MIMO OFDM系统中基于自适应编码调制的跨层设计方案

本文讨论了MIMOOFDM系统中一种基于自适应编码调制的的跨层传输技术。该技术结合传统物理层自适应编码调制和数据链路层自动请求重发,利用信道估计参数和数据链路层的误包率计算自适应编码调制门限,在系统给定时延和误包率约束的基础上最大程度地提高频谱利用率。仿真结果表明,编码系统具有很强的差错控制作用,相比传统物理层自适应编码调制系统和未编码系统都提高了频率利用率;当频率利用率为1时,相比于未编码系统,编码带来的增益为4dB。同时这种强差错控制作用也使得改变系统最大重传次数对切换门限的影响变小,因此跨层自适应编码调制相比传统物理层自适应带来的频谱效率的提高变小,这就使得实际系统能以较小的时延代价换取足够的频谱利用率增益。     

协作通信系统中基于频谱效率优化的跨层设计

为改善协作分集系统的频谱效率,提出一种改进的跨层设计方案,对协作通信系统物理层的自适应调制编码（AMC）技术和链路层的混合自动重传(HARQ)协议进行联合优化设计。该方案利用少量比特的反馈,使得仅当目的节点通过直接信道不能正确译码分组时或者当直接信道处于深度衰落时触发中继节点转发协作伙伴数据。给出了所提方案基于瑞利衰落信道、解码转发（DF）条件下频谱效率的表达式,然后搜索在任意信噪比条件下使频谱效率最大化的调制与编码方案。通过计算机仿真对理论分析进行了验证。理论分析和仿真表明该跨层设计在中低信噪比(SNR)可进一步提升协作通信系统的频谱效率。      

一种基于网络编码的重传算法研究

无线网络中,节点发送的数据分组传输失败后,执行重传机制。传统的重传机制ARQ由于在一次重传中只能发送一个丢失的数据分组,因此传输效率比较低。利用网络编码技术和AQR重传机制,我们可以在重传中使用网络编码,广播发送由多个丢失数据分组编码得到的编码分组,从而提高重传效率。本文中我们提出一种将网络编码应用于多个发送方多个接收方(MSMR)无线网络中的算法RMBNC。理论推导和仿真分析验证了我们提出的算法的有效性。     

基于网络编码的P2P流媒体传输技术

提出了一种基于网络编码的P2P流媒体传输方案,将分层编码后的数据包在基本层和增强层分别进行网络编码后传输,接收节点可根据带宽情况控制发送节点是否发送增强层数据,提高了播放质量和稳定性.仿真试验表明,基于网络编码的传输方案,接收节点的播放质量能够得到提高,当网络中的节点间带宽剧烈变化的情况下,对比提高尤为明显.     

物理层网络编码的调制技术

网络编码理论提高了数据的传输效率,而物理层网络编码进一步缩短了数据交互的时隙,它认为:只要中继节点R采取了恰当的调制解调技术,可以使电磁信号的叠加映射到数字域上的运算操作,实现两时隙交互信息。可见选择恰当的调制解调技术对实现物理层网络编码至关重要。该文从调制技术这一角度出发,综合目前在物理层网络编码中所应用的调制解调技术,对其进行分析总结,并展望了下一步物理层网络编码中调制技术的发展。     

LTE下行链路中基于网络编码的多进程联合HARQ

长期演进系统的下行链路物理层采用了以传输块为基本单元的混合自动重传请求机制,保证移动终端在各种测试场景下都能获得较好的数据通过率。在对传输块的每次接收处理过程中,单个传输块中的多路独立码块数据解码结果差异较大。接收端可以通过对各个码块数据做循环冗余校验测试,标志各个码块的译码结果。然而由于传统重传机制中反馈链路的容量限制,即便只有一个码块解码错误,接收端也将反馈请求发送端重传整个传输块。网络编码技术通过对不同传输块数据做组合编码,可以有效地提高整个网络的传输效率。针对长期演进系统下行链路的传输机制,我们提出一种基于网络编码的多进程联合重传方案。新的重传方案联合多个传输块的重传进程,在每个传输块完成首次传输后,新增一个网络编码重传反馈比特,指示后续是否启动一个新进程传输多个传输块的网络编码数据包。我们给出了具体的重传协议流程,收发端都可以高效地根据网络编码重传反馈比特指示,完成相应的发送和接收操作。结合长期演进系统标准中的典型测试场景,我们给出了数据通过率的仿真结果比较。在仅新增一个反馈比特的条件下,采用网络编码技术的多进程联合重传方案在数据通过率方面有较为明显的性能增益。      

