• 한국정보과학회논문지:정보통신
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• 제28권4호
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• pp.508-516
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• 2001

본 논문에서는 VOD(Video On Demand)나 실시간 방송 데이타 전송과 같은 비상호적 멀티미디어 응용을 위한 멀티캐스트 라우팅 알고리즘을 제안한다. 비상호적 멀티미디어 응용에서는 송신자가 다수로 존재할 수 있다. 송신자가 다수로 존재할 때 적용할 수 있는 멀티캐스트 라우팅 알고리즘을 설계하고 기존의 단 하나의 송신자가 존재할 때와 비교하여 지연 시간 및 비용 측면에서 현저한 성능향상을 가져올 수 있음을 실험적으로 증명한다. 제안된 알고리즘의 성능 검증을 위해 광역 ATM(Asynchronous Transfer Mode)네트워크를 모델링하고 모의 실험을 수행하였다. 모의 실험 결과 다양한 네트워크 환경하에서 다수의 송신자 존재시 하나의 송신자만 존재할 때와 비교하여 현저한 성능향상을 보였다.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제11권3호
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• pp.1441-1461
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• 2017

Nowadays, Dynamic Adaptive Streaming over HTTP (DASH) has become very popular in streaming multimedia contents. In DASH, a client estimates current network bandwidth and then determines an appropriate video quality with bitrate matching the estimated bandwidth. Thus, estimating accurately the available bandwidth is a significant premise in the quality of video streaming, especially when network traffic fluctuates substantially. To cope with this challenge, researchers have presented various filters to estimate network bandwidth adaptively. However, experiment results show that current schemes either adapt slowly to network changes or adapt fast but are very sensitive to delay jitter and produce sharply changed estimation. This paper presents a novel bandwidth estimation scheme based on Moving Average Convergence Divergence (MACD). We applied an MACD indicator and its two thresholds to classifying network states into stable state and agile state, based on the network state different filters are applied to estimate network bandwidth. In the paper, we studied the performance of various MACD indicators and the threshold values on bandwidth estimation. Then we used a DASH proxy-based environment to compare the performance of the presented scheme with current well-known schemes. The simulation results illustrate that the MACD-based bandwidth estimation scheme performs superior to existing schemes both in the speed of adaptively to network changes and in stability in bandwidth estimation.

• 한국멀티미디어학회논문지
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• 제21권3호
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• pp.415-422
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• 2018

Advances of telecommunication technology bring various changes in journalism field. Reporters started to gather, edit, and transmit content to main server in media company using hand-held smart media and notebook computer. This paper tried to testify valid bit-rate of visual news content using LTE network and mobile phone. Field news like natural disasters need real-time transmission of video content. But broadcasting company normally use heavy ENG system and transmission satellite trucks. We prepared and experimented different types of visual content that has different bit-rates. Transmission tool was LU-60HD mobile system of LiveU Corporation. Transmission result shows that bit-rate of 2Mbps news content is not suitable for broadcasting and VBR (Variable Bit Rate) transmission has better definition quality than CBR (Constant Bit Rate) method. Three different bit-rate of VBR transmission result shows that 5Mbps clip has better quality than 1Mbps and 3Mbps. The higher bit-rate, the better video quality. But if the content has much movements, that cause delay and abnormal quality of video. So optimizing the balance between stability of signal and quality of bit-rate is crucial factor of real-time broadcasting news gathering business.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2009년도 IWAIT
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• pp.184-188
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• 2009

To interactively share High Definition (HD)-quality visualization over emerging ultra-high-speed network infrastructure, several lossless and low-delay real-time media (i.e., uncompressed HD video and audio) transport systems are being designed and prototyped. However, most of them still rely on expensive hardware components. As an effort to reduce the building cost of system, in this paper, we propose the integration of both transmitter and receiver machines into a single bi-directional transport system. After detailed bottleneck analysis and subsequent refinements of embedded software components, the proposed integration can provide Real-time Transport Protocol (RTP)-based bi-directional transport of uncompressed HD video and audio from a single machine. We also explain how to interface the Gbps-bandwidth display output of uncompressed HD media system to the networked tiled display of 10240 $\times$ 3200 super-high-resolution. Finally, to verify the feasibility of proposed integration, several prototype systems are built and evaluated by operating them in several different experiment scenarios.

• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2007년도 춘계종합학술대회
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• pp.175-178
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• 2007

3GPP에서는 이동망에서 QoS를 관리하기 위하여 트래픽 조절기가 포함된 QoS 구조에 대해서 정의하고 있다. 본 논문에서는 3GPP에서 정의하는 트래픽 조절기에 적용 가능한 영상 전송 알고리즘을 제안한다. 제안하는 알고리즘은 토큰 버킷을 이용하여 가변적인 트래픽을 제어하는 트래픽 조절기에서 영상의 품질 변화를 최소화 한다. 제안하는 알고리즘은 반복적인 최적화 방법을 사용하지 않고 프레임 레이어에서 전송률을 제어하는 방법으로 영상 프레임간의 왜곡의 변화를 최소화한다. 그리고 전처리가 필요하지 않은 슬라이딩 윈도우 기법을 사용하기 때문에 영상을 압축할 때 추가적인 지연을 발생시키지 않는다. 따라서 제안하는 알고리즘은 낮은 계산량을 필요로 하는 실시간 영상 코덱에 적당한 알고리즘이다.

