• Title/Summary/Keyword: 전자해도 표현 표준

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A Study on The RFID/WSN Integrated system for Ubiquitous Computing Environment (유비쿼터스 컴퓨팅 환경을 위한 RFID/WSN 통합 관리 시스템에 관한 연구)

  • Park, Yong-Min;Lee, Jun-Hyuk
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.49 no.1
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    • pp.31-46
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    • 2012
  • The most critical technology to implement ubiquitous health care is Ubiquitous Sensor Network (USN) technology which makes use of various sensor technologies, processor integration technology, and wireless network technology-Radio Frequency Identification (RFID) and Wireless Sensor Network (WSN)-to easily gather and monitor actual physical environment information from a remote site. With the feature, the USN technology can make the information technology of the existing virtual space expanded to actual environments. However, although the RFID and the WSN have technical similarities and mutual effects, they have been recognized to be studied separately, and sufficient studies have not been conducted on the technical integration of the RFID and the WSN. Therefore, EPCglobal which realized the issue proposed the EPC Sensor Network to efficiently integrate and interoperate the RFID and WSN technologies based on the international standard EPCglobal network. The proposed EPC Sensor Network technology uses the Complex Event Processing method in the middleware to integrate data occurring through the RFID and the WSN in a single environment and to interoperate the events based on the EPCglobal network. However, as the EPC Sensor Network technology continuously performs its operation even in the case that the minimum conditions are not to be met to find complex events in the middleware, its operation cost rises. Moreover, since the technology is based on the EPCglobal network, it can neither perform its operation only for the sake of sensor data, nor connect or interoperate with each information system in which the most important information in the ubiquitous computing environment is saved. Therefore, to address the problems of the existing system, we proposed the design and implementation of USN integration management system. For this, we first proposed an integration system that manages RFID and WSN data based on Session Initiation Protocol (SIP). Secondly, we defined the minimum conditions of the complex events to detect unnecessary complex events in the middleware, and proposed an algorithm that can extract complex events only when the minimum conditions are to be met. To evaluate the performance of the proposed methods we implemented SIP-based integration management system.

ROUTE/DASH-SRD based Point Cloud Content Region Division Transfer and Density Scalability Supporting Method (포인트 클라우드 콘텐츠의 밀도 스케일러빌리티를 지원하는 ROUTE/DASH-SRD 기반 영역 분할 전송 방법)

  • Kim, Doohwan;Park, Seonghwan;Kim, Kyuheon
    • Journal of Broadcast Engineering
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    • v.24 no.5
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    • pp.849-858
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    • 2019
  • Recent developments in computer graphics technology and image processing technology have increased interest in point cloud technology for inputting real space and object information as three-dimensional data. In particular, point cloud technology can accurately provide spatial information, and has attracted a great deal of interest in the field of autonomous vehicles and AR (Augmented Reality)/VR (Virtual Reality). However, in order to provide users with 3D point cloud contents that require more data than conventional 2D images, various technology developments are required. In order to solve these problems, an international standardization organization, MPEG(Moving Picture Experts Group), is in the process of discussing efficient compression and transmission schemes. In this paper, we provide a region division transfer method of 3D point cloud content through extension of existing MPEG-DASH (Dynamic Adaptive Streaming over HTTP)-SRD (Spatial Relationship Description) technology, quality parameters are further defined in the signaling message so that the quality parameters can be selectively determined according to the user's request. We also design a verification platform for ROUTE (Real Time Object Delivery Over Unidirectional Transport)/DASH based heterogeneous network environment and use the results to validate the proposed technology.

Translation of 3D CAD Data to X3D Dataset Maintaining the Product Structure (3차원 CAD 데이터의 제품구조를 포함하는 X3D 기반 데이터로의 변환 기법)

  • Cho, Gui-Mok;Hwang, Jin-Sang;Kim, Young-Kuk
    • The KIPS Transactions:PartA
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    • v.18A no.3
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    • pp.81-92
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    • 2011
  • There has been a number of attempts to apply 3D CAD data created in the design stage of product life cycle to various applications of the other stages in related industries. But, 3D CAD data requires a large amount of computing resources for data processing, and it is not suitable for post applications such as distributed collaboration, marketing tool, or Interactive Electronic Technical Manual because of the design information security problem and the license cost. Therefore, various lightweight visualization formats and application systems have been suggested to overcome these problems. However, most of these lightweight formats are dependent on the companies or organizations which suggested them and cannot be shared with each other. In addition, product structure information is not represented along with the product geometric information. In this paper, we define a dataset called prod-X3D(Enhanced X3D Dataset for Web-based Visualization of 3D CAD Product Model) based on the international standard graphic format, X3D, which can represent the structure information as well as the geometry information of a product, and propose a translation method from 3D CAD data to an prod-X3D.

