UWB脉冲信号的时域波束形成方法

分析了超宽带（Ultra Wideband）脉冲信号阵列空时处理的一般原理,比较了它和传统正弦阵列的差别,给出了一种高精度的时域波束形成方法。精确的可变时延电路实现是时域波束形成的主要问题,采用数字延迟线和FIR延迟补偿滤波器相结合的办法实现UWB信号波束形成的数字化处理。设计了因果稳定的高精度的具有分数时延补偿特性的FIR滤波器,解决了UWB信号在小样本采样时时间量化对波束形成器性能影响的问题,相对于频域处理,在有效提高阵列方向图性能同时具有设计简单,运算量小等优点。     

A digital signal processing approach to interpolation

In many digital signal precessing systems, e.g., vacoders, modulation systems, and digital waveform coding systems, it is necessary to alter the sampling rate of a digital signal Thus it is of considerable interest to examine the problem of interpolation of bandlimited signals from the viewpoint of digital signal processing. A frequency dmnain interpretation of the interpolation process, through which it is clear that interpolation is fundamentally a linear filtering process, is presented, An examination of the relative merits of finite duration impulse response (FIR) and infinite duration impulse response (IIR) digital filters as interpolation filters indicates that FIR filters are generally to be preferred for interpolation. It is shown that linear interpolation and classical polynomial interpolation correspond to the use of the FIR interpolation filter. The use of classical interpolation methods in signal processing applications is illustrated by a discussion of FIR interpolation filters derived from the Lagrange interpolation formula. The limitations of these filters lead us to a consideration of optimum FIR filters for interpolation that can be designed using linear programming techniques. Examples are presented to illustrate the significant improvements that are obtained using the optimum filters.     

基于子带化的宽带数字波束形成延时补偿新方法

采用宽带信号的相控阵雷达可获得高的距离分辨率,但也面临挑战:宽带数字波束合成和自适应抗干扰。典型的宽带自适应数字波束合成架构中,首先,在基带采用分数延时滤波器实现多通道的延时补偿;然后,将宽带信号分解为许多子带,在每个子带内做传统的窄带自适应数字波束合成;最后,合成为宽带波束输出。该信号处理方法,在宽带条件下,通过宽带延时补偿实现了精确的波束指向,取得了较好的抗干扰性能。文中基于子带化方法,提出了一种新的架构,将延时补偿合并到窄带波束合成中,即用窄带的附加相移,代替了原有的多通道延时补偿单元。结果是该架构中不再需要分数延时滤波器,大大降低了计算量节约硬件资源。同时,仍然保证了宽带阵列雷达波束的精确指向。结合相控阵雷达阵列实例,文中分别采用传统架构及所提出的新架构完成宽带波束合成,给出仿真结果以供对比,证明了新架构的有效性。     

An oversampled digital PWM linearization technique for digital-to-analog conversion

An algorithmic-based linearization process for uniformly sampled digital pulsewidth modulation (PWM) is described. It is shown that linearization of the intrinsic distortion resulting in uniformly sampled PWM can be achieved by using a fractional delay digital filter embedded within a noise shaping re-quantizer. A technique termed direct PWM mapping is proposed as a pre-compensation filter scheme for applications in high-resolution digital-to-analog conversion.     

基于分数时延滤波器的宽带数字信号时延的实现

阐述了基于分数时延滤波器的时延方法设计原理,对其中4种代表性的设计方法的时延性能进行了评估,给出宽带数字线性调频信号的时延仿真结果,该方法可用于宽带雷达数字信号处理。     

Recovering Periodically Spaced Missing Samples [DSP Tips & Tricks]

We presented a novel method to recover periodically missing samples from appropriately bandlimited signals. We used filterbanks with fractional delay filters as building blocks, and described a windowed-sine method to approximate the fractional delays with FIR subfilters. Because the implementation of filterbanks and the approximation of fractional delay filters are well-studied problems, this makes the proposed recovery method flexible and easy to implement.     

A simple and efficient design of variable fractional delay FIR filters

In this paper, a simple and efficient approach for designing one-dimensional variable fractional delay finite impulse response digital filters is proposed. Two matrix equations, based respectively on the weighted least-squares function of the optimum fixed fractional delay filter and the filter coefficient polynomial fitting, are formulated in tandem to form the design algorithm, which only has the computation complexity comparable with that of designing fixed finite impulse response digital filters. A design example is also given to justify the effectiveness and advantages of the proposed design method.     

基于改进的Tustin算子的分数阶数字积分器设计

根据传统的Tustin积分算子的积分原理,改变Tustin积分算子的采样间隔,提高积分精度,以此得到了改进的新算子,然后通过Lagrange FIR分数阶延迟滤波器把新积分算子中的分数阶项转换为整数阶,最后采用连续分数展开(CFE)的方法实现了分数阶数字积分器设计.实验证明,设计的分数阶数字积分器在幅频特性上面明显优于传统方法设计的分数阶数字积分器.     

