• 한국정보통신학회논문지
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• 제10권10호
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• pp.1914-1919
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• 2006

모바일에서의 영상전송의 문제점은 전송속도와 제어이다. 영상을 전송하기 위해서는 압축기법이 필요한데 압축을 위해서 H.263 코덱을 사용하였으며, 원격지에서 카메라를 효율적으로 제어하고, 실시간 접속 사용자 수를 증가시켰다. 본 연구에서는 통해 기존의 RF를 이용한 영상 전송시의 거리 제한 문제를 해결하였고, 최적의 영상과 오디오를 전송할 수 있었다.

• 방송과미디어
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• 제7권3호
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• pp.75-82
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• 2002

본 연구에서는 ITU-T와 MPEG에서 공동개발중인 JVT압축/복원방식에 대한 복잡도 분석이 수행된다. 압축률을 향상시키는 측면U떠 부호화 할 때 3가지의 configuration을 설정하여 각각의 configuration이 부호기와 복호기 구현시 필요한 메모리 및 계산시간에 대한 분석을 수행한다. 복잡도분석을 위하여 ATOMIUM 분석기를 이용한 메모리 대역폭 및 처리시간의 2가지 요인에 대한 결과를 소개한다. 또한 부호기 및 복호기의 각 비디오 툴들에 대한 상대적인 계산량을 정량적으로 도시한다.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2005년도 학술대회 논문집 정보 및 제어부문
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• pp.532-535
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• 2005

These days, many image processing techniques have been studied for effective image compression. Among those, 2D image filtering is widely used for 2D image processing. The 2D image filtering can be implemented by performing ID linear filtering separately in the direction of horizontal and vertical. Efficiency of image compression depends on what filtering method is used. Generally, circular convolution is widely used in the 2D image filtering for image processing. However it doesn't consider correlations at the region of image boundary, therefore filtering can not be performed effectively. To solve this problem. I proposed new convolution technique using Symmetric-Mirroring convolution, satisfying the 'alias-free' and 'error-free' requirement in the reconstructed image. This method could provide more effective performance than former compression methods. Because it used very high correlative data when performed at the boundary region. In this paper, pre-processing filtering in H.264 codec was adopted to analyze efficiency of proposed filtering technique, and the simulator developed by Matlab language was used to examine the performance of the proposed method.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2008년도 하계종합학술대회
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• pp.401-402
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• 2008

This paper presents an efficient architecture of unified transform and quantization circuit for H.264/JPEG CODEC. The proposed unified transform circuit shares adders required for all transform operations. The proposed unified quantization circuit uses four multipliers. Our transform circuit and quantization circuit consist of 33,711 gates and 9,650 gates respectively. The maximum operating frequency is 100MHz with 130nm standard cells.

• 대한전자공학회논문지TC
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• 제42권11호
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• pp.13-20
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• 2005

현대의 디지털통신기기나, 오디오/비디오 전자기기엔 항상 비바이나리 에러정정복부호기가 사용되는데 그중 필수적으로 사용되는 Reed-Solomon 복부호화기기의 설계에 대해 기술했다. 2,3 symbol RS 복호기설계법을 설명 후, 새로운 RS 부호화기의 설계법을 제시한다. 각각의 복부호화기기의 동작여부를 예를들어 test해보고 잘 동작함을 확인했다.

