• Proceedings of the Korean Society of Broadcast Engineers Conference
• /
• 1997.06a
• /
• pp.43-48
• /
• 1997

The video transmission for real-time viewing over the Internet is a core operation for the multimedia services. However, its realization is very difficult because the Internet has two major problems, namely, very narrow endpoint-bandwidth and the network jitter. We already proposed a scalable video transmission method in [8] which used MPEG-4 video VM(Verification Model) 2.0[3] for very low bit rate coding and an adaptive temporal rate control of video objects to overcome the network jitter problem. In this paper, we present the improved adaptive temporal rate control scheme for the scalable transmission. Experimental results for three test video sequences show that the adaptive temporal rate control can transfer the video bitstream at source frame rate under variable network condition.

• Journal of KIISE:Computing Practices and Letters
• /
• v.15 no.2
• /
• pp.114-124
• /
• 2009

In this paper, we propose a practical TCP friendly rate control scheme that considers the minimum channel bandwidth of the network when transporting SVC (scalable video coding) video over IP netowrks such as Internet. RTP and RTCP is mainly designed for use with UDP (User Datagram Protocol) for real-time video transport over the Internet. TCP-friendly rate control was proposed to satisfy the demands of multimedia applications while being reasonably fair when competing for bandwidth with conventional TCP applications. However the rate control model of the conventional TCP-friendly rate control scheme does not consider the minimum channel bandwidth of the network. Thus the estimated channel bandwidth by the conventional rate control model might be quite different from the real channel bandwidth when the packet loss ratio of the network is very large. In this paper, we propose a modified TCP-friendly rate control scheme that considers the minimum channel bandwidth of the network. Based on the modified TCP-friendly rate control, we assign the minimum channel bandwidth to the base layer bitstream of SVC video, and remaining available bandwidth is allocated to the enhancement layer of SVC video for the TCP friendly scalable video transmission. It is shown by simulations that the modified TCP-friendly rate control scheme can be effectively used for a wider range of controlled bit rates depending on the packet loss ratio than the conventional TCP-friendly control scheme. Furthermore, the effectiveness of the proposed scheme in terms of objective video quality is proved by comparing PSNR performance with the conventional scheme.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.34 no.5B
• /
• pp.517-525
• /
• 2009

We propose a quality guaranteed scalable video streaming service over the Internet using a new rate adaptation algorithm. Because video data requires much more bandwidth rather than other types of service, therefore, quality of video streaming service should be guaranteed while providing friendliness with other service flows over the Internet. To successfully provide this, we propose a framework for providing quality-guaranteed streaming service using two-channel transport layer and rate adaptation of scalable video stream. In this framework, baseline layer for scalable video is transmitted using TCP transport for minimum qualify service. Enhancement layers are delivered using TFRC transport with layer adaptation algorithm. The proposed framework jointly uses the status of playout buffer in the client and the encoding rate of layers in media contents. Therefore, the proposed algorithm can remarkably guarantee minimum quality of streaming service rather than conventional approaches regardless of network congestion and the encoding rate variation of media content.

• Journal of Communications and Networks
• /
• v.10 no.3
• /
• pp.338-350
• /
• 2008

The differentiated services (DiffServ) is proposed to provide packet level service differentiations in a scalable manner. To provide an end-to-end service differentiation to users having a connection over multiple domains, as well as a flow marker, an intermediate marker is necessary at the edge routers, and it should not be operated at a flow level due to a scalability problem. Due to this operation requirement, the intermediate marker has a fairness problem among the transmission control protocol (TCP) flows since TCP flows have intrinsically unfair throughputs due to the TCP's congestion control algorithm. Moreover, it is very difficult to resolve this problem without individual flow state information such as round trip time (RTT) and sending rate of each flow. In this paper, to resolve this TCP fairness problem of an intermediate marker, we propose a fair scalable marker (FSM) as an intermediate marker which works with a source flow three color marker (sf-TCM) operating as a host source marker. The proposed fair scalable marker improves the fairness among the TCP flows with different RTTs without per-flow management. Through the simulations, we show that the FSM can improve TCP fairness as well as link utilization in multiple domain DiffServ networks.