基于D2D通信系统中的一种跨层中继选择算法

在D2D通信系统与蜂窝网络共存的场景下,引入中继节点可有效提高D2D链路的吞吐量和D2D用户对蜂窝用户的干扰。文中基于译码转发模式,结合跨层协作通信的思想,提出了一种基于物理层和数据链路层的跨层中继选择算法。该算法结合物理层的信道状态信息和数据链路层的队列状态信息,两个参数进行最优中继节点的选择。并最终通过仿真验证表明,基于跨层中继选择算法可提高通信系统的吞吐量,同时降低了通信系统的平均时延和数据包错误率。     

基于Syntax级分组和多线程处理的HEVC熵编码并行算法

新一代视频编码标准HEVC获得了较高的编码效率,但是同时需要较大的计算量。 HEVC并行算法能够提高编码速度,如何开发适用于多核处理器的并行编码算法对于满足高清视频实时传输和大规模共享具有十分重要的意义。提出了一种基于Syntax级分组和多线程处理的HEVC熵编码并行算法。该算法首先将HEVC中一个编码树单元的编码信息按照语法元素进行分组;其次,根据编码块数据间的相关性构建Syntax级并行编码器;然后结合多线程技术实现HEVC帧级编码的并行计算。实验结果表明,在编码图像的主客观质量上没有太大损失的情况下,该并行算法框架与传统的串行算法框架相比具有65%~70%的加速效果。     

MIMO OFDM系统中基于物理层与链路层的跨层优化设计

该文讨论了基于空时编码发射分集正交频分复用(OFDM)系统的跨层传输技术.该技术结合了传统物理层自适应调制和链路层自动请求重发,利用链路层的误包率和信道估计参数计算自适应调制门限,在系统给定时延和误包率约束的基础上最大程度地提高了频谱利用率.仿真结果表明,该算法相比传统物理层自适应传输,在频谱利用率性能上有1.5dB以上的增益.但随着最大重传次数的增加频谱利用率的提高越来越小,这就使得实际系统能以较小的时延代价换取足够的频谱利用率增益.     

物理层网络编码分组的机会中继

为提升物理层网络编码方案的抗衰落性能,该文提出了一种基于物理层网络编码的机会中继方案(Opportunistic Relaying based-on Physical-layer Network Coding,PNC-OR),该方案利用物理层网络编码的基本思想、有效提升网络吞吐的同时,通过中继节点的分布式选择,也能够使系统获得多用户分集增益,提高了系统的抗衰落性能。针对双向无线中继信道中端到端信息交换的情形,推导了准静态衰落环境下PNC-OR中多个目的节点接收信息的和容量。数值结果显示：和机会中继、传统网络编码两种方案相比,PNC-OR具有更高的频谱效率,并且随着中继节点的增多,频谱效率也越高。     

中继协作多播网络中基于最小集合覆盖的编码重传方案

为提高单中继协作多播传输效率,本文提出一种基于最小集合覆盖的分类网络编码重传方案.该方案充分利用中继节点协作传输的优势,将接收端的丢包按中继节点的接收状态分为两类,并按类先后进行编码重传.在各类丢包内部,根据对应的状态反馈矩阵寻找编码机会生成编码包,并将编码包的选择过程抽象为集合覆盖问题,通过求最小集合覆盖使重传次数逼近最小值.此外,在不增加所得前类编码包数的前提下,利用两类丢包之间的编码机会进一步生成新编码包,以减少重传编码包数,从而提高重传效率.分析与仿真结果表明了该方案的有效性.     

深空通信中基于喷泉码的前向纠删方法

基于喷泉码的前向纠删编码方法能够解决深空信道长时延、高误码率及链路易中断等特殊环境下的可靠数据传输问题。该方法同时采用系统Raptor码和Turbo码作为应用层与物理层编码方案,利用喷泉码的无率和一致特性克服了ARQ机制在深空通信应用中的瓶颈。仿真结果表明：该方法可以在无需反馈信道的情况下为系统提供误帧率保证（10-4～10-5）,同时,相比于单纯使用Turbo码的方案,该方法在一定程度上提高了系统频带利用率。     

中继协作无线网络中不完美反馈下基于网络编码的重传方案

无线广播网络传输过程中,目的节点反馈信息丢失或部分丢失导致发送节点不能了解目的节点的真实接收状态.为提高不完美反馈下无线网络的重传效率,本文提出中继协作无线网络中不完美反馈下基于网络编码的重传方案.本方案基于部分可观察马尔科夫决策过程对不完美反馈下的重传过程进行建模.发送节点根据系统观测状态和最大置信度更新系统估计状态,根据数据包发送顺序,优先选择最早丢失且能够恢复最多丢包的编码包重传.目的节点缓存不可解编码包以提升编解码机会.重传过程中源节点关注目的节点请求包需求,相同情况优先选择传输可靠性较高的中继节点,以提升传输有效性.仿真结果表明,在不完美反馈下相对于传统方案,本方案可有效提高重传效率.     

