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1.
The paper proposes a low‐complexity concurrent constant modulus algorithm (CMA) and soft decision‐directed (SDD) scheme for fractionally spaced blind equalization of high‐order quadrature amplitude modulation channels. We compare our proposed blind equalizer with the recently introduced state‐of‐art concurrent CMA and decision‐directed (DD) scheme. The proposed CMA+SDD blind equalizer is shown to have simpler computational complexity per weight update, faster convergence speed, and slightly improved steady‐state equalization performance, compared with the existing CMA+DD blind equalizer. Copyright © 2004 John Wiley & Sons, Ltd.  相似文献   

2.
We propose an iterative blind interference reduction strategy for short‐burst coded DS‐CDMA systems. The blind strategy works by creating a set of ‘training sequences’ in the receiver that are used as input to an interference reduction algorithm whose task is to produce a corresponding set of equalizers that attempt to recover the desired signal. To maintain a reasonable complexity level we develop a semi‐blind interference reduction algorithm that is capable of equalizing the received signal with a relatively small training sequence length (thus maintaining a small training sequence set). The objective then becomes to determine which equalizer from the generated set gives the best performance (smallest bit error). It is demonstrated that the success of this scheme depends greatly on the ability to find an appropriate criterion for picking the best equalizer. Of the tested criteria, one based on feedback from the decoder (essentially using trellis information) is shown to achieve nearly optimal performance. Copyright © 2005 John Wiley & Sons, Ltd.  相似文献   

3.
高阶QAM实时多域测试多模式自适应盲均衡技术研究   总被引:4,自引:1,他引:3  
提出了一种全新的宽带通信信号实时多域分析通用架构,详细介绍了该架构下信号分析的基本原理。在这种架构的基础上,通过加载不同的算法,不仅能够实现各种宽带通信信号高精度实时宽带频谱分析,而且还能同时实现宽带通信信号时域、调制域等多域联合分析。针对宽带高阶正交幅度调制(QAM)通信信号实时多域分析,详细讨论了面向测试的基于GMMA和DDLMS双模自适应盲均衡算法。系统仿真结果证明:相比GMMA自适应盲均衡算法,双模自适应盲均衡算法收敛速度明显提高,256QAM信号均衡后输出残余码间串扰(ISI)改善提高了10dB;同时通过实验验证,采用20MHz实时分析带宽对码率为6.4MSps的宽带256QAM信号进行实时多域分析,误差矢量幅度(error vectorm agnitude,EVM)测试误差小于2%。  相似文献   

4.
In this paper we present fractionally spaced adaptive equalization techniques and space diversity combined receiver and evaluate their performance for the downlink of S‐UMTS system. The conventional ‘training’ (or non‐blind) and the ‘unsupervised’ (or blind) adaptive equalization algorithms are both investigated. Simulation results show that the equalizers are robust to Doppler shift and non‐linearity effects due to TWT amplifiers aboard the satellite. It is also shown that even with a moderate array size of two antenna elements, a significant improvement in terminal performance is achieved. Copyright © 2002 John Wiley & Sons, Ltd.  相似文献   

5.
Nonlinear equalisers based on minimum BER are proposed for the equalisation of nonlinear time‐varying channels. To train the equalisers online, a sliding‐window‐based hybrid quasi‐Newton algorithm is proposed. Switching between sliding‐window stochastic gradient algorithm and sliding‐window quasi‐Newton algorithm makes the new algorithm significantly stabler with a fast convergence rate. Results from extensive simulation tests show that performance of nonlinear equalisers based on minimum BER is better than the equaliser based on minimum mean square error. The proposed algorithm demonstrates high efficiency as well. Copyright © 2013 John Wiley & Sons, Ltd.  相似文献   

6.
针对数字通信系统中符号间干扰(ISI)问题,提出了一种适用于π/4 DQPSK解调的自适应盲均衡和载波恢复算法。T/4 CMA盲均衡器利用接收信号的所有采样点进行迭代,解决了采样相位敏感的问题;改进的载波恢复算法简单易于实现,减小了接收机设计的复杂度和难度。仿真结果表明,这种算法复杂度小、性能好,具有一定的实用价值。  相似文献   

