Evaluation of packet loss impairment on streaming video

Video compression technologies are essential in video streaming application because they could save a great amount of network resources. However compressed videos are also extremely sensitive to packet loss which is inevitable in today's best effort IP network. Therefore we think accurate evaluation of packet loss impairment on compressed video is very important. In this work, we develop an analytic model to describe these impairments without the reference of the original video (NR) and propose an impairment metric based on the model, which takes into account both impairment length and impairment strength. To evaluate an impaired frame or video, we design a detection and evaluation algorithm (DE algorithm) to compute the above metric value. The DE algorithm has low computational complexity and is currently being implemented in the real-time monitoring module of our HDTV over IP system. The impairment metric and DE algorithm could also be used in adaptive system or be used to compare diffeient error concealment strategies.

Secure Packet Video Transmission over Wireless Multimedia Sensor Networks

This paper proposes a packet scheduling scheme thatoptimizing the coded video transmission overmultipath wireless multimedia sensor networks interms of received video distortion and power efficiencyenhances the security aspects of the underlyingsystem.When the aggregate transmission rateavailable at the network cannot support the requiredtransmission rate,the scheduling algorithm can selectivelydrop combinations of video packets prior totransmission to adapt the rate of the sender to thelimitations of the wireless channel capacity.Twoscheduling algorithms are proposed.The Baselinescheme utilizes a novel distortion prediction modeland decides upon which packet can be dropped priorto transmission based on the packet’s impact on thevideo distortion.In addition to the bandwidthlimitations,the Power aware packet scheduling is an extension of the Baseline capable of estimating thepower that will be consumed by each node during thetransmission;hence it can control the power consumptionby selectively drop packets of low importanceto the decoded video.Simulation results indicatethe efficiency of the proposed scheduling schemein terms of received video distortion(PSNR)andpower consumption.

A channel distortion model for video over lossy packet networks

Error-resilient video communication over lossy packet networks is often designed and operated based on models for the effect of losses on the reconstructed video quality. This paper analyzes the channel distortion for video over lossy packet networks and proposes a new model that, compared to previous models, more accurately estimates the expected mean-squared error distortion for different packet loss patterns by accounting for inter-frame error propagation and the correlation between error frames. The accuracy of the proposed model is validated with JVT/H.264 encoded standard test sequences and previous frame concealment, where the proposed model provides an obvious accuracy gain over previous models.

Multiple description video coding using GOB alternation and low quality macroblock update

To combat packet loss and realize robust video transmission over Intemet and wireless networks, a new multiple description （MD） video coding method is proposed. In the method, two descriptions for each video frame is first created by group of blocks （GOB） alternation. Motion information is then duplicated in both the descriptions and a process called low quality macroblock update is designed to redundantly encode textures in each frame using standard bit stream syntax. In this way, the output bit streams are standard compliant and better trade-offs between redundancy and single charmel reconstruction distortion are achieved. The proposed method has much better performance than the well-known MD transform coding （MDTC） method both in terms of redundancy rate distortion, and in the packet loss scenario.

Coping with handover effects in video streaming over cellular networks

The 3rd generation partnership project (3GPP) has defined the protocols and codecs for implementing media streaming services over packet-switched 3G mobile networks. The specification is based on IETF RFCs on audio/video transport.It also adds new features to achieve better adaptation to the mobile network environment. In this paper, we propose an algorithm for handover detection and fast buffer refill that is based on the existing feedback and signaling mechanisms. The proposed algorithm refills the receiver buffer at a faster pace during a limited time frame after a hard handover is detected in order to achieve higher video quality.

