• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2015.07a
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• pp.19-22
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• 2015

현 3D 실험 방송은 좌영상과 우영상의 듀얼스트림을 전송하는 서비스 호환 방식으로 제공되고 있다. 본 논문에서는 3D 실험방송에서 발생하는 제한된 대역폭에 따른 화질의 저하를 해결하는 방법을 제안하였다. MPEG-2 TS 와 MMT 의 GFD Mode 를 사용하여 상기 두 영상을 지상파와 IP 망으로 전송 받아 화질의 열화없이 안정적으로 제공할 수 있는 Hybrid Contents Player 를 개발 하였다. 제안된 시스템은 MMTP Packets 을 우 영상, TS file 을 좌 영상으로 하여 두 영상의 동기화를 통해 고화질의 스테레오스코픽 영상을 구성한다. 이를 위해 본문에서는 MPEG-2 TS 와 MMT 의 GFD mode 의 파일에 대해 분석하고, 동기화 방법에 대해 설명하였으며 실험을 통해 사용자가 3D 시청을 원할 때 스테레오스코픽 영상이 재생되는 Hybrid Contents Player 를 검증하였다.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.10 no.3
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• pp.29-36
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• 2000

It is very import role to increase communication quality that fast detection of random sequence synchronization fail in synchronous stream cipher system using initial synchronization mode. Generally it sends additional information to detect random sequency synchronization fail. But we can't transmit additional informations to decide synchronization fail in a system using RMVD to correct channel error. In this paper we propose a method to detect synchronization fail in the receiver even though a system using RMVD has no margin to send additional information, For detecting random sequency synchronization fail we decipher receiver data analyze probability of transition rate for pre-determined period and decide synchronization fail using calculated transition rate probability. This proposed method is fast very reliable and robust in noisy channel and is easily implemented with hardware.

• Journal of Broadcast Engineering
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• v.22 no.6
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• pp.744-754
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• 2017

This paper compares transport overhead of MPEG-2 TS, MMT and ROUTE in order to compare transport efficiency between the DTV and UHDTV. The MPEG-2 TS standard, widely used, was established for multiplexing and synchronizing encoded audio and video, additional information. In recent years, MMT and ROUTE was established as a next generation multimedia transport standard for the new broadcasting communication environment. In this paper, we compare and analyze transport overhead about three protocol. In order to analysis, we captured the UHD A/V stream in real-time broadcasting service using ROUTE and MMT, and we calculated and analyzed transport overhead using the overhead analysis program which was developed in our laboratory. Furthermore, for comparison under the same conditions, we assumed the MPEG-2 TS stream by extracting ES of UHD A/V stream based on the DTV standard. In this paper, we show the results of protocol transport efficiency in case of basic A/V stream except for additional services. And result show that MMT and ROUTE have similar overhead and MPEG-2 TS is relatively small overhead. However, since MPEG-2 TS result does not consider null packets, it is expected that the relative overhead difference will be reduced.

• The KIPS Transactions:PartC
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• v.11C no.6 s.95
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• pp.747-754
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• 2004

Today, wireless LANs are widely deployed in various places such as corporate office conference rooms, industrial warehouses, Internet-ready classrooms, etc. However, new concerns have been raised regarding suity. Currently, both virtual private network(VPN) and WEP are used together as a strong authentication mechanism. While security is increased by using VPN and WEP together, unnecessary redundancy occurs causing power consumption increase and authentication speed decrease in the authentication process. In this paper a new synchronization protocol for authentication is proposed which allows simple authentication, minimal power consumption at the mobile station, and high utilization of authentication stream. This is achieved by using one bit per a frame authentication, while main authentication process including synchronization is handled by access points. Computer simulation reveals that the proposed scheme significantly improves the authentication efficiency in terms of the number of authenticated frames and authentication speed compared with an earlier protocol employing a similar authentication approach.

