• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.4B
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• pp.604-612
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• 2000

In order to satisfy the desired QoS level associated with each existing connection, ATM networks require traffic policing during a connection. Users who respect the contract should receive the function of transparent traffic policing without any interruption. However, contract violations should be detected and mediated immediately. So we propose a CAM type policing controller to allow user cell streams to minimize additional delay. The proposed policing scheme controls policing actions including traffic shaping by suitably spacing cells on each virtual circuit. This policing action is based on parallel processing of multiple cell stream which arrive in ATM multiplexed virtual circuits. We have developed an analytical model of the proposed policing scheme to examine the amount of cell loss and delay, which depends on traffic load, the size of policing buffers and minimum spacing cell time.

• 한국HCI학회:학술대회논문집
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• 2006.02a
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• pp.1093-1098
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• 2006

VR 엔진은 일부 입력장치에 대해서만 제한적으로 지원하기 때문에, 개발자가 원하는 입력장치를 사용하지 못하는 경우가 있으며, 가격 또한 고가이기 때문에 특수한 입력장치를 사용하기 위해, 다른 VR 엔진이나 별도의 옵션을 구매하기에는 경제적인 부담이 많이 든다. 이러한 문제를 해결하기 위해 본 논문에서는 개발자가 사용하고자 하는 입력장치와 VR 엔진의 호환을 위한 랩퍼 클래스를 제안한다. 개발한 랩퍼 클래스는 VR 엔진에서 조이스틱을 제어할 수 있는 조이스틱 클래스와 USB 캠을 통하여 영상을 획득하기 위한 USB 캠 클래스이다. 조이스틱 클래스는 입력장치 클래스를 상속받은 후 DirectX 를 이용하여 입력장치를 셋업 하고, 입력장치의 데이터 값을 처리한 후 VR 엔진의 API 로 값을 넘겨주기 전에 후킹하여 조이스틱을 제어할 수 있다. USB 캠 클래스는 VFW(Video for Window)를 사용하여 캠의 영상을 획득하여 버퍼에 저장한 후 VR 엔진의 디스플레이 버퍼에 값을 넘겨서 캠의 영상을 VR 엔진에서 디스플레이 할 수 있다. 이러한 방법을 통해 조이스틱, USB 캠 같은 입력장치를 VR 엔진과 호환할 수 있으며, 다른 종류의 입력장치에 대하여서도 본 연구에서 개발한 랩퍼 클래스를 상속받아 사용할 수 있다. 본 논문에서 사용한 VR 엔진은 Vega Prime 엔진이며, Vega Prime 엔진의 API 에 개발한 랩퍼 클래스를 추가하여 드라이빙, 영상인식 시뮬레이터를 개발한 결과, 효과적이고 경제적으로 입력장치의 연동이 가능함을 확인할 수 있었다.

• Journal of IKEEE
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• v.13 no.4
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• pp.9-14
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• 2009

This paper describes measurement of an on-ship power supply noise in mixed-signal integrated circuits. To measure the on-chip power supply noise, we can check the effects of analog circuits and compensate it. This circuit consists of two independent measurement channels, each consisting of a sample and hold circuit and a frequency to digital converter which has a buffer and voltage controlled oscillator(VCO). The time-based voltage information and frequency-based power spectrum density(PSD) can be achieved by a simple analog to digital conversion scheme. The buffer works like a unit-gain buffer with a wide bandwidth and VCO has a high gain to improve resolution. This circuit was fabricated in a 0.18um CMOS technology and has 2.06mV/count. The noise measurement circuit consumes 15mW and occupies $0.768mm^2$.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7B
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• pp.1322-1330
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• 1999

The major source of errors in high-speed networks such as Broadband ISDN(B-ISDN) is buffer overflow during congested conditions. These congestion errors are the dominant sources of errors in high-speed networks and result in cell losses. Conventional communication protocols use error detection and retransmission to deal with lost packets and transmission errors. As an alternative, we have presented a method to recover consecutive cell losses using forward error correction(FEC) in ATM(Asynchronous Transfer Mode) networks to reduce the problem. The method finds the lost cells by observing new cell sequence number($SN^{\ast}$). We have used the LI field together with SN and ST fields to consider the $SN^{\ast}$ which provides more correcting coverage than SN in ATM standards. The $SN^{\ast}$ based on the additive way such as the addition of LI capacity to original SN capacity is numbered a repeatedly 0-to-80 cycle. Another extension can be based on the multiplicative way such that LI capacity is multiplied by SN capacity. The multiplicative $SN^{\ast}$ is numbered in a repeatedly 0-to-1025 cycle.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.27 no.8
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• pp.717-724
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• 2016

