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.      

Use of digital filters to determine the frequency of a sinusoidal signal

An interference-protection-optimum method of measuring the frequency of sinusoidal signals using digital filters, based on the root mean square deviation criterion, is investigated. An algorithm for the digital processing of the input signal is proposed, and a model of a digital filter for achieving this and the principle of the construction of a digital measuring instrument for determining the parameters of a sinusoidal signal are proposed.Translated from Izmeritel'naya Tekhnika, No. 8, pp. 60–63, 1994.     

基于动网格技术的正弦压力研究

正弦压力是一种典型的周期性动态压力标准信号,而正弦压力发生器正是产生正弦压力信号的关键。以转盘式正弦压力发生器为例,采用动网格技术、Reynolds-averaged Navier-Stokes方程和Spalart Allmaras湍流模型,对不同的压力发生腔尺寸和工作频率进行三维瞬态流场的数值仿真。结果表明:工作频率不变时,正弦压力动静幅值比和谐波失真度随腔体宽度增大而减小,压力均值随腔体宽度增大而增大;腔体宽度不变时,正弦压力动静幅值比随工作频率增大而减小,压力均值随工作频率增大而增大,谐波失真度呈现先减小后增大的规律;通过动网格数值仿真技术,能有效实现正弦压力研究中各种模型结构的效果预测。     

Deconvolution from instrumental devices and source effect inacoustic experiments

Two methods to deconvolve experimental data from the distortions introduced by instrumental devices or source effects are presented. Considering a total acquisition system (emission-reception line, amplifier, pre-amplifier) as a global experimental filter, we can define its properties (module and phase) experimentally from the generation of a family of source signals dilated in time. The estimation of this filter allows the deconvolution of the recorded output signal. The first approach is based on the simple reconstruction formula of the continuous wavelet transform (CWT). The second method is based on the construction of a normalized family of a finite number of specific filters, independent of the frequency range used. In both cases, experimental results in an acoustic tank are presented. We show that after deconvolution, the source signal is correctly reconstructed from the recorded output signal and the global instrumental filter     

A statistical approach to estimate state variables in flow-accelerated corrosion problems

Sequential Monte Carlo or Particle Filter Methods have been widely used to deal with sequential Bayesian inference problems in several fields of knowledge. This technique involves approximation of probability sequences distributions of interest, by means of a large set of random samples, i.e. particles that are propagated along time with a simple Sampling Importance distribution, SI. A re-sampling technique is also used to improve the predictive probability. In this study, a methodology is proposed: apply the Bayesian filters to a state estimation problem involving the corrosion amount-time in a contraction–expansion geometry with the aid of Computational Fluid Dynamics to improve the accuracy of the results. The following filters were applied and compared: Sampling Importance Re-sampling filter (SIR filter) and Auxiliary Sampling Importance Re-sampling filter (ASIR filter). The corrosion model adopted is based on a double resistance due to the oxygen diffusion towards the wall through the hydrodynamic boundary layer and the oxide layer. Mass loss data over time are obtained from the literature to compare corrosion rates. Also, the influence of the corrosion products in rates of corrosion is discussed . Best results in corrosion damage estimation were obtained using the ASIR filter.     

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.     

Periodic orbit analysis of two dynamical systems for electrical engineering applications

Mathematical properties of two nonlinear adaptive filters for electrical engineering applications are presented. These filters are designed to extract a desired sinusoidal component of a given periodic signal and estimate its amplitude, phase angle and frequency. Two sets of non-autonomous ordinary differential equations govern the dynamics of the filters. It is shown that each of the filters possesses a unique and stable periodic orbit. The averaging theorem is used to prove the uniqueness and stability of the periodic orbit of one of the filters. Uniqueness and stability of the periodic orbit of the other filter are proven using the Poincaré map theorem. Computer simulations and numerical results are presented to provide numerical verification of the theoretical proofs, and finally experimental results of the laboratory implementation of the filters are presented.     

