A Digital Power Spectrum Analyzer

In this paper we develop a new algorithm for a digital power spectrum analyzer. The advantage of this new algorithm is that it makes possible the lowering of the average sampling rate without incurring large errors.     

基于复解析带通滤波器的复调制细化谱分析的算法研究

在采用复解析带滤波器宽度为fs／2D、隔D点选抽一点、移频和作N点谱分析的方法进行细化谱分析的原理基础上，研究了基于复解析带通滤波器复调制细化谱分析的算法。讨论了算法的核心部分－复解析带通滤波器的构造，滤波器外扩的原理，一级选抽及二级选抽滤波器的设计。仿真研究表明，采用一级FIR数字滤波器，当半阶数为500时，最大细化倍数可达150，仍能保证幅值分析误差在1％以内，采用二级FIR滤波器最大细化倍数可达2000倍以上，运算速度也有较大幅度的提高。     

另一类幅度非均匀取样信号的数字频谱及补偿算法研究

研究了高速并行数字化系统中由ADC增益的不一致性造成的幅度非均匀取样信号的特点,推导出了这类幅度非均匀采样信号的数字频谱表达式,详细分析了输入正弦信号时的数字频谱,得到了系统的信噪比表达式。提出了一种利用IFFT运算对输出信号的频谱进行重构的补偿方法,计算机仿真验证了该方法的有效性。     

Digital spectra of nonuniformly sampled signals: a robust samplingtime offset estimation algorithm for ultra high-speed waveformdigitizers using interleaving

The author presents an algorithm for estimating the sampling time offsets encountered at the parallel sampling paths of an ultra-high-speed waveform digitizing system which is realized by interleaving many sample-hold and A/D (analog/digital) converter modules. One obvious application of this algorithm is to feed these estimates back to adjustable delay units in all the sampling paths to compensate the sampling-time offsets. Simulation results indicate that with an 8-bit quantizer the residual timing error can be reduced to just about 0.05% of the sampling period     

Underwater digital holography for studies of marine plankton

Conventional and digital holographies are proving to be increasingly important for studies of marine zooplankton and other underwater biological applications. This paper reports on the use of a subsea digital holographic camera (eHoloCam) for the analysis and identification of marine organisms and other subsea particles. Unlike recording on a photographic film, a digital hologram (e-hologram) is recorded on an electronic sensor and reconstructed numerically in a computer by simulating the propagation of the optical field in space. By comparison with other imaging techniques, an e-hologram has several advantages such as three-dimensional spatial reconstruction, non-intrusive and non-destructive interrogation of the recording sampling volume and the ability to record holographic videos. The basis of much work in optics lies in Maxwell's electromagnetic theory and holography is no exception: we report here on two of the numerical reconstruction algorithms we have used to reconstruct holograms obtained using eHoloCam and how their starting point lies in Maxwell's equations. Derivation of the angular spectrum algorithm for plane waves is provided as an exact method for the in-line numerical reconstruction of digital holograms. The Fresnel numerical reconstruction algorithm is derived from the angular spectrum method. In-line holograms are numerically processed before and after reconstruction to remove periodic noise from captured images and to increase image contrast. The ability of the Fresnel integration reconstruction algorithm to extend the reconstructed volume beyond the recording sensor dimensions is also shown with a 50% extension of the reconstruction area. Finally, we present some images obtained from recent deployments of eHoloCam in the North Sea and Faeroes Channel.     

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.     

基于希尔伯特变换的信号解调算法及其在飞机供电特性参数测试系统中的应用

电压调制幅度、频率调制幅度和畸变频谱是对飞机机载电源系统供电特性参数进行测试时与信号调制相关的重要参数。为了更好地测试这些参数,提出了一种基于希尔伯特变换原理的数字解调算法,可以将采集的幅度或频率调制信号进行解调分析;同时,构建了一套飞机供电特性参数测试系统,将解调程序与硬件设备结合,通过程序控制数据采集卡对机载电源信号采样与解调分析;设计了实验对波形发生器产生的调制信号进行采样与解调,验证了该系统具有良好的效果,可以应用于飞机供电特性参数的测试中。     

Analysis of a Reentrant Delay Line Real Time Speed-Up Device

The output spectrum of an idealized version of a reentrant delay line speed-up device is calculated. Study of this spectrum yields considerable insight into the rather peculiar way in which this device achives a speed-up in real time. The analysis is then modified to show the effects of time sampling and boxcarring on the output spectrum. Finally, the use of digital instead of analog delay lines is discussed, and an expression for the power density spectrum of the output quantization noise is derived.     

