音频无人机定位的时延估计模拟分析

针对无人机非平稳音频信号时差定位中,广义互相关时延估计算法抗噪性差和时延估计值精度低等问题,文章采用了一种基于广义二次相关时延估计的改进算法。算法对叠加了实际噪声（如风声、雨声、汽车鸣笛声等）的无人机音频信号进行频谱细化的广义二次相关,有效抑制了噪声干扰,融合相关峰精确插值算法,提高了互相关函数的分辨率,使得时延峰值更加明显。仿真实验结果表明,改进的广义二次相关方法在不同信噪比时,比广义互相关和广义二次相关算法的时延估计精度更高,稳定性更好。改进的广义二次相关算法对无人机定位中的时延估计具有更好的性能优势,具有较强的实际应用性。     

泄漏检测应用中突发干扰对LMS时延估计性能的影响

文章中把LMS时延估计算法用于地下管网的泄漏检测和漏点定位，避免了经典时延估计算法一广义互相关法(GCC方法)需要知道信号和噪声统计特性等先验知识的不足，但是泄漏检测应用中，管道外在环境造成的突发性强干扰，导致了传感器接收到的信号是非平稳的，而非平稳信号对LMS的时延估计性能有不利影响，文章中分析了这种由突发性强干扰导致的非平稳信号对LMS时延估计收敛性、收敛速度和时延估计值的影响，提出了消除突发干扰的方法。实验表明，在地下管网泄漏检测应用中，该方法能够有效地消除强突发干扰噪声，使得估计性能得到显著的改善。     

基于高阶累积量的高精度时延估计算法

针对海洋探测中由于接收信号信噪比低并存在各种噪声干扰导致时延估计精度低的问题,提出一种基于二次相关和高阶累积量的具有多种噪声抑制能力的高精度时延估计新方法——SC-HOCS法。该方法首先对两路接收信号进行自相关和互相关处理,抑制部分高斯噪声,然后利用高阶累积量一维切片法对信号进行处理,抑制相关高斯噪声和非高斯色噪声,通过对接收信号的上述处理提高信噪比,最后结合希尔伯特变换对相关峰进行锐化处理,进一步提高时延估计精度。与广义相关法、二次相关法及高阶累积量一维切片法相比,该方法能很好地抑制相关噪声并且能在更低的信噪比下获得较好的时延估计精度,同时该算法计算量较小,可满足对数据实时处理的需求。计算机仿真和水池实验验证了该方法的有效性。该方法为海洋探测中低信噪比信号的高精度时延估计提供一种新的技术途径。     

相关Hilbert差值时延估计方法误差分析

被动探测的关键技术之一就是通过有效的信号处理方法提取出多接收器的时间差,而确定时差最基本的方法是进行信号相关以获得估计信息,相关算法虽然计算过程简单,但是算法对信号和噪声频谱特性要求比较高。基于希尔伯特变换的时延估计是广义相关时延估计方法的拓展,把相关函数的对称性由偶函数转化成奇函数,将相关法中对峰值的检测转化为对过零点的检测。通过对两种算法的误差比较分析,信号的截至频率越高,希尔伯特变换法的均方误差与基本相关法的均方误差相差越明显,估计误差越理想,证明Hilbert算法的优越性。     

基于时延估计改进的主动隔声耳罩语音增强算法

为实现高噪声场所中的听力保护和不同角度入射声源的语音增强，文章采用基于谱减法改进的广义互相关（Generalized Cross Correlation, GCC）时延估计方法对麦克风阵列接收信号进行同步处理，将同步后的阵列信号与基于广义旁瓣抵消器（Generalized Sidelobe Canceller, GSC）的主动隔声耳罩语音增强算法结合，实现噪声抑制和语音增强。首先以GCC算法为基础，引入多窗谱估计的谱减法作为时延估计信号的预处理环节，提高传声器接收信号的信噪比对算法进行改进。随后对加入谱减法的GCC时延估计算法，比较分析了取不同加权函数的仿真结果。最后由时延估计结果对麦克风阵列信号进行时延补偿，对改进后的主动隔声耳罩语音增强算法的噪声抑制和语音增强效果进行了分析。与改进前算法相比，在不同噪声环境下输出的语音信号质量有明显提升。     