Noncoherent spectral analysis of ADC using filter Bank

The aim of this paper is to propose a new spectral analysis method for an on-chip analog-to-digital converter (ADC) dynamic test. ADC characterization by spectral analysis has traditionally been done with discrete Fourier transform. This method imposes restrictions to optimize results; one of these is coherent sampling. Recently, some filter structures have been used for spectral analysis of a sinusoidal signal corrupted by harmonics and noise. In this paper, we present a new filter bank structure used for decomposing a signal into its main spectral components. The main application examined is ADC spectral parameter estimation, like signal-to-noise and distortion ratio, signal to noise ratio, total harmonic distortion, and so on, in noncoherent sampling. Computer simulations are used to demonstrate the performance of the proposed filter bank scheme. This structure is a promising built-in self-test (BIST) approach for ADC ICs.     

Energy-based effective length of the impulse response of arecursive filter

A measure for the effective length of the impulse response of a stable recursive digital filter based on accumulated energy is proposed. The new measure finds applications in several fields of digital signal processing, including estimation of the extent of attack transients for filters with dynamically varying inputs, elimination of transients in variable recursive filters, and design and implementation of linear-phase IIR systems. A general definition and a simple algorithm to evaluate it are introduced, and closed-form expressions are derived for first and second-order all-pole filters. The effect of zeros on the effective length is analyzed. An upper bound for the effective length of higher-order filters is derived using results for low-order filters, which is illustrated for classical digital lowpass filters. The use of the measure is demonstrated with examples of implementation of linear-phase IIR systems and estimation of transients in variable IIR filters     

Optimal Estimation of the Acceleration of a Car Under Performance Tests

In this paper, a recursive least-squares lattice (RLSL) adaptive filter was used to carry out the optimal estimation of the relevant signal coming from an accelerometer placed in car under performance tests. Here, the signal of interest is buried in a broadband noise background where we have little knowledge of the noise characteristics. In addition, due to the fact that the noise and the relevant information sometimes share the same or a very similar frequency spectrum, it is very difficult to cancel the noise that corrupts the relevant information without causing that information to deteriorate. The results of the experiment are satisfactory and, in order to compare classical filtering with optimal adaptive filtering, the signal coming from the accelerometer was also filtered by using a third-order lowpass digital Butterworth filter. The results of comparing the aforementioned filters show that the optimal adaptive filter is superior to the classical filter. Here, a significant improvement of 22.4 dB in the signal-to-noise ratio (SNR) at the RLSL adaptive filter output was achieved. However, the improvement in the SNR at the Butterworth filter output was 6.1 dB, which shows very clear that the optimal adaptive filter performs much better than the classical one     

Active Filters for Measuring Time-Domain Frequency Stability

The instrumentation for measuring the filter variance is simple and low cost. An assembly of the test set known as PCC-1 for evaluaig the time-domain frequency stability of signal generators was developed in the Chengdu Institute of Radio Engineering. It also includes specially designed active filters for measuring the filter variance. Experimental results are presented which show that the high-pass filter variance is a good estimation of the Allan variance.     

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.     

时域模态参数辨识的状态滤波算法

本文针对结构模态参数辨识的噪声干扰问题,采用一种状态滤波方法削弱测量噪声的影响,避免了传统Kalman 滤波法中对系统模型的较高精度要求,并与特征系统实现算法(ERA)相结合,有效地克服了ERA 方法在信噪比较低的情况下对非零奇异值判断的困难,并更精确地识别出结构的模态参数。     

用于增强超声检测信号的分离谱处理性能分析

若干研究表明，分离谱处理方法是增强粗晶材料超声探伤信号的有效手段，但这种处理方法成功与否受诸如滤波器无中滤波器数等信号处理参数的影响，本文通过实验探讨了信号处理方式与参数的选取问题，实验数据取是奥氏体钢。     

On the realization of multimemory block structure digital adaptive filter using distributed arithmetic

Abstract

In this paper, the hardware complexity and system performance of the two multimemory structures (parallel and cascade) of adaptive digital filters are studied. These structures are based on the use of a distributed arithmetic technique without a multiplier in the realization of the filter function. The results from computer simulations and implementations by an MC 68000 16‐bit microprocessor demonstrate that adaptive filters implemented by these two structures have comparable performances on the average. However, the hardware implementation of the parallel structure is simpler and can achieve higher operation speed with fewer IC components. A prototype of a 16‐tap parallel structure (divided into two blocks) adaptive filter is implemented by Schottky TTL circuits. Experimental result of this adaptive filter for noise cancellation has demonstrated its capability for high speed signal processing.     

