Convergence of partitioned adaptive filters for systems with unknown biases

A convergence theorem on performance bounds for partitioned adaptive estimation is extended to include biases among the unknown parameters. The importance and implications of this rarely considered case are briefly discussed.     

Convergence of continuous-time partitioned adaptive state estimators

The asymptotic behaviour of Bayes optimal adaptive state estimation schemes (also called the partitioned adaptive estimation algorithms) for continuous-time linear dynamic Gauss-Markov systems with unknown parameters is investigated. The unknown system parameters are asssumed to belong to a finite set. The results are developed through, weak consistency of the maximum likelihood and the maximum a posteriori probability estimates of the unknown parameters.     

Generalized spline nonlinear adaptive filters

A new nonlinear filter, which employs an adaptive spline function as the basis function is designed in this paper. The input signal to this filter is used to generate suitable parameters to update the control points in a spline function. The update rule for updating the control points have been derived and a mean square analysis has been carried out. The output of the spline functions are suitably combined together to obtain the filter response. This filter is called the generalized spline nonlinear adaptive filter (GSNAF). The proposed GSNAF is similar to a functional link artificial neural network (FLANN), considering a functional expansion using spline basis functions. GSNAF has been shown to offer improved accuracy in benchmark classification scenarios and provide enhanced modeling accuracy in single input single output as well as in multiple input multiple output dynamic system identification cases.     

Secure computation with horizontally partitioned data using adaptive regression splines

When several data owners possess data on different records but the same variables, known as horizontally partitioned data, the owners can improve statistical inferences by sharing their data with each other. Often, however, the owners are unwilling or unable to share because the data are confidential or proprietary. Secure computation protocols enable the owners to compute parameter estimates for some statistical models, including linear regressions, without sharing individual records’ data. A drawback to these techniques is that the model must be specified in advance of initiating the protocol, and the usual exploratory strategies for determining good-fitting models have limited usefulness since the individual records are not shared. In this paper, we present a protocol for secure adaptive regression splines that allows for flexible, semi-automatic regression modeling. This reduces the risk of model mis-specification inherent in secure computation settings. We illustrate the protocol with air pollution data.     

Secure computation with horizontally partitioned data using adaptive regression splines

When several data owners possess data on different records but the same variables, known as horizontally partitioned data, the owners can improve statistical inferences by sharing their data with each other. Often, however, the owners are unwilling or unable to share because the data are confidential or proprietary. Secure computation protocols enable the owners to compute parameter estimates for some statistical models, including linear regressions, without sharing individual records’ data. A drawback to these techniques is that the model must be specified in advance of initiating the protocol, and the usual exploratory strategies for determining good-fitting models have limited usefulness since the individual records are not shared. In this paper, we present a protocol for secure adaptive regression splines that allows for flexible, semi-automatic regression modeling. This reduces the risk of model mis-specification inherent in secure computation settings. We illustrate the protocol with air pollution data.     

On-line identification scheme using adaptive filters

An on-line scheme for identifying a linear process is proposed which consists of a linear time-varying filter and a parameter update algorithm. The disturbances affecting the process, its input and its output, belong to a general class of signals which are a mixture of stochastic and deterministic signal processes generated by some linear time-invariant system excited by white noise and the Dirac delta function, respectively. The process and the disturbance signal models are not restricted to be asymptotically stable. Either a probing input signal or a normal operating input signal can be employed. The probing signal consists of a finite number of sinusoidal signals (exponentially increasing sinusoidal signals for unstable processes) of distinct frequencies. When a normal operating signal is used, an adaptive scheme is employed to tune the parameters of the filters to the distinct frequency components of the signal. The convergence of the parameter estimates to their true value is established.     

Optimization of adaptive identification for time-varying filters

Adaptive identification in a time-varying context is studied when controlled by the LMS algorithm with constant gain μ, under the assumption of correlated successive input vectors, it is well known by experience that the tracking mean square error (MSE)epsilon(mu)results from the tradeoff between the gradient part which is μ-increasing and the lag contribution which is μ-decreasing. In this note we clarify the relative roles of the gradient and lag errors by proving their decoupled character. This property relies upon independence between the additive noise at the output of the plant to be identified and the information vector at the plant input. Convergence of the MSE is established rather than assumed. Quantitative evaluations of upper and lower bounds allow an approximate optimization of the gain. In two important cases the optimum is exact. One of these cases is "slow-variations." It is defined in a quantitative manner thanks to the ratio of the "variation" noise to the output additive noise.     

