基于Hanning自卷积窗函数的短时傅里叶变换

针对频谱泄漏对短时傅里叶变换(STFT)产生的影响,将卷积窗函数引入到STFT中.本文研究了STFT的算法、卷积窗函数的定义及特性、Hanning自卷积窗函数的性能,通过实验比较了Hanning窗函数、2阶Hanning自卷积窗函数及4阶Hanning自卷积窗函数的STFT分析性能.仿真结果表明使用Hanning自卷积窗函数提高了STFT的时间和频率分辨率,且随着卷积窗函数阶数的提高分析效果越明显.     

音乐信号的时频分析

对已用于音乐处理上的主要时频（TF）分析方法进行了综述,并描述了它们的应用范围。在Cohen类框架下对技术方法进行检验,有助于新方法的设计和比较。所述内容涵盖了基音同步分析,短时傅立叶变换（STFT）,相位声码器．恒Q变换和小波变换,Wigner分布和模式分布。详细描述了窗函数法的局限性和对稳态假设、无限持续正弦曲线的依赖性。与窗函数法相反,Wigner分布用解析信号定义瞬时频率和功率参数,而模式分布是对优化的Wigner分布的线性变换。应用领域涉及分析、再合成、记谱、分类和识别、可视化。     

基于DTFT的一种低频振动信号相位差测量新方法

现有基于DFT频谱分析的相位差测量方法在测量低频振动信号的相位差时，精度明显下降，甚至无法测量，其主要原因之一是忽略了频谱中的负频率成分贡献。基于离散时问傅里叶变换（DTFT），提出了一种计及负频率影响的低频振动信号相位差测量方法，分别给出了加矩形窗和加汉宁窗对应的相位差计算公式。仿真及实测结果表明，在无噪声背景下，该方法具有很高的精度，误差接近双精度运算的下限；在噪声背景下，其测量精度仍高于传统的DFT频谱分析方法。提出的测量方案简单实用，特别适用于频率很低或接近奈奎斯特频率的振动信号的相位差测量。     

电力参数间谐波测量算法的研究

DFT的泄漏和栅栏效应不能测量电力系统的间谐波和分析电力系统谐波误差。一般通过加窗插值DFT能测量间谐波,然而需要构造窗函数使得分析及实际运行变得复杂。本文通过简单的频域变换结合多谱线插值并且给出了一种相应的算法,仿真结果表明该方法运行简单便于分析,精度高,因此为电力系统谐波分析提供了一种满意的工具。     

基于多特征融合的乐器声品质评价方法研究

在乐器声品质研究中,以往研究主要是从乐器的结构以及物理特性出发,忽略了乐音本身的重要性以及客观评价测量的不易实现性.为此,提出了一种基于多特征融合的乐器声品质评价方法.通过主观评价法获取乐器琵琶的声品质评价结果,建立具有主观评价的乐音信号库作为实验对象;提取乐音信号的相关系数(CC)、常数Q变换(CQT)和梅尔频率倒谱系数(MFCC)作为单一特征输入,以及多特征融合输入;并构建基于BP神经网络的乐器声品质评价模型.实验结果表明,该方法可以很好地应用于乐器声品质评价.     

用双窗法减小FFT谱分析估算误差

本文简述了频谱分析中能量泄漏和栅栏效应误差产生的原因,指出了能量泄漏在克服栅栏效应误差中的作用,并从减小栅栏效应带来的幅值估计误差的角度,设计了一个具有平顶窗的窗函数,分析对比了它和矩形窗函数的性能差异,在此基础上,提出了用矩形窗和本文设计的平顶窗谱窗函数联合进行普分析的双窗函数频谱估计算法,可在不同降低频谱频率分辨率的条件下,提高其幅值估计精度。     

平面近场声全息中指数滤波器窗函数设计优化

对基于空间二维傅里叶变换法的平面近场声全息算法中指数滤波器窗函数的合理设置进行了分析和优化。根据优化效果给出一定声源频率条件下,指数滤波器窗函数参数合理设置的建议。同时研究了指数滤波器参数最优设置与声源频率之间的对应关系,并寻找出对应不同声源频率的指数滤波器参数的最优设置。结果表明：相比较窗函数陡度系数的取值变化,滤波器截止波数的取值变化对重建效果影响更加明显;当声源频率在100～1 500 Hz之间变化时,随着声源频率的增加,指数滤波器窗函数的陡度系数的最优值逐渐减小,滤波器截止波数的最优值逐渐增大。     

