• Journal of KIISE:Computer Systems and Theory
• /
• v.26 no.12
• /
• pp.1493-1503
• /
• 1999

ATM Clustering System과 같이 SAN(System Area Network) 환경에서 동작하는 시스템은 낮은 지연시간과 넓은 대역폭의 네트워크가 필수적이나 기존의 에러 복구 프로토콜들은 이러한 요구를 충족시키기에는 큰 오버헤드를 가지고 있다. 제안된 새로운 에러 복구 프로토콜은 ATM Clustering System 환경에서 최적의 성능을 나타내는 light-weight 프로토콜로 에러가 없는 상황과 에러 복구가 진행중인 상황에 따라 acknowledgement 주기를 적응적으로 변화시키는 adaptive acknowledgement scheme를 제안하여 적용하였다. 제안된 프로토콜은 상용 툴인 SDT를 이용한 논리 검증 받았고, DEVSim++ 환경에서의 성능 분석을 통해 프로토콜이 최상의 성능을 보이기 위한 파라메터 값을 찾았고, 이 값을 적용하였을 때의 성능을 기존의 프로토콜과 비교하여 제안된 프로토콜이 더 우수함을 확인하였다.Abstract While a system working with SAN, such as ATM Clustering System, requires a network with low latency and wide bandwidth, the previous error recovery protocols have a serious network overhead to satisfy this requirement. The suggested error recovery protocol is a light-weight protocol which can shows its best performance at ATM Clustering System and uses a newly suggested adaptive acknowledgement scheme. In the adaptive acknowledgement scheme, the period of acknowledgement is dynamically changed depending on the state of the network. We proved the logical correctness of our protocol with SDT and did performance analysis with DEVSim++. From the analysis, we found the optimal parameter values for best performance and showed that our protocol works better than the previous error recovery protocols.

• Journal of KIISE:Information Networking
• /
• v.35 no.3
• /
• pp.227-234
• /
• 2008

The development of wireless network technology allows high data rate seamless communication irrespective of the place and time in various emerging mobile service environment. Unlike wired networks, however, wireless networks utilize expensive limited bandwidth. MBS(Multicast Broadcast Service), which is supported by mobile WiMAX system based on IEEE802.16e, overcomes this problem using a shared downlink channel for efficiently supporting a number of users. However. the coverage and throughput of the system are significantly affected by the channel condition. In this paper we propose on MBS system employing Mobile Multi-Hop Relay(MMR) and adaptive modulation and coding(AMC) scheme. The result of NS-2 computer simulation shows that the throughput and transmission time are substantially improved by the proposed approach compared to the existing MBS system.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.32 no.5A
• /
• pp.433-440
• /
• 2007

In this paper, we have proposed WiBro receiver structure employing joint adaptive antenna scheme at the base station (BS) and the mobile station (MS) and evaluated its performance. In WiBro system, the joint use of transmit beamforming at BS and receive beamforming at MS can improve the SINR significantly and increase the capacity compared to the conventional system. Moreover, power allocation level can be decreased because channel variation can be reduced by using the jointly updated weight. However, it needsthe calculation of the antenna weights in every subcarrier for performance improvement. This could imply an enormous computational burden. However, the computational complexity can be reduced significantly by using the same set of the antenna weights for the adjacent subcarriers instead of calculating antenna weights for every subcarrier. We have analyzed the impact of subcarrier grouping for weight calculation on the system performance.

• Journal of KIISE
• /
• v.45 no.2
• /
• pp.175-186
• /
• 2018

Recently, HAS (HTTP Adaptive Streaming) has been the subject of much attention to improve the QoE (Quality of Experience). In a high latency network, HAS degrades the QoE due to the lost RTT cycle since it replies with a response of one segment to the request of one segment. The server-push based HAS schemes of downloading multiple segments in one request cause QoE degradation due to the buffer underflow. In this paper, we propose a VSSDS (Video Streaming Scheme based on Dynamic Server-push) scheme to improve the QoE in a high latency network. The proposed scheme adjust video quality by estimating available bandwidth and determine the number of segments to be downloaded for each segment request cycle. Through the simulation, the proposed scheme not only improves the average video bitrate but also alleviates the buffer underflow.

• Journal of the Institute of Electronics Engineers of Korea CI
• /
• v.47 no.1
• /
• pp.51-57
• /
• 2010

In this paper, It propose methods compressing BTC image utilizing data hiding technique. BTC is used to compress general digital image into binary image and applied into application such as printer. Additional information, transferred with binary image, is as big as the size of binary image. Therefore, we wish to reduce the total transmission bandwidth by decreasing the additional information with sustaining the small image degradation. Because typical BTC image doesn't have enough space for data hiding, we adopt Adaptive AMBTC (Absolute Moment BTC) algorithm to produce the binary image, and calculate virtual histogram from created binary image and modify this histogram for reducing the additional information. The proposed algorithm can reduce about 6-11 % of the image file size, compared with the existing BTC algorithm, without making perceptible image degradation.

