无线视频通信中基于队列预测的自适应缓存管理算法

无线网络中实时视频业务对网络环境的变化非常敏感,因此通常需要在接收端采用合理的缓存管理策略来缓解网络波动对用户观看视频造成的影响。提出一种基于队列预测的自适应缓存播放管理机制,该方法通过判断当前缓存队列的状态,并根据实时视频到达率和端到端时延对缓存长度以及播放速率进行综合调整。实验结果表明,所提算法可以随着网络环境的波动自适应地调整缓存大小和播放速率,有效降低视频业务的中断频率和丢帧率。     

移动网络上的流媒体业务

在移动网络上开展流媒体业务，最大的问题是无线环境的时变性容易引起丢包时延，导致缓冲过多、图像质量差和播放中断。因此，应使流媒体系统根据网络带宽的变化．适当调整服务器传送的音视频流。介绍了相应的关键帧技术和动态多码率技术，以及在终端上实现无线流媒体业务的整个流程。     

面向ICN的动态自适应流媒体算法

信息中心网络(information-centric networking,ICN)与传统IP网络的重要区别在于,前者的路由器节点具备缓存能力,能将热门内容推送到更靠近客户的网络边缘,从而降低客户获取内容的时延。然而ICN中的节点缓存特性会导致传统的自适应流媒体协议出现视频抖动甚至播放中断,降低客户的观看体验。提出了ICN中的动态自适应流媒体算法,在保证客户端播放流畅的同时,尽可能地提升视频片段的码率,提升客户的观看体验。通过在现有的ICN平台中进行仿真实验,证明了该自适应算法在性能上优于传统的算法。     

网络环境下视频连续播放控制算法

网络的时延抖动会造成媒体播放的不连续,本文从信宿端的角度来解决视频媒体的同步连续播放,提出了一种自适应的动态媒体播放算法.该方法在接收端设置缓冲区,根据缓冲区的占用情况动态地调整媒体的播放速率,播放速率是根据具体的媒体内同步要求选取的.另外,分析了马尔可夫调制的泊松到达情况下的排队模型,给出了缓冲区门限的选取原则.最后,用不连续性和播放失真的方差衡量同步性能,计算机模拟结果表明,该算法的同步性能优于Yuang的算法.通过选择合适的参数,该算法可以有效地防止缓冲区下溢造成的播放停顿以及缓冲区上溢造成的播放跳跃,从而实现同步平滑播放.     

移动网络中基于两级缓存的流媒体调度算法

提出一种适用于移动通信网的两级缓存流媒体系统结构2CMSA（two—level cache mobile streaming architecture），它突破了移动流媒体系统中终端缓存空间小、无线接入网带宽窄的局限；针对2CMSA结构设计了基于两级缓存的移动流媒体调度算法2CMSS（two—level cache based mobile streaming scheduling algorithm），建立数学模型分析了其性能；仿真实验证明，与原有的移动流媒体系统相比，使用2CMSS调度算法能够有效地节省网络传输开销，降低用户启动时延。     

基于自然数分段的流媒体主动预取算法

提出了基于自然数分段的流媒体主动预取算法,代理服务器向用户传送已被缓存的数据,同时,提前预取没被缓存的数据,提高了流媒体传送质量,减少了播放抖动.根据提出的自然数分段方法,理论分析了代理服务器预取点的位置和代理服务器为此所需要的最小缓存空间.仿真实验表明,在缓存空间相同的情况下,自然数分段方法比指数分段方法具有更高的字节命中率和更低的代理服务器抖动率,而与相同分段方法接近.     

流媒体代理缓存技术的研究

随着Internet技术的发展和普遍应用,流媒体技术在Internet上得到了广泛的应用.对流媒体对象的访问,需要高且稳定的传送速率,网络带宽消耗大且持续时间长,容易给其他类型文件的访问带来影响,若用户过多,还会使初始流媒体服务器过载.代理缓存技术可帮助解决上述问题.文中介绍了流媒体代理缓存的特点,流媒体代理缓存的算法,流媒体代理缓存的评价指标和影响流媒体代理缓存效果的因素.     

