• 대한공업교육학회지
• /
• v.33 no.2
• /
• pp.287-299
• /
• 2008

HTTP is one of the most widely used protocols of the WWW. Currently it uses TCP as the transport layer protocol to provide reliability. The HTTP uses separate TCP connection for each file request and adds unnecessary head-of-line blocking overhead for the file retrieval. The web application is short sized and affected by the increased handover latency of TCP in wireless environment. SCTP has attractive features such as multi-streaming and multi-homing. SCTP's multi-streaming and multi-homing avoid head-of-line blocking problem of TCP and reduce handover latency of TCP in wired and wireless environment. Mean response time is the important measure in most web application. In this paper, we present the comparison of mean response time between HTTP over SCTP with that of HTTP over TCP in wired and wireless environments using NS-2 simulator. We measured mean response time for varying packet loss rate, bandwidth, RTT, and the number of web objects in wired environment and mean response time and packet loss rate for varying moving speed and region size in wireless environment. Our experimental result shows that SCTP reduces the mean response time of TCP based web traffic.

• Journal of Korea Multimedia Society
• /
• v.13 no.10
• /
• pp.1525-1533
• /
• 2010

With the digital terrestrial television broadcasting transition, terrestrial television broadcasting have required the replacement of retransmission facilities for the analog broadcasting installed in the existing apartment, building, cable TV station, MATV system and so on. In addition, new standards have been enacted for retransmission of the digital television broadcasting in MATV system. To deal with this issue, in this paper, we propose a new 8-VSB remodulator that can retransmit signals of the terrestrial digital television broadcasting. Moreover, we present a standard and the process composition of the 8-VSB remodulator, and an experimental environment configuration for performance evaluation. To achieve this, we have implemented the 8-VSB remodulator with the sequential process components comprised of the RF signal retransmission, the TS stream modulator, the RF signal reception and demodulation. Through the simulation, we analyze the performance standard from the measured data such as spurious and phase noise. And then, we measure SNR and EVM of each attenuation step of the signal obtained by the signal processor and the 8-VSB remodulator with the same retransmission environment and conditions. Experimental results show that both the 8-VSB remodulator and the signal processor can be used as equipment for the retransmission of the terrestrial digital television broadcasting. In addition, the 8-VSB remodulator performed well to improve the transmission efficiency for the digital broadcasting signal, compare to the existing signal processor.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2007.10a
• /
• pp.693-697
• /
• 2007

The SSB modem performs the modulation process which converts the digital voltage level to the audible frequency band signal and the demodulation process which converts reversely the audible frequency signal to the digital voltage level. The modulator and the demodulator are implemented with a single DSP chip. Because of the SSB specific character, the distortion occurs when the frequency is changed. This distortion has no effect on voice communication, but it has an significant effect on data communication. In other words, it is impossible to send data stream with adjacent 2 periods. Therefore, in case of using 2-tone FSK, it is needed to send at least 3 periods to transmit 1 bit. Therefore we implemented the modem using modified phase-delay shift keying to transmit 1 tone signal for high speed transmission. In the 1200[bps] mode, it generates 0, $187{\mu}s$ delay time at 1.3kHz symbol frequency, and in the 2400[bps] mode, 0, $70{\mu}s$, $130{\mu}s$, $200{\mu}s$ delay time at 1.5kHz symbol frequency. Finally, in the maximum 3600[bps] mode, it generates 0, $100{\mu}s$, $160{\mu}s$, $250{\mu}s$ delay time at 2.0kHz symbol frequency. The measured results of the implemented SSB modem shows a good transfer functional characteristic by spectrum analyzer, almost same bandwidth in pass band and 20dB higher SNR comparing the German PACTOR and American CLOVER and in the experimental transmitting test, we verified the transmitted data is received correctly in platform.

