FOS: A Funnel-Based Approach for Optimal Online Traffic Smoothing of Live Video

Traffic smoothing is an efficient means to reduce the bandwidth requirement for transmitting a variable-bit-rate video stream. Several traffic-smoothing algorithms have been presented to offline compute the transmission schedule for a prerecorded video. For live video applications, Sen present a sliding-window algorithm, referred to as$SLWIN(k)$, to online compute the transmission schedule on the fly.$SLWIN(k)$looks ahead$W$video frames to compute the transmission schedule for the next$k$frametimes, where$kleq w$. Note that$W$is upper bounded by the initial delay of the transmission. The time complexity of$SLWIN(k)$is$O(Wast N/k)$for an$N$frame live video. In this paper, we present an$O(N)$online traffic-smoothing algorithm and two variants, denoted as$FOS$,$FOS1$and$FOS2$, respectively. Note that$O(N)$is a trivial lower bound of the time complexity of the traffic-smoothing problem. Thus, the proposed algorithm is optimal. We compare the performance of our algorithms with$SLWIN(k)$based on several benchmark video clips. Experiment results show that$FOS2$, which adopts the aggressive workahead heuristic, further reduces the bandwidth requirement and better utilizes the client buffer for real-time interactive applications in which the initial delays are small.     

存储视频带宽平滑算法的研究比较

文章综述了存储VBR视频带宽平滑的各种算法,比较了这些算法的各自特点,归纳、形式化了这些算法模型的更一般化表示。     

基于MPEG-4的流媒体服务器端缓冲区管理

服务器端缓冲技术是流媒体系统中常用的一种应用控制技术，能够平滑网络抖动，减少数据丢失，提高客户端视频播放质量。该文的缓冲区管理增加了码率控制的功能，控制策略能进一步提高视频质量，特别适合于系统资源有限、带宽窄或强实时性的视频流式传输。     

An optimal bandwidth allocation strategy for the delivery of compressed prerecorded video

The transportation of prerecorded, compressed video data without loss of picture quality requires the network and video servers to support large fluctuations in bandwidth requirements. Fully utilizing a client-side buffer for smoothing bandwidth requirements can limit the fluctuations in bandwidth required from the underlying network and the video-on-demand servers. This paper shows that, for a fixed-size buffer constraint, the critical bandwidth allocation technique results in plans for continuous playback of stored video that have (1) the minimum number of bandwidth increases, (2) the smallest peak bandwidth requirements, and (3) the largest minimum bandwidth requirements. In addition, this paper introduces an optimal bandwidth allocation algorithm which, in addition to the three critical bandwidth allocation properties, minimizes the total number of bandwidth changes necessary for continuous playback. A comparison between the optimal bandwidth allocation algorithm and other critical bandwidth-based algorithms using 17 full-length movie videos and 3 seminar videos is also presented.     

基于视点预测的全景视频码率自适应策略

为解决全景视频传输中存在的视频卡顿多、用户体验质量（quality of experience,QoE）低等问题,研究当前主流的视点自适应传输方案,提出一种基于视点预测的码率自适应策略（VPBAS）。首先,构建了一种基于长短期记忆网络和全卷积网络的视点预测模型,模型将视点数据和视频显著性信息进行特征融合,实现不同模态数据的相互补充和修正,提高视点预测的准确率;然后,客户端采用随机森林算法预测当前的可用带宽,并根据视点预测结果和可用带宽信息为视频分块选择码率。最后,客户端把选择的码率信息定期发送给服务器,服务器根据反馈的信息向客户端推送最佳码率的全景视频流,这种交互过程在视频播放期间不断地重复,直至客户端观看完毕。实验结果表明,与现有传输方案相比,VPBAS能有效提高带宽受限情况下的视频观看体验。     

视频传输自适应网络带宽控制策略的研究

为有效利用网络资源,提高视频服务的实时性,视频的传输码率应根据网络可用带宽的变化进行调整;对视频传输自适应可用带宽相关技术展开了研究,探讨了自适应实现方法中需要解决的关键技术,通过分析视频种码率控制方法,提出了一种根据可用带宽自适应调节量化步长,优化视频输出码率的策略;实验结果表明,根据网络中链路的带宽状况,自适应改变视频量化步长可以有效地控制视频输出码率,并且能快速而平滑地将视频输出码率调整到一个与网络带宽相适应的值,保证视频网络传输的平稳性和实时性,提高视频服务质量.     

