High-accuracy Fourier analysis based on synchronous samplingtechniques

The lack of synchronization between the sampling rate and the signal frequency represents the main source of errors in the frequency analysis of periodic signals performed by means of digital techniques. Several algorithms have been proposed in the literature to reduce these errors, at the cost of an increment in the computational burden of the instrument. However, the complete elimination of these errors can be achieved only when the sampling rate is synchronized to the fundamental frequency of the signal to be analyzed. It is shown how this can be achieved. The results of the fast Fourier transform (FFT) of multifrequency signals under synchronous sampling conditions are given. The accuracy of the measurements is discussed     

Uniform sampling of periodic signals using successive conversions

This paper presents a new approach to the generation of uniform sampling signals which is based on multiple signal conversions: from pulse to digital, then from digital to analog and afterwards from analog back to pulse. It is shown that a conventional analog-to-frequency converter is suitable to be used as a controllable pulse generator providing for equidistant sampling pulses the frequency of which is proportional to the sampled signal frequency. This paper describes two methods for designing fast response, jitter-free sampling-time generators allowing accurate, uniform sampling of periodic signals. The prototypes have been built and tested     

基于FPGA的激光外差干涉仪反正切解调技术

针对高频声学应用中多普勒信号软件解调的局限,提出了一种基于FPGA的数字硬件信号解调方案.基于AD9467设计了高速ADC采集系统,以实现原始多普勒信号的奈奎斯特采样或带通采样;基于Xilinx Zynq-7000片上芯片系统设计数字信号处理系统,实现了 DDS合成参考信号与多普勒信号混频、低通滤波产生I&Q基带信号对...     

High-Frequency Digital Lock-In Amplifier Using Random Sampling

A high-frequency digital lock-in amplifier (LIA) that uses a random-sampling scheme is proposed and tested experimentally in this paper. By using this sampling strategy, it is possible to process, without aliasing effects, periodic signals of frequencies that are several times higher than the Nyquist frequency. Analytical and numerical analyses that show the advantages and limitations of the proposed scheme are presented. A high-frequency digital LIA implementation is also described. The prototype maximum sampling frequency is 150 kHz, and its maximum signal frequency without aliasing is 2.5 MHz, limited only by the random-sampling period quantization. Experimental results that validate the proposal are presented.     

稀疏信号的周期非均匀采样研究

为了实现稀疏信号的有效采样,根据稀疏信号仅有部分频带有能量的特点,建立了周期非均匀采样与重构的模型;利用联合子空间理论将采样与重构转换为向量运算。借助多测量向量模式确定非零行向量的位置,并通过插值器实现信号完整重构,使其能够在数字系统中应用。实验与仿真结果表明该方法能够实现稀疏信号的采样与重构。     

带通滤波后语音可懂度的实验研究

语音信号在空气中远距离传播时,由于大气的低通滤波效应,高频分量衰减严重,能够到达远端的量很少．其占据的信号源部分能量不能有效利用,从而使语音传播距离受到很大限制。在某些要求距离的场合,需要牺牲部分语音清晰度,传播时保留最低限度的可懂度以提升语音传播距离。语音信号中所含频率成分丰富,但对语音可懂起决定作用的只是其中有限范围的频段。以数字滤波和主观评价的方法对采样频率为44．1kHz的汉语语音可懂的极限频率进行了研究。结果表明,在满足语音内容完全可懂的要求时,汉语语音频率的最低上限应取在1kHz,最高下限应处于300Hz。该结论可以为调制信号源频率以提高能量利用效率,提升语音传播距离提供重要依据。     

Frequency Measurement Techniques

This paper considers the problem of digital frequency measurement of band-limited sampled data signals. The signal is assumed to be a fixed sine wave in noise over the measurement interval. The advent of low-cost A/D converters and the drastic price reductions predicted for IC's in the near future make techniques of this type appear practical for special-purpose digital hardware as well as a general-purpose computer. The probability density function is determined for frequency measurements based on the nth root of the ratio of the derivative of the signal to the signal itself, or its quadrature component. The results are compared with the instantaneous rate of change of phase technique. In addition, attention is paid to techniques for computing the derivatives of a signal. It is shown that a Lagrange formulation yields an excellent approximation to the first and second derivatives as long as the signal is sampled at no less than twice the Nyquist rate. Finally, it is shown that for a given error, sampling at approximately twice the Nyquist rate minimizes the number of arithmetic operations required per unit time and bandwidth.     

