基于AVS的广播式手机电视系统

提出并实现了一种基于AVS的广播式手机电视系统,包括音视频数据的实时编码、系统层复用、T-DMB信号发射与接收、PC及手机平台的解码回放等,重点介绍了音视频编码器中的AVS视频编码算法的优化和复用器的实现,通过实验对编码器的性能进行了测试,对播放端的播放效果进行了分析。实验结果表明,AVS能够很好地应用于手机电视。     

流媒体在电视信号传输中的应用

随着互联网的快速发展和流媒体技术的日趋成熟,人们从网络上获取的信息已不仅仅局限于文字和图形,音频和视频等多媒体信息的传播越来越得到关注.电视信号在网络上的传输,尤其是电视信号的实时广播和点播成为了目前电视节目进行传送的主要方向.介绍了流媒体技术的特征、分类以及其在电视数据传输中的应用.     

Dimensioning the capacity of true video-on-demand servers

The need to reduce the huge bandwidth demand of video-on-demand (VoD) services has led to the conception of both multicast and broadcast based techniques for the deployment of such services on a large scale. Interactivity, a desirable feature for video services, includes the capacity to perform VCR operations. However, whenever a viewer requests the performance of a VCR operation, his/her video stream becomes unsynchronized with that of his/her multicast group, and a new channel must be allocated for the performance of this operation. The present paper introduces a novel approach for determining the number of video channels needed to support such interactivity. Moreover, it investigates the performance of interactive systems with a pool of channels reserved for the support of VCR operations. Systems with both batching and piggybacking are analyzed. Results indicate that for a medium to high number of users performing VCR operations the number of channels required to achieve target levels of quality-of-service is lower for systems with no pool of reserved channels than it is for systems with such a pool.     

基于流媒体Location Free电视的设计及实现

本文提出了一种基于流媒体Location Free电视的解决方案,既可以实现一般的电视收看功能,而且直接通过内置的视频网络传输模块连接在异地的电视机,收看在异地的电视频道。在解决方案中,视频采用了目前压缩率最高的H.264编解码算法,音频采用目前使用最广的G.711的编解码算法,并使用网络的流媒体传输协议,进行实时处理,从而实现异地电视信号实时传输通信。     

基于JSP的网络视音频播放系统设计与实现

根据流媒体传输原理,在局域网的基础上模拟基于Web的视频和音频播放系统,实现用户信息管理、听音频、看视频、文件的添加、删除、修改、上传及搜索功能等,从而设计出符合现在人们需求的视音频播放系统,为网络时代的人们提供方便、快捷的视音频点播节目,提供更加人性化设置。     

Modeling and evaluating IPTV applications in WiMAX networks

Internet Protocol-based Television (IPTV) is a digital television service which delivers television content via an IP network. The rapid growth of wireless network technology in recent years has changed, the way people access the Internet. Adding mobility to IPTV can create a truly compelling ubiquitous service which spans different network domains and varied IP-enabled terminals and devices, such as set-top boxes, PCs and cell phones. However, extending IPTV service to wireless networks requires overcoming bandwidth bottlenecks and high packet loss rates. Following the IEEE 802.16 standard, worldwide interoperability for microwave access (WiMAX) features high data rates and large service coverage, offering a wireless broadband solution for IPTV services. While previous research has focused on creating a broadband IPTV service few studies have practically evaluated IPTV applications in a wireless broadband network environment. In this paper, we model and evaluate a common constant bit rate (CBR)¹ based IPTV application and an IPTV live streaming (PPStreaming)² application while retrieving IPTV content via a WiMAX network. We also use the NS2 simulation tool to evaluate the performance of these two IPTV applications. The evaluation metrics include latency, packet loss, data rate and throughput statistics when the two IPTV applications are run in the WiMAX network. ¹The simplest IPTV solution is to convey video content by CBR. IPTV operators and content delivery networks relay CBR streaming content to control the demand for network capacity. Broadcasters prefer CBR video as it conserves bandwidth resources, but CBR delivery can degrade video quality result in higher overall demand on network capacity. ²PPStreaming (also referred to as P2P streaming Internet TV) is a network for live media streaming. In principle it’s similar to BitTorrent (BT) in that it provides stable and smooth broadcast of TV programs to broadband users. Unlike traditional streaming media, PPStreaming adopts P2P-streaming technology and supports full-scale visits with tens of thousands of simultaneous users. Its client software can be used in the browser or as a standalone executable.     

