实现基于RTCP协议的参考图象法的探讨

目前视频图象在 IP分组网中的传输正被日益广泛地应用 (如 :视频电话、视频会议 ) .但是 ,由于 IP分组网固有的特点 ,致使 IP分组包丢失的现象不可避免 ,并极大地影响了视频传输的服务质量 .由于参考图象选择法RPS(Reference Picture Selection)是一种较好的视频图象差错复原的方法 ,因此 ,为了改善因分组包丢失而对视频传输带来的质量下降问题 ,利用 RTCP协议提供反馈信息和利用参考图象选择法进行视频图象复原 ,实现了一种在 IP分组网中进行视频传输的差错控制机制 .实践证明 ,通过这种方法不仅可减小因分组包丢失而对视频图象信息带来的损害 ,而且可提高视频图象的传输质量 ,此方法适用于基于分组网的视频传输系统 .     

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。     

基于GA-BP的实时视频通信自适应前向纠错码

针对在动态的网络环境中,实时视频在基于数据报协议的传输中遇到大量丢包的问题,提出了一种基于遗传算法改进的BP神经网络自适应前向纠错码的方法。该方法通过设计自适应的前向纠错编码来恢复丢失的数据包。首先,设计一个基于帧级别的前向纠错的框架,主要包括视频编码解码器、RS前向纠错码编解码模块和前向纠错码冗余度计算模块,用来模拟一般视频传输的环境;然后,设计GA-BP模型,将其应用到RS-FEC的冗余度计算,实现一种帧级不均等的保护方案;最后,基于GE信道模拟网络,生成丢包数据,训练离线模型,进行实验验证。实验结果表明,相比StaticRS和DeepRS,所提方法能够在较低冗余度下较高地恢复丢失的数据包,得到更高质量的视频。     

一种增强的因特网语音工具传输控制机制

包丢失引起的音质不高是目前语音工具的关键问题,提出的一种同时具有速率调节和丢失恢复功能的因特网语音工具传输控制机制,在速率调节过程中增加了前向纠错功能,能够解决最容易出现的单个包的丢失问题,如果带宽许可,也能够恢复多个丢失的语音包。与现有的传输控制机制相比,最主要的特点是不因为具有前向纠错过程而增加时延,且易于实现。     

视频传输中的差错控制技术综述

文章对视频传输中的差错控制算法进行了分析研究,重点讨论了前向纠错编码、自动重传、错误弹性编码、交叉打包和错误隐藏等差错控制技术的差错控制原理、研究进展情况及优缺点,并指出对于实时性要求很强的视频数据通信来说,前向纠错技术是一种非常适用的保证传输服务质量的方法。     

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。     

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.     

视频传输中的分组调度方法

数据报丢失控制是在因特网中传输视频时的基本问题,缓冲区管理是防止视频数据报丢失的传统网络技术.为了解决视频通信中数据报丢失的问题,提出了一种新的分组调度方法,并建立视频通信系统环境,进行了编程实验.对运行在路由器内,能够保证视频传过程中视频流内部和视频流之间数据报丢失要求的分组调度方法进行了仿真和分析.结果表明了该分组调度方法在控制数据报丢失和效率方面的优点.     

自适应前向纠错增强TCP在无线链路上性能研究

针对无线链路引发的TCP性能缺陷,提出一种适用于TCP端对端前向纠错的自适应算法,并在此基础上利用Reed Solomon码实现一种新的TCP ARS。TCP ARS利用包一级的前向纠错在传输层恢复丢失的分组,并能自适应地根据网络丢包率、往返时延等参数调整前向纠错的冗余信息度,较好地解决了TCP在无线环境中的性能缺陷问题。     

基于IP无线网络FGS视频传输的多乘积码方案

研究基于IP无线网络中精细粒度可伸缩性(FGS)视频的传输。基于包交换的IP无线网络通常由两段链路组成:有线链路和无线链路。为了处理这种混合网络中不同类型数据包的丢失情况,对FGS视频增强层数据运用了一个具有比特平面间不平等差错保护(BPUEP)的多乘积码前向纠错(MPFEC)方案进行信道编码。对FGS增强层每一个比特平面(BP),在传输层,采用里德—索罗蒙码(RS)提供比特平面间的保护;而在链路层,则运用循环冗余校验码(CRC)串联率兼容穿孔卷积码(RCPC)提供数据包内保护。还提出了一个率失真优化的信源—信道联合编码的码率配置方案,仿真结果显示出该方案在提高接收端视频质量方面的优势。     

Packet loss resilience using unequal forward error correction assignment for video transmission over communication networks

Forward error correction (FEC) methods have been developed for packet loss resilience in application layer for real-time video transmission over communication networks. In this paper, an efficient packet loss resilience method is proposed using closed form solution for unequal FEC assignment based on a new packet distortion model. We first derive the packet distortion model by investigating the error concealment property and error propagation effect in H.264. To select the source and channel rate minimizing the overall distortion, we present a model-based rate allocation algorithm using the packet distortion model and rate-distortion function. Then we propose the closed form solution for unequal FEC assignment, which uses the packet distortion model and considers channel status information. Simulation results show that the proposed method gives substantial improvement for the received video quality in packet-lossy Internet and wireless network environments, while it requires much less computational complexity compared to the previous scheme.     