基于SKA1-low的数字波束形成技术

为配合我国参与平方公里射电阵( SKA)项目,深入理解数字波束形成条件下的射电干涉观测和相应数据处理算法,提出了基于多相滤波器组信道化数字移相波束形成和基于分数阶时延滤波器数字波束形成两个技术算法,前者使用互相关法求解相移量来提高波束指向精度,后者是传统算法与射电观测理论的有效结合。首先,详细介绍了算法的理论推导并分析了它们的优缺点;然后,通过仿真实验验证了这两个算法的正确性,仿真结果表明基于多相滤波器组信道化数字移相波束形成算法相比传统算法具有一定的优越性;最后,分析了不同阶数时延滤波器对波束形成的影响并得出了相应结论。这些仿真结果及结论可为从事SKA项目的科研人员提供参考。     

基于遗传算法的FIR可变分数延迟滤波器设计

分数延迟滤波器广泛用于语音处理,回声消除,多速率信号处理等方面.文中设计的FIR可变分数延迟滤波器用于解决全数字接收机的时钟同步问题.首先用传统的加权最小平方误差方法设计出滤波器参数,然后通过遗传算法对参数进行优化并通过matlab仿真验证算法的有效性,仿真结果表明所设计的滤波器有很好的幅度特性和相位延迟特性.     

Reconstruction of nonuniformly sampled bandlimited signals by means of digital fractional delay filters

This paper considers the problem of reconstructing a class of nonuniformly sampled bandlimited signals of which a special case occurs in, e.g., time-interleaved analog-to-digital converter (ADC) systems due to time-skew errors. To this end, we propose a synthesis system composed or digital fractional delay filters. The overall system (i.e., nonuniform sampling and the proposed synthesis system) can be viewed as a generalization of time-interleaved ADC systems to which the former reduces as a special case. Compared with existing reconstruction techniques, our method has major advantages from an implementation point of view. To be precise, (1) we can perform the reconstruction as well as desired (in a certain sense) by properly designing the digital fractional delay filters, and (2) if properly implemented, the fractional delay filters need not be redesigned in case the time skews are changed. The price to pay for these attractive features is that we need to use a slight oversampling. It should be stressed, however, that the oversampling factor is less than two as compared with the Nyquist rate. The paper includes error and quantization noise analysis. The former is useful in the analysis of the quantization noise and when designing practical fractional delay filters approximating the ideal filters.     

一种Farrow结构数字延时滤波器的设计

普通数字延时滤波器虽然结构简单,但系数计算过程复杂,在延时参数快速变化时,系数更新速度无法满足实时性要求,在工程应用上受限制。采用Farrow结构数字延时滤波器能够更加灵活高效地进行分数延时滤波,延时参数改变时,无需重新计算滤波器系数,更容易在现场可编程门阵列(FPGA)上实现。介绍了一种Farrow结构数字延时滤波器,提出采用基于对称结构的滤波器系数求解方法,并经过加权优化,获得最终Farrow滤波器的系数。系数计算过程中,通过对设计所得Farrow滤波器延时精度和误差的分析,调整加权因子的取值和滤波器阶数,进而提高延时精度。计算机仿真结果证明了加权对称系数求解Farrow滤波器系数方法的有效性和实用性。     

针对级联型ⅡR滤波器群延时的均衡优化技术

为了减小由非恒定群延时所引起的滤波器的输出信号失真,本文提出一种适用于级联型无限长脉冲响应数字滤波器的群延时均衡优化方法.通过在级联型ⅡR数字滤波器每一级的输出插入全通均衡器,减小群延时在通带范围内的变化,进而减小滤波器的输出信号失真.对于本文提出的群延时优化方法,当采用1阶和2阶均衡器进行电路优化时,在0~100Hz的通带范围内,分别将群延时的变化量减小了28.19%和49.93%.基于0.18μm CMOS标准单元库进行逻辑综合与版图设计,最终得到整个滤波电路IP核版图的面积为0.1747mm².相比于已有文献方法,本文方法在群延时优化上效果显著,电路实现上功耗和面积较小,非常适合片上系统应用.     