• Journal of Broadcast Engineering
• /
• v.18 no.2
• /
• pp.204-214
• /
• 2013

Recently various research have been conducted relating to the implementation of signal processing or communication system by software using the massively parallel processing capability of the GPU. In this work, we focus on reducing software simulation time of 2K/8K FFT in DVB-T by using GPU. we estimate the processing time of the DVB-T system, which is one of the standards for DTV transmission, by CPU. Then we implement the FFT processing by the software using the NVIDIA's massively parallel GPU processor. In this paper we apply stream process method to reduce the overhead for data transfer between CPU and GPU, coalescing method to reduce the global memory access time and data structure design method to maximize the shared memory usage. The results show that our proposed method is approximately 20~30 times as fast as the CPU based FFT processor, and approximately 1.8 times as fast as the CUFFT library (version 2.1) which is provided by the NVIDIA when applied to the DVB-T 2K/8K mode FFT.

• Journal of KIISE:Information Networking
• /
• v.33 no.4
• /
• pp.333-342
• /
• 2006

The cdma2000 lxEV - DO mobile communication system provides broadcast and multicast services (BCMCS) to meet an increasing demand from multimedia data services. The servicing of video streams over a BCMCS network must, however, face a challenge from the unreliable and error-prone nature of the radio channel. The BCMCS network uses Reed-Solomon coding integrated with the MAC protocol for error recovery. We analyze this coding technique and show that it is not effective in the case of slowly moving mobiles. To improve the playback quality of an MPEG-4 FGS video stream, we propose the Hybrid error recovery scheme, which combines Reed-Solomon with ARQ, using slots which are saved by reducing the Reed-Solomon coding overhead. The target packets to be retransmitted are prioritized by a utility function to reduce the packet error rate in the application layer within a fixed retransmission budget. This is achieved by considering of the map of the error control block at each mobile node. The proposed Hybrid error recovery scheme also uses the characteristics of MPEG-4 FGS (fine granularity scalability) to improve the video quality even when conditions are adverse: slow-moving nodes and a high error rate in the physical channel.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.31 no.4A
• /
• pp.421-430
• /
• 2006

In this paper, we propose a practical implementation of network-adaptive HD(high definition) MPEG-2 video streaming with a cross-layered channel monitoring(CLM) over the IEEE 802.11a WLAN(wireless local area network). For wireless channel monitoring, AP(access point) periodically measures the MAC(medium access control) layer transmission information and sends the monitoring information to a streaming server. This makes that the streaming server reacts more quickly as well as efficiently to the fluctuated wireless channel than that of the end-to-end monitoring(E2EM) scheme for the video adaptation. The streaming sewer dynamically performs the priority-based frame dropping to adjust the video sending rate according to the measured wireless channel condition. For this purpose, our streaming system nicely provides frame-based prioritized packetization by using a real-time stream parsing module. Various evaluation results over an IEEE 802.11a WLAM testbed are provided to verify the intended QoS adaptation capability The experimental results show that the proposed system can effectively mitigate the quality degradation of video streaming caused by the fluctuations of time-varying wireless channel condition.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.35 no.3B
• /
• pp.380-389
• /
• 2010

Since the internet is intend to be the best effort service, the system that stream a large amount of high quality medias need a techniques to overcome the network status for implementation. In this paper, we design and implement a method that estimate quickly whether network permits the needed bandwidth of media and a method that control QoS through that. Presented system uses Relative One-Way Delay(ROWD) trend in the case of the former, and leverages temporal encoding among Scalable Video Coding(SVC) that is apt to apply real time comparatively in the case of the latter. The streaming server classifies the medias by real time to several rates and begins transmission from top-level and is reported ROWD trend periodically from the client. In case of the server reported only 'Increase Trend', the sever decides that the current media exceeds the available bandwidth and downgrades the next media level. The system uses probe packet of difference quantity of the target level and the present level for upgrading the media level. In case of the server reported only 'No Increase Trend' by the ROWD trend response of the probe packet from client, the media level is upgraded. The experiment result in a fiber to the home(FTTH) environment shows progress that proposed system adapts faster in change of available bandwidth and shows that quality of service also improves.