7.
A new adaptive algorithm with fast convergence and low complexity is presented. By using the calculation structure of the dual Kalman variables of the fast transversal filter algorithm and a simple decorrelating technique for the input signal, we obtain an algorithm that exhibits faster convergence speed and enhanced tracking ability compared with the normalized least‐mean‐square algorithm with similar computational complexity. Copyright © 2013 John Wiley & Sons, Ltd.  相似文献   

8.
When input data are contaminated with noise, the least mean square time‐delay estimation (LMSTDE) actually gives a biased estimate. In this paper, we propose and analyse a new adaptive filter structure for time‐delay estimation (TDE), which can eliminate this bias. A new adaptive criterion is then constructed. We determine the corresponding analytical solution, and develop the stochastic gradient algorithm to calculate the optimum solution. Convergence of the stochastic gradient algorithm is established, and upper bounds on the step sizes are deduced for guaranteed convergence. Simulation results are included to illustrate the superiority of the new model and corroborate the theoretical developments. Copyright © 2007 John Wiley & Sons, Ltd.  相似文献   

9.
Series connections of energy storage cells, such as lithium‐ion cells and electric double‐layer capacitors (EDLCs), require cell‐voltage equalizers to ensure years of operation. Conventional equalizers require multiple switches, magnetic components, and/or secondary windings of a multiwinding transformer in proportion to the number of series connections, which usually makes them complex, expensive, bulky, and less extendable with increasing series connections. A double‐switch series‐resonant equalizer using a voltage multiplier is proposed in this paper. The double‐switch operation without a multiwinding transformer achieves simplified circuitry and good modularity at reduced size and cost, compared to conventional equalizers. Operational analyses were separately performed for the following two functional parts of the proposed equalizer: a series‐resonant inverter and a voltage multiplier. The mathematical analyses derived a dc‐equivalent circuit of the proposed equalizer, with which simulation analyses of even an hour's duration can be completed in an instant. Simulation analyses were separately performed for both the original and equivalent circuits. The simulation results of the derived circuit correlated well with those of the original circuit, thus verifying the derived dc‐equivalent circuit. A 5‐W prototype of the proposed equalizer was built for eight cells connected in series and an experimental equalization was performed for series‐connected EDLCs from an initially voltage‐imbalanced condition. The voltage imbalance was gradually eliminated over time, and the standard deviation in the cell voltages decreased to approximately 5 mV at the end of the experiment, thus demonstrating the equalization performance of the proposed equalizer.  相似文献   

10.
An optimal adaptive control technique for the discrete linear systems is discussed in this paper. The system parameters are unknown and one‐step‐ahead adaptive control design is based on the input matching approach and the weighted least‐squares (WLS) algorithm. It is shown that the adaptive stochastic system is globally closed‐loop stable and the system identification is consistent. The adaptive controller converges to the one‐step‐ahead optimal controller. Finally, some simulation examples are given to demonstrate the reliability of the new optimal adaptive control algorithm. Copyright © 2008 John Wiley & Sons, Ltd.  相似文献   

11.
In high-bit-rate optical transmission systems, distortions due to dynamic chromatic dispersion, polarization mode dispersion, and power changes are larger than the distortion tolerances of the system. To meet the tolerances and the desired quality of service, an adaptive equalizer is necessary. We demonstrate the capabilities of planar lightwave circuit integrated optical finite impulse response filters for mitigating distortions of the transmission channel, and we investigate two adaptive equalization approaches. The first approach uses an adaptive feedback generated from electrical spectrum monitoring; the second one uses intersymbol interference minimization with a least mean square error algorithm. We successfully demonstrated adaptive equalization of chromatic dispersion, self-phase modulation, and polarization mode dispersion, as well as combinations of these distortions.  相似文献   

