Analysis of Harmonics in Power Systems Using the Wavelet-Packet Transform

This paper proposes a new algorithm based on the wavelet-packet transform for the analysis of harmonics in power systems. The proposed algorithm decomposes the voltage/current waveforms into the uniform frequency bands corresponding to the odd-harmonic components of the signal and uses a method to reduce the spectral leakage due to the imperfect frequency response of the used wavelet filter bank. This paper studies the selection of the mother wavelet, the sampling frequency, and the frequency characteristics of the wavelet filter bank for the two most common wavelet functions used for harmonic analysis and compares the performance of the proposed method with the results obtained using the discrete Fourier transform (DFT) analysis and the harmonic-group concept introduced by the International Electrotechnical Commission (IEC) under different measurement conditions.     

采用预充开关运放的低功耗12位40MS/s流水线模数转换器

设计了一个采用新型预充快速开启开关运放的低功耗12位40MS/s流水线模数转换器(ADC)。该转换器通过采用新型预充开关运放技术、采样保持电路消去结构、动态比较器和优化采样电容,大大降低了电路的功耗。电路设计采用1.8V 1P6M 0.18μmCMOS工艺,仿真结果表明,在40MS/s采样速率下,输入信号为19MHz时,无杂散动态范围(SFDR)为90.15dB,信噪失真比(SNDR)为72.98dB,功耗为27.9mW。     

Analysis of second-order harmonic distortion of ADC usingbispectrum

This paper presents a new frequency-domain approach to test the second-order harmonic distortion of an analog-to-digital converter (ADC). It uses triple correlation and its Fourier transform, bispectrum, to suppress quantization noise and improve the testing accuracy of weak second-order harmonics. Our approach is independent of the triggering delay that is a known weakness of the spectral averaging method. The new method is also less sensitive to the quantization noise than the power spectrum averaging method. Both theoretical analyses and illustrative computer simulations are provided     

适用于助听器语音信号处理的DFT调制滤波器组设计

提出了一种近似重构、过采样DFT调制滤波器组的设计方法。根据滤波器组的系统失真和输出混迭方面的要求,构造一个简单的目标函数,并采用等纹波法优化设计原型滤波器。与现有方法相比,该设计方法将复杂的非线性优化问题转化为对单一参数的选取,简化了设计过程,并且可在满足DFT调制滤波器组设计准则的条件下使得通带范围内误差逼近最小。利用加权叠接相加（Weighted Overlap and Add,WOLA）的快速实现结构,使整个系统更加高效、易行,可用于助听器语音信号分频带处理与综合。     

基于MMSE和谱峭度的滚动轴承故障诊断方法

共振解调作为滚动轴承常用的故障诊断方法,存在带通滤波器参数难以确定的缺点,针对此缺点以及滚动轴承早期故障信号信噪比低的问题,结合最小均方误差估计方法(MMSE)和基于谱峭度的共振解调方法,并将其应用于滚动轴承早期微弱故障诊断中。首先用MMSE方法抑制白噪声来提高信噪比,然后利用谱峭度自适应确定最优带通滤波器参数,最后对带通滤波后的信号进行能量算子解调谱分析,得出诊断结果。数字仿真信号和实验信号验证了该方法的有效性。     

INL and DNL estimation based on noise for ADC test

This paper presents the linearity characterization of an analog-to-digital converter (ADC). The input signal is noise, which allows low analog area overhead for built-in self-test (BIST). The linearity error estimation is proposed based on the spectral analysis of only the output of the converter. This paper presents the underlying theory and practical results supporting the effectiveness of the proposed method.     

水下小孔径阵列远程目标检测方法研究

匹配滤波器频域自适应线谱增强方法是一种基于递归算法的非线性滤波技术,它大大提高了匹配滤波器的检测性能。针对当前该技术使用窄带信号作为发射信号存在可利用的带宽有限,不能充分发挥自适应线谱增强器性能的问题,文章提出将该技术与宽带信号相结合来检测远程目标。仿真显示,该方法在低信噪比条件下获得了较高的信噪比增益和检测概率。海试数据处理结果表明,该技术的处理增益较传统方法高6.24 dB。该方法的高处理增益适合应用在水下无人平台上,弥补了小孔径阵列空间增益不足的缺点,可以实现远程目标的检测。     

Histogram measurement of ADC nonlinearities using sine waves

This paper gives results concerning the measurement of differential and integral nonlinearity of ADC's using the histogram method with a sine wave input signal. We specify the amount of overdrive required as a function of the noise level and the desired accuracy and the number of samples required as a function of the desired accuracy, the desired confidence level, and the noise level. An analysis of the effect on the results of harmonic distortion of the applied signal is given. The error analysis assumes a mixture of coherent and random sampling rather than pure random sampling     

