The authors previously proposed an improved multiband‐structured subband adaptive filter (IMSAF) algorithm. In this contribution, they first present two delayless structures of the IMSAF algorithm to remove delay in the signal path. Then, they study the transient and steady‐state behaviour of the IMSAF algorithms based on the energy conservation arguments and paraunitary condition imposed on the analysis and synthesis filter banks. The analysis does not require a model for the input signal. Simulation results show that the proposed delayless IMSAF algorithm has a faster convergence rate than the traditional delayless subband adaptive filtering algorithms. Theoretical analysis of the IMSAF algorithm is in good agreement with computer simulation results.

Recently, the general Volterra filter (VF) has been adopted for the modeling of parametric loudspeakers. However, the computation complexity of the VF is too high for real-time implementation. In this paper, a one-dimension Volterra filter (ODVF) with much lower complexity is introduced to model and compensate for the nonlinearity of parametric loudspeakers. A theoretical framework for ODVF model identification is established and a method of measuring the ODVF kernels using the exponential swept-sine signal is provided. The validity of modeling the nonlinearity of the parametric loudspeaker using the ODVF is verified theoretically and experimentally. Based on the established ODVF model, an inverse filter is designed to compensate for the 2nd harmonic distortion of the parametric loudspeakers. To further reduce the 3rd harmonic distortion, an improved compensation method is also proposed. Experimental results show that the performance of the ODVF-based compensation is comparable to that of the Volterra-filter based compensation.

Recently, a multiband-structured subband adaptive filter (MSAF) algorithm was proposed to speed up the convergence of the normalized least-mean-square (NLMS) algorithm. In this letter, we extend this work and propose an improved multiband-structured subband adaptive filter (IMSAF) algorithm to increase the convergence speed of the MSAF, which can also be regarded as a unifying framework for the NLMS, MSAF, and affine projection (AP) algorithms. The proposed optimization criterion is based on the principle of minimal disturbance, canceling the most recentPa posteriori errors in each of theNsubbands. The stability condition and the computational complexity are also analyzed. Computer simulations in the context of system identification demonstrate the effectiveness of the new algorithm.

Based on the Berktay's farfield solution, various preprocessing methods were proposed to reduce the distortion of the highly directional audible signal in the parametric loudspeaker. However, the Berktay's farfield solution is an approximated model of nonlinear acoustic propagation. To determine the effectiveness of these methods, we analyze various preprocessing methods theoretically for directional speech reproduction using the Khokhlov-Zabolotskaya-Kuznetsov (KZK) equation, which provides a more accurate model of nonlinear acoustic propagation. In order to reduce the distortion effectively in the parametric loudspeaker with these preprocessing methods, the initial sound pressure level of the carrier frequency is found to be less than 132 dB according to the KZK equation. Unlike the Berktay' farfield solution that results in a +12 dB/octave gain slope, different gain slopes are derived using the KZK equation and appropriate equalizers are proposed to improve the frequency response of the parametric loudspeaker. The optimal preprocessing method for directional speech reproduction is established based on the KZK equation, which has a relatively flat frequency response of the desired speech signal and the best total harmonic distortion performance.

A highly directional speech signal can be generated using parametric loudspeaker. The generation of highly directional sound beam is due to the nonlinear interaction of amplitude-modulated ultrasound waves in air. However, severe distortion is also generated during the reproduction of directional speech and several preprocessing techniques based on the Berktay's farfield model have been proposed by researchers to reduce the distortion. In this paper, we carried out a thorough investigation on the analytical performance of the recursive single-sideband amplitude modulation (RSSB-AM) technique, which has been found to perform well for directional speech reproduction. Several important characteristics of the performance of the RSSB-AM are observed and optimal parameters of the RSSB-AM are also presented.

We have developed a directional audible sound reproduction system using ultrasonic transducers. This novel directional audio is a result of the nonlinear propagation and subsequent self-demodulation of high-intensity amplitude modulated (AM) ultrasonic waves. In this paper, the propagation characteristic of the directional sound generated by parametric acoustic arrays is investigated using numerical algorithm in the time-domain and three preprocessing methods are employed to reduce the distortions of audio beams. Based on the theoretical analysis and computation simulations, a directional loudspeaker has been designed and prototyped. Furthermore, two directional loudspeakers are applied to the sound reproduction in a target zone. The scattering of sound by sound is implemented in the vicinity of interaction region of two sound beams and verified by the experiments carried out in an anechoic chamber.