Joint power-playout control for media streaming over wireless links

Media streaming applications over wireless links face various challenges, due to both the nature of the wireless channel and the stringent delivery requirements of media traffic. In this paper, we seek to improve the performance of media streaming over an interference-limited wireless link, by using appropriate transmission and playout control. In particular, we choose both the power at the transmitter and the playout scheduling at the receiver, so as to minimize the power consumption and maximize the media playout quality. We formulate the problem using a dynamic programming approach, and study the structural properties of the optimal solution. We further develop a justified, low-complexity heuristic that achieves significant performance gain over benchmark systems. In particular, our joint power-playout heuristic outperforms: 1) the optimal power control policy in the regime where power is most important and 2) the optimal playout control policy in the regime where media (playout) quality is most important; furthermore, this heuristic has only a slight performance loss as compared to the optimal joint power-playout control policy over the entire range of the investigation.     

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.     

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.     

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.     

无线视频传输服务质量研究

在下一代无线通信网络中,以视频传输为主的流媒体业务将成为主要业务之一。着眼于无线视频流服务质量的研究,首先简要介绍了以网络为中心的服务质量控制策略,然后重点分析以终端为中心的速率控制机制,并介绍了目前常用的分组调度算法。     

基于流媒体相关的视频安全组播协议(MSMP)研究

随着无线网络的快速发展和Internet中流媒体视频的巨大成功,无线网络中的视频服务有望在不久的将来得到大规模部署,无线网络上的实时流媒体传输技术已成为研究热点,而其中视频安全组播协议是一个关键问题。但是,由于无线网络中有限的带宽和有限的存储空间,外部的攻击与自适应机制带来的安全性问题在无线流媒体视频中是不可避免的。一个精心设计的密钥管理算法不仅可以明显地提高流媒体视频的性能,还能够保证可靠的数据嵌入以及实时视频应用提供安全支持。如何设计一个高效的密钥管理算法,是当前流媒体视频应用中一个备受关注的问题。针对无线视频应用,针对一个处于开放和不安全的网络环境中的自适应视频应用的密钥管理算法进行了研究,并进行了算法评价。     

An efficient playout smoothing mechanism for layered streaming in P2P networks

Layered video streaming in peer-to-peer (P2P) networks has drawn great interest, since it can not only accommodate large numbers of users, but also handle peer heterogeneity. However, there’s still a lack of comprehensive studies on chunk scheduling for the smooth playout of layered streams in P2P networks. In these situations, a playout smoothing mechanism can be used to ensure the uniform delivery of the layered stream. This can be achieved by reducing the quality changes that the stream undergoes when adapting to changing network conditions. This paper complements previous efforts in throughput maximization and delay minimization for P2P streaming by considering the consequences of playout smoothing on the scheduling mechanisms for stream layer acquisition. The two main problems to be considered when designing a playout smoothing mechanism for P2P streaming are the fluctuation in available bandwidth between peers and the unreliability of user-contributed resources—particularly peer churn. Since the consideration of these two factors in the selection and scheduling of stream layers is crucial to maintain smooth stream playout, the main objective of our smoothing mechanism becomes the determination of how many layers to request from which peers, and in which order. In this work, we propose a playout smoothing mechanism for layered P2P streaming. The proposed mechanism relies on a novel scheduling algorithm that enables each peer to select appropriate stream layers, along with appropriate peers to provide them. In addition to playout smoothing, the presented mechanism also makes efficient use of network resources and provides high system throughput. An evaluation of the performance of the mechanism demonstrates that the proposed mechanism provides a significant improvement in the received video quality in terms of lowering the number of layer changes and useless chunks while improving bandwidth utilization.     

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。     

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.     

A Multilayered Audiovisual Streaming System Using the Network Bandwidth Adaptation and the Two-Phase Synchronization

Synchronous audiovisual streaming and playout are two of the major issues in the multimedia communication network. However, the past corresponding researches of media synchronization mainly focused on the mono-quality and single-layer (nonscalable) audiovisual data. To overcome challenges of ubiquitous multimedia streaming, a scalable audiovisual coder that can provide flexible scalabilities and adaptive streaming control to adapt to complicated network situations are both required. This paper proposes a multilayered audiovisual streaming scheme to deliver layered audiovisual data synchronously, which is called ML-AVSS. Fine-granular scalability (FGS) and bit-sliced arithmetic coding (BSAC) techniques are used to segment video and audio data into one base-layer and multiple enhancement-layer bitstreams. With advantages of audiovisual layer coding, a de-jitter procedure, a conditional retransmission mechanism and a playout synchronization mechanism are designed to transmit hybrid multilayered audiovisual bitstreams in consideration of the result of a network bandwidth adaptation and the distinct decoding time-complexity. Experimental results show that the proposed ML-AVSS is a feasible streaming scheme to overcome challenges of ubiquitous multimedia streaming, e.g., constrained channel bandwidth, quality degradation, unsmooth playout, etc.     

