Video Streaming on Content Delivery Network in Star-Ring Based EPON

VoD （video on demand） service is regard as one of the most important services in next decade. It requires high speed and huge bandwidth to guarantee the QoS （quality of service）. EPON （Ethernet passive optical network） is regarded as one of the best solutions on access network, due to high speed and low cost. The star-ring EPON architecture is an evolution of EPON which provides better local transmission and fault tolerance capability. In addition, the CDN （content delivery network） mechanism, in which the video content is cached at a location closer to the user, is a widely used methodology to reduce the latency of VoD service. Therefore, in this paper, we propose a mechanism which combines the advantage of star-ring based EPON architecture and CDN mechanism to improve QoS. We design a new Sub-OLT （optical line terminal） which includes storage to store video files and serve at local. Thus, it can reduce the bandwidth between OLT and ONU （optical network unit）. Simulation results have shown that our proposed mechanism can improve the system performance and QoS in terms of packet delay and jitter.     

Relay-assisted hierarchical adaptation scheme for multi-user scalable video delivery to heterogeneous mobile devices

In multi-user video (MUV) delivery scenarios,the available resources of receiver devices,such as processing capability,link packet error rate (PER),and bandwidth,are usually different.We propose a relay-assisted hierarchical adaptation (RHA) scheme to maximize the total perceptual quality of all users when transmitting video streams coded via scalable video coding (SVC).First,MUV bitstreams are adaptively extracted under the constraints of network bandwidth and individual decoding capacity.Next,the relay links are introduced as substitutes of possible bad direct links for packets retransmissions.Approximately equal opportunity of transmission is allocated to each stream while the packets inside a stream are scheduled according to their priorities.The priorities are determined by the links states and packets loss distortions.Simulation results show that our RHA scheme has significant performance improvements compared with other schemes.     

Seamless handover between unicast and multicast multimedia streams

With the deployment of heterogeneous networks, mobile users are expecting ubiquitous connectivity when using applications. For bandwidth-intensive applications such as Interact Protocol Television （IPTV）, multimedia contents are typical- ly transmitted using a multicast delivery method due to its bandwidth efficiency. However, not all networks support multicasting. Multicasting alone could lead to service disruption when the users move from a multicast-capable network to a non-multicast network. In this paper, we propose a handover scheme called application layer seamless switching （ALSS） to provide smooth real-time multimedia delivery across unicast and multicast networks. ALSS adopts a soft handover to achieve seamless playback during the handover period. A real-time streaming testbed is implemented to investigate the overall handover performance, espe- cially the overlapping period where both network interfaces are receiving audio and video packets. Both the quality of service （QoS） and objective-mapped quality of experience （QoE） metrics are measured. Experimental results show that the overlapping period takes a minimum of 56 and 4 ms for multicast-to-unicast （M2U） and unicast-to-multicast （U2M） handover, respectively. The measured peak signal-to-noise ratio （PSNR） confirms that the frame-by-frame quality of the streamed video during the handover is at least 33 dB, which is categorized as good based on ITU-T recommendations. The estimated mean opinion score （MOS） in terms of video playback smoothness is also at a satisfactory level.     

Comparisons of the H.264 and H.263 Video Compression Standards

The International Telecommunications Union （ITU-T） and the International Standardization Organization/International Electrotechnical Commission （ISO/IEC） are the only two formal organizations that developed video coding standards. The ITU-T video coding standards called recommendations and are usually optimized for real-time video communication such as videoconference and video telephony while the ISO/IEC standards are mainly designed for storage （DVD） and broadcast （satellite and digital TV）. ITU-T and the ISO/IEC JTC1 have agreed to join their efforts in the development of H.264 standard, which was initiated by ITU-T committee. The ITU-T H.264 video coding standard has been developed to achieve significant improvements over the existing standards in compression performance, although the basic coding framework of the standard is similar to that of the existing standards. H.264 standard is compared with H.263 and test results showed the coding gains obtained by the H.264 encoder is over the H.263 encoder for Common Intermediate Format （CIF） and Quarter Common Intermediate Format （QCIF）sequences, respectively. H.264 achieves an average of 4 dB PSNR（peak signal-to-noise rate） gain for the selected ten CIF sequences at 30 frames per second, and 4.57 dB Peak Signal-to-Noise Rate （PSNR） gain for the selected ten QCIF sequences at 30 frames per second.     

