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Showing papers in "Circuits Systems and Signal Processing in 2006"


Journal ArticleDOI
TL;DR: The differential evolution algorithm is a new heuristic approach with three main advantages: it finds the true global minimum of a multimodal search space regardless of the initial parameter values, it has fast convergence, and it uses only a few control parameters.
Abstract: The differential evolution (DE) algorithm is a new heuristic approach with three main advantages: it finds the true global minimum of a multimodal search space regardless of the initial parameter values, it has fast convergence, and it uses only a few control parameters. The DE algorithm, which has been proposed particularly for numeric optimization problems, is a population-based algorithm like the genetic algorithms and uses similar operators: crossover, mutation, and selection. In this work, the DE algorithm has been applied to the design of digital finite impulse response filters, and its performance has been compared to that of the genetic algorithm and least squares method.

153 citations


Journal ArticleDOI
TL;DR: The results show that the number of adders and subtracters decreases on average 25% for 19-bit coefficients compared with the canonic signed-digit representation.
Abstract: In many digital signal processing algorithms, e.g., linear transforms and digital filters, the multiplier coefficients are constant. Hence, it is possible to implement the multiplier using shifts, adders, and subtracters. In this work two approaches to realize constant coefficient multiplication with few adders and subtracters are presented. The first yields optimal results, i.e., a minimum number of adders and subtracters, but requires an exhaustive search. Compared with previous optimal approaches, redundancies in the exhaustive search cause the search time to be drastically decreased. The second is a heuristic approach based on signed-digit representation and subexpression sharing. The results for the heuristic are worse in only approximately 1% of all coefficients up to 19 bits. However, the optimal approach results in several different optimal realizations, from which it is possible to pick the best one based on other criteria. Relations between the number of adders, possible coefficients, and number of cascaded adders are presented, as well as exact equations for the number of required full and half adder cells. The results show that the number of adders and subtracters decreases on average 25% for 19-bit coefficients compared with the canonic signed-digit representation.

83 citations


Journal ArticleDOI
TL;DR: In this paper, the authors proposed a transmultiplexer design method that can not only achieve close-to-perfect reconstruction but also obtain designable filter length based on a composite distortion measure: the 2-norm of the error transfer matrix of the trans multiplexer.
Abstract: In this paper, motivated by the facts that shorter length filters offer considerable improvements in computation and hardware implementation and that designable filter length brings more design flexibility, we propose a novel transmultiplexer design method that can not only achieve close-to-perfect reconstruction but also obtain designable filter length. The proposed method is based on a composite distortion measure: the 2-norm of the error transfer matrix of the transmultiplexer. Central to the development is providing an efficient means to recursively design the filters in the transmultiplexer and evaluating the composite distortion measure, which greatly improves the computation efficiency. The simulation results show the effectiveness of the proposed algorithms.

57 citations


Journal ArticleDOI
TL;DR: In this article, a general discussion of the inherent difficulties of the problem is given together with a comprehensive study on how the choice of the sampling interval influences the estimation result, and a special focus is given to how the Cramer-Rao lower bound depends on the sample interval.
Abstract: The problem of estimating the parameters in continuous-time autoregressive moving average (ARMA) processes from discrete-time data is considered. Both direct and indirect methods are studied, and similarities and differences are discussed. A general discussion of the inherent difficulties of the problem is given together with a comprehensive study on how the choice of the sampling interval influences the estimation result. A special focus is given to how the Cramer-Rao lower bound depends on the sampling interval.

51 citations


Journal ArticleDOI
TL;DR: In this paper, a high input impedance voltage-mode universal biquadratic filter with one input terminal and five output terminals is presented, which uses three differential voltage current conveyors (DVCCs), four resistors, and two grounded capacitors.
Abstract: A new high input impedance voltage-mode universal biquadratic filter with one input terminal and five output terminals is presented. The proposed circuit uses three differential voltage current conveyors (DVCCs), four resistors, and two grounded capacitors. The proposed circuit can realize all the standard filter functions—lowpass, bandpass, highpass, notch, and allpass—simultaneously, without changing the passive elements. The proposed circuit enjoys the features of high input impedance, orthogonal control of resonance angular frequency and quality factor, use of only grounded capacitors, and low active and passive sensitivities.

