• Journal of the Institute of Convergence Signal Processing
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• v.3 no.2
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• pp.53-57
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• 2002

Recently, digital communication system becomes high speed, as communication demand dose not only increases sharply, but an image, voice various kinds data also comes multimedia. In transmitting data at a high speed, the main problem is fading by multipath. A linear or nonlinear distortion arise In multipath channel fading from ISI(Intersymbol Interference). For restoring this distorted signal, A lot of equalizer and adaptive algorithm is introduced. This paper compares and analysises, for improving communication quality in channel which is long delay spread, performance of decision feedback equalizer by RLS algorithm, a channel estimation by RLS-MLSE and viterbi equalizer Particularly, there Is exactly channel estimation of impluse response and excellent property of equalization about channel, which delay spread is long impluse response comparatively and is property of non-minimun phase.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.5
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• pp.122-129
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• 2009

In this paper, we have proposed the QoS (quality of services) control scheme which can adjust data transfer rates of the session participants in response to their radio channel conditions under the time-varying propagation radio environment. The outage probability is selected as the performance index in order to compare the performance of the proposed scheme with the conventional QoS control scheme. The outage probability of the single and multiple paths is mathematically derived here and its numerical results are investigated. The results show that the significant performance improvement can be obtained compared with the conventional scheme and also the diversity gain can improve the radio link outage as the degrees of the multipath diversity increases. Therefore the overall performance improvement in terms of the QoS of the outage probability can be significantly be achieved by applying the proposed scheme.

• Journal of Broadcast Engineering
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• v.20 no.5
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• pp.667-675
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• 2015

As the interest on the high-definition and high-quality broadcasting is increased, the request on the high quality sound signal is enlarged as well as on the video signal's quality. One factor contributing to the high-quality of audio signal is an expansion of reproduction channels like 10.2channel and 22.2channel, but there is a problem of speaker installation issue of these many channels. One solution to solve this problem, we can use frontal loudspeaker array reproduction technique making virtual surround sound. So in this paper, we introduce theocratical analysis on the Wave Field Synthesis used for speaker array based sound reproduction and also present the result about the subjective listening test of reproduction performance based on this technique to check the perfoemance of this system. As a result, we showed WFS based frontal loudspeaker array reproduction method could provide sufficient performance compared to conventional discrete 5.1 channel reproduction method.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.6
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• pp.719-726
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• 2014

This paper addresses quality of experience(QoE) and how to measure and evaluate QoE including its subjective aspects. Adopting the real measurements on the field, a various quality metric have been measured for VoIP(voice over IP) service provided through a wireless interface of WiBro(Wireless Broadband). By analyzing the measured values and correlation between the metrics, we attempt to find a method to evaluate QoE of the VoIP service in a objective way. As a result, it has been shown that QoE of the VoIP service through WiBro network has close relation to the packet-level end-to-end delay, and the delay has close relation to received signal strength indicator(RSSI).

• Journal of Broadcast Engineering
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• v.17 no.5
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• pp.827-837
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• 2012

As broadcasting environments change rapidly to digital, user requirements for next-generation broadcasting service which surpass current HDTV service become bigger and bigger. The next-generation broadcasting service progress from 2D to 3D, from HD to UHD and from 5.1ch audio to more than 10ch audio for high quality realistic broadcasting service. In this paper, we propose 10.2ch based multichannel audio reproduction system for UHDTV. The 10.2ch-based audio reproduction system add two side loudspeakers to enhance the surround sound localization effect and add two height and one ceiling loudspeakers to enhance the elevation localization effect. To evaluate the proposed system, we used APM(Auditory Process Model) for objective localization test and conducted subjective localization test. As a result of objective/subjective localization test, the proposed system shows the statistically same performance compare with 22.2ch audio system and shows the significantly better performance compared with 5.1ch audio system.

• Proceedings of the Korean Society for Quality Management Conference
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• 2006.04a
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• pp.171-176
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• 2006

본 연구는 새로운 교육채널로 부각되고 있는 사이버대학의 학습시스템에 대해 어떤 요인들이 학생의 이용행위에 영향을 미치는지를 연구하고자 하였다. 이를 위해 초기정보기술의 사용자의 사용태도를 분석하는데 유용한 TAM모형과 인터넷 기반 비즈니스에 많이 적용되는 신뢰(Trust)를 이용하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.505-507
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• 2003

IEEE 802.15.3 High­Rate WPAN(Wireless Personal Area Network)는 10m 내외의 무선환경에서 실시간 비디오, 고품질 오디오 및 대용량 파일 전송을 가능케 하는 기술이다. 하나의 네트워크를 piconet단위로 구성하고 하나의 piconet에는 PNC(piconet coordinator)가 TDMA(Time Division Multiple Access) 방식으로 DEV(device)들에게 채널타임을 할당한다. 본 논문에서는 IEEE 802.15.3 High­Rate WPAN 환경에서 하나의 piconet이 형성된 경우 동시에 여러 개의 채널을 최대한 활용하기 위한 다중 채널 MAC을 제안하고 이를 위한 타임 할당 방식을 연구한다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1999.06b
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• pp.79-86
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• 1999

본 논문에서는 ISO/IEC MPEG-2 AAC 표준을 기반으로 한 멀티채널 오디오 인코더 및 디코더(MASIC) 시스템의 실시간 구현 기술에 대해서 기술한다. MPEG-2 AAC 기술은 멀티채널 오디오 부호화 방식의 국제 표준으로써, 지금까지 개발된 멀티채널 오디오 부호화 방식중 최신의 기술이며, 압축율과 오디오 품질이 가장 우수한 것으로 알려져 있다. MASIC 시스템은 인코딩 및 디코딩 기술의 실시간 처리를 위하여 범용 DSP인 TMS320C6701을 사용하였고, 멀티채널 오디오의 고속 입력과 출력을 위한 디지털 인터페이스를 가지고 있으며, 개인용 컴퓨터와의 인터페이스를 위한 PCI 기술이 적용되어 다양한 입출력 모드를 지원하는 특징을 갖는다.

• Proceedings of the Korea Information Processing Society Conference
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• 2008.11a
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• pp.1359-1362
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• 2008

SVC(Scalable Video Coding)는 다양한 장치와 변화하기 쉬운 대역폭을 가지는 네트워크 환경에서 만족할 수 있는 스트리밍 환경을 제공하기 위해서 디자인되었지만, 본 논문에서는 SVC를 간단하면서도 효율적으로 채널 변경시간을 감소시킬 수 있도록 IPTV시스템에 응용하였다. 제안하는 시스템에서는 IPTV 헤드엔드와 LHR(Last Hop Router)사이의 네트워크 대역폭 일부분에 SVC 기본 계층으로 압축된 채널들을 정적 채널에 할당한다. 따라서 사용자가 정적 채널에 속하는 채널을 선택하면 짧은 네트워크 지연 시간 이내에 기본 계층을 전송 받아 동영상을 볼 수 있으며, 곧바로 향상 계층을 전송 받아 사용자는 정상 품질의 영상을 시청할 수 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.2A
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• pp.174-182
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• 2008

For the UHF EPC C1 Gen2 RFID system, to overcome the performance degradation in wireless channel environments, we proposed a modified tag structure which supports channel coding and its applied methodology. The reader consistently monitors the wireless channel status and sets to use channel coding functions for tags and reader only if wireless channel is corrupted. Compared with previous tag complexity, only negligible gates are needed to implement channel coding. By simulation we showed an obvious performance improvement of the identification reliability.