Active power measurement algorithm for power system signals under non-sinusoidal conditions and wide-range frequency deviations

A simple approach in the design of digital algorithms for active power under non-sinusoidal conditions and wide-range frequency deviations is presented. In order to accurately estimate the fundamental component of the active power, the pure sinusoidal voltage model was assumed. Adaptive finite-impulse-response (FIR) filters are used to decompose out the fundamental components of the input voltage and current signals and to minimise the noise effects. The total active power estimation was carried out by applying adaptive low-pass FIR filters for averaging of the instantaneous power. A simple algorithm for simultaneous frequency and both fundamental and total active power estimation, in a wide range of frequency changes, with benefits in a reduced complexity and computational efforts is obtained. To demonstrate the performance of the developed algorithms, the results of computer simulations and laboratory testing are presented. The algorithm shows a very high level of robustness, as well as high measurement accuracy over a wide range of frequency changes. The algorithm convergence provided fast response and adaptability, thus, this technique provides an accurate active power estimates in about one period. It has been found that the proposed algorithm really meets the needs of online applications.     

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。     

Kalman filter based synchronisation methods

Single- and three-phase synchronisation methods based on optimum filtering theory are proposed. These methods are based mainly on the Kalman filter and are therefore termed Kalman filter-phase locked loop. they explicitly include in the problem formulation the presence of harmonics, voltage unbalance, measurement noise, transients and frequency deviation. such perturbations degrade the performance of many synchronisation structures presented in literature. the formulation presented here makes the synchronisation signals less sensitive to these perturbations. it is also shown that the proposed methods can be helpful by also providing the amplitude, instantaneous phase and frequency of grid voltages that can be useful for the analysis of power quality. furthermore, the Kalman filter provides a way of obtaining the best compromise between transient response and measurement noise rejection for the synchronisation signals. The paper sets out the development of the proposed methods together with the choice of tuning parameters and their physical meaning. simulations and experimental results using a DSP TMS320F2812 are presented to show the effectiveness of the proposed schemes.     

Instantaneous Sequence-Component Resolution of 3-Phase Variables and Its Application to Dynamic Voltage Restoration

A technique for decomposing instantaneous three-phase voltages and currents into sequence components has been described. The technique is based on a recursive weighted least squares procedure. The main advantage of the proposed technique is the use of the same covariance and gain matrices for resolving any number of three-phase variables into their respective sequence components. The proposed technique also estimates the dc offset in the sensed variable, and these are eliminated during the processing. Finally, the proposed technique has been implemented in a digital signal processor (DSP) and has been applied to the open-loop control of a laboratory-scale dynamic voltage restorer.      

Adaptive resonator-based method for power system harmonic analysis

A simple algorithm for the harmonic estimation, in a wide range of frequency changes, with benefits in a reduced complexity and computational efforts is prescribed. This implementation is based on a recently introduced common structure for recursive discrete transforms and contemplated as an implementation of finite-impulse-response (FIR) and infinite-impulse-response (MR) filter transfer functions to reduce computational efforts. This structure consists of digital resonators in a common negative feedback loop. The structure of the estimation algorithm consists of two decoupled modules: the first one for an adaptive filter of input signal with harmonic amplitude and phase calculation, the second one for an external frequency estimation. A very suitable algorithm for frequency and harmonic phasor estimations is obtained. To demonstrate the performance of the developed algorithm, computer-simulated data records are processed. Simulation results show that this algorithm is applicable to detect the harmonic amplitudes of steady-state, varying and decaying sinusoidal signals. It has been found that the proposed method really meets the needs of online applications. This technique provides accurate amplitude estimates in about one period.     

New spectral decomposition based approach for flicker evaluation of electric arc furnaces

A spectral decomposition-based approach is proposed to estimate the light flicker caused by electrical arc furnaces (EAFs) where the system frequency deviates significantly due to the EAF operation. Analytical expressions of the instantaneous light flicker sensation are obtained beginning from a voltage waveform and these expressions are used to obtain a flicker estimation method based on the IEC flickermeter. In the proposed method, the leakage effect of the FFT algorithm due to fundamental frequency variation is reduced by employing a spectral amplitude correction around the fundamental frequency. The eye-brain weighting curve is realised comparing the voltage spectrum with the tabulated normalised IEC flickermeter responses for sinusoidal voltage fluctuations of the IEC standard. The proposed method is tested on both simulated data and field data obtained from three different EAF plants. The comparison with the digital realisation of the IEC flickermeter shows that the method gives satisfactory estimations of both the instantaneous flicker sensation and the short term flicker severity with low computational complexity. The method is especially useful for all other conditions such as disturbances and subsequent system transients where the system frequency deviates without the need for online sampling rate adjustment.     

