• Journal of Broadcast Engineering
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• v.20 no.1
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• pp.16-25
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• 2015

Even if the demands for ultra-high-quality multimedia contents are increasing, it is difficult to produce, encode, play and transport ultra-high-quality contents under the existing broadcasting environment. By the reason, various technologies for the UHD contents have been developed in order to satisfy the user's needs. In this paper, we propose a design methodology of a broadcasting system, which consists of two parts, for UHD services with two parts. At the transmit part of the proposed system can encode a video into several layered-bitstreams hierarchically, and then transport each bitstream over heterogeneous networks. The receiver part can play the received video by composing the separated bitstreams. The proposed system can adaptively provide both HD and UHD contents according to user's reception conditions by using the heterogeneous networks.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.652-663
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• 2002

Recently, many advanced video application services over the mobile wireless networks have required a transcoder which can efficiently reduce the size of compressed video bitstreams. The transcoder can be worked in either the spatial domain or the DCT domain. In this paper, we propose a new fast hybrid-type transcoder which can efficiently reduce the frame size with keeping the visual quality. The proposed scheme consists of two major processes: a transform domain process and a spatial domain process. We also propose a scheme for coding mode selection and motion vector refinement. Experimental results show that our approach can reduce the computational complexity more than any other conventional spatial-domain transcoder with keeping the visual quality.

• Journal of Korea Multimedia Society
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• v.13 no.4
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• pp.575-581
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• 2010

With widespread use of the Internet and improvements in streaming media and compression technology, digital music, video, and image can be distributed instantaneously across the Internet to end-users. However, most conventional Digital Right Management are often not secure and not fast enough to process the vast amount of data generated by the multimedia applications to meet the real-time constraints. The SVC offers temporal, spatial, and SNR scalability to varying network bandwidth and different application needs. Meanwhile, for many multimedia services, security is an important component to restrict unauthorized content access and distribution. This suggests the need for new cryptography system implementations that can operate at SVC. In this paper, we propose a new scrambling encryption for reserving the characteristic of scalability in MPEG4-SVC. In the base layer, the proposed algorithm is applied and performed the selective scambling. And it encrypts various MVS and intra-mode scrambling in the enhancement layer. In the decryption, it decrypts each encrypted layers by using another encrypted keys. Throughout the experimental results, the proposed algorithms have low complexity in encryption and the robustness of communication errors.

• Journal of KIISE:Databases
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• v.37 no.5
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• pp.221-227
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• 2010

Web access sequence mining can discover the frequently accessed web pages pursued by users. Utility-based web access sequence mining handles non-binary occurrences of web pages and extracts more useful knowledge from web logs. However, the existing utility-based web access sequence mining approach considers web access sequences from the very beginning of web logs and therefore it is not suitable for mining data streams where the volume of data is huge and unbounded. At the same time, it cannot find the recent change of knowledge in data streams adaptively. The existing approach has many other limitations such as considering only forward references of web access sequences, suffers in the level-wise candidate generation-and-test methodology, needs several database scans, etc. In this paper, we propose a new approach for high utility web access sequence mining over data streams with a sliding window method. Our approach can not only handle large-scale data but also efficiently discover the recently generated information from data streams. Moreover, it can solve the other limitations of the existing algorithm over data streams. Extensive performance analyses show that our approach is very efficient and outperforms the existing algorithm.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.704-711
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• 2003

Recently, the demands for real-time service and multimedia data are rapidly increasing. There are significant redundancies between header fields both within the same packet header and in consecutive packets belonging to the same packet stream. But there are many overheads in using the current UDP/IP protocol. Header compression is considered to enhance the transmission efficiency for small size of payload. By sending the static field information only once initially and by utilizing dependencies and predictability for other fields, the header size can be significantly reduced for most packets. This work describes an implementation for header compression of the headers of U/UDP protocols to reduce overhead on Ethernet network. Typical UDP/IP Header packets can be compressed down to 7 bytes and the header compression system is designed and implemented on the Linux environment. Using the designed Header compression system between a server and a client have the advantage of effective data throughput in network.

