• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.5
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• pp.45-53
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• 2007

To recover transmitted signal perfectly at DTV receiver, we have to acquire carrier frequency synchronization to compensate pilot signal which located in wrong position and rotated phase. Also, we need a symbol timing synchronization to compensate sampling timing error. Conventionally, to synchronize symbol timing, we use Gardner's scheme which used in multi-level signal. Gardner's scheme is well known for its sampling the timing error signal from every symbol and it makes easy to detect and keep timing sync in multi-path channel. In this paper, to discuss the problem when the received power level is out of range and we cannot get synchronization information. With this problem, we use 2 step procedures. First, we put a received signal power compensation block before Garder's timing error detector. Second, adaptive loop filter to get a fast synchronization information and averaging loop filter's output value to reduce the amount of jitter after synchronization in PLL(Phased Locked Loop) circuit which is used to get a carrier frequency synchronization and symbol timing synchronization. Using the averaging value, we can estimate offset. Based on offset changing ratio, we can adapt adaptive loop filter to carrier frequency and symbol timing synchronization circuit.

• Journal of KIISE:Information Networking
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• v.37 no.5
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• pp.363-373
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• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• The Journal of the Acoustical Society of Korea
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• v.40 no.5
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• pp.417-427
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• 2021

Long-range communication in deep waters must overcome the low data rate due to limited bandwidth. This paper presents the performance of Multiple-Input-Multiple-Output (MIMO) system to increase the data rate. In MIMO system, communication performance is degraded by crosstalk between users and an adaptive passive Time Reversal Processing (TRP) is widely used to eliminate this. In October 2018, long-range underwater acoustic communication experiment was conducted in deep water (1,000 m ~) off the east of Pohang, South Korea. During the experiment, a vertical line array was utilized and communication signals modulated by binary phase shift keying and quadrature phase shift keying with a symbol rate of 512 sps were transmitted. To generate MIMO communication signals, received signals from ranges of 26 km and 30 km is synthesized. Compared to the conventional passive TRP, the adaptive passive TRP eliminates the crosstalk between users and achieves error-free performance with an increase of output signal-to-noise ratio. Therefore, two users separated by 4 km in range achieves an aggregate data rate of 1,024 symbols/s.

• Journal of KIISE:Computing Practices and Letters
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• v.16 no.12
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• pp.1141-1153
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• 2010

This paper introduces ONLIS(Overlay Multicast Network for Live IPTV Service), a novel overlay multicast network optimized to deliver live broadcast IPTV stream. We analyzed IPTV reference model of ITU-T IPTV standardization group in terms of network and stream delivery from the source networks to the customer networks. Based on the analysis, we divide IPTV reference model into 3 networks; source network, core network and access network, ION(Infrastructure-based Overlay Multicast Network) is employed for the source and core networks and PON(P2P-based Overlay Multicast Network) is applied to the access networks. ION provides an efficient, reliable and stable stream distribution with very negligible delay while PON provides bandwidth efficient and cost effective streaming with a little tolerable delay. The most important challenge in live P2P streaming is to reduce end-to-end delay without sacrificing stream quality. Actually, there is always a trade-off between delay & stream quality in conventional live P2P streaming system. To solve this problem, we propose two approaches. Firstly, we propose DSPT(Distributed Streaming P2P Tree) which takes advantage of combinational overlay multicasting. In DSPT, a peer doesn't fully rely on SP(Supplying Peer) to get the live stream, but it cooperates with its local ANR(Access Network Relay) to reduce delay and improve stream quality. When RP detects bandwidth drop in SP, it immediately switches the connection from SP to ANR and continues to receive stream without any packet loss. DSPT uses distributed P2P streaming technique to let the peer share the stream to the extent of its available bandwidth. This means, if RP can't receive the whole stream from SP due to lack of SP's uploading bandwidth, then it receives only partial stream from SP and the rest from the ANR. The proposed distributed P2P streaming improves P2P networking efficiency.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.5
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• pp.40-52
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• 2010

A mobile ad hoc network is instant and heuristic, and it is also vulnerable and volatile. Since topology and route changes are frequent, no single routing protocol designed for a conventional network performs well. Some protocols suffer from significant performance degradation when the number of nodes increases, or when nodes become highly mobile. In this paper we investigate a way to adaptively select a routing protocol that fits to the real-time network conditions. The first phase of our study is to analyze the performances of two classes of routing protocols under various network scenarios. The second phase consists of constructing a routing protocol selection reference. All nodes continue to monitor the status of neighbor nodes and control packets exchanged. Then, the aggregated information is periodically compared against the protocol selection reference. The selected routing protocol is maintained throughout the network until the network property changes substantially. The performance of the proposed algorithm is verified by a set of computer simulations using the OPNET modeler. The experimental results show that selectively changing routing protocol adaptive to the network conditions greatly improves the efficacy of bandwidth utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.2A
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• pp.114-126
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• 2011

