• Title/Summary/Keyword: MPEG 오디오

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Interactive Media Streaming on Mobile Device (이동 단말에서의 인터렉티브 미디어 스트리밍)

  • Ryu, Eun-Seok;Yoo, Chuck
    • Proceedings of the Korean Information Science Society Conference
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    • 2003.04d
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    • pp.61-63
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    • 2003
  • 멀티미디어 기술의 발전은 단일 CD 에 동영상을 저장하기 위한 목적에서 개발되었던 MPEG-1을 비롯하여 최근에는 JVT등의 오디오, 비디오 압축 기술로 이어졌다. 뿐만 아니라 이러한 미디어 데이터를 효과적으로 전달하기 위한 스트리밍 기법, 미디어 데이터 간의 동기화 기법 등 다양한 방면에서 연구되고 있다. 이에 맞춰, 최근에는 사용자와의 상호작용을 위한 User Interaction 에 관한 연구가 활발히 진행되고 있으나 대부분의 연구가 Interaction 자체에 대한 정의 및 이에 대한 기본적인 지원에 관한 연구에 국한되고 있다. 따라서 본 연구에서는 구체적인 성능 향상의 관점에서 interaction 을 효과적으로 제공하기 위한 여러 기법들에 관해 알아보고, 보다 나은 이동 단말에서의 interaction 스트리밍 기법을 제안한다.

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Design of Service Signaling Structure based on MMT for Terrestrial UHD Broadcasting Systems in Heterogeneous Network (이기종망 환경에서의 지상파 UHD 방송을 위한 MMT 기반 서비스 시그널링 구조 설계)

  • Seo, Minjae;Paik, Jong-Ho
    • Journal of Satellite, Information and Communications
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    • v.10 no.2
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    • pp.54-59
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    • 2015
  • UHD(Ultra High Definition) Broadcasting is the focus of attention recent days. However, current broadcasting system faces with difficulty of bandwidth, encoding and so on. To solve this problem, MMT was suggested as the one of solutions. MMT is designed to be based on IP network and it has characteristics that can deliver multimedia through many networks at the same time. UHD Media service can be available with the current broadcasting system that divides media data into hierarchical data based on MMT. To provide this service, information about heterogeneous network should be delivered and signalling should be given to perceive it. For UHD media data service, the information about data from heterogeneous networks should be transported for providing presentation information and service for the receiver models. The present MMT signalling has not much information about heterogeneous services with hierarchical media data. In this paper, we suggest the design of service signaling structure based on MMT for UHD broadcasting systems.

An XMT Authoring System supporting Multiple Presentation Environments (다양한 재생 환경을 지원하는 XMT 저작 시스템)

  • 김희선;임영순
    • Journal of KIISE:Computing Practices and Letters
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    • v.10 no.3
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    • pp.251-258
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    • 2004
  • The XMT standard is MPEG-4 Scene Description of textual format. It can be utilized to edit the audio/video media for broadcasting and develop the user oriented media contents. This paper proposes XMT authoring system that supports exchange among contents in various presentation environment. The XMT authoring system creates two levels of textual syntax and semantics: XMT-$\alpha$ format and XMT-$\Omega$ format. Because XMT-$\alpha$ and XMT-$\Omega$ have different expression method about an object, the authoring tool offers interface for them. the authoring tool offers interface for them. Also, it defines interior data structure that can support two file formats, and offers the function that transforms XMT-$\alpha$ into BIFS and transforms XMT-$\Omega$ into SMIL or XMT-$\alpha$. It offers interoperability among multimedia data in various environment that is XMT's characteristic.

Modeling of Acoustic Echo Canceller Using Subband Adaptive Signal Processing (서브밴드 적응신호처리를 이용한 음향 에코제거기의 모델링)

  • Kim, Chun-Duck;Sim, Dong-Youn;Chung, Ho-Moon;Lee, Jun-Ku;Cha, Kyung-Hwan
    • The Journal of the Acoustical Society of Korea
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    • v.16 no.5
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    • pp.43-49
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    • 1997
  • Generally, echo cancelers of a TV conference system or a audio conference system are to carry out a real time processing in the case of the closed room having long reverberation time because the system requires much time to modify filter coefficients to environmental changes. Therefore this paper proposes a new subband adaptive filtering method using polyphase filter banks of MPEG(Moving Picture Experts Group) audio system to solve the problems. This method divides signal spectra of input and output into several frequency bands, and each band is adaptively filtered by using ES-NLMS (Exponential Step-Normalized Least Mean Square) algorithm. The optimal number of subband is determined by computational simulations. According to the results of simulation, ERLE of the subband model is 2dB smaller than general full band, calculation rate's of the subband model is decreased about 88%.