Network-adaptive HD MPEG-2 video streaming with cross-layered channel monitoring in WLAN

In this paper, we propose a practical design and implementation of network-adaptive high definition (HD) MPEG-2 video streaming combined with cross-layered channel monitoring (CLM) over the IEEE 802.11a wireless local area network (WLAN). For wireless channel monitoring, we adopt a cross-layered approach, where an access point (AP) periodically measures lower layers such as medium access control (MAC) and physical (PHY) transmission information (e.g., MAC layer loss rate) and then sends the monitored information to the streaming server application. The adaptive streaming server with the CLM scheme reacts more quickly and efficiently to the fluctuating wireless channel than the end-to-end application-layer monitoring (E2EM) scheme. The streaming server dynamically performs priority-based frame dropping to adjust the sending rate according to the measured wireless channel condition. For this purpose, the proposed streaming system nicely provides frame-based prioritized packetization by using a real-time stream parsing module. Various evaluation results over an IEEE 802.11a WLAN testbed are provided to verify the intended Quality of Service (QoS) adaptation capability. Experimental results showed that the proposed system can mitigate the quality degradation of video streaming due to the fluctuations of time-varying channel.

RTP payload format for H.264/SVC scalable video coding

The scalable extension of H.264/AVC, known as scalable video coding or SVC, is currently the main focus of the Joint Video Team’s work. In its present working draft, the higher level syntax of SVC follows the design principles of H.264/AVC. Self-contained network abstraction layer units (NAL units) form natural entities for packetization. The SVC specification is by no means finalized yet, but nevertheless the work towards an optimized RTP payload format has already started. RFC 3984, the RTP payload specification for H.264/AVC has been taken as a starting point, but it became quickly clear that the scalable features of SVC require adaptation in at least the areas of capability/operation point signaling and documentation of the extended NAL unit header. This paper first gives an overview of the history of scalable video coding, and then reviews the video coding layer (VCL) and NAL of the latest SVC draft specification. Finally, it discusses different aspects of the draft SVC RTP payload format, in- cluding the design criteria, use cases, signaling and payload structure.

Video quality based link adaptation for low latency video transmission over WLANs

Wireless Local Area Networks (WLANs) such as IEEE 802.11a/g and Hiperlan/2 utilise numerous transmission modes, each providing different throughputs and reliability levels. Many link adaptation algorithms proposed in the literature either maximise the error-free data throughput based on channel conditions or are based on the number of failed transmissions. However, these algo- rithms do not take into account the content of the data stream and strongly rely on the use of Automatic Repeat Requests (ARQs). Low latency video applications such as real-time video transmission may require no retransmission, or only a limited number of retrans- missions. Moreover, completely error-free communication is not essential, especially if robust video compression techniques are applied. In such scenarios, improved decoded video quality can be obtained with a video stream transmitted at a higher bit rate using a higher link speed but with some degree of transmission error, rather than an error-free video stream at a lower bit rate using a lower link speed. In this work, we investigate a link adaptation scheme that improves the Quality of Service (QoS) for video transmission, based on the overall received video quality (Peak Signal to Noise Ratio, PSNR), rather than by maximising the error-free throughput. We also study a practical link adaptation approach that uses PER thresholds at the PHY layer. An empirical study showed that thresholds for switching from one mode to another are much lower (almost error free) than those currently used by throughput based schemes. We show that traditional link adaptation strategies are not appropriate for real-time video transmission with no retransmis- sion. Simulation results using the H.264 video compression standard over IEEE 802.11a are presented.

Joint Optimization Strategy for Video Transmission over Distributed Cognitive Radio Networks

A novel joint optimization strategy for the secondary user( SU) was proposed to consider the short-term and long-term video transmissions over distributed cognitive radio networks( DCRNs).Since the long-term video transmission consisted of a series of shortterm transmissions, the optimization problem in the video transmission was a composite optimization process. Firstly,considering some factors like primary user's( PU's) collision limitations,non-synchronization between SU and PU,and SU's limited buffer size, the short-term optimization problem was formulated as a mixed integer non-linear program( MINLP) to minimize the block probability of video packets. Secondly,combining the minimum packet block probability obtained in shortterm optimization and SU's constraint on hardware complexity,the partially observable Markov decision process( POMDP) framework was proposed to learn PU's statistic information over DCRNs.Moreover,based on the proposed framework,joint optimization strategy was designed to obtain the minimum packet loss rate in long-term video transmission. Numerical simulation results were provided to demonstrate validity of our strategies.

