• Proceedings of the KIEE Conference
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• 2006.10c
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• pp.391-393
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• 2006

In this paper, we develop a portable game device with uncompressed HD video and high quality sound output. Portable game devices support not only game function but also various complex functions recently. It especially supports TV-Out port to play realistic game, connecting a large screen display device. But the video and audio output signals of conventional TV-out port have the low performance and these signals are analog output. So, it is difficult that the users enjoy realistic game with benefit of high resolution digital TV. We propose the game device output with uncompressed digital signal, which has no delay of video/audio signal, also has strong immunity to external noise. Since it supports a high resolution video and high quality sound, users can playa realistic game. First, we implement the HDMI to the game device and we test reliability with the various resolutions video inputs and audio inputs. The proposed method can be applied multimedia devices requiring high performance output function as well as portable devices.

• The KIPS Transactions:PartC
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• v.12C no.2 s.98
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• pp.183-190
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• 2005

In this thesis, first, we propose I-frame encryption techniques of video data for video data itself encryption and propose license agent that processing user's certification and decryption in client system automatically when user execute encrypted video data in system server. License agent runs user's certification, encryption and decryption of video data based on PID(Public Key Infrastructure) using shared key-pool when execute of video data. Also, compose duplex buffer control and propose real time decryption method using efficient buffer scheduling to reduce much playing delay times that happen processing decryption when execute of videoa data of high-capacity.

• ETRI Journal
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• v.32 no.6
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• pp.854-862
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• 2010

One-way delay variation (OWDV) has become increasingly of interest to researchers as a way to evaluate network state and service quality, especially for real-time and streaming services such as voice-over-Internet-protocol (VoIP) and video. Many schemes for OWDV measurement require clock synchronization through the global-positioning system (GPS) or network time protocol. In clock-synchronized approaches, the accuracy of OWDV measurement depends on the accuracy of the clock synchronization. GPS provides highly accurate clock synchronization. However, the deployment of GPS on legacy network equipment might be slow and costly. This paper proposes a method for measuring OWDV that dispenses with clock synchronization. The clock synchronization problem is mainly caused by clock skew. The proposed approach is based on the measurement of inter-packet delay and accumulated OWDV. This paper shows the performance of the proposed scheme via simulations and through experiments in a VoIP network. The presented simulation and measurement results indicate that clock skew can be efficiently measured and removed and that OWDV can be measured without requiring clock synchronization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.5A
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• pp.355-362
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• 2009

In real-time communication services, delay constraints are among the most important QoS (Quality of Service) factors. In particular, it is difficult to guarantee the delay requirement over wireless channels, since they exhibit dynamic time-varying behavior and even severe burst-errors during periods of deep fading. Channel throughput may be increased, but at the cost of the additional delays when ARQ (Automatic Repeat Request) schemes are used. For real-time communication services, it is very essential to predict data deliverability. This paper derives the delay distribution and the successful delivery probability within a given delay budget using a priori channel model and a posteriori information from the perspective of queueing theory. The Gilbert-Elliot burst-noise channel is employed as an a Priori channel model, where a two-state Markov-modulated Bernoulli process $(MMBP_2)$ is used. for a posteriori information, the channel parameters, the queue-length and the initial channel state are assumed to be given. The numerical derivation is verified and analyzed via Monte Carlo simulations. This numerical derivation is then applied to a rate control scheme for real-time video transmission, where an optimal encoding rate is determined based on the future channel capacity and the distortion of the reconstructed pictures.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.8A
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• pp.820-827
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• 2010

In these days, video applications for special purposes such as video conference systems among multiple users and video surveillance systems require multiple video connections and QoS guarantee. The video systems employ IEEE 802.11 Wireless LAN devices to support broadband wireless interfaces and easy internet accesses for cheaper prices. However, according to the current IEEE 802.11e HCCA standard, if more than three video sessions are established in WSTA services, some of them must share the TXOP because the available number of TSIDs for video transmission is two. In order to resolve the problem, we devised a method which can establish up to 15 video sessions by slightly modifying the frame structure while maintaining the compatibility with current standard. Our method is implemented on the NCTUns 4.0 network simulator, and evaluated not only numerically in terms of throughput, delay, and PSNR, but also experimentally in the sense of real video clips that are used as input to our simulation. The results showed that our method sufficiently guarantees the transmission bandwidth requested by each video session.

