• Title/Summary/Keyword: Streaming System

Search Result 677, Processing Time 0.028 seconds

Efficient Transmission of Scalable Video Streams Using Dual-Channel Structure (듀얼 채널 구조를 이용한 Scalable 비디오(SVC)의 전송 성능 향상)

  • Yoo, Homin;Lee, Jaemyoun;Park, Juyoung;Han, Sanghwa;Kang, Kyungtae
    • KIPS Transactions on Computer and Communication Systems
    • /
    • v.2 no.9
    • /
    • pp.381-392
    • /
    • 2013
  • During the last decade, the multitude of advances attained in terminal computers, along with the introduction of mobile hand-held devices, and the deployment of high speed networks have led to a recent surge of interest in Quality of Service (QoS) for video applications. The main difficulty is that mobile devices experience disparate channel conditions, which results in different rates and patterns of packet loss. One way of making more efficient use of network resources in video services over wireless channels with heterogeneous characteristics to heterogeneous types of mobile device is to use a scalable video coding (SVC). An SVC divides a video stream into a base layer and a single or multiple enhancement layers. We have to ensure that the base layer of the video stream is successfully received and decoded by the subscribers, because it provides the basis for the subsequent decoding of the enhancement layer(s). At the same time, a system should be designed so that the enhancement layer(s) can be successfully decoded by as many users as possible, so that the average QoS is as high as possible. To accommodate these characteristics, we propose an efficient transmission scheme which incorporates SVC-aware dual-channel repetition to improve the perceived quality of services. We repeat the base-layer data over two channels, with different characteristics, to exploit transmission diversity. On the other hand, those channels are utilized to increase the data rate of enhancement layer data. This arrangement reduces service disruption under poor channel conditions by protecting the data that is more important to video decoding. Simulations show that our scheme safeguards the important packets and improves perceived video quality at a mobile device.

An Efficient Broadcasting Channel Assignment Scheme for Mobile VOD Services (모바일 VOD 서비스를 위한 브로드캐스팅 채널할당 기법)

  • Choi, Young
    • Journal of Korea Multimedia Society
    • /
    • v.11 no.5
    • /
    • pp.685-691
    • /
    • 2008
  • Recently with the rapid evolution of the mobile computing and communication technologies, mobile VOD service becomes increasingly important for wireless mobile users. The VOD service is being widely used in various areas of application, such as education, entertainment and business, because it provides users convenience in easily having access to video information at any time in any places. However, in reality, the mobile system has many difficulties in providing the smooth VOD service owing to frequent transfers and cutoffs of clients. The importance of a technique to transmit broadcasting is being stressed as a method for providing stabler mobile VOD service to a large number of clients. This paper is aimed at showing how to reduce demands for server bandwidth and delay of earlier service through performance analysis by suggesting an effective VOD broadcasting transmission technique through channel division in the mobile atmosphere. Many researches have been made about regular broadcasting techniques in particular. This study divides the methods used for assigning channels which have been decided by the size of segments into a group of regular channels and assistant channels using wireless gap-fillers to provide effective VOD services to a large number of clients at the mobile environment using small bandwidth resources. The regular channels transfer regular streams, while assistant channels repeatedly transfer the first segment to reduce early service delay time to receive regular streams. In this way, the study suggests a technique to reduce server bandwidth demand and early service delay time. Through the proposed technique, the server bandwidth demand could be reduced by more than 30 percent and the study continuously shows reduced early service delay time through conducting performance analysis.

