we realize the realtime processing system in which phase shifting method is dominant at low frequency and precedence effect is dominant at high frequency. Two sound cards are used to process the audio signal in realtime with 16 bits stereo channel of 44.1 kHz sampling frequency. And the analog circuit is designed to process the phase shifting. In experiments the usefulness of the proposed stereo system is confirmed.

• Proceedings of the IEEK Conference
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• 2009.05a
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• pp.313-315
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• 2009

In this thesis, A methodology of system implement for PID controller, PWM logic, HSC logic, Host Communication and external DAC interface are implemented into single FPGA chip is proposed. The implemented system is used to control the speed of DC servo motor. A DATA block transfers set point value(SV) and P, I, D gain parameters to the corresponding Blocks respectively by the Host Communication to Computer. A HSC block generates process value(PV) by a pulse and $90^{\circ}$ shifted pulse from the encoder A PID block makes error(E) signal from the set value and process value and output manufacture value(MV) through the PID controller. In PWM block using the MV from the PID block, drives H-bridge controlling the Motor. Also DAC interface controls the DAC to graph the digital signal such as SV, PV, E, MV in FPGA onto the Oscilloscope.

• Proceedings of the KSR Conference
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• 2010.06a
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• pp.546-548
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• 2010

This paper proposes a new baseline-free TR-based SHM method in which the time-reversal (TR) property of the guided Lamb waves is utilized. The new TR-based SHM method has two distinct features when compared with the other existing SHM techniques: (1) The measurement- based backward TR process is replaced by the computation-based process (2) In place of the comparison method most commonly used for SHM, the TOF information of the damage signal extracted from the reconstructed signal is utilized for the damage diagnosis. For the damage diagnosis, the imaging method is adopted to efficiently detect damage by representing the damage as an image. The proposed TR-based SHM technique is then validated through the damage diagnosis experiment for an aluminum plate with a damage at different locations.

• Proceedings of the KIEE Conference
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• 1999.07g
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• pp.3221-3223
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• 1999

OAEs are low-level sounds produced by cochlea as part of the normal hearing process. OAEs can be measured with a microphone placed in the outer ear-cannel. We can diagnose cochlea's condition by using OAEs. To diagonose it's condition, however, is difficult by reason of OAEs' tiny. Thus, It need to a method using latency which essential diagnosing a time-element of OAEs. This study proposes a latency detection algorithm using 7-QMF for more effective detection of latency, 7-QMF designed by wavelet theory can process signal without a losses of information. The latency-detector based on 7-QMF is superior to former method.