49 citations


Journal ArticleDOI
TL;DR: In this article, a grounded negative inductance emulator is proposed with full independent control on both the inductance value and the condition, which uses a single operational transresistance amplifier (OTRA), a capacitor, and five resistors, two of which are for independent control.
Abstract: A grounded negative inductance emulator is proposed with full independent control on both the inductance value and the condition. It uses a single operational transresistance amplifier (OTRA), a capacitor, and five resistors, two of which are for independent control. Experimental results, which confirm the theoretical analysis, are presented.

44 citations


Journal ArticleDOI
TL;DR: This work provides a new mixed computational/analytical approach for adaptive compensation of this nonlinear distortion of the high peak-to-average power ratio (PAPR) of OFDM for cases in which the HPA is a traveling wave tube amplifier (TWTA) and solid state power amplifier (SSPA).
Abstract: Orthogonal frequency division multiplexing (OFDM) has several desirable attributes which make it a prime candidate for a number of emerging wireless communication standards. However, one of the major problems posed by OFDM is its high peak-to-average power ratio (PAPR), which seriously limits the power efficiency of the high-power amplifier (HPA) because of the nonlinear distortion resulting from the high PAPR. We provide a new mixed computational/analytical approach for adaptive compensation of this nonlinear distortion for cases in which the HPA is a traveling wave tube amplifier (TWTA) and solid state power amplifier (SSPA). TWTAs are used in wireless communication systems when high transmission power is required as in the case of the digital satellite channel, and SSPAs are generally used in mobile communication systems. Compared to previous predistorter techniques based on LUT (look-up table) or adaptive schemes, our approach relies on the analytical inversion of Saleh's TWTA model and Rapp's SSPA model in combination with a nonlinear parameter estimation algorithm. This leads to a sparse and yet accurate representation of the predistorter, with the capability of tracking efficiently any rapidly time-varying behavior of the HPA. Computer simulations results illustrate and validate the approach presented.

37 citations


Journal ArticleDOI
TL;DR: In this paper, a new current-mode current-controlled three-input single-output universal filter, which employs only five plus-type second-generation current controlled current conveyors (CCCIIs) and two grounded capacitors, is presented.
Abstract: A new current-mode current-controlled three-input single-output universal filter, which employs only five plus-type second-generation current-controlled current conveyors (CCCIIs) and two grounded capacitors, is presented in this paper. The proposed configuration provides lowpass, bandpass, highpass, bandstop, and allpass current responses at a high impedance terminal, which enables easy cascadability of the circuit. The filter also offers independent electronic control of the natural frequency ωo and the quality factor Q through adjusting of the bias current of the CCCII. Derived analytical expressions for the filter parameter deviations due to CCCII nonidealities are also included. PSPICE simulation results are performed to confirm the theoretical analysis.