Multi‐rate digital control for a cascaded system consisting of two subsystems with different input time delays

Abstract

This paper presents a multi‐rate state‐space control scheme for digital control of a cascaded continuous‐time system with fractional time delays. First, a discrete‐time state‐space representation of a continuous‐time system with a fractional input delay is established. Based on the time‐delay digital modelling, an ideal state reconstructor is also presented such that system states are exactly reconstructed via the measurement histories of inputs and outputs without a state observer. Next, a time‐delay subsystem (designated subsystem 1) with a fast sampling rate is designed to form the inner loop of the overall system, then the designed closed‐loop subsystem 1 is cascaded with a time‐delay open‐loop subsystem 2 with a slow sampling rate. A digital modelling of the time‐delay open‐loop subsystem 2, based on a fast‐rate sampling, is also formed for obtaining the digital modelling of the overall cascaded continuous‐time system by using the block‐pulse function approximation. Then, the fast‐rate overall system is converted into a slow‐rate model via the newly developed model conversion technique. Furthermore, subsystem 2 is separated from the slow‐rate overall system via a linear transformation for achieving a reduced‐order subsystem design. As a consequence, a digital control law is determined on some specific goals for the overall system. The proposed method is suitable for digital control of a multivariable, multi‐rate, time‐delay system in which state variables are not accessible.     

Spectral analysis of a sampling optical time-domain reflectometer

A spectral analysis is presented for the backscatter signal of optical time-domain reflectometers (OTDR). Periodic spatial fluctuations in the fiber attenuation produce modulation sidebands in the frequency spectrum of the backscatter signal. Applying the sampling technique to OTDR leads to a considerable improvement of the signal-to-noise ratio (S/N) due to spectrum compression, as a tradeoff with the measurement time. It gives a performance equivalent to digital averaging. Design examples are given at wavelengths of 0.85, 1.3, and 1.55 μm. A practical setup of a sampling OTDR is described     

The effect of accumulation of cathodic decomposition on the anodic conversion of sulfide ions to sulfur in a continuous extractive electrochemical reactor

Abstract

This paper presents an intelligent digital filter synthesis system which consists of a user‐friendly graphic I/O interface, a synthesis program designing filter coefficients, an intelligent hardware generation system and a simulator including time and frequency domain simulation. Users need only to specify the filter type, filter specifications, sampling rate requirement and some parameters through the graphic interface. The filter response and the simulation results can be immediately shown on the screen. The graphic interface is based on the X Window System and therefore provides a network‐transparent and vendor‐independent operating environment for the system. The synthesis system is targeted for two main types of digital filter: FIR and IIR. An automatic partition algorithm on FIR is developed to meet the user's sampling rate requirement. The optimal cascade architecture of biquad‐based IIR with programmable coefficients is also derived. The final circuit designs under economic bitwidth are generated by the GDT system and further layouts can be generated.     

微弱电信号检测方法回顾

系统分析了微弱电信号检测的理论和技术,着重讨论了强噪声背景下微弱电信号现有检测方法的优缺点及应用.以线性和非线性为主线,将线性分析法按时域(相关检测法、锁定放大、取样积分、数字式平均、时域平均)、频域(功率谱法)以及时频域(小波变换谱分析、分数谱分析)检测方法进行梳理归类,而对于非线性分析法主要分析了高阶谱、神经网络、支持向量机、经验模式分解、混沌理论、差分振子以及随机共振方法.最后认为,多种理论与技术的结合以及借助软件技术,如LabVIEW,是弱信号检测的发展趋势.     

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...     

Digital-to-analog (D/A) converters with nonuniformly sampledsignals

In this paper, we investigate the problem of the D/A converter with nonuniformly sampled input data. The input digital data were obtained by sampling the intended analog waveform at nonuniform time intervals, and we look into the question: “Given that the timing offset of each data sample is known, would it be beneficial, in terms of the output signal-to-noise ratio, to use this offset to adjust the playback timing of the D/A converter?” We examine two different timing strategies. The first approach simply plays out the signal at a uniform rate, while the second one uses the known timing offset to adjust the D/A converter playback timing accordingly. The closed-form expressions of the spectrum of the D/A converter output signals are derived. From these expressions, we find that the spectrum structure, for the case where the timing offsets are compensated, is the infinite sum of the weighted shift to the baseband spectrum. For the uniform playout approach, the spectrum structure is much more complicated where each shifted spectrum is modified by a different weighting function of the frequency. Although the spectrum structure may be conceptually simpler for the case where the timing offsets are compensated, it is by no means clear that either method is better than the other in terms of the output waveform quality. We then apply the results to analyze the direct digital synthesis output sine waves. The signal-to-noise ratio, SNR, for both cases is derived in simple closed form. It is found that for the case where the timing offsets are compensated, the SNR can be greatly enhanced by appropriate selection of operation parameters     

