ORCID Profile
0000-0002-5181-1220
Current Organisations
University of Technology Sydney
,
The Adaptive Mind Cluster Project
,
Philipps-Universität Marburg
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Publisher: IEEE
Date: 09-2011
Publisher: IEEE
Date: 03-2016
Publisher: Elsevier BV
Date: 03-2019
Publisher: Elsevier BV
Date: 02-2010
Publisher: Acoustical Society of America (ASA)
Date: 12-2018
DOI: 10.1121/1.5082290
Abstract: 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. The wind noise is found to be irregularly distributed in the spherical harmonic domain, distinct from the sound signal which is concentrated in the first few spherical harmonic modes. This difference is utilized to reduce wind noise without degrading the desired sound pressure level (SPL) by use of a low pass filter method in the spherical harmonic domain. Experimental results with both single-tonal and multi-tonal sound signals demonstrate that the proposed method can reduce wind noise by more than 10 dB in the frequency range below 500 Hz. The SPL of the desired sound signal can be extracted from wind noise with an error within 1.0 dB, even when the sound level is 8 dB lower than wind noise.
Publisher: Acoustical Society of America (ASA)
Date: 07-2020
DOI: 10.1121/10.0001568
Abstract: Unlike the audio sound generated by traditional sources, the directivity of that generated by a parametric array loudspeaker (pal) deteriorates significantly after passing through a thin partition. To study this phenomenon, the pal radiation model based on the Westervelt equation, and the plane wave expansion method are used to calculate the sound fields behind a sheet of aluminum foil and a porous material blanket under the quasi-linear assumption, where the paraxial approximation is assumed only for ultrasonic waves. The audio sounds generated by a point monopole and a traditional directional source are presented for comparison. Both simulation and experiment results show that the transmitted sound from a pal behind the thin partition is small and less focused on the radiation axis because most of the ultrasounds forming the directivity of the pal is blocked by the thin partition which has little effect on the traditional audio sources.
Publisher: Elsevier BV
Date: 12-2015
Publisher: Elsevier BV
Date: 03-2013
Publisher: Elsevier BV
Date: 02-2019
Publisher: Elsevier BV
Date: 06-2018
Publisher: Institute of Noise Control Engineering (INCE)
Date: 03-01-2013
DOI: 10.3397/1.3702011
Publisher: IEEE
Date: 12-2015
Publisher: Acoustical Society of America (ASA)
Date: 2020
DOI: 10.1121/10.0000474
Abstract: 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. One is the spatial harmonic decomposition (SHD) based method and the other is the singular value decomposition (SVD) based method. It is demonstrated that the SVD based method provides a more efficient modal domain decomposition than the SHD method for 2.5 dimensional personal audio design. Simulation results show that the SVD based method outperforms the SHD one by up to 10 dB in terms of acoustic contrast and up to 17 dB in terms of reproduction error for a compact arc array with five loudspeakers, while requiring fewer microphones around the zone boundaries. The SVD based method retains the inherent efficiency of optimizing in a modal domain while avoiding the inherent geometric limitations of using SHD basis functions. Thus, this approach is advantageous for applications with flexible system geometries and a small number of loudspeakers and microphones.
Publisher: Elsevier BV
Date: 08-2013
Publisher: MDPI AG
Date: 25-07-2019
DOI: 10.3390/APP9152973
Abstract: The presence of control signal feedback to the reference microphone in feedforward active control systems deteriorates the control performance. A four-stage method is proposed in this paper to carry out online feedback path modelling with the control signal. It consists of controller initialization, feedback path modelling using decorrelation filters, active control operation, and feedback path change detection for maintaining the control operation. In contrast to the existing auxiliary noise injection method, the proposed method uses five switches and three thresholds to control and maintain the system stability by avoiding the interference between control operation and feedback path modelling, and adaptive decorrelation filters are used to increase the feedback path modelling performance. Simulation results reveal that the proposed method is capable of tracking feedback path changes without injecting any auxiliary noise and maintaining the noise reduction performance and stability of the system.
Publisher: Acoustical Society of America (ASA)
Date: 11-2015
DOI: 10.1121/1.4934267
Abstract: This paper proposes to reduce the radiation of a sound source inside a cavity through the baffled opening by using an array of loudspeakers and microphones. The system is called a planar virtual sound barrier because it acts like a concrete sound barrier to block the transmission of sound but does not affect light and air circulation. An analytical model for the planar virtual sound barrier is developed based on the modal superposition method to calculate the sound field in and outside a rectangular cavity with a baffled opening. After the model is verified with numerical simulations, a performance study of the planar virtual sound barrier is carried out based on the proposed analytical model, and then the results are confirmed by experiments. The mechanisms of the planar virtual sound barrier are investigated and it is found that three mechanisms work together in the system, including changing the impedance of the primary source, modal control, and modal rearrangement. It is also found that there exist some frequencies where the sound cannot be controlled if all the secondary sources are on the same plane parallel to the opening, and the reasons behind the phenomenon are explained.
Publisher: Acoustical Society of America (ASA)
Date: 11-2017
DOI: 10.1121/1.5012740
Abstract: 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. Acoust. Soc. Am. 140, 4178–4182, and the references therein], they cannot be applied directly to the indoor situations because the Reynolds number of wind from fans in indoor environments is usually much smaller than that experienced in atmospheric turbulence. This paper proposes a pressure structure function model that combines the energy-containing and dissipation ranges so that the pressure spectrum for small Reynolds number turbulent flows can be calculated. The proposed pressure structure function model is validated with the experimental results in the literature, and then the obtained pressure spectrum is verified with the numerical simulation and experiment results. It is demonstrated that the pressure spectrum obtained from the proposed pressure structure function model can be utilized to estimate wind noise spectra caused by turbulent flows with small Reynolds numbers.
Publisher: Acoustical Society of America (ASA)
Date: 2020
DOI: 10.1121/10.0000592
Abstract: 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. The limitations of the proposed model and other potential sources for wind noise inside porous windscreens are discussed.
