• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.6
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• pp.53-67
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• 2002

The multimedia synchronization is one of the key issues to be resolved in order to provide a good quality of multimedia related services, such as Video on Demands(VoD), Lecture on Demands(LoD), and tele-conferences. In this paper, we introduce an adaptive multimedia synchronization algorithm using the level of buffers and load of servers, which are modeled and analyzed by ExSpect, a Petri net based simulation tool. In the proposed algorithm, the audio and video buffers are divided to 5 different levels, and the pre-defined play-out speed controller tries to make the buffer level to be normal in different temporal relations between multimedia streams using buffer levels and server loads. Because each multimedia packet is played by the pre-defined play-out speed, the media data can be reproduced within the permissible limit of errors while preserving the level of buffers to be normal. The proposed algorithm is able to handle and support various communication restrictions between providers and users, and offers little jitter play-out to many users in networks with the limited transmission capability. The performance of the developed algorithm is analyzed in various network conditions using a Petri net simulation tool.

• The Transactions of the Korea Information Processing Society
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• v.4 no.7
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• pp.1770-1780
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• 1997

Our problem is improvement of image quality because it is inevitable cell loss of image data when traffic congestion occurs. If cells are discarded indiscriminately in transmission of MPEG video data, it occurs severe degradation in quality of service(QOS). In this paper, to solve this problem, we propose two method. The first, we analyze the traffic characteristics of an MPEG encoder and generate high priority and low priority data stream. During network congestion, only the least low priority cells are dropped, and this ensures that the high priority cells are successfully transmitted, which, in turn, guarantees satisfactory QoS. In this case, the prioritization scheme for the encoder assigns components of the data stream to each priority level based on the value of a parameter ${\beta}$. The second, Number of high priority cells are increased when value of ${\beta}$ is large. It occurs the loss of high priority cell in the congestion. To prevent it, this paper is regulated to data stream rate as buffer occupancy with UPC controller. Therefore, encoder's bandwidth can be calculated renegotiation of the encoder and networks. In this paper, the encoder's bandwidth requirements are characterized by a usage parameter control (UPC) set consisting of peak rate, burstness, and sustained rate. An adaptive encoder rate control algorithm at the Networks Interface Card(NIC) computes the necessary UPC parameter to maintain the user specified quality of service. Simulation results are given for a rate-controlled VBR video encoder operating through an ATM network interface which supports dynamic UPC. These results show that dynamic bandwidth renegotiation of prioritized data stream could provided bandwidth saving and significant quality gains which guarantee high priority data stream.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.10a
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• pp.791-793
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• 2015

Recently, UCC (User Created Contents) and VoD (Video on Demand), and multimedia content are growing, IP-TV, Smart TV, OHTV (Open Hybrid TV) various services such as multi platform (Multi-platform) environment, services and QoS issues. To solve this problem, the network efficiently, and improve the quality of content is necessary for the system. In this paper, the network of channels State and transmission of multimedia data based on dynamic resource usage, TCP and UDP, Adaptive dual-streaming system used for design and analysis. In addition, the existing TCP and UDP streaming system using a single protocol for analysis and verification of the effectiveness of the difference between and. This is a disaster, and medical/first aid system will be utilized in the field of feed, are ubiquitous.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.4
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• pp.940-949
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• 1998

The paper proposes an effective mechanism hiring the adaptive buffer control schemes on the mobile seceiver's side that can resolve the problems of delay spikes caused by the handover process onthe cell boundaries when they transmit real-time moving images through the high speed wireless channels. to confirm the effectiveness of the proposed schemes, we have modeled the characteristics of the random errors on the wireless channels, the burst errors and that of the transmission delay by the handover. We also compared the rate of the delay-adaptation of the receive buffers caused by the delay and delay spikes. the comparison was done by testing the suggested schemes against the existing schemes by applying tehm to the packet delay model. We also could identify the noticable reduction of the casesof buffer delay and overflow compared with the conventional schemes, by applying the suggested algorithm to the video image compressed by H.263.