35 citations


Journal ArticleDOI
TL;DR: In this paper, an efficient coefficient quantization scheme is described for minimizing the cost of implementing fixed parallel linear phase finite impulse response (FIR) filters in the modified Farrow structure introduced by Vesma and Saramaki for generating FIR filters with an adjustable fractional delay.
Abstract: An efficient coefficient quantization scheme is described for minimizing the cost of implementing fixed parallel linear-phase finite impulse response (FIR) filters in the modified Farrow structure introduced by Vesma and Saramaki for generating FIR filters with an adjustable fractional delay. The implementation costs under consideration are the minimum number of adders and subtracters when implementing these parallel subfilters as a very large-scale integration (VLSI) circuit. Two implementation costs are under consideration to meet the given criteria. In the first case, all the coefficient values are implemented independently of each other as a few signed-powers-of-two terms, whereas in the second case, the common subexpressions within all the coefficient values included in the overall implementation are properly shared in order to reduce the overall implementation cost even further. The optimum finite-precision solution is found in four steps. First, the number of filters and their (common odd) order are determined such that the given criteria are sufficiently exceeded in order to allow some coefficient quantization errors. Second, those coefficient values of the subfilters having a negligible effect on the overall system performance are fixed to be zero valued. In addition, the experimentally observed attractive connections between the coefficient values of the subfilters, after setting some coefficient values equal to zero, are utilized to reduce both the implementation cost and the parameters to be optimized even more. Third, constrained nonlinear optimization is applied to determine for the remaining infinite-precision coefficients a parameter space that includes the feasible space where the given criteria are met. The fourth step involves finding in this space the desired finite-precision coefficient values for minimizing the given implementation costs to meet the stated overall criteria. Several examples are included illustrating the efficiency of the proposed synthesis scheme.

33 citations


Journal ArticleDOI
TL;DR: In this paper, a second-order voltage-mode filter with single input and six outputs employing inverting second-generation current conveyors (ICCIIs) is presented, which simultaneously realizes negative and positive gain lowpass, bandpass, and highpass filter responses, and does not require active and passive element matching.
Abstract: In this paper, a novel second-order voltage-mode filter with single input and six outputs employing inverting second-generation current conveyors (ICCIIs) is presented. This filter simultaneously realizes negative and positive gain lowpass, bandpass, and highpass filter responses, and does not require active and passive element matching. The new filter has low active and passive element sensitivities, and offers orthogonal control of the angular resonance frequency (ωo) and quality factor (Q).

26 citations


Journal ArticleDOI
TL;DR: In this paper, the feasibility of the frequency response masking (FRM) technique in array beamforming is investigated in detail, and a novel combination of the concept of effective aperture and the FRM technique does lead to the synthesis of desirable beamformers.
Abstract: The frequency-response masking (FRM) technique is well known to be very efficient in the implementation of finite impulse response (FIR) filters with sharp transition bands. As sensor array beamforming is closely related to FIR filtering, the feasibility of the applications of the FRM technique in array beamforming is investigated in detail in this paper. On one hand, it is shown that there is a limitation in applying the FRM technique in passive array beamforming. On the other hand, for active array beamforming, a novel combination of the concept of effective aperture and the FRM technique does lead to the synthesis of desirable beamformers. These beamformers have effective beampatterns with sharp transition bands and low sidelobes, and can be implemented with fewer sensors than other design techniques.

Journal ArticleDOI
TL;DR: In this article, two classes of frequency-response masking (FRM) linear-phase finite (length) impulse response (FIR) filters for interpolation and decimation by arbitrary integer factors M were proposed.
Abstract: This paper introduces two classes of frequency-response masking (FRM) linear-phase finite (length) impulse response (FIR) filters for interpolation and decimation by arbitrary integer factors M. As they are based on the FRM approach, the proposed filters are low-complexity (efficient) sharp-transition linear-phase FIR interpolation and decimation filters. Compared to previously existing FRM linear-phase FIR filter classes for interpolation and decimation, the new ones offer lower complexity and more freedom in selecting the locations of the passband and stopband edges. Furthermore, the proposed classes of FRM filters can, as special cases, realize efficient Mth-band FRM linear-phase FIR interpolation and decimation filters for all values of M. Previously, only half-band (M = 2) FRM linear-phase FIR filters have appeared in the literature. The paper includes design techniques suitable for the new filters and design examples illustrating their efficiency.

