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1.
多参考输入自适应噪声抵消器的算法与实现   总被引:5,自引:0,他引:5  
针对单参考办入自适应噪声抵消器在多噪声源情况下除噪效果不理想的问题,根据多参考输入自适应噪声抵消器模型,推导了多参考输入最小均方算法,并给出其改进算法。以语音信号为例,给出了多参考输出自适应语音消噪算法及多参考输入自适应语音噪声抵消器的实现方案,并进行了计算机仿真,给出了仿真结果。  相似文献   

2.
高噪声环境下基于自适应滤波语音降噪技术的研究   总被引:2,自引:1,他引:2  
郑海啸  刘珩 《电子测量技术》2007,30(7):16-19,23
为了在高噪声环境下得到较清晰的语音信息,本文提出一种基于线性预测的FIR自适应语音滤波器,对LMS算法作出改进,提高了算法的收敛性。在多种噪声环境下,MATLAB仿真实验结果表明,改进的算法具有更好的降噪效果,显著提高了信噪比,并保持了语音的自然度。  相似文献   

3.
An adaptive noise reduction filter composed of a sandglass‐type neural network (SNN) noise reduction filter (RF) is proposed in this paper. SNN was originally devised to work effectively for information compression. It is a hierarchial network and is symmetrically structured. SNN consists of the same number of units in the input and output layers and a smaller number of units in the hidden layer. It is known that SNN has signal processing performance which is equivalent to Karhunen–Loeve expansion after learning. We proved the theoretical suitability of SNN for an adaptive noise reduction filter for discrete signals. The SNNRF behaves optimally when the number of units in the hidden layer is equal to the rank of the covariance matrix of the signal components included in the input signal. Further we show by applying the recursive least squares method to learning of the SNNRF that the filter can process signals for on‐line adaptive noise reduction. This is an extremely desirable feature for practical application. In order to verify the validity of SNNRF, we performed computer experiments examining how the noise reduction ability of SNNRF is affected by altering the properties of the input pattern, learning algorithm, and SNN. The results confirm that the SNNRF acquired appropriate characteristics for noise reduction from the input signals, and markedly improved the SNR of the signals. © 1999 Scripta Technica, Electr Eng Jpn, 127(4): 39–51, 1999  相似文献   

4.
A novel low computational complexity robust adaptive blind multiuser detector, based on the minimum output energy (MOE) detector with multiple constraints and a quadratic inequality (QI) constraint is developed in this paper. Quadratic constraint has been a widespread approach to improve robustness against mismatch errors, uncertainties in estimating the data covariance matrix, and random perturbations in detector parameters. A diagonal loading technique is compulsory to achieve the quadratic constraint where the diagonal loading level is adjusted to satisfy the constrained value. Integrating the quadratic constraint into recursive algorithms seems to be a moot point since there is no closed‐form solution for the diagonal loading term. In this paper, the MOE detector of DS/CDMA system is implemented using a fast recursive steepest descent adaptive algorithm anchored in the generalized sidelobe canceller (GSC) structure with multiple constraints and a QI constraint on the adaptive portion of the GSC structure. The Lagrange multiplier method is exploited to solve the QI constraint. An optimal variable loading technique, which is capable of providing robustness against uncertainties and mismatch errors with low computational complexity is adopted. Simulations for several mismatch and random perturbations scenarios are conducted in a rich multipath environment with near–far effect to explore the robustness of the proposed detector. Copyright © 2008 John Wiley & Sons, Ltd.  相似文献   

5.
A direct adaptive non‐linear control framework for multivariable non‐linear uncertain systems with exogenous bounded disturbances is developed. The adaptive non‐linear controller addresses adaptive stabilization, disturbance rejection and adaptive tracking. The proposed framework is Lyapunov‐based and guarantees partial asymptotic stability of the closed‐loop system; that is, asymptotic stability with respect to part of the closed‐loop system states associated with the plant. In the case of bounded energy L2 disturbances the proposed approach guarantees a non‐expansivity constraint on the closed‐loop input–output map. Finally, several illustrative numerical examples are provided to demonstrate the efficacy of the proposed approach. Copyright © 2002 John Wiley & Sons, Ltd.  相似文献   

6.
This paper addresses the problem of acoustic echo cancellation. We propose a new version of the fast Newton transversal filter algorithm for stereophonic acoustic echo cancellation applications. This algorithm can be viewed as an efficient implementation of the extended two‐channel fast transversal filter algorithm. Moreover, it fits naturally within the frame of the fast version of the recursive least‐squares (RLS) algorithm, applied to the two‐channel case. To stabilize the proposed two‐channel algorithm, we have adapted and then applied a new numerical stabilization technique that has been proposed recently. The computational complexity of the proposed two‐channel algorithm is less than half the complexity of the fastest two‐channel RLS versions and very close to that of the two‐channel normalized least mean squares algorithm when its predicting part length is chosen to be small. Simulation results and comparisons in term of complexities, convergence speed and tracking with the two‐channel algorithms are presented. Copyright © 2009 John Wiley & Sons, Ltd.  相似文献   

