实时流媒体传输技术研究综述*

为推动实时流媒体传输技术的进步,提高Internet流媒体服务质量,将实时流媒体传输流程分为用户层、编码层、流处理层、传输控制层和网络层,分层总结和综述了实时流媒体传输过程中的主要研究问题和研究进展,阐述了一些目前主流的基于UDP和TCP的实时流媒体传输技术,并且比较分析了它们的优缺点。最后指出低比特率、高丢包率与无线、强干扰环境下保证流媒体服务质量是未来的研究方向,移动实时流媒体业务则是实时流媒体传输技术应用的必然趋势。     

RTCP反馈和缓冲区的传输控制算法研究

为了保证流媒体数据的实时稳定传输,借鉴了和式增加积式减少(AMID)算法,并且结合了接收端缓冲区读取和写入数据速率,利用接收端RTCP的反馈信息提出了一种新的RTP流媒体传输的控制算法--基于缓冲区的和式增加积式减少控制算法(buffer based-additive increase multiplicative decrease,BB-AIMD).该算法通过反馈接收方的缓冲区信息,调整了发送方发送速率的增加幅度和减少幅度.使用NS2软件的模拟结果表明,该算法降低了流媒体传输过程中的丢包率,同时也提高了流媒体实时回放的顺畅性.     

基于RTP的H.264视频流实时打包传输的研究

随着无线宽带网络的快速发展,以及高效的视频压缩技术的应用,流媒体的实时高效传输成为亟待解决的问题。本文从视频传输系统模型入手,分析了最新视频编码标准H.264在算法上的层次结构特点以及音、视频实时传送协议RTP的高效性。并且随着适合H.264流的RTP载荷格式的提出,基于RTP的H.264流媒体无线传输渐渐地得到应用。本文成功实现了一种基于RTP的H.264传输算法,实验通过了TD330无线3G模块测试,并且获得良好的图像质量,实现了低时延、较小丢包率的打包算法。     

面向TCP的流媒体传输编码码率自适应算法

由于网络的时变性和异构性,以及在拥塞情况下的高丢包率,利用TCP传输流媒体数据是Internet流媒体分发系统提高流媒体分发质量的首选方案。由于TCP具有超时或错误重传机制,在网络拥塞情况下,难以保证高码率流媒体数据传输的实时性,因此提出一种面向TCP流媒体传输的编码码率自适应算法(TCP_RA)。该算法根据流媒体发送应用层缓冲区读写指针差值调整流媒体发送端的编码码率适应网络带宽的变化。仿真实验对比分析了该算法与基于UDP之上TFRC协议的流媒体传输码率自适应算法在流媒体传输质量上的差别。结果表明,该算法在网络环境较差的情况下有效地提高了流媒体传输质量。并且该算法容易实现,值得推广。     

一种适用于无线网络的流媒体传输机制

为保证无线网络中多媒体数据的传输质量,提出了一种适用于无线网络的流媒体传输机制（WMTCC）。该机制通过发送探测报文区分网络拥塞丢包和链路误码随机丢包,准确判断网络的拥塞状况,实施发送速率调节,保证了流媒体服务质量(QoS)。由于准确区分出无线链路误码丢包,该机制在链路误码率较高时能维持较高的网络吞吐量。仿真实验结果显示在高误码率无线网络中,该机制可以获得更高的吞吐量和更大的拥塞窗口,并且发送速率的变化更加平滑。     

一种基于流媒体的实时调度控制算法QFEC

Internet视频业务的普及和用户越来越高的服务需求推动了实时流媒体业务的迅速发展,流媒体业务的服务质量(QoS)成为业界研究的热点.由于网络的复杂性,流媒体的实时调度控制算法是解决流媒体QoS的关键.结合FEC编码技术和Kalman数字滤波技术,提出一种基于QoS的改进FEC调度传输控制算法——QFEC.该算法根据接收方的状态合理调度流媒体业务,并结合Kalman滤波器原理完成传输速率控制.通过算法状态分析,以及实验数据和性能分析表明,该调度算法能够维持视频数据良好的连续传输,降低视频流的丢包率,显著改善流媒体业务的QoS.     

无线网络中基于流媒体传输的自适应TFRC机制

在无线网络高误码率的环境下, 经典TFRC机制会将无线误码丢包误认为拥塞丢包, 导致吞吐量过度降低. 针对无线网络实时流媒体业务的传输控制问题, 提出了一种改进型动态自适应TFRC机制(Adaptive-TFRC). 它在接收端利用丢包区分参数来真实反映网络的状态(即拥塞或者误码), 然后反馈至发送端, 同时对经典TFRC机制的吞吐量模型公式进行改进, 最终能够根据实时网络条件动态自适应地调节传输速率. 仿真结果表明, Adaptive-TFRC机制能够有效地提高网络吞吐量, 降低实时业务流的延时抖动, 同时能够进一步改善TCP业务的友好性传输, 从而保证无线网络实时流媒体的服务质量.     

