Linux下通用线程池的改进与实现

对线程池的阻塞唤醒机制,动态调整,线程安全退出,参数处理,系统线程数限制等细节进行研究,保证了其在不同应用场景下的独立性和通用性;同时采用一种基于数组的链表机制来改进线程池的查找分配算法,将其时间复杂度稳定在O（1）,避免了传统线程池当线程数目过大时导致的查询分配性能下降的问题。实验结果表明,改进后的线程池与传统的系统线程分配方式相比在开销上有很大节省。     

KNX/EIB通信协议的实时性分析与改进

KNX/EIB通信协议在其链路层使用CSMA/CA机制解决通信冲突问题,虽然提高了KNX/EIB网络防冲突能力,却造成同等级命令帧数据传输时延抖动非常大,传输实时性大打折扣。提出对KNX/EIB实时性的改进方法KNX/EIB-A,在不改变原有的通信协议栈的基础上,将一个调度程序应用于KNX/EIB通信协议的应用层与用户应用程序之间;将原有的分布式平等结构划分为分级主从结构,对数据命令帧的发送进行调度。最后通过对KNX/EIB-A进行分析及原型实现,证明了该方法通过有效地减轻传输时延的抖动从而改进了协议的实时性。     

利用Winsock编程捕获局域网上所有IP包

在高速网络中捕获所有IP包面临如何快速处理数据包和减少丢包的问题。提出通过建立链表来实现对捕获数据包的批量处理，以减少系统创建线程的开销。同时通过创建线程可以使数据包的分析处理和数据包的捕获并行进行，可减少在处理包数据时可能出现的丢包。在捕获过程中增加了对丢包的统计，可以给出丢包车。同时在创建线程不成功时，通过一个较少延时的重复尝试以尽量减少丢包数。这种方法在100Mbps的局域网中，丢包率可控制在5％之内。     

WRR算法在多类别实时数据流调度中的优化

随着融合型网络的发展,服务质量(包括可用带宽、端到端的时延、抖动和丢包率)对一些实时数据流应用(语音流、视频流等)越来越重要。由于传统的WRR算法只能满足各个应用队列的公平性要求,而不能保证多类别实时数据的低时延和低抖动性要求,所以本文在WRR算法的基础上提出了BSTL-RR调度算法,此算法运用了二层循环和借用时隙的两个思想。BSTLRR调度算法不仅在调度低时延和低抖动的多类别实时数据流帧方面要优于WRR调度算法,而且在一定程度上也保证了各优先级队列调度上的公平性。     

Windows7SP1下DKOM防攻击技术研究

Windows在内存中存储了一些记账信息,用于管理进程、线程、设备驱动程序等对象,并报告给用户,向用户反映系统的运行状况.由于这些信息位于内存中,因此可以直接对其进行修改.文章以微软公司最新的操作系统Windows7 SP1为平台,揭示了Windows7 SP1为进程、线程、驱动等对象建立的一系列结构体和链表,并提出了几种方法,通过修改这些结构体和链表,达到保护、隐藏进程和驱动的目的.     

利用Winsock编程实现局域网上所有IP包的捕获和分析

本文主要讲述了利用Winsock编程方法，通过建立链表和线程，实现对局域网上的所有IP包进行高效的捕获和分析。     

基于抖动测试的AFDX端到端延迟紧性修正

AFDX引入虚拟链路(Virtual Link)实现物理带宽资源的逻辑分隔。由于数据帧的异步到达和多路复用输出造成虚拟链路的时延抖动现象,并最终导致流量端到端延迟分析的不确定性。本文提出了一种基于抖动测试值的网络演算紧缩方法。通过分布式测试,获得虚拟链路在网络中的实际传输抖动,并以此为基础,建立了流量传输精确化模型,通过流量模型的逐级修正,使端到端延迟计算结果逐级精确化。通过将抖动实际值与理论分析结果相结合,提供了网络演算悲观度及其扩散影响度量的直观对比,提高了延迟计算的紧性。     

一种智能开关柜监控系统高效数据传输控制方法

智能开关柜监控系统的数据传输包括综合控制模块与工业触摸屏之间的RS232全双工串行通信,与其他测量模块的RS485半双工串行通信。综合控制模块需要实现数据接收、处理和发送功能。介绍一种在综合控制模块的控制中运用数据缓存技术和多线程处理技术来提高系统的数据传输效率的方法。数据缓存技术主要包括存储算法和提取帧算法。多线程处理包括RS232数据的接收线程、处理线程、发送线程和RS485的数据收发线程和处理线程等。     