保障流媒体时间结构的网络时延分析

流媒体同步对端到端时延和时延抖动提出了确定的要求,而终端抖动缓存一方面能消除时延抖动的影响,一方面却增加了端到端时延,流媒体同步保障对网络时延的要求不明确。论文从概率保障流媒体同步的角度,确定了保障流媒体同步的抖动缓存容量范围,提出了流媒体同步网络保障的充分条件,针对基于Internet VoIP(Voice over IP)业务的实际网络测试结果,给出了应用流媒体同步网络保障充分条件进行同步保障评价的应用实例并验证了其正确性。     

流媒体技术在电视系统中应用

流媒体技术也叫流式传输技术。是指在Inter net和Intranet中使用流式传输技术的连续性媒体 ,如音频、视频或多媒体文件 ,它的核心是串流(Streaming)技术和高压缩技术。流式媒体在播放前并不下载整个文件 ,而只将开始部分内容存入缓存 ,然后就可以实现数据流的随时传送随时播放。当然流式文件也支持在播放前完全下载到硬盘。流媒体技术的主要优势在于 ,它减少了用户的等待时间 ,只是在开始播放时有一些延迟。流媒体实现的关键技术就是流式传输。流式传输定义很广泛 ,现在主要指通过网络传送媒体 (如视频、音频 )的技术总称。其特定含义为通…     

基于缓存窗口和段补丁预取的流媒体动态调度算法

该文提出了一种新的基于缓存窗口和段补丁预取的移动流媒体动态调度算法,采用代理缓存窗口自适应伸缩和分段缓存补丁块方案,实现了移动流媒体对象在代理服务器中缓存的数据量和其流行度成正比的原则。仿真结果表明,对于客户请求到达速率的变化,该算法比传统算法具有更好的适应性,在最大缓存空间相同的情况下,能显著减少通过补丁通道传输的补丁数据,从而降低了服务器和骨干网络带宽的使用,能快速缓存媒体对象到缓存窗口,同时减少了代理服务器的缓存平均占有量。     

一种基于RTCP反馈的3G流媒体速率控制算法

在3G流媒体业务中,缓存数据溢出严重地影响了多媒体画面质量和媒体播放的流畅性,降低了用户对流媒体业务感知的满意度。为了解决这个问题,根据3GPP PSS提出的反馈机制,阐述了一种基于RTCP反馈信息的3G流媒体速率控制算法。通过计算机仿真证明,该算法不仅有效防止了缓存数据上溢,而且保证了发送效率,避免了缓存数据欠载,从而实现了高质量的流媒体服务。     

Media Synchronization and QoS Packet Scheduling Algorithms for Wireless Systems

Wireless multimedia synchronization is concerned with distributed multimedia packets such as video, audio, text and graphics being played-out onto the mobile clients via a base station (BS) that services the mobile client with the multimedia packets. Our focus is on improving the Quality of Service (QoS) of the mobile client's on-time-arrival of distributed multimedia packets through network multimedia synchronization. We describe a media synchronization scheme for wireless networks, and we investigate the multimedia packet scheduling algorithms at the base station to accomplish our goal. In this paper, we extend the media synchronization algorithm by investigating four packet scheduling algorithms: First-In-First-Out (FIFO), Highest-Priority-First (PQ), Weighted Fair-Queuing (WFQ) and Round-Robin (RR). We analyze the effect of the four packet scheduling algorithms in terms of multimedia packet delivery time and the delay between concurrent multimedia data streams. We show that the play-out of multimedia units on the mobile clients by the base station plays an important role in enhancing the mobile client's quality of service in terms of intra-stream synchronization and inter-stream synchronization. Our results show that the Round-Robin (RR) packet scheduling algorithm is, by far, the best of the four packet scheduling algorithms in terms of mobile client buffer usage. We analyze the four packet scheduling algorithms and make a correlation between play-out of multimedia packets, by the base station, onto the mobile clients and wireless network multimedia synchronization. We clarify the meaning of buffer usage, buffer overflow, buffer underflow, message complexity and multimedia packet delay in terms of synchronization between distributed multimedia servers, base stations and mobile clients.     

OFDM系统原理及仿真实现

多媒体系统必须能够限制时延抖动的变化,通过在播放端设置缓冲区以抵消时延抖动,从而保证媒体内及媒体间的同步。根据多源流媒体的分发特点,文章设计了一种分组的媒体缓冲区大小的设置方案,该方案在保证流媒体同步的基础上,巧妙地解决了流媒体不同码率的缓冲区大小设置问题。     

Intelligent Video Traffic Smooth Mechanism for Multimedia Communication

1 Introduction Broad-band integrated service digital networks (B-ISDN) based on the asynchronous transfer mode (ATM) are designed to support a wide variety of multimedia with diverse statistical characteristics and quality of services (QoS). Among the var…     

Intelligent Video Traffic Smooth Mechanism for Multimedia Communication

A principal challenge in supporting real-time video services over ATM is the need to provide synchronous play-out in the face of stochastic end-to-end network delays. In this paper, an intelligent traffic smooth mechanism ( ITSM ) is proposed to meet the continuity requirement which is composed of a back-propagation neural network ( BPNN ) traffic predictor, a play-out buffer, and a fuzzy neural network ( FNN ) based play-out rate determinator. The BPNN traffic predictor online predicts the mean packet rate of the traffic in the future interval ( FI ) and the FNN is designed to adaptively determinate the play-out time according to the number of packets in the buffer and the traffic character predicted. Simulation results show that compared to the window mechanism, ITSM achieves high continuity with accepted delay. Furthermore, ITSM can be adaptively modified to meet the QoS of different kinds of services by FNN parameter training.     

