A double Nyquist digital product detector for quadrature sampling

A technique for digitally obtaining the in-phase (I) and quadrature (Q) components of an IF signal is presented. Initially, the input bandpass signal is mixed to a carrier frequency that is one-fourth of the sampling rate of a single A/D converter. The digitized bandpass signal is converted into its I and Q components at one-half the A/D sample rate by a digital product detector (DPD) composed of a commutator, two sign alternators, and two FIR fractional-phase interpolator filters. This simple structure can yield image performance that is limited by A/D quantization using relatively low interpolator filter orders and IF bandwidths as large as one-half the sampling rate of the A/D converter. The DPD performs Nyquist limit demodulation of the sampled bandpass signal and, therefore, requires a minimal sampling rate. The theory of operation, an analytic proof, design methodology, and simulated performance results are presented. Simulated results show that -86 dB images can be obtained with 8-tap FIR interpolators and a 12 bit A/D converter. A VLSI implementation is also presented     

A quadrature charge-domain sampler with embedded FIR and IIR filtering functions

A circuit technique for integrating built-in complex finite-impulse-response (FIR) and infinite-impulse-response (IIR) filtering functions into operation of a subsampler is presented. Based on integrative multiple sampling in the charge domain, the complex FIR filtering function of the sampler provides internal anti-aliasing and image band suppression prior to quadrature downconversion by subsampling. The complex IIR filtering function, taking place at the output sampling rate of the sampler, performs further first-order channel selection filtering on the downconverted signal. An example 50-MHz IF-sampler implementation in 0.8-/spl mu/m BiCMOS demonstrating the feasibility of the technique is presented in the paper.     

一种基于小波的导数采样算法

研究了小波空间中的一种导数采样理论,给出了定理成立的条件及其突破Nyquist率限制的理论依据,将采样理论扩展到了非带限信号领域。对于紧支尺度函数张成的子波空间中的任意信号,可以利用对信号及其导数采样所得的样本求出其小波系数的估计值,进而得到信号的重建表达式。在信号重建的过程中用到的全是有限冲击响应滤波器,降低了实际物理实现的难度。计算机仿真结果表明该方法是切实有效的,信号重建的相对误差小于0．11％。     

A NEW QUADRATURE SAMPLING APPROACH FOR ARBITRARY BANDPASS SIGNAL WITHIN BASEBAND SAMPLING RATE

A new quadrature sampling technique for arbitrary bandpass signal within baseband sampling rate is presented. The input bandpass signal whose carrier frequency lies in the A/D baseband sampling rate is first decimated by factor 2 and modulated by (- 1)n, and then is interpolated by a linear phase FIR all-pass filter, finally the modulated complex envelope of bandpass signal can be produced.     

基带采样率内任意带通信号的正交采样

提出了一种在基带采样率内对任意带通信号的正交采样技术。载频位于ADC(模/数转换器)基带采样率内的任意带通信号,经1/2抽取和(-1)n调制,再由全通线性多相滤波器内插后,得到其复包络的调制输出XI(n)和XQ(n)。     

The design and multiplier-less realization of software radio receivers with reduced system delay

This work studies the design and multiplier-less realization of a new software radio receiver (SRR) with reduced system delay. It employs low-delay finite-impulse response (FIR) and digital allpass filters to effectively reduce the system delay of the multistage decimators in SRRs. The optimal least-square and minimax designs of these low-delay FIR and allpass-based filters are formulated as a semi-definite programming (SDP) problem, which allows zero magnitude constraint at /spl omega/=/spl pi/ to be incorporated readily as additional linear matrix inequalities (LMIs). By implementing the sampling rate converter (SRC) using a variable digital filter (VDF) immediately after the integer decimators, the needs for an expensive programmable FIR filter in the traditional SRR is avoided. A new method for the optimal minimax design of this VDF-based SRC using SDP is also proposed and compared with traditional weight least squares method. Other implementation issues including the multiplier-less and digital signal processor (DSP) realizations of the SRR and the generation of the clock signal in the SRC are also studied. Design results show that the system delay and implementation complexities (especially in terms of high-speed variable multipliers) of the proposed architecture are considerably reduced as compared with conventional approaches.     

Programmable discrete-time type I and type II FIR filter design on the memristor crossbar structure

A new method for effective realization of discrete-time type I and type II FIR filters on the memristor crossbar structure is developed in this paper. For this purpose, first the analog input signal (to be filtered using the discrete-time filter) is discretized using the classical switched-capacitor circuit and then all of the required delayed samples of this discrete-time signal are generated using the circuit designed for this purpose. Next, the weighted sum of these delayed samples of the original discrete-time signal (which forms the output of the FIR filter under consideration) is produced using the memristor crossbar structure. The proposed structure for FIR filter design is, compared to classical methods, advantageous in the way that it does not need any processors or A/D converter. Moreover, it is fully implemented using analog devices and consequently free of round-off error. Another related contribution of this paper is the circuit proposed for automatic tuning the memristance of the given memristor to the desired value with a high accuracy. Four numerical examples, including the application of the proposed FIR filter for demodulation of AM signals, are studied and HSPICE simulations are presented.     