三相三线制有源电力滤波器谐波检测方法

针对三相三线制有源电力滤波器(Active Power Filter,APF)中的谐波检测,研究了基于瞬时无功理论的谐波电流快速检测算法,结合谐波检测过程中出现的频率混叠现象而搭建的二阶巴特沃斯低通滤波器,共同实现了有源电力滤波器高精度的谐波检测功能.实验结果验证了方法的正确性和可靠性,对于提高APF的补偿性能有很大的帮助.     

Three-phase phase-lock loop for distorted utilities

A novel three-phase phase-locked loop (PLL) structure suitable for phase and angular frequency derivation from distorted ac utility voltages is presented. The proposed PLL has a simple structure; a conventional three-phase PLL followed by a proportional-integral (Pl)-controlled moving average filter together with a phase-locking algorithm. The objective of the proposed technique is to capture the fundamental phase angle and angular frequency of three-phase clean, distorted, balanced or unbalanced ac utilities. The PLL gives fast, accurate angular frequency and phase locking and is robust to utility distortion such as line notching, random noise, voltage imbalance, phase loss, phase imbalance, harmonics, dc offsets and frequency variation. The analysis presented substantiates the immunity of the proposed PLL to unbalanced and distorted utility conditions. The PLL technique is simulated and digital signal processor (DSP)-implemented for a three-phase system to verify the analytical results. The simulated and experimental results, for numerous utility conditions, demonstrate its phase-tracking ability, whereas the conventional technique fails to lock accurately in highly distorted, three-phase grid-connected operation.     

用多项式模型和自适应滤波估计瞬时频率

本文论述了非平稳信号瞬时频率(IF)的估计方法,包括基干最小均方误差(LMS)算法或递推最小二乘(RLS)算法的自适应IF估计方法和基于信号相位多项式模型的IF估计方法。对各种估计方法的估计方差与理论方差下限进行了比较。说明了,基于互Wigner分布的估计方法所需信噪比(SNR)门限值最低。在低SNR、短信号记录的情况下,用多项式相位模型与最大似然法可以获得比较可靠的IF估计。     

An instantaneous amplitude detector for variable frequencythree-phase sinusoidal signals

A fast peak detector for three-phase variable frequency sinusoidal signals is proposed. This new detector fully uses the three-phase characteristic to achieve instantaneous response, frequency and phase sequence independence characteristics. Also, the proposed detector possesses excellent linearity and low sensitivity to phase unbalance and harmonic distortion. Theoretical basis, hardware implementation, and experimental results are all detailed in this context     

On the use of the warblet transform for instantaneous frequency estimation

A new digital signal processing method for instantaneous frequency estimation is proposed here. The attention is mainly paid to signals whose instantaneous frequency trajectories exhibit a periodic evolution versus time. Thanks to an optimized use of the warblet transform, the method assures superior accuracy and resolving capability with respect to other solutions already available in the literature, thus showing itself very attractive in the presence of multicomponent signals characterized by instantaneous frequency trajectories extremely similar and very close to one another. Theoretical notes regarding the warblet transform and its optimized use in the framework of the proposed method are first given. Then, the fundamental steps of the method are described in detail with references to a clarifying example. The results of a number of experiments on emulated and actual signals, aimed at assessing the performance of the method, are finally presented.     

一种新的估计多项式相位信号瞬时频率的参数化时频分析方法

通过多项式非线性核函数取代线性调频小波变换中的线性核函数,提出一种新的参数化时频分析方法：非线性调频小波变换。对瞬时频率是时间任意连续函数的信号而言,选择合适的多项式核特征参数,非线性调频小波变换的时频分布有良好的时频聚集性。应用非线性调频小波变换分析任意阶次多项式相位信号。由于非线性调频小波变换的性能取决于多项式核特征参数,本文还给出非线性调频小波变换的核特征参数估计算法,进一步可实现多项式相位信号的瞬时频率和参量估计。仿真信号验证算法的有效性。     

A Simple Recursive Algorithm for Simultaneous Magnitude and Frequency Estimation

A new approach in the design of digital algorithms for simultaneous local system magnitude and frequency estimation of a signal with time-varying frequency is presented. The algorithm is derived using the maximum likelihood method. The pure sinusoidal voltage model was assumed. The investigation has been simplified because the total similarity to the state of the problem of dc offset and frequency estimation has been noticed. Finite impulse response (FIR) digital filters are used to minimize the noise effect and to eliminate the presence of harmonic effects. The algorithm showed a very high level of robustness, as well as high measurement accuracy over a wide range of frequency changes. The algorithm convergence provided fast response and adaptability. This technique provides accurate estimates in about 25 ms and requires modest computations. The theoretical bases of the technique are described. To demonstrate the performance of the developed algorithm, computer-simulated data records are processed. The proposed algorithm has been tested in a laboratory to establish its feasibility in a real-time environment.     