基于三次相关改进的广义互相关时延估计法在局部放电超声波定位中的研究

在现有的开关柜等电气设备局部放电超声波定位技术中,到达时间差定位法(Time Difference of Arrival,TDOA)在定位精度与技术实现等方面有着一定的优势,得到了广泛的使用,是目前常用的方法,其中的时延估计算法对整个系统起着关键作用。文章首先对目前现有的基本相关、广义相关、二次相关等时延估计算法进行了分析。其次,在二次相关基础上再进行一次相关,并设计了新型的加权函数,将三次相关与广义互相关结合在一起,成为一种新的方法,即广义三次相关时延估计法。最后,搭建了相应的开关柜实验平台并对以上方法进行了实验及对比,分析了各算法的性能。结果表明,广义三次相关时延估计法在相对强噪声环境中较其他算法抗噪性能更强,具有更好的优越性。     

自来水管网相关检漏技术中时延估计方法的选择与综合

管道相关检漏技术中时延估计方法的选择决定了漏点定位的准确性和精确性，本文分析讨论了几种典型的应用环境下时延估计的原理，包括：（１）不相关噪声中各种广义互相关技术和维纳滤波；（２）相关高斯噪声中用多谱技术进行时延估计的方法；（３）混响干扰环境中利用信号重构技术进行时延估计的原理，给出了在设计检漏系统和实际检测时，对时延估计方法进行选择和综合的一般性原则。     

多通道近场广义互相关算法研究

广义互相关时延估计方法(GCC)由于计算简单和易于实现,在被动声源定位系统中得到了广泛应用,在此基础上,提出了多通道近场广义互相关算法。该算法建立时延估计矩阵,得出方位估计函数,用最小方差估计出最优的方位估计量。仿真实验表明,这种方法使估计精度得到提高。     

混响环境中突发声源定向方法及性能

针对突发性声源在混响条件下的定向技术进行了研究。基于传统广义互相关时延估计的定向方法,通过对互相关函数计算结果进行判别与约束、对数据帧采用滑动方式、对时延估计结果进行能量加权选举等措施,有效提高了声源定向结果的稳定性。实验结果表明,改进的算法在混响条件下能有效地对突发性声源进行可靠定向,可用于对枪声、爆炸声等突发性信号进行精确定向和定位。     

相关噪声背景下加权高阶累积量的时延估计方法

时延估计是双基阵被动定位中的重要问题。在非相关噪声干扰下,应用广义互相关法进行时延估计可以取得良好的估计效果,但在相关噪声干扰情况下其估计性能严重下降。使用高阶累积量进行时延估计,可以有效地去除相关高斯噪声的干扰。采用类似广义互相关加权的方法对高阶累积量进行PHAT加权,进一步抑制噪声的影响,提高了高阶累积量法的估计性能。结果表明,在相关噪声背景下加权高阶累积量时延估计效果优于广义互相关法和高阶累积量法。     

广义互相关时延估计声定位算法研究

针对地面目标声定位因信噪比较低而定位精度差的现象,提出了基于广义互相关法的声定位系统,根据平面四元法目标定位计算式,研究了广义互相关算法在实际中的应用。环境噪声、军事目标声和民用目标声频谱范围大多集中,即出现目标声会与背景噪声高度重合的情况,发现一般加窗滤波法在降低噪声的同时亦会将目标声强度大大削弱,而广义互相关时延估计法是通过计算两路信号互相关函数的最大值而求得时延差,其精度高、稳定性好。通过计算机仿真得到在信噪比较低的情况下,加窗滤波因大大削弱目标声强度而造成定位精度较低,而利用广义互相关方法可得到较为精确的声定位坐标。     