• Journal of KIISE:Computing Practices and Letters
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• v.6 no.5
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• pp.505-512
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• 2000

In this paper, an adaptive multimedia synchronization scheme for VOD (Video On Demand) services in internet environments is proposed. This scheme considers the characteristics of MPEG (Moving Picture Expert Group) system stream. Consequently, the intra-synchronization is handled at the pack layer, and the inter-synchronization is handled at the packet layer. The proposed scheme can cope adaptively with variation of packet loss, jitter and client's playback capacity. If there are variations of the packet loss or client's playback capacity, the server will change the transmission rate by selective picture skip. The client can then adjust and control the playback time according to the variation of the network jitter. Our experimental results show that the proposed scheme can quickly adapt to the network condition, and can guarantee a better quality of service than the other existing schemes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.5C
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• pp.464-471
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• 2010

The demand for high quality of multimedia applications, which far surpasses the rapid evolution of transmission and storage technologies, makes better compression coding capabilities ever increasingly more important. In order to provide enhanced video coding performance, this paper proposes an efficient competition-based motion vector coding scheme. The proposed algorithm adaptively forms the motion vector predictors based on the contexts of scene characteristics such as camera motion and nearby motion vectors, providing more efficient candidate predictors than the previous competition-based motion vector coding schemes which resort to the fixed candidates optimized by extensive simulations. Up to 200% of compression gain was observed in the experimental results for the proposed scheme applied to the motion vector selection for skip mode processing.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.6A
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• pp.676-684
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• 2008

In digital video broadcasting-terrestrial (DVB-T), which is the European digital terrestrial television standard, the orthogonal frequency division multiplexing (OFDM) has been adopted for signal transmission. The main reasons using OFDM are to increase the robustness against the frequency selective fading and impulse noise, and to use available bandwidth efficiently. However, channel variation within an OFDM symbol destroys orthogonality between subcarriers, resulting in inter-carrier interference (ICI), which increases an error floor in proportional to maximum Doppler spread. This paper provides an ICI analysis in both time and frequency domains while existing literatures analyze the ICI effects mainly in frequency domain and proposes the algorithms that estimate the channel impulse response and channel variation using least square (LS) algorithm which is the most simple channel estimation technique. And we propose adaptive channel estimation algorithm that estimates the velocity of terminals. The simulation results show that proposed algorithm has similar performance with about 1.5% computational complexity of noise and ICI reduction LS algorithm in low speed environments.

• The KIPS Transactions:PartA
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• v.16A no.5
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• pp.327-338
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• 2009

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, PAP, for guarantee of constant QoS. The PAP prefetches frames by increasing the frame transmission rate while the available network bandwidth is high. The PAP uses the prefetched frames to guarantee the QoS while the available network bandwidth is low and increases the playout interval to prevent buffer underflow. The experiment result shows that the proposed PAP could guarantee the constant QoS by prefetching the frames adaptively to the network bandwidth with the characteristic of video stream.

• Journal of Broadcast Engineering
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• v.24 no.2
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• pp.329-340
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• 2019

Recently, broadcasting technologies have evolved as high-quality AV services such as domestic terrestrial UHD(Ultra-High Definition) broadcasting have been increasing, and broadcasting standards have been newly defined. Also, as network technology develops, contents are consumed not only in the country but also the world. Accordingly, content insertion technology, which is a method of providing suitable contents in accordance with the national and local environments, will be needed. This paper proposes a content insertion service system model and synchronization scheme using ATSC(Advanced Television Systems Committee) 3.0 Event Signaling standard under heterogeneous network environment of broadcasting network and internet network based on transmission standard DASH(Dynamic Adaptive Streaming over HTTP)/ROUTE(Real time Object delivery Over Unidirectional Transport) and MMT(MPEG Media Transport) of terrestrial UHD broadcasting. It also verifies that the service operates in an environment that meets the broadcast standard.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.6
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• pp.1490-1500
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• 1996

One of the important issues for multimedia communications over ATM netowrks is efficient use of net-work resources since transmitting video at low cost requires high utilization of channel bandwidth. In this paper, we propose an effective bandwidth estimation scheme for MPEG sources in ATM networks. It has been known difficult to allocate effective transmission bandwidth to MPEG sources due to its bursty charactersitics. The MPEG traffic stream can bemodeled using five traffic parameters:service rate, burst load ratio, interburst load ratio, burst period, and interburst period. Using these parameters, a transmission bandwidth for each source can be estimated, and the estimated bandwidth is applied synchronously to reduce the queue size in a buffer. For a simple network modelof a high speed link that multiplexes a number of virtual-circuit connections, simulations on adaptive bandwidth allocation were performed, and the results show that the queueing delay is significantly reduced, when compared to a fixed bandwidth allocation.