A simple recursive algorithm for frequency estimation

A new approach to the design of a digital algorithm for local system frequency estimation is presented. The algorithm is derived using the maximum likelihood method. One sinusoidal voltage model was assumed. FIR digital filters used in papers, are used to minimize the noise effect and to eliminate the presence of the harmonics effect. The algorithm showed a very high level of robustness as well as high measurement accuracy over a wide range of frequency changes. The algorithm convergence provided fast response and adaptability. This technique provides accurate estimates with error in the range of 0.005 Hz in about 25 ms and requires modest computations. The theoretical basis and practical implementation of the technique are described. To demonstrate the performance of the developed algorithm, computer simulated data records are processed.     

基于全相位HHT的瞬时频率测量

全相位谱分析能够很好地抑制频谱泄漏、优化系统性能,其关键技术在于对数据进行全相位预处理。将这一技术运用于HHT求解瞬时频率中,提出了一种全相位HHT的瞬时频率测量算法,分析了全相位预处理和HHT求解瞬时频率的算法,将信号首先进行全相位预处理,然后用HHT算法求解信号的瞬时频率。这种算法是一种优化算法性能的算法,不会过多增加系统的计算量。通过对一非平稳信号分析,验证了基于全相位HHT的瞬时频率方差更小,具有更高的精确性。     

A simple analog division scheme

A technique for analog division of two voltage signals that offers high accuracy (0.5%) and relatively good response, especially when the numerator value is nearly constant, is presented. The proposed method is based on the relation of a signal's frequency to its period. The basic idea is to convert input voltage to frequency (through a voltage-controlled oscillator (VCO)) and then produce an output voltage proportional to the period of this signal. The circuit implementing this method is easy to build and may be used in many instrumentation and control applications where the reciprocal of a voltage is needed. Supporting experimental results and performance measurements are also given     

一种基于相关积分的互谱WVD目标方位估计方法

针对水下高速运动目标的被动跟踪问题,将魏格纳-威尔分布(Wigner-Ville Distribution,WVD)算法与互谱法相结合,构建了一种矢量信号处理框架下的目标方位估计方法——基于相关积分的互谱WVD算法。该算法利用了矢量水听器声压通道与振速通道信号的相关特性,首先通过计算两者的互相关函数来提取目标信号的特征信息,然后通过短时积分与傅里叶变换将互相关函数从时域转换到频域,最后在频域提取信号的特征参量,并进行方位估计。仿真研究表明,当目标处于远距离且低速运动时,所提算法的方位估计性能与互谱声强法基本一致;而当目标处于近距离且高速运动时,算法的方位估计性能大大优于互谱声强法。     

Simple and robust analytically derived variable step-size least mean squares algorithm for channel estimation

A new variable step-size least mean squares (VSS-LMS) algorithm for the estimation of frequency selective communications channels is herein presented. In contrast to previous works, in which the step-size adaptation is based on the instantaneous samples of the error signal, this algorithm is derived on the basis of analytical minimisation of the ensemble-averaged mean-square weight error. A very simple rule for step-size adaptation is obtained, using a small number of communication system parameters. This is another significant difference from other proposals, in which a large number of control parameters should be tuned for proper use. The algorithm here proposed is shown to be applicable to both time-varying and time-invariant scenarios. While the lack of a termination rule for step-size adaptation is a common characteristic of other schemes, the algorithm here presented adopts a criterion for stopping the step-size adaptation that assures optimal steadystate performance and leads to large computational savings. A simulation-based performance comparison with other VSS-LMS schemes is provided, including their application to maximum likelihood sequence estimation receivers using per survivor processing (MLSE/PSP). The results show that the algorithm proposed in this work has good performance characteristics and a very low computational cost, specially in the application to MLSE/ PSP receivers. Besides, this algorithm is shown to be robust to changes in the signal-to-noise ratio (SNR).     

A high precision resolver-to-DC converter

A newly developed resolver converter, providing a pseudolinear voltage proportional to the shaft angle, is presented. This converter is based on a new concept involving the absolute values of the demodulated sine and cosine resolver signals together with a dedicated linearization technique. The converter enables instantaneous determination of the mechanical angle with a theoretical error of nonlinearity below 0.011/spl deg/ over the 360/spl deg/ range. The practical performance of this converter is compared to that of a 10/sup 5/ pulses per revolution optical encoder arrangement. The theory of operation, computer simulation, full circuit details, and experimental results are given.     

Novel joint time delay and frequency estimation method

A novel algorithm has been proposed for joint time delay and frequency estimation of sinusoidal signals received at multiple separated sensors. The joint time delay and frequency estimation problem is linked to a trilinear model. Exploiting this link, it derives a trilinear decomposition-based joint time delay and frequency method algorithm. The simulation results reveal that the algorithm performance is close to that of the Cramer?Rao lower bound, and the proposed algorithm even supports small sample sizes.