一种基于光纤激光水听器的PGC改进解调算法

基于分布反馈式（DFB）光纤激光水听器，针对传统相位生成载波（PGC）解调方案中的不足，提出了一种新型的单路微分相除（PGC-SDD）解调算法。与传统的微分交叉相乘（PGC-DCM）和反正切（PGC-Arctan）解调算法相比，该方法仅需要双通道的1路信号作微分处理，以较少的运算步骤和计算量达到解调信号的目的。PGC-SDD算法能够更有效地应对环境引起的光强扰动和调制深度引起的谐波畸变，使解调结果更接近待测信号。对3种解调算法进行仿真和实验验证，结果表明:采用PGC-DCM解调其信噪比约12.4 dB，PGC-Arctan算法的信噪比约13.9 dB，PGC-SDD算法的信噪比达到了17.5 dB。     

采用3×3耦合器的分布反馈式光纤激光传感器解调技术

为了实现分布反馈式光纤激光传感器(DFB FL)大动态范围、稳定解调,建立了基于3×3耦合器的迈克尔逊干涉仪解调系统。对该系统所采用的对称解调算法(NPS)和反正切解调算法进行了深入研究。首先,介绍了基于3×3耦合器解调算法的原理及耦合器不对称时的调校方法。接着,对干涉仪所需最小非平衡路径长度的选取与系统强度噪声、激光器频率噪声的关系进行了分析。最后,针对NPS算法与反正切算法最大可解调信号幅度进行了分析对比,并研究了微分器对对称解调方法解调范围的影响。实验结果表明:NPS算法动态范围高于反正切算法,微分器的幅频特性不理想会减小解调动态范围。在采样频率为125 k Hz、信号频率为1 k Hz、干涉仪非平衡路径为100 m时,NPS算法与反正切算法的动态范围分别达到96 d B和90 d B。用解调前调校的方法,基于3×3耦合的解调方法动态范围大,能够实现稳定解调,满足工程应用要求。     

改进式相位生成载波调制解调方法

在对调制方法、混频原理、调制深度取值等问题深入研究的基础上,为得到更高的抗干扰能力,结合相位生成载波调制解调技术与微分交叉相乘解调算法,引入除法运算,提出一种改进的相位生成载波解调算法,以抑制光强扰动对解调结果的影响,并与传统的微分交叉相乘解调算法在光源产生低频干扰时的解调效果对比,进行仿真实验.对比信号源分别为20 Hz与200 Hz时的解调效果,结果表明,在光源有1Hz低频扰动的情况下,在高频信号解调的结果中,改进的解调算法可以准确地实现对待测信号的解调,解调结果与传统方法相比具有更高的线性度.     

基于声传播算子的声源定位实验模拟方法研究

声传播算子是一种高效的时域声场计算方法,它能够很方便地计算出给定系统参数下任意时刻任意位置的声场变化情况,本文采用这种方法计算所得的二维房间声场信息进行传声器阵列的声源定位仿真实验。计算结果表明,用该方法获取的阵列数据能有效地应用于阵列信号处理算法中,准确地估计出初始高斯脉冲声源的方向。声传播算子声场计算方法能为传声器阵列声源定位的实验提供方便,使得传声器阵列声源定位算法在不同混响时间的鲁棒性实验研究变得更加简捷。     

干涉型光纤传感器的消偏振衰落技术研究

本文对消除干涉型光纤传感器信号偏振衰落的偏振分集接收PDR(Polarization Diversity Receiver)技术进行了理论分析.通过三态PDR方式,对输出的最大有效幅度信号进行选取,能够避免传输光偏振态变化导致干涉信号完全衰落的现象,使干涉信号有效幅度在一定范围内变化.采用基于反正切计算的相位生成载波PGC(Phase Generated Carrier)解调技术的相位测量结果不受由于偏振衰落导致干涉信号有效幅度变化的影响.提出结合三态PDR方式和基于反正切计算的PGC解调技术消除偏振衰落问题的影响,实现干涉型光纤传感器中相位信号的理想解调.     