基于视频发送端速率控制的QoS技术研究

基于尽力而为的网络模式不能提供QoS保证,网络拥塞和分组丢失不可避免。在端到端视频单播结构下,论文提出了一个发送端速率控制框架SRCF,在此框架下首先利用RTCP报文中的字段提出了一种网络参数测量方法,然后设计了一个自适应速率算法SRCA,SRCA利用已得到的网络传输延迟和分组丢失率参数作为初始参数,来调整编码速率,达到充分利用带宽的目的,避免了视频质量由于调整参数带来的剧烈抖动。仿真结果表明,该算法在网络出现一定拥塞的条件下,能跟踪带宽的变化,网络和媒体QoS能保证视频质量较好。     

新媒体广播服务器的设计与实现

随着互联网技术的发展,传统的广播电台行业受网络媒体的影响越来越大。通过对广播行业的调查研究发现,传统广播正向新媒体广播开始转变。针对现有的流媒体服务器不具有多种音频源自动切换,控制功能薄弱,播放延迟大等问题,设计并实现了一个符合新媒体要求的新媒体广播服务器。介绍了流媒体音频服务器的应用及研究现状,阐述了广播行业对新媒体广播服务器的具体要求,描述了新媒体广播服务器的功能,详细分析了环形缓冲线程同步模型、音频源自动切换方法,远程控制方法。最后通过实验测试了服务器的功能和性能,验证了新媒体服务器已满足新媒体广播需求,达到了商用目的。     

浅谈FLV视频格式

视频文件可以分成两大类:一是影音文件,比如说常见的VCD;二是流式视频文件,或称流媒体视频文件。随着多媒体与网络技术的发展,流媒体视频文件已占据视频文件的主要地位,而在众多的流式视频文件格式中,FLV已成为当今网络上最流行的一种流媒体视频格式,它的使用给互联网视频传播带来了极大便利。     

基于MPC的音视频同步码率自适应测试

随着信息化的发展,音视频流媒体技术应用面越来越广,为了使得音视频流媒体技术尤其是在直播方面拥有更好的性能,得到更多用户的好评,采用在原本HTTP的动态自适应流标准的视频流媒体架构下引入MPC控制算法并将MPC模型预测控制与码率自适应算法相结合的方法,进行对AAC优化、确定预测模型、测试音视频同步的影响因素以及PSNR-Y分量、测试切片时长与跳帧时延,计算最终的QoE用户评价指标来进一步检测音视频流媒体技术的优劣。经实验仿真测试可知,相比前人的相关算法,在不同直播场景下以及不同网络环境下均有更加良好的QoE值,平均QoE用户评价指标明显更高,为1237.2826。综上分析可知,MPC的音视频同步码率自适应算法各项性能最好。     

An alternative paradigm for scalable on-demand applications:evaluating and deploying the Interactive Multimedia Jukebox

Straightforward, one-way delivery of audio/video through television sets has existed for many decades. In the 1980s, new services like pay-per-view and video-on-demand were touted as the “killer applications” for interactive TV. However, the hype quickly died away, leaving only hard technical problems and costly systems. As an alternative, we propose a new jukebox paradigm offering flexibility in how programs are requested and scheduled for playout. The jukebox scheduling paradigm offers flexibility ranging from complete viewer control (true video-on-demand), to complete service provider control (traditional broadcast TV). We first describe our proposed jukebox paradigm and relate it to other on-demand paradigms. We also describe several critical research issues, including the one-to-many delivery of content, program scheduling policies, server location, and the provision of advanced services like VCR-style interactivity and advanced reservations. In addition, we present our implementation of a jukebox-based service called the Interactive Multimedia Jukebox (IMJ). The IMJ provides scheduling via the World Wide Web (WWW) and content delivery via the Multicast Backbone (MBone). For the IMJ, we present usage statistics collected during the past couple of years. Furthermore, using this data and a simulation environment, we show that jukebox systems have the potential to scale to very large numbers of viewers     

FESORIA: An integrated system for analysis,management and smart presentation of audio/video streaming services

