• Title/Summary/Keyword: Source Coding

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Get Social and Get Better: How social computing features help open source software projects (소셜 컴퓨팅 요소가 오픈 소스 개발 프로젝트의 성과에 미치는 영향에 대한 연구: 소셜 코딩 플랫폼 Github 사례를 바탕으로)

  • Choi, Joohee;Choi, Junghong;Moon, Jae Yun
    • Journal of the HCI Society of Korea
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    • v.7 no.2
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    • pp.17-24
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    • 2012
  • In this study, we aim to understand how social computing features affect open source project's outcome based on the representative social coding platform, Github (http://github.com). Though there is growing interest regarding the application and effect of employing social computing features, yet empirical evidences related to the subject are still short. To bridge the gap, we conducted our research based on the following research questions: 1) How the system features of social coding platform are classified? 2) How are the use of system features and project performance related to each other? Qualitative and quantitative analysis are performed: The system features of Github are clustered according to their usage in qualitative analysis, and th relation between the feature uses and project outcome is identified by multiple linear regression test. In conclusion, we found that the use of results is also discussed.

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A Study on 8kbps FBD-MPC Method Considering Low Bit Rate (Low Bit Rate을 고려한 8kbps FBD-MPC 방식에 관한 연구)

  • Lee, See-Woo
    • Journal of Digital Convergence
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    • v.12 no.6
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    • pp.271-276
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    • 2014
  • In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and unvoiced consonants in a frame. In this paper, I propose a method of 8kbps Multi-Pulse Speech Coding(FBD-MPC: Frequency Band Division MPC) by using TSIUVC(Transition Segment Including Unvoiced Consonant) searching, extraction and approximation-synthesis method in a frequency domain. I evaluate the 8kbps MPC and FBD-MPC. As a result, SNRseg of FBD-MPC was improved 0.5dB for female voice and 0.2dB for male voice respectively. Compared to the MPC, SNRseg of FBD-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

Research on Open Source Encoding Technology for MPEG Unified Speech and Audio Coding (MPEG 통합 음성/오디오 코덱을 위한 오픈 소스 부호화 기술에 관한 연구)

  • Song, Jeongook;Lee, Joonil;Kang, Hong-Goo
    • Journal of the Institute of Electronics and Information Engineers
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    • v.50 no.1
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    • pp.86-96
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    • 2013
  • Unified Speech and Audio Coding (USAC) is the speech/audio codec with the best quality, approved on Final Draft International Standard (FDIS) at MPEG meeting in 2011. Since MPEG conventionally standardizes only the decoder, it is not easy to study on the encoder technologies. Furthermore, Reference Model(RM) shows extremely poor performance. To solve these problems, the open source project(JAME) proposes the methods to make the improved performance of main encoder technologies in USAC. Especially, this paper introduces the encoder modules: the signal classifier for selective operation between two coders, the psychoacoustic model in frequency domain, and window transition technology. Finally, the results of verification test for FDIS and the performance of Common Encoder are appended.

Adaptive Model-Based Quantization Parameter Decision for Video Rate Control (비디오 비트율 제어를 위한 적응적 모델 기반의 양자화 변수 결정 방법)

  • Kim, Seon-Ki;Ho, Yo-Sung
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.32 no.4C
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    • pp.411-417
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    • 2007
  • The rate control is an essential component in video coding to provide better quality under given coding constraints, such as channel capacity, frame rates, etc. In general, source data cannot be described as a single distribution in a video coding, hence it can cause an exhaustive approximation problem. It drops a coding efficiency under weak channel environments, such as mobile communications. In this paper, we design a new quantization parameter decision model that is based on a rate-distortion function of generalized Gaussian distribution. In order to adaptively express various source data distribution, we decide a shape parameter by observing a ratio of samples, which have a small value. For experiment, the proposed algorithm is implemented into H.264/AVC video codec, and its performance is compared with that of MPEG-2 TM5, H.263 TMN8 rate control algorithm. As shown in simulation results, the proposed algorithm provides an improved quality rather than previous algorithms and generates the number of bits closed to the target bits.