基于现代时频分析技术的地震波时变谱估计

利用短时傅里叶变换（STFT）、Wigner-Ville分布、连续小波变换、S变换以及变复窗口的S变换等现代时频分析技术对地震波进行时变谱估计。推导了基于连续小波变换的小波能量谱的具体表达式,并把近年来提出的变复窗口S变换应用到地震波的时频分析。通过研究表明对于均匀调制非平稳地震波,通过适当选择窗口的长度,利用STFT估计时变谱可得到较好的效果,小波变换与S变换以及变复窗口S变换的估计值在高频段存在能量泄漏现象;各方法估计的时变谱时间边缘函数与目标函数均符合较好;而频率边缘函数,Wigner-Ville分布与目标函数符合得最好,STFT次之,小波变换、S变换以及复窗S变换在高频段存在一定误差。对于频率非平稳性较强的地震波,通过适当地调节窗口参数,利用变复窗口S变换可得到地震波在时频域内有较好分辨率的时变谱;连续小波变换虽然在高频段存在能量泄漏现象,但其在时间域内有较好的分辨率。     

频谱分析中用于相位和频率校正的相位差校正法

提出了一种对连续时域信号分前后两段作傅里叶变换,利用其对应离散谱线的相位差校正出谱峰处的准确频率和相位的新校正方法——相位差校正法,通过窗谱函数的公式还可以校正其幅值,以解决离散频谱分析中由于谱峰谱线没有对正峰顶时所带来的较大误差。该方法原理简单,通用性好,运算速度快,校正精度高,可以在不知道窗谱函数表达式的情况下,直接用其相位差进行求解。仿真研究表明,对单频率成分的频率、相位、幅值进行校正,频率误差小于0.0002个频率分辨率,相位误差小于0.1 度,幅值误差小于0.02% 。     

基于MCSA的异步电机测速系统研究

在满足实时测速的前提下,直接对电机定子电流使用离散傅里叶变换,会带来频谱分辨率差、能量泄漏等问题,从而不能准确获取电机转速信息.为此,提出一种结合快速傅里叶变换、加窗函数、频谱细分和信号识别的综合算法.即将经过调理采样之后的信号加上布莱克曼窗或者凯塞窗,再对其进行复调制高分辨率的快速傅里叶变换,最后有效获取转子信号频率并求出电机速度.实验结果表明,本测速系统测量时间较短且具有较高的测量精度.     

修正倒谱和动态规划的基频估计算法

基音频率是语音信号处理中的一个重要参数。倍频、半频错误以及清浊音判决的可靠性等问题一直是基频估计中的难点问题。在对语音信号的倒谱进行适当修正的基础上,提出了一种高精度的基频估计算法。该算法根据倒谱、短时能量和短时过零率在清音段和浊音段的不同表现,构造了一个清浊音判决函数,大大提高了清浊音判决精度;然后利用动态规划技术进行基频跟踪。在构造代价函数时．充分考虑了基频连续性的影响,从而使该算法既能有效地避免倍频和半频错误,又能体现出基频的自然加倍和减半。通过与现有的几种效果较好的方法进行对比实验,结果表明该算法具有准确率高、基频轨迹平滑的优点,利用该算法得到的基频轨迹基本不需要进行后期平滑处理。     

Time-domain calculation of spectral centroid from backscattered ultrasound signals

Spectral centroid estimation from backscattered ultrasound RF signals is the preliminary step for quantitative ultrasound analysis in many medical applications. The traditional approach of estimating the spectral centroid in the frequency domain takes a long time because discrete Fourier transform (DFT) processing for each RF segment is required. To avoid this, we propose time-domain methods to estimate the spectral centroid in this paper. First, we derive the continuous-time-domain equations for the spectral centroid estimation using Parseval's theorem and Hilbert transform theory. Then, we extend the method to the discrete-time domain to ease the implementation while maintaining the same accuracy as the calculation in the frequency domain. From the result, we observe that it is not practical to apply the discrete-time equations directly, because a high sampling rate is needed to approximate the time derivative in the discrete-time domain. Therefore, we also derive the feasible version of the discrete-time equations using a circular autocorrelation function, which has no constraints on the sampling rate for real RF signals acquired from pulse-echo ultrasound systems. Simulation results using numerical phantoms show that the time-domain calculation is approximately 4.4 times faster on average than the frequency-domain method when the software's built-in functions were used. The average estimation error compared with that of the frequency-domain method using DFT is less than 0.2% for the entire propagation depths. The proposed time-domain approach to estimate the spectral centroid can be easily implemented in real-time ultrasound systems.     

调频初相位补偿的广义Warblet变换时频分析

信号时频分析的长时间窗时频分析法通常可提高输出信噪比和频率分辨率,但对于调频信号,会降低线谱时频能量聚集度并影响瞬时频率估计。对于调频信号广义Warblet变换(Generalized Warblet Transform,GWT),具有较短时傅里叶变换(Short Time Fourier Transform,STFT)更优的时频分析性能,但在长时间窗分析时,调频初相位估计误差会使算法性能下降甚至失效。针对该问题,提出调频初相位补偿的GWT(Frequency Modulation Initial Phase CompensationGWT,FMIPC-GWT)时频分析方法。在调频参数估计时将一半时间窗长所经过的相位补偿到调频初相位中,提高调频参数估计的准确性以增加瞬时频率估计精度。仿真和实验数据验证了,相比STFT法和GWT法,FMIPC-GWT法对于非线性调频信号时频分析性能更优。FMIPC-GWT法在调频信号线谱检测与瞬时频率估计等方面具有应用前景。     

