The performances of two low-complexity IIR adaptive filtering algorithms applied to the equalization of a typical transmission line channel are compared. The algorithms considered are the Simple Hyperstable Adaptive R...
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The performances of two low-complexity IIR adaptive filtering algorithms applied to the equalization of a typical transmission line channel are compared. The algorithms considered are the Simple Hyperstable Adaptive Recursive Filter (SHARF) and the stochastic gradient equation error algorithm. Gradient descent techniques are shown to be ill suited for transmission line channels.
Some properties of a recently proposed unbiased criteria for adaptive IIR filtering, namely the Master-slave Expanded Numerator Algorithm (XNA), is investigated. The XNA algorithm combines the minimization of two crit...
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Some properties of a recently proposed unbiased criteria for adaptive IIR filtering, namely the Master-slave Expanded Numerator Algorithm (XNA), is investigated. The XNA algorithm combines the minimization of two criteria that are quadratic in the parameters. Although the XNA method achieves an unbiased behavior in sufficient-order system identification applications, it generates undesired stationary points. Through the study presented in this paper, explicit expressions for the unwanted stationary points can be obtained. By closely examining the obtained results, a new unbiased adaptive IIR filtering algorithm is presented. Using similar ideas to the Steiglitz-McBride method, the new algorithm is based on the minimization of an off-line error function. After the proposal of the new unbiased (off-line) algorithm, which is quadratic in the parameters, we present the analysis of the possible stationary points. Some examples are shown illustrating the performance of the new algorithm in a system identification application.
A fundamental open problem in linear filtering and estimation is addressed, i.e. what is the steady-state or asymptotic behavior of the Kalman filter, or the Kalman gain, when the observed stationary stochastic proces...
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A fundamental open problem in linear filtering and estimation is addressed, i.e. what is the steady-state or asymptotic behavior of the Kalman filter, or the Kalman gain, when the observed stationary stochastic process is not generated by a finite-dimensional stochastic system, or when it is generated by a stochastic system having higher dimensional unmodeled dynamics? For a scalar observation process, necessary and sufficient conditions are derived for the Kalman filter to converge, using methods from stochastic systems and from nonlinear dynamics, especially the use of stable, unstable and center manifolds. It is shown that, in nonconvergent cases, there exist periodic points of every period p, p>or=3 which are arbitrarily close to initial conditions having unbounded orbits. This rigorously demonstrates that the Kalman filter can also be sensitive to initial conditions.< >
A unified approach is presented for deriving a large class of new and previously known time and order recursive least-squares algorithms with systolic array architectures, suitable for high throughput rate and VLSI im...
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A unified approach is presented for deriving a large class of new and previously known time and order recursive least-squares algorithms with systolic array architectures, suitable for high throughput rate and VLSI implementations of space-time filtering and system identification problems. The geometrical derivation given is unique in that no assumption is made concerning the rank of the sample data correlation matrix. This method utilizes and extends the concept of oblique projections, as used previously in the derivations of the least-squares lattice algorithms. Both the growing and sliding memory, exponentially weighted least-squares criteria are considered.< >
Traditional acoustic source localization techniques attempt to determine the current location of an acoustic source from data obtained at an array of sensors during the current time only. Recently, state-space methods...
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Traditional acoustic source localization techniques attempt to determine the current location of an acoustic source from data obtained at an array of sensors during the current time only. Recently, state-space methods have been proposed that use particle filters to perform recursive estimation of the current source location using all previous data. In this paper we present an overview of these particle filter algorithms, and formulate performance measures for determining their ability to track a moving source. We present results of experiments using reverberant data recorded in a real room, and show that steered beamforming methods have improved performance over GCC-based approaches.
We consider the problem of optimal FIR linear and regression filters for jointly stationary symmetric alpha stable processes with index alpha greater than one. We then consider the signed-error and sign-sign adaptive ...
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We consider the problem of optimal FIR linear and regression filters for jointly stationary symmetric alpha stable processes with index alpha greater than one. We then consider the signed-error and sign-sign adaptive filtering algorithms and provide various convergence results for the signal and tap weights estimation errors.
Three systolic arrays suitable for implementation of order-recursive least-squares (ORLS) adaptive algorithms are considered. It is shown that they can be constructed using two types of elementary cells. A classificat...
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Three systolic arrays suitable for implementation of order-recursive least-squares (ORLS) adaptive algorithms are considered. It is shown that they can be constructed using two types of elementary cells. A classification of systolic implementations of the ORLS adaptive algorithms is given by exploiting the possible variations of the elementary cells. The investigation of the array structures and variations of the elementary cells leads to a systematic approach to designing reconfigurable systolic arrays for implementation of ORLS algorithms. As an application of this approach, a novel least-squares (LS) lattice algorithm based on Givens rotations is derived.< >
A new adaptive algorithm for system identification and adaptive signal processing has been proposed in this paper. The algorithm can be viewed as a combination of the output error type and the equation error type of i...
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A new adaptive algorithm for system identification and adaptive signal processing has been proposed in this paper. The algorithm can be viewed as a combination of the output error type and the equation error type of identifier (i.e. a combination of IIR adaptive filter and FIR adaptive filter in signal processing application). It has been shown both mathematically and by computer simulation that the proposed algorithm can guarantee the globally asymptotical stability under certain regularity conditions on the distribution of signal modes. The transient behaviour of the algorithm is better than conventional output error type of identifier, especially in the case where the system model (signal model) has its eigenvalues on the unit circle. Under certain conditions on the system model (signal model), no moving average (MA) smoothing of the generalized error is needed for the strictly positive realness (SPR) conditions. It gives a reduced bias on the parameter estimate (compare to conventional adaptive FIR filter) when corresponding equation error is a bandlimited process.
This paper presents a novel approach to the analysis of detection signals from optical sensors. The proposed technique is based on the pattern recognition algorithms that belong to the family of composite linear filte...
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This paper presents a novel approach to the analysis of detection signals from optical sensors. The proposed technique is based on the pattern recognition algorithms that belong to the family of composite linear filters and is compared against a MAP-estimate approach. Our study is focused on the information recovery from the reflectivity spectrum of a Bragg sensor measured with an optical spectrum analyzer. However we note that the same scheme would be equally useful for the extraction of information obtained with a polarimetric fibre sensor or an optical chemical sensor, where in general the detection signal is multidimensional and includes data related to several measured parameters.
The paper presents a beat tracking algorithm for musical audio signals. The method firstly extracts musical change points from the help signal and then uses a particle filtering algorithm to associate these to a tempo...
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The paper presents a beat tracking algorithm for musical audio signals. The method firstly extracts musical change points from the help signal and then uses a particle filtering algorithm to associate these to a tempo process. Results are comparable with the current state of the art.
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