• Journal of the Korean Institute of Telematics and Electronics C
• /
• v.36C no.9
• /
• pp.76-84
• /
• 1999

EZW proposed by Shapiro is based on a zerotree constructed in a way that a parent coefficient in a subband is related to four child coefficients in the next finer subband of similar orientation. This fixed treeing based on 1-to-4 parent-child is suitable to exploti hierachical correlations among subbands but not to exploit spatial correlations within a subband. A new treeing by Joo, et al. is suggested to simulatneously exploit those two correlatins by extending parent-child relationship in a flexible way. The flexible treeing leads to increasing the number of symbols and lowering entorpy comparing to the fixed treeing, and therefore a better compression can be resulted. In this paper, we suggest two techniques to suppress the increasing of symbols. First, a probing bit is generated to avoid redundant scan for insignivicant coefficients. Second, since all subbands do not always require the same kind of symbol-set, produced symbols are re-symbolized into binary codes according to a pre-defined procedure. Owing to those techniques, all symbols are generated as binary codes. The binary symbols can be entropy-coded by an adaptive arithmetic coding. Moerover, the binary symbol stream can give comparatively good performances without help of additional entropy coding. Our proposed coding scheme is suggested in two modes: binary coding mode and arithmetic coding mode. We evaluate the effectivenessof our modifications by comparing with the original EZW.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.17 no.11
• /
• pp.1206-1218
• /
• 1992

Token ring systems which control to switch the data stream of networks by passing the token have been widely used to medium access controls in many ring or bus topology LANs. The system could be modeled for analysis as single-server-multi-queue system of the cyclic service method. These concepts could be expanded to multi-token ring systems interconnected with single ring consisting of bridges implemented simply to be stored and transmitted. In the proposal for the performance analysis of the interconnected token ring system, in has been assumed M/G/1 queueing model that frame arrivals are the Poisson process at each station queue and frame sizes are independently and identically distributed. And the average time delays were analyzed mathematically for arbitrary frame transferred from source station to destination area. The time delay of the frame transmission could be explained as the sum of the average time which the token passed from arbitrary position to source station, such as the waiting time in the source station transferring the previous arrival frames, and the propagation time from source station to interdestinated point. These delays were given as the sum of the duration from inner and outer bridge queues, the time delays from inner and outer bridge queues, and the time from outer bridge queue to destination station. These results were investigated by varing parameters effected to total time delays. In the results, those factors to be effected to dominant the total time delays were increased were in the cases of the high arrival rates and the high ration of destination of the other outerring. The system were shown the time delays increased exponentially in spite of the priority service policy. In order to decreasing the number of outerrings and increasing the number of nodes in backbone relatively, so the systems could be decreased the total delay in the interconnected token ring system.

• Journal of the Korea institute for structural maintenance and inspection
• /
• v.8 no.1
• /
• pp.175-181
• /
• 2004

This study is about a section where underground water level occurs at the underground 5m depth by the excavation of the ground, as a stream is adjacent to a excavation section of High Speed Railway ${\bigcirc}{\bigcirc}$ Station construction sections and a reservoir being always full of water is located at the left side of the construction section. Therefore this test is executed for the design and construction of buoyance anchors able to permanently prevent buoyance by the underground water level at working and for the stable construction and permanent smooth maintenance of structures. In this test, bar type anchors are divided according to their length and standard to execute test-anchor test, and In spot test, 9 test-anchors test, proof test to construction process, suitability test and acceptance test are executed 4 times to 9 test-anchors by dividing anchors according to the length of permanent anchor, the outer diameter of bar and boring diameter. Standard motion characteristic centering on load transmission and break mechanism of bar-type anchors for the prevention of buoyance will be showed in the thesis.