摄像技术多媒体网络课程的教学设计与开发

多媒体网络教学是利用网络技术与计算机技术,通过对信息进行收集、传输、处理等,为实现教学目标而服务.利用多媒体网络开展教学工作,其优势体现在不受时间与物理空间限制,载体的信息多样化,将文字、图像、动画、声音等集中于一体,调动学生的多个感官,学习的效率更高,效果也更好.本文就此论述了如何利用摄像技术多媒体网络课程的教学设计与开发提升教学质量.     

安防系统数字化应用探讨

数字化安防系统是采用现代传感技术、数字信息处理技术、数字通信技术、计算机技术、多媒体技术和网络技术来实现各种安防信息的采集、处理、传输、显示和高度集成的系统。它可以使安防各子系统之间以及安防系统与其他信息系统之间实现网络连接和信息的交换与共享，从而提高安全防范的质量与效率的一种科学性系统。[编者按]     

基于Java的网络多媒体制作的关键技术

本文阐述了Java在网络多媒体声像设计中的应用,详细论述了使用类和多线程技术、双缓冲区技术及异常处理技术实现播放声音、编辑图像、动画等。     

浅析数字视频监控系统

数字视频监控系统是将数字信息处理技术、数字通信技术、计算机技术、网络技术、多媒体技术与监控技术有机地结合起来,可以实现各种音视频信息的采集、传输、处理、显示、存储和回放等功能。本文主要分析了数字视频编码技术、磁盘阵列技术,以及介绍了数字视频的实际应用。     

基于全文检索和流媒体技术的素材库设计建设

网络教学需要多媒体素材库的支持。多媒体素材库必须有着强大的多媒体信息存储、管理功能。采用全文检索系统来实现多媒体素材库强大的信息检索功能。采用流媒体技术实现多媒体素材库中流媒体的管理与发布。设计建设一种采用全文检索技术和流媒体技术核心的多媒体素材库，检索精度与效率高，功能十分强大。     

论析IPTV技术与应用

本文主要介绍了IPTV的技术内涵、实现方式、内容提供、接入系统、终端系统、数字家庭多媒体网络,使人们对IPTV 以及数字家庭多媒体网络有新的认识与理解.     

Java在网络多媒体声像设计中的应用

本文阐述Java在网络多媒体声像设计中的应用,内容涉及动画实现、声音的播放、图形图像的编辑等基本技术,以及使用多线程双缓冲区、URL、AWT和异常处理等实现的关键技术。     

浅析虚拟现实关键技术在展示设计中的运用--以江苏软件园展厅设计为例

当今,随着多媒体互动投影系统的飞速发展,展厅设计中采用越来越多先进的计算机视觉技术和投影显示技术来营造现代展厅的的交互交互氛围,如何实现多媒体展项技术实现途径、关键技术、难点与风险处理措施是本文所阐述的重点。     

车载多媒体网络技术综述

对当今车载多媒体网络技术的发展背景、技术特点和未来趋势进行综述。分别介绍了MOST(Media Oriented System Transport)网络技术、IDB-1394网络技术和车载以太网络技术,对其发展历史、基本特点以及研究现状等进行概括、分析和比较,并列举三种车载多媒体网络技术的优点与缺点。最后对车载多媒体网络技术有待深入研究的热点和发展趋势进行展望。     

网络远程教学中流媒体技术的应用

流媒体(StreamingMedia)技术为教学课件提供了丰富的表现力,多媒体课件已经成为网络教学资源不可缺的内容。本文主要介绍了流媒体技术以及在多媒体课件中的应用,Authorware课件在网上发布与播放的具体方案。     

影响流媒体数据传输质量的因素与解决途径

随着Internet的迅猛发展,人们已经不满足于在网络上传输简单的文本图像信息,更加丰富的多媒体信息,特别是连续的媒体内容(视频和音频)已经开始在互联网上普及。通过网络传输连续媒体数据为人们呈现出一个极具吸引力的信息交流场景。目前在Internet上传输音、视频信息还有许多困难,其根本原因在于Internet的无连接每包转发机制主要是为突发性的数据传输而设计的,不适用于对连续媒体流的传输。为了在Internet上有效的、高质量平滑的传输音、视频信息流,还需要多种技术的支持。本文首先分析了流媒体传输的基本原理,讨论分析了影响网络音、视频平滑传输的因素及相关解决途径。     