7.
In this paper, a new adaptive robust stabilization scheme is proposed for uncertain neutral time‐delay systems. No upper bounds on the uncertainties are assumed to be available. An update law is first used to find estimates of these upper bounds. A state‐feedback controller is then designed, which is shown to stabilize the underlying system under some mild conditions. The asymptotic stability of the state trajectories is proved using the Lyapunov–Krasovskii approach. An example is provided, which demonstrates the efficacy of the proposed adaptive control scheme. Copyright © 2012 John Wiley & Sons, Ltd.  相似文献   

8.
A probabilistic indirect adaptive controller is proposed for the general nonlinear multivariate class of discrete time system. The proposed probabilistic framework incorporates input–dependent noise prediction parameters in the derivation of the optimal control law. Moreover, because noise can be nonstationary in practice, the proposed adaptive control algorithm provides an elegant method for estimating and tracking the noise. For illustration purposes, the developed method is applied to the affine class of nonlinear multivariate discrete time systems and the desired result is obtained: the optimal control law is determined by solving a cubic equation and the distribution of the tracking error is shown to be Gaussian with zero mean. The efficiency of the proposed scheme is demonstrated numerically through the simulation of an affine nonlinear system. Copyright © 2010 John Wiley & Sons, Ltd.  相似文献   

9.
The conventional unscented Kalman filter (UKF) requires prior knowledge on system noise statistics. If the statistical characteristics of system noise are not known exactly, the filtering solution will be biased or even divergent. This paper presents an adaptive UKF by combining the windowing and random weighting concepts to address this problem. It extends the windowing concept from the linear Kalman filter to the nonlinear UKF for estimation of system noise statistics. Subsequently, the random weighting concept is adopted to refine the obtained windowing estimation by adjusting random weights of each window. The proposed adaptive UKF overcomes the limitation of the conventional UKF by online estimating and adjusting system noise statistics. Experimental results and comparison analysis demonstrate that the proposed adaptive UKF outperforms the conventional UKF and adaptive robust UKF under the condition without precise knowledge on system noise statistics.Copyright © 2014 John Wiley & Sons, Ltd.  相似文献   

10.
直流配电网包含DC/DC变换器等电力电子器件,非线性特性显著,导致直流输出端电压、电流信号存在大量纹波,需通过滤波降噪处理提升直流电能计量的准确性。针对现有的滤波降噪方法参数设置缺乏优化、滤波降噪效果尚待提升问题,本文提出基于自适应变分模态分解与小波阈值去噪相结合的直流电能计量数据降噪方法。建立输出端直流电压、电流信号变分模态分解的参数最优化模型,并联合互信息分析,实现原始信号的有效模态分量与噪声模态分量的自适应区分。在此基础上,建立以信噪比、均方根误差、平滑度、相关系数复合评价指标最优的小波阈值去噪参数最优化模型,实现噪声模态分量的最优滤波降噪。通过实测数据计算分析,验证所提方法的有效性。  相似文献   

11.
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   

12.
Adaptive filter has been applied in adaptive feedback and feedforward control systems, where the filter dimension is often determined by trial‐and‐error. The controller design based on a near‐optimal adaptive filter in digital signal processor (DSP) is developed in this paper for real‐time applications. The design integrates the adaptive filter and the experimental design such that their advantages in stability and robustness can be combined. The near‐optimal set of controller parameters, including the sampling rate, the dimension of system identification model, the dimension (order) of adaptive controller in the form of an FIR filter, and the convergence rate of adaptation is shown to achieve the best possible system performance. In addition, the sensitivity of each design parameter can be determined by analysis of means and analysis of variance. Effectiveness of the adaptive controller on a DSP is validated by an active noise control experiment. Copyright © 2007 John Wiley & Sons, Ltd.  相似文献   

13.
In real‐time environments a speech recognition system in a car has to receive the driver's voice only while suppressing the background noise. This paper presents a hybrid real‐time adaptive filter that operates within a geometrical zone defined around the head of the desired speaker. Any sound outside of this zone is considered to be noise and is suppressed. As this defined geometrical zone is small, it is assumed that only driver's speech is incoming from this zone. The technique uses three microphones to define a geometric‐based voice–activity detector (VAD) to cancel the unwanted speech coming from outside of the zone. However, when unwanted speech and desired speech are incoming at the same time, the VAD fails to identify the unwanted speech or desired speech. In such a situation an adaptive Wiener filter is switched on for noise reduction. In the case of sole unwanted speech incoming from outside of a desired zone, this speech is muted at the output of the hybrid noise canceller. In the case of desired and unwanted speeches incoming together, the SNR is improved by as much as 20 dB. Copyright © 2009 John Wiley & Sons, Ltd.  相似文献   

14.
A recursive smoothing filter employing a bank of fading‐memory polynomial sub‐filters is presented. Variance estimates are used to mix the outputs of the sub‐filters, imparting variable gain and phase characteristics that permit it to automatically adapt to signal parameter changes. The proposed adaptive technique does not involve the estimation of plant parameters; therefore, it may be used in both open‐loop and closed‐loop configurations. In open‐loop estimation problems, variable gain/bandwidth allows it to reduce the impact of random errors caused by sensor noise and the impact of bias errors caused by model mismatch during ‘maneuvers’. In feedback control problems, variable phase/delay allows it to act as a lag filter for an improved steady‐state response (i.e. greater noise attenuation) and act as a lead filter, or a proportional‐derivative controller, for an improved transient response (i.e. increased closed‐loop damping) for input discontinuities. Copyright © 2015 John Wiley & Sons, Ltd.  相似文献   