12.
Minimum output energy (MOE) algorithm is a widely used adaptive algorithm for blind adaptation of infinite impulse response (IIR) filters. In this paper, we show that the MOE algorithm is not suitable for blind adaptation of the complex‐valued IIR equalizer for digital vestigial sideband signals, whereas the constant modulus algorithm successfully achieves blind adaptation of the IIR equalizers when MOE fails. Because of the difficulty in analyzing IIR equalizers, the analysis is limited to a simple two‐tap channel case. For more general multitap channel cases, the performance of a complex constant modulus algorithm IIR is evaluated through simulation. Copyright © 2011 John Wiley & Sons, Ltd.  相似文献   

13.
A new non‐linear adaptive filter called blind image deconvolution via dispersion minimization has recently been proposed for restoring noisy blurred images blindly. This is essentially a two‐dimensional version of the constant modulus algorithm that is well known in the field of blind equalization. The two‐dimensional extension has been shown capable of reconstructing noisy blurred images using partial a priori information about the true image and the point spread function in a variety of situations by means of simulations. This paper analyses the behaviour of the algorithm by investigating the static properties of the cost function and the dynamic convergence of the parameter estimates. The theoretical results are supported with computer simulations. Copyright © 2006 John Wiley & Sons, Ltd.  相似文献   

14.
This paper proposes a new Steiglitz–McBride (SM) adaptive notch filter (SM‐ANF) based on a robust variable‐step‐size least‐mean‐square algorithm and its application to active noise control (ANC). The proposed SM‐ANF not only has fast convergence but also has small misadjustment. The variable‐step‐size algorithm uses the sum of the squared cross correlation between the error signal and the delayed inputs corresponding to the adaptive weights. The cross correlation provides robustness to the broadband signal, which plays the role of noise. The proposed SM‐ANF is computationally simpler than the existing Newton/recursive least‐squares‐type ANF. The frequency response of the new SM‐ANF has a notch depth of about ?25 dB (for each of the three frequencies considered) and has spectral flatness within 5 dB (peak to peak). This robust notch filter algorithm is used as an observation noise canceller for the secondary path estimation of an ANC system based on the SM method. The ANC with proposed SM‐ANF provides not only faster convergence but also an 11‐dB improvement in noise attenuation over the SM‐based ANC without such a SM‐ANF. Copyright © 2015 John Wiley & Sons, Ltd.  相似文献   

15.
This study addresses the problem of speech quality enhancement by adaptive and nonadaptive filtering algorithms. The well‐known two‐microphone forward blind source separation (TM‐FBSS) structure has been largely studied in the literature. Several two‐microphone algorithms combined with TM‐FBSS have been recently proposed. In this study, we propose 2 contributions: In the first, a new two‐microphone Gauss‐Seidel pseudo affine projection (TM‐GSPAP) algorithm is combined with TM‐FBSS. In the second, we propose to use the new TM‐GSPAP algorithm in speech enhancement. Furthermore, we show the efficiency of the proposed TM‐GSPAP algorithm in speech enhancement when highly noisy observations are available. To validate the good performances of our algorithm, we have evaluated the adaptive filtering properties in computational complexity and convergence speed performance by system mismatch criteria. A fair comparison with adaptive and nonadaptive noise reduction algorithms are also presented. The adaptive algorithms are the well‐known two‐microphone normalized least mean square algorithm, and the recently published two‐microphone pseudo affine projection algorithm. The nonadaptive algorithms are the one‐microphone spectral subtraction and the two‐microphone Wiener filter algorithm. We evalute the quality of the output speech signal in each algorithm by several objective and subjective criteria as the segmental signal‐to‐noise ratio, cepstral distance, perceptual evaluation of speech quality, and the mean opinion score. Finally, we validate the superior performances of the proposed algorithm with physically measured signals.  相似文献   

16.
This paper describes a new solution to channel intersymbol interference and nonlinear distortion by means of an adaptive array antenna and radial basis function (RBF) equalizer. The RBF equalizer can reduce the influence of intersymbol interference and additive noise in the digital communication system. It is also well known to eliminate nonlinear distortion effectively. However, conventional RBF equalizers including our previous proposed equalizer, a spatial and temporal RBF equalizer using adaptive array antenna, need training data to learn channel states, which incurs a decline in the data rate. In this paper, without training data, we construct a blind spatial and temporal RBF equalizer using genetic algorithm. We also describe a method of synthesizing Bayesian decision variables of RBF equalizers in order to obtain better performance. The RBF equalizer based on the proposed method can achieve a remarkable improvement in BER characteristics and its performance is demonstrated by computer simulations. © 2005 Wiley Periodicals, Inc. Electr Eng Jpn, 152(4): 50–56, 2005; Published online in Wiley InterScience ( www.interscience.wiley.com ). DOI 10.1002/eej.20123  相似文献   