• Journal of the Korean Institute of Intelligent Systems
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• v.19 no.4
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• pp.485-492
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• 2009

In the mobile service environment, patterns generated by temporal synchronization are streamed with different instance values. This paper proposed a variable clustering management method, which manages multiple data streaming dynamically, to support flexible clustering. The method manages synchronization effectively and differently with conventional streaming methods in data streaming environment and manages clustering streaming after the structural presentation level and the fitness presentation level. In the structural presentation level, the stream structure is presented using level matching and accumulation matching, and clustering management is carried out by the management of dynamic segment and static segment. The performance of the proposed method is tested by using k-means method, C/S server method, CDN method, and simulation. The test results showed that the proposed method has better performance than the other methods.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2010.11a
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• pp.8-9
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• 2010

ATSC-M/H에서 main stream(좌영상)인 TS-M 패킷은 MPEG-2 전송 시스템을 기반이고, M/H stream(우영상)인 M/H 패킷은 RTP 기반의 전송방식을 사용한다. 이와 같이 dual-stream 시스템을 통해 좌우영상이 전송되는 경우에는 기존에 연구되었던 동종 스트림간의 동기화방안을 적용할 수 없다. 본 논문에서는 3차원 방송 서비스를 하기 위해 ATSC-M/H를 통해 전송되는 좌우영상간의 동기화 방안을 제안한다.

• Journal of the Korea Computer Industry Society
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• v.3 no.9
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• pp.1251-1260
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• 2002

Real-time application programs have synchronization constraints which need to be met between media-data. Synchronization method represents feedback method including virtual client-side buffer. This buffer is used in buffer level method. It is client-leading synchronization that is absorbing variable transmission delay time and that is synchronizing by feedback control. It is the important factor for playback rate and QoS if the buffer level is normal or not. To solve the problems, we can control the start of transmission in multimedia server by appling filtering, control and network evaluation function. Synchronization method is processing for smooth presentation without cut-off while media is playing out. When audio frame which is master media is in high threshold buffer level we decrease play out time gradually, otherwise we increase it slowly.

• The Transactions of the Korea Information Processing Society
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• v.6 no.8
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• pp.2042-2049
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• 1999

In contrast to traditional data communication, the multimedia communication must support variable requirements to multimedia application service. Specially, it is a critical issue that the synchronization supports a continuous and temporal relationship media playout in a distributed environment. But the continuously generated media in source can't guarantee the constant interval on playout for the destination due to the inconsistent network transmission delay and transmission error. In this paper, we introduce a intramedia synchronization mechanism for the continuous and synchronous playout of multimedia stream such as audio and video across a communication network. The proposed mechanism is based on the user defined QoS(Quality of Service) and the playout correction algorithm to guarantee the continuous media playout at the receiver.

• Journal of Korea Multimedia Society
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• v.8 no.10
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• pp.1403-1410
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• 2005

The synchronization problem is directly related to the quality of service in multimedia communication and especially in real-time communication. In this study, we found the cause of clock fluctuation between encoder and decoder in MPEG-2 system layer was that the standard decoder design only considered a fixed time delay component. To solve it, we proposed Extended-SRTS algorithm, which uses STC as service clock by synchronizing transport stream. As the result, we can improve the effect of frequency-drift, time-varying-network-jitter and packing-jitter and so on And by virtue of this algorithm, we can make low the dependency of network clock, which makes easy to synchronize and connect transparently at the ends point, we expect the proposed algorithm can be widely applied to the field of real -time multimedia communications.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.8C
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• pp.1210-1217
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• 2004

LILI-II stream cipher is an upgraded version of the LILI-128, one of candidates in NESSIE. Since the algorithm is a clock-controlled, the speed of the keystream data is degraded structurally in a clock-synchronized hardware logic design. Accordingly, this paper proposes a 4-bit parallel LFSR, where each register bit includes four variable data routines for feedback or shifting within the LFSR. furthermore, the timing of the proposed design is simulated using a Max+plus II from the ALTERA Co., the logic circuit is implemented for an FPGA device (EPF10K20RC240-3), and apply to the Lucent ASIC device (LV160C, 0.13${\mu}{\textrm}{m}$ CMOS ＆ 1.5v technology), and it could achieve a throughput of about 500 Mbps with a 0.13${\mu}{\textrm}{m}$ semiconductor for the maximum path delay below 1.8㎱. Finally, we propose the m-parallel implementation of LILI-II, throughput with 4, 8 or 16 Gbps (m=8, 16 or 32).