空气中声源激发的浅海水下声场传播实验研究

为了了解空气中声源激发的水下声场传播特性,对此开展实验研究.2010年在南海北部海域进行了一次空气中声源激发水下声场实验,在水下成功地接收到远至4 km处大功率扬声器在空中发射的脉冲声信号.本文通过分析实验数据,研究空气中声源激发的浅海水下声场传播特点.针对接收阵拾取的声传播信号信噪比低的特点,综合利用脉冲压缩以及波束...     

光纤水听器相位生成载波解调非线性因素的抑制方法

为了消除非线性因素对相位生成载波解调结果的干扰,降低相位生成载波解调过程中低通滤波器的频响特性对相位生成载波解调结果的影响,该文提出了一种基于扩展卡尔曼滤波椭圆参数估计的相位生成载波微分交叉相乘解调方法。该方法在综合考虑各种非线性因素对相位生成载波微分交叉相乘解调结果的影响的基础上,对传统的相位生成载波微分交叉相乘解调过程进行了改进。经过计算机仿真和实验验证,该文所提出的扩展卡尔曼滤波微分交叉相乘方法能够有效地抑制非线性因素对相位生成载波解调结果的干扰,相比传统的相位生成载波微分交叉相乘方法,其信噪比提高了35.0 d B,总谐波失真降低了30.7 dB,信噪谐波比提高了31.0 dB,且扩展卡尔曼滤波微分交叉相乘方法受解调过程中低通滤波器频响特性的影响较小。     

基于脉冲声分离的刚性球散射声控制

提出了一种基于脉冲声的三维空间中刚性球散射声分离方法,并利用前馈、固定系数控制方式对分离出的散射声进行有源控制,抑制散射声强度,实现了刚性球散射体在观测点处“声学不可见”。该方法利用脉冲信号作为初级噪声,通过有无刚性球时传声器采集脉冲信号的差值确定散射声大小,实现散射声与声源直达声的分离。对分离出的散射声进行多通道有源控制以验证该文所提分离方法及控制系统的有效性。实验结果表明,700~1000 Hz范围内,有源控制开启后,双通道散射声的平均降噪量大于5 dB,多通道散射声的平均降噪量大于8 dB,且误差传声器处采集的残余声场与无刚性球时采集的初级声场信号波形基本一致,实现了刚性球散射体在误差传声器处“声学不可见”。此外,参考传声器布放位置的选取问题也在该文做了详细讨论。     

声脉冲法空间电荷测量系统的研究

气体中空间电荷的分布与电晕放电的机理紧密相关, 获取电晕放电过程中空间电荷分布对深入研究电晕放电起始、自持过程有着重要作用, 但是如何准确获得电晕放电过程中的空间电荷分布一直是国际上尚未解决的难题. 本文基于声脉冲法提出一种电场信号解耦算法, 推导了空间电荷在声场中被调制产生的电场信号与声脉冲信号和空间电荷密度之间的数值关系, 讨论了不同测量情况下声发射系统的设计要求; 搭建了一套可用于实时测量针板电极电晕放电空间电荷分布的非接触式测量系统, 该系统主要包括声脉冲发生模块、空间电荷模块及电场信号解耦算法模块. 运用该系统实现了声脉冲激发作用下电场信号的测量, 通过提出的电场信号解耦算法得到了空间电荷密度, 对其测量结果与电晕电流法测量结果进行比较, 验证了电场信号解耦算法的有效性. 该算法可以应用于空间电荷一维、二维和三维测量系统中.     

超声相控阵距离幅度曲线计算方法

超声相控阵偏转聚焦声场能量分布不均匀,造成不同位置的缺陷回波幅度差异,针对该缺陷量化问题,提出超声相控阵系统距离幅度曲线理论计算方法。在非近轴近似声场模型的基础上,结合缺陷散射模型来建立相控阵系统测量模型,再利用该方法来预测不同位置横通孔缺陷的回波信号,进而获得信号幅度随位置的变化曲线。分析对比了不同偏转聚焦声束理论计算结果与实验测量结果,两者幅度在焦点附近具有较好的一致性,偏差在5%以内,验证了超声测量模型计算方法的有效性。      

Wide-band dereverberation method based on multichannel linear prediction using prewhitening filter

