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1.
本文提出了在丢包网络传输过程中,针对FGS分级编码视频流的优先级差异,提供非均匀FEC编码保护,同时考虑采用一致的信源信道联合统计率失真模型作为约束,根据一定的传输信道带宽和平均丢包率,联合优化信源编码码率和非均匀FEC编码码率的分配.该方法应用于丢包网络的视频传输,可以有效地适应网络传输条件的变化,提高信道带宽利用率,增强视频的传输质量.  相似文献   

2.
提出了一种用于无线实时流媒体传输的数据链路层自适应混合FEC/ARQ控制策略,以显著提高接收方的播放质量。该策略采用跨层设计的方法,基于Kalman滤波器预测当前的网络状态,自适应地调整FEC参数N和ARQ参数Nmax;另一方面,在应用层采用自适应FEC策略,在视频源数据和冗余数据之间动态分配网络带宽。数学分析和仿真验证均表明,该策略能使接收方获得最大的可播放帧率,有效地提高了流媒体传输的可靠性和实时性。  相似文献   

3.
针对网络电视业务的802.11无线局域网数据传输问题,提出一种视频流组播优化机制,基于视频帧区分的方法,对关键帧I帧采用可靠组播机制传输,对非关键的B帧和P帧采用组播机制传输。仿真结果表明,该机制可以在丢帧率和数据延迟之间取得较好的折中,尤其是在无线网络带宽不足的情况下,能够有效提高数据传输后视频的观看质量。  相似文献   

4.
基于内容感知的视频流媒体渐进式流传输方式   总被引:1,自引:0,他引:1       下载免费PDF全文
梁永生  陈旭  柳伟  张基宏 《计算机工程》2010,36(24):220-222
为有效利用传输网络带宽、平衡播出视频质量与用户等待时间之间的矛盾,提出一种基于内容感知的视频流媒体渐进式流传输方式。在视频内容分析和理解的基础上,根据编码关键参数确定视频序列中各帧的重要性,估计帧内片层数据重要性。采用MGS编码,在MobileIP测试平台上进行实验。实验结果表明了该传输方式的正确性和有效性,能够适应网络带宽和质量可伸缩的视频流媒体。  相似文献   

5.
基于GM(1,1)模型的自适应链路层ARQ控制策略   总被引:5,自引:0,他引:5  
靳勇  白光伟 《计算机应用》2008,28(9):2216-2219
提出了一种用于无线实时流媒体传输的自适应链路层ARQ控制策略,用以提高接收方的播放质量。该策略采用跨层设计的方法,基于GM(1,1)模型预测当前的网络状态,考虑GOP可解码帧数的特性,自适应地调整ARQ参数Nmax;另一方面,在应用层采用自适应FEC策略,在视频源数据和冗余数据之间动态分配网络带宽。数学分析和仿真验证均表明,该策略能使接收方获得最大的可播放帧率,有效地提高了流媒体传输的可靠性和实时性。  相似文献   

6.
何伟  王晖 《计算机仿真》2004,21(10):107-110
目前MPEG-4精细颗粒度可伸缩性(Fine Graruldty Scalability,FGS)编码视频正成为视频流服务的一种主要的业务流,因此,针对MPEG-4FGS视频流量进行建模对于网络性能仿真和通信网络设计具有十分重要的意义。该文首先介绍了MPEG-4FGS编码原理,然后对MPEG-4FGS视频流量的统计特性进行了分析,在此基础上,提出了基于MPEG-4FGS的视频流量模型。实验结果表明,该模型能较好地拟合原视频帧序列大小,且能根据网络带宽的动态变化进行适应的码率分配。  相似文献   

7.
《软件工程师》2016,(3):31-32
为了给终端用户提供更好质量的视频,本文提出了一种基于H264/AVC扩展的自适应多路径视频流可伸缩视频编码算法,通过视频流分发网络(VDN)提供路径的多样性,我们的方法最终适应于各种终端用户,并且通过在多路径上的可用带宽的变化来适应网络带宽的波动。实验表明该算法更能有效地估计网络拥塞状况、降低视频丢包率和减少网络延迟,从而更有效地保证视频网络传输质量。  相似文献   