基于区分服务的流媒体传输模型

流媒体在网络上的应用经常会受到数据包丢失或错误以及网络带宽资源不足的干扰,使得接收方播放质量受到严重影响。本文建立了一种适合流媒体传输的区分服务模型,该模型能够使高优先级数据流（实时流媒体数据流）占用更多的带宽。仿真结果表明：该策略能使流媒体数据流获得较高的吞吐量和较低的丢包率,有效提高流媒体传输的可靠性和实时性。     

基于区分服务的流媒体传输模型

流媒体在网络上的应用经常会受到数据包丢失或错误以及网络带宽资源不足的干扰,使得接收方播放质量受到严重影响。本文建立了一种适合流媒体传输的区分服务模型,该模型能够使高优先级数据流(实时流媒体数据流)占用更多的带宽。仿真结果表明:该策略能使流媒体数据流获得较高的吞吐量和较低的丢包率,有效提高流媒体传输的可靠性和实时性。     

基于P2P覆盖网的流媒体高质量传输策略

目前，在Internet上传输流媒体还存在许多困难，因为大量的媒体文件传输要求较高的带宽，所以在网络环境无法满足的情况下，容易造成丢包和延迟现象，降低流媒体的传输质量。该文提出了一个流媒体高质量传输策略，从质量控制的差错控制与拥塞控制两方面来降低丢包率，保证可靠的流媒体传输。     

基于模糊逻辑的Internet视频流拥塞控制

为了提高视频传输质量,在Internet上对视频流进行拥塞控制,即利用包发送和接收间隔时间（IPGs）代替丢包率作为拥塞指示,采用模糊逻辑拥塞控制策略（FLC）调整视频发送速率并用遗传算法优化模糊控制规则,提高了拥塞控制性能。仿真结果表明,与TFRC和RAP拥塞控制相比,由于FLC发送速率更平滑、带宽利用率更高,从而减少了丢包,提高了视频传输质量;另外,FLC能够与竞争的TCP流公平地分享带宽并对路由器缓冲区大小保持了很好的鲁棒性。     

Packet loss in peer-to-peer video streaming over the Internet

Peer-to-peer streaming has recently gained attention as an effective solution to support large scale media streaming applications over the Internet. One of the main challenges of peer-to-peer video streaming is the cumulative impact of the Internet packet loss due to the decoding dependency of the compressed video frames. In this paper we study the impact of the Internet packet loss on the performance of peer-to-peer video streaming systems, and analyze the efficiency of various packet loss recovery policies in such systems. Our analytical and simulation results show how the Internet packet loss can affect the performance of peer- to-peer video streaming systems and how different packet loss recovery policies can be effective for such systems. Our analysis results give us some insights that can be used in designing efficient peer-to-peer video streaming systems.     

Channel adaptive video stream switching for broadband wireless links

This paper exploits an H.264/Advanced Video Coding codec’s smooth stream switching to achieve robust video delivery across an IEEE 802.16 (WiMAX) broadband wireless link. As the wireless channel conditions vary over time, dynamic selection of bitrate and corresponding video quality will reduce the risk of harmful packet loss. The paper investigates choice of stream-switching with Secondary SP-frames or SI-frames relative to the selection of quantization parameter (QP) values. To control the switching points at the WiMAX server, the proposed scheme applies a feedback mechanism that monitors packet loss. As an additional suggestion, the paper considers an adaptive ARQ scheme to protect switching frames against packet loss. In summary, the broadband wireless streaming system gives more protection to higher quality video, reduces delay and packet loss, and improves received video quality. In particular, for the QP values selected, results show that increased quality primary-switching frames together with SI frames bring a significant gain in video quality compared with other switching schemes with secondary SP-frames. The other schemes in turn show an improvement to using ‘no switching’ when streaming takes place over a typical WiMAX channel with burst errors. Link delay is also reduced.     

利用运动强度自适应传输视频内容

在异构的网络环境下,如何自适应地传输满足多种终端设备和不同用户需求的视频内容具有重要的意义.以达尔文流媒体服务器为实验平台,提出一套基于运动强度的视频自适应传输策略.在服务器端通过检测丢包率来控制发送速率等级,避免网络拥塞,并采取一定策略消除因丢帧而产生的马赛克;在自适应策略中加入了运动强度信息,针对不同的运动强度级别进行不同的处理,使得综合视频质量得到提高.实验结果表明,文中的自适应策略不仅可以避免网络拥塞,而且可以改善视频播放质量.     

矿用实时视频传输系统设计

针对现有煤矿井下视频传输系统存在视频清晰度低、传输速率不稳定、兼容性差等问题,设计了一种矿用实时视频传输系统。该系统采用960 nm红外激光作为辅助光源,利用MCCD图像传感器采集视频信号,提高了低光照强度或黑暗环境下视频清晰度;通过视频解码模块TVP5150将采集的PAL制式模拟视频信号转换为YUV数字信号,数字信号经多格式编码器进行H.264压缩编码,并在此基础上添加UDP报文头进行RTP封装,提高了视频数据传输的时效性;通过Live555流媒体服务器进行数据流化,使用ONVIF标准封装RTSP视频流,通过Socket网络编程实现实时视频流数据网络传输,提高了系统兼容性和传输速率稳定性。测试结果表明,该系统视频传输速率为2.190 Mbit/s,丢包率约为1.256%,达到实时视频传输要求。     

