• Title/Summary/Keyword: MPEG Audio

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Test Stream Generation Method for UHDTV Broadcasting Standard (UHD 방송 표준 검증을 위한 시험 스트림 개발에 관한 연구)

  • Kim, Jaeil;Bae, Sungpo;Yang, Jinyoung;Kwon, Donghyun
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.41 no.7
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    • pp.823-832
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    • 2016
  • This paper presents a generation method of test streams for verifying conformance of an UHD broadcasting receiver including decoders for video and audio as well as parsers for PSIP and closed caption data. The proposed test streams for video/audio signals can evaluate conformance of HEVC, AC-3 and DTS-HD standards. Especially, test streams for HEVC video compression standard can be used for testing syntax compliance and error resilience for a HEVC decoder. Moreover, the proposed test streams for system/program and closed caption can be applied for verifying parsers for PSIP and CEA-708 standards.

The QoS Filtering and Scalable Transmission Scheme of MPEG Data to Adapt Network Bandwidth Variation (통신망 대역폭 변화에 적응하는 MPEG 데이터의 QoS 필터링 기법과 스케일러블 전송 기법)

  • 유우종;김두현;유관종
    • Journal of Korea Multimedia Society
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    • v.3 no.5
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    • pp.479-494
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    • 2000
  • Although the proliferation of real-time multimedia services over the Internet might indicate its successfulness in dealing with heterogeneous environments, it is obvious, on the other hand, that the internet now has to cope with a flood of multimedia data which consumes most of network communication channels due to a great deal of video or audio streams. Therefore, for the purpose of an efficient and appropriate utilization of network resources, it requires to develop and deploy a new scalable transmission technique n consideration of respective network environment and individual clients computing power. Also, we can eliminate the waste effects of storage device and data transmission overhead in that the same video stream duplicated according to QoS. The purpose of this paper is to develop a technology that can adjust the amount of data transmitted as an MPEG video stream according to its given communication bandwidth, and technique that can reflect dynamic bandwidth while playing a video stream. For this purpose, we introduce a media scalable media decomposer working on server side, and a scalable media composer working o n a client side, and then propose a scalable transmission method and a media sender and a media receiver in consideration of dynamic QoS. Those methods proposed her can facilitate an effective use of network resources, and provide multimedia MPEG video services in real-time with respect to individual client computing environment.

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MPEG2-TS to RTP Transformation and Application system (MPEG2-TS의 RTP 변환 및 적용 시스템)

  • Im, Sung-Jin;Kim, Ho-Kyom;Hong, Jin-Woo;Jung, Hoe-Kyung
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2010.10a
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    • pp.643-645
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    • 2010
  • The Internet-based multimedia services such as IPTV is being expanded with the development of technology to support the convergence of broadcasting and telecommunications technology for the control seems to be growing larger. Especially for the real-time TV broadcast multicast control technology to support the authentication and resource control, in addition to the technology services that enhance the value of technology for a variety of services in both directions seems to be developed. And, Internet-based transmission system transmit the video content for the video content delivery using RTP(Real Time Transport Protocol). Standardization body, IETF(Internet Engineering Task Force) within the RTP, according to a variety of audio and video formats only transmission format(RTP Payload Format) Establish a separate standard and scalable video content "RTP Payload Format for SVC(Switched Virtual Connection) Video" the standardization is currently processing. In this paper we are improving the quality of broadcasting and telecommunication systems, so that the upper layer by the application can react adaptively to the existing MPEG2-TS and RTP who are provided by a variety of content applied to a variety of devices consumers ETE(End- to-End) QoS(Quality of Service) for enhance the system who was designed and implemented.

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Content Insertion Technology using Mobile MMT with CMAF (CMAF 기반 Mobile MMT를 활용한 콘텐츠 삽입 기술)

  • Kim, Junsik;Park, Sunghwan;Kim, Doohwan;Kim, Kyuheon
    • Journal of Broadcast Engineering
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    • v.25 no.4
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    • pp.560-568
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    • 2020
  • In recent years, as network technology develops, the usage of streaming services by users is increasing. However, the complexity of streaming services is also increasing due to various terminal environments. Even when streaming the same content, it is necessary to re-encode the content according to the type of service. In order to solve the complexity and latency of the streaming service, Moving Picture Experts Group (MPEG) has standardized the Common Media Application Format (CMAF). In addition, as content transmission using a communication network becomes possible, the Republic of Korea's Ultra High Definition (UHD) broadcasting standard has been enacted as a hybrid standard using a broadcasting network and a communication network. The hybrid service enables various services such as transmitting additional information of contents or providing user-customized contents through a communication network. The Republic of Korea's UHD transmission standard utilizes MPEG Media Transport (MMT), and Mobile MMT is an extension of MMT to provide mobile network-specific functions. This paper proposes a method of inserting CMAF contents suitable for various streaming services using signaling messages of MMT and Mobile MMT. In addition, this paper proposes a model for content insertion system in heterogeneous network environment using broadcasting and communication networks, and verifies the validity of the proposed technology by checking the result of content insertion.

Efficient Multi-way Tree Search Algorithm for Huffman Decoder

  • Cha, Hyungtai;Woo, Kwanghee
    • International Journal of Fuzzy Logic and Intelligent Systems
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    • v.4 no.1
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    • pp.34-39
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    • 2004
  • Huffman coding which has been used in many data compression algorithms is a popular data compression technique used to reduce statistical redundancy of a signal. It has been proposed that the Huffman algorithm can decode efficiently using characteristics of the Huffman tables and patterns of the Huffman codeword. We propose a new Huffman decoding algorithm which used a multi way tree search and present an efficient hardware implementation method. This algorithm has a small logic area and memory space and is optimized for high speed decoding. The proposed Huffman decoding algorithm can be applied for many multimedia systems such as MPEG audio decoder.