17.
Recently, sparsity‐aware least mean square (LMS) algorithms have been proposed to improve the performance of the standard LMS algorithm for various sparse signals, such as the well‐known zero‐attracting LMS (ZA‐LMS) algorithm and its reweighted ZA‐LMS (RZA‐LMS) algorithm. To utilize the sparsity of the channels in wireless communication and one of the inherent advantages of the RZA‐LMS algorithm, we propose an adaptive reweighted zero‐attracting sigmoid functioned variable‐step‐size LMS (ARZA‐SVSS‐LMS) algorithm by the use of variable‐step‐size techniques and parameter adjustment method. As a result, the proposed ARZA‐SVSS‐LMS algorithm can achieve faster convergence speed and better steady‐state performance, which are verified in a sparse channel and compared with those of other popular LMS algorithms. The simulation results show that the proposed ARZA‐SVSS‐LMS algorithm outperforms the standard LMS algorithm and the previously proposed sparsity‐aware algorithms for dealing with sparse signals. Copyright © 2015 John Wiley & Sons, Ltd.  相似文献   

18.
An adaptive continuous‐time equalizer for reliable short‐haul high‐speed serial communications is described in this paper. The adaptive equalizer uses the spectrum‐balancing technique to adapt its response to changes in the bandwidth, amplitude, and bit rate of the input signal. In this way, it is able to compensate the frequency response of a 1‐mm diameter step‐index plastic optical fiber, for lengths up to 50 m, and bit rates ranging from 400 Mb/s to 2.5 Gb/s. Experimental results are shown to demonstrate its feasibility. Copyright © 2017 John Wiley & Sons, Ltd.  相似文献   

19.
In this paper, a novel adaptive filter for sparse systems is proposed. The proposed algorithm incorporates a log‐sum penalty into the cost function of the standard leaky least mean square (LMS) algorithm, which results in a shrinkage in the update equation. This shrinkage, in turn, enhances the performance of the adaptive filter, especially, when the majority of unknown system coefficients are zero. Convergence analysis of the proposed algorithm is presented, and a stability criterion for the algorithm is derived. This algorithm is given a name of zero‐attracting leaky‐LMS (ZA‐LLMS) algorithm. The performance of the proposed ZA‐LLMS algorithm is compared to those of the standard leaky‐LMS and ZA‐LMS algorithms in sparse system identification settings, and it shows superior performance compared to the aforementioned algorithms. Copyright © 2013 John Wiley & Sons, Ltd.  相似文献   

20.
When electric double‐layer capacitors (EDLCs) are connected in series, a cell voltage imbalance occurs due to nonuniform cell properties. Cell voltage imbalance should be minimized to prolong cycle lives and maximize the available energy of cells. In this study, we propose a series‐parallel reconfigurable cell voltage equalizer that is considered suitable for energy storage systems using EDLCs instead of traditional secondary batteries as the main energy storage sources. The proposed equalizer requires only EDLCs and switches as its main circuit elements, and it utilizes EDLCs not only for energy storage but also for equalization. An equivalent circuit model using equivalent resistors that can be regarded as an index of equalization speed is developed. Current distribution and cell voltage imbalancing during operation are quantitatively generalized. Experimental charge–discharge tests were performed on the EDLC modules to demonstrate the performance of the cell voltage equalizer. All the cells in the modules could be charged/discharged uniformly even when a degradation‐mimicking cell was intentionally included in the module. The resultant cell voltage imbalances and current distributions were in good agreement with those predicted by mathematical analyses. © 2012 Wiley Periodicals, Inc. Electr Eng Jpn, 181(4): 38–50, 2012; Published online in Wiley Online Library (wileyonlinelibrary.com). DOI 10.1002/eej.21287  相似文献   

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