端到端的流媒体传输控制技术研究综述

随着多媒体应用的发展,因特网上的流媒体传输技术已成为研究热点。在因特网上传输音频或视频流需要有带宽、延迟、丢包率等诸多的QoS要求,但当前的因特网并不提供任何QoS保证,这对流媒体在因特网上的传输提出巨大挑战。该文从端到端的传输控制技术角度入手,给出了一个因特网上流媒体传输的总体框架,然后依此框架为线索,对流媒体传输所必需的协议栈、拥塞控制、自适应速率编码、速率整形、差错控制等技术的研究进展进行了概括总结并进行了对比,同时提出了进一步的研究建议。     

结合信源特性与网络拥塞控制的可靠性视频传输算法

提出了一种用于在无线网络中传输视频的结合信源特性及网络拥塞控制的鲁棒性算法.通过场景建模以及特性分析,将分级编码产生的所有码流层划分成不同的类型,并根据它们对网络拥塞控制的贡献以及对重建图像质量的贡献不同,将其分成两个不同的队列.系统根据不同的网络丢包状态(即丢包是由网络拥塞引起还是由无线信道的不可靠传输引起)动态地调整信源速率、不等错误保护强度以及拥塞控制策略.仿真结果表明,该方法与MPEG-4信源编码加固定速率Turbo码方法以及动态调整信源、信道编码速率加选择性丢I,B,P包的网络拥塞控制方法相比,能够提供更好的性能.     

Adaptive efficient video transmission over the Internet based on congestion control and RS coding

An approach based on adaptive congestion control and adaptive error recovery with RS (Reed-Solomon) coding method is presented for efficient video transmission over the Internet. Featured by weighted moving average rate control and TCP-friendliness, AVSP, a novel adaptive video streaming protocol, is designed with adjustable rate control parameters so as to respond quickly to the QoS status fluctuation during video transmission over the Internet. Combined with congestion control policy, an adaptive RS coding error recovery scheme with variable parameters is presented to enhance the robustness of MPEG video transmission over the Internet with restriction to the total system bandwidth .     

An integrated source transcoding and congestion control paradigmfor video streaming in the Internet

In this work we present an end-to-end optimized video streaming system comprising of synergistic interaction between a source packetization strategy and an efficient and responsive, TCP-friendly congestion control protocol [Linear Increase Multiplicative Decrease with History (LIMD/H)]. The proposed source packetization scheme transforms a scalable/layered video bitstream so as to provide graceful resilience to network packet drops. The congestion control mechanism provides low variation in transmission rate in steady state and at the same time is reactive and provably TCP-friendly. While the two constituent algorithms identified above are novel in their own right, a key aspect of this work is the integration of these algorithms in a simple yet effective framework. This “application-transport” layer interaction approach is used to maximize the expected delivered video quality at the receiver. The integrated framework allows our system to gracefully tolerate and quickly react to sudden changes in the available connection capacity due to the onset of congestion, as verified in our simulations     

Internet视频流传输策略分析

以分组交换和TCP/IP传输协议为主要技术基础的Internet是个只提供Best-Effort的异构网络,带宽随位置和时间随机变化,由拥塞而导致的分组丢失、传输延迟及抖动不可避免。为了在Internet上有效、高质量地传输视频,需要根据信道特性设计网络和编码器接口,即JSCC(信源信道联合编码)。文中在Internet视频传输系统结构的基础上,从拥塞控制、差错控制、面向传输的视频编码等几方面分析了基于JSCC的视频传输控制策略,并对Internet视频流传输前景做了展望。     

AVP: A Highly Efficient Transport Protocol for Low Bit Rate Multimedia Communications

Utilization of Internet communications in distance learning, distributed simulation, and distributed work groups involves multimedia transmission of animation, voice and video clips. Highly compressed audio-video data protocols are required for efficient Internet multimedia communications. Addressing this requirement, a new transport protocol called Audio-Video Protocol (AVP) for highly efficient multimedia communications on the Internet is presented. While providing similar real-time delivery functions as Real-Time Transport Protocol (RTP) and Real-Time Control Protocol (RTCP), AVP adopts a novel audio-based synchronization scheme. This synchronization scheme has two advantages. One is the overhead reduction through eliminating the timestamp in each transmitted data packet. The other is the packet rate reduction by putting multiple audio frames or mixed audio-video frames in a single AVP packet. As a result, the end-to-end media unit delay is reduced while achieving implicit synchronization. Furthermore, AVP provides adaptive quality of service (QoS) by the prioritized packetization scheme. Simulation results are presented to verify the advantages of the AVP protocol.     