• ETRI Journal
• /
• v.45 no.5
• /
• pp.874-886
• /
• 2023

A joint resource-optimization scheme is investigated for nonorthogonal multiple access (NOMA)-enhanced scalable video coding (SVC) multicast in unmanned aerial vehicle (UAV)-assisted radio-access networks (RANs). This scheme allows a ground base station and UAVs to simultaneously multicast successive video layers in SVC with successive interference cancellation in NOMA. A video quality-maximization problem is formulated as a mixed-integer nonlinear programming problem to determine the UAV deployment and association, RAN spectrum allocation for multicast groups, and UAV transmit power. The optimization problem is decoupled into the UAV deployment-association, spectrum-partition, and UAV transmit-power-control subproblems. A heuristic strategy is designed to determine the UAV deployment and association patterns. An upgraded knapsack algorithm is developed to solve spectrum partition, followed by fast UAV power fine-tuning to further boost the performance. The simulation results confirm that the proposed scheme improves the average peak signal-to-noise ratio, aggregate videoreception rate, and spectrum utilization over various baselines.

• ETRI Journal
• /
• v.26 no.4
• /
• pp.351-359
• /
• 2004

Flow aggregation is a scalable method to provide quality of service (QoS) guarantees to a large number of flows economically. A round-robin scheduler is an efficient scheduling algorithm. We investigate flow aggregation using a round-robin scheduler and propose the use of periodic timer interrupts for rate control of the round-robin scheduler. The proposed flow aggregator is a single-stage scheduler compared to Cobb's two-stage flow aggregator consisting of an aggregator and non-aggregating scheduler. It is possible to implement flow aggregation in the existing routers with only a software upgrade. We also present a simulation study showing the delay behaviors of the proposed algorithm.

• Journal of Korea Game Society
• /
• v.2 no.1
• /
• pp.23-29
• /
• 2002

Multicast protocols are efficient methods of group communication such as video conference, Internet broadcasting and On-Line Game, but they do not support the various transmission protocol services like a reliability guarantee, FTP, or Telnet that TCPs do. The Purpose or this Paper is to find a method to utilize multicast routers can simultaneously transport multicast packets and TCP packets. For multicast network scalability and error recovery the existing SRM(Scalable Reliable Multicast)method has been used. Three packets per TCP transmission control window site are used for transport and an ACK is used for flow control. A CBR(Constant Bit Rate) and a SRM is used for UDP traffic control. Divided on whether a UDP multicast packet and TCP unicast packet is used simultaneously or only a UDP multicast packet transport is used, the multicast receiver with the longest delay is measured on the number of packets and its data receiving rate. It can be seen that the UDP packet and the TCP's IP packet can be simultaneously used in a server router.

• Journal of Broadcast Engineering
• /
• v.10 no.4 s.29
• /
• pp.488-504
• /
• 2005

In this paper, we propose a new bitrate control scheme to improve the quality of image encoded by SVC and to resolve the problems of conventional scheme. In JSVM2.0, bitrate of a frame is controlled by an initial quantization parameter and scaling factor that it hasdifferent value according to frame. Itis difficult to get the best of video quality at arbitrary bitrate because the conventional scheme has two defects. One is that we have to know proper initial QP's fur all sequences. Another is that QP's control skill for macroblocks is very inefficient. In this paper, we propose an efficient bit allocation algorithm to reduce the effect of the initial QP and to increase the efficiency of bit allocation by using proper QP's for macroblocks. In simulation results, it can be seen that using the proposed scheme enables the SVC encoder to control the bitrate by the macroblock unit and outperforms the conventional schemes in the respect of rate-distortion.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.2A
• /
• pp.165-172
• /
• 2002

The concept of MPEG-2 transcoding is related to the scaling the bit rate of the previously encoded MPEG bitstream. We proposed a new rate control algorithm that outperforms existing methods and showed the performance by a computer simulation. A rate-quatization model is developed based on a priori knowledge from the given bitstream at macroblock level. The model is independent of the picture coding type, because it is designed in the DCT domain. To increase the accuracy of the model, the activity measurement is considered. By the simulation, the proposed transcoding showed the better result than that of recoding by TM5. We also concluded that MPEG-2 scalable profile is not suitable for the bit rate conversion application.

• Journal of Korea Multimedia Society
• /
• v.15 no.9
• /
• pp.1086-1092
• /
• 2012

In this paper, we propose the rate control scheme reducing frame skipping at low bit-rate. As the method considering lossy parts in the compressing process at the existing scheme, the proposed scheme is improved by subtracting the converted bits from the target bit-rate of the current frame. The converted bits are the value resulted from multiplying the ratio of the current frame MAD to the previous frame MAD by the compressed bits changed from the remaining values after compressing the previous frame.