Publisher: Institution of Engineering and Technology (IET)
Date: 2007
DOI: 10.1049/EL:20071938
Publisher: Elsevier BV
Date: 02-2010
Publisher: Elsevier BV
Date: 12-2008
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 12-2012
Publisher: Elsevier BV
Date: 10-2008
Publisher: Acoustical Society of America (ASA)
Date: 2023
DOI: 10.1121/10.0016819
Abstract: 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. The slit width has little effect on the TL at high frequencies, and the slit depth affects the resonant frequencies significantly even though it has little effect on the overall TL. Compared with the same size slit in the free field, the rigid ground reduces the TL at most frequencies, and that reduction is a constant between 3 and 9 dB in the low frequency range. It is also found that the sound transmitted through the slit is almost omnidirectional at low frequencies, while most of the sound energy at high frequencies falls within the range where the long side of the slit is located. The experimental results demonstrate the validity of the analytical model and the findings in numerical simulations.
Publisher: Acoustical Society of America (ASA)
Date: 04-2011
DOI: 10.1121/1.3553223
Abstract: Different non-exponential decays such as the concave and the convex double sloped decays in the coupled rooms provide distinct sound qualities. These are commonly considered to occur in the less reverberant sub-room and the more reverberant sub-room, respectively. However, numerical simulations and experiments in this paper show that the demarcation line is not located along the physical boundaries (e.g., the partition and the coupling aperture), but in the more reverberant sub-room. The sound field with the concave double sloped decay penetrates into the auxiliary sub-room to an extent which is influenced by the difference between the two natural reverberations of the sub-rooms. Furthermore the sound energy flows in different regions are investigated, demonstrating how energy feedback leads to the concave double sloped decay.
Publisher: Institute of Noise Control Engineering (INCE)
Date: 20-04-2018
DOI: 10.3397/1/376642
Publisher: Elsevier BV
Date: 06-2006
Publisher: Elsevier BV
Date: 09-2006
Publisher: Acoustical Society of America (ASA)
Date: 10-0110
DOI: 10.1121/1.4964752
Abstract: Compact loudspeaker arrays have wide potential applications as portable personal audio systems that can project sound energy to specified regions. It is meaningful to investigate the scattering effects on the array performance since the scattering of the users' heads is inevitable in practice. A five-channel compact endfire array is established and the regularized acoustic contrast control method is evaluated for the scenarios of one moving listener and one listener fixed in the bright zone while another listener moves along the evaluation region. Both simulations and experiments verify that the scattering has limited influence on the directivity of the endfire array.
Publisher: Elsevier BV
Date: 03-2015
Publisher: Institution of Engineering and Technology (IET)
Date: 08-2013
Publisher: Acoustical Society of America (ASA)
Date: 02-2015
DOI: 10.1121/1.4906184
Abstract: 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. The acoustic contrast created by the proposed method increases while the acoustic uniformity decreases with frequency.
Publisher: Acoustical Society of America (ASA)
Date: 02-2020
DOI: 10.1121/10.0000743
Abstract: The problem of the secondary path variation on feedforward active noise control (ANC) systems can be solved by applying a phase shift on the reference signal. The existing algorithms choose the phase shift from several fixed candidates by monitoring the residual noise power, which has the problem of deteriorated convergence speed, especially when the difference between the chosen phase shift and the true phase value is large. In this paper, an improved ANC algorithm is proposed by having a better estimation of the phase shift in each subband with the least squares method. Simulation results validate the faster convergence speed of the proposed algorithm, and its computational load is discussed.
Publisher: Elsevier BV
Date: 12-2011
Publisher: Acoustical Society of America (ASA)
Date: 03-2016
DOI: 10.1121/1.4943547
Abstract: 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. Results show that under the “ideal condition,” the average relative errors in 7 octave bands are less than 2% for white noise and 15% for speech, respectively, when both the anechoic and reverberant signals are available. In contrast, under the second condition, the average relative errors of the blindly identified RIR-based reverberation time estimation are around 20%−30% except the 63 Hz octave band. The fluctuation of reverberation times estimated under the second condition is more severe than that under the ideal condition and the relative error for low frequency octave bands is larger than that for high octave bands under both conditions.
Publisher: Elsevier BV
Date: 03-1996
Publisher: Acoustical Society of America (ASA)
Date: 08-2014
DOI: 10.1121/1.4884760
Abstract: Blind multichannel identification is generally sensitive to background noise. Although there have been some efforts in the literature devoted to improving the robustness of blind multichannel identification with respect to noise, most of those works assume that the noise is Gaussian distributed, which is often not valid in real room acoustic environments. This paper deals with the more practical scenario where the noise is not Gaussian. To improve the robustness of blind multichannel identification to non-Gaussian noise, a robust normalized multichannel frequency-domain least-mean M-estimate algorithm is developed. Unlike the traditional approaches that use the squared error as the cost function, the proposed algorithm uses an M-estimator to form the cost function, which is shown to be immune to non-Gaussian noise with a symmetric α-stable distribution. Experiments based on the identification of a single-input/multiple-output acoustic system demonstrate the robustness of the proposed algorithm.
Publisher: Elsevier BV
Date: 03-2007
Publisher: Elsevier BV
Date: 03-2016
Publisher: Acoustical Society of America (ASA)
Date: 08-2015
DOI: 10.1121/1.4926907
Abstract: Open-sphere microphone arrays are preferred over rigid-sphere arrays when minimal interaction between array and the measured sound field is required. However, open-sphere arrays suffer from poor robustness at null frequencies of the spherical Bessel function. This letter proposes a maximum likelihood method for direction of arrival estimation in the spherical harmonic domain, which avoids the ision of the spherical Bessel function and can be used at arbitrary frequencies. Furthermore, the method can be easily extended to wideband implementation. Simulation and experiment results demonstrate the superiority of the proposed method over the commonly used methods in open-sphere configurations.
Publisher: Elsevier BV
Date: 09-2012
Publisher: Acoustical Society of America (ASA)
Date: 10-2020
DOI: 10.1121/10.0002003
Abstract: The generalized leaky filtered–x least mean square (GLFxLMS) algorithm can reduce the noise lification 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.
