水声通信实验信道仿真研究

水声信道的时变多途特性对其中通信信号传输构成严重影响，为了分析评估现实实验信道条件下水声通信系统的性能，给出了一种基于实验测量的水声信道仿真方法。此方法通过发射线性调频脉冲以检测信道多径结构，同时分别采用自回归模型与多普勒时间压扩来仿真信道中各路径的时变幅度与时变延迟。通过一个具体的直接序列扩频水声通信系统，表明在仿真与实验信道条件下系统性能具有较好的一致性。     

基于高阶统计的水声信道盲辨识

本文给出了基于ARMA模型的水下通信系统模型,将一种基于三阶统计的算法应用到水声信道盲辨识领域,在不需要训练序列的条件下估计得到信道传输函数。与基于二阶统计特性方法相比,该算法具有较强的抗噪性,更适合信噪比低于12dB条件下水声信道的辨识。通过对水声信道盲辨识的计算机仿真,验证了该算法具有较高的辨识精度。     

极化编码调制正交频分复用水声通信中的极化码构造方法

为进一步提高水声通信可靠性和频带利用率,针对极化编码调制的水声通信需求和信道特性,基于蒙特卡洛法提出动态水声信道认知优化统计目标参数、联合译码的判决反馈信道估计两点改进,建立了适用于编码调制水声信道中的极化码构造算法。为验证该算法性能,建立了极化编码调制水声通信的两步应用机制。通过仿真,比对并分析改进前后的极化码构造方法性能、对时变信道的鲁棒性,以及在不同映射规则下的联合比特交织和多级编码的极化编码调制水声通信性能,并与低密度校验(LDPC)编码调制系统进行对比。湖试结果表明,提出的极化编码调制水声通信方案有效保证了信息在浅水水声信道中的可靠传输,在信噪比约为14 dB、通信距离约1 km时,实现无误码传输,性能优于相同条件下的LDPC编码调制系统。     

灰霾粒子与水云粒子不同混合方式对量子卫星通信性能影响

为了研究混合粒子的不同混合方式对星地量子卫星通信的影响,根据灰霾粒子与水云粒子的谱分布函数,以及不同混合方式下的消光系数,提出了外混合方式中星地量子信道衰减的计算关系,建立了内混合方式中的Core-shell信道衰减模型;分析了在不同混合方式下,混合粒子的粒径比与信道平均保真度、信道误码率之间的定量关系.仿真结果表明,当混合粒子的粒径比分别为0.2和0.8时,外混合粒子对应信道容量、信道平均保真度、信道误码率分别为0.39和0.27,0.8和0.8,0.003和0.009;内混合粒子对应信道容量、信道平均保真度、信道误码率分别为0.8和0.21,0.94和0.81,0.018和0.021.由此可见,灰霾粒子和水云粒子的不同混合方式对量子卫星通信性能的影响有显著差别.因此,在实际的量子卫星通信系统中,应根据混合粒子的不同混合方式自适应调整系统的各项参量,以提高星地量子通信链路的可靠性.     

浅水时变多途信道特性分析与模型实验研究

时变多途水声信道是影响水声通信性能的一个重要因素,深入了解水声信道挣性并对水声信道进行建模具有重要的研究意义。本文在分析传统时变信道模型基础上,从时域角度出发,提出采用AR(ρ)模型对已有的浅水时变多途信道模型中的小尺度衰落分量进行修正。然后基于莲花湖实测信道数据,分析了时变信道各个路径的幅度、时延、时间相干系数、系统传递函数等特性,并对修正模型进行了验证正。从仿真结果来看,修正模型与试验结果拟合的很好。本文从多个方面对水声信道特性进行了分析,有助于对水声信道的深入了解,同时为水声通信系统性能评估提供了一种仿真信道模型。     

不同海面风速对量子卫星星舰通信性能的影响

量子卫星星舰通信是量子保密通信的重要应用场景之一,在海面上,由于不同风速所引起的气溶胶粒子浓度发生剧烈变化,而气溶胶粒子浓度的剧变,必然导致星舰量子链路性能的剧烈衰减.然而,有关不同海面风速与量子卫星星舰通信信道参数关系的研究,迄今尚未展开.本文根据海面风速与气溶胶的Gras模型,分别建立了风速与星舰量子信道误码率、信道容量和信道平均保真度的定量关系.仿真结果表明,当风速分别为4 m/s和20 m/s时,海洋大气信道误码率、信道容量、信道平均保真度分别依次为4.62×10^-3和4.91×10^-3、0.957和0.65、0.999和0.974.由此可见,风速对海上量子通信性能有显著的影响.因此,为了提高通信的可靠性,应根据风速大小,自适应调整系统的各项参数.     