  • PDF

FPGA Implementation of a Burst Cell Synchroniser for the ATM-PON Upstream (ATM-PON의 상향에서 버스트 셀 동기장치의 FPGA 구현)

  • Kim, Tae-Min;Chung, Hae;Shin, Gun-Soon;Kim, Jin-Hee;Sohn, Soo-Hyeon
    • Journal of the Institute of Electronics Engineers of Korea TC
    • /
    • v.38 no.12
    • /
    • pp.1-9
    • /
    • 2001
  • In the APON(ATM Passive Optical Network), the transmission of the upstream traffic is based on a TDMA(Time Division Multiple Access) method that an OLT(Optical Line Termination) permits ONUs(Optical Network Units) sending cells by allocating time slots. Because the upstream is not a streaming mode, the cell synchronizer has to be operated in the burst mode. Also, the cell phase monitor is required to prevent collisions between cells which are transmitted by multiple ONUs through a single optical fiber. In this paper, a TDMA burst cell synchroniser is implemented with the FPGA(Field Programmable Gate Array) being used in the APON based on G.983.1 for transmitting upstream cells. It has two main functions which are the upstream data recovery and the phase monitoring. The former is to recover the upstream data and clock in the OLT by seeking the preamble which is the overhead of the upstream time slot and by aligning the phase of the bit and cell with the system clock. The latter is to provide the information to the ONU to compensate for the equalization delay by monitoring continuously the phase difference between adjacent cells to avoid the cell collision on the upstream.

  • PDF

Self-Organizing Middleware Platform Based on Overlay Network for Real-Time Transmission of Mobile Patients Vital Signal Stream (이동 환자 생체신호의 실시간 전달을 위한 오버레이 네트워크 기반 자율군집형 미들웨어 플랫폼)

  • Kang, Ho-Young;Jeong, Seol-Young;Ahn, Cheol-Soo;Park, Yu-Jin;Kang, Soon-Ju
    • The Journal of Korean Institute of Communications and Information Sciences
    • /
    • v.38C no.7
    • /
    • pp.630-642
    • /
    • 2013
  • To transmit vital signal stream of mobile patients remotely, it requires mobility of patient and watcher, sensing function of patient's abnormal symptom and self-organizing service binding of related computing resources. In the existing relative researches, the vital signal stream is transmitted as a centralized approach which exposure the single point of failure itself and incur data traffic to central server although it is localized service. Self-organizing middleware platform based on heterogenous overlay network is a middleware platform which can transmit real-time data from sensor device(including vital signal measure devices) to Smartphone, TV, PC and external system through overlay network applied self-organizing mechanism. It can transmit and save vital signal stream from sensor device autonomically without arbitration of management server and several receiving devices can simultaneously receive and display through interaction of nodes in real-time.

WebRTC-Based Remote Collaborative Learning Platform (WebRTC 기반 원격 협업 학습 플랫폼 기술 연구)

  • Oh, Hyeontaek;Ahn, Sanghong;Yang, Jinhong;Choi, Jun Kyun
    • The Journal of Korean Institute of Communications and Information Sciences
    • /
    • v.40 no.5
    • /
    • pp.914-923
    • /
    • 2015
  • Recently, as the number of smart devices (such as smart TV or Web based IPTV) increases, the way of digital broadcast contents is changed. This change leads that conventional broadcast media accepts Web platform and its services to provide more quality contents. Based on this change, in education field, education broadcasting also follows the trend. The traditional education broadcasting platforms, which just delivered the lecture in one-way, are utilized the Web technology to make interaction between teacher and student. Current education platforms, however, are insufficient to satisfy users' demands for two-way interactions. This paper proposes a new remote collaborative learning platform which able to provide high interactivity among users. Based on new functional requirements from original use case, the platform provides collaborative contents sharing and collaborative video streaming techniques by utilizing WebRTC (Web Real-Time Communication) technology. The implementation demonstrates the operability of proposed system.

A Study on the Relationship between Camera and Subject for Visualization of Image - A Focus on the Status of Watch a Movie with Small Mobile Device - (영상의 시각화를 위한 카메라와 피사체의 상관관계 연구 - 스마트폰 사용자의 영상 시청 현황을 중심으로 -)