Time-Domain Spherical Near-Field Antenna Measurement System Employing a Switched Continuous-Wave Hardware Gating Technique

A time-domain spherical near-field antenna measurement system capable of gating out erroneous signal components, which arise due to multipath propagation in nonideal anechoic chambers, is presented. The developed hardware (HW) gating technique evaluates a switched sinusoidal signal, which is synthesized by an application-specific pulse generator and acquired by either a commercial real-time digitizing oscilloscope or an application-specific equivalent-time sampling receiver developed for this particular purpose. The low-cost measurement system has been optimized for acquisition speed, dynamic range, and resolution. Its operating frequency range covers 1.5–8 GHz, and it is applicable to antennas exhibiting a typical 3-dB bandwidth in excess of 400 MHz. Test measurements of an omnidirectional and a directional antenna, respectively, have been carried out to demonstrate the performance of the novel HW gating technique. It is shown that the HW gating technique can significantly improve the absolute average deviation of the erroneous 3-D far-field pattern.      

Digital Time-Domain Technique for Generating Serial MSK Signal

A digital time-domain technique for the generation of serial minimum shift keying (SMSK) signal is described. Two pairs of synchronized antiphase sinusoidal waveforms, representing the MARK and SPACE signals, are obtained through filtering of digitally generated square waveforms. A synchronous sequential circuit then appropriately multiplexes the four sinusoids to generate the desired SMSK signal. Phase continuity of the SMSK waveform during symbol transitions is inherently maintained. Design procedure and implementation details of this simple digital SMSK generator has been provided.     

Acoustic attenuation estimation for soft tissue from ultrasound echo envelope peaks

A theoretical framework of a modified EP (envelope peak) method is developed. In this method, a wideband echo signal from each A line (one echo sequence) is first filtered in parallel by a bank of narrow-bandpass filters using a split-spectrum processing. The attenuation is then estimated from the EPs of each filtered signal using a narrowband technique. The combination of the split-spectrum processing with the narrowband technique enables the accuracy of the attenuation estimation to be well controlled without the precise measurements of the spectral shape and parameters of the transmitted pulses. On the other hand, the precision of the estimation is still determined by the bandwidth of the original echo signal, and is not affected by the split-spectrum processing. As a result, the modified EP method improves the accuracy of the attenuation estimation while retaining the high precision of the original EP method. Results from phantom experiments supported the theoretical analysis.     

New spin filters for interferometric fringe patterns and grating patterns

The basic spin filter for interferometric fringe patterns is improved and developed into several new versions for different applications. These spin filters can filter off random noise efficiently and have almost no blurring effect and phase distortion for the fringe patterns. First, they find the local fringe tangent direction, and then they apply a one-dimensional low-pass filter on this direction. In this way the spin filters can separate easily and clearly high-frequency noise from a real fringe signal with nearly zero frequency. The new spin filters are suitable not only for various fringe patterns but also for wrapped-phase, line-grating, and cross-grating patterns, which are impossible by common filters.     

混合色噪声背景下的弱信号频率估计：基于互高阶累积量的矩和最小范围—TLS方法

谱估计方法是用于噪声背景下测量微弱正弦信号的一种有效方法。但以往的谱估计都是采用互功率谱估计和高阶自谱估计的各种方法。本在互功率谱估计和高阶统计量理论基础上,提出了互高阶谱估计方法。从理论上建立了基于互四阶累积量的Yule-Walker方程,并在此基础上提出了互高阶谱估计的矩和最小范数－总体最小二乘（TLS）方法。仿真结果表明,在混合色噪声背景下,这两种方法能够有效地抑制噪声,具有良好的频率估计性能。