Stochastic traffic engineering for real-time applications over wireless networks

In this work, we focus on the Stochastic Traffic Engineering (STE) problem arising from the support of QoS-demanding real-time media-streaming applications over fading and congestion affected TCP-friendly/IP multiantenna wireless pipes. First, after recasting the tackled STE problem in the form of a suitable cross-layer nonlinear stochastic optimization problem, we develop a traffic analysis of the overall underlying multiple-input multiple-output (MIMO) wireless pipe that points out the relative effects of both fading-induced errors and congestion-induced packet losses on the goodput offered by the resulting end-to-end connection. Second, we develop an optimal cross-layer resource management policy that allows a joint scheduling of the media encoding rate (i.e., playin rate), transmit energy and delivery rate (i.e., playout rate) of each end-to-end connection active over the considered access network. Salient features of the presented joint scheduling policy are that: (i) it is self-adaptive; (ii) it is able to provide hard (i.e., deterministic) QoS guarantees, in terms of hard limited playout delay and playout rate-jitter; and (iii) it explicitly accounts for the performance interaction of the protocols implemented at all layers of the considered stack.     

Quality adaptive end-to-end packet scheduling to avoid playout interruptions in Internet video streaming systems

In Internet multimedia streaming, the quality of the delivered media can be adapted to the Quality of Service provided by the underlying network, thanks to encoding algorithms. These allow a fine grained enhancement of a low quality base layer at streaming time. The main objective that should be satisfied in such systems is to avoid the starvation of the decoding process and consequent playout interruptions. In this work, we tackle the problem using a control theoretic approach. In particular, we design and implement the novel end-to-end Quality Adaptive Scheduler for properly distributing the network available bandwidth among base and enhancement layers. The developed solution can be adopted in many contexts given that it has been designed without assumptions on the delivered media nor on the protocol stack. Anyway, to test its effectiveness, we have casted it in a H.264/AVC SVC based video streaming architecture for unicast Internet applications. The performance of the scheduler has been experimentally evaluated in both a controlled testbed and several “wild” Internet scenarios, including also UMTS and satellite radio links. Results have clearly demonstrated that our Quality Adaptive Scheduler is able to significantly improve the performance of the video streaming system in all operative conditions.     

An efficient scheduling algorithm for scalable video streaming over P2P networks

During recent years, the Internet has witnessed rapid advancement in peer-to-peer (P2P) media streaming. In these applications, an important issue has been the block scheduling problem, which deals with how each node requests the media data blocks from its neighbors. In most streaming systems, peers are likely to have heterogeneous upload/download bandwidths, leading to the fact that different peers probably perceive different streaming quality. Layered (or scalable) streaming in P2P networks has recently been proposed to address the heterogeneity of the network environment. In this paper, we propose a novel block scheduling scheme that is aimed to address the P2P layered video streaming. We define a soft priority function for each block to be requested by a node in accordance with the block’s significance for video playback. The priority function is unique in that it strikes good balance between different factors, which makes the priority of a block well represent the relative importance of the block over a wide variation of block size between different layers. The block scheduling problem is then transformed to an optimization problem that maximizes the priority sum of the delivered video blocks. We develop both centralized and distributed scheduling algorithms for the problem. Simulation of two popular scalability types has been conducted to evaluate the performance of the algorithms. The simulation results show that the proposed algorithm is effective in terms of bandwidth utilization and video quality.     