基于袋鼠纠缠跳跃模型的量子状态自适应跳变通信策略

自由空间中的量子通信会不同程度上受到雾霾、沙尘等自然环境的干扰.为了研究提升此类干扰下量子通信的性能,本文分析了背景干扰下单量子态信道随时间演化的性能变化,并根据袋鼠纠缠跳跃模型(KEHM),提出了基于KEHM的量子状态自适应跳变通信策略,对其性能参数进行仿真.仿真结果表明,采取量子状态跳变,在背景量子噪声的平均功率与量子信号平均功率的比值为5的情况下,量子误比特率随着量子态跳频率从1增大到15,由0.4524降低到0.1116;当单量子态传输成功率0.95,量子比特率大于200 qubit/s时,不同态跳频率下量子比特的成功传输概率均大于0.97.当发送端信号源平均量子数足够大且接收端接收效率趋近于1时,量子态的通过率也趋近于1;采取量子态跳自适应控制策略,能够进一步降低系统的误码率.     

离散速率条件下的大气激光通信自适应调制编码系统性能

针对大气湍流引起的系统频带利用率下降的问题,研究了一种离散速率条件下的大气激光通信自适应调制编码系统。从信号层角度建立了湍流信道模型,给出了湍流信道下的瞬时信噪比概率密度函数,并采用外场实验验证了该信道模型。理论推导了系统频带利用率和误码率表达式,仿真结果表明大气激光通信自适应调制编码系统较传统单一传输模式系统具有优越性。研究了其频带利用率和误码率曲线特性,分析了误码率要求和湍流强度对系统性能的影响。结果表明降低误码率要求可大幅提高系统的频带利用率,且误码率要求越低,湍流强度越弱,频带利用率越高。     

基于改进卡尔曼滤波的水声通信盲自适应多用户检测算法

针对水声通信网络中遇到的多用户检测中目标用户的多址干扰等检测问题,提出了基于改进Kalman算法的盲自适应多用户检测算法,解决了多用户检测中的多址干扰对水声通信信道用户变动时的干扰抑制问题。仿真分析分别针对同步多用户、异步多用户通信过程,对比了传统Kalman算法及改进的Kalman算法的性能差异,通过仿真对比表明,改进后的Kalman检算法不需要训练序列即可以实现同步和异步通信状态下的多水声目标用户的盲自适应检测,改进后的算法目标检测的信干比比传统算法最大可提高6dB。新算法对于水下多用户检测、区分,准确、稳定的实现基于CDMA协议的快速水声通信具有重要意义。     

空地激光通信链路功率与通信性能分析与仿真

分析了空地激光通信系统中主要器件和信道对通信光功率的影响,并根据接收探测器的信噪比和通信误码率公式,建立了空地激光通信仿真系统.分析了在误码率优于10-7条件下,不同地面大气能见度所对应的最高通信速率;以及要实现通信速率为1.5 GHz,误码率优于10-7时需要的最小发射功率和最长通信距离.结果表明,当发射功率越大时,地面大气能见度对误码率的影响越明显.     

Channel temporal correlation-based optimization method for imperfect underwater acoustic channel state information

The rapid time variations and large channel estimation errors in underwater acoustic (UWA) channels mean that transmitters for adaptive resource allocation quickly become outdated and provide inaccurate channel state information (CSI). This results in poor resource allocation efficiency. To address this issue, this paper proposes an optimization approach for imperfect CSI based on a Gauss–Markov model and the per-subcarrier channel temporal correlation (PSCTC) factor. The proposed scheme is applicable to downlink UWA orthogonal frequency division multiple access systems. The proposed PSCTC factors are measured, and their long-term stability is verified using data recorded in real-world sea tests. Simulation and experimental results show that the optimized CSI effectively mitigates the effects of the temporal variability of UWA channels. It demonstrates that the resource allocation scheme using optimized CSI achieves a higher effective throughput and a lower bit error rate than both imperfect CSI and the CSI predicted by the recursive least-squares (RLS) algorithm.     

