Peifeng Ji

An Improved Multiband-Structured Subband Adaptive Filter Algorithm

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 recent P a posteriori errors in each of the N subbands. 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.

An experimental acoustic cloak for generating virtual images

A two-dimensional acoustic cloak is realized to generate virtual image in air. The cloak is designed to make the backscattering characteristics of a quadrangular prism the same as that of a plate based on transformation acoustics. The required anisotropic parameters of the cloak shell are obtained by acoustic metamaterial made of perforated plates, which are easy to fabricate and unitize. The measurements of the backward and near-backward scattering fields confirm the validity of the proposed cloak. Experimental results show the possibility to hide the shape of an object by changing the reflection properties by using transformation acoustics.

Modeling and compensation for the distortion of parametric loudspeakers using a one-dimension Volterra filter

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.

Transient and steady-state analyses of the improved multiband-structured subband adaptive filter algorithm

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.

On the linearization of a parametric loudspeaker system by using third-order inverse Volterra filters

Due to the nonlinear interaction between the ultrasonic waves emitted by the parametric loudspeaker, a directional sound beam is generated along with harmonic distortion. It is known that the single sideband amplitude modulation (SSB-AM) technique is one of the most effective ways of reducing the harmonics. A 3rd-order inverse Volterra filter (VF) is designed on the basis of the SSB-AM in this paper to further compensate the nonlinearities of the parametric loudspeaker. Meanwhile, the diagonal Volterra filter (DVF) with the advantage of lower complexity compared to the conventional VF is also proposed. The experimental results demonstrate that the inverse DVF is able to reduce the harmonic distortion to meet high fidelity requirement nearly the same as the inverse VF while requiring less computational resources.

Design and demonstration of an underwater acoustic carpet cloak

The carpet cloak, which is designed to hide the objects placed on a reflecting surface, has become a topic of considerable interest. Inspired by those theoretical works, the experimental realization of acoustic carpet cloak in air host has been reported. However, due to the difficulty in obtaining the unit cell in reality, the underwater carpet cloak still remains in simulation thus far. Here, we design and fabricate a realizable underwater acoustic carpet cloak. By introducing a scaling factor, the structure of the carpet cloak, which is comprised of layered brass plates, is greatly simplified at the cost of some impedance match. The experimental results demonstrate a good performance of the proposed carpet cloak in a wide frequency range. Our work paves the way for future applications in the practical underwater devices.

Low-Complexity Implementation of the Improved Multiband-Structured Subband Adaptive Filter Algorithm

Previously, we proposed an improved multiband-structured subband adaptive filter (IMSAF) algorithm to accelerate the convergence rate of the MSAF algorithm. When the projection order and/or the number of subbands is increased, the convergence rate of the IMSAF algorithm improves at the cost of increased complexity. Thus, this paper proposes several approaches to reduce the complexity of the IMSAF algorithm, both in error vector calculation and matrix inversion operation. Specifically, three approaches are developed to efficiently calculate error vector. The first approach gives an approximate filtering, whereas the other two approaches can provide a fast exact filtering with or without update of the weight vector explicitly based on a recursive scheme. The decorrelation property of IMSAF is determined, and two simplified variants are developed to reduce the complexity as by-products, i.e., the simplified IMSAF (SIMSAF) and pseudo IMSAF algorithms. Then, we discuss the problem of solving a linear system of equations. The performance advantages, limitations, and preferable applications of various methods are analyzed and discussed. Computer simulations are conducted in the context of system identification to determine the principle and efficiency of the proposed fast algorithms.

Development of an acoustic filter for parametric loudspeaker using phononic crystals.

The spurious signal generated as a result of nonlinearity at the receiving system affects the measurement of the difference-frequency sound in the parametric loudspeaker, especially in the nearfield or near the beam axis. In this paper, an acoustic filter is designed using phononic crystals and its theoretical simulations are carried out by quasi-one- and two-dimensional models with Comsol Multiphysics. According to the simulated transmission loss (TL), an acoustic filter is prototyped consisting of 5×7 aluminum alloy cylinders and its performance is verified experimentally. There is good agreement with the simulation result for TL. After applying our proposed filter in the axial measurement of the parametric loudspeaker, a clear frequency dependence from parametric array effect is detected, which exhibits a good match with the well-known theory described by the Gaussian-beam expansion technique. During the directivity measurement for the parametric loudspeaker, the proposed filter has also proved to be effective and is only needed for small angles.

Generation of Audible Sound from Two Ultrasonic Beams with Dummy Head

In this study, the nonlinear interaction of two sound beams with a Knowles Electronics manikin for acoustic research (KEMAR) dummy head was investigated experimentally. An audible sound was generated under two conditions: two ultrasonic beams were in parallel and at right angle. It was found that the presence of the dummy head not only increased audible sound levels in KEMAR ears owing to a head scattering effect, but also led to different sound levels in two ears owing to a head shadow effect. Experimental results demonstrated that for the two parallel ultrasonic beams, head effects slightly reduced the audible sound levels behind the dummy head, but increased the audible sound area. Furthermore, it was shown that the audible sound could be generated within the intersection zone of two perpendicular ultrasonic beams, which is useful for localized sound delivery in audio engineering.

Development of a Directional Loudspeaker System for Sound Reproduction

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.