Signal Processing by Generalized Receiver in?DS-CDMA Wireless Communication Systems with?Frequency-Selective Channels

The generalized receiver (GR) based on a generalized approach to signal processing (GASP) in noise is investigated in a direct-sequence code-division multiple access (DS-CDMA) wireless communication system with frequency-selective channels. We consider four avenues: linear equalization with finite impulse response (FIR) beamforming filters; channel estimation and spatially correlation; optimal combining; and partial cancellation. We investigate the GR with simple linear equalization and FIR beamforming filters. Numerical results and simulation show that the GR with FIR beamforming filters surpasses in performance the optimum infinite impulse response beamforming filters with conventional receivers, and can closely approach the performance of GR with infinite impulse response beamforming filters. Channel estimation errors are taken into consideration so that DS-CDMA wireless communication system performance will not be degraded under practical channel estimation. GR takes an estimation error of a maximum likelihood (ML) multiple-input multiple-output (MIMO) channel estimation and GR spatially correlation into account in computation of minimum mean square error (MMSE) and log-likelihood ratio (LLR) of each coded bit. The symbol error rate (SER) performance of DS-CDMA employing GR with a quadrature sub-branch hybrid selection/maximal-ratio combining (HS/MRC) scheme for 1-D modulations in Rayleigh fading is obtained and compared with that of conventional HS/MRC receivers. Procedure of selecting a partial cancelation factor (PCF) for the first stage of a hard-decision partial parallel interference cancellation (PPIC) of the GR employed in DS-CDMA wireless communication system is proposed. A range of optimal PCFs is derived based on the Price’s theorem. Computer simulation results show superiority in bit error rate (BER) performance that is very close to that potentially achieved and surpasses the BER performance of the real PCF for DS-CDMA systems discussed in literature.     

Projected least-squares algorithms for constrained FIR filter design

Constrained finite-impulse response (FIR) filter design with time- and frequency-domain linear constraints can be generally transformed into a, or a series of, constrained least-squares problems, which can be generally reformulated as positive definite quadratic programming (QP) problems. This paper presents a novel algorithm referred to as a projected least-squares (PLS) algorithm for the positive definite QP problems. The PLS algorithm essentially projects the unconstrained (least-squares) minimization solution successively onto the boundaries of active constraints that are identified by an active-set strategy. The PLS algorithm has been applied to the constrained least-squares design of FIR filters directly, and to the constrained Chebyshev design of FIR filters in an iterative fashion. The PLS algorithm is compared with the most widely used interior-point methods and an active-set method through design examples of low-pass filters with specified passband and stopband ripples, Nyquist filter constraints and step response constraints. All these examples demonstrate the high efficiency of the PLS algorithm.     

A new method for adaptive IIR filter design based on tabu search algorithm

Adaptive digital filters have proven their worth in a wide range of applications such as channel equalisation, noise reduction, echo cancelling, and system identification. These filters can be broadly classified into two groups: finite impulse–response (FIR) and infinite impulse–response (IIR) filters. IIR filters have become the target of increasing interest because these filters can reduce the filter order significantly as compared to FIR filters. Tabu search is a heuristic optimisation algorithm which has been originally developed for combinatorial optimisation problems. It simulates the general rules of intelligent problem solving and has the ability of discovering the global minima in a multi-modal search space. In this work, a novel method based on tabu search is described for the design of adaptive IIR filters.     

FIR digital filter design techniques using weighted Chebyshev approximation

This paper discusses the various approaches to designing FIR digital filters using the theory of weighted Chebyshev approximation. The different design techniques are explained and compared on the basis of their capabilities and limitations. The relationships between filter parameters are briefly discussed for the case of low-pass filters. Extensions of the theory to the problems of magnitude and complex approximation are also included, as are some recent results on the design of two-dimensional FIR filters by transformation.     

