嵌入式流媒体客户端缓存技术

介绍了基于嵌入式微处理器S3C2440的嵌入式流媒体系统的硬件结构和工作流程. 服务器端通过RTP/RTCP协议将流媒体数据发送出去,客户端对收到的数据进行解压并实时播放. 将接收缓存分成接收缓冲区、播放缓冲区和DMA缓冲区,三个缓冲区的大小按1：1：2的比例设置,通过平均速率、延时抖动和解码码率等参数来约束缓冲区的容量. 在接收缓冲区设置两个临界点,通过对两个临界点的检测,来辅助调节发送端的数据发送速率. 既可以避免网络拥塞,又可以提高流媒体的传输质量.     

流媒体网络传输中基于速率的控制方法

介绍了一种新的基于速率的拥塞控制方法，重点讨论了速率控制方法在流媒体网络传输中的具体应用。该方法通过监测网络状态变化和客户端缓冲区状态，动态调整服务器端的发送速率，以提高服务质量。研究表明，相对于传统的基于窗口型拥塞控制方法，速率控制方法在流媒体网络传输中可取得更好的效果。     

嵌入式视频点播系统中流媒体数据的缓存策略

基于嵌入式终端的视频点播已逐渐成为当前网络的主流服务,随着视频文件数据量的增加,网络拥堵,服务质量(Qo S)开始下降。为了保证Qo S,视频点播终端中需要分配一定的存储资源作为缓存。但嵌入式系统中可用存储空间有限,通用的数据缓存方式无法适用。本文针对网络延迟和网络抖动问题,提出一种基于终端的延迟回放缓存策略,对流媒体数据的抖动和缓冲区容量的管理给出了解决方案。该策略提高了流媒体播放的顺畅性。     

嵌入式流媒体服务器的设计和实现

为了解决嵌入式环境下的视频传输,提出了一个嵌入式流媒体服务器的应用设计方案.采用了流媒体传送技术、控制协议技术、进程间通信技术,使用开放源码流媒体服务器live555项目设计和实现了一个嵌入式流媒体服务器,解决了四个技术问题:管道读写视频操作问题,服务器传送视频控制字问题,服务器与客户端视频播放控制协议问题,流媒体服务器异常退出后恢复问题.该流媒体服务器能够正确传输视频数据,客户端能够播放实时视频,播放画面图像质量高、时延小、稳定可靠.     

基于客户端缓冲区预警界限的流媒体传输速率控制方法

介绍了一种新的基于速率的网络拥塞控制方法。该控制方法通过监测网络状态变化和客户端缓冲区状态,动态调整服务器端的发送速率,能够很好提高服务质量。通过研究表明,该方法在流媒体网络传输中能够取得很好的效果。     

基于S3C2410与MPEG-4的播放器的设计和实现

流媒体(Streaming Media)是一种新兴的网络传榆技术,广泛应用在互联网上实时传输、播放音、视频等多媒体数据,流媒体技术包括流媒体数据的采集、音/视频编解码、存储、传输以及播放等领域.嵌入式流媒体,顾名思义,需要考虑嵌入式系统以及流媒体技术两方面的特点.本文介绍了一种基于S3C2410与MPEG-4的嵌入式流媒体播放器的软件和硬件的设计与实现.     

流媒体系统中基于速率的拥塞控制方法研究

在比较和研究各类基于速率的拥塞控制机制的基础上．构建出一种新的流媒体系统中的速率型拥塞控制方法。该方法通过监测网络状态变化和客户端缓冲区状态．动态调整服务器端的发送速率,减少包丢失率和缩短延迟时间,提高系统服务质量。此方法表现出很好的灵活性和稳定性,计算复杂度很低,不影响传输效率。和传统的拥塞控制机制相比,此方法在控制的实时性、网络状态震荡抑制、多媒体通信质量等方面均有一定的改善和提高。     

Internet上基于两端的流媒体QoS控制

在Internet这类无法可靠保证服务质量QoS的网络环境中,拥塞和带宽的不足成为流媒体应用的主要瓶颈,所以,流媒体应用的关键是QoS控制机制.QoS控制大致分为两大类:基于路由和基于两端.基于两端的QoS控制技术的基本思想是:网络状况好的时候多传些数据增强播放质量,网络状况不好的时候则只传输能够达到基本播放效果的数据.即QoS技术能够动态地调节传输数据量的多少.主要分析和介绍了相关的流媒体编码技术、基于客户端的QoS控制和基于服务器端的QoS控制技术.     

