Smooth Control of Adaptive Media Playout for Video Streaming

Client-side data buffering is a common technique to deal with media playout interruptions of streaming video caused by network jitters and packet losses of best-effort networks. However, stronger playout interruption protection inevitably amounts to larger data buffering and results in more memory requirements and longer playout delay. Adaptive media playout (AMP), also a client-side technique, can reduce the buffer requirement and avoid buffer outage but at the expense of visual quality degradation because of the fluctuation of playout speed. In this paper, we propose a novel AMP scheme to keep the video playout as smooth as possible while adapting to the channel condition. The triggering of the playout control is based on buffer variation rather than buffer fullness. Experimental results show that our AMP scheme surpasses conventional schemes in unfriendly network conditions. Unlike previous schemes that are tuned for a specific range of packet loss and network instability, the proposed AMP scheme maintains consistent performance across a wide range of network conditions.     

Arrival process-controlled adaptive media playout with multiple thresholds for video streaming

To improve the playout quality of video streaming services, an arrival process-controlled adaptive media playout (AMP) mechanism is designed in this study. The proposed AMP scheme sets three threshold values, denoted by P _n , L and H, for the playout controller to start playback and dynamically adjust the playout rate based on the buffer fullness. In the preroll period, the playout can start only when the buffer fullness n is not less than the dynamic playback threshold P _n ,?which is determined by the jitters of incoming video frames. In the playback period, if the buffer fullness is below L or over H,?the playout rate will slow down or speed up in a quadratic manner. Otherwise, the playback speed depends on the instantaneous frame arrival rate, which is estimated by the proposed arrival process tracking algorithm. We employ computer simulations to demonstrate the performance of the proposed AMP scheme, and compare it with several conventional AMP mechanisms. Numerical results show that our AMP design can shorten the playout delay and reduce both buffer underflow and overflow probabilities. In addition, our proposed AMP also outperforms traditional AMP schemes in terms of the variance of distortion of playout and the playout curve. Hence, the proposed arrival process-controlled AMP is really an outstanding design.     

基于概率估计的自适应媒体播放算法

针对缓冲区下溢造成的视频播放抖动和中断问题,提出一种基于概率估计的自适应媒体播放算法。根据网络信道状态和估计的下溢概率和上溢概率控制视频帧的持续播放时间,适当控制播放速率的变化范围和变化量,减少缓冲区下溢概率和播放时延,实现视频平滑播放。仿真结果证明,该算法性能优于传统的自适应媒体播放算法。     

Joint optimization of continuity and quality for streaming video

This paper aims to reduce the amount of prebuffering required to ensure a maximum video continuity in streaming. Current approaches do this by slowing the playout frame rate of the decoder, this is known as adaptive media playout (AMP). However, doing this introduces playout distortion to the viewers as the video is played slower than its natural playout rate. We approach this by proposing a frame rate control scheme that jointly adjusts the encoder frame generation rate of the encoder and the playout frame rate of the decoder. In addition to using AMP to improve video continuity, we also allow the encoder to increase the encoder frame generation rate. This means the encoder will be sending more frames to the decoder to quickly increase the number of frames available at the playback buffer, thus lowering the chance of buffer underflow which causes discontinuity in video playback. At the same time, the increase in the number of frames at the playback buffer may mean that the decoder does not need to use AMP to delay the playback, thus lowering the playback distortion. However, the increase in encoder frame generation rate comes at a price because frame quality will need to decrease in order to meet the constraint on available network bandwidth. This implies that the scheme needs to find the optimal trade-off between frame quality, playout distortion and video continuity. To do that, we characterize the frame rate control problem using Lyapunov optimization. We then systematically derive the optimization policies. We also show that these policies can be decoupled into separate encoder and decoder optimization policies, thus allowing for a distributed implementation. Simulation results show significant reductions in the prebuffering requirements over a scheme that perform no frame rate control and lower playout distortions compared to the AMP schemes, while exhibiting a modest drop in frame quality.     