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The Development of Multimedia Player Platform for Terrestrial Digital Multimedia Broadcasting (DMB) (지상파 이동 멀티미디어방송용 멀티미디어 재생기 개발)

  • 기명석;서정일;강경옥
    • Journal of Broadcast Engineering
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    • v.8 no.4
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    • pp.465-472
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    • 2003
  • In this paper we propose the structure of MPEG-4 multimedia player platform for Terrestrial Digital Multimedia Broadcasting (DMB) Service. Korea will launch DMB service at next 2004 you based on Eureka-147 Digital Audio Broadcasting (DAB) Service System. This new mobile multimedia broadcasting services provide not only high quality digital audio broadcasting services, but also various multimedia data broadcasting services including high quality video. For the sake of MPEG-4 Systems technologies, it will provide an interactive service to users in the near future. Therefore it terminal shall have various functionalities as well as playing audio-visual contents. However there is no precedence standard for such mobile interactive multimedia broadcasting system. Therefore it is very import to provide the multimedia player platform of DMB service for accelerating the development process of commercial terminal and providing a direction of next DMB terminal structure.

Video Adaptation Model for User-Centric Contents Delivery in Mobile Computing (모바일 환경에서 맞춤형 콘텐츠 전달을 위한 비디오 적응성 모델)

  • Kim, Svetlana;Yoon, Yong-Ik
    • The KIPS Transactions:PartA
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    • v.16A no.5
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    • pp.389-394
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    • 2009
  • Lately the usage of multimedia equipment with small LCD displays is rapidly increasing. Although many people use devices like this, videos intended for TV or HDTV are sent to these mobile devices. Therefore cases where it is hard for the user to view the desired scenes are growing more frequent. Currently, most services simply reduce the size of the content to fit the screen when they offer it for mobile devices. However, especially with sports broadcasts, there are many areas that cannot be seen very well because it was simply reduced in size. We therefore consider this weakness and are researching how to let the user choose an area of interest and then sending it to the user in a way that fits the device. In this paper, we address the problem of video delivery and personalization. For the delivered video content, we suggest the UP-SAM User Personalized Context-Aware Service Adaptation Middleware) model that uses the video content description and MPEG-21 multimedia framework.

Design of Service Signaling Structure based on MMT for Terrestrial UHD Broadcasting Systems (MMT 기반 지상파 UHD 방송을 위한 서비스 시그널링 구조 설계)

  • Seo, Min-jae;Yu, Kyung-A;Paik, Jong-Ho
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2014.06a
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    • pp.126-128
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    • 2014
  • 디지털 방송기술의 비약적인 발전으로 기존 HD(High Definition) 화질의 4~16배까지 지원 가능한 UHD(Ultra High Definition) 방송 서비스가 제공되는 새로운 시대를 맞이하게 되었다. UHD 방송은 초고선명 비디오와 22.2 채널 오디오 서비스가 가능한 차세대 실감방송 기술이다. 이러한 UHD 서비스를 제공하기 위해서는 고압축 영상 코덱 기술인 HEVC(High Efficiency Video Coding), OFDM(Othogonal Frequency Division Multiplexing) 기반 대용량 전송기술과 다양한 멀티미디어 부가서비스 가능한 전송 프로토콜이 필수적으로 요구된다. 최근 UHD 방송 전송 프로토콜로 표준화 추진 중인 MMT(MPEG Media Transport)는 이기종망에서 적용할 수 있으며, 양방향 전환이 가능하여 시청자의 요구사항을 실시간으로 반영할 수 있다는 장점을 지닌다. 한편, 지상파 UHD 방송 서비스를 빠르고 효과적으로 수신하기 위해 필요한 정보인 NIT(Network Information Table), RRT(Rating Region Table) 및 SDT(Service Description Table)가 MMT 시그널링 프로토콜에서는 포함되어 있지 않다. 따라서 본 논문에서는 지상파 UHD 방송 서비스의 효과적 수신이 가능한 MMT 기반 지상파 UHD 방송을 위해 NIT, RRT 및 SDT 정보가 포함된 서비스 시그널링 구조을 제안한다.