Video streaming over wireless multi-hop networks(WMNs) contains the following challenges from channel fading and variable bandwidth of wireless channel, and it cause degradation of video streaming performance. To overcome the challenges, currently, WMNs can use Forward Error Correction (FEC) mechanism. In WMNs, traditional FEC schemes, E2E-FEC and HbH-FEC, for video streaming are applied, but it has long transmission delay, high computational complexity and inefficient usage of resource. Also, to distinguish network status in streaming path, it has limitation. In this paper, we propose monitoring-based coordination of network-adaptive hop-to-end(H2E) FEC scheme. To enable proposed scheme, we apply a centralized coordinator. The coordinator has observing overall monitoring information and coordinating H2E-FEC mechanism. Main points of H2E-FEC is distinguishing operation range as well as selecting FEC starting node and redundancy from monitored results in coordination. To verify the proposed scheme, we perform extensive experiment over the OMF(Orbit Measurement Framework) and IEEE 802.1la-based multi-hop WMN testbed, and we carry out performance improvement, 17%, from performance comparison by existing FEC scheme.

• Journal of KIISE:Databases
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• v.33 no.1
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• pp.86-101
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• 2006

The bandwidth of channel and the power of the mobile devices are limited on a wireless environment. In this case, data broadcast has become an excellent technique for efficient data dissemination. A significant amount of researches have been done on generating an efficient broadcast program of a set of data items with different access frequencies over multiple wireless broadcast channels as well as single wireless broadcast channel. In this paper, an efficient data allocation method over multiple wireless broadcasting channels is explored. In the traditional approaches, a set of data items are partitioned into a number of channel based on their access probabilities. However, these approaches ignore a variation of access probabilities of data items allocated in each channel. In practice, it is difficult to have many broadcast channels and thus each channel need to broadcast many data items. Therefore, if a set of data items broadcast in each channel have different repetition frequencies based on their access frequencies, it will give much better performance than the traditional approaches. In this paper, we propose an adaptive data allocation technique based on data access probabilities over multiple broadcast channels. Our proposed technique allows the adaptation of repetition frequency of each data item within each channel by taking its access probabilities into at count.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.824-833
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• 2004

Due to the bandwidth and storage limitations, medical images are needed to be compressed before transmission and storage. DICOM (Digital Imaging and Communications in Medicine) specification, which is the medical images standard, provides a mechanism for supporting the use of JPEG still image compression standard. In this paper, we explain a method for compressing medical images by JPEG standard and propose two methods for JPEG compression. First, because medical images differ from natural images in optical feature, we propose a method to design adaptively the quantization table using spectrum analysis. Second, because medical images have higher pixel depth than natural images do, we propose a method to design Huffman table which considers the probability distribution feature of symbols. Therefore, we propose methods to design a quantization table and Huffman table suitable for medical images. Simulation results show the improved performance compared to the quantization table and the adjusted Huffman table of JPEG standard. Proposed methods which are satisfied JPEG Standard, can be applied to PACS (Picture Archiving and Communications System).

• Journal of Computing Science and Engineering
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• v.4 no.1
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• pp.80-95
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• 2010

In order to truly integrate e-Learning system into regular curriculum at a university, mobile access to Learning Management Systems has to be enabled. Mobile devices have the potential to be integrated into the classroom, because they contain unique characteristics such as portability, social interactivity, context sensitivity, connectivity and individuality. Adoption of Learning Management Systems by students is still on the low rate, mostly because of poor usability of existing e-Learning systems. Our initial research has confirmed this hypothesis. Usability issue is rising to the higher level on the mobile platform, because of the mobile devices' limited screen size, input interfaces and bandwidth, and also because of the context of use. Our second hypothesis was that it is wrong to consider a mobile device as a surrogate for desktop or laptop personal computer (PC). By just adopting the existing Learning Management System on mobile devices with adaptive technologies such as Google proxy, we do not acquire the satisfactory results. Usability can prove to be even lower compared to desktop application. One possible solution to the problem could be development of rich client applications for today's mobile devices that would raise the usability to a higher level. We developed a PocketPC prototype application by using user-centered design principles, which we presented as a third alternative in usability research conducted among university students. Results gathered in such a way have confirmed that development of e-Learning system, in order to be widely accepted by students, needs to have the user(student) in the center of development process.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.1A
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• pp.8-18
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• 2011

Visible light communication (VLC) using optical sources which can be simultaneously utilized for illumination and communication is currently an attractive option for wireless personal area network. Improving the data rate in optical wireless communication system is challenging due to the limited bandwidth of the optical sources. In this paper, we design the singular value decomposition (SVD)-based multiplexing multi-input multi-output (MIMO) system to support two data streams in optical wireless channels. In order to improve the spectral efficiency, the rate adaptation using multi-level pulse amplitude modulation (PAM) is applied according to the channel condition and we propose the method to allocate the optical power, the offset and the size of modulation scheme theoretically under the constraints of the nonnegativity of the modulated signals, the aggregate optical power and the bit error rate (BER) requirement. The simulation results show that the proposed allocation method gives the better performance than the method to allocate the optical power equally for each data stream.