基于PID控制的自适应媒体播放算法

针对视频播放中因缓冲区下溢带来的抖动问题,提出一种基于PID控制的自适应播放算法。算法结合PID控制和自适应播放算法,根据网络信道状态和缓冲区状态控制视频播放速率,并对播放速率的范围和相邻帧播放速率的突变进行控制,在减少缓冲区下溢概率的同时实现视频的平滑播放。该算法在减小缓冲区下溢概率、实现视频的平滑播放以及减少播放延迟方面均优于固定因子自适应播放算法。     

基于多步下溢概率估计的自适应媒体播放算法

自适应播放(AMP)是一种通过动态调整播放速率减少这种中断的技术。多数AMP算法都基于缓存的满溢度或者其变化调整播放速率,其难点在于如何选择合适的调整门限值。本文算法,利用滑动窗口估计到达帧率的经验分布,再利用统计计算方法估计多步缓存下溢概率,然后根据一步和多步下溢概率调整播放速率。仿真实验证明在一般的信道假设和多种片源的测试中,本算法较其他同类算法有更好的表现。     

Playout in slotted CBR networks: Single and multiple nodes

We consider playout of a constant bit-rate (CBR) traffic after one or several multiplexors in a network with a playout buffer. Probabilistic characteristics of the playout process are found, depending on the traffic characteristics and parameters of the buffer. We present conditions on the buffer parameters that guarantee no jitter (complete playout).     

Content-Aware Adaptive Media Playout Controls for Wireless Video Streaming

Video streaming is one of the killer applications for cellular communications. The MPEG-4 fine-granularity scalability video coding technique can adapt to bandwidth variation and random packet errors. In this paper, to explore the impacts of cellular channel characteristics on the tolerance of buffer performance and quality of service, a novel statistical model-based adaptive media playout (AMP) is proposed by utilizing the statistical assumptions of both arrival and departure processes for a better decision on the dynamic threshold adjustment and frame-rate adjustment. Based on third-generation cellular transmission environment, simulation results will demonstrate that as compared to other AMP schemes, the proposed AMP control provides better visual quality with lower complexity.     

基于H.264可伸缩视频流的自适应媒体播放算法

为了应对H.264可伸缩视频编码（SVC）应用中网络特性的波动,提出了一种预测播放中断与缓冲区溢出风险进行及早调节的自适应媒体播放(AMP)算法。该算法估算网络流量与视频图像组（GOP）结构中各帧长度用于风险预测,通过K步调节过程实现良好的调节平滑性与速度,并利用SVC的可伸缩性尽量减少溢出带来的质量损失。仿真结果表明,该算法在抑制播放中断、处理缓冲区溢出与抖动性能等方面,优于现行的平滑AMP与常规AMP算法。     

一种MPEG1/MPEG2视频流的自适应播放算法

提出了一种MPEG1/MPEG2视频流的自适应播放算法。算法根据播放缓冲区的占用情况来调整视频帧的播放持续时间，并在播放缓冲区上溢时判断到达视频帧的类型，以决定是暂时存储还是丢弃，使得不会造成帧内编码帧和前向预测编码帧的丢失，从而保证视频流平滑地播放。实验结果表明，在播放不连续性和播放失真上新算法都优于Yuang算法，并实现了视频流的平滑播放。     

Dynamic Video Playout Smoothing Method for Multimedia Applications

Multimedia applications including video data require the smoothing of video playout to prevent potential discontinuity. In this paper, we propose a dynamic video playout smoothing method, called the Video Smoother, which dynamically adopts various playout rates in an attempt to compensate for high delay variance of networks. Specifically, if the number of frames in the buffer exceeds a given threshold (TH), the Smoother employs a maximum playout rate. Otherwise, the Smoother uses proportionally reduced rates in an effort to eliminate playout pauses resulting from the emptiness of the playout buffer. To determine THs under various loads, we present an analytic model assuming the Interrupted Poisson Process (IPP) arrival. Based on the analytic results, we establish a paradigm of determining THs and playout rates for achieving different playout qualities under various loads of networks. Finally, to demonstrate the viability of the Video Smoother, we have implemented a prototyping system including a multimedia teleconferencing application and the Video Smoother performing as part of the transport layer. The prototyping results show that the Video Smoother achieves smooth playout incurring only unnoticeable delays.     