Publisher: Elsevier BV
Date: 06-2018
Publisher: Audio Engineering Society
Date: 06-10-2015
Publisher: Elsevier BV
Date: 07-2014
Publisher: Acoustical Society of America (ASA)
Date: 08-2016
DOI: 10.1121/1.4960546
Abstract: Beamformers enable a microphone array to capture acoustic signals from a sound source with high signal to noise ratio in a noisy environment, and the linear microphone array is of particular importance, in practice, due to its simplicity and easy implementation. A linear microphone array sometimes is used near some scattering objects, which affect its beamforming performance. This paper develops a numerical model with a linear microphone array near a rigid sphere for both far-field plane wave and near-field sources. The effects of the scatterer on two typical beamformers, i.e., the delay-and-sum beamformer and the superdirective beamformer, are investigated by both simulations and experiments. It is found that the directivity factor of both beamformers improves due to the increased equivalent array aperture when the size of the array is no larger than that of the scatter. With the increase of the array size, the directivity factor tends to deteriorate at high frequencies because of the rising side-lobes. When the array size is significantly larger than that of the scatterer, the scattering has hardly any influence on the beamforming performance.
Publisher: Acoustical Society of America (ASA)
Date: 06-2018
DOI: 10.1121/1.5040139
Abstract: Previous work has demonstrated that installing secondary sources at the edge of a cavity opening can reduce sound radiation through it, but the mechanisms are not clear, which is investigated in this paper by using the modal decomposition method. It is found that a double layer edge system achieves better performance than a single layer system because secondary sources at the edge of the same layer cannot excite some modes effectively and those at different heights compensate this. There exists an upper limit frequency for the systems with boundary installed secondary sources, which is mainly decided by the length of the short side of the opening. More secondary source layers at the edge will increase the upper limit frequency.
Publisher: Acoustical Society of America (ASA)
Date: 06-2019
DOI: 10.1121/1.5112502
Abstract: Previous work has demonstrated that sound radiation through a cavity opening can be reduced with secondary sources at the edge of the opening, but the error microphones are implemented over the entire opening, which might affect the natural ventilation, lighting, and especially the access through the opening in some applications. A boundary error sensing arrangement is proposed and investigated in this paper. It is found that a double-layer error microphone arrangement achieves better performance than a single-layer one. Although its performance is not as good as the arrangement with error microphones distributed over the entire opening, it is preferable in some applications because it does not block the opening. It is also found that there exists an upper-limit frequency for the systems with error microphones installed at the edge, which is related to the size of the opening and can be increased by adding more layers of error microphones at the edge. This work demonstrates the possibility of developing an almost invisible virtual sound barrier system that can block sound transmission through an opening without affecting its functionalities.
Publisher: Elsevier BV
Date: 2020
Publisher: Acoustical Society of America (ASA)
Date: 08-2015
DOI: 10.1121/1.4927031
Abstract: 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.
Publisher: IOP Publishing
Date: 02-2019
Publisher: Acoustical Society of America (ASA)
Date: 2014
DOI: 10.1121/1.4836215
Abstract: Deep back cavities are usually required for micro-perforated panel (MPP) constructions to achieve good low frequency absorption. To overcome the problem, a close-box loudspeaker with a shunted circuit is proposed to substitute the back wall of the cavity of the MPP constructions to constitute a composite absorber. Based on the equivalent circuit model, the acoustic impedance of the shunted loudspeaker is formulated first, then a prediction model of the sound absorption of the MPP backed by shunted loudspeaker is developed by employing the mode solution of a finite size MPP coupled by an air cavity with an impendence back wall. The MPP absorbs mid to high frequency sound, and with properly adjusted electrical parameters of its shunted circuit, the shunted loudspeaker absorbs low frequency sound, so the composite absorber provides a compact solution to broadband sound control. Numerical simulations and experiments are carried out to validate the model.
Publisher: Elsevier BV
Date: 11-2019
Publisher: Acoustical Society of America (ASA)
Date: 05-2020
DOI: 10.1121/10.0001227
Abstract: This paper investigates the performance of active noise control (ANC) systems with two reflecting surfaces that are placed vertically on ground in parallel. It employs the modal expansion method and the boundary element method to calculate the noise reduction of the systems with infinitely large and finite size reflecting surfaces, respectively. Both experimental and simulation results show that the noise reduction of the system can be significantly increased after optimizing the surface separation distance and their locations with the sound sources. It is found that the sound radiation of the primary source can be completely reduced in principle if the surface interval is less than half the wavelength and the source line is perpendicular to the surfaces for infinitely large reflecting surfaces. Even with finite size ones, the noise reduction performance improvement is still significant compared with those without any reflecting surfaces. For ex le, for an ANC system with a source distance of 0.074 m, experiments achieve an improvement of 8.6 dB at 800 Hz where two 0.2 m × 0.2 m parallel reflecting surfaces are placed with a distance of 0.15 m around the system on ground. The mechanisms for the performance improvement are discussed.
Publisher: Acoustical Society of America (ASA)
Date: 02-2020
DOI: 10.1121/10.0000650
Abstract: Porous materials that are commonly used for sound absorption have poor sound insulation capability. In this paper, rigid scatterers are installed periodically inside porous materials to improve their transmission loss (TL) with the Bragg diffraction. The Delany-Bazley impedance model is used to model the porous material and the transfer matrix method is adopted to calculate the TL of the mixed structure in a duct. Simulation results with a different number of scatterers and porous materials with different airflow resistivity show that the TL of porous materials can be increased significantly with periodically arranged scatterers. The decoupled analysis reveals that the TL of the mixed structure is larger than the sum of the TL of in idual components in most frequency bands, except that around the first Bragg resonance frequency.
