工业以太网实时调度技术的设计与实现

鉴于以太网在工业现场设备控制层的应用前景,有意将以太网引入武器装备的底层控制环境.但它必须满足确定性实时调度的要求.基于以太网的TCP/IP通信机制和IEEE1588时钟同步协议,在数据链路层之上引入实时调度机制以消除现有的CSMA/CD机制在时间行为上的不确定性.相关的分析和测试表明,新设计的通信调度机制能够使得网络...     

OPC UA-TSN传输时延实验方法

OPC UA是一种能提供高带宽、高互操作性和高实时性的工业通信协议,它定义了一套通用的数据描述和语法表达方法(信息模型),每个系统都可以采用OPC UA的信息模型,从而实现系统间数据交互和互操作。TSN具备时间同步、数据调度、负载整形等多种优化机制,可保证数据传输的实时性。将OPC UA与时间敏感网络(TSN)相结合,可提供一个实时、高确定性通信网络,将对网络带宽、安全性、互操作、延迟和同步等方面得到较大改善。与现有的OPC UA over TSN测试仅关心终端间往返时延问题不同,提出了一种包含电机同步的OPC UA over TSN测试框架,引入实际TSN交换设备以及实际应用。本测试架构聚焦TSN标准(IEEE 802.1AS和IEEE 802.1Qbv)以及OPC UA标准IEC 62541-14《OPC UA统一架构14部分：Pub/Sub发布订阅》在IT/OT融合中的端到端传输时延测试问题,在不同调度机制以及不同负载条件下,测试了交换机连接的不同分布式控制计算机间链路层及应用层端到端传输时延。实验结果表明,时间感知整形器(TAS)能保证时间敏感流不受链路中竞争流量的影响,且即使...     

IP网中多媒体表现质量控制的算法研究

提出在目前IP网络设备及技术基础上,一种在通信子网外优化多媒体表现质量的源端最小延迟表现算法。给出并证明了分布式多媒体应用同步表现最优化定理,进而提出最小聚类延迟表现算法作为源端最小延迟表现控制算法的改进。与古典列表调度算法比较得出,上述两种算法能实现优化的分布式多媒体同步表现,同时在目的端缓冲区需求、溢出可能性等方面具有更优的性能。上述算法与底层通信机制无关,未来具有广阔的发展及应用前景。     

多媒体流间同步参数计算

1 引言在分布式多媒体同步控制中,基于有界缓冲区(Boundary Buffer)的同步方式是其中一种主要的强制同步方式。为同步源设置有界缓冲区,将有助于减轻或消除因网络抖动或其他影响所带来的不连续现象,这是保证流内各个媒体单元的实时和连续的重要手段。对于流间同步,设置有界缓冲区的方式也是十分必要的,如快进或重播某些媒体单元等同步操作,都需要缓冲区的支持。     

Linux核心调度机制的实时改造及应用研究

针对网络多媒体系统的实时应用需求，本文对Linux2．4核心的调度机制进行了改造。通过将弱硬实时调度算法、挪用调度算法、反馈调度算法继承到Linux核心调度机制中，提高了核心的实时性能；最后通过实际应用检验了改造后的Linux系统在处理网络多媒体数据时的实时性能。     

实时多媒体流同步机制的研究

实时多媒体流同步是网络环境下流媒体应用的基本要求和主要难点之一。文章介绍了多媒体同步的基本概念，分析了在IP网络环境下实时多媒体流在各个环节失去同步的原因，并针对这些原因提出了相应的解决方法。     

双层多媒体动态同步模型构建方法研究

多媒体同步是分布式多媒体应用的关键技术,解决流内、流间及用户交互同步的矛盾,细化同步粒度,提高同步精度,是学术界一直努力探索但至今尚未解决的问题。提出的双层多媒体动态同步模型构建方法,基于“自然同步点”和“准同步点”两类不同同步点,分两个层次,采用Petri网和Agent两种不同方法,在两种不同的尺度下完成多媒体同步。模型较好地协调了流内同步和流间同步的关系,简化了用户实时交互后的再同步,增强了多媒体的动态协同表现能力。     