Transform Skip Mode Decision and Signaling Method for HEVC Screen Content Coding (HEVC 스크린 콘텐츠의 고속 변환 생략 결정 및 변환 생략 시그널링 방법)

  • Lee, Dahee;Yang, Seungha;Shim, HiukJae;Jeon, Byeungwoo
    • Journal of the Institute of Electronics and Information Engineers
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    • v.53 no.6
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    • pp.130-136
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    • 2016
  • HEVC (High Efficiency Video Coding) extension considers screen content as one of its main candidate sources for encoding. Among the tools already included in HEVC version 1, the technique of using transform skip mode allows transform to be skipped and to perform quantization process only. It is known to improve video coding efficiency for screen contents which are characterized to have much high frequency energy. But encoding complexity increases since its encoder should decide whether transform should be used or not in each $4{\times}4$ transform block. Based on statistical correlation between IBC (Intra block copy) and transform skip modes both of which are known effective in screen contents, this paper proposes a combined method of the fast transform skip mode decision and a modified transform skip signaling which signals transform_skip_flag at CU level as a representative transform skip signal. By simulation, the proposed method is shown to reduce encoding time of $4{\times}4$ transform blocks by about 32%.

A Study on the Hangul Syllables of Unicode System considering Data Transmission Efficiency (데이터전송효율을 고려한 유니코드의 한글글자마디에 대한 연구)

  • Hong, Wan-Pyo
    • The Journal of the Korea institute of electronic communication sciences
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    • v.10 no.1
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    • pp.39-46
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    • 2015
  • The paper studied possibility of improvement of efficient of data processing in the line coder when Hangul syllables in Unicode system is used for the source code. The scrambling in the line coder is to solve the problem happened due to the source code. The study is based on the HDB-3 scrambling method in ITU-T standards that is applied to AMI line coder. The referred data of Hangul syllables and its use frequency which are required to analysis was used the data extracted from the source data of the National Korean Language Institute. According to the analysis, the average 24% scrambling was generated in source code of Hangul syllables in Unicode system. When the referred Hangul syllables was applied to Unicode system, the average 27% scrambling was producted. Total 8,924ea Hangul syllables in 11,172ea Hangul syllables in Unicode system were not scrambled. Therefore the referred Hangul syllables 1,540ea were accepted in the unscrambled code areas. As a result, the existing Unicode Hangul syllable codes can't prevent the scrambling, but it is possible to completely remove the 27% scrambling with new source coding system. And then, it can be improved the data processing efficient upto minimum 27% in line coder by software in presentation layer instead of physical layer.

A Comparative Study on the Infinite NHPP Software Reliability Model Following Chi-Square Distribution with Lifetime Distribution Dependent on Degrees of Freedom (수명분포가 자유도에 의존한 카이제곱분포를 따르는 무한고장 NHPP 소프트웨어 신뢰성 모형에 관한 비교연구)

  • Kim, Hee-Cheul;Kim, Jae-Wook
    • The Journal of Korea Institute of Information, Electronics, and Communication Technology
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    • v.10 no.5
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    • pp.372-379
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    • 2017
  • Software reliability factor during the software development process is elementary. Case of the infinite failure NHPP for identifying software failure, the occurrence rates per fault (hazard function) have the characteristic point that is constant, increases and decreases. In this paper, we propose a reliability model using the chi - square distribution which depends on the degree of freedom that represents the application efficiency of software reliability. Algorithm to estimate the parameters used to the maximum likelihood estimator and bisection method, a model selection based on the mean square error (MSE) and coefficient of determination($R^2$), for the sake of the efficient model, were employed. For the reliability model using the proposed degree of freedom of the chi - square distribution, the failure analysis using the actual failure interval data was applied. Fault data analysis is compared with the intensity function using the degree of freedom of the chi - square distribution. For the insurance about the reliability of a data, the Laplace trend test was employed. In this study, the chi-square distribution model depends on the degree of freedom, is also efficient about reliability because have the coefficient of determination is 90% or more, in the ground of the basic model, can used as a applied model. From this paper, the software development designer must be applied life distribution by the applied basic knowledge of the software to confirm failure modes which may be applied.

The Proposal and Implementation of Wireless Smart Sensor Node and NCAP System based on the IEEE 1451 (IEEE 1451 기반의 Wireless Smart Sensor Node와 NCAP 시스템의 제안과 구현)

  • Heo, Jung-Il;Lim, Su-Young;Seo, Jung-Ho;Kim, Woo-Shik
    • Journal of the Institute of Electronics Engineers of Korea CI
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    • v.44 no.5
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    • pp.28-37
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    • 2007
  • IEEE 1451 standard defines an interface for network and transducer. In this paper, We propose an architectural model to configure data acquisition system and wireless smart sensor node based on IEEE 1451 standard. Proposed Network Capable Application Processor(NCAP) supports the task of data acquisition and communication for smart sensor node and network. The NCAP is able to reconfigure without interrupting the functionality of the wireless sensor node and receives the critical information of transducer using the DB. Smart sensor node is able to provide the basic information of sensor in digital format. This digital format is called Transducer Electronic Data Sheet(TEDS), is capable of plug-and-play capability of wireless sensor node and the NCAP. We simplify the format of TEDS and template to apply to wireless network environment. information of TEDS and template is transmitted using ad-hoc routing. This study system uses body temperature sensor and ECG(Electrocardiogram) sensor to provide the medical information service. The format of template is selected by data sheet of the sensor and reconfigured to accurately describe the property of the sensor. DB of NCAP is possible to register new template and information of the property as developing new sensor.