Journal ArticleDOI
TL;DR: In this paper, the robust H∞ filtering problem for a class of uncertain singular systems with time delays is considered, and a sufficient condition for the existence of a full-order H ∞ filter is given in terms of matrix inequalities.
Abstract: This paper deals with the robust H∞ filtering problem for a class of uncertain singular systems with time delays. The uncertainty under consideration is of a linear fractional form. A sufficient condition for the existence of a full-order H∞ filter is given in terms of matrix inequalities. When a scalar variable in the matrix inequalities is fixed, the condition can be expressed in terms of linear matrix inequalities (LMIs). The desired H∞ filter can be obtained by solving these LMIs. Two numerical examples are given to demonstrate the application of the proposed method.

Journal ArticleDOI
TL;DR: In this paper, the robust H∞ filtering problem for discrete-time singular systems with norm-bounded uncertainties is studied, based on the admissibility assumption of singular systems, and a set of necessary and sufficient conditions for the existence of the desired filters is established.
Abstract: This paper concerns the robust H∞ filtering problem for discrete-time singular systems with norm-bounded uncertainties. Based on the admissibility assumption of singular systems, a set of necessary and sufficient conditions for the existence of the desired filters is established, and a normal filter design method under the linear matrix inequality framework is developed. A numerical example is given to illustrate the application of the proposed method.

Journal ArticleDOI
TL;DR: In this paper, a robust multiuser detector for combating multiple access interference and impulsive noise in code division multiple access (CDMA) communication systems is proposed, which is corroborated with simulation results.
Abstract: In many physical channels where multiuser detection techniques are to be applied, the ambient channel noise is known through experimental measurements to be decidedly non-Gaussian, due largely to impulsive phenomena This is due to the impulsive nature of man-made electromagnetic interference and a great deal of natural noise This paper presents a robust multiuser detector for combating multiple access interference and impulsive noise in code division multiple access (CDMA) communication systems A new M-estimator is proposed for "robustifying" the detector The approach is corroborated with simulation results to evaluate the performance of the proposed robust multiuser detector compared with that of the linear decorrelating detector, and the Huber and the Hampel M-estimator based detectors Simulation results show that the proposed detector with significant performance gain outperforms the linear decorrelating detector, and the Huber and the Hampel M-estimator based detectors This paper also presents an improved robust blind multiuser detection technique based on a subspace approach, which requires only the signature waveform and the timing of the desired user to demodulate that user's signal Finally, we show that the robust multiuser detection technique and its blind adaptive version can be applied to both synchronous and asynchronous CDMA channels

Journal ArticleDOI
TL;DR: In this paper, a robust l2 - l∞ control for uncertain discrete-time switched systems with time delay in the state is proposed. But the control design approach is facilitated by introducing some additional instrumental matrix variables.
Abstract: This paper is concerned with the problem of robust l2 - l∞ control for uncertain discrete-time switched systems with time delay in the state. The uncertainty is assumed to be of structured linear fractional form which includes the norm-bounded uncertainty as a special case. The control design approach is facilitated by introducing some additional instrumental matrix variables. The additional matrix variables decouple the Lyapunov and system matrices, which make the control design feasible. The proposed approach leads to less conservativeness of previous design methods, and the results also generalize earlier works for a more general parametric uncertainty structure. A numerical example is also given to demonstrate the effectiveness and the potential of the proposed techniques.

Journal ArticleDOI
TL;DR: In this article, a hybrid genetic algorithm (HGA) is proposed to jointly optimize all sub-filters in a discrete space, where the simulated annealing technique is introduced into the GA optimization process and effectively prevents the GA from prematurely converging.
Abstract: This paper presents the design of high-speed, arbitrary bandwidth sharp finite impulse response filters with signed powers-of-two coefficients based on a modified frequency-response masking (FRM) structure A novel hybrid genetic algorithm (HGA) is proposed to jointly optimize all subfilters in a discrete space The proposed HGA introduces the simulated annealing technique into the genetic algorithm (GA) optimization process and effectively prevents the GA from prematurely converging It is shown, by means of examples, that FRM filters designed by the HGA achieve a significant reduction in the number of bits

Journal ArticleDOI
TL;DR: A fifth-order modulator is designed to convert audio-band signals with an effective resolution of 20 bits to demonstrate the effectiveness of the proposed analysis and method of design.
Abstract: Single-bit sigma-delta modulators operated in the quasi-sliding mode are investigated. Sufficient conditions for the existence and stability of this mode of operation are derived. The derived stability conditions, along with an accurate prediction of its performance, enable a high-order modulator to be exactly designed. A fifth-order modulator is designed to convert audio-band signals with an effective resolution of 20 bits to demonstrate the effectiveness of the proposed analysis and method of design.