远程教学直播系统中媒体同步的研究与实现

对于远程教学直播系统这样包含不同媒体流的分布式多媒体应用而言，媒体同步是一项重要内容。利用RTP（实时传输协议）传输机制中的时间戳和序列号信息，提出同步控制算法，实现了流内同步和流间同步。考虑到分组网络带来的延时抖动，算法可以动态地适应网络延时变化，从而保证了分布式环境中媒体同步的服务质量。     

工业以太网实时通信中的调度表构建策略研究

介绍了一种工业以太网实时通信机制．在传输层和数据链路层增加了实时通信实体，在不改变以太网通信机制的前提下，通过调度信息的发送解决了以太网的通信冲突问题．工业以太网的实时通信调度需要与功能块调度协调一致；在对功能块调度机制进行分析的基础上，将功能块调度与实时通信调度相结合，提出了工业以太网实时通信中周期性时间调度表和非周期通信调度表的构建方法．     

流媒体传输中的QoS研究及其实现

文中提出的对视频会议的服务质量控制策略是为解决以下问题:视频会议对涉及到的实时数据流传输对网络带宽、时延和丢包率有较高要求,但是,目前得到广泛应用的IP网络是一种尽力而为的网络,它并没有对实时多媒体提供任何服务质量保证。该策略从两个方面对服务质量加以控制:在发送端控制发送流量;在数据再现端通过缓冲机制控制媒体同步。     

Multimedia Multicast in Mobile Computing: Handoff Management

With rapid technological advances being made in the area of wireless communications it is expected that, in the near future, mobile users will be able to access a wide variety of services such as multicast distribution of multimedia streams. These services are characterized by the integrated processing of different media types, such as audio and video. Different multimedia streams must be played out in a synchronized way. However, due to the asynchronous nature of the communication (packets or radio), some problems can disrupt the synchronization such as delay jitter. In this paper, we present a synchronization protocol for multicast of multimedia streams. It allows a mobile host to playback continuously a multimedia stream. When a mobile host moves from cell to cell, the scheme allows continuous playback. The mechanism used is based on the pre-buffering of media units. We present a computation of the handoff time and a mechanism for the restoration of the initial buffer size. Simulation results show that, when resources are already reserved, the handoff time is bounded and the buffer takes again its initial size.     

Synchronization and handoff management schemes for wireless multimedia systems

This paper presents handoff management schemes for synchronization algorithms for wireless multimedia systems. The synchronization and handoff management schemes allow mobile hosts to receive time-dependant multimedia streams without delivery interruption while moving from one cell to another. They also maintain the correct ordering of the media components, through the execution of the wireless multimedia application by a means of timestamped messages passed among mobile hosts, base stations and servers. The timestamp values are used to compute the delay for each multimedia unit for each server. Furthermore, the proposed schemes always search for a quasi-receiver among the base stations with which the mobile hosts can communicate to synchronize multimedia units. We discuss the algorithms and present a set of simulation experiments that evaluate the performance of our schemes, using message complexity and buffer usage at each frame arrival time. Our results indicate that our schemes exhibit no underflow or overflow within the bounded delivery time.     

Adaptive multimedia synchronization in a teleconference system

In this paper, we present an adaptive buffering scheme for implementing intra-stream and inter-stream synchronization in real-time multimedia applications. The essence of the proposed scheme is to dynamically enforce equalized delays to incoming media streams, in order to piece-wise smooth the network delay variations and to synchronize the streams at the sink. An adaptive control mechanism based on an event-counting algorithm is employed to calibrate the PlayOut Clocks (POCs), which manages the presentations of multimedia data. The algorithm does not rely on globally synchronized clock and makes minimal assumption on underlying network delay distribution. Also, the user defined quality of service (QoS) specifications can be directly incorporated into the design parameters of the synchronization algorithm. The proposed synchronization scheme has been experimentally implemented in a teleconference system which consists of separately controllable audio, video, and data channels. The modular structure of the synchronization control provides the flexibility to maintain an arbitrary synchronization group in conjunction with a distributed conference management scheme. This paper also shows the experimental results of the test implementation and the suitability of the proposed scheme with respect to the multimedia traffic across an FDDI/Ethernet network.     

