• Journal of the Korean Operations Research and Management Science Society
• /
• v.24 no.3
• /
• pp.63-72
• /
• 1999

Recent studies on the Constant Bit Rate and Variable Bit Rate transmissions have mainly focused on the frame by frame encoder rate control based on the quantization parameter. With the existing approaches it is difficult to guarantee a consistent video quality. Also, the rate control overhead is too high for the real-time video sources. In this paper, a channel rate allocation scheme based on the control period is proposed to transmit a real-time video, in which the control period is defined by a pre-specified number of frames or group of pictures. At each control period, video traffic information is collected to determine the channel rate at the next control period. The channel rate is allocated to satisfy various channel rate constraints such that the buffer occupancy at the decoder is maintained at a target level. If the allocated channel rate approaches the level at which the negotiated traffic descriptions may be violated, the encoder rate is decreased through adjusting quantization parameters in the MPEG encoder. In the experimental results, the video quality and the overflow and underflow probabilities at the buffer are compared at different control periods. Experiments show that the video quality and the utilization of network bandwidth resources can be optimized through the suitable selection of the control period.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.21 no.9
• /
• pp.2328-2340
• /
• 1996

Until now, the transmission of data for VOD(Video on Demenad) was based on a real time modelling of video data. Markow Modulated Fluid Sources(MMFS) and Markow Modulated Poisson Sources(MMPS) are the most widely used modelling methods. But the charactersitics of the VBR(Variable Bit Rate) signal prevents modelling from actually being "real-time". Also these methods call for the use of large buffers for the abolishment of cell loss. These modelling methods are, of course, useful i case of teleconferences where a real time modelling of video traffic is inevitable, but they are insufficient in cases where the characteristic infomation of video traffic can be obtained beforehand-cases such as VOD. Video data is speial in that if one file is preprocessed all other products can simply be copied from that onepreprocessed file. This characteristic helps reduce the overhead arising from the job of drawing out characteristic information to almost zero. But still, compared to the existing real time modelling method data transmission using characteristic information succeeds in raising the efficiency of data transmission. In tis paper we will outline a method of dta transmission which use the characteristic information of each video stream, and evaluate this method through some experiments.periments.

• Journal of Korean Institute of Industrial Engineers
• /
• v.25 no.2
• /
• pp.162-172
• /
• 1999

For VBR video traffic, Motion-Picture Experts Group(MPEG) coding algorithm was adopted as the standard coding algorithm by International Telecommunication Union(ITU). In this paper, we propose a traffic model of an MPEG coded video traffic in frame level and cell level, and develop an approximate model for evaluating performance of a ATM multiplexer with homogeneous MPEG video sources by considering burst-level variation of aggregated traffics. For homogeneous MPEG video traffics which are frame-synchronized, the performance of the ATM multiplexer is influenced by source correlation at the multiplexing time. When sources are highly correlated, we decompose the aggregated cell streams by the frame-type and model multiplexing process during a frame time as n*D/D/1/K queueing model and suggest an approximate method for obtaining CLP of the ATM multiplexer. In the case that sources are highly correlated, the solution has the meaning of the upper bounds of performance of the ATM multiplexer. For the verification of our model, we compare the solution of our model with simulation resets. As the number of sources increases. The CLP obtained from our model approaches to simulation results, and gives upper bounds of simulation results.

• Journal of KIISE:Computer Systems and Theory
• /
• v.33 no.10
• /
• pp.722-733
• /
• 2006

In scalable streaming application, there are two important knobs to tune to effectively exploit the underlying network resource and to maximize the user perceivable quality of service(QoS): layer selection and packet scheduling. In this work, we propose Semantics Aware Packet Scheduling (SAPS) algorithm to address these issues. Using packet dependency graph, SAPS algorithm selects a layer to maximize QoS. We aim at minimizing distortion in selecting layers. In inter-frame coded video streaming, minimizing packet loss does not imply maximizing QoS. In determining the packet transmission schedule, we exploit the fact that significance of each packet loss is different dependent upon its frame type and the position within group of picture(GOP). In SAPS algorithm, each packet is assigned a weight called QoS Impact Factor Transmission schedule is derived based upon weighted smoothing. In simulation experiment, we observed that QOS actually improves when packet loss becomes worse. The simulation results show that the SAPS not only maximizes user perceivable QoS but also minimizes resource requirements.

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