Channel Estimation and Effective Capacity over Imperfect Rayleigh Channel Environment in Cognitive Radio Networks

In cognitive radio networks, a SU （secondary user） can share the same frequency band with the PU （primary user） as long as the interference introduced to the latter is below a predefined threshold. In this paper, the transmission performance in cognitive radio networks is studied assuming imperfect channel estimation, taking QoS （quality of service） constraints into consideration. It is assumed that the cognitive transmitter can perform channel estimation and send the data at two different rates and power levels depending on the activity of the PU. The existence of the PU can be detected by channel sensing. A two-state Markov chain process is used to model the existence of the PUs. The cognitive transmission is also configured as a state transition model depending on whether the rates are higher or lower than the instantaneous rate values. The maximum capacity of the SU under the delay constraint is investigated. The concept of effective capacity of the channel is applied. An optimization problem for rate and power allocation under interference and power constraints is formulated and solved. Numerical results are presented to illustrate the average effective capacity optimization and the impact of other system parameters.     

An adaptive sub-carrier and power allocation algorithm with QoS guarantee for OFDMA system

This paper presents an adaptive sub-carrier and power allocation scheme for orthogonal frequency division multiple access （OFDMA） systems according to their different quality of service （QoS） requirements and traffic type. The algorithm maximized the transmission data rate while satisfying total power constraint and a certain bit error rate （BER） requirement. A greedy algorithm known to be the most efficient algorithm for this problem can provide a high quality optimal solution, but has the disadvantage of incurring a long computation time. This problem should be solved in a real-time environment. The proposed algorithm not only avoids the high complexity but also provides considerable universality and flexibility for both the fixed rate voice data and variable rate multimedia data of the broadband wireless communication. It mainly consists of two steps. The first is the allocation of sub-carriers and power alternately to the real-time user. The second is the residual resource distribution to the non-real-time users. The simulation results demonstrate that the scheme has computational advantages over the conventional algorithms while providing the QoS guarantee.     

Multi-user MIMO downlink eigen-mode transmission over jointly correlated MIMO channels

The study of multi-user multiple-input multiple-output (MU-MIMO) systems has emerged recently as an important research topic because such systems have the potential to combine the high capacity that can be achieved through MIMO processing with the benefits of spatial-division multiple-access.However,receiver and transmitter channel state information (CSI) is generally required,especially for the downlink channel.In this paper,MU-MIMO downlink transmission systems with statistical CSI at the transmitter are studied for jointly correlated MIMO channels.Eigen-mode space division multiple access transmission is derived based on the maximization of the ergodic sum rate,and two eigen-mode power allocation algorithms are proposed using matrix permanent and convex optimization theory.The new algorithms can overcome the limitations of the existing multi-user downlink transmission algorithm in practical applications.     

A Motion Compensated Lifting Wavelet Codec for 3D Video Coding

A motion compensated lifting (MCLIFT) framework for the 3D wavelet video coding is proposed in this paper. By using bi-directional motion compensation in each lifting step of the temporal direction, the video frames are effectively de-correlated. With the proper entropy coding and bit-stream packaging schemes, the MCLIFT wavelet video coder is scalable at frame rate and quality level. Experimental results show that the MCLIFT video coder outperforms the 3D wavelet video coder without motion by an average of 0.9-1.3dB, and outperforms MPEG-4 coder by an average of 0.2-0.6dB.     

