基于MSS-PDS和单阵元PTRM信道均衡的深海远程水声通信方案

为了提高水声通信的性能,提出了一种联合运用M元扩频（MSS）通信和Pattern时延差编码（PDS）水声通信体制,并采用单阵元被动式时间反转镜（PTRM）来实现声信道均衡的深海远程水声通信方案。该方案的M元扩频-Pattern时延差编码通信技术既能胜任远程水声通信,又能提高通信速率;单阵元被动式时间反转镜信道均衡技术既可抑制多途扩展产生的码间干扰,又能提高信噪比。根据深海声道特性分析,该方案将通信固定节点置于声道轴以获取会聚增益,提高通信距离。计算机仿真结果证明,所提出的深海远程水声通信方案具有很好的鲁棒性和可行性。     

浅海水声信道中的一种带循环均衡的Turbo译码结构

针对浅海水声信道长时延、强多途干扰的特点,建立了一种新的带自适应循环判决反馈均衡的Turbo译码结构,并对其性能进行了仿真研究.该结构对典型的Turbo译码器进行了改进,使之不但能输出信息比特,同时又能输出校验比特,这些比特经硬判决、符号映射及信道交织后反馈回判决反馈均衡器,构成带自适应循环判决反馈均衡的Turbo译码结构.厦门港实测5途浅海水声信道仿真结果显示,该结构具有优良的抗多途性能,信噪比大于4dB时,误比特率小于10-5,比联合判决反馈均衡的Turbo译码结构提高了约2.5dB.     

码率兼容QC-LDPC码在水声通信中的应用

针对水声信道复杂多变、强多途和大起伏干扰等特点,提出采用码率兼容QC-LDPC码的自适应信道编码技术,以提高水声数据传输的可靠性和水声信道的信道吞吐率。建立了码率兼容QC-LDPC码在水声信道中的仿真模型,构造码率为1/2、2/3…5/6等一系列QC-LDPC码,在三种典型的水声信道中对其性能进行仿真。结果表明所设计的码率兼容QC-LDPC码在三种不同信道下可行有效,能较大地提高水声通信系统的性能。由于信道时延越长、信噪比越低,满足通信指标的码率就越低;并给出码率兼容QC-LDPC码在浅海水声信道中不同信噪比下满足通信性能指标的编码码率的查找表。码率兼容QC-LDPC码提高了水声通信的信道利用率,具有灵活的编译码性能,在水声自适应通信中具有很好的应用前景。     

基于FFH/MFSK的深海远程水声通信技术

为实现深海远程水下通信,提出了一种FFH（Fast Frequency Hopping）/MFSK（快跳频/多频移键控）通信方案。该方案利用快跳频技术对已调MFSK信号进行二次调制,借助频率分集,可有效地抑制远程水声信道因多途传播而引起的频率选择性衰落,从而更好地实现远程低信噪比条件下可靠通信。同时采用深海声道作为通信系统水声波导,将通信节点置于声道轴以大幅提高通信距离。仿真及湖试结果证明了所提方案的良好性能。     

STBC-OFDM技术在水声通信中的试验研究

空时分组编码（Space Time Block Coding,STBC）技术能够获得满发射分集增益并且译码简单,而正交频分复用（Orthogonal Frequency Division Multiplexing,OFDM）技术可对抗水声信道中的多径干扰和实现较高的频谱利用率。设计了将空时分组编码与正交频分复用相结合（STBC-OFDM）的高速水声通信系统方案,并进行了湖上试验研究。湖试中,在1200m距离下,带宽为4.32kHz时有效传输速率为3.328kbps,误比特率低于10-5。湖试结果表明,STBC-OFDM技术在高速水声通信中的应用潜力。     

MIMO水声通信中的空时分组扩展编码方案探讨

采用多输入多输出（MIMO）结构和空时编码可以使水声通信系统获得分集增益,改善系统性能。针对水声信道中的频率选择性衰落,研究了一种用于MIMO水声通信系统的空时分组扩展编码方案（STBSC）,利用正交扩频编码克服了水声信道中多径干扰对空时分组编码（STBC）信号正交性的破坏。给出了方案的系统模型和实现框图,采用MIMO水声模型对方案的性能进行了仿真。仿真结果表明,STBSC方案在多径水声信道中可以得到完全发射分集。最后介绍了方案的水库试验,试验结果表明了方案的有效性。     

浅海中低频水声信道仿真研究

在中低频段（通常为数百至数千赫兹频率范围）,传播损失、时空相关性以及时空平稳性等水声信道特性是主、被动目标探测和远程通讯技术研究的重点。为了模拟信道的时空去相关特性,应用KrakenC传播模型得到理想信道传递函数H（r,ω）,并在此基础上,提出了新的基于信道时延的相位加扰方法。通过数值仿真,该方法可以很好地在统计特性上模拟海试LoFAR’08中的实际浅海信道测量数据。测量数据由拖曳式电火花声源（有效带宽为300Hz~2.5kHz）激励浅水信道得到。     

