基于网络的媒体流同步算法研究

同步是多媒体业务中一个重要的服务质量(QoS).分析网络中多媒体传输同步的基本问题,提出一种媒体内自适应缓冲控制方案.该方案采用接收瑞缓存媒体单元补偿网络的时延抖动,缓冲区采用循环队列的结构,当缓冲区达到要求预取数时,开始从缓冲区取帧.测试结果表明,应用该缓冲区方案能够补偿由于网络传输时延特性变化而引起的失步,从而使接收端播放平滑连续,对媒体通信质量有较大提高.     

网络环境下视频连续播放控制算法

网络的时延抖动会造成媒体播放的不连续,本文从信宿端的角度来解决视频媒体的同步连续播放,提出了一种自适应的动态媒体播放算法.该方法在接收端设置缓冲区,根据缓冲区的占用情况动态地调整媒体的播放速率,播放速率是根据具体的媒体内同步要求选取的.另外,分析了马尔可夫调制的泊松到达情况下的排队模型,给出了缓冲区门限的选取原则.最后,用不连续性和播放失真的方差衡量同步性能,计算机模拟结果表明,该算法的同步性能优于Yuang的算法.通过选择合适的参数,该算法可以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,从而实现同步平滑播放.     

通用组帧规程(GFP)帧同步性能研究

通用组帧规程(GFP)是一种具有实现简单和低开销特点的新型数据链路协议,其帧同步采用基于帧标头循环冗余校验(CRC)的多个帧搜捕器方案。采用状态转移法对其帧同步性能进行研究,并给出计算和数值分析结果。针对该方案帧失步平均持续时间过长的缺点,提出了一种GFP帧同步机制的改进方案。理论分析和计算结果表明,此方案可以有效降低GFP帧同步的失步平均持续时间,改善GFP帧同步系统的性能。     

OFDM系统原理及仿真实现

多媒体系统必须能够限制时延抖动的变化,通过在播放端设置缓冲区以抵消时延抖动,从而保证媒体内及媒体间的同步。根据多源流媒体的分发特点,文章设计了一种分组的媒体缓冲区大小的设置方案,该方案在保证流媒体同步的基础上,巧妙地解决了流媒体不同码率的缓冲区大小设置问题。     

媒体PND参考解决方案

RMI与CSR公司和SiGe半导体两家公司共同合作，为其mPND（mediaenabled PDN）参考解决方案提供低成本的GPS实现方案。RMI的mPND参考设计基于其AlchemYAu1250和Au1210媒体处理器，能够加快和简化功能丰富的新型便携式媒体产品的设计。这两款创新的RMI Alchemy处理器现已大批量生产，并付运至各种具有媒体播放能力的多功能导航器件。它们能够实现全帧速率的视频播放，而且无须转码就可以提供DVD品质的视频和动态音频。     

一种新的多路媒体流复用调度算法

文中针对多路媒体流的调度问题,在分析经典的缓冲区反馈控制算法和按DTS调度算法基础上,提出了一种基于权重的调度算法,该算法对各路媒体流均动态分配权重,调度前比较各路媒体流的权重,每次选择权重最大的媒体流进行调度,它吸取了其它多路媒体流调度算法的优点,并充分考虑了多路媒体流调度的优先级问题。试验结果表明该算法是切实可行的和有效的。     

基于细粒度调整的同步播放控制算法

针对网络时延和时延抖动引起缓存波动而造成的流媒体播放不连续,基于接收端播放速率调整,提出一种播放速率自适应调整的播放缓存控制算法。算法依据缓存区的占用水平,通过构造控制序列,动态、细粒度地调整媒体的播放速率。在校园网环境下的模拟结果表明,该算法能够有效地减少缓存上溢或下溢引起的播放跳跃或停顿,从而实现流媒体的平滑播放。     

H.323同步控制实现研究

提出并实现了一种H.323多媒体通信系统音视频同步控制算法,算法以正常语音播放为基准,利用H.263码流语法信息,通过调整视频播放帧率实现媒体间同步, 不增加通信冗余度。实际测试结果表明,算法有效地提高了多媒体通信系统音视频媒体同步性能和多媒体的感知服务质量,并已应用于实际多媒体通信系统。     

基于窄变带宽的实时视频传输算法

根据视频流与实时监视应用的特点,提出一种基于TCP协议的动态双缓冲与双线程视频实时传输算法。在发送方设置视频数据缓存与数据发送缓存,并分别由视频数据输入线程与视频数据发送线程负责管理;视频数据输入线程根据预设的最大等待发送时间与实时计算的网络传输速率,动态调节缓存的大小以及在网络拥塞时有选择性地丢弃视频帧;视频数据发送线程实现视频数据发送与按帧从视频数据缓存获取数据,并实时计算出网络数据传输速率。实验结果表明,本算法能最大限度地利用动态变化的网络带宽,保证视频实时发送至接收方与平稳播放,可有效地应用于窄变带宽网络环境下实时视频监视。     