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A Digital Audio Watermark Using Wavelet Transform and Masking Effect (웨이브릿과 마스킹 효과를 이용한 디지털 오디오 워터마킹)

  • Hwang, Won-Young;Kang, Hwan-Il;Han, Seung-Soo;Kim, Kab-Il;Kang, Hwan-Soo
    • Proceedings of the IEEK Conference
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    • 2003.11b
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    • pp.243-246
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    • 2003
  • In this paper, we propose a new digital audio watermarking technique with the wavelet transform. The watermark is embedded by eliminating unnecessary information of audio signal based on human auditory system (HAS). This algorithm is an audio watermarking method, which does not require any original audio information in watermark extraction process. In this paper, the masking effect is used for audio watermarking, that is, post-tempera] masking effect. We construct the window with the synchronization signal and we extract the best frame in the window by using the zero-crossing rate (ZCR) and the energy of the audio signal. The watermark may be extracted by using the correlation of the watermark signal and the portion of the frame. Experimental results show good robustness against MPEG1-layer3 compression and other common signal processing manipulations. All the attacks are made after the D/A/D conversion.

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Implementation of error-robust software for PC-based T-DMB receivers (오류에 견고한 PC 기반 지상파 DMB 수신기 SW 구현)

  • Kim Kwang Seok;Moon Su Han;Kim Yong Han
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2004.11a
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    • pp.107-111
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    • 2004
  • 본 논문에서는 PC 기반 지상파 DMB(Terrestrial Digital Multimedia Broadcasting, T-DMB) 수신기를 위한 오류에 견고한 SW 구현에 대해 설명한다. 이 수신기는 PC 외부에 T-DMB 신호를 안테나로 수신하여 복조하고 채널 복호하는 프론트 엔드(front-end) 수신 모듈을 이용, USB를 통하여 RS(Reed-Solomon) 부호화된 MPEG-2 TS(Transport Stream) 데이터를 읽어 들여 RS 복호, TS 역다중화, 비디오 보호, 오디오 복호 등의 SW 처리 과정을 거쳐 디스플레이 상에 수신 내용을 표시하게 된다. 수신기가 실용적으로 동작하기 위해서는, 수신된 비트스트림이 예상하지 못한 형태일 때도 오동작하거나 멈춰 서지 않아야 한다. 수신된 비트스트림에는 채널 비트 오류로 인해 훼손된 부분이 있을 수 있으며, 송출 시에 잘못 제작된 비트스트림도 있을 수 있다. 본 논문에서는 우선 RS 복호를 수행하여 정정 가능한 범위까지 오류를 정정하고, 검출되었으나 정정하지 못한 오류 또는 비트스트림 제작 시의 오류로 인한 비트스트림 구문 오류를 검출하여 오류 처리를 수행하며, 이로 인한 화질 손상을 숨기기 위해 프리즈 프레임(freeze-frame)을 수행하도록 하였다.

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Implementation of Internet Terminal using G.729.1 Wideband Speech Codec for Next Generation Network (차세대 통신망을 위한 G.729.1 광대역 음성 코덱을 활용한 인터넷 단말 구현)

  • So, Woon-Seob;Kim, Dae-Young
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.33 no.10B
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    • pp.939-945
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    • 2008
  • Tn this paper we described the process and the results of an implementation of Internet terminal using G.729.1 wideband speech codec for next generation network. For this purpose firstly we chose a high performance RISC application processor having DSP features for speech codec processing and enhanced Multimedia Accelerator(eMMA) function for video codec. In the implementation of this terminal, we used G.729.1 codec recently standardized in ITU-T which is a new scalable speech and audio codec that extends 0.729 speech coding standard. To adopt G.729.1 codec to this terminal we transformed most of the fixed point C codes which require more complexity into assembly codes so as to minimize processing time in the processor. As a result of this work we reduced the execution time of the original C codes about 80% and operated in real time on the terminal. For video we used H.263/MPEG-4 codec which is supported by the eMMA with hardware in the processor. In the SIP call processing test connected to real network we obtained under looms end-to-end delay and 3.8 MOS value measured with PESQ instrument. Besides this terminal operated well with commercial terminals.