水声通信中的信道建模与仿真技术研究

水声信道是水声通信的一个重要组成部分,它的研究可以提高水声信号传输的可靠性。对水声通信中不同信道模型的传播特性进行较为详细的理论分析,并用MATLAB仿真软件对不同的信道模型的传播损失特性进行仿真,为水声通信系统的信道设计提供理论基础。     

水声信道的宽带传播特性仿真分析

以射线模型为基础,在水声信道传输模型中,引入反映信道声吸收频率特性的FIR滤波器,能够更真实地模拟水声信道对宽带信号传输特性的影响,并以白噪声声源的传播为例验证了模型的正确性。     

OFDM水声通信线性最小均方误差算法信道均衡

正交频分复用在高速水声通信中的应用越来越广泛.水声信道是多径、高噪声的衰落信道.由于信道条件差,容易产生码间干扰,信道均衡技术是提高通信系统性能的重要技术.提出了线性最小均方估计在水声通信系统中应用的方案,研究了实际通信中利用调频信号如何确定信号的平均多径时延、最大多径时延,比较了不同调频信号测量信道的性能.该方案基于导频辅助均衡方法,利用调频信号测量信道,获得的信道统计信息,计算信道的自相关矩阵.从而根据线性最小均方误差算法进行信道均衡.通过仿真和湖试进行验证,结果表明该方案效果较好,对系统性能改善比较明显.     

双向无线内窥镜系统的ARQ动态自适应包长确定方法

在对双向数字式无线内窥镜系统通信信道特征分析的基础上,给出了一种适应于超短距离高码率双向数字式无线内窥镜系统的自动请求重传(ARQ)机制及其动态自适应包长的确定方法.试验与仿真结果表明,该ARQ通信机制不仅保证了图像传输的高质量,而且还能保证传输一帧图像的效率和在规定时间内正确传输一帧图像的效率均能达到最优.     

延迟利用的多节点协同预约式水声网络MAC协议

水声网络媒体接入控制(Media Access Control,MAC)层为水声节点提供接入和冲突避免服务。由于水声信道面临着传播延迟高、带宽窄等困难,水声网络MAC协议的性能较低。文章在MACA-U协议的基础上,针对水声网络传播延迟高的特点,对信道预约的长等待时间进行利用,提出了一种RS-MACA协议。该协议在负载较低时与MACA-U协议兼容;在负载较高时,邻近节点可采用"并行发送"的方式发送,显著提高了吞吐量性能。发射节点在信道预约等待时间内,同时接受符合条件的邻节点的发送预约,邻节点不需要消耗额外时间进行信道预约。完成预约的节点按照时隙进行数据发射,避免了数据包的碰撞。NS3软件仿真结果表明,与MACA-U协议相比,改进后的协议吞吐量性能有明显的提升。     

水声通信系统中分布式空时分组码的研究

协作传输可以获得无线信道内在的空间分集,有效改善通信系统的性能,增加通信系统的覆盖范围。对水声通信系统中两中继节点协作的译码转发方案进行了研究,提出了基于译码转发协议(DF)的分布式空时分组扩频编码(DSTBSC)方案,该方案可以克服多径衰落对分布式空时分组码(DSTBC)信号正交性的影响。给出了DSTBSC方案的系统模型,着重分析了在采用两中继译码转发时,中继节点的误码对水声通信系统性能的影响。在接收端不考虑信号冲突的情况进行仿真,结果表明,与直接传输方案相比,采用两中继节点、译码转发的DSTBSC方案,在两中继节点处均无误码和一个中继有误码时在相同的比特误码率(BER)条件下均可获得约4dB的分集增益,增加了传输距离。     

Improved regular and semi-random rate-compatible low-density parity-check codes with short block lengths

Powerful rate-compatible codes are essential for achieving high throughput in hybrid automatic repeat request (ARQ) systems for networks utilising packet data transmission. The paper focuses on the construction of efficient rate-compatible low-density parity-check (RC-LDPC) codes over a wide range of rates. Two LDPC code families are considered; namely, regular LDPC codes which are known for good performance and low error floor, and semi-random LDPC codes which offer performance similar to regular LDPC codes with the additional property of linear-time encoding. An algorithm for the design of punctured regular RC-LDPC codes that have low error floor is presented. Furthermore, systematic algorithms for the construction of semi-random RC-LDPC codes are proposed based on puncturing and extending. The performance of a type-ll hybrid ARQ system employing the proposed RC-LDPC codes is investigated. Compared with existing hybrid ARQ systems based on regular LDPC codes, the proposed ARQ system based on semi-random LDPC codes offers the advantages of linear-time encoding and higher throughput.     

