基于错误传播模型的非均等视频流丢失保护

提出一种适用于丢包网络、面向图像组（GOP）层的非均等视频流丢失保护方案。利用GOP中不同帧之间的非均等显著性,将不同数量前向错误校正包分配到GOP层的不同帧中。采用帧间包交错机制将突发包丢失分散到不同帧上,提高处理突发包丢失时的鲁棒性。仿真结果表明,在不同信道丢失模式下,该方案能提高视频接收质量。     

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.     

Video loss recovery with FEC and stream replication

Packet loss is inevitable in video multicast. In this paper, we propose and study an effective feedback-free loss recovery scheme for layered video which combines forward error correction (FEC) and stream replication. In our scheme, the server multicasts the video in parallel with FEC packets and a number of replicated delayed (ReD) version of the stream. Receivers autonomously and dynamically join the FEC and ReD streams to repair their losses. On the server side, we analyze and optimize the number of replicated streams and FEC packets to meet a certain residual loss requirement (i.e., error after correction). On the receiver side, we analyze the optimal combination of FEC and ReD packets to minimize its loss. We also present a fast yet accurate approximation algorithm for receiver to make such decision. We show that FEC combined with merely one or two replicated streams can effectively reduce the residual error rate (by as much as 50%) as compared with pure FEC or replication alone. Both subjective and objective video measures confirm that our recovery scheme achieves much better visual quality.     

IPv6视频网格中基于用户端的网络性能测量

在具有服务质量保证的IPv6视频网格系统中,视频数据采用TCP协议传输。该文根据TCP协议传输的特点,采用基于客户端的被动测量技术,实现了对用户实际获得的视频流网络性能参数延迟、带宽、抖动的测量,提出了迟到包率的概念和计算方法。     

Estimation of the available bandwidth ratio of a remote link or path segments

Available bandwidth is usually sensitive to network anomalies such as physical link failure, congestion, and DDoS attack. Thus, real-time available bandwidth information can be used to detect network anomalies. Many schemes have been proposed to estimate the end-to-end available bandwidth or end-to-end capacity. However, the problem of estimating the available bandwidth for a specific remote link has not been investigated in detail yet. We propose a new scheme to estimate the available bandwidth ratio of a remote link or remote path segments, a group of consecutive links, without deploying our tool at the remote nodes. The scheme would be helpful in accurately pinpointing anomalous links. Two streams of ICMP timestamp packets are sent to both end nodes of a target link according to a Poisson process, and the available bandwidth ratio for the target link is estimated based on the measured packet delay. Since the proposed scheme needs not incur a short-term congestion, unlike conventional end-to-end available bandwidth estimation mechanisms, the intrusiveness is low and the proposed scheme overcomes the limitation of conventional approaches, inability to probe the links beyond the tight link with the minimum available bandwidth. The performance of the proposed scheme is evaluated by ns-2 simulation.     

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.     

A client-assisted interval caching strategy for video-on-demand systems

In a Video-on-Demand (VoD) system, in order to guarantee smooth playback of a video stream, sufficient resources (such as disk I/O (Input/Output) bandwidth, network bandwidth) have to be reserved in advance. Thus, given limited resources, the number of simultaneous streams can be supported by a video server is restricted. Due to the mechanical nature, the I/O subsystem is generally the performance bottleneck of a VoD system, and there have been a number of caching algorithms to overcome the disk bandwidth limitation. In this paper, we propose a novel caching strategy, referred to as client-assisted interval caching (CIC) scheme, to balance the requirements of I/O bandwidth and cache capacity in a cost-effective way. The CIC scheme tends to use the cache memory available in clients to serve the first few blocks of streams so as to dramatically reduce the demand on the I/O bandwidth of the server. Our objective is to maximize the number of requests that can be supported by the system and minimize the overall system cost. Simulations are carried out to study the performance of our proposed strategy under various conditions. The experimental results show the superior of CIC scheme to the tradition Interval Caching (IC) scheme, with respect to request accepted ratio and average servicing cost per stream.     

基于联邦共识机制的多视频流带宽分配策略

针对瓶颈链路中视频带宽分配不均导致的用户QoE不公平以及带宽利用率低的问题,提出了一种基于联邦深度强化学习的分布式视频流公平调度策略。该策略能够根据客户端网络状态和视频QoE等级动态生成带宽分配权重因子,服务器端的拥塞控制算法则根据带宽分配权重因子为瓶颈链路中的每个视频流分配带宽,以保障瓶颈链路中视频流的公平传输。每个视频终端都运行一个带宽分配agent,且多个agent以联邦学习的方式周期性地训练,以便代理模型能够快速收敛。带宽分配agent通过共识机制同步联邦训练参数,实现了在异步播放请求条件下带宽分配agent模型参数的分布式聚合,并确保了agent模型参数的安全共享。实验结果表明,与最新方案相比,提出策略在QoE公平性和整体QoE效率方面分别提高了10%和7%,这表明提出策略在解决视频流带宽分配不均问题和提升用户体验方面具有潜力和有效性。     

