保障连续媒体流用户层QoS 的缓存控制

研究了缓存控制对媒体流用户层QoS 的影响.多媒体系统信宿端通常采用播放缓存来补偿时延抖动,提高媒体流播放的连续性.缓存控制虽然能够降低时延抖动的影响,却增加了端到端时延.时延或时延抖动是用户可感知的QoS 参数,缓存控制对用户层QoS 的影响究竟如何呢?利用已有的应用层向用户层QoS 映射的研究结果,分析缓存控制参数与端到端QoS 参数、应用层QoS 参数的关系,获得了缓存控制参数与用户层QoS 参数的关系.从理论上深入挖掘缓存控制对用户层QoS 参数的作用,给出了提供确定时延和时延抖动保障的缓存容量值,论证了在网络环境一定时存在提供最佳用户层QoS 的缓存容量值.实验结果验证了分析.     

提高IP电话QoS的综合去抖动同步策略研究

在提供IP电话服务时，时延抖动是一个重要的QoS参数。提出的一种去抖动同步策略综合考虑了如何能达到消除延时抖动的影响，同时又能保证良好的播放实时性。该策略将基于播放时间的同步去抖算法和基于接收缓存数据量控制的再同步算法有机结合并改进，从中获得最佳同步调整，同时还实现了对网络状况的自适应估计。实验证明接收方使用该综合去抖动同步策略能保证接收端语音流平稳连续播放，能使得IP电话的QoS得到很大改善。     

实时流媒体传输中客户终端系统缓冲区设计

网络时延抖动以及时钟偏移等问题,会对媒体流播放是否流畅产生重要影响。从客户终端的角度来讨论如何优化设置和管理缓冲区,以及平滑媒体流的播放,并提出了以SLOW-START为启动模型的动态缓冲控制算法。该算法可以有效地减小起始状态的播放延迟,并有效防止缓冲区上溢造成的播放跳跃以及缓冲区下溢造成的播放停顿。     

空间多媒体通信中音视频同步技术研究

空间多媒体通信具有丢包率高、传输时延长、抖动大、资源受限等特点.为满足音视频同步QoS要求,提出了一种基于RTP/RTCP协议的音视频实时同步算法,以音频流为主媒体流,通过控制视频播放实现音视频同步.该算法无需全网同步时钟与反馈机制,具有较低的算法复杂度,适用于空间多媒体通信.仿真实验结果表明,该同步算法有效提高了空间多媒体通信系统中的音视频媒体间同步性能,满足空间通信的音视频同步QoS性能要求.     

网络环境下视频连续播放控制算法

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.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,最后提出了通过用不连续性和播放失真的方差来衡量同步性能方法,实验结果表明,该算法的同步性能优于Yuang的算法.通过选择合适的参数,可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.     

基于概率估计的自适应媒体播放算法

针对缓冲区下溢造成的视频播放抖动和中断问题,提出一种基于概率估计的自适应媒体播放算法。根据网络信道状态和估计的下溢概率和上溢概率控制视频帧的持续播放时间,适当控制播放速率的变化范围和变化量,减少缓冲区下溢概率和播放时延,实现视频平滑播放。仿真结果证明,该算法性能优于传统的自适应媒体播放算法。     

长期演进系统基于加权平均时延的下行资源调度

针对长期演进（LTE）移动通信系统下行链路传输中多用户的实时（RT）与非实时（NRT）业务传输性能需求问题，提出一种基于用户加权平均时延的改进型的最大加权延时优先（MLWDF）资源调度算法。该算法在考虑信道感知与用户服务质量（QoS）感知的基础上引入反映用户缓冲区状态的加权平均时延因子，该因子通过用户缓冲区中待传输数据与已发送数据的平均时延均衡得到，使具有较大时延和业务量的实时业务优先调度，提升了用户的性能体验。理论分析与链路仿真表明，提出算法在保证各业务时延及公平性的基础上，提升了实时业务的QoS性能，在用户数量达到50的条件下，对比MLWDF算法实时业务的丢包率降低了53.2%，其用户平均吞吐量提升了44.7%，虽牺牲了非实时业务的吞吐量，但仍优于VT-MLWDF算法。实验结果表明，所提算法在多用户多业务传输条件下提升了实时业务的传输性能，并在QoS性能上明显优于对比算法。     

一种新的媒体内同步控制算法

提出了一种新的媒体同步播放方案，该方案是基于发送方的媒体同步控制，根据缓冲区的占用情况来检测失步，并将其反馈给发送方，发送方利用给出的控制函数对发送帧率进行调整，来保持接收方媒体的同步播放，调节算法简单易实现，实验表明该方案能够处理由于网络传输时延特性变化引起的失步，与其它反馈控制方案相比具有更低的数据丢失率，从而使数据表现更为平滑连续。     

一种基于模拟退火方法的多约束QoS组播路由算法

研究了带宽、时延及时延抖动约束最小代价的QoS组播路由问题,提出一种利用模拟退火方法解决该问题的QoS组播路由算法SABDMA。该算法通过选择合适的模拟退火参数迭代求解,以获得满足QoS约束的最小代价组播树。同时,为避免搜索区域的扩大和计算时间的增加,根据时延和时延抖动的关系,提出采用“路径交换”策略在可行解范围内构造邻域集。仿真结果表明该算法具有可行、稳定、收敛快的特点;能根据组播应用对QoS的限制要求,有效地构造代价较低的组播树,具有较强的实时性。     