Journal ArticleDOI
TL;DR: In this article, a multiobjective filtering problem for a class of continuous-time time-delay systems with nonzero initial conditions is investigated, where the solvability conditions are provided via convex computation approaches.
Abstract: A multiobjective filtering problem is investigated for a class of continuous-time time-delay systems with nonzero initial conditions. The plant is supposed to have nonlinear dynamics and multiple time delays in both the state and the measured output. In order to address the compatible mixed H2 and H∞ performance measure for the filtering problem, the generalized H∞ performance index is introduced instead of the standard one. The novel mixed H2 and H∞ filtering problem is formulated as an optimization problem, where the solvability conditions are provided via convex computation approaches. After some algebraic transformations and decompositions, the involved nonlinear constraints can be eliminated and linear matrix inequality-based convex algorithms can be given for the filter design. A simple example is given to illustrate the optimization algorithms.

Journal ArticleDOI
TL;DR: The Aryabhatiya algorithm, which has much in common with the extended Euclidean algorithm (EEA), Chinese remainder theorem (CRT) and Garner's algorithm (GA), is shown to have a complexity comparable to or better than that of the CRT and GA.
Abstract: Public-key crypto-algorithms are widely employed for authentication, signatures, secret-key generation and access control. The new range of public-key sizes for RSA and DSA has gone up to 1024 bits and beyond. The elliptic-curve key range is from 162 bits to 256 bits. Many varied software and hardware algorithms are being developed for implementation for smart-card crypto-coprocessors and for public-key infrastructure. We begin with an algorithm from Aryabhatiya for solving the indeterminate equation a · x + c = b · y of degree one (also known as the Diophantine equation) and its extension to solve the system of two residues X mod mi = Xi (for i = 1,2). This contribution known as the Aryabhatiya algorithm (AA) is very profound in the sense that the problem of two congruences was solved with just one modular inverse operation and a modular reduction to a smaller modulus than the compound modulus. We extend AA to any set of t residues, and this is stated as the Aryabhata remainder theorem (ART). An iterative algorithm is also given to solve for t moduli mi (i = 1, 2,... , t). The ART, which has much in common with the extended Euclidean algorithm (EEA), Chinese remainder theorem (CRT) and Garner's algorithm (GA), is shown to have a complexity comparable to or better than that of the CRT and GA.

Journal ArticleDOI
TL;DR: A new technique is introduced that recasts pairs of the original polyphase components as sums or differences of auxiliary pairs of symmetric and anti-symmetric impulse response filters that results in a factor-of-two reduction in the number of multipliers required to implement the poly phase components.
Abstract: Polyphase implementation of FIR filters effectively reduces the multiplication rate and data storage in a multirate system. However, the coefficients of the polyphase components are no longer symmetric even though the overall filter has a symmetric (or anti-symmetric) impulse response. In this paper, we introduce a new technique that recasts pairs of the original polyphase components as sums or differences of auxiliary pairs of symmetric and anti-symmetric impulse response filters. The coefficient symmetry of these auxiliary polyphase components can be fully exploited to reduce arithmetic complexity without undue complications. Our new technique makes use of the fact that the impulse responses of the non-symmetric polyphase components exist in time-reversed pairs which can be synthesized from pairs of symmetric and anti-symmetric impulse response filters. This results in a factor-of-two reduction in the number of multipliers required to implement the polyphase components.