A new algorithm for improving the accuracy of periodic signalanalysis

Digital periodic signal analysis often requires synchronized sampling with the signal being analyzed. In certain practical situations, however, this condition is difficult to satisfy. As a result, a number of undesirable effect such as the spectral leakage associated with the discrete Fourier transform (DFT), and the truncation errors in digital wattmeters arise and degrade system performance. This paper presents a new approach which attempts to remedy the underlying problem. The basic idea of the proposed method is to modify the actual sampled sequence such that it becomes an ideal sample sequence which is synchronized with the signal subjected to sampling. A simple algorithm for modifying the sampled sequence on-line is derived based on interpolation. The proposed approach requires quite modest additional computational burden which makes it suitable for real-time signal professing. To illustrate the practical applicability of the proposed algorithm, the paper considers two distinct but common cases. First, it shows how the proposed method can be used in the case of DFT analysis of harmonic signals, and secondly, it considers the digital wattmeter application area in electrical power-system measurement. Results show that the proposed algorithm is capable of reducing both the leakage effect in DFT analysis and truncation errors in digital wattmeters     

Chaos-EP-Based Digital Redesign of Uncertain Hybrid Time-Delay Systems With State and Input Constraints

This paper presents a chaos and evolutionary programming (CEP)-based digital redesign scheme. This scheme can determine a practically implementable low-gain digital tracker for global optimization of hybrid uncertain input-delay systems with constraints on states and inputs. The proposed global-optimization searching technique, which was provided in the evolutionary programming together with a chaotic optimization algorithm, is utilized to determine the low-gain digital tracker. This technique takes into consideration the relatively large sampling time and the intersampling behavior for digital control of hybrid input-delay systems with unknown-but-bounded parameter uncertainties. Moreover, the newly developed CEP-based digital redesign scheme provides less conservative results than those obtained by the conventional interval methods. One illustrative example is given to demonstrate the effectiveness of the new scheme compared with the conventional methods.      

A fast and high-precision measurement of distorted power based ondigital filtering techniques

A rapid high-precision measurement of distorted power is described. The measurement is based on digital filtering techniques. The convergence conditions of two previous sampling algorithms are derived, and their frequency responses are analyzed. An adjustable window used for measuring distorted power is also proposed. It is demonstrated that the sampling algorithm using the window leads to significant savings in measuring and computing time, and high accuracy can be obtained     

实时跟踪控制系统中的数字滤波技术

在高精度的实时跟踪控制系统中，由探测器等其他一些因素给系统带来的噪声将不容忽视，数字滤波将是滤除噪声的有效手段，，由于系统实时性和采样频率的要求，本文介绍几种时跟踪系统中常用的数字滤波方法，并对它们在实时跟踪系统中的作用进行了和比较。     

A High Sensitivity Microwave Sampling System with Adaptive Time-Drift Compensation

A new microwave sampling system is presented whose sensitivity has been increased by digital processing of the measurement data. The signal processing gets more complicated due to a time drift caused by electronic circuits in the sampling oscilloscope. An adaptive method to compensate for this time drift has been developed. The influence of noise is discussed, giving the remaining time uncertainty as a function of the signal-to-noise ratio. Finally an example illustrates how this method can be applied in an automatic sampling system.     

一种高效的弱目标线谱检测算法

为实现弱目标线谱检测,在自适应线谱增强(Adaptive Line Enhancement,ALE)算法的基础上,结合频域批处理技术,提出了一种能降低计算量的高效线谱检测算法——归一化频域批处理最小均方(Normalized Frequency-domain Block Least Mean Square,NFBLMS)算法;所提NFBLMS算法在权值迭代过程中,步长参数不受输入信号功率的影响。理论分析和数值仿真结果表明:相比于已有的线谱检测算法,NFBLMS算法能较好地解决ALE算法实时处理运算量问题,并可获得较高的系统增益,且其步长参数具有较强的鲁棒性,能同时兼顾算法的收敛速度和稳态误差。因此NFBLMS算法更适合实际工程应用。