基于RTP/RTCP协议的实时数据传输与同步控制策略

针对分组交换网络中的实时媒体传输,考虑非QOS保证的分组网络可能带来的传输丢包、乱序和抖动等情况,采用基于RTP/RTCP协议的媒体传输和媒体控制机制,在媒体流中添加时间戳等控制信息,通过播放时延控制算法进行媒体内同步,并在媒体内同步的基础上,根据发送方的绝对时间戳和RTP时间戳的对应关系,确定不同媒体流之间的同步点,从而达到多通道媒体间同步的效果。     

Control of perceived quality of service in multimedia retrieval services: prediction-based mechanism vs. compensation buffers

In multimedia systems end-to-end delay jitter has a great impact on the continuity of information playback. Therefore, it is necessary to introduce appropriate mechanisms to compensate for delay variations, so that the intramedia and intermedia temporal relationships can be preserved. In this paper, two methods for compensation of the network delay jitter in a distributed multimedia retrieval service are compared: the first is based on prediction of the network delay jitter suffered by each information unit and retrieval time modification at the source site; the second is based on a compensation buffer at the destination site. Comparison is made by assuming a master/slave relationship between the monomedia streams composing the multimedia data flow.     

流媒体通信传输算法研究

流媒体技术应用越来越广泛，但数据传输中的延迟、抖动，影响了媒体流播放质量。如何提供保证性能的流媒体服务成为推广流媒体技术的关键。提出了最小延迟算法，可以提供高的信道利用率及高的目的端缓冲区数据吞吐量，提高媒体流播放质量。还提出了最小聚类延迟算法作为改进，进一步优化媒体流整体播放性能。上述两种算法在流媒体技术中有一定的应用推广价值。     

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.     

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.     

Performance analysis of multimedia based web traffic with QoS constraints

During the recent years, there has been a tremendous growth in the development and deployment of multimedia based networked applications such as video streaming, IP telephony, interactive games, among others. These applications, in contrast to elastic applications such as email and data sharing, are delay and delay jitter sensitive but can tolerate certain level of packet loss. A vital element of end-to-end delay and delay jitter is the random queueing delays in network switches and routers. Analysis of robust mechanisms for buffer management at network routers needs to be carried out in order to reduce end-to-end delay for traffic generated by multimedia applications. In this context, a threshold based buffer management scheme for accommodating multiple class multimedia traffic in network routers has been analysed. This technique effectively controls the allocation of buffer to various traffic classes according to their delay constraints. The forms of the joint state probabilities, as well as basic performance measures such as blocking probabilities are analytically established at equilibrium. Typical numerical experiments are included to illustrate the credibility of the proposed mechanism in the context of different quality of service (QoS) grades for various network traffic classes. This model, therefore, can be used as a powerful tool to provide a required grade of service to a particular class of multimedia based web traffic in any heterogeneous network.     

Scheduling and admission testing for jitter-constrained periodic threads

We present new admission tests for periodic real-time threads with explicitly stated deadlines scheduled according to the earliest deadline first (EDF) algorithm. In traditional real-time periodic scheduling, the deadline of a periodic thread is conventionally the end of the current period. In contrast, our tests support periodic threads in which the deadline may be earlier than the end of the current period. In the extreme case, the deadline may be specified as identical to the per period execution time, which results in perfectly isochronous periodic threads. The provision of such threads, which we refer to as jitter-constrained threads, helps end-systems to honour jitter as well as throughput-related QoS parameters in distributed multimedia systems. In addition, such threads can reduce end-to-end delay and buffer memory requirements as less buffering is needed to smooth excessive delay jitter.     

Dejittering in the transport of MPEG-2 and MPEG-4 video

When multimedia information is transported over a packet-switched network, the quality of presentation can be degraded due to network delay variation or jitter. This paper presents a dejittering scheme that can be used in the transport of MPEG-4 and MPEG-2 video to absorb any introduced network jitter, thus preserving the presentation quality of transported media streams. The dejittering scheme is based on the statistical approximation of delay variation in the arrival times of video packets carrying encoded clock reference values and a filtering and re-stamping mechanism. In addition, a brief overview of the MPEG-4 system is presented.     

嵌入式流媒体客户端缓存技术

介绍了基于嵌入式微处理器S3C2440的嵌入式流媒体系统的硬件结构和工作流程. 服务器端通过RTP/RTCP协议将流媒体数据发送出去,客户端对收到的数据进行解压并实时播放. 将接收缓存分成接收缓冲区、播放缓冲区和DMA缓冲区,三个缓冲区的大小按1：1：2的比例设置,通过平均速率、延时抖动和解码码率等参数来约束缓冲区的容量. 在接收缓冲区设置两个临界点,通过对两个临界点的检测,来辅助调节发送端的数据发送速率. 既可以避免网络拥塞,又可以提高流媒体的传输质量.     

Delay-jitter control in multimedia applications

The growing needs of multimedia communications are leading to new developments in providing real-time communication with guarantees. Several extensions have been proposed for different layers of the Open Systems Interconnection Reference Model to accomodate these needs. In this paper, we study methods for guaranteeing delay jitter bounds for high-speed networks in the network and application layers of this model. The method proposed for the network layer provides distributed jitter control. The method proposed for the application layer allows the destination application to control delay jitter. We use a simulation to compare the effects on delay jitter in each method for various scenarios, such as constant bit rate, cross traffic, and bursty data. In addition, the buffer space requirements for accommodating real-time channels are monitored at each node in the network.