Journal ArticleDOI
TL;DR: In this paper, the authors proposed a graph-based approach for transient and distortion analysis of nonlinear analog circuits based on a frequency domain Volterra series representation of non-linear circuits.
Abstract: This paper presents a novel approach for transient and distortion analyses for time-invariant and periodically time-varying mildly nonlinear analog circuits. Our method is based on a frequency domain Volterra series representation of nonlinear circuits. It computes the nonlinear responses using a nonlinear current method that recursively solves a series of linear Volterra circuits to obtain linear and higher-order responses of a nonlinear circuit. Unlike existing approaches, where Volterra circuits are solved mainly in the time domain, the new method solves the linear Volterra circuits directly in the frequency domain via an efficient graph-based technique, which can derive transfer functions for any large linear network efficiently. As a result, both frequency domain characteristics, like harmonic and intermodulation distortion, and time domain waveforms can be computed efficiently. The new algorithm takes advantage of identical Volterra circuits for second- and higher-order responses, which results in significant savings in driving the transfer functions. Experimental results for two circuits—a low-noise amplifier and a switching mixer—are obtained and compared with SPICE3 to validate the effectiveness of this method.

Journal ArticleDOI
TL;DR: Simple nonlinearities, easily implementable as electronic circuits, are shown capable of producing an amplification of the signal-to-noise ratio (SNR) by the nonlinearity.
Abstract: A harmonic signal corrupted by an additive white noise is processed by an arbitrary memoryless nonlinear device. The transformation of the signal-to-noise ratio (SNR) by the nonlinearity is explicitly computed and analyzed for Gaussian and non-Gaussian noise. Simple nonlinearities, easily implementable as electronic circuits, are shown capable of producing an amplification of the SNR. Such an amplification is not obtainable with linear filters, whatever their complexity or high order, but becomes easily accessible with simple nonlinear devices.

Journal ArticleDOI
TL;DR: The discrete evolutionary transform provides a time-frequency procedure to obtain a complete characterization of the multipath, fading, and frequency selective channels in OFDM systems.
Abstract: Orthogonal frequency division multiplexing (OFDM) has become a very popular method for high data rate wireless communications because of its advantages over single carrier modulation schemes on multipath, frequency selective fading channels However, intercarrier interference, due to Doppler frequency shifts, and multipath fading severely degrade the performance of OFDM systems Estimation of channel parameters is required at the receiver In this paper, we present a channel modeling and estimation method based on the time-frequency representation of the received signal The discrete evolutionary transform provides a time-frequency procedure to obtain a complete characterization of the multipath, fading, and frequency selective channels Simulations are used to illustrate the performance of the proposed procedure and to compare it with other time-varying channel estimation techniques

Journal ArticleDOI
TL;DR: In this paper, a new transformation method is presented and used to transform voltage-mode op-amp-RC circuits to current-mode Gm-C ones, which enables the generation of high-performance GmC filters that benefit from the advantages of good and well-known Op-AMP-RC structures and the advantage of the currentmode circuits, and at the same time feature electronic tunability, high frequency capability and monolithic integration ability.
Abstract: A new transformation method is presented and used to transform voltage-mode op-amp-RC circuits to current-mode Gm-C ones. The proposed method enables the generation of high-performance Gm-C filters that benefit from the advantages of good and well-known op-amp-RC structures and the advantages of the current-mode circuits, and at the same time feature electronic tunability, high frequency capability, and monolithic integration ability. An attractive feature of the proposed method is that it results in Gm-C structures with only grounded capacitors in spite of the presence of floating capacitors in the original op-amp-RC circuits and it utilizes a small number of transconductors. Moreover, simultaneous multiple outputs are easily available in the transformed current-mode Gm-C circuits.