• IEIE Transactions on Smart Processing and Computing
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• 제4권6호
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• pp.422-433
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• 2015

In this paper, we demonstrate inter-layer prediction tools for scalable video coders. The proposed scalable coder is designed to support not only spatial, quality and temporal scalabilities, but also view scalability. In addition, we propose quad-tree inter-layer prediction tools to improve coding efficiency at enhancement layers. The proposed inter-layer prediction tools generate texture prediction signal with exploiting texture, syntaxes, and residual information from a reference layer. Furthermore, the tools can be used with inter and intra prediction blocks within a large coding unit. The proposed framework guarantees the rate distortion performance for a base layer because it does not have any compulsion such as constraint intra prediction. According to experiments, the framework supports the spatial scalable functionality with about 18.6%, 18.5% and 25.2% overhead bits against to the single layer coding. The proposed inter-layer prediction tool in multi-loop decoding design framework enables to achieve coding gains of 14.0%, 5.1%, and 12.1% in BD-Bitrate at the enhancement layer, compared to a single layer HEVC for all-intra, low-delay, and random access cases, respectively. For the single-loop decoding design, the proposed quad-tree inter-layer prediction can achieve 14.0%, 3.7%, and 9.8% bit saving.

• 한국군사과학기술학회지
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• 제20권2호
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• pp.217-224
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• 2017

This study proposes an evaluation method for time delay function(TDF) of Point Detonation(PD) fuse using underwater explosion and water entry phenomena. Until now, nothing but the naked eyes of an observer or video images have been used to determine whether the TDF of PD fuze is operated or not. The observer has verified the performance of TDF by analysing the shape of the plume formed by underwater explosion. However, it is very difficult to evaluate the TDF of PD fuse by these conventional methods. In order to overcome this issue, we propose a method using underwater sound signal emitted from the underwater explosion of high explosive charge. The result shows that the measured sound signal is in accord with the physical phenomena of water entry of warhead as well as underwater explosion. Also, from the hypothesis test of bubble period, difference on underwater sound analysis between dud event and delay one is proved.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제14권3호
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• pp.1377-1399
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• 2020

In search and rescue mission, micro aerial vehicles (MAVs) are typically used to capture image and video from an aerial perspective and transfer the data to the ground station. Because of the power limitation, a cluster of MAVs are required for a large search area, hence an ad-hoc wireless network must be maintained to transfer data more conveniently and fast. However, the unstable link and the intermittent connectivity between the MAVs caused by MAVs' movement may challenge the packet forwarding. This paper proposes a delay tolerant packet forwarding algorithm based on location estimation for MAV networks, called DTN_est algorithm. In the algorithm, ferrying MAVs are used to transmit data between MAVs and the ground station, and the locations of both searching MAVs and ferrying MAVs are estimated to compute the distances between the MAVs and destination. The MAV that is closest to the destination is selected greedy to forward packet. If a MAV cannot find the next hop MAV using the greedy strategy, the packets will be stored and re-forwarded once again in the next time slot. The experiment results show that the proposed DTN_est algorithm outperforms the typical DTN_geo algorithm in terms of packet delivery ratio and average routing hops.

• 한국정보처리학회논문지
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• 제6권2호
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• pp.459-466
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• 1999

I broadband ISDN every packet will show different result if it would be processed according to its usage by the server. That is, normal data won't show big differences if they would be processed at normal speed. But it will improve the quality of service to process some kinds of data - for example real time video or voice type data or some data for a bid to by something through the internet - more fast than the normal type data. solution for this problem was suggested - priority packets. But the analyses of them are under way. Son in this paper a switching system for an output queuing model in a single server was assumed and some packets were given priorities and analysed. And correlation, simulating real life situation, was given too. These packets were analysed through three cases, first packets having no correlation, second packets having only correlation and finally packets having priority three cases, first packets having no correlation, second packets having only correlation and finally packets having priority and correlation. The result showed that correlation doesn't affect the mean delay time and the high priority packets have improved mean delay time regardless of the arrival rate. Those packets were assumed to be fixed-sized like ATM fixed-sized cell and the contention strategy was assumed to be short bus contention method for the output queue, and the mean delay length and the maximum 버퍼 length not to lose any packets were analysed.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제13권1호
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• pp.291-304
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• 2019

This paper proposes a dynamic-tiling-based bandwidth-efficient (DTBE) virtual reality (VR) streaming scheme. We consider 360-degree VR contents with multiple view points such as the front, back, upper, and bottom sides. At a given time, the focus of a client is always bound to a certain view among multiple view points. By utilizing this perspective, under our proposed scheme, tiles with high encoding rates are selectively assigned to the focused view where multiple view points consist of multiple tiles with different encoding rates. The other tiles with low encoding rates are assigned to the remaining view points. Furthermore, for reducing view adaptation delay, we design a novel rapid view adaptation mechanism that selectively delivers an I-frame during view point updates by using frame indexing. We implement the proposed scheme on a commercial VR test bed where we adopt the MPEG media transport (MMT) standard with high-efficiency video coding (HEVC) tile modes. The measurement-based experiments show that the proposed scheme achieves an average data usage reduction of almost 65.2% as well as average view adaptation delay reduction of almost 57.7%.