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Labeling network applicaion study policy settings for optimized transmission of multimedia internet (멀티미디어 인터넷망의 최적화 전송을 위한 라벨링망 응용 정책설정 고찰)

  • Gu, Hyun-Sil;Hwang, Seong-kyu
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.19 no.8
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    • pp.1780-1784
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    • 2015
  • Traditional IP routing, see only the Destination Address When Forwarding Layer 3 routing and exchange information and Destination-Based Routing Lookup is required for all Hop. Thus, all routers Full Internet routing information, the route information of more than about 120,000 may require. Therefore, the router configuration, which can be dispersed in the environment, the traffic load is required in accordance with this congestion. In this study, a unique characteristic of the Internet in the environment of an existing network Best Effect for QoS guarantee and hardware high speed switching of large multimedia data transmitted using a Labeling for forwarding a packet environment configuration is required. Video Stream Broadcast Transport Labeling rather than in much of the higher performance of the multi-step policy to most of the Video Stream Packet deulim was fixed to Labeling Header Format proposes a method of applying an effective QoS policy to a more simplified policy.

A Bitrate Control considering Interframe Variance of Image for H.264/AVC (화면간 영상 변화량을 고려한 H.264/AVC 비트율 제어 방법)

  • Son Nam-Rye;Lee Guee-Sang
    • The KIPS Transactions:PartB
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    • v.13B no.3 s.106
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    • pp.245-254
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    • 2006
  • In this work, a new rate control algorithm for transmission of H.264/AVC video bit stream through CBR(constant bit rate) channel is proposed. The proposed algorithm predicts target bit rate and MAD(mean of absolute difference) for current frame considering image complexity variance between neighboring backward and current images. In details, respective linear regression analysis for MAD and encoded bit rate against image complexity variance produce correlation parameters. Additionally, it uses frame skip technique to maintain bit stream within a manageable range and protect buffer from overflow or underflow. Implementation and experimental results show that the proposed algorithm can provide accurate bit allocation, and can effectively visual degradation after scene changes. Also our proposed algorithm encodes the video sequences with less frame skipping compared to the existing rate control for H.264/AVC.

A Design and Implementation of the Real-Time MPEG-1 Audio Encoder (실시간 MPEG-1 오디오 인코더의 설계 및 구현)

  • 전기용;이동호;조성호
    • Journal of Broadcast Engineering
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    • v.2 no.1
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    • pp.8-15
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    • 1997
  • In this paper, a real-time operating Motion Picture Experts Group-1 (MPEG-1) audio encoder system is implemented using a TMS320C31 Digital Signal Processor (DSP) chip. The basic operation of the MPEG-1 audio encoder algorithm based on audio layer-2 and psychoacoustic model-1 is first verified by C-language. It is then realized using the Texas Instruments (Tl) assembly in order to reduce the overall execution time. Finally, the actual BSP circuit board for the encoder system is designed and implemented. In the system, the side-modules such as the analog-to-digital converter (ADC) control, the input/output (I/O) control, the bit-stream transmission from the DSP board to the PC and so on, are utilized with a field programmable gate array (FPGA) using very high speed hardware description language (VHDL) codes. The complete encoder system is able to process the stereo audio signal in real-time at the sampling frequency 48 kHz, and produces the encoded bit-stream with the bit-rate 192 kbps. The real-time operation capability of the encoder system and the good quality of the decoded sound are also confirmed using various types of actual stereo audio signals.

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Design and Implementation of Distributed Object Framework Supporting Audio/Video Streaming (오디오/비디오 스트리밍을 지원하는 분산 객체 프레임 워크 설계 및 구현)