Huffman code based error screening and channel code optimization for error concealment in perceptual audio coding (PAC) algorithms

The class of perceptual audio coding (PAC) algorithms yields efficient and high-quality stereo digital audio bitstreams at bit rates from 16 kb/sec to 128 kb/sec (and higher). To avoid "pops and clicks" in the decoded audio signals, channel error detection combined with source error concealment, or source error mitigation, techniques are preferred to pure channel error correction. One method of channel error detection is to use a high-rate block code, for example, a cyclic redundancy check (CRC) code. Several joint source-channel coding issues arise in this framework because PAC contains a fixed-to-variable source coding component in the form of Huffman codes, so that the output audio packets are of varying length. We explore two such issues. First, we develop methods for screening for undetected channel errors in the audio decoder by looking for inconsistencies between the number of bits decoded by the Huffman decoder and the number of bits in the packet as specified by control information in the bitstream. We evaluate this scheme by means of simulations of Bernoulli sources and real audio data encoded by PAC. Considerable reduction in undetected errors is obtained. Second, we consider several configurations for the channel error detection codes, in particular CRC codes. The preferred set of formats employs variable-block length, variable-rate outer codes matched to the individual audio packets, with one or more codewords used per audio packet. To maintain a constant bit rate into the channel, PAC and CRC encoding must be performed jointly, e.g., by incorporating the CRC into the bit allocation loop in the audio coder.     

卫星光网络的虚拟信道和主信道双重优化数据封装方法

针对卫星 光网络中不同类型、不同速率业务统一封装实现难度大和传输效率低的问题,提出利用虚拟 信道(VC) 与主信道(MC)联合优化的数据封装方法,分别给出VC封装与MC处理方法,并针 对其原理进 行分析。在VC中,提出了基于优先级的虚拟信道调度算法,对帧调度的时间冲突进 行优化,提 高虚拟信道处理的帧复用效率。在MC处理中,对传输帧支持的业务类型进行了扩展, 分析了前向 纠错编码对帧传输的性能影响。研究结果表明,采用优先级算法后帧传输效率提升 24.61%;外码为RS 码、内码为卷积码的级联码是MC处理中可选择的最佳编码方式。     

Downlink media streaming with wireless fountain coding in wireline‐cum‐WiFi networks

This paper addresses the problem of streaming packetized media data in a combined wireline/802.11 network. Since the wireless channel is normally the bottleneck for media streaming in such a network, we propose that wireless fountain coding (WFC) be used over the wireless downlink in order to efficiently utilize the wireless bandwidth and exploit the broadcast nature of the channel. Forward error correction (FEC) is also used to combat errors at the application‐layer. We analytically obtain the moment generating function (MGF) for the wireless link‐layer delay incurred by WFC. With the MGF, the expected value of this wireless link‐layer delay is found and used by the access point (AP), who has no knowledge of the buffer contents of wireless receivers, to make a coding‐based decision. We then derive the end‐to‐end packet loss/late probability based on the MGF. We develop an integrated ns‐3/EvalVid simulator to evaluate our proposed system and compare it with the traditional 802.11e scheme which is without WFC capability but equipped with application‐ and link‐layer retransmission mechanisms. Through extensive simulations of video streaming, we show that streaming with WFC is able to support more concurrent video flows compared to the traditional scheme. When the deadlines imposed on video packets are relatively stringent, streaming with WFC also shows superior performance in terms of packet loss/late probability, video distortion, and video frame delay, over the traditional scheme. Copyright © 2011 John Wiley & Sons, Ltd.     