基于网络的多媒体校园电视监控系统

本文针对多媒体数据在网络传输时遇到的问题,提出了把闭路电视网同校园网结合起来的解决方案,并介绍了基于网络的多媒体校园电视监控系统的组成、功能特点及系统实现。     

A Survey of Application Layer Techniques for Adaptive Streaming of Multimedia

Though the integrated services model and resource reservation protocol (RSVP) provide support for quality of service, in the current Internet only best-effort traffic is widely supported. New high-speed technologies such as ATM (asynchronous transfer mode), gigabit Ethernet, fast Ethernet, and frame relay, have spurred higher user expectations. These technologies are expected to support real-time applications such as video-on-demand, Internet telephony, distance education and video-broadcasting. Towards this end, networking methods such as service classes and quality of service models are being developed. Today's Internet is a heterogeneous networking environment. In such an environment, resources available to multimedia applications vary. To adapt to the changes in network conditions, both networking techniques and application layer techniques have been proposed. In this paper, we focus on the application level techniques, including methods based on compression algorithm features, layered encoding, rate shaping, adaptive error control, and bandwidth smoothing. We also discuss operating system methods to support adaptive multimedia. Throughout the paper, we discuss how feedback from lower networking layers can be used by these application-level adaptation schemes to deliver the highest quality content.     

基于因特网的连续媒体交互技术研究

对基于因特网的多媒体会议系统的传输特性、要求和协议标准等进行了系统地研究。通过对因特网上多媒体会议系统的特殊需求和支持多媒体网络协议存在问题的分析,提出了因特网多媒体会议系统的总体设计方案,并在此基础上对连续媒体中音频会议系统的实现进行了试验。     

基于IPv6的视频广播/点播系统

文章阐述了当前开展多媒体数据服务的迫切需要,详述了利用现有的两大主要传输网络———有线电视网络和基于IPv4协议的Internet———进行多媒体数据服务所存在的本质缺陷。在此基础上提出了在IPv6上实现多媒体数据服务的可能性,并提出了基于IPv6组播方式的视频广播/点播系统。     

Cost-based cache replacement and server selection for multimedia proxy across wireless Internet

Multimedia proxy plays an important role in multimedia streaming over wireless Internet. Since wireless network exhibits different characteristics from the Internet, multimedia proxy caching over wireless Internet faces additional challenges. In this paper, we present a study of cache replacement for a single server and server selection for multiple servers across wireless Internet. By considering multiple objectives of multimedia proxy, we design a unified cost metric to measure proxy performance in wireless Internet. Based on the defined unified cost metric, we propose a novel replacement algorithm for single-server and a new server-selection policy for multiple servers to improve the end-to-end performance such as throughput, media quality, and start-up latency. To effectively handle errors occurred on wireless link, channel-adaptive unequal error protection is deployed according to distinct quality of service requirements of layered or scalable media. Simulation results demonstrate that our approaches achieve significantly better performance than the known cache-replacement algorithms and sever selection schemes, respectively.     

An end-to-end robust approach for scalable video over the internet

With the growth of the Internet and abundant network resources, video streaming is becoming one of the increasingly important Internet applications. However, the current IP-based network provides only a single class best effort service. Video packet can be regarded as a packet loss by the video decoder either due to network congestion or due to exceeding the maximum delay threshold. It remains an open challenging task as to how to cope with the packet loss in the video streaming over the Int…     

Internet multimedia: answering basic questions

IP networks, including the public Internet and private corporate intranets, have emerged as the common way to carry data, voice, and video - otherwise known as multimedia. Some technologies and protocols that help multimedia work over IP networks are already well established. Others will become more familiar as Internet multimedia becomes more prevalent.     

Packet Spacing: An Enabling Mechanism for Delivering Multimedia Content in Computational Grids

Streaming multimedia with UDP has become increasingly popular over distributed systems like the Internet. Scientific applications that stream multimedia include remote computational steering of visualization data and video-on-demand teleconferencing over the Access Grid. However, UDP does not possess a self-regulating, congestion-control mechanism; and most best-effort traffic is served by congestion-controlled TCP. Consequently, UDP steals bandwidth from TCP such that TCP flows starve for network resources. With the volume of Internet traffic continuing to increase, the perpetuation of UDP-based streaming will cause the Internet to collapse as it did in the mid-1980's due to the use of non-congestion-controlled TCP.To address this problem, we introduce the counter-intuitive notion of inter-packet spacing with control feedback to enable UDP-based applications to perform well in the next-generation Internet and computational grids. When compared with traditional UDP-based streaming, we illustrate that our approach can reduce packet loss over 50% without adversely affecting delivered throughput.     

因特网多媒体会议系统中音频工具的设计与实现

因特网多媒本会议系统最近几年发展很快,受到了广泛的关注,根据因特网多媒体会议系统中音频应用的特点,分析了处理音频的关键技术,并在此基础上设计实现了网络音频工具NAT（Network Audio Tool)。简单地分析了实现NAT中的缓存管理总是和系统性能。