• Title/Summary/Keyword: frame delivery rate

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Analysis of Several Digital Network Technologies for Hard Real-time Communications in Nuclear Plant

  • Song, Ki-Sang;No, Hee-Cheon;Kim, Dong-Hun;Koo, In-Soo
    • Nuclear Engineering and Technology
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    • v.31 no.2
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    • pp.226-235
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    • 1999
  • Applying digital network technology for advanced nuclear plant requires deterministic communication for tight safety requirements, timely and reliable data delivery for operation-critical and mission-critical characteristics of nuclear plant. Communication protocols, such as IEEE 802/4 Token Bus, IEEE 802/5 Token Ring, FDDI, and ARCnet, which have deterministic communication capability are partially applied to several nuclear power plants. Although digital communication technologies have many advantages, it is necessary to consider the noise immunity from electromagnetic interference (EMI), electrical interference, impulse noise, and heat noise before selecting specific digital network technology for nuclear plant. In this paper, we consider the token frame loss and data frame loss rate due to the link error event, frame size, and link data rate in different protocols, and evaluate the possibility of failure to meet the hard real-time requirement in nuclear plant.

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Modeling and Simulation Framework for Assessing Interference in Multi-Hop Wireless Ad Hoc Networks

  • Woo, Shin-Uk;Hong, Jin-Pyo;Kim, Hwang-Nam
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.3 no.1
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    • pp.26-51
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    • 2009
  • In this paper, we propose an empirical framework for modeling and emulating interference in multi-hop wireless ad-hoc networks. Wireless interference causes wide variation in the frame delivery rate at a link, and thus we cannot represent the state of the link with only two states, connected state and disconnected state, as in wired networks. We first investigate wireless interference in detail, in order to accurately calibrate the interference and identify its underlying attributes, and then we simulate the diverse occurrences and effects of interference, after incorporating the scheme into a simulation tool. Based on these observations, we devise a modeling and simulation framework with several control parameters, and perform an extensive set of simulation studies. The simulation results indicate that the proposed framework enables us to examine various attributes of wireless interferences and their effects on wireless network protocols and systems.

ServerNet and ATM Interconnects: Comparison for Compressed Video Transmission

  • Ashfaq Hossain;Kang, Sung-Mo;Robert Horst
    • Journal of Communications and Networks
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    • v.1 no.2
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    • pp.134-142
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    • 1999
  • We have developed fully functional Video Server and Client applications which can transmit, receive, decompress and display compressed video over various networks. Our video trans-port allows dynamic rate control feedback, loss detection, and repair requests from Clients to the Server. Our experiments show how feedvack-before-degradation scheme for rate adaptation maintains good display frame-rate for video playback. We show how the playback degradation(reduction in display frame-rate) oc-curs and what happens if corrective measures are not taken to im-prove the situation. The degradation is attributed to the increased internal kernel buffering which consumes scarce CPU resources. We demontrate with our experimental results that ServerNet, with improved hardware delivery guarantees, can significantly reduce host CPU resource consumption while serving video streams. We present the maximum number of streams which can be served for each of ATM and ServerNet interconnects. The appropriate user-level packet size for the video server are also determined for each case.

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AFSO: An Adaptative Frame Size Optimization Mechanism for 802.11 Networks

  • Ge, Xiaohu;Wang, Cheng-Xiang;Yang, Yang;Shu, Lei;Liu, Chuang;Xiang, Lin
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.4 no.3
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    • pp.205-223
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    • 2010
  • In this paper, we analyze the impact of different frame types on self-similarity and burstiness characteristics of the aggregated frame traffic from a real 802.11 wireless local area network. We find that characteristics of aggregated frame traffic are affected by both mean frame size and the proportion of specified frame types. Based on this new knowledge, an adaptative frame size optimization (AFSO) mechanism is proposed to improve the transmission efficiency by adaptively adjusting data frame size according to the proportions of different frame types. Simulation results show that our proposed mechanism can effectively regulate the burstiness of aggregated frame traffic and improve the successful delivery rate of data frames when a fixed throughput target is set for 802.11 wireless networks.