Quick Audio Retrieval Using Multiple Featrue Vector (다중 특징 벡터를 이용한 고속 오디오 검색)

  • Ban Ji-hye;Kim Ki-man;Park Kyu-sik
    • Proceedings of the Acoustical Society of Korea Conference
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    • spring
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    • pp.351-354
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    • 2004
  • 최근 MPEG-7 등에서 컨텐츠 내용 기반 검색에 대한 연구가 이루어지고 있다. 내용 기반 검색은 기존의 키워드기반 검색이 아닌 컨텐츠 내의 특징 벡터를 추출하여 이와 일치하는 것을 찾는 작업으로써 차세대 디지털 방송 등에 적응될 예정이다. 본 논문은 긴 오디오 stream에서 찾고자 하는 오디오의 위치를 빨리 찾을 수 있는 고속 검객 방법을 제시한다. 기존의 방법에서는 zero-crossing rate만을 이용하여 검색을 했었으나 본 논문에서는 오디오 신호의 특성을 표현할 수 있는 여러 가지 특징 벡터들을 이용한 고속 검색 방법을 고찰 한다. 본 논문의 가장 중요만 부분은 active search 알고리즘과 히스토그램, 그리고 적절하게 조합된 다중 특징 벡터들을 이용한 오디오 검색의 정확도와 속도를 향상시키는데 있다.

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Audio Transcoding Algorithm for Terrestrial DTV and Terrestrial DMB Systems (지상파 DTV와 지상파 DMB 방송을 위한 오디오 트랜스코딩 알고리듬)

  • Bang Kyoung Ho;Lee Jae Seong;Lee Chang Joon;Park Young Cheol;Seo Jeong Il
    • Proceedings of the Acoustical Society of Korea Conference
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    • autumn
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    • pp.161-164
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    • 2004
  • 본 논문에서는 지상파 DTV 의 저작물을 지상파 DMB 방송에 활용할 수 있는 오디오 트랜스코딩 기법에 대해 제안한다. 지상파 DTV 에서는 오디오 신호를 AC-3 방식으로 압축하는 반면, 지상파 DMB 에서는 MPEG-4 BSAC 방식을 사용한다. 각 알고리듬이 사용하는 주파수 변환 방식과 심리음향모델에 의한 비트할당 기법이라는 유사성을 이용하면, 두 방식간의 트랜스코딩 효율을 향상시킬 수 있다 실시간 변환을 요구하는 경우나 휴대기기를 위한 응용분야에서는 지연시간과 전력소모를 줄일 수 있는 잇점을 갖는다.

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Design of A Downlink Power Control Scheme in Unequal Error Protection Multi-Code CDMA Mobile Medicine System

  • Lin, Chin-Feng;Lee, Hsin-Wang;Hung, Shih-Ii;Li, Ching-Yi
    • Proceedings of the Korea Society of Information Technology Applications Conference
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    • 2005.11a
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    • pp.335-338
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    • 2005
  • In this paper, we propose a downlink power control scheme to apply in the unequal error protection multi-code CDMA mobile medicine system. The mobile medicine system contains (i) blood pressure and body temperature measurement value, (ii) ECG medical signals measured by the electrocardiogram device, (iii) mobile patient's history, (iv) G.729 audio signal, MPEG-4 CCD sensor video signal, and JPEG2000 medical image. By the help of the multi-code CDMA spread spectrum communication system with downlink power control scheme and unequal error protection strategy, it is possible to transmit mobile medicine media and meet the quality of service. Numerical analysis and simulation results show that the system is a well transmission platform in mobile medicine.

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An Audio Coding Technique Employing the Inter-channel Phase Difference Skip (채널 간 위상차 파라미터 생략 기법을 이용한 오디오 부호화)

  • Kim, Hyun-Hwi;Kim, Rin-Chul
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2015.07a
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    • pp.3-4
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    • 2015
  • 본 논문에서는 공간 오디오 부호화 기법인 MPEG 서라운드에서 공간 파라미터 전송 시 위상 파라미터를 생략하는 기법에 대해 다룬다. 기존 방법에서는 한 프레임이 모두 적은 위상차를 가지는 경우에도 정상적으로 처리하여 전송한다. 이러한 경우 위상차 파라미터를 생략하여 비트 효율을 향상시킬 수 있다. 스테레오 복원 과정에서 발생하는 채널 간 시간차에 기반해 설계된 양자화기를 생략 기법에 적용하면 기존에 비해 평균적으로 40 ~ 50% 정도의 위상 파라미터 절감 효과를 얻을 수 있다.

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Complexity Reduction Method for BSAC Decoder

  • Jeong, Gyu-Hyeok;Ahn, Yeong-Uk;Lee, In-Sung
    • ETRI Journal
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    • v.31 no.3
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    • pp.336-338
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    • 2009
  • This letter proposes a complexity reduction method to speed up the noiseless decoding of a bit-sliced arithmetic coding (BSAC) decoder. This scheme fully utilizes the group of consecutive arithmetic-coded symbols known as the decoding band and the significance tree structure sorted in order of significance at every decoding band. With the same audio quality, the proposed method reduces the number of calculations that are performed during the noiseless decoding in BSAC to about 22% of the amount of calculations with the conventional full-search method.

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