自相似网络的时延抖动性能仿真分析

自相似性对网络性能产生了影响是当前的研究热点。建立了一种基于FBM的自相似网络排队时延抖动分析模型,重点讨论了自相似流量作为输入时对排队系统的时延抖动的影响。对理论分形流量和实际测量流量进行了仿真实验,验证了结果的正确性和有效性。实验结果表明：自相似流量长相关强弱的程度对排队系统时延抖动特性具有非常不同的影响,尤其是在缓存较大的情况下。同时,还发现网络流量中长相关发生作用时状态转变与排队系统本身的参数也有关,这是新的发现,对实时业务的网络性能评价具有重要的参考意义。     

基于Transpose规则的无锁自组织链表算法

自组织链表可以依据访问序列动态调整链表结构,提高链表性能。在分析并研究现有自组织链表算法的基础上,结合Transpose规则,提出无锁自组织链表算法。线程可标记被访问的结点并尝试与标记结点前驱相交换,也可直接物理删除已被标记的结点,同时其他线程发现该标记结点时会辅助该线程完成相应操作,从而保证链表的非阻塞特性。实验结果表明,该算法性能与Harris-Michael链表算法相当,并且其无锁实现方式比粗粒度锁算法更具优势。     

AVP: A Highly Efficient Transport Protocol for Low Bit Rate Multimedia Communications

Utilization of Internet communications in distance learning, distributed simulation, and distributed work groups involves multimedia transmission of animation, voice and video clips. Highly compressed audio-video data protocols are required for efficient Internet multimedia communications. Addressing this requirement, a new transport protocol called Audio-Video Protocol (AVP) for highly efficient multimedia communications on the Internet is presented. While providing similar real-time delivery functions as Real-Time Transport Protocol (RTP) and Real-Time Control Protocol (RTCP), AVP adopts a novel audio-based synchronization scheme. This synchronization scheme has two advantages. One is the overhead reduction through eliminating the timestamp in each transmitted data packet. The other is the packet rate reduction by putting multiple audio frames or mixed audio-video frames in a single AVP packet. As a result, the end-to-end media unit delay is reduced while achieving implicit synchronization. Furthermore, AVP provides adaptive quality of service (QoS) by the prioritized packetization scheme. Simulation results are presented to verify the advantages of the AVP protocol.     

适应性查询处理——Internet数据查询面临的挑战与解决技术

随着基于Internet的数据查询系统的发展与普及,适应性查询处理逐渐成为一项重要的技术。目前的Internet可以看作一个庞大的分布式和异构化数据库,各个数据源具有自治性,加上广域网网络传输带宽的限制,各个数据源数据的可访问性以及传输速度是经常变化和不可预测的。传统的采用“停止-进行”方式的查询处理不能很好地处理这种情况。而能够在查询执行过程中动态调整查询计划的适应性查询处理是针对此类应用的最佳选择。文章论述适应性查询处理涉及的研究课题及解决技术,并例举最新的研究成果。     

Generic forward error correction of short frames for IP streaming applications

If the frame size of a multimedia encoder is small, Internet Protocol (IP) streaming applications need to pack many encoded media frames in each Real-time Transport Protocol (RTP) packet to avoid unnecessary header overhead. The generic forward error correction (FEC) mechanisms proposed in the literature for RTP transmission do not perform optimally in terms of stability when the RTP payload consists of several individual data elements of equal priority. In this paper, we present a novel approach for generating FEC packets optimized for applications packing multiple individually decodable media frames in each RTP payload. In the proposed method, a set of frames and its corresponding FEC data are spread among multiple packets so that the experienced frame loss rate does not vary greatly under different packet loss patterns. We verify the performance improvement gained against traditional generic FEC by analyzing and comparing the variance of the residual frame loss rate in the proposed packetization scheme and in the baseline generic FEC.     

Neural and fuzzy computation techniques for playout delay adaptation in VoIP networks

Playout delay adaptation algorithms are often used in real time voice communication over packet-switched networks to counteract the effects of network jitter at the receiver. Whilst the conventional algorithms developed for silence-suppressed speech transmission focused on preserving the relative temporal structure of speech frames/packets within a talkspurt (intertalkspurt adaptation), more recently developed algorithms strive to achieve better quality by allowing for playout delay adaptation within a talkspurt (intratalkspurt adaptation). The adaptation algorithms, both intertalkspurt and intratalkspurt based, rely on short term estimations of the characteristics of network delay that would be experienced by up-coming voice packets. The use of novel neural networks and fuzzy systems as estimators of network delay characteristics are presented in this paper. Their performance is analyzed in comparison with a number of traditional techniques for both inter and intratalkspurt adaptation paradigms. The design of a novel fuzzy trend analyzer system (FTAS) for network delay trend analysis and its usage in intratalkspurt playout delay adaptation are presented in greater detail. The performance of the proposed mechanism is analyzed based on measured Internet delays.     