Several dereverberation algorithms have been studied. The sampling frequencies used in conventional studies are typically 8–16 kHz because their main purpose is preprocessing for improving the intelligibility of speech communication and articulation for automatic speech recognition. However, in next-generation communication systems, techniques to analyze and reproduce not only semantic information of sound but also more high-definition components such as spatial information and directivity will be increasingly necessary. To decompose these sound field characteristics with high definition, a dereverberation algorithm that is useful at high sampling frequencies is an important technique to process sound that includes high-frequency spectra such as musical sounds. The LInear-predictive Multichannel Equalization (LIME) algorithm is a promising dereverberation method. Using the LIME algorithm, however, a dereverberation signal cannot be solved at high sampling frequencies when the source signal is colored, such as in the case of speech and sound of musical signals. Because the rank of the correlation matrix calculated from such a colored signal is not full, the characteristic polynomial cannot be calculated precisely. To alleviate this problem, we propose preprocessing of all input signals with filters to whiten their spectra so that this algorithm can function for colored signals at high sampling frequencies.     

光纤水听器解调技术研究进展

光纤水听器是水声研究领域的关键技术,在实际应用中具有显著优势,广泛应用于AUV导航定位、资源勘探、海水警戒等场景.光纤水听器将声信号转化为光信号传播,在接收信号时就需要对光信号进行解调提取出声压信号.总结了几种常用的信号解调算法及关键技术,包括PGC解调法、3×3耦合器解调法和外差解调法;介绍了各种解调算法的基本原理和...     

An improved arctangent algorithm based on phase-locked loop for heterodyne detection system

We present an ameliorated arctangent algorithm based on phase-locked loop for digital Doppler signal processing,utilized within the heterodyne detection system. We define the error gain factor given by the approximation of Taylor expansion by means of a comparison of the measured values and true values. Exact expressions are derived for the amplitude error of two in-phase quadrature signals and the frequency error of the acousto-optic modulator. Numerical simulation results and experimental results make it clear that the dynamic instability of the intermediate frequency signals leads to cumulative errors, which will spiral upward. An improved arctangent algorithm for the heterodyne detection is proposed to eliminate the cumulative errors and harmonic components. Depending on the narrow-band filter, our experiments were performed to realize the detectable displacement of 20 nm at a detection distance of 20 m. The aim of this paper is the demonstration of the optimized arctangent algorithm as a powerful approach to the demodulation algorithm, which will advance the signal-to-noise ratio and measurement accuracy of the heterodyne detection system.     

不同采样方式下光纤布喇格光栅反射谱寻峰算法的分析

对光纤布喇格光栅反射谱的三种寻峰算法(三次样条插值数值微分法、高斯-多项式拟合法和高斯拟合法)进行了分析和比较;相同采样情况下,得出了高斯拟合法确定的峰值准确度最高的结论;采样点数为250的均匀采样中,三种算法寻峰结果对实际值的误差分别为:3.4pm、13.0 pm和2.6 pm.引入了非均匀数据采集的寻峰思路,分别应用三种寻峰法对实际光栅的反射谱分别进行均匀采集和非均匀采集.结果表明,对于相同寻峰方法在非均匀数据采集下获得的峰值更精确,采样点数为250时,高斯拟合法寻峰时非均匀采集对应的误差比均匀采集减少了38.46％.     

不同采样方式下光纤布喇格光栅反射谱寻峰算法的分析

对光纤布喇格光栅反射谱的三种寻峰算法(三次样条插值数值微分法、高斯-多项式拟合法和高斯拟合法)进行了分析和比较;相同采样情况下,得出了高斯拟合法确定的峰值准确度最高的结论;采样点数为250的均匀采样中,三种算法寻峰结果对实际值的误差分别为: 3.4 pm、13.0 pm和2.6 pm.引入了非均匀数据采集的寻峰思路,分别应用三种寻峰法对实际光栅的反射谱分别进行均匀采集和非均匀采集.结果表明,对于相同寻峰方法在非均匀数据采集下获得的峰值更精确,采样点数为250时,高斯拟合法寻峰时非均匀采集对应的误差比均匀采集减少了38.46%.     

