Yoshinobu Kajikawa

Recent advances on active noise control : open issues and innovative applications

The problem of acoustic noise is becoming increasingly serious with the growing use of industrial and medical equipment, appliances, and consumer electronics. Active noise control (ANC), based on the principle of superposition, was developed in the early 20th century to help reduce noise. However, ANC is still not widely used owing to the effectiveness of control algorithms, and to the physical and economical constraints of practical applications. In this paper, we briefly introduce some fundamental ANC algorithms and theoretical analyses, and focus on recent advances on signal processing algorithms, implementation techniques, challenges for innovative applications, and open issues for further research and development of ANC systems.

Identification and elimination of second-order nonlinear distortion of loudspeaker systems using Volterra filter

Modeling a loudspeaker system with the Volterra series expansion is essential to eliminate the nonlinear distortion. We have proposed a method for measuring the Volterra kernel of a loudspeaker system by multi sinusoidal waves. This method, however, has the problems of not considering the phase property of the nonlinear element and the third-order distortion. Therefore, we propose a novel method measuring the second-order Volterra kernel by multi sinusoidal waves. In this method, the phase property of the nonlinear element and the third-order distortion are considered. Moreover, we off-line eliminate the nonlinear distortion of the loudspeaker system with the Volterra kernel measured by the novel method and show the methods effectiveness.

Active noise control without a secondary path model by using a frequency-domain simultaneous perturbation method with variable perturbation

We propose a frequency-domain active noise control (ANC) system without a secondary path model. The proposed system is based on the frequency-domain simultaneous perturbation (FDSP) method with variable perturbation. In this system, the coefficients of the adaptive filter are updated only by error signals. The conventional ANC system using the filtered-x algorithm becomes unstable due to the error between the secondary path, from secondary source to error sensor, and its model. In contrast, the proposed ANC system has the advantage of not using the model. Furthermore, the variable perturbation brings fast convergence. Simulation results demonstrate the efficiency of the proposed ANC system compared with the conventional ANC systems.

Active noise control system without secondary path model

The proposed system is based on the simultaneous perturbation optimization method with block process. Consequently, the coefficients of the adaptive filter in the proposed system are updated by only error signals. The conventional ANC system using the filtered-x algorithm becomes unstable due to the error between the secondary path and its model. On the other hand, the proposed ANC system has an advantageous property in not using the model. In this paper, we show the principle of the proposed ANC system and examine the efficiency of computer simulations.

An Estimation Method of Parameters for Closed-box Loudspeaker System

In this paper, we propose a method that uses Simulated Annealing (SA) to estimate the linear and nonlinear parameters of a closed-box loudspeaker system for implementing effective Mirror filters. The nonlinear parameters determined by W. Klippels method are sometimes inaccurate and imaginary. In contrast, the proposed method can estimate the parameters with satisfactory accuracy due to its use of SA; the resulting impedance and displacement characteristics match those of an actual equivalent loudspeaker. A Mirror filter designed around these parameters can well compensate the nonlinear distortions of the loudspeaker system. Experiments demonstrate that the method can reduce the levels of nonlinear distortion by 5 dB to 20 dB compared to the before compensation condition.

Identification of the parametric array loudspeaker with a volterra filter using the sparse NLMS algorithm

Volterra filters can be applied to a wide range of nonlinear systems, keeping only the low order kernels to yield a good approximation. The parametric array loudspeaker (PAL), as a weak nonlinear acoustic system, is an attractive directional sound reproduction device. Volterra filters have been adopted in the linearization system of the PAL that efficiently reduces the nonlinear distortion with no need of solving the nonlinear acoustic equation. In this paper, the ultrasound-to-ultrasound Volterra filter is proposed, being inspired by the nonlinear acoustic principle, to provide a better systematic representation of the PAL. Experiment results are presented to prove the effectiveness of the proposed approach, where the sparse NLMS algorithm is carried out in the identification.

A convolution model for computing the far-field directivity of a parametric loudspeaker array

This paper describes a method to compute the far-field directivity of a parametric loudspeaker array (PLA), whereby the steerable parametric loudspeaker can be implemented when phased array techniques are applied. The convolution of the product directivity and the Westervelts directivity is suggested, substituting for the past practice of using the product directivity only. Computed directivity of a PLA using the proposed convolution model achieves significant improvement in agreement to measured directivity at a negligible computational cost.

Linearization of Loudspeaker Systems Using a Subband Parallel Cascade Volterra Filter

In this paper, we propose a low complexity realization method for compensating for nonlinear distortion. Generally, nonlinear distortion is compensated for by a linearization system using a Volterra kernel. However, this method has a problem of requiring a huge computational complexity for the convolution needed between an input signal and the 2nd-order Volterra kernel. The Simplified Volterra Filter (SVF), which removes the lines along the main diagonal of the 2nd-order Volterra kernel, has been previously proposed as a way to reduce the computational complexity while maintaining the compensation performance for the nonlinear distortion. However, this method cannot greatly reduce the computational complexity. Hence, we propose a subband linearization system which consists of a subband parallel cascade realization method for the 2nd-order Volterra kernel and subband linear inverse filter. Experimental results show that this proposed linearization system can produce the same compensation ability as the conventional method while reducing the computational complexity.

Frequency domain active noise control system using optimal step-size parameters

We propose a frequency domain active noise control system using optimal step-size parameters at each frequency. The proposed ANC system can converge faster than the conventional ANC system using the Filtered-x LMS algorithm with the optimal step-size parameter. Moreover, the proposed system can converge by setting the step-size parameters at unstable frequencies to 0 in the case where the phase error of the secondary path model does not satisfy the stable condition, whereas the conventional ANC system cannot converge in this case. In this paper, the theoretical equation of the optimal step-size parameters is derived by using available information during system operation. Next, we present the structure of the ANC system using the optimal step-size parameters obtained from the theoretical equation. Moreover, a control technique determining unstable frequencies is introduced. Finally, simulation results demonstrate the efficiency of the proposed ANC system.

A comparative study of preprocessing methods in the parametric loudspeaker

The parametric loudspeaker is a directional sound reproduction device making use of the parametric sound generation. A sound beam is formed as a result of nonlinear interactions between ultrasonic beams. The parametric loudspeaker is advantageous in transmitting an equally narrow sound beam from a smaller emitter as compared to the conventional loudspeaker. Due to this advantage, parametric loudspeakers are readily applied in a variety of sound field control applications, such as creation of personal listening spots, spatial audio reproduction, and active noise control. However, there is a long concerned drawback of the parametric loudspeaker, whereby harmonic and intermodulation distortions are byproducts of the parametric sound generation. Hence, a comparative study of six preprocessing methods, including two proposed methods from this paper, is carried out. Harmonic and intermodulation distortions are demonstrated by experiments.