Publications by authors named "Sipei Zhao"

The conventional wave field synthesis (WFS) theory is based on the free field assumption and the performance of systems based on it deteriorates significantly in reverberant environments. By introducing an error microphone array to monitor reproduction errors, the adaptive WFS (AWFS) system adjusts the loudspeaker signals to correct the sound field in reverberant environments. The AWFS system utilizes a centralized control strategy with a single processor, which imposes a high computational burden on the processor due to global error estimation, limiting the application scale.

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Idlers are essential to conveyor systems, as well as supporting and guiding belts to ensure production efficiency. Proper idler maintenance prevents failures, reduces downtime, cuts costs, and improves reliability. Most studies on idler fault detection rely on supervised methods, which depend on large labelled datasets for training.

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Recently, distributed active noise control (DANC) algorithms have been explored as a way to reduce computational complexity while ensuring system stability, thereby outperforming conventional centralized and decentralized algorithms. Most existing DANC algorithms assume that each node has only one pair of loudspeaker and microphone, limiting their flexibility in practical applications. In contrast, this paper proposes a DANC algorithm with general multi-device nodes based on the recently developed augmented diffusion strategy, allowing flexible and scalable ANC applications.

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Sound field reconstruction (SFR) augments the information of a sound field captured by a microphone array. Using basis function decomposition, conventional SFR methods are straightforward and computationally efficient but may require more microphones than needed to measure the sound field. Recent studies show that pure data-driven and learning-based methods are promising in some SFR tasks, but they are usually computationally heavy and may fail to reconstruct a physically valid sound field.

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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.

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Acoustic beamforming aims to focus acoustic signals to a specific direction and suppress undesirable interferences from other directions. Despite its flexibility and steerability, beamforming with circular microphone arrays suffers from significant performance degradation at frequencies corresponding to zeros of the Bessel functions. To conquer this constraint, baffled or concentric circular microphone arrays have been studied; however, the former need a bulky baffle that interferes with the original sound field, whereas the latter require more microphones that increase the complexity and cost, both of which are undesirable in practical applications.

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Personal sound systems have received significant research interest in the past two decades due to their promising applications in a variety of scenarios. Various methods have been proposed to generate personal sound zones, most of which are based on sound pressure manipulation in both the acoustically bright and dark zones. Since sound intensity is closely related to human perception of sound localization, this paper proposes an acoustic contrast control method with a sound intensity constraint to increase the spatial planarity in the bright zone.

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Recent studies on diffusion adaptation for distributed active noise control (DANC) systems have attracted significant research interest due to their balance between computational burden and stability compared to conventional centralized and decentralized adaptation schemes. The conventional multitask diffusion FxLMS algorithm assumes that the converged solutions of all control filters are consistent to each other, which is unrealistic in practice hence results in inferior performance in noise reduction. An augmented diffusion FxLMS algorithm has been proposed to overcome this problem, which adopts a neighborhood-wide adaptation and node-based combination approach to mitigate the bias in the converged solution of the multitask diffusion algorithms.

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Wind noise is notorious for its detrimental impacts on audio devices. This letter evaluates the influence of wind noise on the active noise control performance of headphones in a wind tunnel, and the noise reduction is found to decrease with wind speeds. To improve the performance of noise control systems in windy environments, the filtered-x least mean squares algorithm is modified based on the total least squares technique, taking the characteristics of wind noise into account.

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This letter introduces a database of Room Impulse Responses (RIRs) measured in seven different rooms for multizone sound field reproduction research in various acoustic environments. A circular array of 60 loudspeakers was installed in each room, with two microphone arrays placed sequentially in five different zones inside the loudspeaker array. A total of 260 400 RIRs were measured to establish the database.

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Multizone sound field reproduction aims to generate personal sound zones in a shared space with multiple loudspeakers. Traditional multizone sound field reproduction methods have focused on optimizing the source strengths given a preset array configuration. Recently, however, various methods have explored optimization of the loudspeaker locations.

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Various array patterns, such as circular, linear, and arc-shaped arrays, have been used in multi-zone sound field reproduction, but most of them are based on empirical rather than judicious selection. This article proposes an iterative optimization method to select the loudspeaker positions from a large set of candidate locations. Both the number and locations of the loudspeakers can be designed with superior performance.