• 한국정보과학회:학술대회논문집
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• 한국정보과학회 1999년도 가을 학술발표논문집 Vol.26 No.2 (3)
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• pp.12-14
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• 1999

H.263+[1]는 1996년에 비디오압축 표준안으로 확정된 H.263의 확장으로 현재 널리 퍼져 있는 저 전송률의 가정용 전화선을 위한 비디오 압축에 관한 표준안으로 1988년에 확정되었다. H.263+의 기본 알고리즘은 H.263과 같으나 사용상의 편의를 위한 여러 선택모드와 이종의 네트워크환경에서 발생하는 대역폭의 변화나 에러를 위한 scalability(계층부호화)와 같은 새로운 기능이 추가되었다. 새로운 표준안에 따른 실시간 비디오 전송을 처리하기 위해서 필연적으로 하드웨어 코덱의 개발 필요성이 대두되고 있으며 실시간 비디오 코덱은 영상회의 전화기나 멀티미디어 전송시스템의 핵심기술로써 적용될 수 있다. 본 논문에서는 ITU-T H.263+의 기본모드와 새로이 추가된 선택모드 일부를 지원하는 코덱을 하드웨어 기술 언어인 VHDL(VHSIC Hardware Description Language)을 사용하여 기술하고 각 모듈과 최상의 모듈을 시뮬레이션하여 동작을 검증했다.

• 정보와 통신
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• 제25권12호
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• pp.10-18
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• 2008

In this paper, we present a low-power design of H.264 codec based on dedicated hardware and software solution on EMP(ETRI Multi-core platform). The dedicated hardware scheme has reducing computation using motion estimation skip and reducing memory access for motion estimation. The design reduces data transfer load to 66% compared to conventional method. The gate count of H.264 encoder and the performance is about 455k and 43Mhz@30fps with D1(720x480) for H.264 encoder. The software solution is with ASIP(Application Specific Instruction Processor) that it is SIMD(Single Instruction Multiple Data), Dual Issue VLIW(Very Long Instruction Word) core, specified register file for SIMD, internal memory and data memory access for memory controller, 6 step pipeline, and 32 bits bus width. Performance and gate count is 400MHz@30fps with CIF(Common Intermediated format) and about 100k per core for H.264 decoder.

• 디지털산업정보학회논문지
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• 제4권1호
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• pp.41-46
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• 2008

With widespread use of the Internet and improvements in streaming media and compression technology, digital music, video, and image can be distributed instantaneously across the Internet to end-users. However, most conventional Digital Right Management are often not secure and fast enough to process the vast amount of data generated by the multimedia applications to meet the real-time constraints. In this paper, we propose the copyright protection using encryption of DCT coefficients and motion vector in MPEG-4 video codec of mobile device. This paper presents a new Digital Rights Management that modifies the Motion Vector of Macroblock for mobile device. Experimental results indicate that the proposed DRM can not only achieve very low cost of the encryption but also enable separable authentication to individual mobile devices such as Portable Multimedia Player and Personal Digital Assistants. The performance of the proposed methods have low complexity and low increase of bit rate in overhead.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 1998년도 Proceedings of International Workshop on Advanced Image Technology
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• pp.177-182
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• 1998

In this paper, we propose a lossless motion video coding method using motion compensation. For realtime coding and transmission, we developed a lossless video codec based on the proposed method. It was confirmed the codec functions normally in an experiment involving coding and transmitting through an actual ATM network. Furthermore, we proposed a lossless coding method with bit-plane expansion under a constant bitrate. Two approaches, a closed bit-plane approach and a merged bit-plane approach, are considered and characteristics of their compression efficiency are estimated. Simulation results show that the proposed method is suitable for lossless video coding with bitrate control.

• 한국시뮬레이션학회논문지
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• 제10권2호
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• pp.15-24
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• 2001

In this paper we study the delay patterns of voice traffic for end-to-end users Caused by serving the whole bunch of applications traffic at the same time on the Voice over Internet Protocol (VoIP) network. Given the current situation that voice traffic is served along with other application services on the VoIP network, it is quite necessary to figure out how and by what the voice traffic requiring high QoS is delayed. We compare the delay performance of voice traffic on the VoIP network under FIFO with the one under Weighted Fair Queuing(WFQ), and discover the differences of the delay performance resulting from the use of different voice codec algorithms. The results of our study show that using the voice codec algorithm with a higher coding rate nd the queuing algorithm of WEQ can provide users with high-quality voice traffic.