Packet loss probability for DiffServ over IP and MPLS reliable homogeneous multicast networks

Multicasting has become increasingly important with the emergence of Internet-based applications such as IP telephony, audio/video conferencing, distributed databases and software upgrading. IP multicasting is an efficient way to distribute information from a single source to multiple destinations at different locations. In practice IP is considered as a layer 3 protocol. Multiprotocol Label Switching (MPLS) replaces the IP forwarding by a simple label lookup. MPLS combines the flexibility of layer 3 routing and layer 2 switching.In order to provide QoS in group communications for real time applications such as video conferencing, reliable multicasting is used. Miscellaneous efforts have been undertaken to provide reliability on top of IP multicast. Two error control strategies have been popular in practice. These are the FEC (Forward Error Correction) strategy, which uses error correction alone, and the ARQ (Automatic Repeat Request) strategy, which uses error detection, combined with retransmission of data.In this paper, we present a new fair share policy (FSP) that utilizes Differentiated Services to solve the problems of QoS and congestion control when reliable ARQ multicast is used. The results should provide insight into the comparisons of the residual packet loss probability between IP multicast in MPLS networks using FSP and plain IP multicasting using the same policy when DiffServ are adopted and when reliable ARQ multicast is considered.     

Header Detection to Improve Multimedia Quality Over Wireless Networks

Wireless multimedia studies have revealed that forward error correction (FEC) on corrupted packets yields better bandwidth utilization and lower delay than retransmissions. To facilitate FEC-based recovery, corrupted packets should not be dropped so that maximum number of packets is relayed to a wireless receiver's FEC decoder. Previous studies proposed to mitigate wireless packet drops by a partial checksum that ignored payload errors. Such schemes require modifications to both transmitters and receivers, and incur packet-losses due to header errors. In this paper, we introduce a receiver-based scheme which uses the history of active multimedia sessions to detect transmitted values of corrupted packet headers, thereby improving wireless multimedia throughput. Header detection is posed as the decision-theoretic problem of multihypothesis detection of known parameters in noise. Performance of the proposed scheme is evaluated using trace-driven video simulations on an 802.11b local area network. We show that header detection with application layer FEC provides significant throughput and video quality improvements over the conventional UDP/IP/802.11 protocol stack     

Unequal error protection under bitrate constraint for video streaming over internet

This article describes a simple packet-level FEC system suitable for unequal error protection of layered video streams, that we called TAPIOCA (in French, Transport Audiovisuel avec Protection Inégale des Objets et Contrôle d’Admission). It is designed in a way that the FEC overhead induced by redundant packets is perfectly controlled by the sender. In order to achieve that, TAPIOCA calculates on-the-fly the optimal erasure code to be used, video data unit by video data unit, under a given bitrate constraint. In addition, and contrary to the well-known PET (Priority Encoding Transmission) system, the video data units of each layer are encoded separately. This is especially useful when all layers are not output from the video coder at the same time. Simulation results for MPEG-4 video streaming show that the proposed FEC system can be very efficient even if packet losses are due to network congestion. Moreover, comparison with PET system shows that TAPIOCA exhibits better performance, considering criteria including the decodable frame rate, protection system efficiency and computational cost.     

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.     

An Adaptive Motion-Based Unequal Error Protection Approach for Real-Time Video Transport Over Wireless IP Networks

In this work, we consider the delivery of digital video over future 3G wireless IP networks and we propose a low-complexity adaptive motion-based unequal error protection (UEP) video coding and transmission system which efficiently combines three existing error-resilience techniques by exploiting knowledge of the source material as well as the channel operating conditions. Given this information, the proposed system can adaptively adjust the operating parameters of the video source encoder and the forward error correction (FEC) channel encoder to maximize the delivered video quality based upon both application-layer video motion estimates and link-layer channel estimates. We demonstrate the efficacy of this approach using the ITU-T H.264 video source coder. The results indicate that a significant performance improvement can be achieved with enhanced resilience to inaccurate channel feedback information and with substantially reduced computational complexity compared to competing approaches.     

An Integrated Framework for Efficient Transport of Real-Time MPEG Video over ATM Best Effort Service

This paper describes a framework for the transport of real-time multimedia traffic generated by MPEG-2 applications over ATM networks using an enhanced UBR best effort service (UBR+). Based on the factors affecting the picture quality during transmission, we propose an efficient and cost-effective ATM best effort delivery service. The proposed framework integrates three major components: 1) a dynamic frame-level priority assignment mechanism based on MPEG data structure and feedback from the network (DexPAS), 2) a novel audio-visual AAL5 SSCS with FEC (AV-SSCS), and 3) an intelligent packet video discard scheme named FEC-PSD, which adaptively and selectively adjusts cell drop levels to switch buffer occupancy, video cell payload type and forward error correction capability of the destination. The overall best-effort video delivery framework is evaluated using ATM network simulation and MPEG2 video traces. The ultimate goal of this framework is twofold: First, minimizing loss of critical video data with bounded end-to-end delay for arriving cells and second, reducing the bad throughput crossing the network during congestion. Compared to previous approaches, performance evaluation shows a good protection of Predictive coded (P-) and Bi-directional Predictive coded (B-) frames at the MPEG video slice layer.     

实时视频传输的反馈拥塞控制策略

提出了RTP/UDP/IP的视频实时传输体系结构。采用基于实时传输协议（RTP）和实时传输控制协议（RTCP）的反馈拥塞控制算法，控制发送端视频编码器输出的比特率，以使得整个基于IP的网络视频传输系统既能充分利用网络带宽又不引起网络拥塞，从而减少时延，降低丢包率，为视频数据的接收者提供一个较好的图象质量。     

IP网络丢包情况下音视频实时通讯研究

分析了IP网络中音视频实时通讯抗网络丢包的技术现状,并探讨了采用前向纠错方法实现抗丢包的实现机制。对于开发各种网络状况自适应的视频会议系统具有一定参考价值。