  • Ko, Hyun-Wook
    • Journal of Korea Entertainment Industry Association
    • /
    • v.13 no.5
    • /
    • pp.119-126
    • /
    • 2019
  • Watching movies is common on a big screen like a theater or on a big-screen TV. nowadays, small platform such as mobile devices is increasing rapidly for watch a movie. These changes are deeply related to the advent of Internet-based video streaming services such as OTT. OTT's development has provided in free video viewing system without using the set-top box is free from the limitations of time and space. Leading the market is Netflix[1], which started its business with Internet-based DVD rental service. Netflix, which is growing in tandem with the mobile market, had 193.26[2] million members as of the end of 2018. Other OTT participating companies include content-based Pooq, TVing, platform-based Olleh TV Mobile, Oksusu and LTE video portal. The size of such new growth projects has grown gradually, with 25.4 percent of all smartphone users currently watching video content with small mobile devices. Therefore, de-largeization, it is thought that visual language is needed for viewing small mobile devices that are capable of OTT services. To this end, this paper will identify the problem in viewing popular video content with small mobile devices and Survey and study its impact on viewers using the questionnaire.

An Embedding /Extracting Method of Audio Watermark Information for High Quality Stereo Music (고품질 스테레오 음악을 위한 오디오 워터마크 정보 삽입/추출 기술)

  • Bae, Kyungyul
    • Journal of Intelligence and Information Systems
    • /
    • v.24 no.2
    • /
    • pp.21-35
    • /
    • 2018
  • Since the introduction of MP3 players, CD recordings have gradually been vanishing, and the music consuming environment of music users is shifting to mobile devices. The introduction of smart devices has increased the utilization of music through music playback, mass storage, and search functions that are integrated into smartphones and tablets. At the time of initial MP3 player supply, the bitrate of the compressed music contents generally was 128 Kbps. However, as increasing of the demand for high quality music, sound quality of 384 Kbps appeared. Recently, music content of FLAC (Free License Audio Codec) format using lossless compression method is becoming popular. The download service of many music sites in Korea has classified by unlimited download with technical protection and limited download without technical protection. Digital Rights Management (DRM) technology is used as a technical protection measure for unlimited download, but it can only be used with authenticated devices that have DRM installed. Even if music purchased by the user, it cannot be used by other devices. On the contrary, in the case of music that is limited in quantity but not technically protected, there is no way to enforce anyone who distributes it, and in the case of high quality music such as FLAC, the loss is greater. In this paper, the author proposes an audio watermarking technology for copyright protection of high quality stereo music. Two kinds of information, "Copyright" and "Copy_free", are generated by using the turbo code. The two watermarks are composed of 9 bytes (72 bits). If turbo code is applied for error correction, the amount of information to be inserted as 222 bits increases. The 222-bit watermark was expanded to 1024 bits to be robust against additional errors and finally used as a watermark to insert into stereo music. Turbo code is a way to recover raw data if the damaged amount is less than 15% even if part of the code is damaged due to attack of watermarked content. It can be extended to 1024 bits or it can find 222 bits from some damaged contents by increasing the probability, the watermark itself has made it more resistant to attack. The proposed algorithm uses quantization in DCT so that watermark can be detected efficiently and SNR can be improved when stereo music is converted into mono. As a result, on average SNR exceeded 40dB, resulting in sound quality improvements of over 10dB over traditional quantization methods. This is a very significant result because it means relatively 10 times improvement in sound quality. In addition, the sample length required for extracting the watermark can be extracted sufficiently if the length is shorter than 1 second, and the watermark can be completely extracted from music samples of less than one second in all of the MP3 compression having a bit rate of 128 Kbps. The conventional quantization method can extract the watermark with a length of only 1/10 compared to the case where the sampling of the 10-second length largely fails to extract the watermark. In this study, since the length of the watermark embedded into music is 72 bits, it provides sufficient capacity to embed necessary information for music. It is enough bits to identify the music distributed all over the world. 272 can identify $4*10^{21}$, so it can be used as an identifier and it can be used for copyright protection of high quality music service. The proposed algorithm can be used not only for high quality audio but also for development of watermarking algorithm in multimedia such as UHD (Ultra High Definition) TV and high-resolution image. In addition, with the development of digital devices, users are demanding high quality music in the music industry, and artificial intelligence assistant is coming along with high quality music and streaming service. The results of this study can be used to protect the rights of copyright holders in these industries.