Design of a broadband dielectric resonator antenna and its application in low sensitivity frequency beam‐scanning array with wide scanning angle and suppressed sidelobes

In this paper, a broadband dielectric resonator antenna (DRA) with a simple H‐slot feeding structure is proposed. The broad bandwidth (~45%) and uni‐directional radiation enable this DRA suitable to work as a radiation element in the construction of a beam‐scanning array with not only low frequency sensitivity which is important for accurate angle estimation of in‐coming targets, but also a wide scanning range. Moreover, stable coupling coefficients are established around all the above three resonances with respect to its corresponding feeding line, ensuring a low sidelobe level (SLL) across the whole operating frequency range. In order to realize such a DRA, a latticed rectangular structure is utilized to generate the lower two resonances, while the upper resonance is generated by its H‐shaped feeding slot. The shape of the latticed dielectric resonator is also modified according to a 3‐D printed fixing structure for an accurate installation. A 20‐element beam‐scanning prototype is demonstrated and manufactured. The range of scanning angle is from ?36° to +13° within the bandwidth of 10 to 16 GHz, corresponding to a frequency sensitivity of only 122.5 MHz/^o. The achieved SLL are smaller than 19 dB for all scanning beams. The above performance indicates that, this array is very suitable for near‐range radar systems requiring an accurate angle estimation.     

基于Kalman滤波的GSC改进语音增强算法

针对广义旁瓣相消器（Generalized sidelobe canceller,GSC）存在非相干噪声消除性能不佳的缺陷,提出了采用后置Kalman滤波器改进的GSC去噪算法。该算法通过归一化最小均方算法校正自适应噪声对消器,并将滤除方向性干扰噪声后的语音信号输出到Kalman滤波器中,对残余背景噪声进行迭代最小均方误差（Minimum mean square error,MMSE）估计,抑制非相干噪声与麦克风阵元所产生的热噪声。经过在不同信噪比条件下客观语音质量评估（Perceptual evaluation of speech quality,PESQ）及语谱图分析后证明,与传统的GSC以及后置谱减法的改进GSC相比,本算法在噪声消除上的表现更为优越,且增强后信号也更接近目标信号。     

Successive cancellation multibit-flipping decoding algorithm for polar codes

The investigation of the probability of error bits in a codeword shows that single-bit errors and 2-bit errors are main reasons that influence the decoding performance of polar codes with a short or moderate blocklength. Based on the statistical analysis of the error bits, a successive cancellation(SC) multibit-flipping decoding algorithm is proposed. Compared with the SC single-bit-flipping decoding, the proposed scheme can obtain better performance gains for polar codes with a short blocklength. Compared with the SC list decoding by exploring multiple paths simultaneously, the proposed scheme has a lower space complexity. Moreover, the decoding complexity of the proposed decoding algorithm decreases as the signal to noise ratio(SNR) increases. Simulation shows that for polar codes with a short or moderate blocklength, the performance of the proposed decoder can approach that of the SC list decoding with lower decoding complexity in high SNR regions.     

一种低副瓣锐截止波束赋形阵列天线设计

采用联合DFP和BFGS变尺度,结合全局寻优算法,通过优化阵列各单元馈电幅度和相位分布,设计了一种满足低副瓣锐截止波束赋形要求的阵列天线.该阵列由26个印刷振子单元天线组成,各单元馈电电路采用带状线环形桥.馈电电路网络采用带状线结构,整个馈电网络和印刷天线采用一体化设计,具有插入损耗小,结构紧凑的特点.实测表明,3dB波束宽度为11°,水平面锐截止为5.83dB/(°),副瓣低于-20dB.天线设计满足指标要求,并已获得应用.     

Variable step-size affine projection algorithm based on global speech absence probability for adaptive feedbackcancellation简

A novel approach is proposed for improving adaptive feedback cancellation using a variable step-size affine projection algorithm（VSS-APA） based on global speech absence probability（GSAP）.The variable step-size of the proposed VSS-APA is adjusted according to the GSAP of the current frame.The weight vector of the adaptive filter is updated by the probability of the speech absence.The performance measure of acoustic feedback cancellation is evaluated using normalized misalignment.Experimental results demonstrate that the proposed approach has better performance than the normalized least mean square（NLMS） and the constant step-size affine projection algorithms.     

基于M估计的有效成比例仿射投影算法

在回声消除中,针对回声路径具有稀疏性的优势,有学者提出基于μ准则记忆的改进型成比例仿射投影算法（μ‐MIPAPA）,但该算法抗脉冲干扰性能较差。为此将稳健估计中的M估计应用于该算法,提出一种基于M 估计μ准则记忆的改进型成比例仿射投影算法（μ‐MMIPAPA）。为使算法具有更好的稳态性能,采用改良的 Huber M 估计函数,改进阈值参数的选取方式。有色输入下的稀疏回声信道辨识结果表明了μ‐M M IPA PA的良好稳态性能和抗脉冲干扰性能。     

基于预测滤波器的数字降噪耳机研究

通常数字降噪耳机采用的自适应滤波器,一般基于LMS、NLMS等算法,但是由于A/D转换存在一定的延迟,导致实际降噪效果降低,故降噪耳机上的主动降噪系统需要对噪声进行预测.通过线性预测算法和改进滤波器,对已经采样的噪声信号进行算法分析,预测下一时刻的噪声信号,在Matlab中实现对飞机引擎噪声进行预测,并将降噪效果与LMS自适应滤波器的降噪效果进行比较.仿真结果表明,在处理低频噪声时,预测滤波器效果要优于LMS自适应滤波器.     

矿用IP广播对讲系统回声消除技术研究

分析了矿用IP广播对讲系统的回声产生路径,指出其回声具有来源复杂、路径延迟时间长、路径延迟抖动大的特点;提出在广播主机采用软件算法消除回声,在广播终端采用结构设计和硬件电路设计相结合的方法消除回声的方案。实际应用表明,该回声消除方案有效地提高了矿用IP广播对讲系统的通信质量,系统全双工扩音对讲通话效果良好。     

数字可视电话系统中的G.168回声消除技术

由于数字可视电话系统的性能需求和系统中回声的特点,传统的回声消除技术应用于数字可视电话系统时无法满足ITU-T G.168标准要求。该文利用NLMS算法的滑动窗自适应滤波器和简化非线性处理器设计,实现一种数字可视电话系统中的G.168自适应回声消除器,介绍其基本结构和各个功能模块的算法,并以ARM 920T为硬件平台实现该系统。仿真结果显示,简化的回声消除算法运算复杂度小、消除回声有效性高、稳定性强。利用ITU-T G.168标准定义的CSS信号对G.168回声消除器进行测试,测试结果均符合或优于标准要求。     

基于分段块频域自适应滤波算法的长延时回声消除

在网络电话、电话会议、车载电话、IP电话等免提电话系统以及其他一些通信系统中,要想得到高质量的回声消除效果,必须抑制声学回声、尤其是长延时回声.然而对于长延时回声的抑制,如果采用传统的基于LMS/NLMS的自适应算法,则会产生大而复杂的计算量,且收敛速度慢.为解决此问题,本文对分段块频域自适应滤波算法进行了研究,即首先对该算法进行了描述,然后通过录制两个语音信号进行仿真.仿真结果表明,该方法具有非常好的长延时声学回声消除效果.