Ultrasonic velocity and amplitude characterization of magnetorheological fluids under magnetic fields

Variations in velocity of sound and amplitude of the signal of a commercial magnetorheological fluid under different magnetic fields are studied experimentally. Different factors such as orientation, uniformity, geometry and intensity of the magnetic field are investigated. An increase in the change of MR fluid acoustical properties is obtained when the magnetic field intensity is risen. In addition, these properties show an opposite behavior when a magnetic field is applied parallel or perpendicular to the ultrasound propagation. Experiments using an electromagnet and permanent magnets as the source of magnetic field are also compared. Properties such as anisotropy in sound velocity and amplitude make these materials interesting regarding applications.     

无线传感网中基于质心的高效坐标压缩算法

鉴于无线传感网中有限的通信带宽和计算资源, 传感器节点传送压缩后的数据对于系统节能具有十分重要的意义. 本文提出一种利用反正切函数构建的非线性坐标压缩策略来降低系统的数据传输量. 得益于反正切函数的非线性特点, 在质心附近压缩率低, 离质心越远, 压缩率越高. 仿真结果显示, 所提算法特别适应于在质心附近具有高频分量的信号类型. 所提算法具有以下几个特点: 1)采样间隔是非均匀的; 2)可以在采样前实现压缩, 类似于压缩感知; 3)计算复杂度低, 算法简单易实现, 在实际应用中更具有优势. 关键词： 无线传感器网络 坐标压缩 非线性 质心     

Performance of time- and frequency-domain binaural beamformers based on recorded signals from real rooms

Extraction of a target sound source amidst multiple interfering sound sources is difficult when there are fewer sensors than sources, as is the case for human listeners in the classic cocktail-party situation. This study compares the signal extraction performance of five algorithms using recordings of speech sources made with three different two-microphone arrays in three rooms of varying reverberation time. Test signals, consisting of two to five speech sources, were constructed for each room and array. The signals were processed with each algorithm, and the signal extraction performance was quantified by calculating the signal-to-noise ratio of the output. A frequency-domain minimum-variance distortionless-response beamformer outperformed the time-domain based Frost beamformer and generalized sidelobe canceler for all tests with two or more interfering sound sources, and performed comparably or better than the time-domain algorithms for tests with one interfering sound source. The frequency-domain minimum-variance algorithm offered performance comparable to that of the Peissig-Kollmeier binaural frequency-domain algorithm, but with much less distortion of the target signal. Comparisons were also made to a simple beamformer. In addition, computer simulations illustrate that, when processing speech signals, the chosen implementation of the frequency-domain minimum-variance technique adapts more quickly and accurately than time-domain techniques.     

Decoding neural events from fMRI BOLD signal: A comparison of existing approaches and development of a new algorithm

Neuroimaging methodology predominantly relies on the blood oxygenation level dependent (BOLD) signal. While the BOLD signal is a valid measure of neuronal activity, variances in fluctuations of the BOLD signal are not only due to fluctuations in neural activity. Thus, a remaining problem in neuroimaging analyses is developing methods that ensure specific inferences about neural activity that are not confounded by unrelated sources of noise in the BOLD signal. Here, we develop and test a new algorithm for performing semiblind (i.e., no knowledge of stimulus timings) deconvolution of the BOLD signal that treats the neural event as an observable, but intermediate, probabilistic representation of the system's state. We test and compare this new algorithm against three other recent deconvolution algorithms under varied levels of autocorrelated and Gaussian noise, hemodynamic response function (HRF) misspecification and observation sampling rate. Further, we compare the algorithms' performance using two models to simulate BOLD data: a convolution of neural events with a known (or misspecified) HRF versus a biophysically accurate balloon model of hemodynamics. We also examine the algorithms' performance on real task data. The results demonstrated good performance of all algorithms, though the new algorithm generally outperformed the others (3.0% improvement) under simulated resting-state experimental conditions exhibiting multiple, realistic confounding factors (as well as 10.3% improvement on a real Stroop task). The simulations also demonstrate that the greatest negative influence on deconvolution accuracy is observation sampling rate. Practical and theoretical implications of these results for improving inferences about neural activity from fMRI BOLD signal are discussed.