Experimental study of a distributed active noise control system with multi-device nodes based on augmented diffusion strategy.

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.

Sound field reconstruction using a compact acoustics-informed neural network.

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.

An augmented diffusion algorithm with bidirectional communication for a distributed active noise control system.

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.

Transmission loss and directivity of sound transmitted through a slit on ground.

An analytical model is proposed for sound transmission through a slit on a rigid ground based on the modal superposition method to investigate the transmission loss (TL). A simple formula is derived for estimation of the TL for plane waves with and without the ground, which gives a more precise prediction than existing approaches. It is found that a larger slit height generally decreases the TL, except at the resonant frequencies of the slit.

A room impulse response database for multizone sound field reproduction (L).

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.

Evolutionary array optimization for multizone sound field reproduction.

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.

Improving the performance of an active staggered window with multiple resonant absorbers.

The active noise control (ANC) technique has been applied in staggered windows to improve the noise reduction at low frequencies. The control performance of such a system deteriorates significantly at some frequencies where the secondary source cannot radiate effectively due to the reflection at the boundaries of the staggered window. A resonant absorber consisting of a perforated panel and coiled up tubes is proposed to solve the problem.

An experimental study on transfer function estimation using acoustic modelling and singular value decomposition.

Transfer functions relating sound source strengths and the sound pressure at field points are important for sound field control. Recently, two modal domain methods for transfer function estimation have been compared using numerical simulations. One is the spatial harmonic decomposition (SHD) method, which models a sound field with a series of cylindrical waves; while the other is the singular value decomposition (SVD) method, which uses prior sound source location information to build an acoustic model and obtain basis functions for sound field modelling.

MSBOTS: a multiple small biological organism tracking system robust against non-ideal detection and segmentation conditions.

Accurately tracking a group of small biological organisms using algorithms to obtain their movement trajectories is essential to biomedical and pharmaceutical research. However, object mis-detection, segmentation errors and overlapped individual trajectories are particularly common issues that restrict the development of automatic multiple small organism tracking research. Extending on previous work, this paper presents an accurate and generalised Multiple Small Biological Organism Tracking System (MSBOTS), whose general feasibility is tested on three types of organisms.

Broadband noise insulation of windows using coiled-up silencers consisting of coupled tubes.

It has been demonstrated that a staggered window achieves better noise reduction performance than a traditional single glazing one at middle to high frequencies while maintaining a degree of natural ventilation. There is, however, little improvement in the low frequency range. In contrast, this work proposes to apply coiled-up silencers consisting of coupled tubes on the side walls of staggered windows to obtain noise attenuation in a broad band, especially in the low frequency range.

A comparison between two modal domain methods for personal audio reproduction.

Personal audio provides private and personalized listening experiences by generating sound zones in a shared space with minimal interference between zones. One challenge of the design is to achieve the best performance with a limited number of microphones and loudspeakers. In this paper, two modal domain methods for personal audio reproduction are compared.

Mitigating wind noise with a spherical microphone array.

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.

Development of a clarity parameter using a time-varying loudness model.

The perceived sound clarity is often estimated with the clarity index, which is calculated on the basis of physical acoustic measures that can correlate weakly to the way humans perceive sound for certain test conditions. Therefore, this study proposes a clarity parameter based on a binaural room impulse response processed with a time-varying loudness model. The proposed parameter is validated by calculating the correlation coefficient with subject responses collected from previous listening experiments.

Spatial decorrelation of wind noise with porous microphone windscreens.

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.

Wind noise spectra in small Reynolds number turbulent flows.

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.

On the wind noise reduction mechanism of porous microphone windscreens.

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.

Comparison of psychoacoustic-based reverberance parameters.

This study compared psychoacoustic reverberance parameters to each other, as well as to reverberation time (RT) and early decay time (EDT) under various acoustic conditions. The psychoacoustic parameters were loudness-based RT (T), loudness-based EDT [EDT; Lee, Cabrera, and Martens, J. Acoust.

Pressure spectra in turbulent flows in the inertial and the dissipation ranges.

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.

Improving Visual Saliency Computing With Emotion Intensity.

Saliency maps that integrate individual feature maps into a global measure of visual attention are widely used to estimate human gaze density. Most of the existing methods consider low-level visual features and locations of objects, and/or emphasize the spatial position with center prior. Recent psychology research suggests that emotions strongly influence human visual attention.

Uncertainties of reverberation time estimation via adaptively identified room impulse responses.

This paper investigates the reverberation time estimation methods which employ backward integration of adaptively identified room impulse responses (RIRs). Two kinds of conditions are considered; the first is the "ideal condition" where the anechoic and reverberant signals are both known a priori so that the RIRs can be identified using system identification methods. The second is that only the reverberant speech signal is available, and blind identification of the RIRs via dereverberation is employed for reverberation time estimation.

Reverberation time estimation from speech signals based on blind room impulse response identification.

A reverberation time (RT) estimation method is presented which consists of three steps, the anechoic speech is first recovered by maximizing the skewness of the linear prediction residual of the reverberant speech, then room impulse response (RIR) is identified using the recovered anechoic and reverberant speech, finally RIR is truncated to compensate the estimation errors and RT is estimated using the Schroeder's method. Simulations show that the proposed method successfully estimates RT less than 1.4 s and is insensitive to the speech content such as the number of long pauses and sharp offsets.

Acoustic contrast control in an arc-shaped area using a linear loudspeaker array.

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.

Delivering sound energy along an arbitrary convex trajectory.

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.