FIR滤波器设计:基于进化规划的频率采样技术

本文介绍了进化规划在要样技术中的应用,结合ＦＩＲ数字低通、带通滤波器设计的两个例子,给出了算法实现的具体操作步骤和实验结果。实验数据表明采用进化规划确定的频率过渡带样本值是最优的,设计的ＦＩＲ滤波器的性能优于查表法。     

基于Matlab的FIR滤波器在DSP中的实现

本文主要对基于Matlab的FIR数字滤波器的设计与如何在DSPOP得到实现进行了研究,介绍Matlab中实现FIR滤波器设计的窗函数法。并结合具体的实例,介绍了如何将Matlab设计的滤波器完成到DSP的FIR滤波器的转化。     

对FIR数字滤波器频域实现错误观点的讨论

关于FIR数字滤波器的频域实现存在一种错误的观点和做法,即直接在离散傅利叶变换域将输入信号属于阻带的谱线清零,而属于通带的谱线保留,再离散傅利叶反变换到时域,并且认为通过这种方法得到的信号是对输入信号理想滤波的结果.本文针对这一观点,利用频域取样的概念,从时域和频域两个角度分析,指出该做法并不能实现理想滤波,并且滤波性能通常不能达到指标要求.     

有限冲击响应数字滤波器的DSP实现

数字滤波是信号处理过程的主要方式,FIR数字滤波器以其系统稳定和易实现线性相位应用更为广泛。设计了采用窗函数法的FIR数字带通滤波器,在DSP中采用单采样模式,在每一个采样周期内只产生一个信号输出值,实时处理采样后的信号。通过MATLAB进行滤波器的仿真,修改滤波器的参数使其达到设计指标。利用窗函数法设立的FIR数字滤波器,是获得较好的主瓣最大能量和旁瓣衰减意义下的最佳设计方法。     

Design and multiplierless realization of maximally flat FIR digitalHilbert transformers

A unified treatment of approximation and realization of type-3 finite impulse response (FIR) linear-phase Hilbert transformers is presented. A simple method based on Bernstein polynomials and half-band filters is proposed to derive the transfer function of the system, and a triangular array realization based on the de Casteljau algorithm is developed from the Bernstein form of the transfer function. It is shown that the array structure, consisting of multiplierless identical modules, can be realized hierarchically using complex and real signal processing techniques     

直升机自动测试中的数据采集及滤波模块的研究

针对国内某型号直升机自动测试系统的实际应用需求,设计了基于TMS320F2812的可配置参数的实时数据采集及滤波模块。模块能够对实时数据进行FIR滤波、FFT频谱分析,实现CAN通讯。在介绍硬件系统的基础上,研究了上述算法的实现,阐述了系统根据实测信号自动调用相关滤波算法的方法,并结合实际应用进行了系统分析。结果表明,该模块满足测试系统的要求,具有良好的实用性。     

基于FPGA的TDM FIR数字滤波器的设计及硬件实现

介绍了一种可实现对高速率、高采样率信号进行实时处理的滤波器——TDM（Time-Domain Multiplexing）FIR数字滤波器的设计方法,并且给出了一个32阶的高速实时TDM FIR数字滤波器的软件仿真及硬件实现过程。     

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.     

Calibration for Realization Errors of Two-Channel Hybrid Filter Bank Analog-to-Digital Converters Based on Frequency-Dependent Model

Hybrid filter bank (HFB) analog-to-digital converters (ADCs) are used to achieve higher sampling rates for medium-to-high resolution applications in many communication systems. However, the analog filter realization errors and channel mismatches severely degrade the reconstruction performance of HFB ADCs. A general error analysis, which takes analog imperfections and channel mismatches into consideration based on a single-channel filtering (SCF) HFB architecture is presented in this paper. A frequency-dependent model (FDM) for sub-ADCs output is then derived to integrate all the realization errors to an equivalent channel-transfer function (ECTF). Based on the ECTF, the perfect reconstruction condition is derived as the optimization object for the synthesis filter banks. Finally, a two-channel SCF HFB ADC prototype with 12-bit resolution and 200 MHz sampling rate is implemented. After mismatch correction with 64-tap finite impulse response (FIR) filters, 75.75 dB of spurious-free dynamic range (SFDR) is achieved over the whole bandwidth. The experimental results show that the average SFDR is enhanced by about 10–25 dB compared with the traditional calibration method.     

基于并行FIR滤波器结构的数字下变频

对宽带信号进行并行处理,可同时满足低功耗和实时性的要求,已成为目前宽带信号处理的研究热点。本文提出了一种可在FPGA中实现的并行快速FIR滤波器设计方法。该方法通过应用并行多相处理技术中的一种新型分布式处理算法,在滤波器结构上实现了多级级联的形式,增强了中频处理的灵活性和通用性,节省了硬件开销。仿真结果表明,该算法很好的解决了原始低通滤波器速度跟不上A/D采样率的问题,把采样率提高到了320MHz以上。同时该方法应用软件实现并行信号处理,避免了使用DDC专用芯片,具有较强的通用性,可以很好的移植到其他CPLD中。     

Frequency tracking techniques of power systems in coloured noises

A technique for tracking the frequency of power systems in sine noises using numerical differentiation is presented. A voltage or current sinusoidal signal corrupted by sine noises and white noises is considered. For the signal corrupted by one or two sine noises, a central numerical differentiation-based method is proposed. For the signal corrupted by multiple sine noises and white noises, a hybrid method of numerical differentiation and a digital finite impulse response (FIR) filter are proposed. The digital FIR filtering algorithm is used to remove the white noises and the sine noises, and the numerical differentiation algorithm is used to estimate the fundamental frequency of power systems when the fundamental component is decomposed out of the signal. The proposed algorithm shows an advantage in time and speed when compared with other existing techniques and shows better dynamics and higher accuracy in frequency estimation. Carried out in Matlab, the simulation results are satisfactory     

一种新的带通信号采样方法

本文提出一种新的带通信号采样方法,实现对带通信号的“等效”低通信号采样。带通信号实际采样率仅为输出端所获同相分量和正交分量采样率的两倍,可以直接确定采样频率和设计低通抗混叠滤波器。该采样方法使用滤波器的多相结构实现,这种实现方法特别适合于线性相移FIR滤波器。     

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。