多媒体邮件系统中媒体同步的研究与实现

设计并实现了一种基于H.323协议的多媒体邮件系统。为了解决多媒体同步问题、在终端侧,利用媒体同步控制模块根据RTP包头中附带的时间戳标志,对视频、音频信息进行同步控制;在服务器侧,采用基于时间轴的定义方法建立的存储模型,直接对媒体流的RTP包进行存储,以保持发送端的时间戳不变,使发送端的媒体及媒体内同步关系不变。     

Communication Under Strong Asynchronism

A formulation of the problem of asynchronous point-to-point communication is developed. In the system model of interest, the message codeword is transmitted over a channel starting at a randomly chosen time within a prescribed window. The length of the window scales exponentially with the codeword length, where the scaling parameter is referred to as the asynchronism exponent. The receiver knows the transmission window, but not the transmission time. Communication rate is defined as the ratio between the message size and the elapsed time between when transmission commences and when the decoder makes a decision. Under this model, several aspects of the achievable tradeoff between the rate of reliable communication and the asynchronism exponent are quantified. First, the use of generalized constant-composition codebooks and sequential decoding is shown to be sufficient for achieving reliable communication under strictly positive asynchronism exponents at all rates less than the capacity of the synchronized channel. Second, the largest asynchronism exponent under which reliable communication is possible, regardless of rate, is characterized. In contrast to traditional communication architectures, there is no separate synchronization phase in the coding scheme. Rather, synchronization and communication are implemented jointly. The results are relevant to a variety of sensor network and other applications in which intermittent communication is involved.     

一种 1x2x4分布式空时编码异步协作方案

协作分集技术可以有效提高移动通信系统的容量和可靠性。近年来,分布式空时码（DSTC）被广泛应用到协作分集系统以实现空间分集增益。然而,时间异步和频率异步会使正交DSTC的码字结构受到破坏,严重影响系统性能。本文针对时间异步的协作通信系统,提出一种基于分布式Alamouti STBC的1x2x4异步协作方案。对这种方案进行了理论分析和算法推导,并且重新定义了中继节点归一化后的发射信号幅度。仿真结果表明该方案的误比特率性能具有明显优势,能在时间异步的协作通信系统中很好地发挥多天线接收性能。     

Long-term rate control for concurrent multipath real-time video transmission in heterogeneous wireless networks

Concurrent multipath transmission provides an effective solution for streaming high-quality mobile videos in heterogeneous wireless networks. Rate control is commonly adopted in multimedia communication systems to fully utilize the available network bandwidth. This paper proposes a novel rate control for concurrent multipath video transmission. The existing rate control algorithms mainly adapt bit rate in the short-term pattern, i.e., without considering the long-term video transmission quality. We propose a long-term rate control scheme that takes into account the status of both the transmission buffer and video frames. First, a mathematical model is developed to formulate the non-convex problem of long-term quality maximization. Second, we develop a dynamic programming solution for online encoding bit rate control based on buffer status. The performance evaluation is conducted in a real test bed over LTE and Wi-Fi networks. Experimental results demonstrate that the proposed long-term rate control scheme achieves appreciable improvements over the short-term rate control schemes in terms of video quality and delay performance.     

Uncoordinated rate-division multiple-access scheme for pulsed UWB signals

Popular multiple-access schemes for asynchronous users that access the channel at randomly or pseudorandomly chosen time instances are ALOHA and TH-PPM. To cope with the randomness of these signals can be challenging for a receiver. In this paper, an uncoordinated but deterministic multiple-access scheme is proposed that can eliminate some of these drawbacks. The principle of this scheme is that each user transmits with an individual pulse or packet rate, while the duty cycle of the user signals is kept very low. As the access to the channel is asynchronous, collisions will occur. An analytical expression is derived for the collision probability, which depends on the asynchronism between the users. Design rules are derived that make the collision probability independent of the asynchronism. In practical implementations, deviations from the design values for the user rates will occur, and the length of a data packet is limited; the impact of these aspects on the collision probability is discussed. Simulation of a sensor network scenario without transmit power control shows that the design rules that make the collision probability insensitive to the asynchronism also do this for the bit-error rate (BER). The comparison of simulation results for RDMA with binary antipodal modulation and random TH-PSK, which is related to TH-PPM, yields very similar bit-error rates.     