Signal processing for high speed underwater acoustic transmission of image

A signal processing method for high-speed underwater acoustic transmission of image is presented. It has two parts. Part 1 introduces signal processing for underwater acoustic coherent communication. Part 1 includes 3 technical points. （1） Doppler shift compensation. Chirp signals are inserted between data packages. A correlation process between two copy correlation functions gives more accurate estimation of the mean Doppler shift. Then it could be compensated by resampling the data. In adaptive decision feedback equalizer （DFE） an adaptive phase compensator with fast self-optimized least mean square （FOLMS） adaptation algorithm is utilized resulting in better motion tolerance than compensators with 2nd order Phase-Lock Loop algorithm. The performance of the combination of mean Doppler shift compensation and adaptive phase compensator is quite good. （2） A diversity combiner （DC） used in advance of equalizer. Both combiner and adaptive DFE are based on FOLMS adaptation algorithm. This results in reduced computation complexity and better performance. （3） Cascaded equalizer and Turbo-Trellis Coded Modulation （TCM） decoder and the iteration algorithm. A new bitsymbol converter based on Soft Output Viterbi Algorithm （SOVA） is studied. Comparing with the traditional decision, coding and mapping algorithm, the new converter can reduce Bit Error Rate（BER） by nearly 2 orders. Part 2 is mainly around a robust image compression algorithm. Based on Discrete wavelet transform and fixed length coding, a robust compression algorithm for acoustic image is studied. The algorithm includes 4 technical points. （1） Utilizes CDF9/7 wavelet bases to transform the images. （2） Analyses the energy distribution of subband coefficients. Suitable transformation layer number is 3. （3） Applies different quantization steps to different subbands in accordance with their energy distribution. （4） Uses fixed length coding to prevent error propagation. The results show the algorithm achieves a balance among image quality, compression rate, and most important, robustness to BER. The compressed bit rate of gray scale acoustic image is 0.85 bit/pixel. Image quality remains good when BER is lower than 10^-3. There are some small dirty points when BER rises to 10^-2. Based on the signal processing techniques above mentioned, an underwater acoustic commumcation system is built. Its operational frequency band is （7.5-12.5） kHz. Its receiving array is an 8 elements uniform linear array. QPSK and 8PSK modulation and iteration algorithm for cascaded equalizer and 1-Turbo-TCM decoder based on hard SOVA are used. The system has been tested in Qiandao Lake. Low BER is achieved in 5.5 km range when data rate is 10 kbps. One gray scale image can be transmitted in 7 s. The product of its communication distance and data rate is 55 km kbps.     

Pre-processing optimization for IBI mitigation in an OFDM system over channels with large delay spreads

When the cyclic prefix (CP) in an orthogonal frequency division multiplexing (OFDM) system is shorter than the delay span of wireless channels, inter-block interference (IBI) and inter-carrier interference (ICI) will occur. To effectively mitigate interference in this case, pre-processing approaches at transmitters based on full channel state information (CSI) have been proposed. However, feeding full CSI back will occupy significant bandwidth and is impractical sometimes. Therefore, a pre-processing optimization exploiting limited CSI is investigated in this paper; several pre-processing optimization approaches have been developed. Simulation results show that the proposed algorithms can effectively cancel the IBI and ICI and significantly improve the performance in OFDM systems.     

Adaptive and efficient nonlinear channel equalization for underwater acoustic communication

We investigate underwater acoustic (UWA) channel equalization and introduce hierarchical and adaptive nonlinear (piecewise linear) channel equalization algorithms that are highly efficient and provide significantly improved bit error rate (BER) performance. Due to the high complexity of conventional nonlinear equalizers and poor performance of linear ones, to equalize highly difficult underwater acoustic channels, we employ piecewise linear equalizers. However, in order to achieve the performance of the best piecewise linear model, we use a tree structure to hierarchically partition the space of the received signal. Furthermore, the equalization algorithm should be completely adaptive, since due to the highly non-stationary nature of the underwater medium, the optimal mean squared error (MSE) equalizer as well as the best piecewise linear equalizer changes in time. To this end, we introduce an adaptive piecewise linear equalization algorithm that not only adapts the linear equalizer at each region but also learns the complete hierarchical structure with a computational complexity only polynomial in the number of nodes of the tree. Furthermore, our algorithm is constructed to directly minimize the final squared error without introducing any ad-hoc parameters. We demonstrate the performance of our algorithms through highly realistic experiments performed on practical field data as well as accurately simulated underwater acoustic channels.     

An extended neck versus a spiral neck of the Helmholtz resonator

This paper focuses on improving the noise attenuation performance of the Helmholtz resonator (HR) at low frequencies with a limited space. An extended neck or a spiral neck takes the place of the traditional straight neck of the HR. The acoustic performance of the HR with these two types of necks is analyzed theoretically and numerically. The length correction factor is introduced through a modified one-dimensional approach to account for the non-planar effects that result from the neck being extended into the cavity. The spiral neck is transformed to an equivalent straight neck, and the acoustic performance is then derived by a one-dimensional approach. The theoretical prediction results fit well with the Finite Element Method (FEM) simulation results. Without changing the cavity volume of the HR, the resonance frequency shows a significant drop when the extended neck length or the spiral neck length is increased. The acoustic characteristics of HRs with these two different neck types have a potential application in noise control, especially at low frequencies within a constrained space.     

