双重控制策略在流媒体连续播放中的应用

传统的发送速率控制和播放速率控制由于各自的局限性,很难克服网络时延扰动对流媒体连续播放的影响.为了改善播放系统的性能,将双重控制策略应用到流媒体播放中,在发送端发送速率控制采用了内模控制器以克服传输延迟的影响,在接收端播放速率控制采用了简单的PID控制器,两种控制共同调节缓冲区长度.实验表明该方法能够处理传输时延和网络扰动引起的失步对播放的影响,缓冲区长度能够稳定在合理的区间,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.发送速率变化率较小,有利于避免网络拥塞的发生.尤其是在网络出现大的扰动时,与其他方法相比,控制效果更加理想,播放更加流畅.     

基于AVS视频流媒体传输系统的设计与实现

随着新一代视频编码标准AVS的发展,如何在IP网上实时传输AVS视频流对于IPTV,对于视频点播系统的应用有重要意义.利用RTP/RTCP协议设计并实现了一种基于AVS编码标准的实时视频传输系统,通过应用排队论采设计接受端缓冲区并将接收端缓存区的状态信息反馈到发送端控制发送端的发送速率,从而减少由于接收端缓存溢出带来的数据包的丢失,提高传输的QoS质量.     

基于缓冲区控制的TCP友好速率控制机制

TFRC因没有考虑流媒体传输过程中发送速率对接收端缓冲区的影响,导致其在保证TCP友好性的前提下盲目的增长发送速率,造成缓冲区溢出、占用过多的带宽等问题.由此在TFRC基础上提出了基于缓冲区控制的TCP友好速率控制机制BC-TFRC.该机制的发送端在传输过程中根据接收端反馈的缓冲区水位状态和自身检测的网络状态调整发送速率.实验结果表明,BC-TFRC能解决TFRC存在的问题,在保证流媒体播放质量的同时又不多占用网络带宽.     

基于RTP／RTCP的音视频同步方法研究

分析音频和视频同步的标准以及造成音视频不同步的原因,提出一种基于RTP／RTCP协议的音视频同步控制方法。以音频为主媒体,视频为从媒体,在接收端通过设置缓冲区实现音频流的媒体内同步,通过比较音视频数据包的时间戳判断同步关系,对视频采取重复或丢弃的方法实现音视频同步控制。在实验环境下模拟音视频流经网络传输后在接收端经过同步控制后播放和直接播放两种情况,表明,提出的同步方法能很好的纠正音视频不同步的现象。     

面向TCP友好性的IP网络实时流媒体流预测控制

为了解决IP网络实时流媒体流控中既要避免网络拥塞又要将接收端缓冲区长度控制在合理范围的难题, 本文提出了一种面向TCP友好性的IP网络实时流媒体流预测控制算法. 该算法将IP网络实时流媒体流控转化为具有TFRC约束时变不确定条件的预测控制问题, 使用改进的动态矩阵控制算法(DMC)进行优化求解. 仿真实验结果表明该方法能够适应IP网络的复杂环境, 满足了TCP友好性要求, 有效地消除了网络传输抖动对播放带来的不利影响, 具有良好的实用性.     

基于端系统的视频流组播速率控制方法

针对目前Internet未对多媒体应用提供QoS保障的问题,分析了视频流组播的难点,提出了一种基于缓冲区管理的网络自适应组播发送速率控制方法.该机制可以合理控制服务器的发送速率,既能自适应网络状况的变化,又满足流媒体实时播放的需求.实验结果表明,该机制通过控制发送缓冲区占有率,降低了分组丢包率,提高了终端的接收质量,具有良好的实用价值.     

在接收端实现流媒体同步控制的研究

为了更好的解决音视频同步问题,提出了一种在RTP/RTCP协议的基础上,在接收端实现流媒体同步控制的方法。根据音视频接收端缓冲区的占用率,以音频流为主媒体流,视频流为从媒体流,通过比较音频帧和视频帧的播放时间标签,对视频帧采取重复/丢弃方式来取得音视频媒体间的同步。实验表明,该种方法具有良好的同步性,既能满足P-QoS(可感知服务质量)要求又能满足实时性要求。     

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.     

NGN的媒体流自适应控制的实现

多媒体流自适应控制是流媒体应用的基本要求和主要难点之一。RTP提供端到端的数据传输服务,以便接收端实现数据包顺序播放和音视频同步,它需要由实时传输控制协议RTCP（Real-Time Transport Control Protocol）提供质量反馈,达到传输的实时性和传输质量要求。详细设计了RTCP网络传输质量反馈的实现方法,并给出了媒体流自适应控制的实现策略。     

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.     

一种联合速率控制和缓冲管理的VBR视频鲁棒传输方法

In this paper we present an adaptive video transmission framework that integrates rate allocation and buffer control at the source with the playback adjustment mechanism at the receiver. A transmission rate is determined by a rate allocation algorithm which uses the program clock reference (PCR) embedded in the video streams to regulate the transmission rate in a refined way. The server side also maintains multiple buffers for packets of different importance levels to trade off random loss for controlled loss according to the source buffer size, the visual impact, and the playback deadline. An over-boundary playback adjustment mechanism based on proportional-integra (PI) controller is adopted at the receiver to maximize the visual quality of the displayed video according to the overall loss and the receiver buffer occupancy. The performance of our proposed framework is evaluated in terms of peak signal-to-noise ratio (PSNR) in the simulations, and the simulation results demonstrate the improvement of the average PSNR values as well as the better quality of the decoded frames.     