Consideration of CJK Joint Hanja Unicode when is used in AMI/HDB-3 Line Coding (AMI/HDB-3 회선부호화와 한·중·일 한자 유니코드 체계 고찰)

  • Tai, Dong-Zhen;Hong, Wan Pyo
    • The Journal of the Korea institute of electronic communication sciences
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    • v.8 no.7
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    • pp.1011-1015
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    • 2013
  • This paper analyses the violation rate of CJK joint Chines character Unicode to the source code rule. In the paper, Chinese character 150ea in Chinese Unicode which have relatively a higher frequency in use of a character was chosen to study. The frequency rate in use of the 150ea characters is about 50% of the total frequency rate of the Chinese characters. The study was applied the AMI/HDB-3 line coding/scrambling and HDLC protocol, According to the analyses, the number of violated characters were 77ea of 150 ea, frequency rate in use 29%. Therefore, when the violated 77ea characters are replaced to the matched character codes to the source coding rule, the processing rate of the line coder can be improved about 37%.

A Method Enabling Exploitation of Spatial Diversity and Physical Layer Security in an Extreme Case of Source-Wiretapping without a Jamming Beamformer

  • Tran, Truc Thanh;Kong, Hyung Yun
    • Journal of Communications and Networks
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    • v.17 no.5
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    • pp.482-490
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    • 2015
  • This article exploits spatial diversity for jamming to prevent wiretapping in the extreme case in which an eavesdropper is located near the source and a common jamming signal is unavailable. To address this challenge, the jamming signal is allowed to carry a random binary message. Then, it is proposed that the active intermediate node transmits this jamming signal and the decoding of this signal at both source and destination is physically secured as result of using the physical-layer security method. If the source and the destination securely and correctly decode this jamming message, the source transmits another message which is created from combining its information message and the decoded message using the network-coding method. Therefore, this method prevents the transmissions from being eavesdropped upon by the source-wiretapping.

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 Software Vulnerability Analysis System using Learning for Source Code Weakness History (소스코드의 취약점 이력 학습을 이용한 소프트웨어 보안 취약점 분석 시스템)

  • Lee, Kwang-Hyoung;Park, Jae-Pyo
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.18 no.11
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    • pp.46-52
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    • 2017
  • Along with the expansion of areas in which ICT and Internet of Things (IoT) devices are utilized, open source software has recently expanded its scope of applications to include computers, smart phones, and IoT devices. Hence, as the scope of open source software applications has varied, there have been increasing malicious attempts to attack the weaknesses of open source software. In order to address this issue, various secure coding programs have been developed. Nevertheless, numerous vulnerabilities are still left unhandled. This paper provides some methods to handle newly raised weaknesses based on the analysis of histories and patterns of previous open source vulnerabilities. Through this study, we have designed a weaknesses analysis system that utilizes weakness histories and pattern learning, and we tested the performance of the system by implementing a prototype model. For five vulnerability categories, the average vulnerability detection time was shortened by about 1.61 sec, and the average detection accuracy was improved by 44%. This paper can provide help for researchers studying the areas of weaknesses analysis and for developers utilizing secure coding for weaknesses analysis.

Multi-channel Audio Service in a Terrestrial-DMB System Using VSLI-Based Spatial Audio Coding

  • Seo, Jeong-Il;Moon, Han-Gil;Beack, Seung-Kwon;Kang, Kyeong-Ok;Hong, Jae-Keun
    • ETRI Journal
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    • v.27 no.5
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    • pp.635-638
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    • 2005
  • Spatial audio coding (SAC) is an extremely high compact representation of encoded multi-channel audio material. This paper suggests a multi-channel audio service in the terrestrial digital multimedia broadcasting (T-DMB) system using a novel SAC tool, which is called a virtual source location information (VSLI)-based SAC tool. Intensive experiments are presented to evaluate the validity of the proposed VSLI-based SAC tool, and prototypical systems are also presented to demonstrate the reliability of the proposed multi-channel T-DMB system in real applications.

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Joint Channel Coding Based on Principal Component Analysis

  • Hyun, Dong-Il;Lee, Dong-Geum;Park, Young-Cheol;Youn, Dae-Hee;Seo, Jeong-Il
    • ETRI Journal
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    • v.32 no.5
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    • pp.831-834
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    • 2010
  • This paper proposes a new joint channel coding algorithm based on principal component analysis. A conventional joint channel coder using passive downmixing undergoes a reduction of both the primary-to-ambient energy ratio (PAR) of the downmix signal and the panning gain ratio of the primary source. The proposed system preserves the PAR of the downmix signal by using active downmixing which reflects spatial characteristic. The proposed system also improves the accuracy of the panning gain ratio estimation. Computer simulations and subjective listening tests verify the performance of the proposed system.