ISI-free FIR filterbank transceivers for frequency-selectivechannels

Discrete multitone modulation transceivers (DMTs) have been shown to be very useful for data transmission over frequency-selective channels. The DMT scheme is realized by a transceiver that divides the channel into subbands. The efficiency of the scheme depends on the frequency selectivity of the transmitting and receiving filters. The receiving filters with good stopband attenuation are also desired for combating narrowband noise. The filterbank transceiver or discrete wavelet multitone (DWMT) system has been proposed as an implementation of the DMT transceiver that has better frequency band separation, but usually, intersymbol interference (ISI) cannot be completely cancelled in these filterbank transceivers, and additional equalization is required. We show how to use over interpolated filterbanks to design ISI-free FIR transceivers. A finite impulse response (FIR) transceiver with good frequency selectivity can be designed, as demonstrated by the design examples     

应用MATLAB分析语音信号

语音信号处理是语音学与数字信号处理技术相结合的交叉学科.将语音当做一种特殊的信号,即一种“复杂向量”来看待.通过调用处理数字信号工具MATLAB里的命令函数,利用数字信号处理的知识来解决问题.像给一般信号做频谱分析一样,也分析了语音信号的频谱,即利用设计的多种数字滤波器,比如:FIR低通滤波器、高通滤波器、带通滤波器和...     

Passive Arm Filters in I and Q Receivers for MSK-Type Continuous Phase Modulations

The use of Butterworth and Chebyshev low-pass filters in suboptimum in-phase (I) and quadrature phase (Q) receivers is considered for several bandwidth efficient modulations. These modulations are: minimum shift keying (MSK), tamed frequency modulation with rectangular pulse shaping (TFMREC), and MSK with duobinary encoding of the source bits (DMSK). The sensitivity of the modulations with theirIandQreceivers to a noisy phase reference and to timing errors is also investigated. It was found that standard, passive low-pass filter receive filters performed quite well when compared to other nonstandard filters. The use of standard filters should thus be considered in any system design. The modulations, with their simple receivers, can be ranked in order of increasing sensitivity to timing errors as follows: MSK, DMSK, TFMREC. This ranking is the reverse to the one obtained when it is done in order of best bandwidth efficiency.     

Implementation of Digital Unbiased FIR Filters with Polynomial Impulse Responses

This paper addresses the design and implementation of digital unbiased finite impulse response (FIR) filters with polynomial impulse response functions. The transfer function, its fundamental properties, and a general block-diagram are discussed for the impulse response represented with the l-degree Taylor series expansion. As a particular results, we show a fundamental identity uniquely featured to such filters in the transform domain. For low-degree impulse responses, the transfer functions are found in simple closed forms and represented in compact block-diagrams. The magnitude and phase responses are also analyzed along with the group delays. A comparison with predictive FIR filters is given. As examples of applications, filtering of time errors of local clocks is discussed along with the low-pass filter design employing a cascade of the unbiased FIR filters.     

A unified approach to IFIR filter design using B-spline functions

An efficient procedure is presented for the design of interpolated finite impulse response (IFIR) filters with linear phase. The algorithm uses the uniform B-spline function as an interpolator and solves the optimal Chebyshev approximation problem on the appropriate subinterval. The technique can be used for the design of general low-pass, high-pass and bandpass filters. Although the number of multiplications of the IFIR filter is dependent on the bandwidth and the center frequency of the desired filter, this approach nearly always provides a substantial reduction in complexity when compared to other FIR and IFIR design procedures     

Turbocharging Interpolated FIR Filters [DSP Tips & Tricks]

Interpolated finite impulse response (IFIR) filters drastically improve the computational efficiency of low-pass finite impulse response (FIR) filters designed using the traditional Parks-McClellan design method. This article presents a trick to further improve the computational efficiency of traditional IFIR filters.     

Filter design for MIMO sampling and reconstruction

We address the problem of finite impulse response (FIR) filter design for uniform multiple-input multiple-output (MIMO) sampling. This scheme encompasses Papoulis' generalized sampling and several nonuniform sampling schemes as special cases. The input signals are modeled as either continuous-time or discrete-time multiband input signals, with different band structures. We present conditions on the channel and the sampling rate that allow perfect inversion of the channel. Additionally, we provide a stronger set of conditions under which the reconstruction filters can be chosen to have frequency responses that are continuous. We also provide conditions for the existence of FIR perfect reconstruction filters, and when such do not exist, we address the optimal approximation of the ideal filters using FIR filters and a minmax l/sub 2/ end-to-end distortion criterion. The design problem is then reduced to a standard semi-infinite linear program. An example design of FIR reconstruction filters is given.     