Due to the elevated consumption of resources, the high cost of the production of contents and the quality of service required in audio/video streaming services, it is extremely important to optimize all the elements involved in the deployment of these services. With this goal in mind, provider companies have developed their management and presentation tools. At the same time, some specific tools for audio/video streaming analysis have appeared. Data are collected from servers and proxies by analyzing their log files in order to generate different types of reports. In spite of their utility, there is a disconnection between these types of tools. In this way, several important relationships between collected data are lost and the influence of other important aspects such as the behaviour of the users and their relationship with the subject or the length of the contents is not considered. This generates inaccurate analyses and the impossibility to improve the presentation, for example by generating recommendations using the information gathered from the analysis tool. Fesoria is a system which combines both characteristics. It is an analysis tool and, at the same time, a system to manage the whole audio/video service. Fesoria is able to process the logs gathered from the streaming servers and proxies, and combine the extracted information with other types of data, such as content metadata, content distribution networks architecture, user preferences, etc. All this information is analyzed in order to generate reports on service performance, access evolution and users’ preferences, and thus to improve the presentation of the services. The system has been used in real audio/video services since 2001 with satisfactory results.

P-chaining: a practical VoD service scheme autonomically handling interactive operations

Providing scalable VoD streaming services has recently become a hot issue, and many approaches have been proposed. Because video streaming services through the Internet are widely used, the need to support VCR operations also increases. However, there are few approaches to supporting VCR operations on the Internet. We propose a service scheme based on chaining, in which clients as well as the server provide streaming services. In the proposed scheme, services are provided by unicast and managed locally using node lists. In addition, our scheme can support frequent VCR operations without incurring significant overhead in the server workload. We have evaluated our scheme through simulation with real traces from a content distribution network (CDN) company and with various parameters. The results show that the proposed scheme reduces server workload significantly. The results also verify that frequent VCR operations can be served smoothly without causing too much overhead.

浅论广电机房设备串口服务器设计

基于嵌入式WEB的串口服务器可以将广电机房串口设备方便地接入互联网实现资源共享,该文结合作者的项目开发经历,探讨了如何利用嵌入式WEB技术实现音视频切换器等广电机房设备串口服务器的设计。     

Seamless handover between unicast and multicast multimedia streams

With the deployment of heterogeneous networks, mobile users are expecting ubiquitous connectivity when using applications. For bandwidth-intensive applications such as Interact Protocol Television （IPTV）, multimedia contents are typical- ly transmitted using a multicast delivery method due to its bandwidth efficiency. However, not all networks support multicasting. Multicasting alone could lead to service disruption when the users move from a multicast-capable network to a non-multicast network. In this paper, we propose a handover scheme called application layer seamless switching （ALSS） to provide smooth real-time multimedia delivery across unicast and multicast networks. ALSS adopts a soft handover to achieve seamless playback during the handover period. A real-time streaming testbed is implemented to investigate the overall handover performance, espe- cially the overlapping period where both network interfaces are receiving audio and video packets. Both the quality of service （QoS） and objective-mapped quality of experience （QoE） metrics are measured. Experimental results show that the overlapping period takes a minimum of 56 and 4 ms for multicast-to-unicast （M2U） and unicast-to-multicast （U2M） handover, respectively. The measured peak signal-to-noise ratio （PSNR） confirms that the frame-by-frame quality of the streamed video during the handover is at least 33 dB, which is categorized as good based on ITU-T recommendations. The estimated mean opinion score （MOS） in terms of video playback smoothness is also at a satisfactory level.     

一种基于P2P的视频点播系统设计

在计算机发展领域中,目前有一项新兴的技术———视频点播技术(VOD),它是一种根据观众的要求播放节目的系统,随时提供交互式视频服务,主要用于实现多用户对网络多媒体文件的共享播放。文中将P2P与Internet上重要的媒体传输技术———流媒体技术相结合提出了一个基于P2P的视频点播系统的具体设计与实现方案,讨论了系统设计和实现过程中采用的相关技术与方法,采用点组技术,重点提出了一种新的发现机制。     

Low bandwidth desktop and video streaming for collaborative tiled display environments