  • Ban, Deok-Hun;Kim, Dong-Seong;Park, Yeon-Sang;Lee, Heon-Ju
    • Journal of KIISE:Computing Practices and Letters
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    • v.5 no.4
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    • pp.440-448
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    • 1999
  • 본 논문은 객체지향형 분산처리 환경 하에서 오디오나 비디오 등과 같은 실시간(real-time) 스트림(stream) 데이타를 처리하는 데 필요한 소프트웨어 기반구조를 설계하고 구현한 내용을 기술한다. 본 논문에서 제시한 DAViS(Distributed Object Framework supporting Audio/Video Streaming)는, 오디오/비디오 데이타의 처리와 관련된 여러 소프트웨어 구성요소들을 분산객체로 추상화하고, 그 객체들간의 제어정보 교환경로와 오디오/비디오 데이타 전송경로를 서로 분리하여 처리한다. 분산응용프로그램 작성자는 DAViS에서 제공하는 서비스들을 이용하여, 기존의 분산프로그래밍 환경이 제공하는 것과 동일한 수준에서 오디오/비디오 데이타에 대한 처리를 표현할 수 있다. DAViS는, 새로운 형식의 오디오/비디오 데이타를 처리하는 부분을 손쉽게 통합하고, 하부 네트워크의 전송기술이나 컴퓨터시스템 관련 기술의 진보를 신속하고 자연스럽게 수용할 수 있도록 하는 유연한 구조를 가지고 있다. Abstract This paper describes the design and implementation of software framework which supports the processing of real-time stream data like audio and video in distributed object-oriented computing environment. DAViS(Distributed Object Framework supporting Audio/Video Streaming), proposed in this paper, abstracts software components concerning the processing of audio/video data as distributed objects and separates the transmission path of data between them from that of control information. Based on DAViS, distributed applications can be written in the same abstract level as is provided by the existing distributed environment in handling audio/video data. DAViS has a flexible internal structure enough to easily incorporate new types of audio/video data and to rapidly accommodate the progress of underlying network and computer system technology with very little modifications.

IEEE 802.22 WG에서의 CR응용: WRAN MAC설계

  • Go Gwang-Jin;Hwang Seong-Hyeon;Song Myeong-Seon;Kim Chang-Ju;Gang Beop-Ju
    • The Proceeding of the Korean Institute of Electromagnetic Engineering and Science
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    • v.17 no.2 s.58
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    • pp.38-49
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    • 2006
  • In order to increase the spectrum efficiency, recently, there is the number of studies for CR technology. For instance, IEEE 802.22 WRAN(Wireless Regional Area Network) WG considered the CR technology as a solution of WRAN system to serve the high speed internet service(1.5 Mbps down stream and 384 kbps up stream) in 100 km overall coverage and 54 MHz-746 MHz band. Basically, in MAC point of view, the WRAN system have been standardizing based on the IEEE 802.16 MAC layer features such as Data transmission method, QoS provision and Bandwidth request schemes. Additionally, the WRAN system further include CR nature functions such as incumbent user protection, self coexistence which would be importantly considered. Also, the inherent WRAN functions are added such as channel bonding and fractional bandwidth usage. This paper mainly explained frame structure, IU protection, self coexistence which are key functions of WRAN system. Finally, in this paper, we expressed a prospect of IEEE 802.22 WRAN standardization.

MPLS and Video Stream broadcast multicast transport optimization through convergence (MPLS와 멀티캐스트 융합을 통한 Video Stream 방송 전송 최적화)

  • Hwang, Seong-Kyu;Han, Seung-Jo
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.18 no.6
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    • pp.1330-1336
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    • 2014
  • QoS techniques and transmitted in real-time communication with the advancement of technology a variety of applications and services are available these days, mobile devices bogeuphwa LTE technology to the development of multimedia services with high quality can be realized. In order to satisfy this condition simply with a router with an increased bandwidth expansion by considering the increase in the routing table of the network scalability problems included. Burst traffic data to be distributed according to the environment is to be centered. To do this, the destination -based routing method to transmit the current paper -based (Source routing) routing settings are required. In this paper, published by the IETF, IP switching system based on standardized protocol Label Switching Multi-Protocol Label Switching (MPLS) network by using the existing Best Effect is difficult to guarantee QoS for multimedia transmission in MPLS network environment using optimized QoS guarantees to transmit the multicast.

Implementation of HL7 Interface Engine for Medical Information Exchange (의료정보 공유를 위한 HL7 인터페이스 엔진 구현)

  • Hwang, Deuk-Young
    • Journal of the Korea Society of Computer and Information
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    • v.15 no.8
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    • pp.89-98
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    • 2010
  • Recently supply of Internet is bringing a important change in medical environments. The hospitals which had a different system is required the system that can efficiently share and exchange medical information. In order to transmission medical information between systems, the Health Level Seven(HL7) interface engine development that can convert medical data to HL7 messages is necessary. The HL7 is a standard protocol for data exchange in healthcare environments. In this paper, I implemented HL7 interface engine for Alzheimer's disease in elderly care facility. The interface engine is composed of the client system and the server system. The client system inputs user's medical care data for the aged, and builds them into HL7 message stream. HL7 messages in the client system transmitted over TCP/IP protocol to the server system. The server system parses and validates this messages stream to the segments and fields and then transmits acknowledgement to the client system. I implemented it using the Java and JavaCC. The study of interface engine implementation can be used meaningfully in electronic health record, telemedicine system, and medical information sharing among various healthcare institutions.