Rate-distortion optimized distributed packet scheduling of multiple video streams over shared communication resources

We consider the problem of distributed packet selection and scheduling for multiple video streams sharing a communication channel. An optimization framework is proposed, which enables the multiple senders to coordinate their packet transmission schedules, such that the average quality over all video clients is maximized. The framework relies on rate-distortion information that is used to characterize a video packet. This information consists of two quantities: the size of the packet in bits, and its importance for the reconstruction quality of the corresponding stream. A distributed streaming strategy then allows for trading off rate and distortion, not only within a single video stream, but also across different streams. Each of the senders allocates to its own video packets a share of the available bandwidth on the channel in proportion to their importance. We evaluate the performance of the distributed packet scheduling algorithm for two canonical problems in streaming media, namely adaptation to available bandwidth and adaptation to packet loss through prioritized packet retransmissions. Simulation results demonstrate that, for the difficult case of scheduling nonscalably encoded video streams, our framework is very efficient in terms of video quality, both over all streams jointly and also over the individual videos. Compared to a conventional streaming system that does not consider the relative importance of the video packets, the gains in performance range up to 6 dB for the scenario of bandwidth adaptation, and even up to 10 dB for the scenario of random packet loss adaptation.     

Multiple TFRC Connections Based Rate Control for Wireless Networks

Rate control is an important issue in video streaming applications for both wired and wireless networks. A widely accepted rate control method in wired networks is equation based rate control , in which the TCP friendly rate is determined as a function of packet loss rate, round trip time and packet size. This approach, also known as TCP friendly rate control (TFRC), assumes that packet loss in wired networks is primarily due to congestion, and as such is not applicable to wireless networks in which the bulk of packet loss is due to error at the physical layer. In this paper, we propose multiple TFRC connections as an end-to-end rate control solution for wireless video streaming. We show that this approach not only avoids modifications to the network infrastructure or network protocol, but also results in full utilization of the wireless channel. NS-2 simulations, actual experiments over 1$times$RTT CDMA wireless data network, and and video streaming simulations using traces from the actual experiments, are carried out to validate, and characterize the performance of our proposed approach.     

基于视频发送端速率控制的QoS技术研究

基于尽力而为的网络模式不能提供QoS保证,网络拥塞和分组丢失不可避免。在端到端视频单播结构下,论文提出了一个发送端速率控制框架SRCF,在此框架下首先利用RTCP报文中的字段提出了一种网络参数测量方法,然后设计了一个自适应速率算法SRCA,SRCA利用已得到的网络传输延迟和分组丢失率参数作为初始参数,来调整编码速率,达到充分利用带宽的目的,避免了视频质量由于调整参数带来的剧烈抖动。仿真结果表明,该算法在网络出现一定拥塞的条件下,能跟踪带宽的变化,网络和媒体QoS能保证视频质量较好。     

Channel Aware Multiuser Scalable Video Streaming Over Lossy Under-Provisioned Channels: Modeling and Analysis

In this paper, we analyze the performance of media-aware multiuser video streaming strategies in capacity limited wireless channels suffering from latency problems and packet losses. Wireless video streaming applications are characterized by their bandwidth-intensity, delay-sensitivity, and loss-tolerance. Our main contributions include i) a rate-minimized unequal erasure protection (UXP) scheme, ii) an analytical expression for packet delay and play-out deadline of UXP protected scalable video, iii) a loss-distortion model for hierarchical predictive video coders with picture copy concealment, iv) an analysis of the performance and complexity of delay-aware, capacity-aware, and optimized UXP streaming scenarios, and v) we show that the use of unequal error protection causes a rate-constrained optimization problem to be nonconvex. Performance evaluations using a 3GPP network simulator show that, for different channel capacities and packet loss rates, delay-aware nonstationary rate-allocation streaming policies deliver significant gains which range between 1.65 dB to 2 dB in average Y-PSNR of the received video streams over delay-unaware strategies. These gains come at a cost of increased offline computation which is performed prior to the start of the streaming session or in batches during transmission and therefore, do not affect the run-time performance of the streaming system.      

Distributed Collaboration for Enhanced Sender-Driven Video Streaming

We propose a sender-driven system for adaptive streaming from multiple servers to a single receiver over separate network paths. The servers employ information in receiver feedbacks to estimate the available bandwidth on the paths and then compute appropriate transmission schedules for streaming media packets to the receiver based on the bandwidth estimates. An optimization framework is proposed that enables the senders to compute their transmission schedules in a distributed way, and yet to dynamically coordinate them over time such that the resulting video quality at the receiver is maximized. To reduce the computational complexity of the optimization framework an alternative technique based on packet classification is proposed. The substantial reduction in online complexity due to the resulting packet partitioning makes the technique suitable for practical implementations of adaptive and efficient distributed streaming systems. Simulations with Internet network traces demonstrate that the proposed solution adapts effectively to bandwidth variations and packet loss. They show that the proposed streaming framework provides superior performance over a conventional distortion-agnostic scheme that performs proportional packet scheduling on the network paths according to their respective bandwidth values.