• Journal of Broadcast Engineering
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• v.12 no.3
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• pp.203-213
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• 2007

DMB video/audio service was successfully launched. DMB data service is a matter of primary concern now. The TTI(Traffic and Travel Information) service is considered as a killer application of the data service. Tn this paper, we propose a method for design and implementation of the broadcasting system for TTI service. The proposed TTI service system consists of an authoring tool, the DB module, the Transmission module and Monitoring module. This system satisfies the national TTA standards of the Traffic and Travel Information data services for terrestrial DMB; TPEG(Transport Protocol Experts Group) platform. The system was designed to support real-time automatic transmission for CTT and CTT SUM, and non real-time authoring & automatic transmission for POI and SDI, and was focused on the capability to make high-quality contents efficiently and to send them to the data inserter reliably. The performance of the implemented system was proven through the conformance tests with the various commercial receivers. After the continuous upgrade, the system is being used in commercial service.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.8B
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• pp.613-619
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• 2012

The Long Term Evolution (LTE) system is already able to provide a background of variety services for mobile users with multimedia services such as audio, video, and data. In fact, the High Speed Packet Access plus (HSPA+) solution can greatly enhance bit rates on down-link. However, the supporting for multimedia applications with different QoS (Quality of Service) requirements is not devised yet. Hence, in this paper we propose an effective packet scheduling algorithm based on Proportional Fairness (PF) scheduling algorithms for the LTE. In this proposed packet scheduling scheme, we optimized instantaneous user data rates and the traffic class weight which prioritize user's packets. Finally, we evaluated and showed the performance of the proposed scheduling algorithm through simulations of multimedia traffics being transmitted to users over LTE links in a multi-cell environment.

• The KIPS Transactions:PartC
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• v.16C no.1
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• pp.101-108
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• 2009

The data broadcasting service, which is growing remarkably today, provides viewers with useful information as well as high quality video and audio. Service information is a kind of additional data that contains a wide range of information such as channel list and/or program title. Each service information is transmitted in the form of a table. And most standard committees have specified their own table list used for carrying the service information. Consequently, It causes incompatibility among services that each broadcast operators produce because the tables that they use differ from each other. In this paper, we propose a general-purpose service information processing system for an integrated data broadcasting middleware that is compatible with heterogenous broadcasting environments. The system is able to change its target table list dynamically without any code modification. Futhermore, we also adopted a thread pool model for efficient parsing and event dispatching.

• The Journal of the Acoustical Society of Korea
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• v.36 no.1
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• pp.70-77
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• 2017

Bandwidth extension is a technique to improve speech quality, intelligibility and naturalness, extending from the 300 ~ 3,400 Hz narrowband speech to the 50 ~ 7,000 Hz wideband speech. In this paper, an Artificial Bandwidth Extension (ABE) module embedded in the Opus audio decoder is designed using the information of narrowband speech to reduce the computational complexity of LPC (Linear Prediction Coding) and LSF (Line Spectral Frequencies) analysis and the algorithm delay of the ABE module. We proposed a spectral envelope extension method using DBN (Deep Belief Network), one of deep learning techniques, and the proposed scheme produces better extended spectrum than the traditional codebook mapping method.

• Journal of Digital Convergence
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• v.12 no.3
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• pp.269-275
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• 2014

The DAB system was designed to provide CD quality audio and data services for fixed, portable and mobile applications with the required BER below $10^{-4}$. However for the T-DMB system with the video service of MPEG-4 stream, BER should go down $10^{-8}$ by adding FEC blocks which consist of the Reed-Solomon (RS) encoder/decoder and convolutional interleaver/deinterleaver. In this paper we propose two types of turbo coded T-DMB system without altering the puncturing procedure and puncturing vectors defined in the standard T-DMB system for compatibility. One(Type 1) can replace the existing RS code, convolutional interleaver and RCPC code by a turbo code and the other one (Type 2) can substitute the existing RCPC code by a turbo code. Simulation results show that two new turbo coded systems are able to yield considerable performance gain after just 2 iterations. Type 2 system is better than type 1 but the amount of performance improvement is small.

