• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.1899-1906
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• 1998

In ATM switching system link bandwidth is one of the improtant resources. Thus ATM switching system must allocate the bandwidth to the users efficiently and guarantee the QoS. Especially to guarantee the QoS of the VBR traffic source such as video, users must declare the traffic characteristics and QoS expectations using UPC parameters. The CAC can use two multiplexig models in variation to link capacity and connection number. One is loseless multiplexing model; the second is a statistical multiplexing model. The loseless multiplexing model performs best when the number of mutiplexed sources is small, but as the number of multiplexed sources is increase the CAC must use the statistical multiplexing model. In this paper, the statistical multiplexing models are studied, which are suitable for high capacity ATM link on VBR traffic sources. The satistical multiplexing model and the loseless multiplexing model are combined. In statistical model we map the UPC parameters provided by new VBR connection to appropriate source traffic model. In the high capacity ATM link, as the connection number increases, the statical multiplexing gain increases.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.7A
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• pp.994-1004
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• 2000

In this paper, bitstreams are composed of using H.263 for a moving picture coding in the band-limited and error-prone environment such as wireless environment. EREC sub-frames are implemented by applying the proposed EREC algorithm in order to be UEP for the real data parts of implemented bitstreams. Because those are able to do resynchronization with a block unit, propagation of the error can be minimized, and the position of the important bits such as INTRADC and MVD can be known. Class is separated using the position of these important bits, and variable puncturing tables are designed by the class informations and the code rates of turbo codes are differently designed in according to the class. Channel coding used the turbo codes, and an interleaver to be designed in the turbo codes does not eliminate redundancy bits of the important bits in applying variable code rates of EREC sub-frames unit and is always the same at the transmitter and the receiver although being variable frame size. As a result of simulation, UEP with the code rate similar to EEP is obtained a improved result in the side of bit error probability. And the result of applying it to image knows that the subjective and objective quality have been improved by the protection of important bits.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.1A
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• pp.80-88
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• 2002

Visual media(image, graphic, and video) processing poses challenge from several perpectives, specifically from the point of view of real-time implementation and scalability. There have been several approaches to obtain speedups to meet the computing demands in multimedia processing ranging from media processors to special purpose implementations. A variety of parallel processing strategies are adopted in these implementations in order to achieve the required speedups. We have investigated a parallel processing system for improving the processing speed o f visual media related applications. The parallel processing system we proposed is similar to a pipelined memory stystem(MAMS). The multi-access memory system is made up of m memory modules and a memory controller to perform parallel memory access with a variety of combinations of 1${\times}$pq, pq${\times}$1, and p${\times}$q subarray, which improves both cost and complexity of control. Facial recognition, Phong shading, and automatic segmentation of moving object in image sequences are some that have been applied to the parallel processing system and resulted in faithful processing speed. This paper describes the parallel processing systems for the speedup and its utilization to three time-consuming applications.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.2042-2051
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• 1998

VBR compressed, pre-recorded video is known to exhibit significant, multiple time-scale bit rate variability. To reduce the variability and burstiness of the aggregated transmission rate as low as possible, in this paper, we present an adaptive bandwidth smoothing algorithm, that can be effectively applicalble for VoD services over a shared-medium channel. For these environments where many clients are connected to a single server, by introducing the conventional MVBA(minimum variability bandwidth allocation) algorithm and controlling adaptively the conventional MVBA(minimum variability bandwidth asllocation) algorithm and controlling adaptively the aggregated transmission rate whenever a new clients request is arrived at the server side, the proposed algorithm effectively reduces the burstiness and variability of the aggregated transmission rate. Through computer experiments, it is shown that the proposed method perporms better than the convertional non-aggregated bandwidth smoothing schemes in terms of the peak rate, standard deviation, number of rate changes for the aggregated traffic.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.1B
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• pp.80-86
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• 2003