15.
This paper is devoted to robust adaptive sliding mode control for time‐delay systems with mismatched parametric uncertainties. Sufficient conditions for the existence of linear sliding surfaces are given in terms of linear matrix inequalities, by which the corresponding adaptive reaching motion controller is also designed. Simulation studies show the effectiveness of the control scheme. Copyright © 2008 John Wiley & Sons, Ltd.  相似文献   

16.
In this contribution, a steady‐state approach for determining the optimal size and control of a shunt hybrid filter (SHF), to control harmonic current mitigation and to provide reactive power compensation, is proposed. The SHF topology is formed by a shunt active power filter (APF) and a shunt capacitor. The APF current injections are determined from the solution of a nonlinear programming problem formulated to meet permissible operation limits, with an optimal APF size. The formulation and control theory for the SHF is developed in the abc reference frame. An important practical aspect such as the application of SHF compensation in non‐stiff systems is included in the analysis and solution of the nonlinear programming problem, as well as in the current control technique, maintaining stringent performance requirements on the tracking of the filtering currents, by allowing the use of the shunt capacitor also as a filter for draining the ripple current inherent to the APF injection currents. Results obtained with matlab /Simulink (MathWorks, Inc., Natick, MA, USA) show that the proposed theory and the control of the optimal SHF compensation constitute an effective system to compensate reactive power and to control harmonic distortion under selected/permissible limits with an optimal APF injection current reducing the APF size. Copyright © 2011 John Wiley & Sons, Ltd.  相似文献   

17.
The convergence problems of conventional DC analysis can be partly avoided by using piecewise‐linear analysis. This paper proposes a piecewise‐linear DC analysis method that can efficiently handle arbitrary couplings between non‐linear circuit elements. Piecewise‐linear modelling of the non‐linear circuit elements is automatically performed during simulation, using simplicial subdivisions. The number of linear regions, and thereby iterations, is considerably reduced by combining the common parts of separate simplicial subdivisions. Due to these reasons and since the method is formulated with the commonly used modified nodal approach, it has been possible to implement the method in the general‐purpose circuit simulator APLAC. The correct operation of the method is demonstrated with three examples. Copyright © 1999 John Wiley & Sons, Ltd.  相似文献   

18.
We consider the problem of distributed state estimation over a sensor network in which a set of nodes collaboratively estimates the state of a continuous‐time linear time‐varying system. In particular, our work focuses on the benefits of weight adaptation of the interconnection gains in distributed Kalman filters. To this end, an adaptation strategy is proposed with the adaptive laws derived via a Lyapunov‐redesign approach. The justification for the gain adaptation stems from a desire to adapt the pairwise difference of state estimates as a function of their agreement, thereby enforcing an interconnection‐dependent gain. In the proposed scheme, an adaptive gain for each pairwise difference of the interconnection terms is used in order to address edge‐dependent differences in the state estimates. Accounting for node‐specific differences, a special case of the scheme is also presented, where it uses a single adaptive gain in each node estimate and which uniformly penalizes all pairwise differences of state estimates in the interconnection term. The filter gains can be designed either by standard Kalman filter or Luenberger observer to construct the adaptive distributed Kalman filter or adaptive distributed Luenberger observer. Stability of the schemes has been shown, and it is not restricted by the graph topology and therefore the schemes are applicable to both directed and undirected graphs. The proposed algorithms offer a significant reduction in communication costs associated with information flow by the nodes. Finally, numerical studies are presented to illustrate the performance and effectiveness of the proposed adaptive distributed Kalman filters. Copyright © 2015 John Wiley & Sons, Ltd.  相似文献   

19.
A method for the linear least‐squares estimation of random signals contaminated with random noise that uses a new method of spectral factorization is shown. It is shown that the optimal filter can be written entirely in terms of the two spectral factors of signal plus noise and noise‐alone, and can be applied to the general case of coloured and white additive noise. The method of spectral factorization used is novel and uses control‐system methodology. Copyright © 2012 John Wiley & Sons, Ltd.  相似文献   

20.
In this paper, an adaptive switching control algorithm is proposed for the stabilization of uncertain discrete‐time systems with time‐varying delay. It is assumed that the time delay is unknown and time varying, nonetheless bounded with a known bound. It is supposed that the system is highly uncertain, and that a set of controllers are designed (off‐line) to stabilize the system in the whole uncertain parameter space; subsequently, a switching scheme is developed to stabilize the uncertain time‐delay system. A thorough stability analysis for the uncertain time‐delay system under the mentioned control scheme is provided. Furthermore, an upper bound on the allowable rate of change of the system parameters and delay is obtained. Simulation results are presented to show the efficacy of the proposed switching scheme. Copyright © 2008 John Wiley & Sons, Ltd.  相似文献   

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