• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.591-601
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• 2002

Motion compensation is a powerful method to compress an image sequence. Its main drawback is that once an error is occurred, the error propagates through the frames. Recently, the intra update method was proposed to stop the error propagation at the expense of reduction in compression efficiency. This paper proposes an intra update method based on a rate-distortion optimization in error prone environments. The rate and the distortion are estimated using the Lagrangian optimization to select the coding mode and the quantization step size. The proposed method is applied to MPEG-4 codec, and the experimental results show that it is robust to the error such as packet losses comparing with the conventional ones.

• Journal of KIISE:Information Networking
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• v.28 no.2
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• pp.236-241
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• 2001

A new scheme is proposed to speed up a known bandwidth measuring method which employs potential bandwidth for filtering out noiscs (in cstirna60nl from time compression caused by a packet queueing ahead of two probe packets. Instead of inerementing the potential bandwidth by a fixed amount as in the original method we increase the potential bandwidth exponentially for faster convergence. To retain its filtering capability as well as its agility to adapt to new bottleneck bandwidth, each trial potential bandwidth(PB) is adjusted using MAX and MIN as upper bound and lower bound. An experiment using known bandwidths shows 45~g9% improvement in conVergence time.

• Journal of the Korea Society of Computer and Information
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• v.4 no.4
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• pp.155-162
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• 1999

Various applications of video conferencing are required real-time transmission in order to offer service of best effort in internet. Because the bandwidth of internet changes dynamically, appropriated QoS could not be guaranteed To resolve the problem. available bandwidth between sender and receiver is measured. And according to measured bandwidth, the transmission of multimedia data is controlled In this paper, we propose algorithm of efficient transmission for best QoS in internet According to a present status of network, we measure available bandwidth using feedback RTCP information and change a compression rate to reduce a producing CODEC data. And according to the priority that is measured by packet loss for received RTCP information, we abandon frames indicated as lower weight in transmission buffer of sender.

• The KIPS Transactions:PartB
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• v.16B no.2
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• pp.101-108
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• 2009

As content serviced for IP set-top boxes is streamed over IP network, the existing hacking tools for IP network can be used to capture the streamed content. Until recently, most of the content serviced on IP set-top boxes has been MPEG-2 TS. However, this content will be gradually moved to WMV, MPEG-4 or H.264 because of the relatively low compression efficiency and overhead of the TS packet. In this paper, we propose a DRM scheme other than WMRM for streamed WMV content. Our approach is to design a DRM scheme independent to the existing WMV streaming system. We also design this scheme in order to provide the feature for controlling the DRM processing time considering device performance. We verified it through the experiment.

• Journal of the Institute of Convergence Signal Processing
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• v.3 no.1
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• pp.21-26
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• 2002

Video stream invoke a variety of traffic with the structure of compression algorithm and image complexity. Thus, it is difficult to allocate the resource on the both sides of sender and receiver, and playout on the Internet such as a packet switched network. Thus, in this paper we proposed video stream smoothing using multistream for the effective transmission of video stream. This method specifies the type of LDU(logical data unit) according to the type of original stream, and then makes a large number of streams as a fixed size, and transfers them. So, the proposed method can reduce the buffering time which occurs during the process of the smoothing and prefetch be robust to the jitter on network, as well. Consequently, it has the effective transmission characteristics of fully utilizing the clients bandwidth.

• Proceedings of the KSRS Conference
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• 2002.10a
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• pp.228-228
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• 2002

The DLS(Data Link System) is located in the PDTS(Payload Data Transmission Subsystem) of KOMPSAT-II, and its main function is to provide communication link with Ground Segment as a space segment. DLS receive the data of MSC, OBC from DCSU(Data Compression Storage Unit) and transmit to the Ground Station by X-Band RF link. DLS is consist of CCU(Channel Coding Unit), QTX(QPSK Transmitter, ASU(Antenna Switch Unit) CCU makes a packet for communication after several kind of data processing such like Ciphering, RS Coding. QTX transmit PDTS data by OQPSK. Modulation. ASU is the unit for reliability of antenna switching. So, DLS's function is consists of ciphering, RS coding, CCSDS packetizing, randomizing, modulation and switching to antenna. These DLS's functions are controlled by PMU(Payload Management Unit). All commands to DLS are sent by PMU and all telemetries of DLS are sent to the PMU. The PMU receives commands from OBC and sends telemetries to the OBC. The OBC communicates with Ground Station by S-Band RF link. This paper presents the on-orbit DLS operation concept through the ground segment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.5B
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• pp.437-442
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• 2011

As an Internet Engineering Task Force (IETF) Working Group, 6LoWPAN is standardizing the IPv6 packet transfer technology in accordance with IEEE 802.15.4. It has completed two Request for Comments (RFC) documents, one of which, RFC 4944, addresses fragmentation, reassembly, and header compression technologies. In this paper, a communication mechanism is proposed to provide efficient communication between 6LoWPAN and external nodes. In this mechanism, the gateway between 6LoWPAN and external networks serves as the proxy gateway between nodes. The simulation was conducted using QualNet to compare the performance of the proposed mechanism and the existing RFC 4944 method. The comparative analysis of the proposed mechanism and the existing method showed that the proposed method performed better.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.2 s.332
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• pp.89-96
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• 2005

In this paper, we propose a customer premise platform for processing multimedia data service on the ATM network. The proposed platform has a specific AAL2 processor that includes AAL2 protocol and scheduler algorithm so as to off-load large potion of burden from host processor and make it easy to process multimedia data from the ATM network in real time compared with conventional platform in which AAL/ATM tasks are processed by software. The ATS scheduler that is implemented based on 2-level time slot ring provides a simple and efficient method for scheduling data of VBR-rt, UBR and CBR traffics. TMS320C5402 DSP is used to process voice-related tasks such as voice compression and voice packet manupulation and AAL2 processor is implemented on $0.35\;{\mu}m$ process line. We implemented the customer premise equipment for VoDSL service and tested the proposed platform on a test bed network. The experimental results show that the proposed equipment has the call success rate of $97\%$ at least and provides voice service of toll-qualify.

• The Journal of the Korea Contents Association
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• v.8 no.5
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• pp.29-37
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• 2008

To transmit video bit stream over low bandwidth such as wireless channel, high compression algorithm like H.264 codec is exploited. In transmitting high compressed video bit-stream over low bandwidth, packet loss causes severe degradation in image quality. In this paper, a new algorithm for recovery of missing or erroneous motion vector is proposed. Considering that the missing or erroneous motion vectors in blocks are closely correlated with those of neighboring blocks. Motion vector of neighboring blocks are clustered according to average linkage algorithm clustering and a representative value for each cluster is determined to obtain the candidate motion vector sets. As a result, simulation results show that the proposed method dramatically improves processing time compared to existing H.264/AVC. Also the proposed method is similar to existing H.264/AVC in terms of visual quality.

• The KIPS Transactions:PartA
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• v.15A no.6
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• pp.317-324
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• 2008

In this paper, we propose a system which easily controls the existing serially connected multi-channel speakers in a general personal computer by using a USB(Universal Serial Bus) interface. The personal computer as a host of the USB interface analyzes a sound source and sends audio data in a real-time fashion by the use of the isochronous transmission, one of four transmission methods provided by the USB interface. In addition, a channel is assigned by means of the bulk transmission, one of four transmission methods provided by the USB interface. Transmitted data from the USB host are sent to each speaker through compression and packet generation process. Each speaker detects corresponding digital data and regenerates audio signals through DAC(Digital-to-Analog Converter). A user can easily select a sound source file and a channel by the use of a GUI environment in a personal computer.