基于集成FEC和层次传输的可靠组播的流控技术

端系统能力和网络带宽的异构性给大规模组播的流控带来很大困难。本文将前向纠错（ＦＥＣ）技术与层次传输相结合,较好地解决了异构环境下可靠组播的流控问题。我们给出了传输调度和信道速率分配的算法,讨论了差错控制问题。性能分析和模拟表明,该方法对大规模、异构组播组可显著减少平均传输时间并且有效地利用网络带宽。只需较少数目的组播组就能得到性能的很大提高。软件ＦＥＣ编码器的速度能够匹配当前的网络条件,算法易于实     

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.     

MPSSF:一种低失序的缓存转发移动切换方案

全IP无线移动网络的微移动协议在无线接入网采用快速切换技术降低移动切换时延，同时采用了数据包缓存转发技术来解决切换过程中的丢包。多流转发方案存在较多的失序，使上层的TCP协议不适当地启动拥塞控制机制而降低吞吐量。单流转发方案虽然没有失序，但是会占用较多的网络资源并增加数据包的时延。提出一种多径单流转发方案MPSSF，较好地解决了移动切换过程中数据包的丢失与失序问题，同时网络资源消耗以及数据包时延也比单流转发方案显著减小。网络模拟实验表明，MPSSF在移动切换时避免了数据包失序，保持了TCP的拥塞窗口，对TCP性能的改善效果优于多流转发方案及单流转发方案。     

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.     

Impact of video scheduling on bandwidth allocation for multiplexed MPEG streams

We present efficient schemes for scheduling the delivery of variable-bit-rate MPEG-compressed video with stringent quality-of-service (QoS) requirements. Video scheduling is being used to improve bandwidth allocation at a video server that uses statistical multiplexing to aggregate video streams prior to transporting them over a network. A video stream is modeled using a traffic envelope that provides a deterministic time-varying bound on the bit rate. Because of the periodicity in which frame types in an MPEG stream are typically generated, a simple traffic envelope can be constructed using only five parameters. Using the traffic-envelope model, we show that video sources can be statistically multiplexed with an effective bandwidth that is often less than the source peak rate. Bandwidth gain is achieved without sacrificing the stringency of the requested QoS. The effective bandwidth depends on the arrangement of the multiplexed streams, which is a measure of the lag between the GOP periods of various streams. For homogeneous streams, we give an optimal scheduling scheme for video sources at a video-on-demand server that results in the minimum effective bandwidth. For heterogeneous sources, a sub-optimal scheduling scheme is given, which achieves acceptable bandwidth gain. Numerical examples based on traces of MPEG-coded movies are used to demonstrate the effectiveness of our schemes.     

一种基于NS-2的Internet视频码流传输研究方案

该文提出了一种基于NS-2网络仿真器的Internet视频码流传输研究方案，该方案通过修改NS-2的业务数据源发生器，将从视频码流中得出的IP分包信息注入NS-2虚拟网络进行传输，能将仿真和研究过程直接与真实码流发生联系。将在目标节点接收到的所有数据包按照时间戳信息重新拼接成一个视频文件，在码流播放器上回放，还可以直观看到网络传输效果，从而能直观论证网络通讯性能。利用该方案还可以进一步对视频和音频的多媒体混合码流传输算法或多媒体传输控制协议进行仿真，对于复杂网络拓扑及相关协议设计与研究将具有极为重要的意义和价值。     

基于TCP友好速率控制和前向纠错的MPEG-2视频传输

针对Internet视频传输面临拥塞控制和数据包丢失的问题，结合TCP友好的速率控制算法和前向纠错机制建立视频传输的分层体系构架和控制策略。传输体系同时采用以GOP为基本分析单元的视频帧速率预测模型，实现根据网络丢包率的变化动态地优化配置前向纠错的冗余信息。实验证明，传输体系采用动态优化的前向纠错能实时地适应带宽的变化，有效地降低数据包丢失带来的影响，从而改善视频回放质量。     

Multipoint-to-point communications for SHE surveillance with QoS and QoE management