Publisher: Acoustical Society of America (ASA)
Date: 07-2009
DOI: 10.1121/1.3147491
Abstract: A ray-based method is presented for evaluating multiple acoustic diffraction by separate rigid and parallel wide barriers, where two or more neighboring ones are of equal height. Based on the geometrical theory of diffraction and extended from the exact boundary solution for a rigid wedge, the proposed method is able to determine the multiple diffraction along arbitrary directions or at arbitrary receiver locations around the diffracting edges, including the positions along the shadow or reflection boundaries or very close to the edges. Comparisons between the results of the numerical simulations and the boundary element method show validity of the proposed method.
Publisher: Springer Science and Business Media LLC
Date: 23-05-2013
Publisher: Elsevier BV
Date: 06-2021
Publisher: MDPI AG
Date: 03-06-2021
DOI: 10.3390/S21113866
Abstract: 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. This paper investigates the performance and challenges of using a piece of retro-reflective film directly as an acoustic membrane pick-up with an LDV to sense its vibration to form a remote acoustic sensing apparatus. Due to the special properties of the retro-reflective material, the LDV beam can be projected to the target over a wide range of incident angles. Thus, the location of the LDV relative to the pick-up is not severely restricted. This is favourable in many acoustic sensing and control applications. Theoretical analysis and systematic experiments were conducted on the membrane to characterise its performance. One design has been selected for sensing sound pressure level above 20 dB and within the 200 Hz to 4 kHz frequency range. Two ex le applications—remote speech signal sensing/recording and an active noise control headrest—are presented to demonstrate the benefits of such a remote acoustic sensing apparatus with the retro-reflective material. Particularly, a significant 22.4 dB noise reduction ranging from 300 Hz to 6 kHz has been achieved using the demonstrated active control system. These results demonstrate the potential for such a solution with several key advantages in many applications over traditional microphones, primarily due to its minimal invasiveness.
Publisher: Elsevier BV
Date: 08-2017
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 11-2008
Publisher: Acoustical Society of America (ASA)
Date: 10-2020
DOI: 10.1121/10.0002161
Abstract: The reflection of audio sounds generated by a parametric array loudspeaker (PAL) is investigated in this paper. The image source method and the non-paraxial PAL radiation model under the quasilinear approximation are used to calculate the reflected audio sound from an infinitely large surface with an arbitrary incident angle. The effects of the surface absorption in the ultrasound frequency range are studied, and the simulation and experiment results show that the reflection behavior of audio sounds generated by a PAL is different from those generated by traditional audio sources. The reason is that the reflected sound generated by the PAL consists of the reflection of audio sounds generated by incident ultrasounds and the audio sounds generated by the reflected ultrasound, and it is the latter that determines the directivity of the reflected audio sound.
Publisher: Elsevier BV
Date: 05-2020
Publisher: Elsevier BV
Date: 11-2014
Publisher: Acoustical Society of America (ASA)
Date: 03-2022
DOI: 10.1121/10.0009750
Abstract: 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.
Publisher: Elsevier BV
Date: 04-2004
Publisher: Springer International Publishing
Date: 2021
Publisher: Acoustical Society of America (ASA)
Date: 07-2011
DOI: 10.1121/1.3596457
Abstract: The feasibility of applying active noise control techniques to attenuate low frequency noise transmission through a natural ventilation window into a room is investigated analytically and experimentally. The window system is constructed by staggering the opening sashes of a spaced double glazing window to allow ventilation and natural light. An analytical model based on the modal expansion method is developed to calculate the low frequency sound field inside the window and the room and to be used in the active noise control simulations. The effectiveness of the proposed analytical model is validated by using the finite element method. The performance of the active control system for a window with different source and receiver configurations are compared, and it is found that the numerical and experimental results are in good agreement and the best result is achieved when the secondary sources are placed in the center at the bottom of the staggered window. The extra attenuation at the observation points in the optimized window system is almost equivalent to the noise reduction at the error sensor and the frequency range of effective control is up to 390 Hz in the case of a single channel active noise control system.
Publisher: Elsevier BV
Date: 11-2016
Publisher: Springer International Publishing
Date: 2021
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 03-2013
Publisher: Elsevier BV
Date: 11-2019
Publisher: Elsevier BV
Date: 02-2006
Publisher: Elsevier BV
Date: 06-2015
Publisher: Elsevier BV
Date: 07-2012
Publisher: Elsevier BV
Date: 04-1995
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 2022
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 11-2008
Publisher: Elsevier BV
Date: 03-2001
Publisher: Acoustical Society of America (ASA)
Date: 06-2018
DOI: 10.1121/1.5040480
Abstract: 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. Results show that the parameter outperforms the clarity index in most of the tested conditions, but its performance is less robust than parameter for clarity (PCLA).
Publisher: Elsevier BV
Date: 04-2016
Publisher: Elsevier BV
Date: 2013
Publisher: Audio Engineering Society
Date: 19-03-2018
Publisher: Elsevier BV
Date: 2013
Publisher: Acoustical Society of America (ASA)
Date: 02-2022
DOI: 10.1121/10.0009587
Abstract: 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.
Publisher: Informa UK Limited
Date: 04-05-2018
Publisher: IEEE
Date: 09-2006
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 2008
Publisher: Acoustical Society of America (ASA)
Date: 12-2016
DOI: 10.1121/1.4968881
Abstract: 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.
Publisher: Acoustical Society of America (ASA)
Date: 2018
DOI: 10.1121/1.5021335
Abstract: 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 & D0/ξ & 4), regardless of the wind speed and windscreen diameter. The spatial coherence between the wind noise outside and inside a porous microphone windscreen is compared with that without the windscreen, and the coherence is found to decrease significantly when the windscreen diameter is approximately 2 to 4 times the turbulence wavelengths, corresponding to the most effective wind noise reduction frequency range of the windscreen. Experimental results with a fan are presented to support the simulations. It is concluded that the wind noise reduction mechanism of porous microphone windscreens is related to the spatial decorrelation effect on the wind noise signals provided by the porous material and structure.