联合拥塞控制与H.264容错编码的视频传输

视频压缩码流在传输过程中经常会遇到因各种原因引起的误码丢包等网络传输错误。由于目前常用的编码器普遍采用“预测-残差”方式进行编码，因此一旦传输发生错误，不仅会影响当前帧的重建，更会造成空间及时间上的误差蔓延。在传输系统中，较为实际的做法是由系统各个部分共同分担这种抗网络传输差错的“责任”，通过前后部分之间不断的呼应协作，合理地满足用户对于视觉质量、实时性等各个不同方面的QoS要求。为了对抗网络传输错误对视频重建质量的影响，在结合了H．264容错编码、信道保护、打包策略、拥塞控制、差错掩盖等算法的基础上，提出了一个集成拥塞控制的端对端视频传输系统框架，并对其中关键算法进行了分析与设计，实验证明，该视频传输系统框架在不同网络情况下，均可收到较好的效果。     

基于实时流媒体传输系统的H.264组包算法研究

有线和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组包策略的选取是其中的一个关键问题.本文首先设计了一种基于RTP的H.264实时流媒体传输系统模型,在此基础上提出了适用于H.264实时视频流传输的RTP组包算法HMP.该算法针对H.264 NAL视频流特点,在考虑到视频流关联性的同时加入了对重要信息的保护.实验结果证明,HMP算法在网络丢包率较大的情况下仍能获得良好视觉质量.     

Cost-based cache replacement and server selection for multimedia proxy across wireless Internet

Multimedia proxy plays an important role in multimedia streaming over wireless Internet. Since wireless network exhibits different characteristics from the Internet, multimedia proxy caching over wireless Internet faces additional challenges. In this paper, we present a study of cache replacement for a single server and server selection for multiple servers across wireless Internet. By considering multiple objectives of multimedia proxy, we design a unified cost metric to measure proxy performance in wireless Internet. Based on the defined unified cost metric, we propose a novel replacement algorithm for single-server and a new server-selection policy for multiple servers to improve the end-to-end performance such as throughput, media quality, and start-up latency. To effectively handle errors occurred on wireless link, channel-adaptive unequal error protection is deployed according to distinct quality of service requirements of layered or scalable media. Simulation results demonstrate that our approaches achieve significantly better performance than the known cache-replacement algorithms and sever selection schemes, respectively.     

基于TCP友好速率控制和前向纠错的MPEG-2视频传输

针对Internet视频传输面临拥塞控制和数据包丢失的问题，结合TCP友好的速率控制算法和前向纠错机制建立视频传输的分层体系构架和控制策略。传输体系同时采用以GOP为基本分析单元的视频帧速率预测模型，实现根据网络丢包率的变化动态地优化配置前向纠错的冗余信息。实验证明，传输体系采用动态优化的前向纠错能实时地适应带宽的变化，有效地降低数据包丢失带来的影响，从而改善视频回放质量。     

流媒体自适应传输控制的研究

流媒体是指多媒体数据流在网络上一边传输一边播放的一种多媒体通信服务.提供尽力而为服务的Internet不能为流媒体保证网络带宽、传输延迟、分组丢失以及分组错误等,而自适应传输控制机制能够提高流媒体服务的传输服务质量和传输服务的公平性.本文探讨流媒体自适应传输控制技术所涉及的各个方面,包括拥塞控制、质量自适应和错误控制.     

AVS视频的分层重传优化策略与实现

针对网络抖动等因素会影响图像质量的问题,采用差错控制与拥塞控制相结合的恢复技术,改进AVS视频数据在传输层的RTP打包,提出提高重传效率的AVS视频分层重传策略,通过令牌发放的方式控制数据包重传的次数解决重传导致网络拥塞问题。实验结果表明,在低带宽的网络下,该策略对AVS视频传输相对于无令牌发放的重传方式图像质量有一定的提高。     

TCP友好的多媒体通信流控制机制

该文给出了一个端到端的适应性多媒体流控制策略,称为基于丢失延时的算法LDA(loss-delaybasedalgo-rithm),它调整多媒体流的发送行为,以符合网络的当前拥塞状况。LDA算法利用实时传输协议RTP(real-timetransportprotocol)来收集分组丢失和延时统计信息,并用来调整发送端的发送行为,使它和TCP的拥塞控制有类似的统计特性,是TCP友好的。仿真的结果显示LDA算法对于网络资源的利用、拥塞避免和TCP流的公平性都是比较好的。     

基于小波EBCOT的图像IP网络传输控制策略

为了使当前“尽力而为”的网络提供视频流服务时满足QoS要求，文章提出一种基于小波EBCOT的图像IP网络传输控制策略．通过采用基于小波EBCOT的渐进可分级编码方法，对压缩后的比特流按其重要性分层打包传输，同时根据对当前网络可用带宽的估计及信道状态的判断，区分网络拥塞及不可靠传输两种不同情况进行自适应不等重丢包保护AUPLP．软件仿真表明，该文算法可大大增强小波EBCOT编码后数据的抗误码能力，在发生数据拥塞时有助于缓解网络的过负载状况，在发生不可靠传输时接收端解码图像能平均提高1．2dB的PSNR。     