Enhanced adaptive RTCP-based Inter-Destination Multimedia Synchronization approach for distributed applications

Newer social multimedia applications, such as Social TV or networked multi-player games, enable independent groups (or clusters) of users to interact among themselves and share services within the context of simultaneous media content consumption. In such scenarios, concurrently synchronized playout points must be ensured so as not to degrade the user experience on such interaction. We refer to this process as Inter-Destination Multimedia Synchronization (IDMS). This paper presents the design, implementation and evaluation of an evolved version of an RTCP-based IDMS approach, including an Adaptive Media Playout (AMP) scheme that aims to dynamically and smoothly adjust the playout timing of each one of the geographically distributed consumers in a specific cluster if an allowable asynchrony threshold between their playout states is exceeded. For that purpose, we previously had also to develop a full implementation of RTP/RTCP protocols for NS-2, in which we included the IDMS approach as an optional functionality. Simulation results prove the feasibility of such IDMS and AMP proposals, by adopting several dynamic master reference selection policies, to maintain an overall synchronization status (within allowable limits) in each cluster of participants, while minimizing the occurrence of long-term playout discontinuities (such as skips/pauses) which are subjectively more annoying and less tolerable to users than small variations in the media playout rate.     

基于延迟预测的自适应媒体播放算法

针对AMP-Live模型中存在的问题,提出一种基于报文延迟预测的自适应媒体播放算法（NEWAMP）,采用未来信道和缓冲状态的预测值作为视频报文播放速率调整的依据,将速率变化的程度进一步细化,同时考虑应用要求的最大端到端延迟,提高算法性能,与传统播放算法相比,NEWAMP在保证报文因下溢和上溢而丢弃的概率足够小的前提下,缓冲延迟减小了约50%,而与普通AMP-Live方法相比,NEWAMP不仅减小了报文因下溢和上溢而丢弃的概率,还将缓冲延迟减小了约40%。实验结果证明了该算法的有效性。     

Voice quality prediction models and their application in VoIP networks

The primary aim of this paper is to present new models for objective, nonintrusive, prediction of voice quality for IP networks and to illustrate their application to voice quality monitoring and playout buffer control in VoIP networks. The contributions of the paper are threefold. First, we present a new methodology for developing perceptually accurate models for nonintrusive prediction of voice quality which avoids time-consuming subjective tests. The methodology is generic and as such it has wide applicability in multimedia applications. Second, based on the new methodology, we present efficient regression models for predicting conversational voice quality nonintrusively for four modern codecs (G.729, G.723.1, AMR and iLBC). Third, we illustrate the usefulness of the models in two main applications - voice quality prediction for real Internet VoIP traces and perceived quality-driven playout buffer optimization. For voice quality prediction, the results show that the models have accuracy close to the combined ITU PESQ/E-model method using real Internet traces (correlation coefficient over 0.98). For playout buffer optimization, the proposed buffer algorithm provides an optimum voice quality when compared to five other buffer algorithms for all the traces considered.     

Adaptive VoIP playout scheduling: assessing user satisfaction

Delay and packet loss dramatically affect the quality of voice-over-IP (VoIP) calls and depend on the playout buffer scheme implemented at the receiver. The choice of playout algorithm can't be based on statistical metrics without considering the perceived end-to-end conversational speech quality. The authors present a method for evaluating various playout algorithms that extends the E-model concept by estimating user satisfaction from time-varying transmission impairments. This article evaluates several playout algorithms and shows a correspondence between the authors' results and those obtained via statistical loss and delay metrics.     