Multimedia streaming gateway with jitter detection

This paper investigates a novel active buffer management scheme, "Jitter Detection" (JD) for gateway-based congestion control to stream multimedia traffics in packet-switched networks. The quality of multimedia presentation can be greatly degraded due to network delay variation or jitter when transported over a packet-switched network. Jitter degrades the timing relationship among packets in a single media stream and between packets from different media streams and, hence, creates multimedia synchronization problems. Moreover, too much jitter will also degrade the performance of the streaming buffer in the client. Packets received by the client will be rendered useless if they have accumulated enough jitter. The proposed active buffer management scheme will improve the quality of service in multimedia networking by detecting and discarding packets that accumulated enough jitter, such as to maintain a high bandwidth for packets within the multimedia stream's jitter tolerance. Simulation results have shown that the proposed scheme can effectively lower the average received packet jitter and increase the goodput of the received packets when compared to random early detection (RED) and DropTail used in gateway-based congestion control. Furthermore, simulation results have also revealed that the proposed scheme can maintain the same TCP friendliness when compared to that of RED and DropTail used for multimedia streams.     

多媒体数据库系统中的同步表示管理

支持多媒体应用的扩展数据库系统必须具有新的机制以确保多种媒体数据流的同步表示。为了使用户同时具有多媒体数据流表示的灵活性和高效率,必须将媒体数据流分割成媒体对象,并且定义和保持其间的时间限制。为此要在现有数据库系统上采用能高效管理多媒体数据时间传性的新管理工具。文中讨论了多媒体数据库系统中同步表示管理。当考虑延迟的影响时,提出了多种媒体数据流同步表示的原则。基于这些原则,可以设计出各种同步机制。     

弱同步及其在WWW上的应用

在当今世界,WWW已经成为新颖而强大的媒体,但是,现在运行于Web上的应用仍然不是完全多媒体化的.这主要是因为Internet网络是不可靠、尽力型的,无法提供数据实时传输的保证,尤其对那些大数据量的连续媒体(如视频与音频数据).因此,在Web环境中多媒体对象间的同步就变得格外复杂.该文提出了一种新颖的同步模型：弱同步.它使多媒体文档即使在多变的环境下也能获得最佳效果.在弱同步中,不仅可以定义对象间确定的时间关系,也可以定义非确定的时间关系.此外,媒体间的同步关系,如严格同步和松散同步,也可以在文档的同步描述中定义.     

Delay compensation protocols for synchronization of multimedia datastreams

Temporal synchronization of various data streams in multimedia information supporting voice, video, graphics and text, that are exchanged between users over a high speed network are discussed. During delivery of such data, maintaining the required association between data units across various streams in real-time is necessary to sustain quality of service in the presence of data loss and/or delay in the network. Solving this synchronization problem requires framing of data streams whereby various points in the data streams deliverable simultaneously to a user are identified. A solution in which the temporal axis of an application is segmented into intervals in which each interval is a unit of synchronization and holds a data frame is presented. Simultaneous data delivery involves delivering all data segments belonging to an interval within a certain real-time delay, as specifiable by the application. Based on the approach, end-to-end transport protocols that compensate for the data skew that may arise due to data loss/delay is described. Simulation experiments confirm the viability of the protocols. Major uses of the protocols are in broadband ISDNs and metropolitan area networks (MANs)     

扩展时间流Petri网多媒体同步模型

多媒体同步的一个主要问题是多媒体同步时间要求的描述与建模。文中基于时间流Ｐｅｔｒｉ网模型（ＴＳＰＮ）,通过分析ＴＳＰＮ模型的不足之处,提出了一种新的多媒体同步模型－－扩展时间流Ｐｅｔｒｉ网模型（ＳＴＳＰＮ）,ＳＴＳＰＮ不但可以完全、准确地描述媒体流内和流间的时间行为,而且通过引入容许阻塞时间、问题算法等对ＴＳＰＮ进行扩展,可以有铲地克服网络随机延迟、阻塞和丢包等因素的不良影响,提高了模型的建模能力     

基于节的分布式多媒体资源管理

分布式多媒体系统面临的最大挑战是支持稳定的连续媒体流和媒体流间同步，分布式多媒体资源管理模型是保证它们的关键，本文提出了一种基于节的资源分配和管理模型，在资源分配时，它通过建立媒体流的线程，Ｉ／Ｏ处理，缓冲区和网络处理的平衡方程，计算保证稳定的连续媒体所必需的资源，在资源管理时，它通过调整媒体流的速率达到媒体流间同步，本文还提出了一种实时的节资源请求方案，它能系统资源开销最小，最后给出了在交换式１