• Journal of KIISE:Computer Systems and Theory
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• v.35 no.9_10
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• pp.466-475
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• 2008

Owing to the improved wireless communication technologies, it is possible to provide streaming service of multimedia with PDAs and mobile phones in addition to desktop PCs. Since mobile client devices have low computing power and low network bandwidth due to wireless network, the transcoding technology to adapt media for mobile client devices considering their characteristics is necessary. Transcoding servers transcode the source media to the target media within corresponding grades and provide QoS in real-time. In particular, an effective load balancing policy for transcoding servers is inevitable to support QoS for large scale mobile users. In this paper, the transcoding load estimation algorithm is proposed for load balance on the distributed transcoding environments. The proposed algorithm estimates transcoding time from transcoding server information, movie information and target transcoding bit-rate. The estimated transcoding time is proved based on experiments.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.4 s.334
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• pp.9-18
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• 2005

This paper proposes the novel synchronous segmented bus architecture that has the pipeline bus architecture based on OCP(open core protocol) and the memory-oriented bus for MPEG system. The proposed architecture has bus architectures that support the memory interface for image data processing of MPEG system. Also it has the segmented hi-directional multiple bus architecture for multitasking processing by using multi -masters/multi - slave. In the scheme address of masters and slaves are fixed so that they are arranged for the location of IP cores according to operational characteristics of the system for efficient data processing. Also the bus architecture adopts synchronous segmented bus architecture for reuse of IP's and architecture or developed chips. This feature is suitable to the high performance and low power multimedia SoC systum by inherent characteristics of multitasking operation and segmented bus. Proposed bus architecture can have up to 3.7 times improvement in the effective bandwidth md up to 4 times reduction in the communication latency.

• Journal of Broadcast Engineering
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• v.11 no.1 s.30
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• pp.107-115
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• 2006

In the environment of single application data broadcasting, only one application can be serviced at a specific time on one channel. To overcome this, we developed the structure and the method of operation for multi-applications which are fully conformant to the ACAP (Advanced Common Application Platform), and modified data broadcasting system to support multi-applications. In multi-application environment, broadcasters can service multiple applications simultaneously at a specific airtime on one channel so users can enjoy services selectively according to their preferences. In this paper, we present an example of multi-application service which was developed to make an experiment before servicing them to users on the air. The core of the multi-application is a manager application which manages other ordinary applications, so we describe the function and structure of the manager application, and then present the experimental results to show that the proposed method is the proper model for multi-applications.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.8
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• pp.984-998
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• 1999

최근 개인 휴대 통신에 대한 관심도가 증가하면서 B-ISDN (Broadband Integrated Services Digital Network)과 같은 기존의 유선 망에서 제공하던 다양한 멀티미디어 응용 지원을 무선 망으로 확장시키기 위한 연구가 활발히 진행되고 있다. 그러나 기존의 유선 망에서는 멀티미디어 응용 지원을 위해 QoS (Quality of Service) Provisioning에 관한 많은 연구가 되어 있으나 무선 망에서는 이동성과 무선 전파의 열악한 전송으로 인해 새로운 QoS Provisioning 방법에 관한 연구가 필수적이다. 본 논문에서는 이러한 무선 망의 특수성으로 인해 발생할 수 있는 서비스의 질 저하와 강제 종료를 줄임으로써 지속적인 QoS를 보장해 주고 한정된 무선 자원을 효율적으로 사용하며 처리에 의한 오버헤드를 줄이기 위하여 다음과 같은 세 가지 방법을 제안하였다. 첫째, 핸드오프 강제 종료율을 줄이기 위하여 대역폭 예약 방법을 사용하되 특정 셀의 트래픽 특성에 맞게 또한 시간대에 따른 트래픽 특성에 따라 예약 대역폭의 양을 조절하는 적응적 대역폭 예약 방법이다. 둘째, 많은 경우 각 셀의 트랙픽 변화는 일정한 주기로 변화한다는 특성에 따라 과거의 트래픽 정보를 이용하는 예측 기반의 대역폭 예약 방법이다. 마지막으로 호의 종류, 트래픽 특성, 단말기의 이동 속도에 따라 다른 우선 순위에 의해 호 수락 제어를 수행하는 우선 순위 기반의 호 수락 제어를 제안하였다. 시뮬레이션을 통하여 기존에 제안된 방법과 성능 비교하여, 요구되는 수준의 QoS 보장과 효율적인 자원의 사용, 요구되는 처리비용의 최소화를 통해 전체 시스템의 성능 향상을 입증하였다.Abstract As interest in wireless hand-held terminals and in personal communications services increases recently, there have been broad studies on the ways to support multimedia applications provided in wired networks such as B-ISDN (Broadband Integrated Services Digital Network) in wireless networks. However, since many studies have focused on Quality of Service (QoS) Provisioning in wired networks to provide multimedia applications, new methods of QoS Provisioning are needed in wireless networks to resolve the problem of wireless channel fading and the difficulty of mobility occurred in wireless networks. This paper proposes three schemes of QoS Provisioning in wireless networks which will make continuous QoS guarantee and efficient use of limited wireless resources possible. The first scheme reserves bandwidth in proportion to the amount of real-time traffic in the neighbor cells to decrease the handoff dropping rate of delay sensitive real-time connections, adapting reserved bandwidth for efficient resource utilization. The second scheme is predictive bandwidth reservation scheme that utilizes the past handoff information. It can decrease overheads required to adapt bandwidth reservation. The last scheme is priority-based call admission control prioritizing traffic type (real-time traffic/ non-real-time traffic), connection type (new connection /handoff connection), and mobile terminal speed (fast mobile/slow mobile). Simulation results show that the proposed QoS Provisioning schemes improve the total system performance by achieving three goals - required QoS guarantee, higher bandwidth utilization and less overhead.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.41 no.10
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• pp.95-104
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• 2004

