• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.668-672
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• 2009

In wireless ad-hoc network, knowing the available bandwidth of the time varying channel is imperative for live video streaming applications. This is because the available bandwidth is varying all the time and strictly limited against the large data size of video streaming. Additionally, adapting the encoding rate to the suitable bit-rate for the network, where an overlarge encoding rate induces congestion loss and playback delay, decreases the loss and delay. While some effective rate controlling methods have been proposed and simulated well like VTP (Video Transport Protocol) [1], implementing to cooperate with the encoder and tuning the parameters are still challenging works. In this paper, we show our result of the implementation experiment of VTP based encoding rate controlling method and then introduce some techniques of our parameter tuning for a video streaming application over wireless environment.

• Journal of Korea Society of Digital Industry and Information Management
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• v.6 no.2
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• pp.11-17
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• 2010

In this paper, we propose efficient rate control algorithm for video near-lossless compression. The proposed method adjusts pixel error range using amounts of previous encoded bits and target bits at a slice encoding interval. Thus, encoded bits by the proposed method are transmitted well through limited bandwidth without overflow. Also, target bits and encoded bits can be controlled according to transmission policy. Experimental results show that our proposed method not only almost fits compression ratio, but also has better image quality.

• JSTS:Journal of Semiconductor Technology and Science
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• v.11 no.1
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• pp.33-39
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• 2011

H.264 decoders usually have pipeline architecture by a macroblock or a 4 ${\times}$ 4 sub-block. The period of the pipeline is usually fixed to guarantee the operation in the worst case which results in many idle cycles and higher data bandwidth. Adaptive pipeline architecture for H.264 decoders has been proposed for efficient decoding and lower the requirement of the bandwidth for the memory bus. However, it requires a controller for the adaptive priority control to utilize the advantage. We propose a smart bus arbiter that replaces the controller. It is introduced to adjust the priority adaptively the QoS (Quality of Service) control of the decoding process. The smart arbiter can be integrated the arbiter of bus systems and it works when certain conditions are met so that it does not affect the original functions of the arbiter. An H.264 decoder using the proposed architecture is designed and implemented to verify the operation using an FPGA.

• Journal of Communications and Networks
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• v.9 no.3
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• pp.236-246
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• 2007

This paper investigates the performance of the transmission control protocol (TCP) transport protocol over IEEE 802.11 infrastructure based wireless networks. A wireless link is generally characterized by high transmission errors, random interference and a varying latency. The erratic packet losses usually lead to a curbing of the flow of segments on the TCP connection and thus limit TCP's performance. This paper examines the impact of the lossy nature of IEEE 802.11 wireless networks on the TCP performance and proposes a scheme to improve the performance of TCP over wireless links. A negative acknowledgment scheme, selective negative acknowledgment (SNACK), is applied on TCP over wireless networks and a series of ns-2 simulations are performed to compare its performance against that of other TCP schemes. The simulation results confirm that SNACK and its proposed enhancement SNACK-S, which incorporates a bandwidth estimation model at the sender, outperform conventional TCP implementations in 802.11 wireless networks.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.149-158
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• 2005

This paper proposes a congestion avoidance algorithm which provides stable throughput and transmission rate regardless of buffer size by limiting the TCP congestion window in fixed bandwidth networks. Additive Increase, Multiplicative Decrease (AIMD) is the most commonly used congestion control algorithm. But, the AIMD-based TCP congestion control method causes unnecessary packet losses and retransmissions from the congestion window increment for available bandwidth verification when used in fixed bandwidth networks. In addition, the saw tooth variation of TCP throughput is inappropriate to be adopted for the applications that require low bandwidth variation. We present an algorithm in which congestion window can be limited under appropriate circumstances to avoid congestion losses while still addressing fairness issues. The maximum congestion window is determined from delay information to avoid queueing at the bottleneck node, hence stabilizes the throughput and the transmission rate of the connection without buffer and window control process. Simulations have performed to verify compatibility, steady state throughput, steady state packet loss count, and the variance of congestion window. The proposed algorithm can be easily adopted to the sender and is easy to deploy avoiding changes in network routers and user programs. The proposed algorithm can be applied to enhance the performance of the high-speed access network which is one of the fixed bandwidth networks.

