• KIPS Transactions on Computer and Communication Systems
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• v.2 no.9
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• pp.381-392
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• 2013

During the last decade, the multitude of advances attained in terminal computers, along with the introduction of mobile hand-held devices, and the deployment of high speed networks have led to a recent surge of interest in Quality of Service (QoS) for video applications. The main difficulty is that mobile devices experience disparate channel conditions, which results in different rates and patterns of packet loss. One way of making more efficient use of network resources in video services over wireless channels with heterogeneous characteristics to heterogeneous types of mobile device is to use a scalable video coding (SVC). An SVC divides a video stream into a base layer and a single or multiple enhancement layers. We have to ensure that the base layer of the video stream is successfully received and decoded by the subscribers, because it provides the basis for the subsequent decoding of the enhancement layer(s). At the same time, a system should be designed so that the enhancement layer(s) can be successfully decoded by as many users as possible, so that the average QoS is as high as possible. To accommodate these characteristics, we propose an efficient transmission scheme which incorporates SVC-aware dual-channel repetition to improve the perceived quality of services. We repeat the base-layer data over two channels, with different characteristics, to exploit transmission diversity. On the other hand, those channels are utilized to increase the data rate of enhancement layer data. This arrangement reduces service disruption under poor channel conditions by protecting the data that is more important to video decoding. Simulations show that our scheme safeguards the important packets and improves perceived video quality at a mobile device.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.2
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• pp.171-178
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• 2016

An aircraft in flight and ATC(: Air Traffic Controllers) working in the Ground Control Center carry out a voice communication using the radio. Voice signal to be transmitted from the aircraft is received to a plurality of terrestrial sites around the country at the same time. The ATC receives the various quality of voice signal from the aircraft depending on the distance, speed, weather conditions and adjusted condition of the antenna and the radio. The ATC carries out a voice communication with aircraft in the optimal conditions finding the best voice signal. However, the present system chooses the values of the CD(: Carrier Dectect) which is determined to be superior to, based on the input voice level, as optimal channel. Thus this system can not be seen to select the optimal channel because it doesn't consider the effect of the noise which influences on the communication quality. In this paper, after removing the noise in the voice signal, we could give the digitized information and an improved voice signal quality, so that users can select an optimal channel. By using it, when operating the training eavesdropping system or the aircraft control, we can expect prevention accident and improvement of training performance by selecting the improved quality channel.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.10 no.12
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• pp.3775-3781
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• 2009

This aims of this study is to examine the factors influencing on the channel satisfaction and preference of the mobile commerce. This model tests various theoretical research relating to technology acceptance model(TAM), transaction cost theory and service quality. The model is tested Structural Equation Modeling(SEM) using AMOS 7.0 analysis on the sample collected from 320 respondents. The result of hypothesis testing is as follows. First, usefulness, asset specificity, uncertainty, trust and responsiveness were influenced by channel satisfaction. Second, channel satisfaction was influenced positively by channel preference.

• Journal of Korean Society of Industrial and Systems Engineering
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• v.22 no.51
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• pp.89-97
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• 1999

Recently, due to the rapid technological progress, CDMA cellular system is widely used for voice, data and multimedia services. But the analysis for voice-data integrated traffic has not been done satisfactorily. Moreover, there is few research results for voice-data integrated traffic. In this paper, we propose an effective channel allocation algorithm for CDMA cellular system which serves voice-data integrated traffic. As for the proposed channel allocation algorithm, new calls and hand-off calls first attempt to connect micro-cell. We model the channel allocating scheme as a Birth-and-Death process. We reserve a few hand-off dedicated channels and calculate the blocking probability of hand-off calls varying the number of hand-off dedicated channels. Then we decide the number of hand-off dedicated channels satisfying the proper QoS(Quality of Service) and minimum blocking probability.

• Proceedings of the Korean Information Science Society Conference
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• 2012.06d
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• pp.334-336
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• 2012

비디오 스트리밍 서비스의 품질을 보장하기 위한 연구가 활발히 진행되고 있다. 그러나 기존 연구들은 모바일 단말의 이동에 따른 무선 채널 상태 변화에 대한 고려가 부족하다. 본 논문에서는 비디오 품질을 향상시키기 위하여, 새로운 비디오 전송 기법을 제안한다. 제안한 기법은 모바일 단말의 이동에 따른 무선 네트워크 상태 변화와 네트워크 혼잡상태를 고려하여 비디오 전송률을 조절한다. 시뮬레이션 결과를 통해 제안한 기법의 성능을 확인하였다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.11a
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• pp.239-242
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• 2008