Reducing bandwidth requirement for delivering video over wide area networks with proxy server

Due to the high bandwidth requirement and rate variability of compressed video, delivering video across wide area networks (WANs) is a challenging issue. Proxy servers have been used to reduce network congestion and improve client access time on the Internet by caching passing data. We investigate ways to store or stage partial video in proxy servers to reduce the network bandwidth requirement over WAN. A client needs to access a portion of the video from a proxy server over a local area network (LAN) and the rest from a central server across a WAN. Therefore, client buffer requirement and video synchronization are to be considered. We study the tradeoffs between client buffer, storage requirement on the proxy server, and bandwidth requirement over WAN. Given a video delivery rate for the WAN, we propose several frame staging selection algorithms to determine the video frames to be stored in the proxy server. A scheme called chunk algorithm, which partitions a video into different segments (chunks of frames) with alternating chunks stored in the proxy server, is shown to offer the best tradeoff. We also investigate an efficient way to utilize client buffer when the combination of video streams from WAN and LAN is considered.     

高速移动场景下的平稳宽带车载视频点播系统设计

针对列车乘客在接入路侧蜂窝移动网络时因多普勒效应而引起的带宽稳定性差的问题,设计了一种高速移动场景下的平稳宽带车载视频点播系统,用户可在手机客户端通过WiFi连接固定于车厢内的车载服务器以下载或上传本地视频资源;此设计中,通信两端均在车厢内部,处于相对静止状态,因此可有效避免多普勒效应;同时,使用了定向增益天线来增强信号在车厢窄长矩形区域的有效覆盖;最终,实验数据表明：该系统在30*10 m的室内场景下,移动终端访问的实时传输带宽基本在35 Mbps到56 Mbps之间且波动较小,可实现稳定流畅的视频服务。     

多视点视频无线资源优化

多视点视频是指在场景中放置多台摄像机,记录下多个视点数据,提供给用户视点选择和场景漫游的交互式媒体应用.多个摄像机从不同视角同时拍摄同一场景得到的一组视频信号,每一个摄像机代表一个不同的视角.可同时传输多个空间角度的视频流到用户端,并合成用户所需要的视域图像.多视点视频是一种新型的具有立体感和交互操作功能的视频,是未来一种极具应用前景的多媒体应用.然而,当前多视点的无线网络带宽分配机制中,都没有考虑大量的、不同解码能力的用户共存时的效率问题.一般情况下,合成视域往往需要至少左右两边(两条以上)的参考视频同时传输到用户端,才能使合成的视域质量不低于直接传输的视点质量,使得网络数据量成倍增加.同时,用户的设备性能影响用户感知质量.网络中手机屏幕和高清大屏显示对网络传输视频数据率的要求不同,必须考虑用户设备解码能力的限制,才能真正提供用户满意的感知质量.本文通过考虑移动无线网络带宽约束,考察不同用户端硬件的解码能力、视域大小及带宽消耗,利用博弈理论,使整体网络资源收益最大.本文分别考虑了几种特定场景下多视点视频传输的无线网络的资源分配.第一,已知网络用户的满意度参数,不考虑带宽的限制(带宽充足),如何确定每个用户需要支付的单位价格.第二,在用户个数不确定的条件下,如何判断出可以接入的用户个数和用户需提供的单位价格.第三,同时考虑用户设备对最大场景复杂度解码能力受限以及网络带宽受限两个约束条件,同时进行用户接入控制和多视点的视频质量优化,使得网络的收益和用户的效用得到最大化.本文对提出的算法进行了理论分析,证明了本文参数设置的合理性.在多视点移动网络资源调度中,本文提出的算法可以方便设置所需的价格参数.从视域大小、价格、用户效用、网络收益等各方面对实验性能进行比较.仿真结果显示本方法在同等实验条件下,多视点用户效用提升分别为5%和12%,网络总体收益增加32%.本文算法可以同时满足网络收益和用户整体效用最优,提高多视点视频在多用户下的网络资源利用率.     