• Journal of Advanced Marine Engineering and Technology
• /
• v.34 no.8
• /
• pp.1165-1171
• /
• 2010

The OFDM system has better characteristics in transmission rate, power efficiency, bandwidth efficiency, impulse-noise immunity, and narrow band interference immunity etc. in comparison with other conventional systems. However, high PAPR of an OFDM signals causes some serious non-linear processing of RF amplifier. And performance of the communication system gets worse. Therefore, various methods reducing PAPR of an OFDM skills such as the clipping method, block coding method, and phase rotation method etc. have been researched. In this paper, we propose a high-speed adaptive PTS method which eliminates high PAPR. And we compare the proposed method with other conventional methods. The proposed method has decreased quantity of calculation compare with an adaptive PTS method. Of course, The more its calculation amount is decreased, the more its BER characteristic is not better than an adaptive PTS method. However, keeping up satisfactory BER performance, we highly improved calculation amount of a PTS method.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.17 no.7
• /
• pp.1618-1626
• /
• 2013

Virtualization service processes all operation including the data creation, storing, and disposal in a server and transmits processed data as the streaming media form. Therefore, client can use the same environment as the traditional desktop environment without considering the type of device. Virtualization service should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived Quality of Service(QoS) of streaming service. In this paper, we propose a network-adaptive transmission architecture that focuses on guaranteeing QoS requirements for virtualization service. In order to provide those, the proposed architecture have the transmission rate adaptation function based on available bandwidth and the content bit-rate control function based on sender buffer state. Through each function, proposed architecture guarantee the delay bounds and continuity of virtualization contents playback. The simulation results show that proposed network-adaptive transmission architecture provides a improve performance of throughput and transmission delay.

• The Journal of the Acoustical Society of Korea
• /
• v.15 no.1
• /
• pp.46-53
• /
• 1996

Adaptive LMS algorithm has been used in many application areas due to its low complexity. In this paper input signal is transformed into the subbands with arbitrary bandwidth. In each subbands the dynamic range can be reduced, so that the independent filtering in each subbands has faster convergence rate than the full band system. The DCT transform domain LMS adaptive filtering has the whitening effect of input signal at each bands. This leads the convergence rate to very high speed owing to the decrease of eigen value spread Finally, the filtered signals in each subbands are synthesized for the output signal to have full frequency components. In this procedure wavelet filter bank guarantees the perfect reconstruction of signal without any interspectra interference. In simulation for the case of speech signal added additive white gaussian noise, the suggested algorithm shows better performance than that of conventional NLMS algorithm at high SNR.

• Journal of KIISE:Information Networking
• /
• v.30 no.2
• /
• pp.155-166
• /
• 2003

Mobile computing becomes a new issue of researches in computing due to the advances of mobile equipment and the connection with Internet. In mobile environment, there are many constraints such as limited bandwidth, intermittent disconnection, limited battery life, and so on. By these reasons, broadcasting has been generally used to disseminate data efficiently by the mobile applications. This paper proposes an adaptive broadcasting method which logically divides broadcast channel into the periodic broadcast channel and the on-demand broadcast channel and dynamically assigns the bandwidths of both channel. The former disseminates data that are selected based on both the popularity and the size of each datum, the latter disseminates data that are selected based on the requests of mobile clients. When selecting data to be disseminated, the proposed broadcasting method considers the mobility of a mobile client and also considers the size of each datum by using SF(size factor) proposed in this paper. This paper also evaluates the proposed broadcasting method by measuring the energy expenditure of mobile client in experiments.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.41 no.7
• /
• pp.9-19
• /
• 2004

WLAN (Wireless Local Area Networks) needs error recovery and flow control schemes to support reliable multicast protocol. Limited wireless bandwidth, as well as queuing losses caused by the asymmetric wired/wireless interactions, demands more effective approaches for reducing packet losses. Moreover, since the wireless channel is a shared broadcast medium, if sender receives feedback information simultaneously from several receivers, the feedback delays data frame transmission of forward direction by introducing channel congestion and burden at the sender. Therefore, it is important to minimize the amount of feedback information from receivers. In this paper, we propose an ARS(Adaptive Redundancy Scheme) that combines FEC(Forward Error Correction) using channel state estimation and ARQ(Automatic Repeat Request) both to reduce the amount of feedback information and the number of retransmissions and to guarantee high data reliability in a WLAN multicast environment. Performance of the proposed scheme is evaluated by means of analysis and simulations in AWGN and Rayleigh fading channels. The results show that the proposed scheme reduces the amount of feedback information and the number of retransmissions and guarantees high data reliability, while keeping throughput efficiency similarly with the conventional FEC and ARQ scheme.