Differentiated cooperative multiple access for multimedia communications over fading wireless networks

The quality of service (QoS) support for multimedia communications faces a big challenge in a fading wireless network. On one hand, conventional automatic repeat request (ARQ) schemes are not effective for small-scale fading channels with correlated errors due to consecutive retransmission failures. On the other hand, large-scale fading due to propagation loss or shadowing severely limits transmission range. A novel differentiated cooperative medium access control (MAC) protocol, called DC-MAC, is proposed to enhance the QoS support for multimedia communications while supporting service differentiation based on the IEEE 802.11e architecture. By enabling cooperative ARQ, the retransmission is initiated from an appropriate transmission queue of an appropriate relay node instead of the original source. Since unnecessary and useless retransmissions may intensify the node contention and degrade the system performance contrarily, a novel negative acknowledgement feedback mechanism is introduced for loss distinguishing and channel estimation such that cooperative retransmission will be employed only when necessary and only by competent nodes. Extensive simulations are conducted on the OPNET platform to analyse the performances of DC-MAC under both small-scale and large-scale fading. Simulation results show that the proposed scheme significantly improves the performances of both multimedia applications and best-effort data applications in terms of throughput, delay and coverage with moderate user contention.     

虚拟时间反转镜PDS水声通信双向均衡方案

虚拟时间反转镜（Virtual Time Reversal Mirror,VTRM）引导了空间聚焦和信道均衡,将其应用在Pattern时延差编码（Pattern Time Delay Shift Coding,PDS）水声通信体制,构成VTRM-PDS系统,可增强抗多途干扰的能力。但当通信节点间存在缓慢运动,即信道缓慢时变时,在一帧信号中,信道均衡的效果会随时间逐渐降低,产生失配现象。提出了双向均衡方案,在没有信道先验知识的前提下,通过前向和后向信道均衡,充分利用探测信号和信道信息,提高了信道均衡效果。湖试结果证明了该方案的优越性。     

Interaction between radio link level truncated ARQ, and TCP in multi-rate wireless networks: a cross-layer performance analysis

A complete queueing model for radio link layer performance analysis is developed assuming adaptive modulation and coding (AMC) at the physical layer and truncated automatic repeat request (ARQ)-based error control at the link layer. From the model, queue length distribution and average queueing delay can be calculated. The average queueing delay is then used to estimate transmission control protocol (TCP) throughput performance using a fixed-point approach. Using the model, we are able to choose signal-to-noise ratio thresholds of different transmission modes for AMC at the physical layer for different persistence levels of ARQ at the link layer so that TCP throughput is maximized. We observe that channel correlation negatively impacts the TCP throughput performance. Also, throughput enhancement of TCP NewReno over TCP Reno is observed to be non-negligible only if no ARQ-based error recovery is employed at the link layer.     

基于正交频分复用的水下转发器的设计

为解决长距离高速水声数据传输问题,采用接力式水声数据转发传输是一个很好的解决方案。结合正交频分复用技术的特点设计实现了一个数字式的水下转发器系统。系统功耗低、结构简单,可以很好地解决水声通信中存在的多途干扰和窄带宽问题,实现高速、可靠的水下数据转发的目的。湖试中,在6000m的距离内,3kHz带宽下,转发速率为4.32kb/s,误码率低于10-3。湖试结果表明,基于正交频分复用的水下转发器可以达到较高的转发速率和较低的误码率。     

Effective transmission opportunity allocation scheme for real-time variable bit rate traffic flows with different delay bounds

The medium access control of IEEE 802.11e defines a novel coordination function, namely, hybrid coordination function (HCF), which allocates transmission opportunity (TXOP) to stations taking their quality of service (QoS) requirements into account. However, the reference TXOP allocation scheme of HCF controlled channel access, a contention-free channel access function of HCF, is only suitable for constant bit rate traffic. For variable bit rate traffic, packet loss may occur seriously. The authors propose a TXOP allocation scheme to efficiently allocate bandwidth and meet the QoS requirements in terms of both delay bound and packet loss probability. To achieve high bandwidth efficiency, the authors take advantage of not only intra-flow multiplexing gain of traffic flows with large delay bounds, but also inter-flow multiplexing gain of multiple traffic flows with different delay bounds. According to numerical results obtained by computer simulations, the proposed TXOP allocation scheme results in much higher bandwidth efficiency than previous algorithms under the same constraints of delay bounds and packet loss probability.