• Journal of KIISE:Databases
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• v.27 no.1
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• pp.13-21
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• 2000

Online analytical processing (OLAP) systems are introduced to support decision support systems. Many researches focussed on efficient processing of aggregate functions that usually occur in OLAP queries. However, most previous researches in the literature are deal with the situation in which aggregate functions arc applied to all the values in a given range. Since those approaches utilize characteristic of aggregate functions applied to a range, they are difficult to be applied to a muitipoint query that is a query considering only some points in a given range. In this paper, we propose the Ranking Index and the flanking Decision Tree (RDT) for efficient evaluation of multipoints MAX/MIN queries. The ranking of possible MAX/MIN values are computed with RDT Then MAX/MIN values can be acquired from the Ranking Index. We show through experiments that our method provides high performance in most situations. In other words, the proposed method is robust as well as efficient. A single common set of precomputed results for both MAX and MIN values is another advantage of the proposed method.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.967-970
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• 2002

광역 네트워크상에서 침입탐지는 백본망에서의 기가비트를 처리할 수 능력이 시스템에 필요로 하고 있다. 하지만 많은 loss 와 시스템 미치는 부하로 시스템 자체에 큰 영향을 미친다. 따라서 본 논문에서는 이러한 단점을 보완하기 위하여 백본망에 있는 각 local network 에 분산 에이전트를 설치하고, 여기에서 발생한 데이터를 다중회귀분석의 회귀계수를 메인 시스템에서 보내 처리함으로써 전체 및 각 Local 네트워크에 대한 밸런스를 조절하고, 감시하는 기능을 제공하는 시스템의 설계방법을 제시한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.9A
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• pp.2173-2183
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• 1998

We consider a frequency-hopped multiple-access communication system that employs reed-solomon code over GF(Q) and M-ary FSK signaling ($M{\leq}Q$) in rayleigh fading channel. We investigate the tradeoff among the modulation symbol size (M), the number of frequency slots, and the code rate in maximizing the average number of successfully transmitted information bits per unit time and unit bandwidth (called normalized throughput). We find that it is desirabel to use a large M in noise-limited environment. In interference-limited environment, it is more improtant to prevent errors (hits) by increasing the number of frequency slots than to correct them with formward error correction techniques or to reduce the error rate by increasing M.

• The Transactions of the Korea Information Processing Society
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• v.5 no.1
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• pp.24-32
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• 1998

The bus contention among bus-based shared-memory multiprocessors limits their performance. In addition, under split bus transaction environment, multiprocessors may make some memory requests unnecessary stand by in the memory access buffer, which makes system performance worse. This unnecessary stand-by can be eliminated by maintaining the bitmap table which contains the status bit for each memory block. However, this mechanism requires a great size of SRAM for the status information, which is fully mapped from the whole memory blocks. To solve this problem, we propose a bitmap cache which exploits partial mapping and locality of references. The simulation results show that the proposed system can greatly reduce the capacity of SRAM for the status information with little deteriorating its performance.

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.30 no.7
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• pp.144-149
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• 2002

This paper presents a technique for real time multiprocessor parallel processing to develop an avionic system computer(ASC) which integrates the avionics control, navigation and fire control, cursive and raster graphic symbol generation into one line replaceable unit. The proposed method has optimal performance by adopting a logically asymmetric structure between four 32bit RISC processors based on the master-slave multiprocessing, a tightly coupled interaction level with the time shared common bus and global memory, and an efficient bus arbitration algorithm. The ASC has been verified through a series of flight tests. The relevant tests also have been rigorously conducted on the prototype ASC such as electrical test, environmental test, and electromagnetic interference test.

• The Magazine of the IEIE
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• v.25 no.5
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• pp.71-78
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• 1998

디지털 TV 방송을 수신하기 위한 디지털 TV 수신기의 핵심 부품을 5개의 IC로 개발하였다. 5개의 칩 세트는 VSB로 변조된 신호를 수신하기 위한 2개의 채널 디코더 IC와 3개의 비디오 신호처리 IC로 구성되어 있다. VSB 수신용 IC는 Syne 및 Timing 복구와 채널 등화 기능을 수행하는 SyncEq와 전송시의 오류를 정정하는 VSB 채널 디코더로 구성되며, 비디오 신호 처리부는 MPEG2 다중화 방식의 Transport Stream을 디코딩하기 위한 역다중화용 IC와 MPEG2 비디오 압축/신장 규격의 MP@HL의 비트스트림을 디코딩하기 위한 비디오 디코더 및 18가지 비디오 포맷을 단일한 출력 포멧으로 변환하여 주며 OSD 등 디스플레이 기능을 위한 비디오 디스플레이 처리용 IC로 구성되어 있다. 이 칩 세트는 VHDL로 설계되었으며 RTL 시뮬레이션과 하드웨어 Emulator로 시스템 레벨에서 검증되었고 0.6u, TLM, CMOS 공정을 이용하여 제작되었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.11C
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• pp.673-681
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• 2011