IEEE 802.21-based seamless multicast streaming with dynamic playback control

With the explosion of video streaming available on the Internet, online multimedia applications become more popular in our days and the video quality of the Internet multimedia applications is directly affected by the network transmission state, which will be worse while switching the ongoing network connection from one wireless interface to another heterogeneous wireless technology, such as IEEE 802 families, UMTS (universal mobile telecommunications system) network and 3GPP LTE (3rd generation partnership project-long term evolution). In order to perform seamless handover between heterogeneous wireless networks, IEEE group proposed the “IEEE 802.21 standard” and defined a middleware function called “media independent handover function (MIHF)” to smooth the handover. In this paper, a dynamic playback control for multicasting streaming based on IEEE 802.21 is proposed to reduce the influence of handover between heterogeneous networks. In addition, we evaluate three different rate modes for seamless multicasting streams. The simulation results show that different playback frame rate modes, including the usual, incremental and linear modes, can achieve different video quality and can extend the playing time for handover video playback.     

Video Decompression Estimation and Playout Scheme Over the Internet

This paper examines problems associated with display of live continuous media. Under the assumption that the network cannot guarantee the required bounds on delay and jitter and the operating system scheduling is non-realtime, there is a need to accommodate the delay and jitter in the end systems in order to maintain a desirable Quality of Service (QoS). We propose a method of video playback which requires accurate estimation of display cycle time of video frames and the delay suffered by frames in the packet network. We apply various deterministic forecasting methods used in time series analysis on experimental data collected from video transmission. Suitable methods are recommended for display cycle time and delay estimation.     

请求下降邻域叠加选取分布式P2P 视频点播调度

为实现对等架构的低成本视频流传输和实时播放要求,提出基于请求下降叠加选 取的分布式P2P 视频点播调度算法。首先,基于叠加技术构建P2P 视频点播的技术指标,充分 考虑输入邻域节点、输出邻域节点和媒体服务器负载3 组优化指标,构建叠加架构和分布式算 法流程;其次,利用请求下降策略对发送节点和服务节点选取进行改进,解决可能出现的带宽 低利用率和无效的视频播放问题;最后,通过BitTorrent 视频点播系统对所提算法的有效性进 行了验证。     

Priority-based Media Delivery using SVC with RTP and HTTP streaming

Media delivery, especially video delivery over mobile channels may be affected by transmission bitrate variations or temporary link interruptions caused by changes in the channel conditions or the wireless interface. In this paper, we present the use of Priority-based Media Delivery (PMD) for Scalable Video Coding (SVC) to overcome link interruptions and channel bitrate reductions in mobile networks by performing a transmission scheduling algorithm that prioritizes media data according to its importance. The proposed approach comprises a priority-based media pre-buffer to overcome periods under reduced connectivity. The PMD algorithm aims to use the same transmission bitrate and overall buffer size as the traditional streaming approach, yet is more likely to overcome interruptions and reduced bitrate periods. PMD achieves longer continuous playback than the traditional approach, avoiding disruptions in the video playout and therefore improving the video playback quality. We analyze the use of SVC with PMD in the traditional RTP streaming and in the adaptive HTTP streaming context. We show benefits of using SVC in terms of received quality during interruption and re-buffering time, i.e. the time required to fill a desired pre-buffer at the receiver. We present a quality optimization approach for PMD and show results for different interruption/bitrate-reduction scenarios.     

Strategies for adaptive smoothing and rebuffering via dynamic network calculus

This paper proposes an adaptive video smoothing and rebuffering algorithm for video transmission in best-effort, low-bandwidth networks. Our algorithm, which is derived from a “dynamic” extension of the network calculus, (1) detects whether playback disruptions are inevitable and, if so, schedules a rebuffering action with minimal rebuffering delay and (2) computes a conservative transmission schedule that ensures continuous playback while minimizing consumption of network available bandwidth. NS-2 simulation results show that, compared to an aggressive transmission strategy, our method can greatly reduce bandwidth consumption and, compared to a simple rebuffering strategy with the same bandwidth consumption, our method can effectively reduce total playback delay while also reducing the number of rebuffering events.     

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。     

基于磁盘系统的多用户视频准入算法

分析了基于磁盘系统的视频服务器的准入策略,提出了基于磁盘系统的为保证视频服务器接纳更大数目用户的准入算法,即在每个服务周期中每个用户要求检索到的数据块的数目与其对应的回放速率成比例,视频服务器通过过量预约资源来允许更大数量的客户数。这种算法允许视频服务器接纳更大数目的用户,满足多用户媒体回放的稳态性和迁移态连续性的要求。     

Low cost pre-stored video transmission across networks

This paper proposes a simple, practical and low-cost dynamic segmentation and bandwidth allocation scheme, called middle point, to transmit pre-stored video from source to playback destination across networks. The proposed scheme is based on two concepts, playback tunnel and bandwidth tunnel, to determine the segmentation boundary and transmission bandwidth with the objective of reducing transmission cost while guaranteeing the playback buffer neither underflows nor overflows. The performance of the proposed scheme is evaluated by a set of real-life MPEG video traces and also compared with the well-known optimal smoothing scheme. The obtained results show that the proposed scheme can be easily implemented with low complexity and low transmission cost.