Novel approaches for online playout delay prediction in VoIP applications using time series models

Voice over IP (VoIP) applications requires a buffer at the receiver to minimize the packet loss due to late arrival. Several algorithms are available in the literature to estimate the playout buffer delay. Classic estimation algorithms are non-adaptive, i.e. they differ from more recent approaches basically due to the absence of learning mechanisms. This paper introduces two new formulations of adaptive algorithms for online learning and prediction of the playout buffer delay, the first one being based on the standard Box-Jenkins autoregressive model, while the second one being based on the feedforward and recurrent neural networks. The obtained results indicate that the proposed algorithms present better overall performance than the classic ones.     

多媒体播放中的动态双门限自适应调整算法

针对无线网络存在的自相似特性会影响视频流的播放质量问题,提出了基于滑动窗口的接收端播放缓存调整算法,根据网络流量的变化,动态地调整双门限,并利用播放缓存的占用率来控制视频流的播放速度,平滑时延抖动.仿真实验证明,无论网络流量处于平稳状态还是处于突发状态,本文设计的算法都能够较好地保证视频流的连续播放,提高视频流的播放质量,为用户提供良好的视觉效果.     

一个负载自适应的实时音频播放算法

研究了Windows操作系统中网络电话软件的实时播放音频的策略。在非实时系统中，音频播放程序不能严格地被定时执行，播放缓冲区被耗尽而产生播放空隙。采用DirectSound技术，以ms为单位来控制音频的播放，并根据负载的变化动态地调整每一个话音期的门限值来减少播放空隙。实验结果表明，该算法能够以较小的时延为代价来获取平滑的播放效果。     

Resource-aware and quality-fair video-streaming using multiple adaptive TCP connections

In this paper, we present a resource-aware and quality-fair video content sharing system. When a video sharing server has insufficient uplink bandwidth and needs to serve multiple video content sharing services via streaming or downloading to other client peers using TCP transport, each service shares the limited uplink bandwidth equitably, due to the fair sharing characteristics inherent in TCP. However this bandwidth fair sharing cannot always guarantee quality fairness among the services, due to the specific requirements for video-streaming services, such as the playout rate and the size of the playout buffer. In our system, the server uses multiple TCP connections adaptively, depending on the anticipated status of each client playout buffer, to guarantee the bandwidth of each video-streaming session. By guaranteeing the quality of each video-streaming session, without the quality loss of other service sessions, the proposed system can successfully achieve service quality fairness. Simulation results show that our proposed algorithm can dramatically enhance the quality of each streaming session and thus provide service quality fairness among simultaneous multiple heterogeneous video-streaming services and content download services.     

Playout buffering of speech packets based on a quality maximization approach

To combat jitter problems in voice streaming over packet networks, playout buffering algorithms are used at the receiver side. Most of the proposed solutions rely on two main operations: prediction of delay statistics for future packets; setting of the end-to-end delay so as to limit or avoid packet losses. In recent years, a new approach has been presented, which is based on using a quality model to evaluate the impact of both packet loss and delay on the voice quality. Such a model is used to find the buffer setting that maximizes the expected quality. In this paper, we present a playout buffering algorithm whose main contribution is the extension of the new quality-based approach to the case of voice communications affected by bursty packet losses. This work is motivated by two main considerations: most of IP telephony applications are characterized by bursty losses instead of random ones; the human perception of the speech quality is significantly affected by the temporal correlation of losses. To this purpose, we make use of the extensions proposed in the ETSI Tiphon for the ITU-T E-Model so as to incorporate the effects of loss burstiness on the perceived quality. The resulting playout algorithm estimates the characteristics of the loss process varying the end-to-end delay, weights the loss and the delay effects on the perceived quality, and maximizes the overall quality to find the optimal setting for the playout buffer. The experimental results prove the effectiveness of the proposed technique.