Efficient algorithms of clustering adaptive nonlinear filters

This paper proposes a new class of efficient adaptive nonlinear filters whose estimation error performance (in a minimum mean square sense) is superior to that of competing approximate nonlinear filters, e.g., the well-known extended Kalman filter (EKF). The proposed filters include as special cases both the EKF and previously proposed partitioning filters. The new methodology performs an adaptive selection of appropriate reference points for linearization from an ensemble of generated trajectories that have been processed and clustered accordingly to span the whole state space of the desired signal. Through a series of simulation examples, the approach is shown significantly superior to the classical EKF with comparable computational burden     

The effect of noise on adaptive envelope-constrained filters

The performance of the envelope-constrained filters designed by the newly developed adaptive algorithms may deteriorate when the input signal is contaminated by noise. The average window with a growing window size, and the exponentially weighted average window are presented. It is shown that the noise-induced bias in the augmented cost function will be reduced because applying such an averaging scheme helps increase values of the signal-to-noise ratio at the filter input channel. With this, the noise effect on the adaptive envelope-constrained filters can be depressed. Simulation results are given to illustrate the improved performance of the adaptive envelope-constrained filters designed in conjunction with the average window.     

模型参考自适应IIR递归滤波器辨识新算法

针对自适应IIR滤波器算法容易陷入局部极小点的缺陷,提出了一种新的自适应递归滤波辨识算法.该算法采用模型参考自适应系统设计了辨识参数自适应律,基于Lyapunov理论保证了自适应递归算法的稳定性,而且辨识参数收敛.仿真结果表明了该算法的可行性和滤波器结构的正确性.     

Noisy speech processing by recurrently adaptive fuzzy filters

Two noisy speech processing problems-speech enhancement and noisy speech recognition-are dealt with. The technique we focus on is by using the filtering approach; a novel filter, the recurrently adaptive fuzzy filter (RAFF), is proposed and applied to these two problems. The speech enhancement is based on adaptive noise cancellation with two microphones, where the RAFF is used to eliminate the noise corrupting the desired speech signal in the primary channel. As to the noisy speech recognition, the RAFF is used to filter the noise in the feature domain of speech signals. The RAFF is inherently a recurrent multilayered connectionist network for realizing the basic elements and functions of dynamic fuzzy inference, and may be considered to be constructed from a series of dynamic fuzzy rules. As compared to other existing nonlinear filters, three major advantages of the RAFF are observed: 1) a priori knowledge can be incorporated into the RAFF, which makes the fusion of numerical data and linguistic information possible; 2) owing to the dynamic property of the RAFF, the exact lagged order of the input variables need not be known in advance; 3) no predetermination, like the number of hidden nodes, must be given since the RAFF can find its optimal structure and parameters automatically Several examples on adaptive noise cancellation and noisy speech recognition problems using the RAFF are illustrated to demonstrate the performance of the RAFF     

A class of adaptive directional image smoothing filters

The gray level distribution around a pixel of an image usually tends to be more coherent in some directions compared to other directions. The idea of adaptive directional filtering is to estimate the direction of higher coherence around each pixel location and then to employ a window which approximates a line segment in that direction. Hence, the details of the image may be preserved while maintaining a satisfactory level of noise suppression performance. In this paper we describe a class of adaptive directional image smoothing filters based on generalized Gaussian distributions. We propose a measure of spread for the pixel values based on the maximum likelihood estimate of a scale parameter involved in the generalized Gaussian distribution. Several experimental results indicate a significant improvement compared to some standard filters.     

On state variables filters for adaptive system design

An alternative way of using the state variable filters, coupled with the Kalman-Yacubovich lemma or Popov's hyperstability theorem, in the design of model reference adaptive systems is proposed. The improvement over previous designs is the increased flexibility in the selection of the filter parameters to reduce the effects of noise or initial conditions.     

