Yoichi Hinamoto

Equivalence between frequency-domain blind source separation and frequency-domain adaptive beamforming for convolutive mixtures

Frequency-domain blind source separation (BSS) is shown to be equivalent to two sets of frequency-domain adaptive beamformers (ABFs) under certain conditions. The zero search of the off-diagonal components in the BSS update equation can be viewed as the minimization of the mean square error in the ABFs. The unmixing matrix of the BSS and the filter coefficients of the ABFs converge to the same solution if the two source signals are ideally independent. If they are dependent, this results in a bias for the correct unmixing filter coefficients. Therefore, the performance of the BSS is limited to that of the ABF if the ABF can use exact geometric information. This understanding gives an interpretation of BSS from a physical point of view.

Equivalence between frequency domain blind source separation and frequency domain adaptive beamforming

Frequency domain Blind Source Separation (BSS) is shown to be equivalent to two sets of frequency domain adaptive microphone arrays, i.e., Adaptive Beamformers (ABFs). The minimization of the off-diagonal components in the BSS update equation can be viewed as the minimization of the mean square error in the ABF. The unmixing matrix of the BSS and the filter coefficients of the ABF converge to the same solution in the mean square error sense if the two source signals are ideally independent. Therefore, the performance of the BSS is limited by that of the ABF. This understanding. gives an interpretation of BSS from physical point of view.

Analysis of the filtered-X LMS algorithm and a related new algorithm for active control of multitonal noise

In the presence of tonal noise generated by periodic noise source like rotating machines, the filtered-X LMS (FXLMS) algorithm is used for active control of such noises. However, the algorithm is derived under the assumption of slow adaptation limit and the exact analysis of the algorithm is restricted to the case of one real sinusoid in the literature. In this paper, for the general case of arbitrary number of sources, the characteristic polynomial of the equivalent linear system describing the FXLMS algorithm is derived and a method for calculating the stability limit is presented. Also, a related new algorithm free from the above assumption, which is nonlinear with respect to the tap weights, is proposed. Simulation results show that in the early stage of adaptation the new algorithm gives faster decay of errors.

A Filtered-X LMS Algorithm for Sinusoidal Reference Signals—Effects of Frequency Mismatch

The objective of this letter is to analyze the effects of frequency mismatch for an adaptive algorithm that becomes the Filtered-X LMS algorithm when the reference signals are purely sinusoidal. The Filtered-X LMS algorithm is often used for active control of acoustic noise. For the case of sinusoidal noise sources, if there is a deviation between the frequency used in the adaptive algorithm and its true value (frequency mismatch), the performance of the Filtered-X LMS algorithm might degrade considerably. In this letter, using the equivalent transfer function method, the effects of frequency mismatch are analyzed precisely. Finally, computer simulations are presented to demonstrate the obtained results

Interface for barge-in free spoken dialogue system based on sound field reproduction and microphone array

A barge-in free spoken dialogue interface using sound field control and microphone array is proposed. In the conventional spoken dialogue system using an acoustic echo canceller, it is indispensable to estimate a room transfer function, especially when the transfer function is changed by various interferences. However, the estimation is difficult when the user and the system speak simultaneously. To resolve the problem, we propose a sound field control technique to prevent the response sound from being observed. Combined with a microphone array, the proposed method can achieve high elimination performance with no adaptive process. The efficacy of the proposed interface is ascertained in the experiments on the basis of sound elimination and speech recognition.

Convergence analysis of linear and nonlinear filtered-X LMS algorithms for active control of multitonal noise

In the presence of tonal noise generated by periodic noise source like rotating machines, the filtered-X LMS algorithm is used for active control of such noises. However, the algorithm is derived under the assumption of slow adaptation limit and the exact analysis of the algorithm is restricted to the case of one real sinusoid in the literature. In this paper for the general case of arbitrary number of sources, the characteristic polynomial of the equivalent linear system describing the filtered-X LMS algorithm is derived and a method for calculating the stability limit is presented. Also, a new nonlinear algorithm free from the above assumption is proposed. Simulation results show that in the early stage of adaptation the nonlinear algorithm gives faster decay of errors.

Normal-form state-space realization of single frequency IIR notch filters and its application to adaptive notch filters

This paper deals with the normal-form state-space realization of second-order IIR notch filters. A method for approximately realizing a normal-form state-space model from the transfer function of a single frequency notch filter is developed. The pole and zero sensitivity of the normal-form state-space model is then analyzed, and it is shown that the normal-form state-space notch filter has minimum pole sensitivity. In addition, a single frequency adaptive notch filter is constructed by applying the normal-form state-space model. In this connection, a method for generating gradients required to adjust the coefficients of a normal-form state-space adaptive notch filter is explored. A numerical example is presented to demonstrate the validity and effectiveness of the proposed adaptive notch filter.

Convergence analysis of an unbiased modeling algorithm in hearing aids

Abstract The convergence analysis is presented for an unbiased modeling algorithm in hearing aids recently proposed by Spriet, Moonen and Proudler. Using the ODE (ordinary differential equation) method the identi ability condition derived by them is shown to be also the convergence condition of the corresponding adaptive algorithm. The asymptotic covariance matrix of the parameter estimate is then derived and its validity is examined by simulations.

Weighted roundoff noise and pole sensitivity minimization subject to l 2 -scaling constraints for 2-D separable-denominator digital filters

The problem of minimizing roundoff noise and pole sensitivity simultaneously subject to l2-norm scaling constraints of dynamic range for 2-D digital filters with separable denominator is investigated. A novel measure for the evaluation of roundoff noise and pole sensitivity is introduced and then an effective method for minimizing the measure is explored by converting the constrained optimization problem into an unconstrained optimization problem and by employing an efficient quasi-Newton method. A numerical example is included to illustrate the effectiveness of the present method.

Minimized roundoff noise and pole sensitivity subject to L 2 -Scaling constraints for IIR filters

This paper investigates the minimization problem of weighted roundoff noise and pole sensitivity subject to l2-scaling constraints for state-space digital filters. A new measure for evaluating roundoff noise and pole sensitivity is proposed, and an efficient technique for minimizing this measure is developed. It is shown that the problem can be converted into an unconstrained optimization problem by using linear-algebraic techniques. The unconstrained optimization problem at hand is then solved iteratively by employing an efficient quasi-Newton algorithm with closed-form formulas for key gradient evaluation. Finally a numerical example is presented to demonstrate the validity and effectiveness of the proposed technique.