Greedy Precedent Frame Transmission Technique in VOD System (VoD 시스템에서 탐욕적 선행 전송 기법)

  • Lee, Joa-Hyoung;Jung, In-Bum
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.14 no.3
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    • pp.603-612
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    • 2010
  • Recently, with the advance of computing and networking technique, the high speed internet becomes widespread, however, it is still hard job to do streaming the media which requires high network bandwidth over the internet. Previous VoD system researches for streaming over the internet mainly proposed techniques that controls the QoS(Quality of Service) of the media in proportion to the network status. Though, this could be the solution for the service provider while the service user who wants constant QoS may not satisfy with variable QoS. In the paper, we propose greedy precedent frame transmission technique, GPFT, for guarantee of constant QoS. In GPFT, Streaming VoD server prefetches precedent frames and transmits the frame greedily by increasing the frame transmission rate while the available network bandwidth is high. The GPFT uses the prefetched precedent frames to guarantee the QoS while the available network bandwidth is low. The experiment result shows that the proposed GPFT could guarantee the constant QoS by prefetching the frames adaptively to the network bandwidth with the characteristic of video stream.

Multiplexing of UHDTV Based on MPEG-2 TS (MPEG-2 TS 기반의 UHDTV 다중화)

  • Jang, Euy-Doc;Park, Dong-Il;Kim, Jae-Gon;Lee, Eung-Don;Cho, Suk-Hee;Choi, Jin-Soo
    • Journal of Broadcast Engineering
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    • v.15 no.2
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    • pp.205-216
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    • 2010
  • In this paper, a method of MPEG-2 Transport Stream (TS) multiplexing for Ultra HDTV (UHDTV) and its design and implementation as a SW tool is described. In practice, UHD video may be divided into several HD videos and each video is encoded in parallel. Therefore, it is necessary to synchronize and multiplex multiple bitstreams encoding each HD video for transmitting and storing UHD video. In this paper, it is assumed that 4 HD videos partitioning a UHD spatially are encoded as H.264/AVC and two 5.0 channel audios are encoded by AC-3. Therefore, 4 H.264/AVC elementary streams (ESs) and 2 AC-3 ESs is mainly considered in the TS multiplexing of UHD. For the carriage of H.264/AVC and AC-3 over MPEG-2 TS, PES packetization and TS multiplexing are designed and implemented based on the extended specification of the MPEG-2 Systems and ATSC (Digital audio compressed standard), respectively. The implemented UHD TS multiplexing tool emulates real time HW operation in the time unit corresponding to the duration of one TS packet transmission in a given TS rate. In particular, in order to satisfy the timing model, the buffers defined in the TS System Target Decoder (T-STD) are monitored and their statuses are considered in the scheduling of TS multiplexing. For UHD multiplexing, two kinds of multiplexing structures, which are UHD re-multiplexing and UHD program multiplexing, are implemented and their strength and weakness are investigated. The developed UHD TS multiplexing tool is tested and verified in terms of the syntax and semantics conformance and functionalities by using a commercial analyzer and real-time presentation tools.

Building Low Delay Application Layer Multicasting Trees for Streaming Services (스트리밍 서비스를 위한 적은 지연의 응용계층 멀티캐스트 트리 구축)

  • Kim, Jong-Gyung
    • The Journal of the Korea Contents Association
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    • v.8 no.10
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    • pp.20-26
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    • 2008
  • The quality of stream remaking is decided the load of a server and Jitter through the traffic of the transmission path between end to end. In order to improve these problems in this paper, I propose tree construction method of low-delay-level-multicast. In this case which the network congestion will be occurred by streaming quality, I also propose the technique which dynamically changes the transmission path. This technique first constructs the overlay structure for relaxing the overload of server. Secondly, in order to decrease Jitter of client, it makes upload bandwidth and low latency balanced. In the evaluation of the performance, this paper showed better enhancement of about 15 than P2CAST[4] in the simulation about node average join count, average bandwidth, service request refusal ratio, RTT measurement of nodes, and node average join count by defect ratio.