Multichannel adaptive filtering with a feedback convergence function

The fundamental structure of linear‐combined multichannel adaptive digital filters which can process a multiple‐input single‐output time series, and their tapcoefficient correction algorithms with fast convergence are described. These are fundamental circuits required for multichannel echo‐cancellers, and echo‐cancellers with noise‐cancelling facilities. The structure of linear‐combined multichannel adaptive digital filters using the learning identification (LI) was introduced, and their characteristics were analyzed. Adaptive algorithms using the per‐channel normalization of power were proposed to improve the convergence speed. With a minor increase in the amount of computations, this method is a significant improvement of the convergence speed which, inherently, is degraded by the differences of average powers among channels in a linear‐combined multichannel adaptive filter. For further improvement in the convergence speed, another tap‐coefficient correction equation (the projection method) also was introduced. In the tap‐coefficient equation using the projection algorithms, the convergence speed can also be improved significantly by applying the per‐channel power normalization.

Adaptive filters are prevalent in many real‐time signal processing applications. Many adaptive algorithms already exist, but most of them assume white Gaussian noise as disturbance. However, for many applications such as electrocardiogram, foetus heart rate measurement, low frequency atmospheric noise, underwater acoustic noise and signal measurement in instrumentation, the impulsive noise is more common. This study presents a modified robust mixed norm (MRMN) adaptive filter algorithm robust to impulsive noise with higher convergence rate and lower steady state error (SSE). MRMN adaptive filter algorithm has been simulated using Matlab and Xilinx system generator high level synthesis tool and a significant improvement in SSE and convergence speed is obtained compared with the existing adaptive filter algorithms for similar specifications. The proposed algorithm is also described using VHDL and synthesised using Xilinx synthesiser tool in order to implement on field‐programmable gate array (FPGA). The FPGA post route and place implementation results show nearly 90% reduction in resource utilisation and nearly 2.6 times improvement in clock frequency as compared with the existing FPGA based implementation for similar specifications.

This paper aims to apply adaptive filtering algorithms in sensor array processing for underwater surveillance and detection systems with linear arrays. Adaptive filters are iterative in nature and continuously update the filter taps to minimize the error signal based upon certain criterion. Adaptive filters are used in beamforming to place the main beam in the direction of desired signal and place nulls in the direction of unwanted signals. Adaptive filters are also used in spectral analysis of acoustic signals for background noise cancellation. This paper evaluates the performance of three adaptive filtering algorithms with respect to sensor array design parameters i.e. number of sensors, sensor spacing, number of incident signals and their angular separation. The three adaptive filtering algorithms namely least mean square (LMS), normalized LMS (NLMS), and sample matrix inversion (SMI) have been compared in the context of linear array beamforming. The paper provides detailed discussions on the simulation results of these adaptive algorithms in terms of convergence speed, beamwidth, null depths and maximum side lobe levels. In addition the paper illustrates a practical example of applying adaptive filters in underwater sensor array processing systems for detection of target’s acoustic signatures.

Given the complex traffic factors and strong mobility of a transit vehicles, it is difficult to predict its travel time with the conventional prediction models. The emphasis of this research effort is to improve the adaptability of the Kalman filter model with regard to transit bus travel time prediction. In order to resolve the filter divergence and calculated divergence in the prediction process, an adaptive fading Kalman filter algorithm is put forward in this paper. Compared with the conventional Kalman filter prediction model, a "forgotten factor" is added to the adaptive fading Kalman filter algorithm to restrain the influence of the old data on the filter. Furthermore, it can correct the predictive value and the model parameters with the latest measured value, hence higher adaptability and predictive accuracy. Finally, the GPS data of transit vehicles in the city of Yichun is used in the paper to verify the effectiveness of the new model. The results reveal that the adaptive fading Kalman filter algorithm is more effective than the conventional one. It performs better in terms of convergence and dynamic adaptability, implying a brighter application prospect.