Publisher: IEEE
Date: 09-2016
Publisher: Springer Science and Business Media LLC
Date: 17-10-2017
DOI: 10.1038/S41598-017-13546-2
Abstract: We propose a virtual sound barrier system that blocks sound transmission through openings without affecting access, light and air circulation. The proposed system applies active control technique to cancel sound transmission with a double layered loudspeaker array at the edge of the opening. Unlike traditional transparent glass windows, recently invented double-glazed ventilation windows and planar active sound barriers or any other metamaterials designed to reduce sound transmission, secondary loudspeakers are put only along the boundaries of the opening, which provides the possibility to make it invisible. Simulation and experimental results demonstrate its feasibility for broadband sound control, especially for low frequency sound which is usually hard to attenuate with existing methods.
Publisher: Elsevier BV
Date: 2007
Publisher: Elsevier BV
Date: 02-2019
Publisher: Elsevier BV
Date: 04-2009
Publisher: Elsevier BV
Date: 2017
Publisher: Springer Science and Business Media LLC
Date: 15-10-2014
DOI: 10.1038/SREP06628
Publisher: Elsevier BV
Date: 10-2007
Publisher: Acoustical Society of America (ASA)
Date: 03-2020
DOI: 10.1121/10.0000793
Abstract: It has been reported that audible sounds can be heard behind a parametric array loudspeaker in free field, which cannot be predicted by existing models. A non-paraxial model is developed in this paper for the finite size and disk-shaped parametric source based on quasilinear approximation and disk scattering theory. The sounds on both front and back sides are calculated numerically and compared with the existing non-paraxial model for the parametric source installed in an infinitely large baffle. Both simulation and experiment results show that audible sound exists on the back side. The mechanism of the phenomenon is explored.
Publisher: Acoustical Society of America (ASA)
Date: 06-2020
DOI: 10.1121/10.0001401
Abstract: Due to their low computational complexity, reduced wiring cost, and flexibility of scaling up, decentralized multiple channel active control systems are attractive in many applications. In a decentralized multiple channel active control system, a number of small subsystems are constructed, which are updated independently with only the associated error signals. In this letter, a time domain two channel decentralized control algorithm is proposed to achieve the similar noise reduction performance as the centralized one. Auxiliary filters are introduced to filter the reference signal for control filter update and a unique design method is proposed to shape the frequency response of the auxiliary filters. The simulation results using the measured impulse responses demonstrate the efficacy of the proposed algorithm for broadband noise control.
Publisher: Acoustical Society of America (ASA)
Date: 08-2018
DOI: 10.1121/1.5049145
Abstract: Near-field error sensing is beneficial to the compactness and stability of an active noise control system. This paper proposes an error sensing strategy based on the spatial Fourier transform to achieve active directivity control of radiated sound. The error microphone array is located on a plane close to the primary source and the cost function is the weighted sum of the error signals from the microphones. The weighting factor is related to the phase shift from the error microphones to the plane perpendicular to the direction where noise reduction is required. The geometric configurations of the error microphone array for effective directivity control are investigated. It is found that the distance between neighboring error microphones must be less than approximately half the wavelength of the frequency of interest and the equivalent size of the microphone array should be larger than twice the size of the primary source. Numerical simulations and experiments demonstrate the feasibility of the proposed strategy.
Publisher: IEEE
Date: 05-2008
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 15-05-2019
Publisher: Acoustical Society of America (ASA)
Date: 02-2017
DOI: 10.1121/1.4976095
Abstract: For micro-speakers in a closed box, commonly used nonlinear compensation methods only compensate the distortion caused by the force factor and the stiffness. In this letter, a method to compensate the distortion with consideration of the nonlinear mechanical resistance is proposed based on the feedback linearization criterion. The proposed method is further improved by minimizing the variation of the output power spectrum after compensation. The simulations and experiments show that the total harmonic distortion and the intermodulation distortion of the sound pressure can be reduced significantly with little influence on the sound pressure level.
Publisher: Acoustical Society of America (ASA)
Date: 12-2019
DOI: 10.1121/1.5134062
Abstract: This paper investigates the feasibility of increasing the noise reduction performance of active noise control (ANC) systems on ground by introducing two vertical reflecting surfaces with an included angle. By using the image source method, the theory of sound wave propagation in a wedge-shaped reflector and the integral equation method, the noise reduction of the ANC systems with two infinitely large or finite size reflecting surfaces with different included angles are studied. It is demonstrated that the noise reduction of the system can be increased significantly with two reflecting surfaces after optimizing their included angle and size. The simple empirical formulas for the optimal included angle of the surfaces and the noise reduction are presented. It is found that the noise reduction at 500 Hz increases by 13.6 dB when two vertical reflecting surfaces are arranged with an optimal angle of 125° and the source distance is 0.1 m. By optimizing the size of the reflecting surfaces to about 0.35 of the wavelength, the noise reduction can be further increased by approximately 2.8 dB. The mechanisms for the performance improvement are disclosed, and the experiments are conducted to validate the results.
Publisher: Elsevier BV
Date: 06-2014
Publisher: Acoustical Society of America (ASA)
Date: 12-1999
DOI: 10.1121/1.428193
Abstract: When designing an active control system to globally control the far-field sound radiation from a vibrating surface, a challenging problem is to properly define the near-field acoustic sensing strategy and the type of cost function to be minimized by the controller. The strategy of sensing and minimizing the near-field active intensity at discrete locations in the active control of free field radiation from a vibrating plate is investigated in this paper. The optimal minimization of the sum of the near-field, normal active sound intensities at the error sensor locations using acoustic control sources is derived for this problem, and the results obtained are compared to the minimization of the sum of the near-field squared pressures. Some of the difficulties associated with sound intensity minimization are pointed out.