• Journal of Korea Multimedia Society
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• v.5 no.6
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• pp.697-703
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• 2002

In this paper, we propose the adaptive multi stream transmission technique based on SPIHT video signal for the highest quality service over the current Internet that does not guarantee QoS. In addition to the reliable transmission of the video stream over the asynchronous packet network, the proposed approach provides the transmission using the adaptive frame pattern control and multi steam over the TCP for continuous replay. The adaptive frame pattern control makes the transmission date scalable in accordance with the client's buffer status. Apart from this, the multi stream transmission improves the efficiency of video stream, and is robust to the network jitter problem, and maximally utilizes the bandwidth of the client's. As a result of the experiment, the DR(delay ratio) in the proposed adaptive multi-stream transmission is more close to zero than in the existing signal stream transmission, which enables the best-efforts service to be implemented.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.8
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• pp.1571-1579
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• 2009

This paper proposes a P2P(Peer-to-Peer) streaming protocol which allows various QoS(Qualify of Service) requirements of video streaming to be supported for various personal IPTV applications in ubiquitous Internet environments. The proposed P2P streaming protocol takes fairly short startup delay, guarantees end-to-end delay bound and bandwidth requirements, and supports reliability level of video streaming for an IPTV application. The QoS-based P2P streaming protocol can properly use not only the advantages of scalability, availability, and ubiquity of P2P streaming but also complement its disadvantages of performance and reliability so that various types of personal IPTV applications can be properly implemented in the existing broadband Internet environments.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1618-1626
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• 2013

Virtualization service processes all operation including the data creation, storing, and disposal in a server and transmits processed data as the streaming media form. Therefore, client can use the same environment as the traditional desktop environment without considering the type of device. Virtualization service should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived Quality of Service(QoS) of streaming service. In this paper, we propose a network-adaptive transmission architecture that focuses on guaranteeing QoS requirements for virtualization service. In order to provide those, the proposed architecture have the transmission rate adaptation function based on available bandwidth and the content bit-rate control function based on sender buffer state. Through each function, proposed architecture guarantee the delay bounds and continuity of virtualization contents playback. The simulation results show that proposed network-adaptive transmission architecture provides a improve performance of throughput and transmission delay.

• Journal of Broadcast Engineering
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• v.12 no.6
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• pp.574-584
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• 2007

As the real time multimedia service become more popular, the needs of transmission with low delay and high quality are getting more stronger. Among those video compression technologies, the rate control method dose an important role in getting the effective data transmitting and the high image quality. In this paper, we combined the feature of CBR and VBR coding methods to propose a new bit-rate control method witch allows each frame to generates bits in the defined fluctuation range and applies a scene change detection at a part of frame and also can maintain low-delay and high quality even if the perfect VBR transmission environment is not guaranteed. The experiment result shows the proposed algorithm provides more effective method than TMN8 in real time application.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9A
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• pp.1285-1294
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• 1999

In this paper we consider the performance of TCP video traffic over ABR with Long-Range Dependent VBR traffic. As compressed coded video traffics are increasing rapidly over Internet, lots of studies are being done for transmitting those traffics efficiently using limited network resources. We consider here the transmitting video service over ABR service in ATM networks, especially satellite networks. CBR or VBR services are suggested in transmitting the video traffic in ATM Forum TM 4.0. But ABR service connection, which is considered as appropriate service for data traffic, can be established with a small amount of bandwidth, MCR (Minimum cell rate). Furthermore ABR service can control the source's transmitting rate using feedback mechanism. Using this feature ABR service can be used in some applications which can control their quality of services corresponding to network loads. Compressed video sources with MPEG-2 are used for Long-Range Dependent VBR traffic here. We model the compressed video source to resemble the MPEG-2 transport streams. These compressed video traffic streams are consisted of three different frames, I-frame, P-frame, and B-frame. So when a network are overloaded, we can control the quality of service using this traffic features. TCP Traffics over ABR need large buffers in ATM switch to satisfy their QoS with background VBR traffics, which have high deviations in bandwidth. Furthermore satellite ATM networks with large feedback delay need large buffers corresponding RTT delay. The performance comparisons among EFCI and ER switch (ERICA+) switches in the network circumstances described above were shown in this paper. We also considered the case with ON-OFF VBR traffics.