The benefits of optical current transducers (OCT) have been intensively investigated in recent years. However, temperature and stress-induced linear birefringence in sensing materials cause error and instability, which limits severely the practical application of OCTs. In order to take advantages of both the conventional CT and OCT, the paper presents a new kind of hybrid optical CT formed by the conventional CT with high accuracy in the steady state and the optical CT with no saturation under fault current conditions. By means of the adaptive Kalman filter algorithm derived in this paper, the hybrid optical CT is adjusted adaptively to improve the accuracy to satisfy the need of measurement and protection in the power systems. In addition, applying the noise estimation with unknown prior statistics to the basic Kalman adaptive filter equations, the adaptive Kalman filter algorithm with noise estimation is derived in the paper. The test results show the validity of the hybrid optical CT and the adaptive Kalman filter algorithm.

The LMS adaptive filter algorithm can be viewed as the application of a simple iterative linear equation solver to an estimated time variantWiener-Hopf equation.With this interpretation we can utilize the “bag-of-tricks” available in numerical linear algebra in devising new adaptive filter algorithms. Making use of the preconditioning paradigm, we present in this paper a new family of NLMS-type adaptive filter whose performance can be tailored to possible a priori knowledge of the autocorrelation properties of the input signal. The design of circulant preconditioning matrices for such adaptive filter algorithms is the main contribution of the paper.

Noise and interference are the two most common and basic problems in wireless communication systems. The noise in wireless communication channels has the characteristics of randomness and impulsivity, so the performance of adaptive filtering algorithms based on geometric algebra (GA) and second-order statistics is greatly reduced in the wireless communication systems. In order to improve the performance of adaptive filtering algorithms in wireless communication systems, this paper proposes two novel GA-based adaptive filtering algorithms, which are deduced from the robust algorithms based on the minimum error entropy (MEE) criterion and the joint criterion (MSEMEE) of the MEE and the mean square error (MSE) with the help of GA theory. The noise interference in wireless communication is modeled by alpha-stable distribution which is in good agreement with the actual data in this paper. Simulation results show that for the mean square deviation (MSD) learning curve, the GA-based MEE (GA-MEE) algorithm has faster convergence rate and better steady-state accuracy compared to the GA-based maximum correntropy criterion algorithm (GA-MCC) under the same generalized signal-to-noise ratio (GSNR). The GA-MEE algorithm reduces the convergence rate, but improves the steady-state accuracy by 10–15 dB compared to the adaptive filtering algorithms based on GA and second-order statistics. For GA-based MSEMEE (GA-MSEMEE) algorithm, when GA-MSEMEE and the adaptive filtering algorithms based on GA and second-order statistics keep the same convergence rate, its steady-state accuracy is improved by 10–15 dB, and when GA-MSEMEE and GA-MEE maintain approximately steady-state accuracy, its convergence rate is improved by nearly 100 iterations. In addition, when the algorithms are applied to noise cancellation, the average recovery error of the two proposed algorithms is 7 points lower than that of other GA-based adaptive filtering algorithms. The results validate the effectiveness and superiority of the GA-MEE and GA-MSEMEE algorithms in the alpha-stable noise environment, providing new methods to deal with multi-channel interference in wireless networks.

This paper describes a method for noise removal from ultrasonic signals used in NDE (Non Destructive Evaluation) applications. The method used is FPGA based adaptive noise cancellation. Indeed adaptive filtering is one of the core technologies in digital signal processing. We can find numerous application areas in science as well as in industry. Adaptive filtering techniques are used in a wide range of applications, including system identification, adaptive equalization, adaptive noise cancellation and echo cancellation. Characteristics of signals that are generated by systems of the above application are not known a priori. Under this condition, a significant improvement in performance can be achieved by using adaptive rather than fixed filters. An adaptive filter is a self-designing filter that uses a recursive algorithm (known as adaptation algorithm or adaptive filtering algorithm) to design itself. Here the algorithm used is normalized least mean square algorithm, which has good convergence and stability. Keywords; Adaptive noise cancellation, NLMS algorithm, NDE applications, ultrasonic grain noise.