High-resolution display environments built on networked, multi-tile displays have emerged as an enabling tool for collaborative, distributed visualization work. They provide a means to present, compare, and correlate data in a broad range of formats and coming from a multitude of different sources. Visualization of these distributed data resources may be achieved from a variety of clustered processing and display resources for local rendering and may be streamed on demand and in real-time from remotely rendered content. The latter is particularly important when multiple users want to concurrently share content from their personal devices to further augment the shared workspace. This paper presents a high-quality video streaming technique allowing remotely generated content to be acquired and streamed to multi-tile display environments from a range of sources and over a heterogeneous wide area network.The presented technique uses video compression to reduce the entropy and therefore required bandwidth of the video stream. Compressed video delivery poses a series of challenges for display on tiled video walls which are addressed in this paper. These include delivery to the display wall from a variety of devices and localities with synchronized playback, seamless mobility as users move and resize the video streams across the tiled display wall, and low latency video encoding, decoding, and display necessary for interactive applications. The presented technique is able to deliver 1080p resolution, multimedia rich content with bandwidth requirements below 10 Mbps and low enough latency for constant interactivity. A case study is provided, comparing uncompressed and compressed streaming techniques, with performance evaluations for bandwidth use, total latency, maximum frame rate, and visual quality.     

基于Linux的网络直播平台设计与实现

不同带宽用户对流媒体直播服务器的音频/视频质量有不同需求,为此,设计并实现一个基于Linux平台的网络直播平台。该平台采用MPEG4IP编码库的方法实现MPEG4软编码,利用实时传输协议传输数据的方法实现客户端直播请求,通过调用简单媒体接口库的方法实现视频在Linux平台的显示播放。实验结果证明,该网络直播平台传输效果较好,并且其稳定性和负载均衡性均达到设计要求。     

Extracting viewer interests for automated bookmarking in video-on-demand services

Video-on-demand (VoD) services have become popular on the Internet in recent years. In VoD, it is challenging to support the VCR functionality, especially the jumps, while maintaining a smooth streaming quality. Previous studies propose to solve this problem by predicting the jump target locations and prefetching the contents. However, through our analysis on traces from a real-world VoD service, we find that it would be fundamentally difficult to improve a viewer’s VCR experience by simply predicting his future jumps, while ignoring the intentions behind these jumps. Instead of the prediction-based approach, in this paper, we seek to support the VCR functionality by bookmarking the videos. There are two key techniques in our proposed methodology. First, we infer and differentiate viewers’ intentions in VCR jumps by decomposing the interseek times, using an expectation-maximization (EM) algorithm, and combine the decomposed inter-seek times with the VCR jumps to compute a numerical interest score for each video segment. Second, based on the interest scores, we propose an automated video bookmarking algorithm. The algorithm employs the time-series change detection techniques of CUSUMandMB-GT, and bookmarks videos by detecting the abrupt changes on their interest score sequences. We evaluate our proposed techniques using real-world VoD traces from dozens of videos. Experimental results suggest that with our methods, viewers’ interests within a video can be precisely extracted, and we can position bookmarks on the video’s highlight events accurately. Our proposed video bookmarking methodology does not require any knowledge on video type, contents, and semantics, and can be applied on various types of videos.     

Real‐time multicast tree visualization and monitoring

The exponential growth of the Internet combined with the increasing popularity of streaming audio and video are pushing Internet bandwidth constraints to their limits. Methods of managing and more efficiently utilizing existing bandwidth are becoming increasingly vital. Using IP multicast to deliver content, especially streaming audio and video, can provide enormous bandwidth savings. A decade of effort at deploying multicast, combined with the rising need for better traffic management for bandwidth‐hungry applications has led to significant momentum for multicast use and deployment. One of the remaining barriers to widespread adoption is the lack of multicast monitoring and debugging tools. To address this need we introduce MHealth, a graphical, near real‐time multicast monitoring tool. MHealth utilizes existing tools to collect comprehensive data about Realtime Transport Protocol (RTP) based streaming audio/video sessions. By using a combination of application‐level protocol data for participant information and a multicast route tracing tool for topology information, MHealth is able to present a multicast tree's topology and information about the quality of received data. In this paper we describe the design and implementation of MHealth and include an example analysis of multicast tree statistics. Copyright © 2000 John Wiley & Sons, Ltd.