Publisher: Elsevier BV
Date: 05-2002
Publisher: Elsevier BV
Date: 08-2015
Publisher: ASA
Date: 2013
DOI: 10.1121/1.4800209
Publisher: Elsevier BV
Date: 05-2000
Publisher: Elsevier BV
Date: 2006
Publisher: Elsevier BV
Date: 07-2012
Publisher: Elsevier BV
Date: 12-2015
Publisher: Elsevier BV
Date: 03-2019
Publisher: Institute of Electronics, Information and Communications Engineers (IEICE)
Date: 2009
Publisher: Elsevier BV
Date: 06-2010
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 07-2016
Publisher: IEEE
Date: 10-2011
Publisher: Elsevier BV
Date: 12-2018
Publisher: Elsevier BV
Date: 08-1998
Publisher: MDPI AG
Date: 04-12-2018
DOI: 10.3390/APP8122484
Abstract: It has been demonstrated that a single shunted loudspeaker can be used as an effective low frequency sound absorber in a duct, but many shunted loudspeakers have to be used in practice for noise reduction or reverberation control in rooms, thus it is necessary to understand the performance of an array of shunted loudspeakers. In this paper, a model for the parallel shunted loudspeaker array for multi-tone sound absorption is proposed based on a modal solution, and then the acoustic properties of a shunted loudspeaker array under normal incidence are investigated using both the modal solution and the finite element method. It was found that each shunted loudspeaker can work almost independently where each unit resonates. Based on the interaction analysis, multi-tone absorbers in low frequency can be achieved by designing multiple shunted loudspeakers with different shunt circuits respectively. The simulation and experimental results show that the normal incidence sound absorption coefficient of the designed absorber has four absorption peaks with values of 0.42, 0.58, 0.80, and 0.84 around 100 Hz, 200 Hz, 300 Hz, and 400 Hz respectively.
Publisher: Acoustical Society of America (ASA)
Date: 10-2009
DOI: 10.1121/1.3206660
Abstract: Loudspeakers in virtual sound imaging systems are usually modeled as omnidirectional monopole sources. These models are, however, only an approximation for the low frequency range. This paper presents an analytical model of crosstalk cancellation systems in a free field which takes into account the scattering and spatial characteristics of the sound sources. Based on the proposed model, the effects caused by the spatial characteristics of the sound source and its misalignments on the performance of the crosstalk cancellation system are studied numerically. It is found that although the factors such as the directivity of the sound sources and the distance between the sound sources and receiver affect the performance of the system to a certain extent, the channel separation of the crosstalk cancellation system, however, is most sensitive to the misalignment of the subtended angle of the sound sources. Therefore, if highly accurate binaural cues are required in practical applications, the type and characteristics of the playback sound sources, their locations, and orientations all should be considered carefully.
Publisher: Acoustical Society of America (ASA)
Date: 03-2022
DOI: 10.1121/10.0009765
Abstract: 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. By designing a combination of different absorbers, a proper sound absorption coefficient is achieved around the ineffective frequency. Numerical simulations show that the active sound power reduction increases by 13.5 dB at the frequency with the absorbers attached on one end of the staggered window, and the overall sound power reduction between 100 and 500 Hz increases from 25.9 to 31.2 dB. Attaching the sound absorbers elsewhere in the upstream of the secondary source, for ex le, on the side walls of the duct also works. The active sound power reduction at 435 Hz increases by 6.3 dB after attaching the absorbers in the experiments, and the noise reduction increment at the evaluation point is 13.6 dB, which agrees with simulation results and demonstrates the feasibility of the proposed sound absorbers.
Publisher: Acoustical Society of America (ASA)
Date: 06-2001
DOI: 10.1121/1.1367247
Abstract: A single input, single output active noise control system using the time-domain Filtered-X LMS algorithm with output constraint is investigated. The constraint on the output of the control filter is applied by three different methods: the leakage algorithm based on the transformation method using a penalty function the re-scaling algorithm based on the active set method and the simple practical (clipping) algorithm which just clips the output if a constraint is encountered. A comparison of the three algorithms shows that the re-scaling algorithm can usually work successfully under the constraint, while the leakage algorithm usually needs a large leakage coefficient to satisfy the constraint with a resulting performance loss. The clipping algorithm has potential problems both with the stability and convergence speed.
Publisher: Acoustical Society of America (ASA)
Date: 03-2010
DOI: 10.1121/1.3295691
Abstract: An approach for predicting the reflected sound pressure is proposed in one-dimensional sound field. For a duct ended with a rigid reflective surface, only one microphone is required to measure the total sound pressure on the surface, which is further used as the error sensing strategy in an active noise control (ANC) system to reduce the in-duct reflection. Experiments are carried out to validate the prediction method, and a broadband feedforward ANC system is implemented to suppress the impulsive reflection. It is found that the ANC system based on the reflected sound prediction is effective, and 12.2 dB attenuation of the one-dimension impulsive reflection is obtained after the active control.
Publisher: Springer International Publishing
Date: 20-12-2021
Publisher: Acoustical Society of America (ASA)
Date: 09-2015
DOI: 10.1121/1.4929933
Abstract: 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 ide 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. Compared to the MacDonald method and other existing methods, the proposed method is a rigorous solution for whole space and is also much easier to understand.
Publisher: Elsevier BV
Date: 11-2007
Publisher: Wissenschaftliche Verlagsgesellschaft mbH
Date: 03-2009
DOI: 10.3813/AAA.918160
Publisher: Elsevier BV
Date: 03-2017
Publisher: IOP Publishing
Date: 02-2018
Publisher: Institute of Noise Control Engineering (INCE)
Date: 2000
DOI: 10.3397/1.2827968
Publisher: Wiley
Date: 25-10-2013
DOI: 10.1002/ASJC.813
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 2019
Publisher: Acoustical Society of America (ASA)
Date: 03-2021
DOI: 10.1121/10.0003606
Abstract: The near and far fields of traditional loudspeakers are differentiated by whether the sound pressure litude is inversely proportional to the propagating distance. However, the audio sound field generated by a parametric array loudspeaker (PAL) is more complicated, and in this article it is proposed to be ided into three regions: near field, Westervelt far field, and inverse-law far field. In the near field, the audio sound experiences strong local effects and an efficient quasilinear solution is presented. In the Westervelt far field, local effects are negligible so that the Westervelt equation is used, and in the inverse-law far field, a simpler solution is adopted. It is found that the boundary between the near and Westervelt far fields for audio sound lies at approximately a2/λ – λ/4, where a is transducer radius and λ is ultrasonic wavelength. At large transducer radii and high ultrasonic frequencies, the boundary moves close to the PAL and can be estimated by a closed-form formula. The inverse-law holds for audio sound in the inverse-law far field and is more than 10 meters away from the PAL in most cases. With the proposed classification, it is convenient to apply appropriate prediction models to different regions.