Ergodic mutual information of OFDMA-based Selection-Decode-and-Forward cooperative relay networks with imperfect CSI

This paper presents a novel approach to investigate ergodic mutual information of OFDMA Selection-Decode-and-Forward (SDF) cooperative relay networks with imperfect channel state information (CSI). Relay stations are either dedicated or non-dedicated (i.e., subscriber stations assisting other subscriber stations). The CSI imperfection is modeled as an additive random variable with known statistics. Numerical evaluations and simulations demonstrate that by considering the CSI imperfection based on a priori knowledge of the estimation error statistics, a substantial gain can be achieved in terms of ergodic mutual information which makes channel adaptive schemes closer to practical implementations.     

Performance analysis of overlay cognitive NOMA network with imperfect SIC and imperfect CSI

In this paper, we investigate a multiple users cooperative overlay cognitive radio non-orthogonal multiple access (CR-NOMA) network in the presence of imperfect successive interference cancellation (SIC) and imperfect channel state information (CSI). In the context of cellular network, cell-center cognitive secondary users act as relays to assist transmission from the primary user (PU) transmitter to the cell-edge PU receiver via NOMA. According to the received signals between the primary transmitter and multiple cognitive secondary center users, the best cell-center cognitive SU with the maximum signal to noise ratio (SNR) is selected to transmit the PU’s signals and its own signal to cell-edge users through NOMA principle. Then, the PU cell-edge user combine the signals received from direct transmission in the first phase and relay transmission from the best cell-center cognitive SU in the second phase by selection combining (SC). To measure the performance of the system quantitatively, we derive the end-to-end outage probability and capacity for the primary and secondary networks by taking the imperfect SIC and CSI into consideration. Finally, the performance analysis is validated by the simulations, and show that serious interference caused by imperfect SIC and (or) imperfect CSI reduce the system performance.     

Performance of multi-user transmitter preprocessing assisted MIMO system over correlated frequency-selective channels

In this contribution we present the performance of a multi-user transmitter preprocessing (MUTP) assisted multiple-input multiple-output (MIMO) space division multiple access (SDMA) system, aided by double space time transmit diversity (DSTTD) and space time block code (STBC) processing for downlink (DL) and uplink (UL) transmissions respectively. The MUTP is invoked by singular value decomposition (SVD) which exploits the channel state information (CSI) of all the users at the base station (BS) and only an individual user’s CSI at the mobile station (MS). Specifically, in this contribution, we investigate the performance of multi-user MIMO cellular systems in frequency-selective channels from a transmitter signal processing perspective, where multiple access interference (MAI) is the dominant channel impairment. In particular, the effects of three types of delay spread distributions on MUTP assisted MIMO SDMA systems pertaining to the Long Term Evolution (LTE) channel model are analyzed. The simulation results demonstrate that MUTP can perfectly eliminate MAI in addition to obviating the need for complex multi-user detectors (MUDs) both at the BS and MS. Further, SVD-based MUTP results in better achievable symbol error rate (SER) compared to popularly known precoding schemes such as block diagonalization (BD), dirty paper coding (DPC), Tomlinson–Harashima precoding (THP) and geometric mean decomposition (GMD). Furthermore, when turbo coding is invoked, coded SVD aided MUTP results in better achievable SER than an uncoded system.     

Bit error rate evaluation of relay-aided cooperative NOMA with energy harvesting under imperfect SIC and CSI

In this paper, we investigate a decode-and-forward relay-assisted cooperative non-orthogonal multiple access (NOMA) scheme, where the relay implements a time-switching (TS) based energy harvesting (EH). The impacts of the imperfect channel state information (CSI), inter-cell interference (ICI) and imperfect successive interference cancellation (SIC) are taken into account. We derive the end-to-end bit error rate (BER) expressions under imperfect CSI and ICI for both users. The effect of the EH parameters under the imperfect CSI on users’ BER performance is also examined. Furthermore, we discuss the impact of ICI on BER performance. Computer simulations are used for numerical analysis validation. The results reveal that the CSI deterioration reduces the SIC performance in each node despite the increase in EH parameters and causes an error floor at the higher signal-to-noise ratio (SNR). Furthermore, the BER performance of the users increases by increasing the EH parameters. Also, the ICI affects the SIC and degrades the BER of users.     

Variable step-size based method for acoustic feedback modeling and neutralization in active noise control systems

This paper investigates the issue of the acoustic feedback during online operation of active noise control (ANC) systems. In the existing approach, two FIR filters are used for this task: adaptive for feedback path modeling (FBPM) and fixed for feedback neutralization (FBN). Previously, a simplified method is proposed where these two tasks of modeling and neutralization have been combined into one feedback path modeling and neutralization (FBPMN) adaptive filter. Here we introduce an intuition based variable step size (VSS) parameter, for LMS equation of FBPMN filter. This VSS is motivated from the fact that the error signal of FBPMN filter contains a disturbance-component that is decreasing in nature. The computer simulations are carried out for single-channel and multichannel ANC systems. It is demonstrated that the proposed method achieves better performance than the existing methods.