Joint Source Coding and Network-Supported Distributed Error Control for Video Streaming in Wireless Multihop Networks

Real-time video communication over wireless multihop networks has gained significant interest in the last few years. In this paper, we focus our attentions on the problem of source coding and link adaptation for packetized video streaming in wireless multihop networks when network nodes are media-aware. We consider a system where source coding is employed at the video encoder by selecting the encoding mode of each individual macro-block, while error control is exercised through application-layer retransmissions at each media-aware network node. For this system model, the contribution of each communication link on the end-to-end video distortion is considered separately in order to achieve globally optimal source coding and ARQ error control. To reach the globally optimal solution, we formulate the problem of joint source and distributed error control (JSDEC) and devise a low-complexity solution algorithm based on dynamic programming. Extensive experiments have been carried out on the basis of H.264/AVC codec to demonstrate the effectiveness of the proposed algorithm over the existing joint source and channel coding (JSCC) algorithm in terms of PSNR perceived at the decoder under time-varying multihop wireless links.     

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.     

Adaptive interface selection over cloud-based split-layer video streaming via multi-wireless networks

As mobile devices such as tablet PCs and smartphones proliferate, the online video consumption over a wireless network has been accelerated. From this phenomenon, there are several challenges to provide the video streaming service more efficiently and stably in the heterogeneous mobile environment. In order to guarantee the QoS of real-time HD video services, the steady and reliable wireless mesh is necessary. Furthermore, the video service providers have to maintain the QoS by provisioning streaming servers to respond the clients’ request of different video resolution. In this paper, we propose a reliable cloud-based video delivery scheme with the split-layer SVC encoding and real-time adaptive multi-interface selection over LTE and WiFi links. A split-layer video streaming can effectively scale to manage the required channels on each layer of various client connections. Moreover, split-layer SVC model brings streaming service providers a remarkable opportunity to stream video over multiple interfaces (e.g. WiFi, LTE, etc.) with a separate controlling based on their network status. Through the adaptive interface selection, the proposed system aims to ensure the maximizing video quality which the bandwidth of LTE/WiFi accommodates. In addition, the system offers cost-effective streaming to mobile clients by saving the LTE data consumption. In our system, an adaptive interface selection is developed with two different algorithms, such as INSTANT and EWMA methods. We implemented a prototype of mobile client based on iOS particularly by using iPhone5S. Moreover, we also employ the split-layer SVC encodes in streaming server-side as the add-on module to SVC reference encoding tool in a virtualized environment of KVM hypervisor. We evaluated the proposed system in an emulated and a real-world heterogeneous wireless network environments. The results show that the proposed system not only achieves to guarantee the highest quality of video frames via WiFi and LTE simultaneous connection, but also efficiently saves LTE bandwidth consumption for cost-effectiveness to client-side. Our proposed method provides the highest video quality without deadline misses, while it consumes 50.6% LTE bandwidth of ‘LTE-only’ method and 72.8% of the conventional (non-split) SVC streaming over a real-world mobile environment.     

A systematic rate controller for MPEG-4 FGS video streaming

This paper proposes a systematic rate controller (SRC) for content-aware streaming of MPEG-4 FGS video over the Internet. An active layer dropping technique is proposed to provide both coarse-grain and fine-granularity scalability of smooth quality adaptation to bandwidth fluctuations and bit-rate variations of streamed video over a general time-scale. The smooth quality adaptation is realized through the mode and state transition of a state machine that implements the SRC. The SRC effectively uses available bandwidth and client buffer by forward-shifting the FGS video stream. It provides protection to video segments with important content by introducing a content-aware priority-based layer model for the MPEG-4 FGS video stream. RID="*" ID="*" The work reported in this paper was performed when this author was working at Microsoft Research Asia as a research intern.     

Practical design of a proxy agent to facilitate adaptive video streaming service across wired/wireless networks

Thanks to the growing of the wireless networks, the video streaming application becomes a ubiquitous joyful service. In a wireless communication network environment, the service traffic spans across the wired and wireless domains. In this article, we propose a practical design of a proxy agent - SPONGE (Stream Pooler Over a Network Graded Environment) sitting between the wireless User Equipments (UEs) and the video streaming server to facilitate the adaptive video streaming service across wired/wireless networks. To make the wireless streaming service more efficient, an input video session would be encoded as multiple qualities of video streams so that UEs with a similar receiving condition can share streams with the same service quality via SPONGE. SPONGE can alleviate the direct load on the original stream broadcasting server. Meanwhile, it can make each UE get an adaptive streaming service according to the network conditions of the UE by a reduced network condition feedback latency. Our theoretical analysis and simulation results show that SPONGE can help wireless streaming users get a smooth and better playback quality by a quick and accurate reaction to the network condition.