Publisher: Acoustical Society of America (ASA)
Date: 06-2004
DOI: 10.1121/1.1736654
Abstract: Active control of the sound radiated from a piston set in a rigid sphere with a set of control point sources around is considered in this paper, where the scattering sound field of the control sound from the rigid sphere has been taken into account to minimize the total radiated sound power. Analytic results of the sound power are obtained and numerical simulations show that it is possible to reduce the radiation from a small piston set in a rigid sphere similar to the size of a human head up to a certain frequency. It is found that the introduction of the scattering object makes significant differences from the active control without scattering objects. This being the case, the scattering object makes the active noise control easier. To increase the global reduction of sound-power output, the optimal number and locations of the control sources and the optimal number and locations of error sensors are discussed. Finally, experiments with one control source and one error sensor around a head simulator have been carried out to verify the simulation results.
Publisher: SAGE Publications
Date: 03-2000
Abstract: The filtered-x LMS algorithm (FXLMS) has been successfully applied to the active control of periodic and random noise and vibration. This paper presents a modified algorithm for active control of periodic noise based on the FXLMS algorithm which uses random noise for on-line cancellation path transfer function (CPTF) estimation. In the proposed algorithm, another two short adaptive filters are introduced. One is an adaptive noise cancellation filter, which is used to improve the convergence speed of the CPTF modelling filter in the presence of very large litude primary noise by cancelling the component of the error signal that is correlated with the primary noise. The other is an adaptive estimator, which is used to re-estimate the obtained CPTF (long FIR filter estimated by random noise) with a short FIR filter by using the periodic reference signal as the input. The traditional FXLMS algorithm is then used with the shortened FIR filter to filter the reference signal, thus providing significant processing flexibility in practical situations where the primary path transfer function changes much faster than the CPTF. Simulation results demonstrate the effectiveness of the proposed algorithm.
Publisher: Elsevier BV
Date: 02-2017
Publisher: Acoustical Society of America (ASA)
Date: 06-2012
DOI: 10.1121/1.4714338
Abstract: In-plane waves in a waveguide made from a thin plate are described by a superposition of a set of orthogonal functions that satisfy the edge conditions of the waveguide. Due to the Poisson and shear effects, the displacement components of the in-plane waves along the two in-plane orthogonal coordinates are coupled and this coupling affects the propagation and spatial properties of the waveguide modes. The orthogonal functions and their associated wavenumbers represent the characteristics of the uncoupled modes of the waveguide where the above mentioned couplings are ignored. This study demonstrates that the characteristics of the waveguide modes are determined by the couplings of the uncoupled mode pairs, which become significant when the pairs satisfy the conditions of spatial coincidence. At some frequencies, certain waveguide modes can be determined by a single pair of uncoupled modes. For this case, the analytical solution for the waveguide modes exists and provides both a qualitative and quantitative interpretation of the characteristics of the waveguide mode.
Publisher: Institute of Electronics, Information and Communications Engineers (IEICE)
Date: 12-2007
Publisher: AIP Publishing
Date: 24-07-2017
DOI: 10.1063/1.4995966
Abstract: Acoustic metasurfaces manipulate waves with specially designed structures and achieve properties that natural materials cannot offer. Similar surfaces work in audio frequency range as well and lead to marvelous acoustic phenomena that can be perceived by human ears. Being intrigued by the famous Maoshan Bugle phenomenon, we investigate large scale metasurfaces consisting of periodic steps of sizes comparable to the wavelength of audio frequency in both time and space domains. We propose a theoretical method to calculate the scattered sound field and find that periodic corrugated surfaces work as spatial filters and the frequency selective character can only be observed at the same side as the incident wave. The Maoshan Bugle phenomenon can be well explained with the method. Finally, we demonstrate that the proposed method can be used to design acoustical landscapes, which transform impulsive sound into famous trumpet solos or other melodious sound.
Publisher: Elsevier BV
Date: 02-2003
Publisher: IEEE
Date: 07-2008
Publisher: Acoustical Society of America (ASA)
Date: 10-2017
DOI: 10.1121/1.5008860
Abstract: 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. Simulation results indicate that the wind noise reduction first increases and then decreases with both viscous and inertial coefficients after reaching a maximum. Experimental results conducted on five porous microphone windscreens with porosity from 20 to 60 pores per inch (PPI) show that the 40 PPI windscreen has the highest wind noise reduction performance, and this supports the simulation results. The existence of the optimal values for the viscous and inertial coefficients is explained qualitatively and it is shown that the design of the porous microphone windscreens should take into account both the turbulence suppression inside and the wake generation behind the windscreen to achieve optimal performance.
Publisher: Institute of Electronics, Information and Communications Engineers (IEICE)
Date: 07-2005
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 03-2019
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 03-2011
Publisher: Acoustical Society of America (ASA)
Date: 2003
DOI: 10.1121/1.1529665
Abstract: In some situations of active noise control, infinite impulse response (IIR) filters are more suitable than finite impulse response (FIR) filters owing to the poles in the transfer function. A number of algorithms have been derived for applying IIR filters in active noise control however, most of them use the direct form IIR filter structure, which faces the difficulties of checking stability and relatively slow convergence speed for noise composed of narrow-band components with large power disparity. To overcome these difficulties along with using the direct form IIR filters, a new adaptive algorithm is proposed in this paper, which uses and updates the lattice form adaptive IIR filter in an active noise control system. Full mathematical derivations of the proposed algorithm are presented, and the comparison between the proposed algorithm and the commonly used filtered-u LMS and filtered-v LMS algorithms shows the superiority of the proposed algorithm.