Two bit correlation-an adaptive time delay estimation

Time delay estimation is a very important operation in ultrasound time-domain flow mapping and correction of phase aberration of an array transducer. As the interest increases in the application of one and a half-dimensional (1.5-D) and two-dimensional (2-D) array transducers to improving image quality and three-dimensional (3-D) imaging, the need of simple, fast, and sufficiently accurate algorithms for real-time time delay estimation becomes exceedingly crucial. In this paper, we present an adaptive time-delay estimation algorithm which minimizes the problem of noise sensitivity associated with the one bit correlation while retaining simplicity in implementation. This algorithm converts each sample datum into a two bit representation including the sign of the sample and an adaptively selected threshold. A bit pattern correlation operation is applied to find the time delay between two engaged signals. By using the criterion of misregistration as an indicator, we are able to show that the proposed algorithm is better than one bit correlation in susceptibility to noise level. Analytical results show that the improvement in reducing misregistration of the two bit correlation over its counterpart is consistent over a wide range of noise level. This is achieved by an adaptive adjustment of the threshold to accommodate signal corruption due to noise. The analytical results are corroborated by results from simulating the blood as a random distribution of red blood cells. Finally, we also present a memory-based architecture to implement the two bit correlation algorithm whose computation time does not depend upon the time delay of the signals to be correlated     

汽车鸣笛声定位系统仿真

提出了一种汽车鸣笛声定位系统用以协助执法部门锁定违章鸣笛车辆。该系统采用分布在立体空间的五元麦克风阵列,整个定位系统由三个模块组成。首先根据鸣笛声的频域特性检测出信号,然后基于广义互相关理论计算到达各麦克风信号的时延差,并通过对多次估计求均值提高时延估计的准确性。最后利用闭式球形插值算法得到声源位置信息。仿真结果表明该系统可以有效地定位汽车鸣笛声,为实际的系统实现提供了一次初步探讨。     

A spline-based algorithm for continuous time-delay estimation using sampled data

Time delay estimation (TDE) lies at the heart of signal processing algorithms in a broad range of application areas, including communications, coherent imaging, speech processing, and acoustics. In medical ultrasound for example, TDE is used in blood flow estimation, tissue motion measurement, tissue elasticity estimation, phase aberration correction, and a number of other algorithms. Because of its central significance, TDE accuracy, precision, and computational cost are of critical importance. Furthermore, because TDE is typically performed on sampled signals-and delay estimates are usually desired over a continuous domain-time delay estimator performance should be considered in conjunction with associated interpolation. In this paper we present a new time-delay estimator that directly determines continuous time-delay estimates from sampled data. The technique forms a spline-based, piecewise continuous representation of the reference signal then solves for the minimum of the sum squared error between the reference and the delayed signals to determine their relative time delay. Computer simulation results clearly show that the proposed algorithm significantly outperforms other algorithms in terms of jitter and bias over a broad range of conditions. We also describe a modified version of the algorithm that includes companding with only a minor increase in computational cost.     

自适应时延估计在蛙人超短基线定位中的应用

蛙人超短基线(Ultra-Short Base Line, USBL)定位设备实现浅水环境下高精度定位的关键技术之一是精确的时延估计。由于自适应时延估计方法具备环境自适应能力强的特点,文章将混合调制的拉格朗日直接时延估计方法应用于蛙人USBL定位时的高精度测向上,它可以在信标信号中心频率已知的情况下将小数时延滤波器调制到信号中心频率处,以较低的阶数提供更高的时延估计精度;并根据USBL阵型和信号自身的特点,对混合调制的拉格朗日直接时延估计在低信噪比下的具体使用模式进行了探讨和仿真验证;结果显示,所采用的自适应时延估计方法可以在中低信噪比下提供1°～3°的定位测向精度。     

Sweep signal time-delay estimation with time-frequency cross-correlation algorithm

Sweep signal is usually employed as a source signal in active detection such as radar and sonar. Since the frequency spectrum of the sweep signal varies against time, a novel algorithm, namely a time-frequency cross-correlation (TFCC) algorithm based on wavelet packet transform (WPT), is proposed to estimate the time delay of sweep signal. In this algorithm, the source sweep and the received signals are decomposed with WPT to obtain their time-frequency representations and the TFCC between the source sweep and the received signals is performed. Each reflected sweep in the received signal is converted into a time-frequency correlation peak whose position can indicate its time delay. The TFCC algorithm can suppress ambient noise effectively and improve the performance of sweep extraction and can match more precisely the source and the reflected sweeps to their known time-frequency characters. Numerical experiments were performed to compare the performance of the TFCC algorithm with that of the conventional cross- correlation and phase-data algorithms. The results proved that the TFCC algorithm can extract the reflected sweeps effectively and its performance is better than that of the conventional algorithms.