Bargaining game theoretic power control in selfish cooperative relay networks

Wireless cooperative communications require appropriate power allocation (PA) between the source and relay nodes. In selfish cooperative communication networks, two partner user nodes could help relaying information for each other, but each user node has the incentive to consume his power solely to decrease its own symbol error rate (SER) at the receiver. In this paper, we propose a fair and efficient PA scheme for the decode-and-forward cooperation protocol in selfish cooperative relay networks. We formulate this PA problem as a two-user cooperative bargaining game, and use Nash bargaining solution (NBS) to achieve a win-win strategy for both partner users. Simulation results indicate that the NBS is fair in that the degree of cooperation of a user only depends on how much contribution its partner can make to decrease its SER at the receiver, and efficient in the sense that the SER performance of both users could be improved through the game.     

Impact of Network Dynamics on User's Video Quality: Analytical Framework and QoS Provision

We develop an analytical framework to investigate the impacts of network dynamics on the user perceived video quality. Our investigation stands from the end user's perspective by analyzing the receiver playout buffer. In specific, we model the playback buffer at the receiver by a G/G/1/? and G/G/1/N queue, respectively, with arbitrary patterns of packet arrival and playback. We then examine the transient queue length of the buffer using the diffusion approximation. We obtain the closed-form expressions of the video quality in terms of the start-up delay, fluency of video playback and packet loss, and represent them by the network statistics, i.e., the average network throughput and delay jitter. Based on the analytical framework, we propose adaptive playout buffer management schemes to optimally manage the threshold of video playback towards the maximal user utility, according to different quality-of-service requirements of end users. The proposed framework is validated by extensive simulations.     

基于无线视频传感器网络的失真优化路由算法

针对无线视频传感器网络链路不稳定、重建质量要求不高的特点,提出一种适应多描述编码（MDC）可靠传输的路由算法EDLOR。该算法充分考虑了视频编码速率、时延受限、网络丢包等因素,以多描述峰值信噪比（PSNR）作为优化目标,使视频总体失真最小化;然后根据计算结果,将多描述编码分配到指定的路径进行传输。实验结果表明,EDLOR路由算法能够提高平均PSNR,降低了网络丢包率,提升了总体视频质量。     

Video streaming over the internet with optimal bandwidth resource allocation

In this paper, an adaptive framework for video streaming over the Internet is presented. The framework is a joint design of packet scheduling and rate control with optimal bandwidth resource allocation. The transmission rate is dynamically adjusted to obtain maximal utilization of the client buffer and minimal allocation of the bandwidth. Under the constraint of the transmission rate, a prioritized packet scheduling is designed to provide a better visual quality of video frames. The packet scheduling is a refined bandwidth allocation which takes into account of varying importance of the different packets in a compressed video stream. Moreover, the proposed approach is scalable with increasing multimedia flows in the distributed Internet environment. Comparisons are made with the most current streaming approaches to evaluate the performance of the framework using the H.264 video codec. The extensive simulation results show that the average Peak Signal to Noise Ratio (PSNR) increases in our proposed approach. It provides a better quality of the decoded frames, and the quality of the decoded frames changes more smoothly. The achieved video quality among different users also has a lower fluctuation, which indicates a fair sharing of network resources.

基于状态机的视频码率自适应算法

由于链路带宽存在随机性,已有的基于超文本传输协议的动态自适应流媒体传输技术（DASH）的码率自适应算法不能很好解决播放流畅性和视频质量之间的矛盾。为解决该问题,提出一种基于状态机的DASH（SDASH）算法,将码率切换过程用状态机进行分析与控制。首先充分考虑客户端观看体验质量（QoE）的影响因素,对影响因素进行数值分析,并设定6个码率等级状态;然后将视频码率与影响因素的数值变化之间的联系作为状态转移条件;最后在保证播放缓存和码率偏移率处于一定阈值的条件下将视频码率切换至视频质量和播放流畅性整体性能相对最佳的码率等级上。实验结果表明,该算法与基于模糊逻辑控制的码率自适应算法相比能够提高客户端请求视频的平均码率,且尽量避免出现码率骤降等情况,从而较好地平衡播放流畅性和视频质量之间的关系,提升了视频观看过程的体验质量。     

