• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.06a
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• pp.754-757
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• 2007

In this paper, we present an implementation and it's real-sea test of an underwater acoustic data communication system, which allows the system to reduce complexity and increase robustness in time variant underwater environments. For easy adaptation to complicated and time-varying environments of the ocean, all-digital transmitter and receiver systems were implemented. For frame synchronization the CAZAC sequence was used, and QPSK modulation/ demodulation method with carrier frequency of 25kHz and a bandwidth of 5kHz were applied to generate 10kbps transmission rate including overhead. To improve transmission quality, we used several techniques and algorithms such as adaptive beamforming, adaptive equalizer, and convolution coding/Viterbi decoding. for the verification of the system performance, measurement of BER has been done in a very shallow water with depth of 8m at JangMok, Geoje. During the experiment, image data were successfully transmitted up to about 7.4km.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6A
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• pp.667-667
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• 2004

We must design the buffer algorithm that protects traffic congestion and decreasing throughput at satellite communication network. It is important that buffer algorithm is satisfied with the good performance of transmission packet, responsibility of many connecting traffic and the QOS for connecting character. Old buffer algorithms are not the suitable algorithms when we have the satellite communication network environment. RED buffer algorithm is proposed by Floyd. It has a better performance than old buffer algorithm. But this algorithm is not well adapted a number of connecting TCP packet and changing network, so this algorithm has a bad performance on satellite communication network that is many of connecting user at same time. This paper propose the TARED(Target Adaptive RED). It has a good performance, adaptation and stability on satellite communication network and has not overflow and underflow of the buffer level.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6A
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• pp.666-676
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• 2004

We must design the buffer algorithm that protects traffic congestion and decreasing throughput at satellite communication network. It is important that buffer algorithm is satisfied with the good performance of transmission packet, responsibility of many connecting traffic and the 005 for connecting character. Old buffer algorithms are not the suitable algorithms when we have the satellite communication network environment. RED buffer algorithm is proposed by Floyd. It has a better performance than old buffer algorithm. But this algorithm is not well adapted a number of connecting TCP packet and changing network, so this algorithm has a bad Performance on satellite communication network that is many of connecting user at same time. This paper Propose the TARED(Target Adaptive RED). It has a good performance, adaptation and stability on satellite communication network and has not overflow and underflow of the buffer level.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1618-1626
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• 2013

Virtualization service processes all operation including the data creation, storing, and disposal in a server and transmits processed data as the streaming media form. Therefore, client can use the same environment as the traditional desktop environment without considering the type of device. Virtualization service should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived Quality of Service(QoS) of streaming service. In this paper, we propose a network-adaptive transmission architecture that focuses on guaranteeing QoS requirements for virtualization service. In order to provide those, the proposed architecture have the transmission rate adaptation function based on available bandwidth and the content bit-rate control function based on sender buffer state. Through each function, proposed architecture guarantee the delay bounds and continuity of virtualization contents playback. The simulation results show that proposed network-adaptive transmission architecture provides a improve performance of throughput and transmission delay.

• Journal of Broadcast Engineering
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• v.8 no.1
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• pp.30-36
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• 2003

According to an evolution of image and video compression technologies, most digital images are in the compressed form. Resizing of these compressed images have various applications such as transmission of resized image according to varying bandwidth, content adaptation for display purpose and etc. Discrete Cosine Transform (DCT) is the most popular transformation for image compression. Recently, several researches have been performed to obtain the reconstructed image of original size in the DCT domain after downsampling and upsampling in the DCT domain. Main focus of these researches is to improve quality of the reconstructed image after downsampling and upsampling in the DCT domain In this paper, we present an modified IDCT method to downsize DCT-encoded image. Furthermore, we propose an efficient scheme for image downsampling and upsampling in the DCT domain With these modified IDCT method. The proposed scheme Provides higher PSNR values than the existing schemes In terms of the reconstructed image after halving and doubling in the DCT domain.