Multiple User Variable Rate Coding for TASI and Packet Transmission Systems

In this paper we examine the use of variable rate coding concepts for TASI and packet speech transmission systems. The paper is divided into three major parts. In the first part, the theoretical performance of variable rate coding is analyzed for multiple user (TASI) applications. Potential gains are experimentally determined from twoparty telephone conversation data for up to 12 shared conversations on a channel. In the second part of the paper, the buffer control mechanism for a dynamic buffer scheme for coupling a variable rate coder to a fixed rate (or slowly varying rate) channel is analyzed. It is shown that the buffer control can be modeled as a second-order control system and, under adverse parameter settings, the system can be unstable. By an appropriate design and parameter setting, the buffer control can be stabilized, The insight developed from this particular buffer control mechanism may also lead to a better understanding of other buffer control problems in variable rate transmission or packet systems. In the third part of the paper, a practical method is analyzed for implementing a variable rate ADPCM system for multiple user applications. Examples of computer simulations of the system are presented.     

视频监控网络中目标特征的虚拟同步

视频监控网络已经成为计算机视觉的研究热点之一。文章分析了视频监控网络中数据不同步的原因．提出了使用目标缓冲池链进行虚拟同步的方法。在此基础上讨论了对目标特征数据进行处理的三层结构．解决了视频监控网络系统中的数据不同步问题。实验结果表明得到的虚拟同步数据真实可靠，实现简单，具有较强的实用性。     

Delay constrained throughput optimisation with imperfect CSI using discrete adaptive power

Link adaptation is an effective tool to overcome fading effects in wireless links. However, time-varying adaptive transmission rate leads to queueing delay, and moreover, in practise, imperfect channel state information (CSI) is available for the transmitter to adapt its transmission rate and power. This article aims to consider practical constraints to enhance the link adaptation scheme. We reformulate and optimise buffer delay constrained throughput of a wireless link based on outdated noisy CSI. Discrete power adaptation is proposed, in which a limited number of feedback bits (just the index of transmission power level) is required, while the performance is improved compared to the constant power and is close to continuous adaptive power. A unified scheme is set-up, where constant, discrete or continuous adaptive power transmission is utilised considering average or instantaneous bit error rate constraints based on imperfect CSI. The effectiveness of our designs is evaluated by numerical evaluations.     

The ready-to-go virtual circuit protocol: a loss-free protocol formultigigabit networks using FIFO buffers

The ready-to-go virtual circuit protocol (or RGVC) is an immediate transmission protocol, in which the source need not wait for an end-to-end roundtrip delay for reservations to be made before transmitting the data. The protocol is designed to handle the lossless transport of ABR traffic, and will be used in the 40 Gb/s Thunder and Lightning testbed being prototyped at the University of California at Santa Barbara (UCSB). An important advantage of the RGVC protocol over previous connection and flow control protocols is that it is suitable for networks in which the switches use FIFO buffers that are shared by multiple sessions. The RGVC protocol ensures lossless communication by coupling link capacity with buffer space, so that when a portion of a buffer at a node is occupied, a proportional fraction of the incoming capacity to that buffer is frozen. Given the constraints on the frozen capacity, an algorithm is executed at each node to allocate the transmission rate to each FIFO buffer so as to maximize capacity utilization. The requirement that the protocol operate with FIFO buffers at the network nodes poses some unique challenges in the design that are not present in rate- and credit-based schemes. Briefly, since several sessions share a common FIFO buffer, per-VC flow control is no longer possible so control over the rate of an individual session is lost. Also, since the contents of the buffers change dynamically, the buffer composition becomes difficult to determine. For the rate-allocation algorithm of the RGVC protocol to be executed, however, the contents of the FIFO buffers at a node must be known, To implement the bookkeeping required, we present two schemes: the measurement-based scheme, where the bookkeeping function is implemented via measurements, done essentially in hardware; and the estimation-based scheme, where the bookkeeping is done analytically via the exchange of control packets between nodes     

Intelligent Flow Control Technique of ABR Service in ATM Networks Based on Fuzzy Neural Networks

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An adaptive congestion control scheme for real time packet videotransport

We show that modulating the source rate of a video encoder based on congestion signals from the network has two major benefits: the quality of the video transmission degrades gracefully when the network is congested and the transmission capacity is used efficiently. Source rate modulation techniques have been used in the past in designing fixed rate video encoders used over telephone networks. In such constant bit rate encoders, the source rate modulation is done using feedback information about the occupancy of a local buffer. Thus, the feedback information is available instantaneously to the encoder. In the scheme proposed, the feedback may be delayed by several frames because it comes from an intermediate switching node of a packet switched network. The paper shows the proposed scheme performs quite well despite this delay in feedback. We believe the use of such schemes will simplify the architecture used for supporting real time video services in future nationwide gigabit networks