Multirate digital filters, filter banks, polyphase networks, andapplications: a tutorial

The basic concepts and building blocks in multirate digital signal processing (DSP), including the digital polyphase representation, are reviewed. Recent progress, as reported by several authors in this area, is discussed. Several applications are described, including subband coding of waveforms, voice privacy systems, integral and fractional sampling rate conversion (such as in digital audio), digital crossover networks, and multirate coding of narrowband filter coefficients. The M-band quadrature mirror filter (QMF) bank is discussed in considerable detail, including an analysis of various errors and imperfections. Recent techniques for perfect signal reconstruction in such systems are reviewed. The connection between QMF banks and other related topics, such as block digital filtering and periodically time-varying systems, is examined in a pseudo-circulant-matrix framework. Unconventional applications of the polyphase concept are discussed     

Digital integrator design using Simpson rule and fractional delay filter

The IIR digital integrator is designed by using the Simpson integration rule and fractional delay filter. To improve the design accuracy of a conventional Simpson IIR integrator at high frequency, the sampling interval is reduced from T to 0.5T. As a result, a fractional delay filter needed to be designed in the proposed Simpson integrator. However, this problem can be solved easily by applying well-documented design techniques of the FIR and all-pass fractional delay filters. Several design examples are illustrated to demonstrate the effectiveness of the proposed method.     

A New Adaptive Subsample ETDE for Large-Range Time-Varying Delay Estimation Using Low-Order Interp olators

We present a new adaptive subsample Ex-plicit time delay estimator (ETDE) that is well suitable for determining and tracking the large-range time-varying delay between two noisy bandlimited signals only using low-order interpolators. Unlike conventional ETDE where a Fractional delay filter (FDF) is used, the new ETDE is based on a novel on-line interpolator which has the ability to optimally synthesize any continuously variable noninte-gral sample delay. The theoretical derivations for our pro-posed on-line interpolator and the new ETDE algorithm are provided. Simulations are presented to evaluate the performance of the new estimator and to compare the per-formance with other estimators. Simulation results show that the new ETDE significantly outperforms FDF-based ETDE when the delay is large or varies in a large range, and the same filter orders are used.     

Closed-form discrete fractional and affine Fourier transforms

The discrete fractional Fourier transform (DFRFT) is the generalization of discrete Fourier transform. Many types of DFRFT have been derived and are useful for signal processing applications. We introduce a new type of DFRFT, which are unitary, reversible, and flexible; in addition, the closed-form analytic expression can be obtained. It works in performance similar to the continuous fractional Fourier transform (FRFT) and can be efficiently calculated by the FFT. Since the continuous FRFT can be generalized into the continuous affine Fourier transform (AFT) (the so-called canonical transform), we also extend the DFRFT into the discrete affine Fourier transform (DAFT). We derive two types of the DFRFT and DAFT. Type 1 is similar to the continuous FRFT and AFT and can be used for computing the continuous FRFT and AFT. Type 2 is the improved form of type 1 and can be used for other applications of digital signal processing. Meanwhile, many important properties continuous FRFT and AFT are kept in the closed-form DFRFT and DAFT, and some applications, such as filter design and pattern recognition, are also discussed. The closed-form DFRFT we introduce has the lowest complexity among all current DFRFTs that is still similar to the continuous FRFT     

The optimization of bandlimited systems

Bandlimited functions play important roles in the design of signal transmission systems; antennas; digital, analog, and optical filters; tapered transmismon lines, etc. The basic problem arising in all the above-mentioned applications consists in approximating a given nonbandlimited response with a bandlimited system function. This paper presents 1) a unified treat-merit of the mathematical properties of bandlimited system functions; 2) analytical and numerical methods for the optimization of bandlimited systems; and 3) the application of the theoretical results to practical engineering problems.     

一种自适应语音增强均衡器的研究

在参考滤波器组的基础上，提出一个用于语音和音频信号进行时不变或自适应谱修正的数字滤波器结构，主要用于音频信号均衡和降噪。在频域计算滤波器系数的同时，信号在时域滤波。与通常频域处理相比，在信号延迟、原型滤波器设计、复杂性等方面，该结构有良好的特性。该算法既适用于均匀频率分辨率，也适用于非均匀频率分辨率。     

分数阶傅里叶域两通道滤波器组

 基于两通道滤波器组构建的子带信号处理方法已在图像、语音信号处理中得到广泛的应用.本文从分数阶傅里叶域多抽样率信号处理基本理论和分数阶卷积定理出发,推导了分数阶傅里叶域两通道滤波器组准确重建的基本条件,并基于传统傅里叶域有限长标准正交镜像滤波器组和共轭正交镜像滤波器组的原型滤波器设计了分数阶傅里叶域标准正交镜像滤波器组和共轭正交镜像滤波器组.本文所提出的结论为分数阶傅里叶域滤波器组理论的建立提供了基本依据,同时也为分数阶傅里叶变换在图像、语音信号处理等工程实践中的应用奠定了理论基础.最后,仿真实验验证了所提分数阶傅里叶域滤波器设计方法的有效性.