基于Kalman滤波的无线流媒体自适应混合FEC/ ARQ控制策略

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。     

无线流媒体自适应链路层HARQ控制策略

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;提出了一种用于无线实时流媒体传输的自适应链路层HARQ控制策略,针对不同的信道状况动态选择混合ARQ方案;该策略采用跨层设计的方法,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,有效地提高流媒体传输的可靠性和实时性。     

一种无线实时流媒体增强型自适应FEC控制机制

无线网络动态的信道特性和带宽有限等特点,使得在无线环境下为流媒体应用提供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增强型自适应前向纠错控制策略,以提高接收方的播放质量。该策略采用跨层设计的方法,根据当前的网络状态,自适应地调整MPEG视频帧的发送速率,在视频源数据和冗余数据之间动态分配网络带宽。仿真结果表明,该策略能使接收方获得最大的可播放帧率,有效提高流媒体传输的可靠性和实时性。     

Kalman滤波的自适应链路层FEC控制策略

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。     

无线实时流媒体传输性能的跨层优化设计

提出一种用于无线实时流媒体传输的优化设计策略,以提高接收方的播放质量。该策略采用跨层设计的方法,利用泊松过程分析链路层数据帧的丢失,同时把链路层最大重传次数映射到端到端时延和丢包率的计算中,自适应地调整MPEG视频帧的发送速率,在视频源数据和冗余数据之间动态分配网络带宽。仿真实验结果表明,该策略能使接收方获得最大的可播放帧率,有效提高流媒体传输的可靠性和实时性。     

基于自适应FEC的区分服务流媒体传输模型

实时流媒体在目前的因特网中传输经常会因为网络带宽的不足或数据包丢失严重,使得接收方播放质量受到严重影响.基于跨层设计的思想,在应用层使用自适应前向纠错算法,即使流媒体数据有一定的丢包率,接收方仍然能完整地恢复出原视频序列.在数据链路层采用有利于流媒体传输的区分服务模型,用以增大高优先级数据流(如实时流媒体数据流)传榆过程的吞吐量.数学分析和仿真结果均表明,该策略能使接收方获得最大的可播放帧率,从而有效提高流媒体传输的可靠性和实时性.     

基于节点状态模型的传输模式自适应选择算法

针对现有的流媒体传输方法不能在高效分发数据的同时保证数据接收质量的问题, 提出了一种传输模式自适应选择算法。该算法首先根据节点状态模型, 将节点状态分为数据接收质量和综合状态值两部分, 对各节点本身稳定状态和节点重要性进行综合判定, 再结合现有数据缓冲模型进行传输模式自适应选择。仿真结果表明, 该算法有效地提高了数据在传输过程中的分发效率, 同时通过对服务器负载一定程度的控制, 在保证数据接收质量的同时, 提高了系统的可扩展性。     

基于区分服务的流媒体传输模型

流媒体在网络上的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,使得接收方播放质量受到严重影响。本文建立了一种适合流媒体传输的区分服务模型,该模型能够使高优先级数据流（实时流媒体数据流）占用更多的带宽。仿真结果表明：该策略能使流媒体数据流获得较高的吞吐量和较低的丢包率,有效提高流媒体传输的可靠性和实时性。     

基于区分服务的流媒体传输模型

流媒体在网络上的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,使得接收方播放质量受到严重影响。本文建立了一种适合流媒体传输的区分服务模型,该模型能够使高优先级数据流(实时流媒体数据流)占用更多的带宽。仿真结果表明:该策略能使流媒体数据流获得较高的吞吐量和较低的丢包率,有效提高流媒体传输的可靠性和实时性。     

一种基于Internet的短波接入数据链路传输协议的改进方法

讨论了数据链路数据包长度与传输出错率的关系以及数据包合理长度的范围,进而对无帧序号自适应选择式ARQ传输方式的改善进行了分析,提出采用无帧序号自适应选择式ARQ传输方式节省时间的定量分析方法,得出了节省时间的仿真测试结果。采用改进的数据传送方式可以有效减少链路上的传送和应答开销,提高了短波系统的传输效率。     

一种MPEG1/MPEG2视频流的自适应播放算法

提出了一种MPEG1/MPEG2视频流的自适应播放算法。算法根据播放缓冲区的占用情况来调整视频帧的播放持续时间，并在播放缓冲区上溢时判断到达视频帧的类型，以决定是暂时存储还是丢弃，使得不会造成帧内编码帧和前向预测编码帧的丢失，从而保证视频流平滑地播放。实验结果表明，在播放不连续性和播放失真上新算法都优于Yuang算法，并实现了视频流的平滑播放。     

基于GOP取帧和跳数的自适应协作ARQ

为给多媒体数据传输提供有效QoS保障,提出一种适用于无线多媒体传感器网络数据传输的自适应协作ARQ机制。采用跨层设计的方法,分析ARQ机制协作传输跳数与能效的规律,根据节点间通信距离自适应选取最优协作传输跳数,结合数据帧跳数,自适应调整最大重传次数,根据快进快退码率基于帧间依赖关系进行GOP取帧。分析结果表明,该机制在丢包率和端到端平均时延方面与传统ARQ相比具有较优的性能,能有效提高无线多媒体传感器网络通信的可靠性和实时性。     

无线流媒体主动弃帧策略的仿真研究

仿真研究IEEE802.11g无线网络环境下实时流媒体的性能,在分析和探讨支持实时流媒体应用时无线网络性能瓶颈的基础上,提出一种改进策略--主动弃帧.仿真结果表明,这一策略显著改善了网络性能,为实时流媒体在WLAN上的应用提供更好的服务质量.     