8.
细粒度扩展视频均等质量流化算法   总被引:2,自引:0,他引:2  
梅峥  李锦涛  宁华 《软件学报》2006,17(12):2499-2507
细粒度扩展(fine-granularity-scalability,简称FGS)编码具有很强的灵活性和较好的视频流化性能,故已被MPEG-4和H.26L等标准所采用.FGS编码的一个突出特点是可以随意裁减以适应网络带宽的变化.但是,简单的裁减方法容易造成连续图像质量抖动过大,而用户通常希望流化视频的图像质量尽可能地平滑.在相关研究的基础上,针对非实时流化应用,分别讨论了在无损和有损条件下进行FGS编码均等质量流化的算法.该算法基于FGS的分段线性率失真模型和滑动窗口,在无损条件下,使用二分法,在当前窗口中的所有帧之间进行速率分配,以实现均等质量流化;在有损条件下,通过自适应的启发式算法,并结合前向纠错(forward error correction,简称FEC)技术来达到同样的目的.实验结果表明:在两种情况下,该算法均可以获得较好的流化效果,使流化视频的图像质量更加平滑.  相似文献   

9.
协同设计工具中视频数据的多方传输   总被引:3,自引:0,他引:3  
视频工具是协同设计环境下人-人交互的重要手段之一,为了支持群体工作,视频工具应该具有多方和多路视频的特征,这将使普通视频工具原本存在的网络带宽资源占用量大的问题表现的尤其突出,基于multicast传输,提出一种混合式的数据处理与传输方法,使系统中视频数据的网络带宽占有量明显下降,并且为视频源数据流引入了自适应控制机制,根据网络负载大小调整视频的充理,使视频流的质量和网络带宽分配达到折中,另外,采  相似文献   

10.
Kalman滤波的自适应链路层FEC控制策略   总被引:2,自引:0,他引:2       下载免费PDF全文
无线网络动态的信道特性、高误码率和带宽有限等特点,使得在无线环境下为实时流媒体传输提供QoS保证面临更大的挑战。提出一种用于无线实时流媒体传输的自适应链路层FEC控制策略,以显著提高接收方的播放质量。该策略采用跨层设计的方法,基于Kalman滤波器预测当前的网络状态,考虑物理带宽限制和GOP可解码帧数的特性自适应地调整FEC参数N;另一方面,在应用层采用自适应FEC策略,在视频源数据和冗余数据之间动态分配网络带宽。数学分析和仿真验证均表明,该策略能使接收方获得最大的可播放帧率,有效地提高了流媒体传输的可靠性和实时性。  相似文献   

11.
基于TCP友好速率控制和前向纠错的MPEG-2视频传输   总被引:2,自引:0,他引:2  
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。  相似文献   

12.
Wireless multimedia home servers are the next generation of home entertainment systems. The provision of high quality time-critical video streaming applications in indoor environments is very challenging due to high attenuation and interference caused by the walls and the contention in the shared wireless channel. This paper proposes a Delay-sensitive Multipath Forward Error Correction (DM-FEC) mechanism involving an estimation of available bandwidth and a mathematical analytical model with which the appropriate transmission rate, the FEC block length, and the FEC redundancy on each path in a multipath environment can be determined to solve previous problems. The DM-FEC mechanism not only selects the appropriate transmission rate and FEC redundancy on each path but also adaptively adjusts the FEC block length and concurrently sends data over multiple paths to overcome burst packet losses which otherwise can cause the video frame to be unplayable and impede the timely recovery of video information.  相似文献   

13.
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.  相似文献   

14.

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.

  相似文献   

15.
基于尽力而为的网络模式不能提供QoS保证,网络拥塞和分组丢失不可避免。在端到端视频单播结构下,论文提出了一个发送端速率控制框架SRCF,在此框架下首先利用RTCP报文中的字段提出了一种网络参数测量方法,然后设计了一个自适应速率算法SRCA,SRCA利用已得到的网络传输延迟和分组丢失率参数作为初始参数,来调整编码速率,达到充分利用带宽的目的,避免了视频质量由于调整参数带来的剧烈抖动。仿真结果表明,该算法在网络出现一定拥塞的条件下,能跟踪带宽的变化,网络和媒体QoS能保证视频质量较好。  相似文献   

16.
If the frame size of a multimedia encoder is small, Internet Protocol (IP) streaming applications need to pack many encoded media frames in each Real-time Transport Protocol (RTP) packet to avoid unnecessary header overhead. The generic forward error correction (FEC) mechanisms proposed in the literature for RTP transmission do not perform optimally in terms of stability when the RTP payload consists of several individual data elements of equal priority. In this paper, we present a novel approach for generating FEC packets optimized for applications packing multiple individually decodable media frames in each RTP payload. In the proposed method, a set of frames and its corresponding FEC data are spread among multiple packets so that the experienced frame loss rate does not vary greatly under different packet loss patterns. We verify the performance improvement gained against traditional generic FEC by analyzing and comparing the variance of the residual frame loss rate in the proposed packetization scheme and in the baseline generic FEC.  相似文献   

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