Accurate spectral estimation based on measurements with adistorted-timebase digitizer

The timebase distortion present in an equivalent-time sampling oscilloscope introduces errors in the estimation of the values of the spectral components of a microwave signal when a classical discrete Fourier transform is used. A method is developed here to avoid these errors. The method is tested both in practice and with simulations. Two parts can be distinguished. At first, the timebase distortion is measured. This is done by digitizing a sinusoidal signal applied at the oscilloscope's input, and by calculating the phase of the analytical signal of the digitized waveform. Other possible methods to measure the timebase distortion are discussed, and it is shown why the method used is the most appropriate for our specific application. The knowledge of the timebase distortion is then used to build a least squares error estimator for the values of the spectral components of a digitized microwave signal. An experimental verification is done, from which is concluded that the method effectively removes the spectral estimation errors due to a timebase distortion     

交流增益的测量不确定度

介绍了用四参数正弦波最小二乘拟合法评价线性测量系统交流增益的不确定度分析和评价过程。讨论了主要的不确定度来源，包括信号源误差、采集序列的谐波失真、采集序列的噪声及非谐波失真、采集序列中信号周期的抖动、软件拟合运算误差的影响等等，给出了减小不确定度的主要措施。并结合一个实例，给出了评价结果。     

一种低信噪比环境下的语音端点检测算法

端点检测技术是语音信号处理的关键技术之一，为提高低信噪比环境下端点检测的准确率和稳健性，提出了一种非平稳噪声抑制和调制域谱减结合功率归一化倒谱距离的端点检测算法。该算法首先通过抑制非平稳噪声再采用调制域谱减消除残余噪声来提升信噪比，减少语音失真。然后再提取每帧信号的功率归一化倒谱系数，计算每帧信号与背景噪声的功率归一化倒谱距离。最后将该倒谱距离作为检测参数，采用双门限判决方法进行端点检测。实验结果表明，该端点检测算法对语音帧和噪声帧具有较好的区分性。此外，在低信噪比环境下，所提出的算法对于不同类型的噪声都具有较好的稳健性。     

Random Errors in Time Interval Measurement Based on SAW Filter Excitation

This paper deals with the time interval measurement method that makes use of a surface acoustic wave (SAW) filter as a time interpolator. The method is based on the fact that a transversal SAW filter excited by a short pulse can generate a finite signal with highly suppressed spectra outside a narrow frequency band. If the responses to two excitations are sampled at clock ticks, they can be precisely reconstructed from a finite number of samples and then compared to determine the time interval between the two excitations. We have analyzed noise contributions to the overall uncertainty of the interval measurement. The following four noise sources have been considered: (1) noise of excitation; (2) noise of amplifier; (3) quantization noise of the analog-to-digital converter; and (4) clock and sampling jitter. The main goal of the analysis was to determine the uncertainty of the measurement in dependence on the variances of contributing noises, and the magnitude of the frequency response of the filter. First, each uncertainty component has been analyzed under general assumptions. Second, a general narrow-band filter has been investigated. Finally, we have found the solution for an ideal narrow-band filter where we have obtained simple illustrative results. The results have been compared to Monte-Carlo simulations. Good agreement between theory and simulation has been demonstrated except for the outliers due to the false ambiguity solution that occurs with large measurement errors and small relative filter bandwidth.     

一种基于滤波器组设计的多频带组合信号增强方法

针对低信噪比情况下未知多频带组合信号增强及其检测问题,提出了一种基于滤波器组设计的多频带组合信号增强方法。首先,依据耳蜗时频分解原理设计伽马通滤波器组,对传感器采集数据进行分频带处理;然后,根据信号与背景噪声在强度和稳定性上的差异性,对各频带数据进行方差归一化处理,建立判决统计量;最后,依据判决统计量对各频带数据进行判决处理,实现对多频带组合信号增强。数值仿真结果表明,在仿真条件下,该方法能够将数据中未知多频带组合信号信噪比提高6 dB以上,在信号频带未知情况下实现了对多频带组合信号的增强,提升了能量累积检测方法对未知多频带组合信号的检测效果。     

正弦波拟合参数的不确定度评定

介绍了用4参数正弦波最小二乘拟合法拟合参数不确定度分析和评价过程。讨论了主要的不确定度来源,包括信号源误差、采集序列谐波失真、采集序列噪声及非谐波失真、采集序列中信号周期抖动、软件拟合运算误差的影响等,给出了减小不确定度的主要措施。结合一个幅度拟合实例给出了评价结果。     