An efficient delay-constrained ARQ scheme for MMT packet-based real-time video streaming over IP networks

The MPEG has recently Querydeveloped a new standard, MPEG media transport (MMT), for the next-generation hybrid media delivery service over IP networks considering the emerging convergence of digital broadcast and broadband services. On account of the heterogeneous characteristics of broadcast and broadband networks, MMT provides an efficient delivery timing model to enable inter-network synchronization, measure various kinds of transmission delays and jitters caused by the transmission delay, and re-adjust the timing relationship between the MMT packets to ensure synchronized playback. By exploiting the delivery timing model, it is possible to accurately estimate the round-trip time (RTT) experienced during MMT packet transmission. Based on the measured RTT, we propose an efficient delay-constrained automatic repeat request (ARQ) scheme, which is applicable to MMT packet-based real-time video streaming service over IP networks. In the proposed ARQ scheme, the receiver buffer fullness at the time of packet loss detection is used to compute the arrival deadline, which is the maximum allowed time for completing the requesting and retransmitting of the lost MMT packet. Simulation results demonstrate that the proposed delay-constrained ARQ scheme can not only provide reliable error recovery, but it also achieves significant bandwidth savings by reducing the number of wastefully retransmitted packets that arrive at the receiver side and exceed the allowed arrival deadline.     

Effective client-driven three-level rate adaptation (TLRA) approach for adaptive HTTP streaming

Multimedia streaming allows consumers to view multimedia content anywhere. However, quality of service is a major concern amid heightened levels of network traffic caused by increasing user demand. Accordingly, media streaming technology is adopting a new paradigm: adaptive HTTP streaming (AHS). AHS is widely used for real-time streaming content delivery in the Internet environment. In streaming, selection of appropriate bitrate is crucial for adapting media rate to network variations and client processing capabilities while ensuring optimal service for the consumer. We evaluate a proposed client-driven three-level optimized rate adaptation algorithm for adaptive HTTP media streaming. In the first stage, the algorithm chooses a suitable starting bitrate close to the available channel capacity. Next, the algorithm monitors the client parameters in real time, precisely detecting network variations and choosing a likely available bit representation for the next download segment. The algorithm controls and minimizes the effects of buffer stalls and overflow resulting from the brief network variations occurring between consecutive segments. The proposed algorithm is implemented in Dynamic Adaptive Streaming over HTTP (DASH) player and its performance compared to that of commercially available Gstreamer-based HTTP Live Streaming (HLS) and DASH players which use conventional segment fetch time–based adaptation and throughput-based adaptation algorithms respectively. This evaluation uses a real-time cloud server client and test bed streaming setup. The resulting analysis shows that the client-driven three-level rate adaptation (TLRA) approach allows adaptive streaming clients to maximize use of end-to-end network capacity, delivering an ideal user experience by precisely predicting network variations and rapidly adapting to available bandwidth, minimizing rebuffering events and bitrate level changes.     

Design and Implementation of a Streaming System for MPEG-1 Videos

A new streaming application has been developed for the Internet environment. The system has client-server structure together with multithreaded architecture and pipelining. RTP protocol is used to transmit packets belonging to MPEG videos. RTCP protocol collects transmission statistics. System is adaptive in the sense that it reacts to dynamic network conditions. A feedback mechanism controls both the frame interval and frame rate depending on the frame-loss statistics and buffered video level at the client. A flow control module at the client side controls buffer underflows and overflows. Performance results of the implementation are reported and discussed. The performance of the proposed buffering strategy is compared with other proposed methods from the literature. The comparisons showed that the proposed strategy is more robust than other methods.     

XOR-based frame loss recovery scheme for video streaming

High coding dependencies among video frames suffer from vulnerability to packet loss, which impacts the playback quality of video streaming. In this paper, according to the characteristics of MPEG4/H.264 encoding methods, we propose a simple and low-complexity XOR-based FEC frame loss recovery scheme. Within an entire Group of Pictures (GOP), the proposed scheme shows the ability to recover simultaneously I-frame loss and one P-frame loss. The high frame loss resilience improves the playback QoS of compressed video streaming. The mathematical analysis reveals that the proposed scheme has 72.7% performance improvement than no frame loss protection in term of full GOP frames successful decoding rate.