This paper presents design of a 40 GHz CMOS PLL frequency synthesizer for a 60 GHz sliding-IF RF transceiver. For stable locking over a wide bandwith for a injection-locked frequency divider, an inductive-peaking technique is employed so that it ensures the PLL can safely lock across the very wide tuning range of the VCO. Also, Injection-locked type LC-buffer with low-phase noise and low-power consumption is added in between the VCO and ILFD so that it can block any undesirable interaction and performance degradation between VCO and ILFD. The PLL is designed in 65 nm CMOS precess. It covers from 37.9 to 45.3 GHz of the output frequency. and its power consumption is 74 mA from 1.2 V power supply.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.3 s.345
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• pp.126-134
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• 2006

Traditional TCP implementations have the under-utilization problem in large bandwidth delay product networks especially during the startup phase. In this paper, we propose a delay-based congestion control(DCC) mechanism to solve the problem. DCC is subdivided into linear and exponential growth phases. When there is no queueing delay, the congestion window grows exponentially during the congestion avoidance period. Otherwise, it maintains linear increase of congestion window similar to the legacy TCP congestion avoidance algorithm. The exponential increase phase such as the slow-start period in the legacy TCP can cause serious performance degradation by packet losses in case the buffer size is insufficient for the bandwidth-delay product, even though there is sufficient bandwidth. Thus, the DCC uses the RTT(Round Trip Time) status and the estimated queue size to prevent packet losses due to excessive transmission during the exponential growth phase. The simulation results show that the DCC algorithm significantly improves the TCP startup time and the throughput performance of TCP in large bandwidth delay product networks.

• The Journal of the Korea Contents Association
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• v.10 no.3
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• pp.80-91
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• 2010

Due to the convergence of broadcasting and communications, IPTV services are spotlighted as the that next-generation multimedia services. IPTV services should have functionality such as unlimited channel capacity, extension of media, QoS awareness and are required increasing traffic and quality control technology to adapt the attributes of IPTV service. Consequently, flow based quality control techniques are needed. Therefore, many studies for providing Internet QoS are performed at IETF (Internet Engineering Task Force). As the buffer management mechanism among IP QoS methods, active queue management method such as RED(Random Early Detection) and modified RED algorithms have proposed. However, these algorithms have difficulties to satisfy the requirements of various Internet user QoS. Therefore, in this paper we propose the Flow based AQM(Active Queue Management) algorithm for the multimedia services that request various QoS requirements. The proposed algorithm can converge the packet loss ratio to the target packet loss ratio of required QoS requirements. And we present a performance evaluation by the simulations using the ns-2.

• Journal of Korea Multimedia Society
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• v.7 no.1
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• pp.54-63
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• 2004

This paper describes a hybrid QoS guarantee scheme for high quality audio streaming services on the Internet. The continuous playback of the audio data requires the isochronous transmission of the audio data packet through the Internet. In order to retain the QoS at the ultimate destination (client) as the same as servers provide, the transmission protocols should consider the error conditions such as packet loss, and out of order delivery. Generally, the protocols supporting the transmission of continuous media data do not try to recover the errors. The protocols are working somehow for the toll quality multimedia streaming services, but rot for the high quality streaming services, such as the DVD sound/music payback. The hybrid QoS guarantee scheme includes the three mechanisms to overcome the problem. The selective retransmission for the lost packet, the adaptive buffering at client-side, and the adaptive transmission rate at server-side are totally adopted to recover the packet loss with the minimal overhead, to prevent from the buffer starvation during the retransmission, and to maintain the isochronous transmission even after the retransmission. The experiments have shown good results for the high Quality audio streaming services on the Internet.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.5B
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• pp.413-420
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• 2006

HMIPv6 is designed to reduce the signaling load to external network and improve handover speed of MN by including Mobility Anchor Point(MAP) in local handover. However in this case of macro handover, it's just used pervious MIPv6 handover algorithm. So, it occurs packet loss and transmission delay problem. In this paper, we propose the mechanism applying the HMIPv6 for Fast Handover to choose suitable to the condition buffering handover. The condition for the selection is result distance measurement between MN and CN, between MN and NAR. Furthermore, using F-SNOOP protocol, it is possible to improve wireless network performance. Wireless network has high Bit Error Rate(BER) characteristic because of path loss, fading, noise and interference. TCP regards such errors as congestion and starts congestion control. This congestion control makes packet transmission rate low. However, F-SNOOP improves TCP performance based on SNOOP and Freeze TCP that use Zero Window Advertisement(ZWA) message when handoff occurs in wireless network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5B
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• pp.448-456
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• 2003

In this parer, an ABR (Available Bit Rate) congestion control algorithm for voice transmission in ATM networks was proposed. To deal with the network congestion problem, not only the buffer level of a switch but also the variation of the buffer level were considered. Also, to resolve the unfairness among sources where the bit transfer rates vary, a loading factor that is used to adjust the bit rate was introduced. To show the superiority of this paper over others, simulation was done with a network of 7 voice sources and 4 switches, which was represented by Petri net model. ExSpect was used for simulation. The simulation results showed that there was improvement in network utilization and that unfairness among sources were resolved a lot.