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Broadband adjustable sound absorbers are desired for controlling the acoustic conditions within enclosed spaces. Existing studies on acoustic absorbers, either passive or active, aim to maximize the sound absorption coefficients over an extended frequency band. By contrast, this paper introduces a tunable acoustic absorber, whose working frequency band and sound absorption characteristics can be defined by users for different applications. The approach leverages an error signal that can be synthesized using a standing wave separation technique. The error signal encodes different target reflection coefficients, leading to arbitrary absorption coefficients between 0 and 1. Experimental validation is conducted in a one-dimensional standing wave tube, demonstrating that the proposed active absorber achieves near-perfect absorption within the 150–1600 Hz frequency range, boasting an average absorption coefficient of 0.98. Adjustable absorption is demonstrated across three octave bands, aligning closely with theoretical predictions. Furthermore, when coupled with a shaping filter, the absorber exhibits spectrally tunable broadband absorption capabilities, selectively reflecting specific frequency bands while effectively absorbing others. These outcomes underscore the versatile tunability of the proposed active acoustic absorber, which is expected to pave the way for personalized regulating of the indoor acoustic environment. [ABSTRACT FROM AUTHOR]

A recent paper by Zhang, Wang, Duan, Tao, Zou, and Qiu [(2022). J. Acoust. Soc. Am. 152, 2840–2848] proposed that the average distance between physical microphones in remote acoustic sensing should be no larger than half the wavelength to ensure the coherence between the virtual signal and physical signals no less than 0.9 in complex sound fields. In this letter, the effects of the sound sources' distance on the coherence between the virtual signal and physical signals are further investigated, and it is found that the required average distance among physical microphones becomes smaller when sound sources are closer to the microphones. [ABSTRACT FROM AUTHOR]

The prediction of reverberant sound fields generated by a directional source is of great interest because practical sound sources are not omnidirectional, especially at high frequencies. For an arbitrary directional source described by cylindrical and spherical harmonics, this paper developed a modal expansion method for calculating the reverberant sound field generated by such a source in both two-dimensional and three-dimensional rectangular enclosures with finite impedance walls. The key is to express the modal source density using the cylindrical or spherical harmonic expansion coefficients of the directional source. A method based on the fast Fourier transform is proposed to enable the fast computation of the summation of enclosure modes when walls are lightly damped or rigid. This makes it possible to obtain accurate reverberant sound fields even in a large room and/or at high frequencies with a relatively low computational load. Numerical results with several typical directional sources are presented. The efficiency and the accuracy of the proposed method are validated by the comparison to the results obtained using the finite element method. [ABSTRACT FROM AUTHOR]

As environmental changes are inevitable, online algorithms that can track the variation of acoustic transfer functions are preferred in sound zone control (SZC) systems. In this paper, it is found that for the time domain SZC, the nonuniqueness problem caused by the coupling in multichannel transfer function modeling can be solved with time-varying control filters. To keep a stable acoustic contrast (AC) performance irrespective of environmental changes and also avoid the extreme distortion introduced by maximizing the AC in the time domain, the objective of maintaining a desired level of AC is used and an adaptive control algorithm is developed. The simulations validate that the proposed method tracks the environmental changes and adapts the control filters to give a stable AC performance. [ABSTRACT FROM AUTHOR]

The far field directivity is a straightforward indicator to describe the radiation pattern of the audio sound generated by a parametric array loudspeaker (pal), but its accurate and computationally efficient prediction is still challenging at present. This paper derives two-dimensional (2D), three-dimensional (3D), and 3D axisymmetric convolution models for calculating the far field directivity based on the quasilinear solution of Westervelt equation. The obtained expressions are expressed as linear and spherical convolutions of the ultrasound directivity and Westervelt directivity for 2D and 3D models, respectively. To improve prediction accuracy, the obtained expression is multiplied by an effective directivity resulted from the aperture factor of audio sound. The calculated directivities are compared against the exact solution obtained using the cylindrical and spherical wave expansions for 2D and 3D models, respectively. Numerical results with piston, apodized, and steerable profiles in both 2D and 3D models show that the proposed modified convolution model agrees well with the exact solution. It is also found that sidelobes appear in the audio sound directivity at large aperture sizes and high audio frequencies due to the aperture factor of audio sound, which can be predicted with the proposed method with a relatively low computational expenditure. [ABSTRACT FROM AUTHOR]

Remote acoustic sensing can be used to estimate the error signals in human ears without placing any physical microphones there. In this paper, the coherence between the signals picked up by physical microphones over a sphere surface and the signal obtained at the sphere center is investigated. Based on the multiple channel coherence formulas in the time domain and frequency domain, the relationship between the coherence and the placement of physical microphones is analyzed by numerical simulations first, then the experimental results obtained in a reverberation chamber and a car cabin are presented to verify the simulation results. Finally, a placement of physical microphones for active control of road noise in car cabins is discussed. Both the numerical and experimental results show that an upper limit frequency exists for accurate sound pressure estimation at the center of a sphere with the sound pressure on the sphere surface. For a sufficiently complex sound field such as that in a reverberation room or in a car, half the wavelength of the upper limit frequency is about the average distance among the physical microphones. [ABSTRACT FROM AUTHOR]

A steerable parametric array loudspeaker (PAL) can electronically steer highly directional audio beams in the desired direction. The challenge of modelling a steerable PAL is to obtain the audio sound pressure in both near and far fields with a low computational load. To address this issue, an extension of the spherical wave expansion is proposed in this paper. The steerable velocity profile on the radiation surface is expanded as Zernike polynomials which are an orthogonal and form a complete set over a unit circle. An expression for the radiated audio sound is then obtained using a superposition of Zernike modes. Compared to the existing methods, the proposed expansion is computationally efficient and provides a rigorous transformation of the quasilinear solution of the Westervelt equation without paraxial approximations. The proposed expansion is further extended to accommodate local effects by using an algebraic correction to the Westervelt equation. Numerical results for steering single and dual beams are presented and discussed. It is shown that the single beam can be steered in the desired direction in both near and far fields. However, dual beams cannot be well separated in the near field, which cannot be predicted by the existing far field models. [ABSTRACT FROM AUTHOR]