谐波窗方法及其在裂纹故障识别中的应用

分析了谐波小波的优点。在研究谐波小波的频段分解的基础上，提出了不分层分析的谐波窗方法。推导了谐波窗的频段分析表达式，并给出了实现过程。该方法显示了小波分析的窗口伸缩功能．但避免了在基于二进的小波变换过程中，由于隔点采样导致的信息丢失现象。利用该方法对3个数字信号进行分析，体现了该方法对近频信号和微弱瞬态信号的识别能力，也体现了该方法对淹没在强噪声中微弱周期信号的提取能力。利用该方法，对某汽车齿轮箱的裂纹故障产生的原因进行振动信号的频域识别，得到了满意结果。     

A Digital Method of Frequency Multiplication and Its Application to Numeric Spectrum Analysis of Periodic Signals

A digital way to produce an integral multiple of the frequency of periodic trigger pulses is described. This procedure is used to generate the sampling pulses for the ADC of a time-series analyzer. The device which implements this method is called sampling frequency generator (SFG). A comparison is made between this digital method and the analog frequency multiplication method by means of a PLL.     

Radiometers, autocorrelators, and matched-filter receivers by means of acousto-optic spectrum analysis

Acousto-optic spectrum analyzers provide a convenient means of separating wide-bandwidth signals into their frequency components. By a change in the rf input signal into the spectrum analyzer and by the provision of additional digital postprocessing, it is possible to perform radiometry, signal autocorrelation, and matched-filter reception. Although the acousto-optic device has a space-integrating architecture, the matched-filter receiver can be implemented for signals having time durations much longer than the acousto-optic cell. The resulting signal-to-noise ratio improvements from the receiver are consistent with the time-bandwidth product of the waveform, rather than the time-bandwidth product of the acousto-optic device. A mathematical foundation of the processor is presented along with specific receiver implementations. Computer-simulation and experimental results demonstrate key findings. In one experimental example, a linear-frequency-modulated waveform is matched-filter processed to recover a signal that is -24 dB with respect to the input noise floor.     

基于数字锁相环的科氏质量流量计信号处理方法

针对科里奥利质量流量计信号处理中存在的问题 ,即信号频率在小范围内变化和信号易受谐波干扰 ,采用基于数字锁相环的方法处理科氏质量流量计的信息 ,跟踪信号频率的变化 ,计算相位差。对美国专利提出的方法进行了改进 ,做了仿真。仿真结果表明 ,在有谐波干扰的情况下 ,数字锁相环方法可以准确地跟踪信号频率的变化和计算相位差。本文还对计算误差进行了分析     

Definition of Instantaneous Dielectric Loss Factor and Digital Algorithm for Online Monitoring

Among the digital algorithms currently available for online monitoring of the dielectric loss factor of power apparatus, there are unavoidable limitations, such as that most of them must strictly meet the synchronous sampling criterion simultaneously with a precondition that the power frequency must be invariable in the sampling periods. This paper presents a unique definition and a corresponding digital algorithm of the instantaneous dielectric loss factor. The theoretical basis of the algorithm is also deduced. In a nonstationary signal environment where power frequencies are randomly variable, the algorithm can accurately track the instantaneous frequency of power signals, recursively calculate the parameters of parallel equivalent circuit, and, finally, figure out the waveform of the instantaneous dielectric loss factor. Simulations and experimental results show that the proposed algorithm renders good application performance without being affected by the limitations incurred from synchronous sampling and harmonics.      

Low order anti-aliasing filters for sparse signals in embedded applications

Major emphasis, in compressed sensing (CS) research, has been on the acquisition of sub-Nyquist number of samples of a signal that has a sparse representation on some tight frame or an orthogonal basis, and subsequent reconstruction of the original signal using a plethora of recovery algorithms. In this paper, we present compressed sensing data acquisition from a different perspective, wherein a set of signals are reconstructed at a sampling rate which is a multiple of the sampling rate of the ADCs that are used to measure the signals. We illustrate how this can facilitate usage of anti-aliasing filters with relaxed frequency specifications and, consequently, of lower order.     

齿轮故障振动信号检波解调分析中的混频效应

在齿轮故障诊断中,振动信号解调分析是一种很有效的方法。本文在分析调幅调相信号检波解调分析原理的基础上,讨论了该类算法软件实现过程中产生的混频效应,指出信号在检波解调过程中经过非线性变换后将包含高频成分,有可能发生明显的混频,导致得出错误的解调信号。通过合理地选择采样频率及分析算法中低通滤波器的截止频率或采用较高的采样频率能有效地消除这种混频效应对解调的影响,最后以仿真数据及实验论证了这些结论。     

涂层、钢筋混凝土电化学阻抗现场测试仪的研制

当今,实际工程中缺乏现场快速简便测量涂层、钢筋混凝土状态的仪器。利用高速单片机、数字信号发生器、双通道同步高速AD和高输入阻抗模拟电路,设计了一套电池供电、适于现场应用的便携式电化学阻抗（EIS）测试仪。考虑到涂层或钢筋混凝土的高阻抗以及现场测试过程易受外部噪声干扰的特点,仪器电路设计中采取了双通道8阶低通滤波器和前置...     