• Journal of Korea Multimedia Society
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• v.13 no.3
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• pp.349-358
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• 2010

Adaptability is an important issue in video streaming over mobile environments, since the clients may request videos with great differences in their workload. In this paper, we propose the issues in limited bandwidth scheduling for efficient MPEG-4 video stream transmission over a mobile or wireless network. In the phase of admission control, the amount of bandwidth allocated to serve a video request is the mean bandwidth requirement of its requested video. The dynamic allocation of bandwidth in the phase of scheduling depends on the playback buffer levels of the clients with an objective to make it more adaptive to the playback situation of individual clients. In the proposed RTA scheduling protocol, more bandwidth may be allocated temporarily to the client whose buffer level is low. By employing the buffer level based scheduling policy, this protocol attempts to maximize the real-time performance of individual playback while minimizing the impact of transient overloading. Extensive simulation experiments have been performed to investigate the performance characteristics of the RTA protocol as comparing with BSBA protocol. This RTA protocol shows the better performance by transferring more frames than BSBA protocol.Computer simulations reveals that the standard deviation of the bit rate error of the proposed scheme is 50% less than that of the conventional method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.6
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• pp.1387-1394
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• 2005

In this paper, we propose and the call admission control mechanism that can improve the capacity of the wireless system for the non-uniform traffic load distribution based intelligent information in multiple cellular CDMA environment. The number of mobile stations that can be served simultaneously in a base station is limited by the amount of total interference received in CDMA system. Further, the average number of mobile stations in each cell may not be uniformly distributed. In this paper, considering this factors, the call admission control mechanism using the effective bandwidth concept is adapted to improve the system capacity of non-uniform traffic load distribution based intelligent information. Thus, the bandwidth for a new call can be varied dynamically for reducing the blocking rate of new calls and the dropping rate of handoff calls. The suggested call admission control mechanism is experimented through simulation by dynamically assigning the bandwidth to new and handoff calls. The simulation results show that the proposed call admission control mechanism can accommodate more mobile stations than the other methods in multiple cellular CDMA environment.

• The Journal of the Korea Contents Association
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• v.8 no.8
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• pp.57-64
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• 2008

In this paper, to resolve the problem caused by a network state information inaccuracy the slow delay time that conclusion of network state of one way delay time which accuracy delay time information, according to network state changes on the based TFRC flow control, and suggest that flow control mechanism to adjust transfer rate fit of real time multimedia data. In simulation, to measure of netowork state information that on the average about 12% difference of compared RTT and $OWD{\times}2$. When used RTT, used fair bandwidth TFRC much better than TCP about 32%, when used OWD, difference about 3% used fair bandwidth. Thus, conclusion of accuracy network state that used fair bandwidth according to network state changes on the based TFRC, users can support service of high quality that flow control mechanism to adjust transfer rate fit of real time data.

• Journal of Korea Multimedia Society
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• v.7 no.5
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• pp.737-743
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• 2004

In this paper, we propose the call control scheme that can improve the capacity of the wireless system for the non-uniform traffic load distribution and the multiple types of services in multiple cells CDMA system. The number of mobile stations that can be served simultaneously in a base station is limited by the amount of total interference received in CDMA system. Further, the average number of mobile stations in each cell may not be uniformly distributed. Considering this factors, the call admission control scheme using the effective bandwidth concept is adapted in this paper. Thus, the bandwidth for a new call can be varied dynamically for reducing the blocking rate of new calls and the dropping rate of handoff calls. The suggested call control scheme is experimented through a simulation by dynamically assigning the bandwidth to new and handoff calls. The simulation results show that the proposed call control scheme can accommodate more mobile stations than the other methods in multiple cells environment.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.2
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• pp.310-323
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• 2004

In this paper, we extend the multiple time scale control framework to window-based congestion control, in particular, TCP This is performed by interfacing TCP with a large tine scale control nodule which adjusts the aggressiveness of bandwidth consumption behavior exhibited by TCP as a function of "large time scale" network state. i.e., conformation that exceeds the horizon of the feedback loop as determined by RTT Our contribution is threefold. First, we define a modular extension of TCP-a function call with a simple interface-that applies to various flavors of TCP-e.g., Tahoe, Reno, Vegas and show that it significantly improves performance. Second, we show that multiple time scale TCP endows the underlying feedback control with preactivity by bridging the uncertainty gap associated with reactive controls which is exacerbated by the high delay-bandwidth product in broadband wide area networks. Third, we investigate the influence of three traffic control dimensions-tracking ability, connection duration, and fairness-on performance. Performance evaluation of multiple time scale TCP is facilitated by a simulation bench-mark environment which is based on physical modeling of self-similar traffic.