Variable Time Scale Multimedia Streaming Over IP Networks

This paper presents a comprehensive analysis of a variable time-scale streaming technique, VTSS, according to which rate changes are obtained by varying the inter-packet transmission interval, rather than altering, as in most cases, the source coding rate. Instead of constraining the transmitter to operate in real-time, the time scale of the packet scheduler can vary between zero, when the network is congested, to as faster than real-time as the channel bandwidth allows, when the network is lightly loaded. Although this approach is reportedly used in commercial streaming products, so far the technique has not yet been analyzed in a rigorous fashion, nor it has been compared to other state-of-the-art streaming techniques. This work first presents a theoretical analysis of the performance achievable by the VTSS approach, and it shows that, for the same channel conditions, VTSS yields a total distortion which is lower or, in the worst case, equal than the distortion of the standard real-time source-rate adaptive approach. A lower bound on receiver buffer size is also derived. Network simulations then analyze the performance of a TCP-friendly test implementation of VTSS compared with an ideal real-time source rate-adaptive technique, whose performance, being ideal, represents the upper bound of any transmission scheme based on source rate adaptation. The simulation results, also based on actual network traces, show that the VTSS approach delivers higher perceptual quality (up to 1.2 dB PSNR in the considered scenarios) and reduced video quality fluctuations (1.6 dB standard deviation PSNR, instead of 4.9 dB) for a wide range of standard video sequences. Perceptual quality evaluation by means of PVQM confirms such results. The gains, as expected, are even more pronounced (7.6 dB PSNR on average) if compared to real-time constant bit-rate video transmission.      

An enhanced algorithm for improving real-time video transmission for tele-training education

Tele-training in surgical education has not been effectively implemented. There is a stringent need for a high transmission rate, reliability, throughput, and reduced distortion for high-quality video transmission in the real-time network. This work aims to propose a system that improves video quality during real-time surgical tele-training. The proposed approach aims to minimise the video frame’s total distortion, ensuring better flow rate allocation and enhancing the video frames’ reliability. The proposed system consists of a proposed algorithm for Enhancing Video Quality, Distorting Minimization, Bandwidth efficiency, and Reliability Maximization called (EVQDMBRM) algorithm. The proposed algorithm reduces the video frame’s total distortion. In addition, it enhances the video quality in a real-time network by dynamically allocating the flow rate at the video source and maximizing the transmission reliability of the video frames. The result shows that the proposed EVQDMBRM algorithm improves the video quality with the minimized total distortion. Therefore, it improves the Peak Signal to Noise Ratio (PSNR) average by 51.13 dB against 47.28 dB in the existing systems. Furthermore, it reduces the video frames processing time average by 58.2 milliseconds (ms) against 76.1, and the end-to-end delay average by 114.57 ms against 133.58 ms comparing to the traditional methods. The proposed system concentrates on minimizing video distortion and improving the surgical video transmission quality by using an EVQDMBRM algorithm. It provides the mechanism to allocate the video rate at the source dynamically. Besides that, it minimizes the packet loss ratio and probing status, which estimates the available bandwidth.

     

一种自适应的视频流化前向纠错算法

网络视频应用经常会受到数据包丢失或错误以及网络带宽资源不足的干扰.相关研究表明:在多数情况下,动态变化的网络带宽和丢包率是影响视频流化质量的关键因素.因此,为了保证视频质量,可以采用前向纠错(forward error correction,简称FEC)编码来提高视频数据传输的可靠性;同时,为了适应网络状态的变化,发送端可以调节视频数据的发送速率,并在视频源数据与FEC数据之间合理分配网络传输带宽.首先通过对视频流结构的分析,在充分考虑帧之间的依赖关系和帧类型的基础上提出了一种帧的解码模型.在此基础上,建立了用于在视频源数据和FEC数据之间分配网络带宽资源的优化算法.实验表明,该模型可以有效地适应网络状态的变化,并通过优化分配网络带宽资源来使接收端获得最大的可播放帧率.     