Journal ArticleDOI
TL;DR: In this paper, a sine-to-triangular wave converter based on operational transconductance amplifiers (OTAs) is proposed, which exploits a hyperbolic tangent behavior of an OTA to convert from a SINR wave to a triangular wave.
Abstract: A sine-to-triangular wave converter based on operational transconductance amplifiers (OTAs) is proposed in this paper. The realization method exploits a hyperbolic tangent behavior of an OTA to convert from a sine wave to a triangular wave. The proposed circuit provides a simple configuration and employs only OTAs as the active elements. This scheme can be operated in both a voltage-controlled mode and a current-controlled mode. The performances of the proposed circuit are discussed in detail and confirmed by the experimental results.

Journal ArticleDOI
TL;DR: In this paper, a channel split-and-add method for designing oversampled transmultiplexers and filter banks is presented. The proposed method is based on an initial design with an additional number of bands, which is then reduced to the desired value by the proper combination of adjacent and/or nonadjacent bands (subchannels).
Abstract: In this paper, a new structure, called the channel split-and-add method, for designing oversampled transmultiplexers and filter banks is presented. The proposed method is based on an initial design with an additional number of bands. The band number is then reduced to the desired value by the proper combination of adjacent and/or nonadjacent bands (subchannels). With the proposed approach it is always possible to perform the filtering tasks at the lowest data rate of the system. An example illustrates the design flexibility achieved with the proposed structure.

Journal ArticleDOI
TL;DR: In this article, a sliding mode control algorithm for a third-order uncertain, nonlinear, and time-varying dynamic system subject to unknown disturbance and input constraint is proposed.
Abstract: In this paper a new sliding mode control algorithm for a third-order uncertain, nonlinear, and time-varying dynamic system subject to unknown disturbance and input constraint is proposed. The algorithm employs a time-varying switching plane. At the initial time t = t0, this plane passes through the point determined by the system initial conditions in the error state space. Afterwards, the switching plane moves with a constant velocity to the origin of the space. In order to select the switching plane parameters, the integral of the time multiplied by the absolute error is minimised. Two types of input signal constraints are considered. First a conventional constraint expressed by an inequality is analysed, and then an elastic (or stretchable) constraint is taken into account. In the second case, we assume that the threshold value of the system input signal is known and that exceeding this value is undesirable, however not definitely forbidden. Exceeding this value is acceptable if it leads to an essential improvement of the system performance. In both cases the switching plane is chosen in such a way that the reaching phase is eliminated, insensitivity of the system to the external disturbance and the model uncertainty is guaranteed from the very beginning of the proposed control action, and fast, monotonic error convergence to zero is achieved.

Journal ArticleDOI
TL;DR: In this article, a Gaussian mixture model-based Bayesian analysis for blind source separation of an underdetermined model that has more sources than sensors is proposed, which follows a hierarchical learning procedure and alternative estimations for sources and the mixing matrix.
Abstract: This paper proposes a Gaussian mixture model-based Bayesian analysis for blind source separation of an underdetermined model that has more sources than sensors. The proposed algorithm follows a hierarchical learning procedure and alternative estimations for sources and the mixing matrix. The independent sources are estimated from their posterior means, and the mixing matrix is estimated by the maximum likelihood method. Because each source is conditionally correlated with others in its Markov blanket, the correlations between them are approximated by using linear response theory; this is based on the factorized approximation to the sources' true posteriors. In this framework, each source is modeled as a mixture of Gaussians to fit its actual distribution. Given enough Gaussians, the mixture model can learn any distribution. The algorithm provides a good identification of the mixing system, and its flexibility speeds up the convergence. The iterative learning for Gaussians leads to a parametric density estimation for all hidden sources as well as their recovery in the end. The major advantages of this algorithm are its flexibility and its fast convergence. Simulations using synthetic data validate the effectiveness of the algorithm.

Journal ArticleDOI
Abstract: Kibble's bivariate gamma distribution has been around since the 1940s and has been applied in several areas of electrical and electronic engineering. However, it seems that no explicit expressions for its product moments are available. In this short note, we derive a general expression for the product moments and also provide several particular forms.