Broadband Beamforming Using TDL-Form IIR Filters

Conventional broadband beamforming structures make use of finite-impulse-response (FIR) filters in each channel. Large numbers of coefficients are required to retain the desired signal-to-interference-plus-noise-ratio (SINR) performance as the operating bandwidth increases. It has been proven that the optimal frequency-dependent array weighting of broadband beamformers could be better approximated by infinite-impulse-response (IIR) filters. However, some potential problems, such as stability monitoring and sensitivity to quantization errors, of the IIR filters make the implementation of the IIR beamformers difficult. In this paper, new broadband IIR beamformers are proposed to solve these problems. The main contributions of this paper include 1) the Frost-based and generalized sidelobe canceller (GSC)-based broadband beamformers utilizing a kind of tapped-delay-line-form (TDL-form) IIR filters are proposed; 2) the combined recursive Gauss-Newton (RGN) algorithm is designed to compute the feedforward and feedback weights in the Frost-based implementation; and 3) in the GSC-based structure, the unconstrained RGN algorithm is customized for the TDL-form IIR filters in the adaptive beamforming part. Compared with the beamformer using direct-form IIR filters, the new IIR beamformers offer much easier stability monitoring and less sensitivity to the coefficient quantization, while comparable SINR improvement over the conventional FIR beamformer is achieved     

Design of transmit FIR filters for FDM data transmission systems

The design of transmit finite-impulse response (FIR) filters with few coefficients in frequency division multiplexing data transmission systems is considered. In these systems, quality objectives are imposed for transmission over channels affected by additive white Gaussian noise (AWGN) and over channels affected by AWGN plus adjacent channel interference (ACI). The goal of this letter is: given that an adaptive receive filter, possibly cooperating with an adaptive decision feedback equalizer, is used for the AWGN channel and for the AWGN plus ACI channel, what is the best fixed FIR transmit filter to use for both channel cases? This goal is achieved by optimizing the compromise between the performance of the system on the two mentioned channels. Also, the advantages of the proposed design over a rival method based on a fixed receive filter are demonstrated.     

1D and 2D economical FIR filters generated by Chebyshev polynomials of the first kind

Christoffel–Darboux formula for Chebyshev continual orthogonal polynomials of the first kind is proposed to find a mathematical solution of approximation problem of a one-dimensional (1D) filter function in the z domain. Such an approach allows for the generation of a linear phase selective 1D low-pass digital finite impulse response (FIR) filter function in compact explicit form by using an analytical method. A new difference equation and structure of corresponding linear phase 1D low-pass digital FIR filter are given here. As an example, one extremely economic 1D FIR filter (with four adders and without multipliers) is designed by the proposed technique and its characteristics are presented. Global Christoffel–Darboux formula for orthonormal Chebyshev polynomials of the first kind and for two independent variables for generating linear phase symmetric two-dimensional (2D) FIR digital filter functions in a compact explicit representative form, by using an analytical method, is proposed in this paper. The formula can be most directly applied for mathematically solving the approximation problem of a filter function of even and odd order. Examples of a new class of extremely economic linear phase symmetric selective 2D FIR digital filters obtained by the proposed approximation technique are presented.     

A Reconfigurable Channel Filter for Software Defined Radio Using RNS

This paper presents a high-speed FIR channel filter using residue number system (RNS) whose frequency response can be reconfigured to adapt to a multitude of channel filtering specifications of a multi-standard software defined radio (SDR) receiver. The channel filters in the channelizer of an SDR extract multiple narrowband channels corresponding to different communication standards from the wideband input signal. The proposed architecture has been synthesized on TSMC 0.18 μm CMOS standard cell technology. Synthesis result shows that the proposed reconfigurable FIR channel filter, for a Digital Advanced Mobile Phone Systems (D-AMPS) example, offers speed improvement of 42% and AT complexity reduction of 26% over existing reconfigurable FIR method.     