지상파 DMB는 이동 환경에서 QVGA 급의 영상과 스테레오 오디오를 제공하는 방송 서비스로서 2005년 12월부터 본격적으로 서비스되고 있는데, 최근에는 DMB 환경에서 고품질의 영상과 오디오를 제공하려는 기술에 대한 연구가 이루어지고 있다. 지상파 DMB 환경에서 고품질의 영상 또는 오디오를 제공하기 위해서는 기존의 DMB 서비스에 추가적인 데이터들을 전송하는 것이 필요한데, 하나의 지상파 DMB 방송 채널에 할당되는 전송 비트율이 높지 않다는 점을 감안하면, 이러한 추가적인 데이터들을 효율적으로 전송하는 것이 서비스의 상용화 입장에서는 중요한 요소가 될 수 있다. 본 논문에서는 지상파 DMB 환경에서 멀티채널 오디오 서비스를 제공하고자 할 때, 추가적으로 전송되어야 하는 부가정보 스트림의 효율적인 전송을 위한 패킷화 방법을 제안한다. 지상파 DMB 환경에서 멀티채널 오디오 서비스를 제공하기 위한 부가정보 스트림은 일반 오디오 스트림과 마찬가지로 프레임 단위로 생성이 되며, 약 12kbps의 비트율을 가진다. 그러나, 부가정보 스트림을 지상파 DMB 환경에서 전송하기 위하여, MPEG-2 TS로 패킷화하여 전송하게 되면, 부가정보 스트림의 비트율보다 훨씬 높은 약 32kbps의 전송율을 가지게 된다. 본 연구에서는 이와 같은 문제점을 해결하기 위하여, 멀티채널 오디오 서비스를 위해 필요한 부가정보 스트림의 비트율을 분석하고, 이를 바탕으로 하나의 TS 패킷에 하나 이상의 부가정보 프레임을 포함하여 전송하는 방법을 제안한다. 제안한 방법의 성능 검증을 위해 제안한 방법에 따라 하나 이상의 부가정보 프레임을 하나의 TS 패킷에 포함하여 패킷화하는 것을 시뮬레이션하고, 그 결과를 제시하였다.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.15 no.5
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• pp.3100-3106
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• 2014

This paper develops power measurement system with multiple sensor input channels (voltage-8 channels and current-10 channels) that simultaneously can monitor power components for both supply and load side of power system. The hardware implementation of the proposed system is based on TMS320C42 DSP and signal processing program algorithm to calculate power components use the definition of IEEE Standard 1459-2010 related power quality. The performance of the developed system is tested by using standard ac power source device, and the test results showed that accuracy of the developed system is less than 0.2 %. Also, field test of the proposed system in the three-phase and four-wire power system was implemented. Simultaneous multiple channel measurement and analysis of power components in commercial and industrial electrical power system using the proposed system will be necessary to reduce power quality problems.

• The Transactions of the Korea Information Processing Society
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• v.7 no.8S
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• pp.2679-2687
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• 2000

위성 ATM 망에서 TCP/IP 기반의 응용 서비스를 제공하기 위해서는 위성의 긴 전송지연과 이성 채널의 버스트 오류 특성에 따른 정확한 전송 성능의 영향과 분석이 요구되며, 아울러 성능 목표치가 제시되어야 한다. 본 논문은 2Mbps급 KOREASAT위성을 이용한 위성망과 지상망이 상호 연동된 실제 위성/지상 ATM LANE(LAN Emulation)을 구현하고, 위성 채널의 Eb/No와 TCP 소켓 버퍼 크기, 그리고 전송 메시지 크기를 바꾸어 가면서 그때의 TCP/IP 트래픽과 FTP의 전송성능을 비교 분석하였다. 아울러 IPoA(IP over ATM)기반의 영상회의 서비스에 대한 전송 품질 및 메시지 전달 특성 등을 분석하였다. 연구결과 TCP 소켓 버퍼 크기가 증가할 때 TCP의 전체 성능이 증가함을 알 수 있으나 제한된 최대 윈도우 크기 64kbytes와 위성채널의 오류 특성으로 인해 실제 최대 전송속도는 이론상의 목표치를 달성하지 못함을 보여 주었다. 또한 2Mbps 위성링크에서 Eb/No가 6dB 이하에서는 실제적으로 제공할 수 있는 최대 전송성능의 50%이하로 떨어져 TCP/IP over ATM 서비스가 거의 불가능함을 보여주었다. FTP의 경우, 파일 크기가 작으면 최대 전송속도에 도달하기 전에 파일 전송이 완료되어 대역폭 이용의 효율성 측면에서 TCP/IP의 slow start 메케니즘이 적절하지 않음을 보여주었다. 이울러 IPoA영상회의 서비스에서도 Eb/No가 약 6.5dB 이하에서는 영상이 거의 정지되고, 메시지 전달시간도 ITU에서 권고하는 목표치를 훨씬 상회하여 서비스가 거의 불가능함을 보였다.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.55-60
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• 1996

This paper describes the subjective assessment of the multi-channel audio quality according to the bitrate changing and evaluates the predistortion effect to avoid the unmasked noise after matrixing/dematrxing process in transmission and regeneration of the multi-channel audio. The simulation is processed by the perceptual coding that is MPEG-2 Audio layer II algorithm. We evaluate the quality improvement about predistortion using or not by 384, 320, 256, 128kbps. As the result of the double blind subjective assessment, 5 Grade-Impairment Scale is scored under minus one to 320kbps and so audio quality is evaluated to be perceptible, but not annoying in 3/2 channel. The effect of the predistortion is improved one level in 128kbps and especially speech test material I better improved than music test materials.

• Journal of Internet Computing and Services
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• v.4 no.5
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• pp.61-68
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• 2003

In this paper, we propose a wireless MAC protocol named APRMA, which is capable of supporting the ABR type data service and Maximizing channel utilization. Data terminals with random data packets are not provided slot reservation with PRMA protocol. That is, slot reservation is applicable to the time constraint voice packet exclusively. But the reservation scheme have to be performed for loss sensitive data packet, and contended their quality of service, Therefore, in wireless MAC, reservation technique has to be used for both voice and data services. So the terminal which wants to request for ABR type service, is allocated a minimum bandwidth from system for the first time, If the system have some extra available bandwidth, ABR terminals would acquire additional bandwidth slot by slot, As a result, APRMA protocol can support the data service with loss sensitivity and maintain their channel utilization high.