无线网络视频传输差错控制技术研究

随着多媒体、网络技术以及移动通信的发展,视频通信的应用成了必然的趋势。传输的视频需要进行压缩,冗余信息的丢失使视频数据在传输中抵抗信道误码的能力变得十分脆弱。不幸的是,无线网络信道中,误码的产生、数据的丢失总是难以避免。当网络拥塞时更容易造成突发性的分组丢失现象,引起图像质量严重下降,必须采用有效的差错控制技术进行处理。本文提出并实现了一种适合于无线网络环境下视频传输的差错控制方法,它包括一个基于丢包检测的流量自适应算法,编码端的宏块重排序的算法和解码端的自适应的差错掩藏算法。实验结果表明,在无线网络环境下,采用本文提出的差错控制方法能够有效提升视频传输的质量。     

Optimized H.264 Video Encoding and Packetization for Video Transmission Over Pipeline Forwarding Networks

Previous works showed that the quality-of-service (QoS) requirements of multimedia applications can be optimally satisfied by pipeline forwarding (PF) by providing end-to-end delay guarantees as well as high network resource utilization. However, the unavoidable mismatch between reserved resources and the unpredictable traffic profile of a video stream has an impact on the resulting application layer quality. Therefore, a new low-complexity H.264 video encoding and packetization scheme based on a distortion-optimized macroblock grouping technique is designed here to maximize the performance of video transmission on PF networks. The scheme considers the perceptual importance of the different parts of the video data to group the most important information in few packets that are the natural candidates to receive the deterministic service provided by PF. Results show peak signal-to-noise ratio (PSNR) gains up to 2.5 dB over traditional video encoding and packetization schemes, as well as more graceful degradation in case of high network load.     

视频实时传输中的速率控制研究

视频在互联网上的实时传输常因其要求高带宽、低延迟而造成网络拥塞。传统TCP基于窗口的拥塞控制已经不适用于实时传输中的拥塞控制,该文讨论了基于速率的网络拥塞控制方法,它采用自适应码率和分层的视频编码技术,详细说明了基于源端和收端的各种速率控制方法。     

SVC在无线信道传输中的非均衡差错保护

针对H.264的可伸缩视频编码扩展标准(SVC)在噪声信道中的传输,采用低密度奇偶校验码(LDPC)提出一种非均衡差错保护的方案。在所提的方案中,根据时间、分辨率和质量把原视频序列按重要性分成不同的层。由于不同层的数据对错误的敏感性不同,对其进行不同码率的LDPC信道编码,实现非均衡差错保护。根据视频流中每一帧不同层的PSNR增量不同,和不同信道码率下正确解码的概率不同,反复计算每一帧所有码率组合的PSNR增量值并找出最大组,从而进行信道编码并传输。实验表明,在相同的平均码率条件下,提出的方案相比其他方案的PSNR值增加了2.8 dB,更适合无线信道的传输。     

Resource allocation for multimedia streaming over the Internet

This paper addresses the resource allocation problem for multiple media streaming over the Internet. First, we present an end-to-end transport architecture for multimedia streaming over the Internet. Second, we propose a new multimedia streaming TCP-friendly protocol (MSTFP), which combines forward estimation of network conditions with information feedback control to optimally track the network conditions. Third, we propose a novel resource allocation scheme to adapt media rate to the estimated network bandwidth using each media's rate-distortion function under various network conditions. By dynamically allocating resources according to network status and media characteristics, we improve the end-to-end quality of services (QoS). Simulation results demonstrate the effectiveness of our proposed schemes     

Two-Dimensional Channel Coding Scheme for MCTF-Based Scalable Video Coding

The motion-compensated temporal filtering (MCTF)-based scalable video coding (SVC) provides a full scalability including spatial, temporal and signal-to-noise ratio (SNR) scalability with fine granularity, each of which may result in different visual effect. This paper addresses a novel approach of two-dimensional unequal error protection (2D UEP) for the scalable video with a combined temporal and quality (SNR) scalability over packet-erasure channel. The bit-stream is divided into scalable subbitstreams based on the structure of MCTF. Each subbitstream is further divided into several quality layers. Unequal quantities of bits are allocated to protect different layers to obtain acceptable quality video with smooth degradation under different transmission error conditions. Experimental results are presented to show the advantage of the proposed 2D UEP scheme over the traditional one-dimensional unequal error protection (1D UEP) scheme. Comparing the proposed method with the 1D UEP scheme on SNR layers, our method gives up to 0.81-dB improvement for some video sequences