UHD 4K end to end broadcast solution over DVB-T2 SFN network using HEVC real time encoding

  • Dimitrakopulos, Nik
    • Broadcasting and Media Magazine
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    • v.19 no.2
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    • pp.36-45
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    • 2014
  • With major sports events such as FIFA World Cup and The Olympic Games coming up, UHDTV technology has been given a major push by broadcasters, test and measurement (T&M), and consumer electronics (TV and set-top box) manufacturers. The increase in picture resolution ($3840{\times}2160$) for 4K and ($7680{\times}4320$) for 8K compared to current HDTV ($1920{\times}1080$), as well as the need to deliver these services in higher frame rates (50/60 up to 100/120fps) translate to a big challenge in UHDTV content delivery to households over terrestrial transmission due to higher data rates. DVB-T2 has been favored by many countries across the world as it is proven to give the best spectral efficiency for terrestrial broadcasting. On the other hand MPEG-4 appears to be the bottleneck for UHDTV since a much higher data rate is needed to deliver such services. Therefore HEVC becomes mandatory for UHDTV delivery over DVB-T2. In this paper, we will give an overview about a complete end-to-end solution for UHDTV delivery over DVB-T2 based on a realistic SFN scenario in Seoul metropolitan area.

Maritime English vs Maritime English Communication

  • Choe, Seung-Hui
    • Proceedings of the Korean Institute of Navigation and Port Research Conference
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    • 2015.07a
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    • pp.272-274
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    • 2015
  • Success of communication at sea is directly linked with clear and complete delivery and receipt of the target message between interlocutors. It can be said that speakers' effective delivery of their intended message and listeners' precise decoding and accurate understanding are the keys to successful maritime communication. From this perspective, the scope of maritime English education and training needs to be reconceptualized and expanded into the area of communication itself, beyond the simple acquisition of, and familiarization with, IMO Standard Maritime Communication Phrases (SMCP). Therefore, in order to make learners' acquisition of marine communication knowledge more feasible, and the knowledge learned more practically applicable, training on effective and clear oral delivery should be also considered within the frame of maritime English education. Thus, critical training elements to realize this goal need to be suggested as guidelines. In this presentation, the theoretical background on this will be introduced in terms of English as a Lingua Franca (ELF) and Lingua Franca Core (LFC), which are the current mainstream forms of English communication in the international business context. Based on this, six key training elements will be discussed; that is, speech rate, word groups, pauses, nuclear stresses, consonants (including consonant clusters), and vowels (specifically long and short vowels). Finally, the practical pedagogical methods of each element, and its actual application into a real ESP classroom, will be suggested.

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Dependent Quantization for Scalable Video Coding

  • Pranantha, Danu;Kim, Mun-Churl;Hahm, Sang-Jin;Lee, Keun-Sik;Park, Keun-Soo
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2006.11a
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    • pp.127-132
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    • 2006
  • Quantization in video coding plays an important role in controlling the bit-rate of compressed video bit-streams. It has been used as an important control means to adjust the amount of bit-streams to at]owed bandwidth of delivery networks and storage. Due to the dependent nature of video coding, dependent quantization has been proposed and applied for MPEG-2 video coding to better maintain the quality of reconstructed frame for given constraints of target bit-rate. Since Scalable Video Coding (SVC) being currently standardized exhibits highly dependent coding nature not only between frames but also lower and higher scalability layers where the dependent quantization can be effectively applied, in this paper, we propose a dependent quantization scheme for SVC and compare its performance in visual qualities and bit-rates with the current JSVM reference software for SVC. The proposed technique exploits the frame dependences within each GOP of SVC scalability layers to formulate dependent quantization. We utilize Lagrange optimization, which is widely accepted in R-D (rate-distortion) based optimization, and construct trellis graph to find the optimal cost path in the trellis by minimizing the R-D cost. The optimal cost path in the trellis graph is the optimal set of quantization parameters (QP) for frames within a GOP. In order to reduce the complexity, we employ pruning procedure using monotonicity property in the trellis optimization and cut the frame dependency into one GOP to decrease dependency depth. The optimal Lagrange multiplier that is used for SVC is equal to H.264/AVC which is also used in the mode prediction of the JSVM reference software. The experimental result shows that the dependent quantization outperforms the current JSVM reference software encoder which actually takes a linear increasing QP in temporal scalability layers. The superiority of the dependent quantization is achieved up to 1.25 dB increment in PSNR values and 20% bits saving for the enhancement layer of SVC.