一种基于深度报文检测的合成视频质量评估方法

移动视频业务快速发展,用户感知与网络KPI指标发生偏离,传统视频运维支撑手段存在端到端问题定界难的问题。本文对国内电信运营商移动视频业务特性进行分析,提出一种基于深度报文检测的视频合成算法,将事务级的DPI单据进行关联,以一个完整的视频业务为单位输出相关体验评估信息,进而研究以业务级为基础的DPI指标,用于评估用户观看完整视频的全过程感知,提升用户感知保障。此方法能够覆盖主流的视频网站资源,本文所提出的方法经过实际应用验证,可真实评估视频质量及用户感知体验,助力实现国内电信运营商通信网络问题定界与网络自动驾驶。

一种嵌入视频流的丢包自测量方法

现有的丢包主动测量方法是通过探测流的丢包信息去推测网络的丢包特性,进而推测特定应用流的丢包,测量结果不能准确获知某一给定应用流的丢包情况.由于丢包通常属于短时间、小概率事件,要更加准确地测量丢包就意味着需延长测量时间,或者提高探测流的发送速率以及时发现丢包,这将不可避免地增加网络的额外负载.分析了不同类型帧损伤的影响,并以MPEG-4,H264视频为研究对象,通过对其码流结构特点及RTP封装策略的分析,提出一种将测量信息嵌入到视频用户数据域(User_Data)的丢包测量方法 PLBU(packet loss measurement based on User_Data).该方法利用视频码流信息完成对丢包的探测,不影响视频的正常播放,不需要注入新的探测流,极大地降低了因测量而引入的额外负载.NIST Net及Planetlab等实验结果表明,该算法不仅丢包测量准确性高,且可提供丢包所属视频帧类型等信息,如视频中I,P,B帧的数据包丢失的情况.借助该测量方法,服务提供商可评测网络视频流丢包,并分析视频体验质量(QoE)变化情况,且不受视频流在网络传输中的优先级影响.

一种网络丢包的无参考视频质量评估方法

针对网络视频质量评估需要满足实时性高、资源开销小等要求,在研究不同类型帧的丢包对视频质量影响的基础上,提出一种不需要对视频解码的无参考网络视频质量评估模型.该方法不必对接收到的码流解码,只通过码流的包头分析即可得到视频的时域复杂度;并通过研究不同帧类型的丢失对视频质量的影响,提出一种无参考的网络视频质量评价模型来预测网络失真视频质量.该方法不仅考虑到人眼视觉系统的空域掩盖效应和时域掩盖效应对视频质量的影响,而且也考虑了不同类型的帧的丢包对视频质量的影响.提出的模型不需要对视频进行完全解码,具有资源开销小、实时性好的特点,适合对实时传输的视频流进行质量评估.实验结果表明,使用该视频质量评估方法测得的视频丢包失真质量评分与其主观质量评分有很好的一致性,相比于国际标准G.1070中的丢包视频质量评价模型,无参考的网络视频质量评价模型与主观质量评价的相关性平均提高了6.96%.

一种视频质量监测系统的设计与实现

网络服务提供商希望能从用户的角度了解目前网络所提供的服务质量,而用户也希望获得定量的指标来评价当前网络服务质量。为此,以视频质量监测为研究对象,提出一种面向用户体验质量的网络监测系统。通过实验分析了网络传输过程中QoS参数对视频QoE的影响;提出一种将视频流转化为测试序列的视频丢包测量方法,该方法能低入侵、准确测量视频传输过程中的丢包情况;基于以上的研究成果,通过对MIB库的扩展和对MIB库轮询机制的研究,构建了面向QoE的视频服务监测系统,该监测系统结构简单、可行性强,实验表明可实时对网络中的视频服务质量进行监测。

一种新的IP网络视频通信丢包错误纠正方案

分组丢失是IP交换网络常见的现象。该文针对下一代网络多媒体通信,提出了一套IP网络抗分组丢失的方案。包括纠删编码、打包和发送以及接收端错误掩盖。与常见的纠删编码相比,该文提出的[9,5,3]码具有构造简单,编解码时间短,纠删能力高的优点。打包和发送方案根据[9,5,3]码的纠删性能对视频数据打包,并以一定顺序发送,提高了纠删码抗突发错误的能力。采用新的视频编码方案H.264所提供的抗误码工具——灵活的宏块次序(FMO),对纠删失效的视频数据在解码端进行错误掩盖。