A Technique for Synthesizing Digital Filters

In designing digital systems, one often faces the task of replacing a given analog filter by an equivalent digital filter. This paper proposes a method for synthesizing such digital filters in the time domain. It is assumed that the pulsed transfer function of the digital filter is a ratio of two rational polynomials. The coefficients are then determined by least-square fitting the digital filter to the analog filter's sampled input and output data. The resulting equations for computing the coefficients are linear. It is shown that the digital filter is essentially related to the analog filter, the sampling time, and the power spectrum of the signal being processed. If the signal is band-limited and the sampling frequency is sufficiently high, the digital filter can then be simply approximated by the Z transform of the analog filter multiplied by the sampling period.     

Five-Point Amplitude Estimation of Sinusoidal Signals: With Application to LVDT Signal Conditioning

A simple algorithm for the amplitude estimation of sinusoidal signals is proposed. It is first shown that three well-spaced samples are sufficient to calculate the amplitude of a harmonic signal. However, the three-point estimator is sensitive to deviation of the sampling rate. To deal with errors that were caused by noncoherent sampling, a modified algorithm that uses two more samples is developed. With the five-point algorithm, estimation errors fall below 0.04% for 1% of frequency deviation. As a demonstration of its simplicity, the five-point estimator is applied to the signal conditioning of a linear variable differential transformer (LVDT) using moderate-speed devices. By implementing the proposed algorithm with a low-cost 16-b microcontroller, a digital LVDT signal conditioner is built and tested with satisfactory performance.      

EVM可控的数字调制信号的产生方法研究

针对数字信号发生器检定装置和信号分析仪期间核查缺乏产生EVM值可控的数字信号标准装置的情况,在理论推导加性高斯白噪声环境下的EVM与SNR之间的解析关系基础上,提出了基于噪声发生器和数字信号发生器的EVM值可设置的数字调制信号产生系统方案。对系统输出的BPSK信号和64QAM信号展开重复性和稳定性测试,测试结果表明,尽管理论设置值与实际值之间存在一定的差距,但并不影响任意EVM值数字调制信号产生系统的稳定输出,其重复性和稳定完全可满足数字信号发生器校准装置或者矢量信号分析仪的期间核查的需要。     

Helium plasma source time-of-flight mass spectrometry: off-cone sampling for elemental analysis

In this paper, an atmospheric pressure, helium microwave-induced plasma (MIP) ion source coupled with an orthogonal acceleration time-of-flight mass spectrometer (TOFMS) is explored for elemental analysis. Studies of the relationship between ion signals and sampling distance of the MS reveal that background signals can be suppressed dramatically without sacrificing the signal intensities of analytes when the microwave plasma plume is off the tip of the sampler orifice. This "off-cone" ion sampling mode provides a technique to obtain nearly "clean" background spectra and, thus, eliminates the spectral interference from entrainment air and the working-gas species, making it possible to sensitively determine isotopes that suffered from spectral interference in ICPMS and MIPMS (such as 40Ca, 52Cr, 55Mn, and 56Fe). On the other hand, since the high-temperature plasma is kept away from the sampler aperture, off-cone sampling places little demand on the cooling device and the lifetime of the sampler plate can be extended. The instrumental system can provide a fairly good mass resolution of 1100 (fwhm). The detection limits (3sigma) in the tens of picograms per milliliter level for the elements studied can be achieved with a digital oscilloscope. These detection limits can be easily improved with an advanced detection system, which is currently available in commercial markets.     

The implementation of a digital sine wave oscillator using theTMS320C25: distortion reduction and applications

The implementation of a digital sine wave oscillator using the TMS320C25 digital signal processor (DSP) is described. The system is implemented with the Dalanco Spry model 25 DSP board, and a software system is designed whereby an IBM PC host computer provides control of the waveform generator functions and parameters. Waveforms are generated using the lookup-table (LUT) method. The methods of direct LUT and linear interpolation of missing samples are implemented and compared with a method that uses a trigonometric identity to reduce the harmonic distortion of the sine wave by effectively increasing the table length of the direct LUT method. The results of an experiment performed in the digital domain are presented without consideration of the problem of analog reconstruction. The oscillator can produce a sine wave without the nonuniform sampling distortion associated with fractional addressing, over the range of 7.2 Hz to 58.8 kHz, with a resolution of 7.2 Hz using 384 data words. The trigonometric identity method uses less data memory than other methods for the same distortion levels. Waveforms for the sum of two sine waves, a frequency swept sine wave, amplitude modulation (AM), and frequency modulation (FM) signals are shown as applications of the waveform generator