• Journal of the Korea Society of Computer and Information
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• v.15 no.7
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• pp.27-32
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• 2010

We can cut and paste portions of MPEG coded bitstream efficiently to rearrange the audio and video sequences using our proposed method. The proposed method decodes the MPEG stream within just only one GOP(Group of Picture), edits the decoded video frames, and encodes it back to a MPEG stream. In this method, precise editing is possible. A pre-processing step is specially designed to provide easy cut and paste processing. In the pre-processing step for editing MPEG streams, the detail information is extracted. In addition, video quality is not degraded after the proposed editing process is applied. Consequently, the experimental results show significant improvements compared with traditional algorithms for video editing method in terms of the efficiency and exactness.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.4
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• pp.9-16
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• 2011

This paper represents an interworking architecture for keeping the VLC audio quality between Worldwide Interoperability for Microwave Access (WiMAX) and IEEE 802.11 Mobile Ad hoc Network (MANET) where both mobile routers and mobile nodes are moving dynamically. Systematic performance analysis on the interworking architecture has been conducted by using OPNET simulator to show the results such as Packet Delivery Ratio (PDR) and throughput. Based on simulation results, when the number of MANET nodes is small, PDR remains relatively stable even though data packets increase. However, with the many MANET nodes, PDR decreases as data traffic increases. Throughput is affected by the number of MANET nodes. Especially when the MANET node density has increased further, throughput is much higher, but it is not affected by the mobility speed. However, FTP download and upload response time is not affected much by both the number of MANET nodes and the mobility speed.

• The Korean Journal of Rehabilitation Nursing
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• v.5 no.2
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• pp.217-225
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• 2002

Purpose : This research is aimed at developing a substantive theory related to the experience of adjustmented hemodialysis patients by identifying and analyzing the nature, process, and types of adjustment experienced by them. For this purpose, this study takes a grounded theory approach. Method : Data were collected from April, 2002 through September, 2002 through in-depth interviews and close observations of eleven hemodialysis patients who have experienced adjustment. With their consent, the interviews were recorded by audio tapes and later transcribed. Observation memos were also prepared on the subjects' behavior during the interviews. Data collection continued until saturated. The data were analyzed into concepts, subcategories, and categories with the open coding process. The axial coding was done to identify the relationships of the concepts and categories. And the selective coding was done to develop a core category, which is the central phenomenon of the hemodialysis patients who experienced adjustment. Result : This process resulted in 88 concepts, which may be grouped into 24 subcategories and 6 core categories. The 6 categories, in fact, depict the process of changes the patients experience from the sense of crisis, self-control, new life meaning, support system, coping ability, and quality of life improvement. And its core is self-control, which encompasses the dynamic stages from recognition, change attempts, belief solidification, and empowerment. Conclusion : The results of the research offer the following implications. First, research on adjustment is a formative stage in nursing, it is imperative for nursing researchers to develop them, which may be more relevant to nursing. Second, it is important to develop nursing intervention techniques that may be most effective in adjustment of hemodialysis patients and at the same time for each stage of changes taking place in adjustmented hemodialysis patients.

• Journal of KIISE:Computer Systems and Theory
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• v.31 no.9
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• pp.527-535
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• 2004

Handling mixed workload in digital set-top box or streaming server becomes an important issue as integrated file system gets momentum as the choice for the next generation file system. The next generation file system is required to handle real-time audio/video playback while being able to handle text requests such as web page, image file, etc. Legacy file system provides only best effort I/O service and thus cannot properly support the QoS of soft real-time I/O. In this paper, we would like to present our experience in developing the file system which fan guarantee the QoS of multimedia stream. We classify all application I/O requests into two category: periodic I/O and sporadic I/O. The QoS requirement of multimedia stream could be guaranteed by giving a higher priority to periodic requests than sporadic requests. The proto-type file system(Qosfs) is developed on Linux Operating System.

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.4
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• pp.214-219
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• 2009

This paper proposes a watermarking algorithm considering the frequency characteristics of Haegeum sounds for copyright protection of digital Haegeum sound contents. The harmonics of Haegeum sounds commonly have large magnitude values in 1500Hz~2000Hz and 2800Hz~3500Hz so that those bands are selected to embed a watermark. The proposed method computes the FFT (fast Fourier transform) of the original sound signal and embeds the watermark bits generated by PN (pseudo noise) sequence into the harmonics in the selected bands. Furthermore, the proposed method is robust to lowpass filter, bandpass filter, cropping, noise addition, MP3 compression attacks and the maximum BER (bit error rate) is 1.41% after lowpass filter attack. To measure the quality of the watermarked sound, subjective listening test, MUSHRA (multiple stimuli with hidden reference and anchor), was conducted. The mean value of MUSHRA listening test is bigger than 98 and 96.67 for every Haegeum sounds and Korean classical music with Haeguem, respectively.