The bandwidth-hunger applications of SHE (Smart Home Environment) can take advantage of the multipoint-to-point (MPP) connections to aggregate more bandwidth to gain user-perceived Quality of Experience (QoE) and network Quality of Service (QoS). The receiver-centric transport-layer R²CP (Radial Reception Control Protocol) was proposed to resolve the incapability of the MPP communication in conventional TCP and UDP. However, R²CP has no consideration to discriminate the importance in a packet payload which is critical to QoE and brings an issue for critical data packets that may be dropped in great risk of network congestion. In this paper, we thus present P-R²CP (Prioritized R²CP) to effectively decrease the loss ratio of critical data packets in MPP video streaming while the network is congested. P-R²CP is a cross-layer protocol that considers both the transport-layer issues and the media content’s properties in application-layer. Then, an example on MPP-UVS (MPP ubiquitous video surveillance) is presented as UVS is now a very important Internet application that requires QoS/QoE management to protect lives and assets especially in SHE. Our experiments are conducted on different kinds of surveillance videos over MPP links with different bandwidth and packet loss inserted. The experimental results demonstrate that, as the loss of critical packets is decreased by an order and much less critical data packets are dropped, P-R²CP can highly guard not only QoS but also QoE of SHE surveillance video streaming.     

面向对象分配带宽的极低比特率视频编码器

在低比特率环境下,如何提高图象编码质量是许多人感兴趣的课题,针对此问题,提出了一种面向对象分配带宽（OOBA）的视频编码器,用以提高低比特率环境下编码图象的主观质量,在充分考虑人的主观视觉特性的基础上,该编码器把图象序列分成不同的视频对象,而每个对象以宏块为单位构成,并且对每个对象单独分配大小不同的带宽,以把更多的比特分配人主观感兴趣的对象,来达到在低比特率环境下提高编码图象主观视觉质量的目的。虽然每个对象独立编码,但在比特流的组织上却能与H．263兼容。为了合理地在多个对象间分配带宽,还给出了基于速率－失真（R－D）模型、基于序列分析和比值加权法等3种带宽分配方法。实验结果表明,该编码器与传统的基于帧的编码器相比,在极低的速率下,峰值信噪比虽略有降低,但图象的视觉质量却得到较大的提高。从而表明了该编码器的有效性,也说明在低比特率环境下,仅以客观准则作为评价编码器好坏的标准是不够的。     

基于网络编码和多访问点协作的视频多播方案

随着智能手机和平板等无线设备的普及,基于互联网的视频浏览成为无线设备的重要功能。然而,由于无线网络的链路间存在干扰、可靠性较低,为了提高视频多播传输的可靠性,文中分析了覆盖区域发生重叠的访问点间相互协作对于提高系统增益和系统公平性的作用,提出一种基于网络编码和多访问点协作的视频多播方案。该方案首先将每个视频划分为大小相同的报文段,并利用RLNC机制对每个段的报文进行编码。然后以使用户接收到的报文数量的期望值最大化的同时,实现公平调度为目标,考虑了访问点间的完全干扰和非完全干扰两种模型,将访问点间的传输范围发生重叠时的视频多播问题建模为线性规划优化问题,提出了双阶段启发式算法来进行求解,该算法通过利用多个访问点来获取空间和时间分集增益,提高了数据传输的可靠性,通过允许发生干扰的访问点并行传输数据,提高了系统效用。最后的仿真实验也验证了本文方案在接收报文总量、解码报文数量和公平性方面的有效性。     

Optimized H.264 Video Encoding and Packetization for Video Transmission Over Pipeline Forwarding Networks

Previous works showed that the quality-of-service (QoS) requirements of multimedia applications can be optimally satisfied by pipeline forwarding (PF) by providing end-to-end delay guarantees as well as high network resource utilization. However, the unavoidable mismatch between reserved resources and the unpredictable traffic profile of a video stream has an impact on the resulting application layer quality. Therefore, a new low-complexity H.264 video encoding and packetization scheme based on a distortion-optimized macroblock grouping technique is designed here to maximize the performance of video transmission on PF networks. The scheme considers the perceptual importance of the different parts of the video data to group the most important information in few packets that are the natural candidates to receive the deterministic service provided by PF. Results show peak signal-to-noise ratio (PSNR) gains up to 2.5 dB over traditional video encoding and packetization schemes, as well as more graceful degradation in case of high network load.     

Real-time Internet video using error resilient scalable compressionand TCP-friendly transport protocol

We introduce a point to point real-time video transmission scheme over the Internet combining a low-delay TCP-friendly transport protocol in conjunction with a novel compression method that is error resilient and bandwidth-scalable. Compressed video is packetized into individually decodable packets of equal expected visual importance. Consequently, relatively constant video quality can be achieved at the receiver under lossy conditions. Furthermore, the packets can be truncated to instantaneously meet the time varying bandwidth imposed by a TCP-friendly transport protocol. As a result, adaptive flows that are friendly to other Internet traffic are produced. Actual Internet experiments together with simulations are used to evaluate the performance of the compression, transport, and the combined schemes