This paper propose a motion estimator architecture to reduce the power consumption of the most-power-consuming motion estimation method when designing multimedia SoC with deep submicron technologies below 0.13${\mu}{\textrm}{m}$. The proposed architecture considers both dynamic and static power consumption so that it is suitable for large leakage process technologies, while conventional architectures consider only dynamic power consumption. Consequently, it is suitable for mobile information terminals such as mobile videophone where efficient power management is essential. It exploits full search method for simple hardware implementation. It also exploits early break-off method to reduce dynamic power consumption. To reduce static power consumption, megablock shutdown method considering power line noise is also employed. To evaluate the proposed architecture when applied multimedia SoC, system-level control flow and low-power control algorithm are developed and the power consumption was calculated based on thor From the simulation results, power consumption was reduced to about 60%. Considering the line width reduction and increased leakage current due to heat dissipation in chip core, the proposed architecture shows steady power reduction while it goes worse in conventional architectures.

• Proceedings of the Korean Information Science Society Conference
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• 2008.06b
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• pp.298-302
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• 2008

주어진 작은 크기의 칩 내부에 많은 기능 (예: 멀티미디어, 음성/영상 등)을 작동시키기 위해서는 고집적(high-integration)의 회로가 구현되게 된다. 이러한 고집적 회로는 작동할 때 상당한 양의 전력 소모를 유발하게 되어 결국 배더리 수명을 단축시키는 상황을 가지게 한다. 더욱 심각한 상황은 고 밀도의 칩 안에서의 많은 전력 소모는 열의 발생을 더욱 가속화 시키게 되며, 결국 칩 작동의 신뢰성(reliability)을 상당히 잃게 만든다. 본 연구에서는 칩의 작동에 따른 열 발생으로 유발되는 칩의 온도 상승을 감지하는 센서회로 구현에 관한 것이다. FPGA 칩은 주 목적의 기능을 수행하는 회로들을 구현함과 동시에 추가적으로 열 감지 센서 회로를 구현할 자원을 FPGA가 제공을 해 주어야 하는데, 주목적의 회로 공간(즉, 자원) 사용으로 인해 열 센서 회로 구현 자원이 충분하지 않을 경우나 여러 지역에 사용 가능한 자원이 소규모로 흩어진 경우 등 센서 구현을 위한 자원 탐색 및 구현 가능성에 대해 점검하는 알고리즘이 필요하다. 본 연구는 이러한 알고리즘을 개발하여 그 효용성을 실험을 통해 보이고 있다. 제안한 알고리즘의 특징은 Branch-and-Bound에 기반을 두고 있으며, 알고리즘의 수행 시간 단축을 위한 효과적인 search tree pruning 기법을 제안하고 있다.

• Journal of KIISE:Computing Practices and Letters
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• v.13 no.3
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• pp.153-164
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• 2007

The Access Grid (AG) provides collaboration environments over the IP multicast networks by enabling efficient exchange of multimedia contents among remote users; however, since lots of current networks are still multicast-disabled, it is not easy to deploy this multicast-based multi-party AG. For this problem, the AG provides multicast bridges as a solution by putting a relay server into the multicast networks. Multicast-disabled clients make UDP connections with this relay server and receive forwarded multicast traffics in unicast UDP packets. This solution is facing several limitations since it requires duplicate forwarding of the same packet for each unicast peer. Thus, in this paper, we propose an alternate solution for the multicast connectivity problem of the AG based on the UMTP (UDP multicast tunneling protocol). By taking advantage of flexibilities of UMTP, the proposed solution is designed to improve the efficiency of network and system utilization, to allow reuse of multicast-based AG applications without modification, and to partially address the NAT/firewall traversal issues. To verify the feasibility of proposed solution, we have implemented a prototype AG connectivity tool based on the UMTP, named as the AG Connector.

• The Transactions of the Korea Information Processing Society
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• v.5 no.12
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• pp.3047-3062
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• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.11
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• pp.21-26
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• 2005

We propose a fully converged Ethernet and TDM transport system. Developed Ethernet and TDM convergence system can support not only L2 VPN service and premium multimedia service based on MPLS protocol but also TDM leased line service, simultaneously. Developed convergence system can provide high reliability for Ethernet packet due to support protection and restoration function of circuit based networks. Evaluation for Ethernet and TDM path was successfully performed to show the typical application of the proposed system in the legacy networks.

• Journal of Broadcast Engineering
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• v.24 no.6
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• pp.1076-1084
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• 2019

In this paper, we introduced a system that extracts metadata by recognizing characters and objects in media using deep learning technology. In the field of broadcasting, multimedia contents such as video, audio, image, and text have been converted to digital contents for a long time, but the unconverted resources still remain vast. Building media archives requires a lot of manual work, which is time consuming and costly. Therefore, by implementing a deep learning-based metadata generation system, it is possible to save time and cost in constructing media archives. The whole system consists of four elements: training data generation module, object recognition module, character recognition module, and API server. The deep learning network module and the face recognition module are implemented to recognize characters and objects from the media and describe them as metadata. The training data generation module was designed separately to facilitate the construction of data for training neural network, and the functions of face recognition and object recognition were configured as an API server. We trained the two neural-networks using 1500 persons and 80 kinds of object data and confirmed that the accuracy is 98% in the character test data and 42% in the object data.