Video Decompression Estimation and Playout Scheme Over the Internet

This paper examines problems associated with display of live continuous media. Under the assumption that the network cannot guarantee the required bounds on delay and jitter and the operating system scheduling is non-realtime, there is a need to accommodate the delay and jitter in the end systems in order to maintain a desirable Quality of Service (QoS). We propose a method of video playback which requires accurate estimation of display cycle time of video frames and the delay suffered by frames in the packet network. We apply various deterministic forecasting methods used in time series analysis on experimental data collected from video transmission. Suitable methods are recommended for display cycle time and delay estimation.     

Fast Bitstream Switching Algorithms for Real-Time Adaptive Video Multicasting

Bitstream switching among multiple bitstreams encoded at different bit rates is an effective way to address the bandwidth variation issue in transmitting multimedia over the Internet or wireless networks. This paper proposes two new fast real-time bitstream switching algorithms that aim to minimize the drifting error, while avoiding the problems of long delay, high complexity and bit-rate overhead for storage and transmission that often occur in prior solutions. The basic idea is to choose a switching point in a neighborhood with the highest encoding quality, within a switching window determined by the switching delay constraint. We show that they can significantly outperform a simple switching algorithm, and achieve performance that is closer to an offline mean-square-error-optimized bitstream switching solution, when compared to our previous work based on the similarity of the reference frames. The proposed schemes are especially useful in the scenario of real-time multicasting over dynamic heterogeneous networks, where multiple bitstreams with different bit rates are generated on the fly and dynamic bitstream switching is required for individual clients     

网络时延预测研究

网络时间延迟是影响网络控制系统性能的一个主要因素,提前知道网络时延对提升网络控制系统的性能有一定的重要作用,网络时延预测准确度的高低直接影响到网络控制系统性能,为了更好的预测数据在网络上传输的时间延迟,满足网络控制系统需要,该文针对互联网中网络时延的预测问题进行了分析,分别用AR模型和Elman神经网络预测网络时延,通过仿真表明,平稳时延的预测AR模型要稍好,但扰动时延的预测Elman神经网络预测准确度及自适应性优于AR模型。     

预估控制下的实时网络遥操作移动机器人

构建了能使操作者通过Internet远程实时控制的移动机器人系统．为了补偿网络时延和抵消其对遥操作系统的影响,基于我们以前提出的改进型Smith预估器原理,采用了预估控制策略．为了保证系统稳定性和透明性,基于主从端的传感器信息交换,设计了一个动态模型管理器,其中模型和力反馈误差调节通过模糊控制实现．除了力反馈外,为了增强遥操作的实时性,引入了预估的虚拟显示．为了精确地预测网络时延,提出了一个新颖的时钟同步算法．为了降低时延抖动,结合我们提出的两个算法,实现了数据缓冲策略．最后,通过长距离的网络遥操作实验验证了系统和控制策略的实用性和有效性．     

基于时槽预定的加权公平调度策略

面向以太网的物理帧时槽交换(Ethernet-oriented physical frame timeslot switching,简称EPFTS)技术以用户域内使用最为广泛的以太网MAC(media access control)帧为运载对象、以定长物理层帧EPF(Ethernet-oriented physical frame)的传输时间为时槽,作为数据传输与交换的基础.针对EPFTS交换技术的特点,提出了一类新的调度策略--时槽加权的公平调度原则(timeslot-reservation based weighted fair scheduling,简称TRWFS),以解决EPFTS交换机中的业务数据调度问题.TRWFS以连接建立阶段各业务流预定的时槽数为基础,控制交换矩阵仲裁过程中各输入端向输出端请求转发信元的时刻,借用一般轮询算法的二相迭代机制来解决端口冲突问题.还给出了TRWFS的3种实现算法,表明TRWFS的实现复杂度可与一般Round-Robin调度算法相当.仿真实验结果进一步表明,即使在重负载条件下,TRWFS仍可有效保障EPFTS交换机各端口对上的预定时槽数,并在平均传输时延和吞吐率保障方面优于其他经典调度算法.     

Neural network prediction model for a real-time data transmission

Both the real-time transmission and the amount of valid transmitted data are important factors in real-time multimedia transmission through the Internet. They are mainly affected by the channel bandwidth, delay time, and packet loss. In this paper, we propose a predictive rate control system for data transmission, which is designed to improve the number of valid transmitted packets for the transmission of real-time multimedia over the Internet. The one-step-ahead round-trip delay time and packet loss are predicted using a prediction algorithm and then these predicted values are used to determine the transmission rate. A real-time multimedia transmission system was implemented using a TCP-friendly algorithm, in order to obtain the measurement data needed for the proposed system. Neural network modeling was performed using the collected data, which consisted of the round-trip time (RTT) delay and packet loss rate (PLR). In addition, the performance of the neural network prediction model was verified through a validation process. The transmission rate was determined from the values of RTT delay and PLR, and a data transmission test for an actual system was performed using this transmission rate. The experiment results show that the algorithm proposed in this study increases the number of valid packets compared with the TCP-friendly algorithm.