复音音调提取及其在目标识别中的应用

提出了基于自相关原理提取音调的具体方法和步骤,以乐音和舰船辐射噪声为例,采用3层BP神经网络目标分类器进行仿真实验。仿真结果证明音调是有效的识别特征,且音调的识别率与目标辐射声信号的频率特性的可分性有关。频率特性的可分性越高,则目标识别率越高;当可分性不高时,可以将音调和其他主观量结合共同作为识别特征。实验表明,将特性响度作为目标特征,与音调特征结合起来,目标识别率得到进一步提高。     

水声信号多参量估计的PWVD改进方法

Pseudo Winger-Ville分布（PWVD）是处理非平稳信号的有力工具,因其具有较高的时频聚集性更适合于实时处理,且选择合适的窗函数可降低旁瓣大小,提高分辨率。传统的PWVD可以对信号瞬时频率进行无偏估计,但其频率分辨率和时间分辨率不能同时兼顾;提出了一种插值PWVD法可克服此不足,极大地减小了频谱泄漏的负面影响,有效提高了频率估计与时间估计的精度。仿真和实测数据的分析结果验证了插值PWVD法的可行性与有效性。     

An approach for the Pasternak elastic foundation parameters estimation of beams using simulated frequencies

As a first attempt, a mix of differential quadrature (DQ) and simplex (S) method for the Pasternak elastic foundation parameters estimation of beams using simulated frequencies is presented. The method is applied for the parameters estimation of functionally graded (FG) beams. The first-order shear deformation theory is employed to derive governing equations of FG beam on the Pasternak elastic foundation. The equations are discretized by utilizing the DQ method and frequencies of the beam are calculated. Then, the simulated frequencies are obtained by applying random error to the calculated frequencies. The simulated frequencies are used as input data for estimating parameters of the problem. An objective function as a root-mean-square error between the calculated and simulated frequencies is defined. The DQ method and simplex technique as a classical optimization technique are coupled to minimize the function via finding the best foundation parameters, iteratively. Some examples are solved to show applicability, robustness and accuracy of the mixed method for elastic foundation parameters estimation of the beam. Also, it has been found that using only the first simulated frequency of the beam cannot give correct parameters of the foundation.     

A Simple Method for Design of Adaptive Filters for Sinusoidal Signals

Filtering of input signals in algorithms for measurement of power system electrical parameters is very important. Filters are used to minimize the noise effect and eliminate the presence of higher order harmonics. In addition to that, a number of measurement algorithms apply orthogonal signal components obtained by two orthogonal finite-impulse response filters. The frequency response of the filters must have nulls at the higher order harmonic frequencies that are expected to be present in the signal and must have a unity gain at the main harmonic frequency. In the case of a time-varying frequency, the filter parameters have to be adapted during frequency estimation. In this paper, a simple method for online design of digital filters for sinusoidal signals is proposed. It is based on closed-form solutions for calculating filter coefficients. A simple linear algorithm for frequency estimation was used, and a derived algorithm for online adaptation of the filter coefficients is computationally very efficient. The number of subsections in the cascade and data window lengths can also be changed, depending on the frequency variations during measurement.      

Optimal training design for linearly time-varying MIMO/OFDM channels modelled by a complex exponential basis expansion

Time- and frequency-selective fading of propagating channels degrades the performance of multiple-input multiple-output/orthogonal frequency-division multiplexing (MIMO/OFDM) systems extensively by introducing double convolutions in both time-domain and frequency- domain. The author addresses the problem of linearly time-varying (LTV) channel estimation of MIMO/OFDM systems. First, the time-varying coefficients of the LTV channel are modelled by complex exponential basis expansions. Secondly, the LTV channel is estimated and the optimal pilot symbols are derived following the minimum mean square error criterion. It is shown that the optimal pilot strategy is to group consecutive pilot tones together as a pilot cluster and to distribute uniformly all pilot clusters in frequency-domain. In addition, the LTV channel estimation is further improved by imposing a window function on received signals. Finally, through simulations, it is shown that the new channel estimator can provide a considerable performance improvement in estimating MIMO-LTV channels, especially for the rapidly time-varying channel of a large Doppler frequency.     

Estimation of ultrasound attenuation from broadband echo-signals using bandpass filtering

A problem with video signal analysis for estimating frequency-dependent ultrasonic attenuation is that relative echogenicity versus depth curves are distorted when broadband pulses are used. In this correspondence, we present results that demonstrate improved accuracy of attenuation estimates computed from B-mode or envelope data derived after narrowband (1 MHz BW) filtering at different frequencies around the center frequency of the broadband echo signal. Based on the premise that transducer center frequencies are selected in part on penetration or imaging depth requirements, simulation and experimental results for a typical ultrasound imaging system show that narrowband video signal analysis for frequencies lower than or at the center frequency of the broadband pulse provide unbiased attenuation estimation over this depth. Filtered signals above the center frequency of the pulse yield accurate results only at shallow depths.