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Microphones have been extensively studied for many decades and their related theories are well-established. However, the physical presence of the sensor itself limits its practicality in many sound field control applications. Laser Doppler vibrometers (LDVs) are commonly used for the remote measurement of surface vibration that are related to the sound field without the introduction of any such physical intervention.

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Static pressure tubes are widely used to measure the static pressure in turbulent flows. Existing work focuses on the alteration of the static pressure tubes to the flow field. This paper investigates the effects of the geometric properties of a static pressure tube on the frequency response.

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Simultaneous measurements of wind velocity and pressure fluctuations were conducted in a wind tunnel to investigate the wind noise source inside compact spherical open celled porous windscreens. The existing outdoor wind noise models are found to be inadequate to predict the wind noise inside a wind tunnel. This paper proposes a model to predict the interior stagnation pressure, which agrees with the wind noise measured inside the windscreen within a bandwidth, where the exterior turbulence-turbulence interaction pressure overestimates the wind noise level.

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This paper utilizes a rigid spherical microphone array to reduce wind noise. In the experiments conducted, a loudspeaker is used to reproduce the desired sound signal and an axial fan is employed to generate wind noise in an anechoic chamber. The sound signal and wind noise are measured separately with the spherical microphone array and analyzed in the spherical harmonic domain.

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This paper explores the wind noise reduction mechanism of porous microphone windscreens by investigating the spatial correlation of wind noise. First, the spatial structure of the wind noise signal is studied by simulating the magnitude squared coherence of the pressure measured with two microphones at various separation distances, and it is found that the coherence of the two signals decreases with the separation distance and the wind noise is spatially correlated only within a certain distance less than the turbulence wavelength. Then, the wind noise reduction of the porous microphone windscreen is investigated, and the porous windscreen is found to be the most effective in attenuating wind noise in a certain frequency range, where the windscreen diameter is approximately 2 to 4 times the turbulence wavelengths (2 < D/ξ < 4), regardless of the wind speed and windscreen diameter.

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Wind noise spectra caused by wind from fans in indoor environments have been found to be different from those measured in outdoor atmospheric conditions. Although many models have been developed to predict outdoor wind noise spectra under the assumption of large Reynolds number [Zhao, Cheng, Qiu, Burnett, and Liu (2016). J.

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This paper investigates the wind noise reduction mechanism of porous microphone windscreens. The pressure fluctuations inside the porous windscreens with various viscous and inertial coefficients are studied with numerical simulations. The viscous and inertial coefficients represent the viscous forces resulting from the fluid-solid interaction along the surface of the pores and the inertial forces imposed on the fluid flow by the solid structure of the porous medium, respectively.

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Based on existing studies that provide the pressure spectra in turbulent flows from the asymptotic pressure structure function in the inertial range, this paper extends the pressure spectrum to the dissipation range by proposing a pressure structure function model that incorporates both the inertial and dissipation ranges. Existing experiment results were used to validate the proposed pressure structure function model first, and then the obtained pressure spectrum was compared with the simulation and measurement data in the literature and the wind-induced noise measured outdoors. All comparisons demonstrate that the pressure spectrum obtained from the proposed pressure structure function model can be used to estimate the pressure spectra in both the inertial and dissipation ranges in turbulent flows with a sufficiently large Reynolds number.

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This paper proposes a different method for calculating a sound field diffracted by a rigid barrier based on the integral equation method, where a virtual boundary is assumed above the rigid barrier to divide the whole space into two subspaces. Based on the Kirchhoff-Helmholtz equation, the sound field in each subspace is determined with the source inside and the boundary conditions on the surface, and then the diffracted sound field is obtained by using the continuation conditions on the virtual boundary. Simulations are carried out to verify the feasibility of the proposed method.

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This paper proposes a method of creating acoustic contrast control in an arc-shaped area using a linear loudspeaker array. The boundary of the arc-shaped area is treated as the envelope of the tangent lines that can be formed by manipulating the phase profile of the loudspeakers in the array. When compared with the existing acoustic contrast control method, the proposed method is able to generate sound field inside an arc-shaped area and achieve a trade-off between acoustic uniformity and acoustic contrast.

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Accelerating beams have attracted considerable research interest due to their peculiar properties and various applications. Although there have been numerous research on the generation and application of accelerating light beams, few results have been published on the generation of accelerating acoustic beams. Here we report on the experimental observation of accelerating acoustic beams along arbitrary convex trajectories.

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