Periodic broadcast and patching services - implementation,measurement and analysis in an internet streaming video testbed

Multimedia streaming applications can consume a significant amount of server and network resources. Periodic broadcast and patching are two approaches that use multicast transmission and client buffering in innovative ways to reduce server and network load, while at the same time allowing asynchronous access to multimedia streams by a large number of clients. Current research in this area has focussed primarily on the algorithmic aspects of these approaches, with evaluation performed via analysis or simulation. In this paper, we describe the design and implementation of a flexible streaming video server and client test bed that implements both periodic broadcast and patching, and explore the issues that arise when implementing these algorithms using laboratory and internet-based test beds. We present measurements detailing the overheads associated with the various server components (signaling, transmission schedule computation, data retrieval and transmission), the interactions between the various components of the architecture, and the overall end-to-end performance. We also discuss the importance of an appropriate server application-level caching policy for reducing the needed disk bandwidth at the server. We conclude with a discussion of the insights gained from our implementation and experimental evaluation.Subhabrata Sen: The work of this author was conducted while he was at the University of Massachusetts.     

异构网络路径上视频的流式平滑传输算法

压缩的视频具有变比特率特性，在流式传输中需要进行率平滑,在Internet上，存储视频在端到端的流式传输中要经过不同特性的异构路径段，这对率平滑算法提出了新的要求。讨论了这种异构传输路径上存储视频的流式传输与率平滑问题，利用异构路径之间提供的缓冲代理，提出了一个考虑异构路径特性的率平滑调度模型及算法，并分析了算法的结果是优化的,随后给出了问题的应用实例和相关实验结果，说明了算法的有效性。     

使用特殊复合距离的选播路由算法

选播成员都是等价的服务器,服务数据的服务质量比作为请求的选播数据报更为重要．使用特殊复合距离的选播路由算法(ASCD)使用跳数、逆向传输延迟、逆向可用带宽以及服务器负载合成的距离来选择路径．不同于其他算法,ASCD使用度量在路径逆向上的值,即从选播数据报目标节点(服务器)到选播数据报源节点(客户)方向,而不是常规从选播数据报的源节点到目的节点方向．ASCD定位的路径和选播成员使选播数据报请求的服务数据能够得到更多路径资源．ASCD还能够在一定程度上平衡服务器负载．     

基于P2P的视频点播系统的设计与实现

目前基于IP的各种互联网应用层出不穷,其中对视频节目的下载需求也逐渐增大,但是传统的视频下载服务主要是采用客户端/服务器模式,服务器以单播的方式和每个客户建立连接。随着客户数目的快速增加,服务器的带宽等资源成为系统瓶颈。P2P技术作为解决这种问题的一个方案,应用日益广泛。本文在对当前流行的几种P2P结构进行研究的基础上,提出下载排序以及播放控制两种算法优化,并以开源软件JTorrent为基础,开发出适合流媒体播放的点播系统。     

On noncausal weighted least squares identification of nonstationary stochastic systems

In this paper, we consider the problem of noncausal identification of nonstationary, linear stochastic systems, i.e., identification based on prerecorded input/output data. We show how several competing weighted (windowed) least squares parameter smoothers, differing in memory settings, can be combined together to yield a better and more reliable smoothing algorithm. The resulting parallel estimation scheme automatically adjusts its smoothing bandwidth to the unknown, and possibly time-varying, rate of nonstationarity of the identified system. We optimize the window shape for a certain class of parameter variations and we derive computationally attractive recursive smoothing algorithms for such an optimized case.     

Online smoothness with dropping partial data based on advanced video coding stream

Most studies of smoothing video stream compute the required bit rate of video transmission to satisfy all the transmitted data. In this paper, our proposed online smoothing with tolerable data dropping algorithm can adjust the bit rate as smooth as possible. Several multimedia encoding schemes, such as advanced video coding (AVC), can support partial data dropping to adapt to available bandwidth network. The AVC stream can be adapted by smoothing algorithm to ensure video quality for a given set of constraints where these constraints may be either static after the session set up or may dynamically change over the session duration. Our algorithm is based on the online minimum variance bandwidth allocation algorithm to look ahead a window of frames, dynamically adjusting the required bit rate such that ensuring smoothness when the buffer encounters underflow or overflow for video stream. Furthermore, we add the scheme of data dropping into this algorithm to increase the possibility of smoothing bit rates. The experimental results show the peak rate, the average ratio of dropped data, and the coefficient of variation for five test sequences with different content characteristics such as the average frame size, the peak/mean ratio of frame size, and the average frame bit rate. Experimental parameters are varied by window sizes and tolerable dropping ratios. The algorithm can significantly reduce the peak rate and the coefficient of variation when the transmitted packets are allowed dropping by a user-defined dropping ratio.     

