• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.23 no.10
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• pp.1188-1195
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• 2012

The trend for the development of radar is the active phased-array radar system. The trade-off between the processing speed and the number of the signal process board for the real time signal processing has to be optimized particularly in multichannel radar system. This paper introduces a transmission technique in order to transmit a large amount of received data from an Antenna Part to Signal Process Part. As a result, the number of the S/L board(COTS board) is reduced to one half, and the margin of the data transmission rate is about 2 times higher than the original method.

• 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.

• 제어로봇시스템학회:학술대회논문집
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• 1992.10a
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• pp.829-833
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• 1992

This paper presents the design of an automated EEG analyzing system. The design considerations including processing speed, A/D conversion, filtering, and waveforms detection, are overviewed with the description of the associated EEG characteristics. The architecture of the currently implemented system consists of a p-controller based front-end signal processing unit and a host computer system. The data acquisition procedures are described along with a couple of illustrations of the acquired EEG/EOG signal.

• The Journal of Engineering Geology
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• v.1 no.1
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• pp.85-108
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• 1991

Marine seismic processing is characterized by a great amount of data, several professional processing steps, and various parameters to be decided in each step. In general, adequate processing sequence and optimum parameters are obtained through test processing with sample set of data representing the whole group. The sequence and parameters are then applied in processing the whole data. In this paper, optimum processing sequence and parameters for the data acquired in Korean continental shelf are examined through test processing with real data. Finally, a good-quality migration section is produced using those sequence and parameters decided on the basis of the test results.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2006.05a
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• pp.45-48
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• 2006

The system that can generate biological brain information processing mechanism more precisely may have several abilities such as exact cognition, situation decision, learning and inference, and output decision. In this paper, to implement high level information processing and thinking ability in a complex system, the information processing layer based on the biological brain is introduced. The biological brain information processing mechanism, which is analyzed in this paper, provides fundamental information about intelligent engineering system, and the design of the layer that can mimic the functions of a brain through engineering definitions can efficiently introduce an intelligent information processing method having a consistent flow in various engineering systems. The applications proposed in this paper are expected to take several roles as a unified model that generates information process in various areas, such as engineering and medical field, with a dream of implementing humanoid artificial intelligent system.

• Journal of Radiation Protection and Research
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• v.20 no.2
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• pp.85-95
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• 1995

Radiation measuring system using wireless communication method with single channel has been diveloped and tested. In this system, radiation signals from GM tube are transformed into digital pulses in pulse processing circuit and modulated in FSK (frequency shift keying) circuit for digital communication and then wirelessly transmitted to a receiving unit. The digital pulses received are then demodulated in FSK circuit and converted into radiation dose/dose rate in the data acquisition unit to display on the screen of a personal computer. The performance of this system was evaluated by using both a pulse generator and a standard radiation source(Cs-137). In both cases, digital pulses with 5V were observed in pulse processing circuit without distortion of their shape through wireless communication system. The experimental results of radiation measurement by this system after several test-irradiation of GM detector to a standard radiation source(Cs-137), showed good agreement with irradiation dose rate within 10% difference, and proved that this system could be effectively utillized as radiation measuring instrument. It is expected that this wireless radiation measuring system developed for the first time in Korea, can be used as a radiation monitor as well as a personal dosimeter if we can further improve this system to adopt wireless multichannel communication system.

• Journal of Sensor Science and Technology
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• v.10 no.4
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• pp.239-249
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• 2001

LAPS is a device which is dependent on the bias potential between a pH sensitivity and alternating photocurrent. We implemented the multichannel LAPS device and the detection system which was able to effectively measure the sensor's output by a synchronized detection circuit and multiple methods. The implemented LAPS was structured the multiple sensing sites for analyzing a various components simultaneously. And the system included a time-division method using one pre-amplifier being able to detect the multichannel pH concentration preserving a high S/N ratio and a control part. System hardware consists of a pre-amplifier, digital unit and sensor unit, and software consists of a system program and PC program. As results, we verified the successful operations of system including an implemented pre-amplifier and signal processing units.

• Journal of Broadcast Engineering
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• v.2 no.1
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• pp.56-64
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• 1997

An HDTV frame has 4~6 times more pixels than a DTV frame. In order to encode the HDTV image in real time, parallel processing architectures have been widely used in many HDTV codec developments. That is, an HDTV Image is divided into several subbands and each subband is encoded in parallel using some DTV level encoders. In this paper, we adopt an HDTV codec architecture which divides an HDTV frame into 4 subbands and propose a new scene change detection algorithm using local variance. In addition, a new adaptive multichannel rate control scheme which allocate target bits adaptively to each subband of the HDTV image based on the activities of subband images is suggested in this paper. The activities of subband images are calculated at scene change detection part and reused at the adaptive rate control part. The simulation results show that the proposed scene change detection algorithm detects the scene change of HDTV video very accurately. Also the suggested adaptive multichannel rate control scheme shows better performance than the rate control method which allocates target bits equally to each subbands of the HDTV image.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.39-42
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• 1998

본 논문에서는 방송용 오디오 기기가 갖는 다채널의 특성과 각 채널에 대한 다양한 신호처리 기능의 특성을 고려하여 다채널 디지털 오디오 신호 처리기의 구조를 제안하고 범용 DSP를 이용하여 실시간 병렬 처리 시스템을 구현하였다. 구현된 시스템은 32비트 부동수소점 DSP를 이용하였으며 스테레오 채널의 48KHz 표본화 주파수를 지원하고 20비트 해상도를 갖는 시스템이다. 다채널 디지털 오디오 신호 처리 시스템의 구조는 디지털 신호 처리 과정을 수행하는 디지털 오디오 데이터 처리 부분과 시스템을 제어하기 위한 제어 정보 처리 부분으로 제안하였다. 이러한 구조에 적합한 실시간 시스템을 구현하기 위해 전체 시스템은 4부분의 모듈로 구성된다.

• Journal of the korean Society of Automotive Engineers
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• v.14 no.5
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• pp.95-106
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• 1992

Active control of acoustic noise is an application area of adaptive digital signal processing with increasingly interest along the last year. This work studies the implementation of the multichannel LMS filter and the application of this algorithm for the reduction of the noise inside a vechicle cabin using a number of 'secondary sources' drived by adaptive filtering of a reference noise source. Firstly, we propose the use of an adaptive method for the time-varient optimal convergence factor. Secondly, we propose the use of adaptive delayed inverse model to estimate the elastic-acoustic transfer function presented in vechicle cabin. The original, primary source is often periodic, with a known fundamental frequency. A suitably filtered reference signal can thus be used to drive the secondary sources. An algorithm is presented for adapting the coefficients of an FIR filter feeding such a secondary source in such a way as to minimize the output of a suitably placed microphone. In this algorithm, the coefficients of adaptive filter driving an array of secondary sources can be adapted to minimize the sum of the squares of the outputs of a number of error microphones. The multichannel LMS algorithm displays that such an algorithm is considered suitable to used for the global suppression of noise in vehicle cabin.