In this paper, we design multiple antenna relay transmission schemes of low complexity to enhance the spatial diversity in orthogonal frequency division multiplexing (OFDM) systems. The relay scheme underlined, can provide space time block coding (STBC) of OFDM signals in the time domain without IFFT and FFT operations with much reduced complexity. In this paper, we modify the conventional low-complexity STBC-OFDM relaying scheme to be compatible to the existing OFDM systems. In addition, we extend the proposed scheme for multiple antenna relays and provide performance enhancement strategies according to the channel quality information available at the relay. The proposed scheme is shown to improve the diversity and thereby to reduce the outage probability and coded bit error rate. Therefore, the proposed scheme will be promising for service quality improvement or coverage extension based on OFDM like wireless LANs and maritime communications.

• The Journal of the Acoustical Society of Korea
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• v.14 no.2E
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• pp.91-97
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• 1995

This paper describes the implementation of a multi-channel audio codec for HETV. This codec has the features of the 3/2-stereo plus low frequency enhancement, downward compatibility with the smaller number of channels, backward compatibility with the existing 2/0-stereo system(MPEG-1 audio), and multilingual capability. The encoder of this codec consists of 6-channel analog audio input part with the sampling rate of 48 kHz, 4-channel digital audio input part and three TMS320C40 /DSPs. The encoder implements multi-channel audio compression using a human perceptual psychoacoustic model, and has the bit rate reduction to 384 kbit/s without impairment of subjective quality. The decoder consists of 6-channel analog audio output part, 4-channel digital audio output part, and two TMS320C40 DSPs for a decoding procedure. The decoder analyzes the bit stream received with bit rate of 384 kbit/s from the encoder and reproduces the multi-channel audio signals for analog and digital outputs. The multi-processing of this audio codec using multiple DSPs is ensured by high speed transfer of date between DSPs through coordinating communication port activities with DMA coprocessors. Finally, some technical considerations are suggested to realize the problem of real-time operation, which are found out through the implementation of this codec using the MPEG-2 layer II sudio coding algorithm and the use of the hardware architecture with commercial multiple DSPs.

• The Transactions of the Korea Information Processing Society
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• v.4 no.10
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• pp.2562-2570
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• 1997

This paper deals with the design and performance of the adaptive equalizer using high order statistics in order to improve the transmission characteristics of multipath fading channel. The multipath propagational phenomenon occurred in digital radio transmission causes the distortion and ISI of receiving signal. These are main reasons to increase the bit error rate and degrade the performance of receivers. In this paper, the adaptive equalization algorithm using high order statistics of received signal is used instead of CMA algorithm, Bussgang and Godard which are known widely. The performance of this algorithm (residualisi, recovered constellation, calculation) is presented varing SNR. As the result of the computer simulation, equalizer algorithm using high order statistics is better than CMA in the range of low SNR, $10{\sim}20dB$. Therefore, considering the actual communication systems which use the range of $14{\sim}20$ SNR, the adaptive equalizer using high order statistics can be used in the real multipath fading environment.

• Journal of Digital Contents Society
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• v.9 no.1
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• pp.1-6
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• 2008

In traditional wireless communication systems the main power consumption is due to the actual transmissions power. Therefore, energy-constrained wireless networks have gained considerable research attention in recent years. Multiple-input-multiple-output (MIMO) structure, or multiple antenna communication is one of the techniques that has gain considerable importance in wireless systems and networks. In this paper, BER and throughput performance of MISO system with Go-back-N ARQ(Automatic Repeat Request) technique in wireless networks are analyzed and the energy consumption of MISO-based wireless networks is compared with conventional SISO-based wireless networks. Obtained results show the applicability of MISO system with Go-back-N ARQ technique in wireless networks with smart system design.