基于对象的H.264视频编码抗分组丢失方法

为了改善视频在IP网络传输时分组丢失造成的质量下降,提出了一种计算复杂度低且简单易行的视频对象划分新方法,以宏块编码类型作为视频对象划分的依据,将图像划分为前景和背景两个视频对象.结合H.264新的抗误码技术———灵活的宏块次序,将前景和背景对象分别置于不同的Slice组中,使视频对象可独立解码并采用不同的抗分组丢失方案.在此基础上提出了基于视频对象的信源-信道联合编码方法,加强对前景对象的强制Intra刷新率,并根据信道丢包率采用不同码率的信道编码对视频对象分别进行保护,有效地提高了分组丢失网络中的视频通信质量.

基于视觉注意力变化的网络丢包视频质量评估

针对网络中受丢包损伤的视频提出了一种基于视觉注意力变化的全参考客观质量评估方法.该方法基于视觉显著性检测在视频数据上的应用,考察受网络丢包失真影响的视频数据与标准参考数据在空间和时间上引起的视觉注意力变化,并根据此变化相应的视觉显著性在空间和时间上的差异,提出了一组客观质量评估方法.文中采用17个受丢包损伤的视频数据进行测试,并实施了主观评价实验作为评价标准.与传统的没有考虑人眼视觉显著特性的质量评估方法,以及目前主流的基于视觉显著区域/感兴趣区域对失真像素进行加权的方法进行对比,实验结果表明,基于视觉注意力变化的方法较后两者与主观质量评估结果有更好的相关性,能够更有效地评估丢包损伤视频的质量.

视频会议中的同步缓冲设计

接收缓冲区对于保证视频流的播放质量起着关键的作用。针对接收实时视频流这种特殊需求,本文提出了一种时间驱动的具有同步功能的缓冲区设计方案。经过实验测试,缓冲区在平滑视频呈现、解决恶劣网络环境下的丢包乱序等问题有良好的效果,已经应用到某大型网通视频直播接收系统中,可以广泛应用于任何接收实时视频流的客户端系统中。

基于丢包率预测的视频传输纠错算法

为解决视频图像在互联网中进行传输时,其质量易受网络丢包率、时延等因素的影响而显著降低的问题,提出了一种基于丢包率预测的视频传输纠错算法。该算法采用隐马尔可夫模型预测网络丢包率,根据丢包率的大小自适应地选择FEC或ARQ对视频图像进行纠错操作。当预测出的丢包率较高时,为避免FEC算法在丢包率较高时降低带宽利用率,采用选择性ARQ算法恢复丢失的视频数据包,并通过限制其重传次数使视频传输的实时性得到了保证;当预测出的丢包率较低时,则采用优化了RS冗余值的FEC算法进行纠错操作。在OPNET modeler中进行的仿真实验表明,与HARQ算法相比,使用该纠错算法,视频图像的PSNR的平均值提高了1.6 dB,平均时延减少了0.24 s左右。该算法不但降低了视频传输的平均时延和丢包率,而且提高了接收端视频图像的重建质量,具有复杂度低、实现简单的特点。

基于TMS320C6201的数字视频采集卡设计

传统的数字视频采集卡通常只具备视频采集功能,而不具备视频处理功能。介绍了一块基于TI公司C6000系列DSP芯片的数字视频采集卡的设计,该卡具备自主的数字视频处理功能,通过PCI接口与主机通信。

网络丢包对多视点3D视频质量的影响

为研究网络传输差错对多视点立体视频质量的影响,分析多视点视频在丢包网络中的差错传播并推导多视点视频差错传播的理论计算公式,通过仿真实验研究丢包和误码对立体视频播放效果的影响。根据人眼所能忍受的图像质量和立体感减弱的程度,近似确定出可接受的立体视频在IP网络传输中的丢包率和错误率。