基于MPEG-4的流媒体服务器端缓冲区管理

服务器端缓冲技术是流媒体系统中常用的一种应用控制技术，能够平滑网络抖动，减少数据丢失，提高客户端视频播放质量。该文的缓冲区管理增加了码率控制的功能，控制策略能进一步提高视频质量，特别适合于系统资源有限、带宽窄或强实时性的视频流式传输。     

一种具有双重QoS保证的VOD流媒体系统结构

文中提出了一种基于IP网络具有双重QoS保证的VOD流媒体系统结构.此结构使用RSVP协议对流媒体预留资源,资源预留采用基于节的资源管理方法;服务器端通过RTP协议传输实时多媒体数据流,RTCP协议监视传输服务质量进行动态QoS控制,因此对基于IP网络的VOD系统的流媒体传输QoS提供了双重保证.其客户端使用SDP协议取得流媒体文件描述信息和组播地址,通过RTSP协议与流媒体文件服务器交互,建立和控制多媒体会话.     

Design and implementation of MP3-music on demand system using streaming technology

The success of the Internet and the use of broadband in homes have caused a gradual shift in traffic on the Internet from data to multimedia communication. Multimedia applications typically include a large quantity of video/audio information. Streaming technology is normally adopted to handle the transmission of multimedia traffic and thus reduce the buffer requirement on the client side and the service request/response time. This work focuses on the transmission of MP3 music which has a constant bit rate characteristic. The design of both the server side and the client side of the MP3-music on demand (MoD) system with streaming technology, is considered to meet the quality of service (QoS) requirements of MP3 music. A stream buffering technique is used and an adaptive rate control mechanism is applied in combination with a client feedback packet to prevent stream buffer overflow or underflow on the client side, and thereby accommodate the network delay, jitter, and timing deviation between the server machine and the client host. A server self-timing revision scheme is used to reduce the network overhead of the feedback mechanism. The adaptive rate control mechanism is developed and verified using a computer simulation. Finally, for completeness a MoD system is constructed with a low-cost embedded network system to which an Altera FPGA is applied to provide cut-through data movement and an adaptive rate control mechanism is realized to evaluate QoS.     

一种多方通话动态控制技术

近些年多媒体技术和数据通信技术发展迅速,多方音视频通话逐渐兴起.多方音视频通话能够解决人们远距离通信问题,但在媒体流控制方法上,现存通话系统并未将集中式和散布式两种媒体流控制方式进行有效结合,无法发挥两种控制方法的各自优势.针对上述问题,提出了一种多方通话动态控制技术,以客户端网络带宽和用户数量作为变换条件,通过对星状网络和网状网络两种拓扑结构的结合,使多方通话系统能够自适应调节媒体流分发方式.本系统以SIP会议控制为基础并运用WebRTC技术提高通话质量,实验结果表明,该技术可有效掌握系统内各客户端网络拥塞状况和资源占用率,对系统媒体流控制方式能够适时进行调整,使媒体流传输通畅高效,提高通话质量.     

A unified benchmarking and model-based framework for building QoS-aware streaming media services

A number of technology and workload trends motivate us to consider the appropriate resource allocation mechanisms and policies for streaming media services in shared cluster environments. We present MediaGuard – a model-based infrastructure for building streaming media services – that can efficiently determine the fraction of server resources required to support a particular client request over its expected lifetime. The proposed solution is based on a unified cost function that uses a single value to reflect overall resource requirements such as the CPU, disk, memory, and bandwidth necessary to support a particular media stream based on its bit rate and whether it is likely to be served from memory or disk. We design a novel, time-segment-based memory model of a media server to efficiently determine in linear time whether a request will incur memory or disk access when given the history of previous accesses and the behavior of the server's main memory file buffer cache. Using the MediaGuard framework, we design two media services: (1) an efficient and accurate admission control service for streaming media servers that accounts for the impact of the server's main memory file buffer cache, and (2) a shared streaming media hosting service that can efficiently allocate the predefined shares of server resources to the hosted media services, while providing performance isolation and QoS guarantees among the hosted services. Our evaluation shows that, relative to a pessimistic admission control policy that assumes that all content must be served from disk, MediaGuard (as well as services that are built using it) deliver a factor of two improvement in server throughput.     