Method of a Multi-mode Low Rate Speech Coder Using a Transient Coding at the Rate of 2.4 kbit/s (전이구간 부호화를 이용한 2.4 kbit/s 다중모드 음성 부호화 방법)

  • Ahn Yeong-uk;Kim Jong-hak;Lee Insung;Kwon Oh-ju;Bae Mun-Kwan
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.42 no.2 s.302
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    • pp.131-142
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    • 2005
  • The low rate speech coders under 4 kbit/s are based on sinusoidal transform coding (STC) or multiband excitation (MBE). Since the harmonic coders are not efficient to reconstruct the transient segments of speech signals such as onsets, offsets, non-periodic signals, etc, the coders do not provide a natural speech quality. This paper proposes method of a efficient transient model :d a multi-mode low rate coder at 2.4 kbit/s that uses harmonic model for the voiced speech, stochastic model for the unvoiced speech and a model using aperiodic pulse location tracking (APPT) for the transient segments, respectively. The APPT utilizes the harmonic model. The proposed method uses different models depending on the characteristics of LPC residual signals. In addition, it can combine synthesized excitation in CELP coding at time domain with that in harmonic coding at frequency domain efficiently. The proposed coder shows a better speech quality than 2.4 kbit/s version of the mixed excitation linear prediction (MELP) coder that is a U.S. Federal Standard for speech coder.

Extraction of Renal Glomeruli Region using Genetic Algorithm (유전적 알고리듬을 이용한 신장 사구체 영역의 추출)

  • Kim, Eung-Kyeu
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.46 no.2
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    • pp.30-39
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    • 2009
  • Extraction of glomeruli region plays a very important role for diagnosing nephritis automatically. However, it is not easy to extract glomeruli region correctly because the difference between glomeruli region and other region is not obvious, simultaneously unevennesses that is brought in the sampling process and in the imaging process. In this study, a new method for extracting renal glomeruli region using genetic algorithm is proposed. The first, low and high resolution images are obtained by using Laplacian-Gaussian filter with ${\sigma}=2.1$ and ${\sigma}=1.8$, then, binary images by setting the threshold value to zero are obtained. And then border edge is detected from low resolution images, the border of glomeruli is expressed by a closed B-splines' curve line. The parameters that decide the closed curve line with this low resolution image prevent the noises and the border lines from breaking off in the middle by searching using genetic algorithm. Next, in order to obtain more precise border edges of glomeruli, the number of node points is increased and corrected in order from eight to sixteen and thirty two from high resolution images. Finally, the validity of this proposed method is shown to be effective by applying to the real images.

Nonnegative Matrix Factorization Based Direction-of-Arrival Estimation of Multiple Sound Sources Using Dual Microphone Array (이중 마이크로폰을 이용한 비음수 행렬분해 기반 다중음원 도래각 예측)

  • Jeon, Kwang Myung;Kim, Hong Kook;Yu, Seung Woo
    • Journal of the Institute of Electronics and Information Engineers
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    • v.54 no.2
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    • pp.123-129
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    • 2017
  • This paper proposes a new nonnegative matrix factorization (NMF) based direction-of-arrival (DOA) estimation method for multiple sound sources using a dual microphone array. First of all, sound signals coming from the dual microphone array are segmented into consecutive analysis frames, and a steered-response power phase transform (SRP-PHAT) beamformer is applied to each frame so that stereo signals of each frame are represented in a time-direction domain. The time-direction outputs of SRP-PHAT are stored for a pre-defined number of frames, which is referred to as a time-direction block. Next, In order to estimate DOAs robust to noise, each time-direction block is normalized along the time by using a block subtraction technique. After that, an unsupervised NMF method is applied to the normalized time-direction block in order to cluster the directions of each sound source in a multiple sound source environments. In particular, the activation and basis matrices are used to estimate the number of sound sources and their DOAs, respectively. The DOA estimation performance of the proposed method is evaluated by measuring a mean absolute error (MAE) and the standard deviation of errors between the oracle and estimated DOAs under a three source condition, where the sources are located in [$-35{\circ}$, 5m], [$12{\circ}$, 4m], and [$38{\circ}$, 4.m] from the dual microphone array. It is shown from the experiment that the proposed method could relatively reduce MAE by 56.83%, compared to a conventional SRP-PHAT based DOA estimation method.