The Design of a High-Performance Error-Correcting Coding Scheme for the Canadian Broadcast Telidon System Based on Reed-Solomon Codes

Error correction can greatly improve the performance and extend the range of broadcast teletext systems. In this paper, the requirements for an error-correcting scheme for broadcast teletext in North America (NABTS) are set down. An error-correction scheme which meets all these requirements is then described. The simplest case employs the one parity bit in each 8 bit byte and no suffix of parity check bits at the end of each data block. The next level also uses a single byte of parity check bits at the end of each data block. Adding a second byte of parity checks at the end of each data block results in a Reed-Solomon code, called theCcode, for each data block. Adding one data block of parity checks afterh - 1data blocks results in a set ofhdata packets being encoded into a bundle, in which verticalCcodes provide powerful interleaving. In a final alternative, two data blocks hold the check bytes for the vertical codewords, and the most powerful coding scheme, the double bundle code, results. The detailed mathematical definitions of the various codes are referred to or described, formulas for performance calculations are referred to, and performance curves are presented for the AWGN channel as well as for measured field data. These performance curves are discussed and compared to the performance of a difference set cyclic code, originally designed for the Japanese teletext system, which corrects any 8 bits in error in a packet.     

Interleaved coding and sequential transmission scheme for packet‐level forward error correction over burst loss channels

Burst packet loss is a common problem over wired and wireless networks and leads to a significant reduction in the performance of packet‐level forward error correction (FEC) schemes used to recover packet losses during transmission. Traditional FEC interleaving methods adopt the sequential coding‐interleaved transmission (SCIT) process to encode the FEC packets sequentially and reorder the packet transmission sequence. Consequently, the burst loss effect can be mitigated at the expense of an increased end‐to‐end delay. Alternatively, the reversed interleaving scheme, namely, interleaved coding‐sequential transmission (ICST), performs FEC coding in an interleaved manner and transmits the packets sequentially based on their generation order in the application. In this study, the analytical FEC model is constructed to evaluate the performance of the SCIT and ICST schemes. From the analysis results, it can be observed that the interleaving delay of ICST FEC is reduced by transmitting the source packets immediately as they arrive from the application. Accordingly, an Enhanced ICST scheme is further proposed to trade the saved interleaving time for a greater interleaving capacity, and the corresponding packet loss rate can be minimized under a given delay constraint. The simulation results show that the Enhanced ICST scheme achieves a lower packet loss rate and a higher peak signal‐to‐noise‐ratio than the traditional SCIT and ICST schemes for video streaming applications.     

基于LT码的LBCMP迭代纠错方法

针对删除信道中发生错误的数据包,提出联合信道编码的LBCMP迭代纠错方法,该方法充分利用错误数据包中含有的正确信息,将LT编码包作为冗余纠错包与线性分组码相结合,并采用MP迭代译码方法进行纠错.理论分析及实验结果表明,采用LBCMP迭代方法可以减少为恢复错误数据包所需要的信源编码包数量.     

Packet level acknowledgement and Go‐Back‐N protocol performance in infrared wireless LANs

Infrared wireless LANs may employ repetition rate (RR) coding to increase the symbol capture probability at the receiver. This paper examines the effectiveness of RR coding to utilization for infrared LANs using the physical and link layer parameter values proposed in the Advanced Infrared (AIr) protocol standard, which is developed by the Infrared Data Association (IrDA). Infrared LANs employ a Go‐Back‐N (GBN) automatic repeat request (ARQ) retransmission scheme at the Link Control (LC) layer to ensure reliable information transfer. To efficiently implement RR coding, the receiver may return after every DATA packet a suggestion for the suitable RR value to be used by the transmitter and implement a Stop‐and‐Wait (SW) ARQ scheme at the medium access control (MAC) layer. The effectiveness of employing this optional SW ARQ scheme at the MAC layer is discussed. Analytical models for the ARQ retransmission schemes are developed and employed to compare protocol utilization for different link parameter values such as window size, packet length and LC time out periods. This analysis identifies the ARQ protocol that maximizes performance for the specific link quality and the implemented link layer parameters. The effectiveness of the proposed RR coding to LAN utilization for different ARQ scheme implementation is finally explored. This analysis identifies the link quality level at which RR should be adjusted for maximum performance. It is concluded that if the packet error rate is higher than 0.1–0.4 (depending on the implemented ARQ protocol), the receiver should advise the transmitter to double the implemented RR for maximum performance. These error rate values are high and can be effectively estimated by the transmitter based on packet retransmissions. Thus, the usefulness of the receiver indicating to the transmitter to adjust RR is questionable, as the transmitter can effectively implement the suitable RR value based on packet retransmissions. Copyright © 2003 John Wiley & Sons, Ltd.