Rate control for consistent visual quality of H.264/AVC encoding

In this paper, a novel rate control scheme with sliding window basic unit is proposed to achieve consistent or smooth visual quality for H.264/AVC based video streaming. A sliding window consists of a group of successive frames and moves forward by one frame each time. To make the sliding window scheme possible for real-time video streaming, the initial encoder delay inherently in a video streaming system is utilized to generate all the bits of a window in advance, so that these bits for transmission are ready before their due time. The use of initial encoder delay does not introduce any additional delay in video streaming but benefits visual quality as compared to traditional one-pass rate control algorithms of H.264/AVC. Then, a Sliding Window Buffer Checking (SWBC) algorithm is proposed for buffer control at sliding window level and it accords with traditional buffer measurement of H.264/AVC. Extensive experimental results exhibit that higher coding performance, consistent visual quality and compliant buffer constraint can be achieved by the proposed algorithm.     

Multi-path content delivery: Efficiency analysis and optimization algorithms

The paper studies the benefits of multi-path content delivery from a rate-distortion efficiency perspective. We develop an optimization framework for computing transmission schedules for streaming media packets over multiple network paths that maximize the end-to-end video quality, for the given bandwidth resources. We comprehensively address the two prospective scenarios of content delivery with packet path diversity. In the context of sender-driven systems, our framework enables the sender to compute at every transmission instance the mapping of packets to network paths that meets a rate constraint while minimizing the end-to-end distortion. In receiver-driven multi-path streaming, our framework enables the client to dynamically decide which packets, if any, to request for transmission and from which media servers, such that the end-to-end distortion is minimized for a given transmission rate constraint. Via simulation experiments, we carefully examine the performance of the scheduling framework in both multi-path delivery scenarios. We demonstrate that the optimization framework closely approaches the performance of an ideal streaming system working at channel capacity with an infinite play-out delay. We also show that the optimization leads to substantial gains in rate-distortion performance over a conventional content-agnostic scheduler. Through the concept of error-cost performance for streaming a single packet, we provide another useful insight into the operation of the optimization framework and the conventional scheduling system.     

无线网络中TCP友好流媒体传输改进机制

为保持无线网络中多媒体业务对TCP的友好性，提出了一种适用于无线网络的动态自适应的流媒体传输速率调节机制。该机制通过在接收端区分网络拥塞丢包和链路错误随机丢包，准确判断网络的拥塞状况结合接收端缓存区占用程度，自适应实施多级速率调节，实现了TCP流友好性和流媒体服务质量（QoS）的折中。由于准确区分出无线链路误码丢包和动态调整流媒体QoS要求，该机制能维持较高的网络利用率。仿真实验结果显示在连接数为2和32，链路误码率从0到0．1变化时TCP,TFRC和吞吐量幅度下降幅度较大，WTFCC幅度下降相对较慢，最大相差达2M；在网络负载重时，尽管链路误码率较低，WTFCC区分链路错误与拥塞丢包，因此，端到端丢包率高于TCP和TFRC，但整体传输吞吐量也高于两者。归一化吞吐量显示WTFCC对TCP流友好。     

一种基于罚因子的DASH调度算法

随着移动互联网的普及,基于DASH的流媒体传输协议的应用越来越广泛。如何在带宽波动较大的移动互联网环境中保证用户实现流媒体的流畅点播,提高用户的体验度是DASH调度算法最主要研究的问题。以提高用户体验度为出发点,结合带宽和缓存深度两方面因素,对带宽预测模型的置信度进行评价。在高置信度情况下,大胆地对网络带宽估计模型的模型参量进行调整;在低置信度情况下,以保护缓冲区深度为目的,谨慎地对模型参量进行调整。这种调整势必会对QoE造成相应的影响,该影响作为"罚因子"反馈回模型置信度的评价,以获得模型参数的动态最优解,得到一种较好的DASH调度算法。     

基于RTP的流媒体传输系统的设计与实现

旨在解决流媒体实时传输的效率和系统的可拓展性问题。介绍了流媒体传输的基本协议RTP,并针对实时传输过程中的特殊要求,设计出了一种流媒体传输的结构模型,在流量控制、组包算法和缓存管理等方面进行研究,给出了详细的实现过程。