Real-time data dissemination in mobile peer-to-peer networks

Mobile peer-to-peer networks have found many uses such as streaming of audio and video data. There are circumstances, such as emergency situations and disaster recovery, when real-time delivery is a fundamental requirement. The problem is challenging due to the limited network capacity, the variable transmission rates and the unpredictability with respect to the network conditions in the mobile peer-to-peer network.In this paper we address the problem of real-time data dissemination of multimedia streams in mobile peer-to-peer networks. Four routing algorithms are proposed based on a packet's deadline, priority or a combination of these metrics. They are simulated under different setups in a mobile peer-to-peer network with Bluetooth connectivity and nodes broadcasting audio and video streams using different priorities. We compare the performance of the algorithms using a number of metrics. Detailed experimental results are presented. Based on these results, propositions on the usage of the algorithms and the design of network requirements are presented.     

基于Directshow的流媒体视频混合及网络传输系统

本文基于Directshow和Windows Socket技术,建立了一种流媒体视频混合及网络传输系统,实现了两路视频流媒体的混合,给出了混合视频流网络传输的解决方案。即在一台视频流媒体服务器、一台终端和一条低带宽线路的条件下同时传输两路/多路动态视频流,并且可以随时调整叠加视频的位置和Alpha值;具有客户端简单,节约带宽,传输效率高,传输信号准确清晰的优点。     

端到端MPEG-4 FGS视频TCP友好的平滑传输

着重研究了Internet上MPEG-4 FGS(fine grained scalable)视频流的自适应平滑传输,其主要目的在于,在网络带宽变化的情况下,提供稳定的视频回放质量.提出了一种新的基于TFRC(TCP-friendly rate control)的MPEG-4 FGS端到端视频流传输系统框架,在此框架的基础上,首先假设完整的可用带宽变化已知,并且提出了一种离线的自适应平滑算法.此后,给出一种基于改进的ARAR(autoregressive autoregressive)预测技术的在线自适应平滑算法.最后,以NS-2为实验平台进行了模拟实验.模拟实验表明,提出的离线和在线自适应平滑算法可以充分利用可用网络带宽,并且能够在可用网络带宽持续波动的情况下保证接收方的回放尽可能地平稳,从而达到获得最佳视觉效果的目的.     

Efficient Selective Frame Discard Algorithms for Stored Video Delivery Across Resource Constrained Networks

Video delivery from a server to a client across a network is an important component of many multimedia applications. While delivering a video stream across a resource constrained network, loss of frames may be unavoidable. Under such circumstances, it is desirable to find a server transmission schedule that can efficiently utilize the network resources while maximizing the perceived quality-of-service (QoS) at the client. To address this issue, in this paper we introduce the notion ofselective frame discard at the server and formulate the optimal selective frame discard problem using a QoS-based cost function. Given network bandwidth and client buffer constraints, we develop an O (N log N) algorithm to find the minimum number of frames that must be discarded in order to meet these constraints. The correctness of the algorithm is also formally established. We present a dynamic programming based algorithm for solving the problem of optimal selective frame discard. Since the computational complexity of the optimal algorithm is prohibitively high in general, we also develop several efficient heuristic algorithms for selective frame discard. These algorithms are evaluated using JPEG and MPEG video traces.     

Rate-smoothed schedule with tolerable data dropping for video coding stream

Most algorithms of smoothing schedule compute the required bit rate of video transmission to satisfy all the transmitted data. In this paper, our proposed tolerable data dropping algorithm can adjust transmitting data to fit available bit rate. MPEG-4 with fine grained scalability (FGS) can support partial data dropping to adapt to available bandwidth network. The algorithm is based on the minimum variance bandwidth allocation (MVBA) algorithm proposed by Salehi et al. to compute the bit rate such that still ensuring that the buffer never underflows and overflows for MPEG-4 FGS streams under the limited bandwidth resource. We prove that our proposed algorithm, named MVBADP, is smoother than the MVBA algorithm. The experimental results show the peak rate, the number of rate changes, and the ratio of total dropping data, and the PSNR for four test sequences with different content characteristics. They are varied by buffer sizes and tolerable dropping ratios. We found that the MVBADP algorithm can reduce the peak rate and the number of changes when the transmitted data are dropped by tolerable dropping ratio, especially on the video sequences with the high motion and complex texture characteristic and larger size change of the consecutive frame.