Extended adaptive lattice filters for disturbance input processes

The adaptive lattice filter is characterized as an input-dependent filter lying in an impulse disturbance. A modified recursive least-squares algorithm for bad data input is obtained by ‘dead zone’ control of the so-called reflection coefficients. The constraint-input constraint-output for stability analysis of the adaptive filter is specified as a sufficient condition.     

Modelling of brain consciousness based on collaborative adaptive filters

A novel method for the discrimination between discrete states of brain consciousness is proposed, achieved through examination of nonlinear features within the electroencephalogram (EEG). To allow for real time modes of operation, a collaborative adaptive filtering architecture, using a convex combination of adaptive filters is implemented. The evolution of the mixing parameter within this structure is then used as an indication of the predominant nature of the EEG recordings. Simulations based upon a number of different filter combinations illustrate the suitability of this approach to differentiate between the coma and quasi-brain-death states based upon fundamental signal characteristics.     

基于HVS特性的自适应K近邻均值滤波算法

通过充分考虑宿主图像亮度、纹理、边缘等特征,提出一种改进的图像自适应K近邻均值滤波算法。该方法首先利用基于人眼视觉特性的临界噪声阈值来确定噪声点,然后根据噪声密度自适应调整滤波窗口大小与参与滤波的像素数K值,采用自适应K近邻均值滤波对检测出的噪声点进行处理。该算法能有效去除噪声,并较好地保留图像边缘细节,仿真实验结果表明,提出算法比传统中值滤波、均值滤波和K近邻均值滤波算法有更好的去噪能力。     

Distractor-aware visual tracking using hierarchical correlation filters adaptive selection

Applied Intelligence - In recent years, the ensembled trackers composed of multi-level features from the pre-trained Convolutional Neural Network (CNN) have achieved top performance in visual...     

Fuzzy adaptive filters, with application to nonlinear channelequalization

Two fuzzy adaptive filters are developed: one uses a recursive-least-squares (RLS) adaptation algorithm, and the other uses a least-mean-square (LMS) adaptation algorithm. The RLS fuzzy adaptive filter is constructed through the following four steps: (1) define fuzzy sets in the filter input space Rn whose membership functions cover U; (2) construct a set of fuzzy IF-THEN rules which either come from human experts or are determined during the adaptation procedure by matching input-output data pairs; (3) construct a filter based on the set of rules; and (4) update the free parameters of the filter using the RLS algorithm. The design procedure for the LMS fuzzy adaptive filter is similar. The most important advantage of the fuzzy adaptive filters is that linguistic information (in the form of fuzzy IF-THEN rules) and numerical information (in the form of input-output pairs) can be combined in the filters in a uniform fashion. The filters are applied to nonlinear communication channel equalization problems     

Output error convergence of adaptive filters with compensation foroutput nonlinearities

Output error convergence of a Wiener model-based nonlinear stochastic gradient algorithm is analyzed. The normalized scheme estimates the parameters of a linear finite impulse response model in cascade with a known output nonlinearity. The algorithm can be interpreted as a normalized least mean square algorithm with compensation for an output nonlinearity. Linearizing inversion of the nonlinearity is not utilized. Global output error convergence is then proved, provided that the nonlinearity is monotone (not strictly monotone), and provided that a previously observed mechanism resulting in deadlock does not occur. The algorithm and the analysis include important practical cases like sensor saturation and dead zones that must be excluded when global parametric convergence is studied     

基于滤波器的局部自适应全变分图像去噪模型

综合利用冲击滤波器和非线性各向异性扩散滤波器对含噪图像做预处理,然后基于边缘检测函数建立反映图像局部特征的自适应权函数,构建能同时兼顾图像平滑去噪与边缘保留的局部自适应性的全变分模型,并建议用本原对偶算法快速求解。实验结果表明,同传统的全变分图像去噪模型相比,该局部自适应全变分模型在消除噪声的同时能很好地保持图像的边缘轮廓和纹理等细节特征,得到的复原图像在客观评价标准和主观视觉效果方面均有所提高。