Publisher: Acoustical Society of America (ASA)
Date: 08-2019
DOI: 10.1121/1.5120260
Abstract: The bin-normalized frequency domain block least mean square (NFBLMS) algorithm is a good choice in the active noise control system due to its benefit of high convergence speed. However, it suffers from a biased steady-state solution with insufficient adaptive filter length, often unavoidable due to the influence of the secondary path. A modified frequency domain block least mean square (MFBLMS) algorithm with guaranteed optimal steady-state performance has been proposed recently. However, its convergence speed is generally lower than that of the NFBLMS algorithm. In this paper, a mixed algorithm combining the NFBLMS and MFBLMS algorithms is proposed based on the analysis of the initial convergence trajectory of the NFBLMS algorithm. An effective switching strategy is designed, enabling the MFBLMS algorithm after the NFBLMS algorithm approaches its steady state and shifting back to the NFBLMS algorithm when the change of environment is detected. The mixed algorithm has both the high convergence speed and the optimal steady-state performance, and its effectiveness is validated by simulations using measured acoustic transfer functions.
Publisher: Springer Science and Business Media LLC
Date: 10-12-2016
Publisher: Acoustical Society of America (ASA)
Date: 09-2020
DOI: 10.1121/10.0001816
Abstract: 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. A theoretical formulation is developed to describe the relationship between the sound pressure inside and outside the tube. The numerical simulation results show that the peaks in the frequency response move to lower frequencies when the tube diameter, tube length, and orifice depth increase and when the orifice diameter decreases. Experiments with a 3D-printed static pressure tube were conducted to verify the analytical results. The proposed model can be used to optimize the static pressure tube in the design stage or to correct the measurement results afterwards instead of cumbersome experimental calibration.
Publisher: Elsevier
Date: 2016
Publisher: Acoustical Society of America (ASA)
Date: 10-2019
DOI: 10.1121/1.5127179
Abstract: Noise reduction performance of a compact active sound radiation control system is significantly affected by locations of the error microphones which are required to be installed near the primary source. In this paper, near-field error sensing for multi-channel active radiation control systems in free field is investigated, and it is found that the optimal locations of error sensors for minimizing the sum of squared sound pressure are between the primary source and the secondary sources distributed uniformly on a sphere surface surrounding the primary source. Both simulation and experiment results show that the optimal locations of error microphones are independent of the type of primary source when there are sufficient secondary sources. These optimal locations remain unchanged at low frequencies and move toward secondary sources when the secondary source number increases. Therefore, for active radiation control applications in low frequency range, a compact multi-channel system can be developed by locating error microphones between the primary source and secondary sources.
Publisher: Elsevier BV
Date: 11-2018
Publisher: Acoustical Society of America (ASA)
Date: 11-2022
DOI: 10.1121/10.0015140
Abstract: 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.
Publisher: IOP Publishing
Date: 15-01-2019
Publisher: Acoustical Society of America (ASA)
Date: 05-2020
DOI: 10.1121/10.0001261
Abstract: The existing non-paraxial expression of audio sounds generated by a parametric array loudspeaker (pal) is hard to calculate due to the fivefold integral in it. A rigorous solution of the Westervelt equation under the quasilinear approximation is developed in this paper for circular PALs by using the spherical harmonics expansion, which simplifies the expression into a series of threefold summations with uncoupled angular and radial components. The angular component is determined by Legendre polynomials and the radial one is an integral involving spherical Bessel functions, which converge rapidly. Compared to the direct integration over the whole space, the spherical expansion is rigorous, exact, and can be calculated efficiently. The simulations show the proposed expression can obtain the same accurate results with a speed of at least 15 times faster than the existing one.
Publisher: Institute of Electrical and Electronics Engineers (IEEE)
Date: 05-2011
Publisher: Elsevier BV
Date: 10-2007
Publisher: Acoustical Society of America (ASA)
Date: 10-2017
DOI: 10.1121/1.5005508
Abstract: 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 (TN), loudness-based EDT [EDTN Lee, Cabrera, and Martens, J. Acoust. Soc. Am. 131, 1194–1205 (2012a)], and parameter for reverberance [PREV van Dorp Schuitman, de Vries, and Lindau., J. Acoust. Soc. Am. 133, 1572–1585 (2013)]. For the comparisons, a wide range of sound pressure levels (SPLs) from 20 dB to 100 dB and RTs from 0.5 s to 5.0 s were evaluated, and two sets of subjective data from the previous studies were used for the cross-validation and comparison. Results of the comparisons show that the psychoacoustic reverberance parameters provided better matches to reverberance than RT and EDT however, the performance of these psychoacoustic reverberance parameters varied with the SPL range, the type of audio s le, and the reverberation conditions. This study reveals that PREV is the most relevant for estimating a relative change in reverberance between s les when the SPL range is small, while EDTN is useful in estimating the absolute reverberance. This study also suggests the use of PREV and EDTN for speech and music s les, respectively.
Publisher: Elsevier BV
Date: 08-2006
Publisher: Acoustical Society of America (ASA)
Date: 08-2011
DOI: 10.1121/1.3608124
Abstract: A coherent image source method is presented for evaluating single frequency sound propagation from a point source in a flat waveguide with two infinite and parallel locally reactive boundaries. The method starts from formulating reflections of the spherical sound radiation into integrals of plane wave expansion, and the analytical evaluation of the integrals is simplified by introducing a physically plausible assumption that wave front shapes remain the same before and after each reflection on a reflective boundary. The proposed model can determine coherently the sound fields at arbitrary receiver locations in a flat waveguide, even when one boundary is highly sound absorptive. Being compared with the classical wave theory and the existing coherent ray-based methods, it is shown that the proposed method provides considerable accuracy and advantages to predict sound propagation in flat waveguides with a sound absorptive ceiling and a reflective floor over a broad frequency range, particularly at large distances from the source where the existing methods are problematic.
Publisher: Elsevier BV
Date: 2015
Location: United States of America
No related grants have been discovered for Xiaojun Qiu.