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A User Driven Adaptable Bandwidth Video System for Remote Medical Diagnosis System (원격 의료 진단 시스템을 위한 사용자 기반 적응 대역폭 비디오 시스템)

  • Chung, Yeongjee;Wright, Dustin;Ozturk, Yusuf
    • Journal of Information Technology Services
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    • v.14 no.1
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    • pp.99-113
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    • 2015
  • Adaptive bitrate (ABR) streaming technology has become an important and prevalent feature in many multimedia delivery systems, with content providers such as Netflix and Amazon using ABR streaming to increase bandwidth efficiency and provide the maximum user experience when channel conditions are not ideal. Where such systems could see improvement is in the delivery of live video with a closed loop cognitive control of video encoding. In this paper, we present streaming camera system which provides spatially and temporally adaptive video streams, learning the user's preferences in order to make intelligent scaling decisions. The system employs a hardware based H.264/AVC encoder for video compression. The encoding parameters can be configured by the user or by the cognitive system on behalf of the user when the bandwidth changes. A cognitive video client developed in this study learns the user's preferences (i.e. video size over frame rate) over time and intelligently adapts encoding parameters when the channel conditions change. It has been demonstrated that the cognitive decision system developed has the ability to control video bandwidth by altering the spatial and temporal resolution, as well as the ability to make scaling decisions

An Adaptive Multimedia Synchronization Scheme for Media Stream Delivery in Multimedia Communication (멀티미디어 통신에서 미디어스트림 전송을 위한 적응형 멀티미디어 동기화 기법)

  • Lee, Gi-Sung
    • The KIPS Transactions:PartC
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    • v.9C no.6
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    • pp.953-960
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    • 2002
  • Rel-time application programs have constraints which need to be met between media-data. It is client-leading synchronization that is absorbing variable transmission delay time and that is synchronizing by feedback control and palyout control. It is the important factor for playback rate and QoS if the buffer level is normal or not. This paper, The method of maintenance buffer normal state transmits in multimedia server by appling feedback of filtering function. And synchronization method is processing adaptive playout time for smooth presentation without cut-off while media frame is skip. When audio frame which is master media is in upper threshold buffer level we decrease play out time gradually, low threshold buffer level increase it slowly.

Frame/Slot Allocation Method for Real-time Traffic in IEEE 802.15.4 Wireless Sensor Networks (IEEE 802.15.4 무선센서네트워크의 실시간 트래픽 처리를 위한 프레임/슬롯 할당방법)

  • Kim, Hyung-Seok
    • Journal of the Institute of Electronics Engineers of Korea CI
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    • v.44 no.3
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    • pp.49-56
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    • 2007
  • The IEEE 802.15.4 low rate wireless personal area network is a protocol suitable for wireless sensor networks. Thus there has been growing interest in the IEEE 802.15.4 protocol in the home automation networks. This paper proposes a scheduling scheme obtaining optimal parameters regarding the IEEE 802.15.4 frame and subframes in the home automation networks. Guaranteed time slots (GTS) are exploited for the delivery of real-time traffic. Given a set of the requirements for bandwidths and periods assigned to nodes in the network, the beacon interval and the active/inactive subframe duration satisfying the requirements are selected considering the low duty cycle by the proposed scheme. Based on these parameters, analytic results provide the way to efficient use of network resources including energy.