Wide-Band High-Resolution Determination of Absolute Frequency

In the determination of absolute frequency, conventional approaches have been either frequency or period counting, both of which are subject to resolution-bandwidth tradeoffs and frequency distortion. This paper presents a new technique that avoids such tradeoffs and separates the filtering and sampling functions, giving greater flexibility and enabling one to work down to the limits established by the noise included with the signal. The technique is the use of a commercial frequency synthesizer as the local oscillator in a phase-locked loop. The paper describes this approach, discusses certain modifications that must be made to the synthesizer if it is to work in this configuration, and discusses the design of a type 2, sixth-order feedback system that produces a better approximation to ideal filter response than is obtained by conventional second-order designs.     

组合声源管路信号失真的自适应修正方法研究

针对使用外置声源模拟汽车进气、排气噪声进行空气声传递路径分析的试验中,现有低频声源无法发出进排气时频率为30 Hz的低频噪声,同时由于低频声源体积过大需要过渡管道才能将噪声信号传输至排气尾管.为了解决声源系统引入过渡管道带来的声阻抗变化,提出了一种自适应预滤波的噪声修正方法.该方法采用最小均方误差自适应算法对系统逆传递...     

基于滤波器组和高阶累积量的信号特征检测

结合高阶累积具有对加性高斯和对称非高斯噪声不敏感的特性，提出了一种基于滤波器组和高阶累积量的特征检测方法。该方法既不需待检信号做高斯和平稳的假设，也不需要有信号的先验知识。其原理是首先通过滤波器组将检测信号在频域上进行分离，选取输出能量较大的一组子频带信号近似给出信号的时频描述；然后在各个选中的子频带内分别计算三阶累积量的短时估计，从而抑制有色噪声，将信号的特征检测出来。仿真信号和实验信号验证了该方法的有效性和适应性，即使信号特征完全淹没在噪声中，也能检测出来。     

Doppler ultrasound spectral enhancement using the Gabor transform-based spectral subtraction

Most of the important clinical indices of blood flow are estimated from the spectrograms of Doppler ultrasound (US) signals. Any noise may degrade the readability of the spectrogram and the precision of the clinical indiCes, so the spectral enhancement plays an important role in Doppler US signal processing. A new Doppler US spectral enhancement method is proposed in this paper and implemented in three main steps: the Gabor transform is used to compute the Gabor coefficients of a Doppler US signal, the spectral subtraction is performed on the magnitude of the Gabor coefficients, and the Gabor expansion with the spectral subtracted Gabor coefficients is used to reconstruct the denoised Doppler US signal. The different analysis and synthesis windows are examined in the Gabor transform and expansion. The signal-to-noise ratio (SNR) improvement together with the overall enhancement of spectrograms are examined on the simulated Doppler US signals from a femoral artery. The results show the denoising method based on the orthogonal-like Gabor expansion achieves the best denoising performance. The experiments on some clinical Doppler US signals from umbilical arteries confirm the superior denoising performance of the new method.     

Testing an ADC linearized with pseudorandom dither

When a pure sinewave is digitized by an analog-to-digital converter (ADC), the errors are determined by the input voltage and, hence, the phase of the sinewave. The errors generate signal harmonics coherently. One technique used to reduce the harmonic distortion is dithering by combining a pseudorandom wide bandwidth dither signal with the input signal. When pseudorandom dither is added to a sinusoidal signal, it randomizes the ADC errors with respect to the sinewave so that the errors cannot add coherently. The dominant effect of the dither component is to reduce large spurious harmonic distortion components by spreading them into many smaller ones. This paper presents a test method for testing an ADC linearized with pseudorandom dither. We present results of testing a 12-bit, 5 MHz converter and a state-of-the-art, 14-15-bit, 10 MHz converter     

A theoretical framework for performance characterization of elastography: the strain filter

This paper presents a theoretical framework for performance characterization in strain estimation, which includes the effect of signal decorrelation, quantization errors due to the finite temporal sampling rate, and electronic noise. An upper bound on the performance of the strain estimator in elastography is obtained from a strain filter constructed using these limits. The strain filter is a term used to describe the nonlinear filtering process in the strain domain (due to the ultrasound system and signal processing parameters) that allows the elastographic depiction of a limited range of strains from the compressed tissue. The strain filter predicts the elastogram quality by specifying the elastographic signal-to-noise ratio (SNR(e)), sensitivity, and the strain dynamic range at a given resolution. The dynamic range is limited by decorrelation errors for large tissue strain values, and electronic noise for low strain values. Tradeoffs between different techniques used to enhance elastogram image quality may also be analyzed using the strain filter.