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Kernel adaptive filter

About: Kernel adaptive filter is a research topic. Over the lifetime, 8771 publications have been published within this topic receiving 142711 citations.


Papers
More filters
Journal ArticleDOI
TL;DR: This paper is concerned with the realization of a given arbitrary filter transfer function as a network of resistively interconnected integrators using a new technique called intermediate function (IF) synthesis, based on the selection of a set of functions to serve as either the transfer functions from the filter input to the integrator outputs or the transfer function from the Integrator inputs to the filter output.
Abstract: This paper is concerned with the realization of a given arbitrary filter transfer function as a network of resistively interconnected integrators. These state-space realizations are synthesized using a new technique called intermediate function (IF) synthesis. The technique is based on the selection of a set of functions to serve as either the transfer functions from the filter input to the integrator outputs or the transfer functions from the integrator inputs to the filter output. Relationships between the filter sensitivity and dynamic range and the intermediate functions are derived. A number of results are also given to aid in the selection of a set of IF's that yields structures with optimum performance.

89 citations

Journal ArticleDOI
TL;DR: This work derives a different form of the Kalman filter by considering, at each iteration, a block of time samples instead of one time sample as it is the case in the conventional approach.
Abstract: The Kalman filter is a very interesting signal processing tool, which is widely used in many practical applications. In this paper, we study the Kalman filter in the context of echo cancellation. The contribution of this work is threefold. First, we derive a different form of the Kalman filter by considering, at each iteration, a block of time samples instead of one time sample as it is the case in the conventional approach. Second, we show how this general Kalman filter (GKF) is connected with some of the most popular adaptive filters for echo cancellation, i.e., the normalized least-mean-square (NLMS) algorithm, the affine projection algorithm (APA) and its proportionate version (PAPA). Third, a simplified Kalman filter is developed in order to reduce the computational load of the GKF; this algorithm behaves like a variable step-size adaptive filter. Simulation results indicate the good performance of the proposed algorithms, which can be attractive choices for echo cancellation.

88 citations

Journal ArticleDOI
01 Feb 2011
TL;DR: A low-complexity Kalman filter implementation in which the filter gain is approximated by its steady-state form, computed offline before real-time decoding commences, which reduces the computational load for decoding the firing rates of 25±3 single units by a factor of 7.9.
Abstract: The Kalman filter is commonly used in neural interface systems to decode neural activity and estimate the desired movement kinematics. We analyze a low-complexity Kalman filter implementation in which the filter gain is approximated by its steady-state form, computed offline before real-time decoding commences. We evaluate its performance using human motor cortical spike train data obtained from an intracortical recording array as part of an ongoing pilot clinical trial. We demonstrate that the standard Kalman filter gain converges to within 95% of the steady-state filter gain in 1.5 ± 0.5 s (mean ±s.d.). The difference in the intended movement velocity decoded by the two filters vanishes within 5 s, with a correlation coefficient of 0.99 between the two decoded velocities over the session length. We also find that the steady-state Kalman filter reduces the computational load (algorithm execution time) for decoding the firing rates of 25±3 single units by a factor of 7.0±0.9. We expect that the gain in computational efficiency will be much higher in systems with larger neural ensembles. The steady-state filter can thus provide substantial runtime efficiency at little cost in terms of estimation accuracy. This far more efficient neural decoding approach will facilitate the practical implementation of future large-dimensional, multisignal neural interface systems.

88 citations

Journal ArticleDOI
TL;DR: This paper presents a sensorless technique for permanent-magnet synchronous motors (PMSMs) based on high-frequency pulsating voltage injection, using a neural network based adaptive variable-band filter instead of a fixed-bandwidth filter, needed for catching the speed information from the sidebands of the stator current.
Abstract: This paper presents a sensorless technique for permanent-magnet synchronous motors (PMSMs) based on high-frequency pulsating voltage injection. Starting from a speed estimation scheme well known in the literature, this paper proposes the adoption of a neural network (NN) based adaptive variable-band filter instead of a fixed-bandwidth filter, needed for catching the speed information from the sidebands of the stator current. The proposed NN filter is based on a linear NN adaptive linear neuron (ADALINE), trained with a classic least mean squares (LMS) algorithm, and is twice adaptive. From one side, it is adaptive in the sense that its weights are adapted online recursively. From another side, its bandwidth is made adaptive during the running of the drive, acting directly on the learning rate of the NN filter itself. The immediate consequence of adopting a variable-band structure is the possibility to enlarge significantly the working speed range of the sensorless drive, which can be increased by a factor of five. The proposed observer has been tested experimentally on a fractional horsepower PMSM drive and has been compared also with a fixed-bandwidth structure.

88 citations

Patent
13 Mar 1997
TL;DR: In this paper, the phase and magnitude response of the adaptive equalizer filter are constrained at a predetermined frequency using an optimal orthogonal projection operation as a modification to a least mean square (LMS) adaptation algorithm.
Abstract: A sampled amplitude read channel for magnetic disk recording which asynchronously samples the analog read signal, adaptively equalizes the resulting discrete time sample values according to a target partial response, extracts synchronous sample values through interpolated timing recovery, and detects digital data from the synchronous sample values using a Viterbi sequence detector is disclosed. To minimize interference from the timing and gain control loops, the phase and magnitude response of the adaptive equalizer filter are constrained at a predetermined frequency using an optimal orthogonal projection operation as a modification to a least mean square (LMS) adaptation algorithm. Further, with interpolated timing recovery, the equalizer filter and its associated latency are removed from the timing recovery loop, thereby allowing a higher order discrete time filter and a lower order analog filter.

87 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
202322
202251
202113
202020
201931
201844