ARTP: A buffer-aware rate control protocol for media streaming

We present adaptive real-time transport protocol (ARTP), a media streaming transport protocol that implements a congestion control mechanism. With this mechanism, the sender adapts its sending rate to network conditions and to the buffering capacity of the receiver. This adaptiveness takes into account the real-time constraints of media streaming. It aims at ensuring media playback continuity, and at achieving a low packet loss rate during media streaming sessions. ARTP ensures the continuity of media playback by buffering media packets during congestion-free periods and reduces the loss rate by reducing the transmission rate during congestion periods. This protocol considers the size of the buffer that the receiver dedicates to rate control in order to avoid overflow or underflow of the buffer. This approach allows limited memory devices such as cellular phones and PDAs to take advantage of rate control.ARTP is based on the feedback that the real-time control protocol (RTCP) reports give with the addition of two new parameters that we define in this paper: the steady state loss event rate and the duration to the next feedback report. It also requires that the real-time streaming protocol (RTSP) provide the server with the size of the buffer that the client dedicates to rate control.Our NS-2 simulations show that, besides buffer protection, ARTP significantly reduces the loss rate. Compared to Additive Increase Multiplicative Decrease (AIMD) rate control techniques, ARTP provides a better media quality by ensuring the continuity of media playback and, compared to equation-based rate control techniques, it achieves a better loss rate and reduces the bandwidth used for feedback.     

802.11e中失真驱动的视频传输跨层优化

针对H.264视频在802.11e无线网络中传输时,由丢包和编码量化引起的接收端失真问题,提出了一种以失真为驱动的跨层优化算法,以减少接收端的失真。通过率失真模型得到量化参数(QP)和量化失真之间的关系后,根据不同视频数据分区的丢包率,估计出接收端的传输失真和总体失真;然后,以这个总体失真为依据,提出一个求最优量化参数的选择算法。实验结果表明,所提方法相比只针对H.264视频不同数据分区赋予不同传输优先级的由上向下跨层架构,或者只考虑根据传输丢包率调整编码器量化参数的由下到上的跨层架构,平均峰值信噪比(PSNR)提高了1~2dB,具有更小的接收端失真。     

Error-Resilient Video Encoding and Transmission in Multirate Wireless LANs

In this paper, we present a cross-layer approach for video transmission in wireless LANs that employs joint source and application-layer channel coding, together with rate adaptation at the wireless physical layer (PHY). While the purpose of adopting PHY rate adaptation in modern wireless LANs like the IEEE 802.11a/b is to maximize the throughput, in this paper we exploit this feature to increase the robustness of wireless video. More specifically, we investigate the impact of adapting the PHY transmission rate, thus changing the throughput and packet loss channel characteristics, on the rate-distortion performance of a transmitted video sequence. To evaluate the video quality at the decoder, we develop a cross-layer modeling framework that considers jointly the effect of application-layer joint source-channel coding (JSCC), error concealment, and the PHY transmission rate. The resulting models are used by an optimization algorithm that calculates the optimal JSCC allocation for each video frame, and PHY transmission rate for each outgoing transport packet. The comprehensive simulation results obtained with the H.264/AVC codec demonstrate considerable increase in the PSNR of the decoded video when compared with a system that employs separately JSCC and PHY rate adaptation. Furthermore, our performance analysis indicates that the optimal PHY transmission rate calculated by the proposed algorithm, can be significantly different when compared with rate adaptation algorithms that target throughput improvement.