Revisiting the doppler filter of LEO satellite GNSS receivers for precise velocity estimation

The theoretical aspects of the precise velocity determination of Low Earth Orbit (LEO) satellites’on board Global Navigation Satellite Systems (GNSS) receivers are derived. It shows that the receiver’s Phase Lock Loop (PLL) is required to feature extremely small group delay within its low frequency band, which is in contrast to existing work that proposed wide band linear phase filters. Following this theory, a Finite Impulse Response (FIR) filter is proposed. To corroborate, the proposed FIR filter and an Infinite Impulse Response (IIR) filter lately proposed in literals are implemented in a LEO satellite onboard GNSS receiver. Tests are conducted using a third party commercial GPS signal generator. The results show that the GNSS receiver with the proposed FIR achieves 11 mm/s R.M.S precision, while the GNSS receiver with the IIR filter has a filter-caused velocity error that can not be ignored for space borne GNSS receivers.     

A 1-V 3.5-mW CMOS switched-opamp quadrature IF circuitry for Bluetooth receivers

A 1-V switched-capacitor (SC) quadrature IF circuitry for Bluetooth receivers is demonstrated using switched-opamp technique. To achieve double power efficiency while maintaining low sensitivity to finite opamp gain effects for the SC IF circuitry, half-delay integrator-based filters and /spl Sigma//spl Delta/ modulator have been proposed. The proposed quadrature IF circuitry employs a seventh-order IF filter for channel selection and a third-order /spl Sigma//spl Delta/ modulator for analog-to-digital conversion. A noise-shaping extension technique is employed to enhance the resolution of the low-pass /spl Sigma//spl Delta/ modulator by 16 dB while operating at the same oversampling ratio and power consumption. At a 1-V supply, the quadrature IF circuitry achieves a measured IIP3 of -3 dBm at a nominal gain of 24 dB with a 48-dB variable gain control while consuming a total power dissipation of 3.5 mW.     

An FIR Channel Estimation Filter with Robustness to Channel Mismatch Condition

This letter proposes a finite impulse response (FIR) channel estimation filter that has robustness against the channel mismatch due to the FIR structure. The channel impulse response is described with a complex state space model and then estimated from received data on the recent time interval. Numerical results show that the FIR channel estimation filter can provide more robust performance than conventional Kalman IIR filters when channel model parameters are not correct.     

A Generic Open-Loop Algorithm for Three-Phase Grid Voltage/Current Synchronization With Particular Reference to Phase, Frequency, and Amplitude Estimation

This paper presents a new open-loop architecture for three-phase grid synchronization based on moving average and predictive filters, where accurate measurements of phase, frequency, and amplitude are carried out in real time. Previous works establish that the fundamental positive sequence vector of a set of utility voltage/current vectors can be decoupled using Park's transformation and low-pass filters. However, the filtering process introduces delays that impair the system performance. More specifically, when the input signal frequency is shifted above the nominal, a nonzero average steady-state phase error appears in the measurements. To overcome such limitations, a suitable combination of predictive and moving average finite impulse response (FIR) filters is used by the authors to achieve a robust synchronization system for all input frequencies. Moving average filters are linear phase FIR filters that have a constant time delay at low frequencies, a characteristic that is exploited to good effect to design a predictive filter that compensates such time delays, enabling zero steady-state phase errors for shifted input frequencies. In summary, the main attributes of the new system are its good frequency adaptation, good filtering/transient response tradeoff, and the fact that its dynamics is independent of the input vector amplitude. Comprehensive experimental results validate the theoretical approach and the high performance of the proposed synchronization algorithm.      

海洋重力测量中滤波方法比较研究

海洋重力测量数据中含有大量高频噪声,需要采用低通滤波器来提取实际的重力异常信息。该文结合实测的海洋重力数据研究了低通滤波器截止频率和滤波器长度的确定方法,设计了不同窗函数的有限脉冲响应(FIR)低通滤波器。经海洋重力测量数据处理试验结果表明,汉宁窗和海明窗的FIR低通滤波器处理的重复线内符合精度和交叉点不符值中误差均优于0.5 mGal(1 Gal=10^-2 m/s²),可满足海洋重力测量要求。