The adaptive kernel algorithms usually achieve a good convergence performance and a tracking performance due to the universal approximator, offering an excellent solution to many problems with nonlinearities. However, as is well known, the convergence rate and steady-state error of adaptive filtering algorithm are a pair of inherent contradictions, and the kernel method is not exceptional. For this problem, a robust kernel adaptive filtering algorithm, called the variable-scaling factor kernel fractional lower power adaptive filtering algorithm based on the Sigmoid function, is developed by creating a new framework of cost function which combines the kernel fractional low power error criterion with the Sigmoid function for system identification of different noise environments. This new cost framework incorporates a scaling factor into the cost function of the Sigmoid kernel fractional lower power adaptive filtering algorithm (VS-SKFLP) in this paper. One of the main features in the new framework is its scaling factor. This scaling factor is used to control the steepness of the Sigmoid function, and the steepness can affect the convergence speed of filtering algorithm. The scaling factor provides a tradeoff between the convergence rate and the steady-state mean square error (MSE), which improves the convergence rate under the same steady-state mean square error. However, it is also an important problem to choose an appropriate scale factor. Therefore, a variable-scale factor SKFLP algorithm is also proposed to improve the convergence rate and steady-state MSE, simultaneously. The proposed variable-scale factor structure consists of a function of error, featuring the adaptive updates of their parameter estimated by making discerning use of the error. In this paper, the nonlinear saturation characteristic of the Sigmoid function and low order norm criterion are used to overcome the performance degradation of training data destroyed by non-Gaussian impulse noise and colored noise. Through the convergence analysis, the parameter estimation sequence of our proposed algorithm proves convergent. Simulation results show that the proposed algorithm (VS-SKFLP) outperforms other kernel adaptive filtering algorithms in system recognition with different noise environments.

Estimation of battery state of charge based on changing window adaptive extended Kalman filtering

Publication in the conference proceedings of EUSIPCO, Antalya, Turkey, 2005

In the extended Kalman filter (EKF) and the unscented Kalman filter (UKF) of non-linear system, the mathematical model and the noise statistics are essential; this limitation was settled by adaptive algorithm. The adaptive extended Kalman filter (AEKF) and adaptive unscented filter (AUKF) were proposed to solve the filtering problem of the system with unknown mathematical model or noise statistics in information fusion. The simulation result shows that the adaptive algorithm in EKF and UKF can improve the performance of filter and decrease the computing time.

A generalized least squares fast transversal filter algorithm for the decision feedback equalization of dispersive channels

A novel method of indoor mobile robot navigation is presented. The proposed approach fuses the data of odometry and electronic compass for navigation. It includes two calibration methods and a fusion algorithm. First of all, calibration method of systematic odometry error is used to reduce the error of navigation and provide the reliable estimate of pose of mobile robot for adaptive extended Kalman filter fusion algorithm later. Secondly, calibration method of electronic compass using an adaptive neural fuzzy inference system provides direction angle of mobile robot as observation for adaptive extended Kalman filter algorithm later. Finally, a fusion algorithm using adaptive extended Kalman filter algorithm fuses data of odometry and electronic compass that provides the position and orientation of mobile robot. In addition, the value of the parameter k of adaptive extended Kalman filter algorithm which is related to the process noise covariance is decided by fuzzy algorithm. In order to illustrate the feasibility of the proposed approach, two types of experiments are done: the first-type experiment is that the mobile robot runs along default path only with odometry, and the mobile robot with odometry and electronic compass in the second-type experiment utilizes the proposed approach for navigation. The results of experiments show that the error of localization and navigation in the first-type experiment is larger than one in the second-type experiment. They prove the good performance of the proposed approach.

A family of adaptive filtering algorithms for processing signals which have energy concentrated in a relatively small number of component subspaces in the spectral domain is introduced. The approach is based on transform domain signal decomposition and linear least squares filtering of the selected subset of transform domain signal components. The derivation is based on the linear least squares adaptive filtering framework introduced in previous work (1997). Fast convergence and computational efficiency are the main characteristics of the resulting algorithms. The method is applied to the problem of adaptive line enhancement comb filtering and DFT is used as a transform method. It is also shown that the resulting adaptive structure is capable of handling the case of non-coinciding frequencies. The performance of the algorithm is evaluated through a series of simulation experiments.