• Journal of Broadcast Engineering
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• v.13 no.5
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• pp.696-706
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• 2008

In this paper we show the performance of the network-adaptive high-definition scalable video streaming using QWLAN board with IEEE 802.11e MAC monitoring and control. Realtime collected MAC parameters are used to determine which video data is extracted for the predicted available bandwidth. To achieve performance, extraction through R-D is proposed instead of the standard video packet extraction. It is shown through experiments that streaming video quality can be enhanced by fast adaptation to network conditions by using the proposed method.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.10a
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• pp.527-530
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• 2014

DASH is a well-known streaming technology, which was proposed in 2010 by MPEG and standardized in 2011. Major multimedia contents service providers, including Apple, Microsoft, and Adobe are all using this technology to support their media streaming services. Whenever a new service is requested to the server, the DASH technology helps servicing the multimedia streaming to client by recognizing the capacity of network and by adapting the quality of the multimedia contents. In DASH, the quality of multimedia contents will be automatically lowered to meet the fluctuating network status, when undesirable breaks interrupt the network. In this paper, we classified and analysed the advantages and disadvantages of DASH researches in three aspects: bit-rate measurement method, bandwidth aggregation method; rate adaptation metrics, algorithms and logics; user's experiences and QoE.

• Journal of KIISE:Computer Systems and Theory
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• v.28 no.12
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• pp.660-673
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• 2001

In Video-on-Demand (VoD) servers, disk throughput is an important system design parameter because it is directly related to the number of user requests that can be served simultaneously. In this paper, we propose an efficient periodic request grouping scheme for disk array-based VoD servers that reduces the disk seek time, thus improving the disk throughput of VoD disk arrays. To reduce the disk seek time, the proposed scheme groups the periodic requests that access data blocks stored in adjacent regions into one, and arranges these groups in a pre-determined order (e.g., in left-symmetric or right-symmetric fashion). Our simulation result shows that the proposed scheme reduces the average disk bandwidth required by a single video stream and can serve more user requests than existing schemes. For a data block size of 192KB, the number of simultaneously served user requests is increased by 8% while the average waiting time for a user request is decreased by 20%. We also propose an adaptation technique that conforms the proposed scheme to the user preference changes for video streams.

• The KIPS Transactions:PartB
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• v.18B no.5
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• pp.249-254
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• 2011

FGS (fine granularity scalability) supporting scalability in MPEG-4 Part 2 is a scalable video coding scheme that provides bit-rate adaptation to varying network bandwidth thereby achieving of its optimal video quality. In this paper, we proposed FGS coding scheme which performs one more bit-plane coding for residue signal occured in the enhancement-layer of the basic FGS coding scheme. The experiment evaluated in terms of video quality scalability of the proposed FGS coding scheme by comparing with FGS coding scheme of the MPEG-4 verification model (VM-FGS). The comparison was conducted by analysis of PSNR values of three tested video sequences. The results showed that when using rate control algorithm VM5+, the proposed FGS coding scheme obtained Y, U, V PSNR of 0.4 dB, 9.4 dB, 9 dB averagely higher and when using fixed QP value 17, obtained Y, U, V PSNR of 4.61 dB, 20.21 dB, 16.56 dB averagely higher than the existing VM-FGS. From results, we found that the proposed FGS coding scheme has higher video quality scalability to be able to achieve video quality from minimum to maximum than VM-FGS.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.1
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• pp.24-36
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• 2003

The digital mobile communication will be developed till getting multimedia service in anyone, any where, any time. Theses requiring items are going to be come true via IMT-2000 system. Transmitting signal bandwidth of IMT-2000 system is 3 times as large as IS-95 system. That is mean peak to average of signal is higher than IS-95A system. So we have to design it carefully not to effect in adjacent channel. HPA(High Power Amplifier) located in the end point of system is operated in 1-㏈ compression point(Pl㏈), then it generates 3rd and 5th inter modulation signals. Theses signals affect at adjacent channel and RF signal is distorted by compressed signal which is operated near by Pl㏈ point. Then the most important design factor is how we make HPA having high linearity. Feedback, Pre-distorter and Feed-forward methods are presented to solve theses problems. Feed-forward of these methods is having excellent improving capacity, but composed with complex structure. Generally, Linearity and Efficiency in power amplifier operate in the contrary, then it is difficult for us to find optimal operating point. In this paper we applied algorithm which searches optimal point of linear characteristics, which is key in Power Amplifier, using minimum current point of error amplifier in 1st loop. And we made 2nd loop compose with new structure. We confirmed fabricated LPA is operated by having high linearity and minimum current condition with ACPR of -26 ㏈m max. @ 30㎑ BW in 3.515㎒ and ACLR of 48 ㏈c max@${\pm}$㎒ from 1W to 40W.