High speed networking applications like video-on-demand require the support of switches that can handle very high multicast loads. Switches studied in the past can support multicast loads of 25${\sim}$30% To overcome this limitation, Tandem Expanded Delta Network(TEDN) multicast switch was proposed to support more higher multicast load by 85${\sim}$90% without degradation in the performance But there is tremendous routing capacity that is not used in the series of routing network blocks. This paper presents modified TEDN multicast switches to save the excessive routing capacity. by recirculating packets in last routing network, we can achieve the acceptable packet loss ratio with the smaller number of routing network blocks. Extensive simulations show that our proposed switches have the acceptable packet loss ratio with reduced hardware cost.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.493-505
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• 2003

A new inter-client synchronization framework for multicast media streaming is proposed employing a server-client coordinated adaptive playout control. The proposed adaptive player controls the playback speed of audio and video by adopting the time-scale modification of audio. Based on the overall synchronization status as well as the buffer occupancy level, the playout speed of each client is manipulated within a perceptually tolerable range. Additionally, the server implicitly helps increasing the time available for retransmission while the clients perform an interactive error recovery mechanism with the assistance of playout control. The network-simulator based simulations show that the proposed framework can reduce the playout discontinuity without degrading the media quality, and thus mitigate the client heterogeneity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5C
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• pp.514-522
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• 2003

Contrary to the previous methods, in this paper, we propose an enhanced shot change detection method using multiple features and binary decision tree. The previous methods usually used single feature and fixed threshold between consecutive frames. However, contents such as color, shape, background, and texture change simultaneously at shot change points in a video sequence. Therefore, in this paper, we detect the shot changes effectively using multiple features, which are supplementary each other, rather than using single feature. In order to classify the shot changes, we use binary classification tree. According to this classification result, we extract important features among the multiple features and obtain threshold value for each feature. We also perform the cross-validation and droop-case to verify the performance of our method. From an experimental result, it was revealed that the EI of our method performed average of 2% better than that of the conventional shot change detection methods.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.10
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• pp.2090-2101
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• 2015

We propose a robust tracking method that combines the merits of ACM(active contour model) and the color-based PF(particle filter), effectively. In the proposed method, PF and ACM track the color distribution and the contour of the target, respectively, and Decision part merges the estimate results from the two trackers to determine the position and scale of the target and to update the target model. By controlling the internal energy of ACM based on the estimate of the position and scale from PF tracker, we can prevent the snake pointers from falsely converging to the background clutters. We appled the proposed method to track the head of person in video and have conducted computer experiments to analyze the errors of the estimated position and scale.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.4C
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• pp.277-282
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• 2012

In this paper, we present a real-time eye contact system for realistic telepresence using a Kinect depth camera. In order to generate the eye contact image, we capture a pair of color and depth video. Then, the foreground single user is separated from the background. Since the raw depth data includes several types of noises, we perform a joint bilateral filtering method. We apply the discontinuity-adaptive depth filter to the filtered depth map to reduce the disocclusion area. From the color image and the preprocessed depth map, we construct a user mesh model at the virtual viewpoint. The entire system is implemented through GPU-based parallel programming for real-time processing. Experimental results have shown that the proposed eye contact system is efficient in realizing eye contact, providing the realistic telepresence.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.8B
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• pp.613-619
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• 2012

The Long Term Evolution (LTE) system is already able to provide a background of variety services for mobile users with multimedia services such as audio, video, and data. In fact, the High Speed Packet Access plus (HSPA+) solution can greatly enhance bit rates on down-link. However, the supporting for multimedia applications with different QoS (Quality of Service) requirements is not devised yet. Hence, in this paper we propose an effective packet scheduling algorithm based on Proportional Fairness (PF) scheduling algorithms for the LTE. In this proposed packet scheduling scheme, we optimized instantaneous user data rates and the traffic class weight which prioritize user's packets. Finally, we evaluated and showed the performance of the proposed scheduling algorithm through simulations of multimedia traffics being transmitted to users over LTE links in a multi-cell environment.