In sound zone control, the pressure matching (PM) algorithm is generally considered to achieve relatively low acoustic contrast (AC) regardless of the desired sound field. In this paper, we propose a generalized PM framework to generate a desired sound field with variable AC. The framework indicates that the choice of the desired sound field solely determines the AC performance of PM when the physical setting and acoustic environment remain unchanged. In the extreme, PM obtains the same AC as the acoustic contrast control method, which is widely accepted as the algorithm leading to the maximum AC. Based on the proposed framework, we design a variant coordinate descent algorithm to adjust the desired sound field by maximizing the AC under the constraint of the reconstruction error. Simulation results validate the efficacy and flexibility of the proposed framework in constrained sound zone control. [ABSTRACT FROM AUTHOR]

This work investigates the scattering by a rigid sphere of audio sound generated by a parametric array loudspeaker (pal). A computationally efficient method utilizing a spherical harmonic expansion is developed to calculate the quasilinear solution of audio sound fields based on both Kuznetsov and Westervelt equations. The accuracy of using the Westervelt equation is examined, and the rigid sphere scattering effects are simulated with the proposed method. It is found the results obtained using the Westervelt equation are inaccurate near the sphere at low frequencies. Contrary to conventional loudspeakers, the directivity of the audio sound generated by a pal severely deteriorates behind a sphere, as the ultrasounds maintaining the directivity of the audio sound are almost completely blocked by the sphere. Instead, the ultrasounds are reflected and generate audio sound on the front side of the sphere. It means that a listener in front of the pal will hear the audio sound scattered back after introducing the sphere as if it is reflected by the sphere. The experiment results are also presented to validate the numerical results. [ABSTRACT FROM AUTHOR]

This paper investigates the feasibility of remotely generating a quiet zone in an acoustic free field using multiple parametric array loudspeakers (PALs). A primary sound field is simulated using point monopoles located randomly in a two-dimensional plane, or three-dimensional (3D) space, whereas the secondary sound field is generated by multiple PALs uniformly distributed around the circumference of a circle sitting on the same plane as the primary sources, or on the surface of a sphere for 3D space. A quiet zone size is defined as the diameter of the maximal circular zone within which the noise reduction is greater than 10 dB. The size of this quiet zone is found to be proportional to 0.19λN for N secondary sources with a wavelength λ when the primary and secondary sources are in the same plane, whereas it is found to be 0.55λN1/2 for the 3D case. The size of the quiet zones generated by PALs is similar to that observed with traditional omnidirectional loudspeakers; however, the effects of using PALs on the sound field outside the target zone is much smaller due to their sharp radiation directivity and slow decay rate along the propagation distance. Experimental results are also presented to validate these numerical simulations. [ABSTRACT FROM AUTHOR]

In this work, a cylindrical expansion for the audio sound generated by a steerable baffled parametric array loudspeaker (PAL) based on the phased array technique is derived from the Westervelt equation. The expansion is a series of twofold summations with uncoupled angular and radial components in the cylindrical coordinate system. The angular component is determined by the trigonometric functions, and the radial component is an integral containing the Bessel functions and an arbitrary excitation velocity profile. The numerical results for a typical steerable PAL are presented and compared to those obtained using the convolution model. It is found that the prediction of the audio sound using the proposed cylindrical expansion improves the agreement with the experimental results when compared to the existing models. This is because no further approximations are required in the cylindrical expansion of the quasilinear solution of the Westervelt equation, whereas the complex near field nonlinear interactions between the ultrasonic waves cannot be correctly captured in a convolution model. The proposed cylindrical expansion does, therefore, provide an alternative approach to modeling a phased array PAL and high accuracy with a relatively low computational cost. [ABSTRACT FROM AUTHOR]

The phase-shifting based active noise control (ANC) algorithm without secondary path modeling is appealing due to its simple implementation and high computational efficiency. However, the algorithm suffers from the influence of an implicit circular convolution caused by directly applying the most appropriate phase shift to the reference signal in the frequency domain. The implicit circular convolution results in a poor linear relationship between the phase-shifted reference signal and the error signal, which considerably degrades the convergence performance of the ANC system. In this paper, two methods are proposed to deal with this problem. In the first method, the circular wraparound artifacts are removed by the time-domain constraining operation. In the second method, a noncausal phase-shifting filter is used to regularize the phase of the reference signal. Compared with the standard frequency-domain implementation, the two proposed algorithms both eliminate the influence caused by the wraparound artifacts of the circular convolution and improve the linear relationship between the phase-shifted reference signal and the error signal, leading to a higher level of noise attenuation with a tolerable increase in computational complexity. Simulation results validate the better noise attenuation ability of the proposed algorithms. [ABSTRACT FROM AUTHOR]

The generalized leaky filtered–x least mean square (GLFxLMS) algorithm can reduce the noise amplification caused by the waterbed effect in feedback active control systems effectively; however, it suffers from a high computation complexity. Hence, a frequency band constrained filtered–x least mean square algorithm is proposed to reduce the computation complexity of the GLFxLMS algorithm by replacing the penalty term containing a symmetric Toeplitz matrix in the cost function with the mean square of a penalty signal. The simulation results based on the measured transfer functions of an active headrest system show that the proposed algorithm has the same performance as the GLFxLMS algorithm, but with much lower computation complexity. [ABSTRACT FROM AUTHOR]