基于网格的流媒体服务QoS管理框架及实现

流媒体服务是Internet上一类高带宽需求和高实时性约束的应用，对服务质量（Quality of Service，QoS）有较高的要求。流媒体服务的发展导致传统的QoS管理框架难以适应平台的异构性和复杂性。本文提出了一种基于网格的流媒体服务QoS管理框架，为由异构的系统构成的流媒体服务提供集成的、平台无关的QoS管理机制。在谈框架的基础上，我们设计了一个基于网格的流媒体服务QoS管理系统。     

一种TCP友好的Internet DTV组播发送速率控制机制

当前的Internet没有对流媒体应用提供足够的QoS的保证,同时Internet DTV的组播发送速率必须满足以下两点要求:1)能够自适应网络拥塞的变化;2)能够适应节目码率的要求,因此需要在服务器端进行组播发送速率控制.分析了Internet DTV组播的难点,提出了一种基于缓冲区管理的具有网络自适应特性的组播发送速率控制方法.通过合理地控制发送方的发送速率,既能自适应网络状况的变化,又能满足流媒体实时播放的需求.仿真实验结果表明,该算法在保证服务器端发送缓冲区不溢出前提下,与单纯采用TFMCC方法比较,减少了接收端的平均丢包率,提高了网络电视节目流的传输质量.     

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.     

基于实时流协议的流媒体客户端

介绍了实时流和实时流协议，针对基于实时协议的流媒体客户端播放器的特殊要求，提出了一种适用于视频点播（Video On Demand)的流媒体客户端结构模型，研究并设计了相应的流量控制策略、组包算法和缓存管理策略，试验结果表明，该客户端播放器占用系统资源少，具有良好的实时性、容错性和同步性。     

Rate-sensitive leverage of QoS and QoP for ubiquitous video streaming via buffer-aware feedback control

Providing real-time Internet video streaming anytime, anywhere and using any devices from different access networks preserves more challenges to equilibrate the quality of service (QoS) and security protection (QoP). Because encryption/decryption for video packets are time-consuming processes to protect real-time video streaming services from eavesdropping, our observation is that the playback buffer occupancy (PBO) can simply indicate time availability to adjust security level to affect the packet sending rate. In this paper, we present an end-to-end buffer-aware feedback control from client PBO for effectively securing media streaming for heterogeneous clients over ubiquitous Internet. That is, security-level adjustments can be applied further to keep PBO running away from overflow and underflow to pursue an effective leverage between QoS and QoP. To further boost the protection, we also apply the Diffie-Hellman key negotiation method to provide the dynamic key changes. Moreover, since the running PBO will vary on the dynamics of Internet from access time, client devices and access networks, the different applied security levels and key changes during the video streaming session will make eavesdropper more difficult to recover all the encrypted videos delivered in public networks. We demonstrate the leverage performance in preserving both QoS and QoP for ubiquitous video streaming in our proposed schemes by comprehensive experiments on a true VoD system. The experimental results show our secure VoD scheme can achieve cost-effective leverage of QoS and QoP from different inserted network dynamics, even if client buffer size is limited to 256 KB only.     

A Statistical Admission Control Scheme for Continuous Media Servers Using Caching

In continuous media servers, disk load can be reduced by using buffer cache. In order to utilize the saved disk bandwidth by caching, a continuous media server must employ an admission control scheme to decide whether a new client can be admitted for service without violating the requirements of clients already being serviced. A scheme providing deterministic QoS guarantees in servers using caching has already been proposed. Since, however, deterministic admission control is based on the worst case assumption, it causes the wastage of the system resources. If we can exactly predict the future available disk bandwidth, both high disk utilization and hiccup-free service are achievable. However, as the caching effect is not analytically determined, it is difficult to predict the disk load without substantial computation overhead. In this paper, we propose a statistical admission control scheme for continuous media servers where caching is used to reduce disk load. This scheme improves disk utilization and allows more streams to be serviced while maintaining near-deterministic service. The scheme, called Shortsighted Prediction Admission Control (SPAC), combines exact prediction through on-line simulation and statistical estimation using a probabilistic model of future disk load in order to reduce computation overhead. It thereby exploits the variation in disk load induced by VBR-encoded objects and the decrease in client load by caching. Through trace-driven simulations, it is demonstrated that the scheme provides near-deterministic QoS and keeps disk utilization high.