• The KIPS Transactions:PartC
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• v.15C no.6
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• pp.547-556
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• 2008

Recently, with the advance of computing and networking technique, the high speed internet becomes widespread, however, it is still hard job to do streaming the media which requires high network bandwidth over the internet. Previous researches for streaming over the internet mainly proposed techniques that controls the QoS(Quality of Service) of the media in proportion to the network status. Though, this could be the solution for the service provider while the service user who wants constant QoS may not satisfy with variable QoS. In the paper, we propose a network adaptive prefetching technique, NAP, for guarantee of constant QoS. The NAP prefetches frames by increasing the frame transmission rate while the available network bandwidth is high. The NAP uses the prefetched frames to guarantee the QoS while the available network bandwidth is low. The experiment result shows that the proposed NAP could guarantee the constant QoS by prefetching the frames adaptively to the network bandwidth with the characteristic of video stream.

• The Journal of the Korea Contents Association
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• v.8 no.6
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• pp.43-53
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• 2008

Hyper text transfer protocol (HTTP) over transmission control protocol (TCP) is currently used to transfer objects in the Internet. Stream control transmission protocol (SCTP), an alternative to TCP, which allows for independent delivery among streams, and can thus reduce the mean response delay of web object. We present an analytical model to find the mean response delay for HTTP over SCTP, therefore, estimate the effectiveness of SCTP over TCP. Typical TCP delay models assume the wired environment. On the contrary, the proposed model in this paper assumes the multiple packet losses and wireless environment where fast retransmission is not possible due to small window. The estimated mean response time can be used the benchmark to meet quality of service (QoS) at end-user. We validate the accuracy of our model using experiments. It is shown that the differences between the results from model and those from experimental are very small below 6 % on average. We also find that the mean response delay for HTTP over SCTP is less than that for HTTP over TCP.

• Journal of Korea Multimedia Society
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• v.4 no.4
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• pp.297-305
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• 2001

Recently advances in high-speed networks and multimedia computer technologies allow new types of multimedia applications to manipulate large volumes of multimedia data. However, in the real time and/or the heterogeneous data transmissions, there are many difficulties such as network transmission delay, the implementation difficulties, and so on. To solve these problems, in this paper, we extend the method of the multimedia service design which is proposed by OMG. To do this, we suggest an efficient real time audio/video stream framework, called Smart Explorer, based un CORBA and JMF Java Media API. And we separate the transmission path of control data from that of media data and use RTP/RTCP protocol for efficient real time audio/video transmission. Also we show the appropriate implementation of the audio/video stream system based on our suggested framework Smart Explorer. In the future, we expect our audio/video stream system to be applied to the real time communication software such as broadcasting, distance learning, and video conferencing.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.40 no.9
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• pp.702-713
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• 2003

The importance of protection for data and information is increasing by the rapid development of information communication and network. And the concern for protecting private information is also growing due to the increasing demand for lots of services by users. Asymmetric cryptosystem is the mainstream in encryption system rather than symmetric cryptosystem by above reasons. But asymmetric cryptosystem is restricted in applying fields by the reason it takes more times to process than symmetric cryptosystem. In this paper, encryption system which executes authentication works of asymmetric cryptosystem by means of symmetric cryptosystem. The proposed cryptosystem uses an algorithms combines that combines block cipherment with stream cipherment and has a high stability in aspect of secret rate by means of transition of key sequence according to the information of plaintext while symmetric/asymmetric cryptosystem conducts encipherment/deciphermeent using a fixed key. Consequently, it is very difficult to crack although unauthenticator acquires the key information. So, the proposed encryption system which has a certification function of asymmetric cryptosystem and a processing time equivalent to symmetric cryptosystems will be highly useful to authorize data or exchange important information.

• The Journal of Korean Association of Computer Education
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• v.11 no.2
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• pp.107-117
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• 2008

The agent system that can adjust the workload dynamically through thc periodical monitoring of the VoD system comprises the agency part interfacing the VoD system and the intelligence part reasoning or learning the facts required for the adjustment of workload. This paper proposes a learning method that can apply to the intelligence part of the agent system. The proposed method can adjust the workload more efficiently by the rule's reasoning process and case-based learning process. An experiment of implementing a simulator was conducted to see whether or not application of the proposed method to VoD systems is efficient. As a result of the experiment, it was found that the throughput and the average waiting time of the VoD server were relatively improved when the proposed method was applied compared to existing means.