Sound propagation in closed test section wind tunnels suffers from reflections and diffraction, which compromise acoustic measurements. In this article, it is proved possible to improve the post-processing of phased-array microphone measurements by using an approach based on the combination of numerical acoustic simulations and beamforming. A Finite Element Method solver for the Helmholtz equation is used to model the acoustic response of the experimental facility. The simulations are compared with acoustic experiments performed at TU Delft's Low Turbulence Tunnel, using both fully reflective (baseline) and lined test sections. The solver accurately predicts the acoustic propagation from a monopole sound source at the centre of the test section to the microphones in the phased-array, for frequencies in the range 500Hz<f<2000Hz. It is shown that a (lower fidelity) geometric modelling method is unable to precisely predict the acoustic response of the Low Turbulence Tunnel at these frequencies, due to strong acoustic diffraction. The numerical results are used to implement corrections to the post-processing of experimental data. A corrected version of the Source Power Integration method is able to increase the accuracy of the source's noise levels calculation, based on a single numerical simulation with the source at the same location as in the experiment. A Green's function correction increases the beamforming resolution and the source's noise levels estimation accuracy from the beamforming maps, without a priori knowledge of the source's location. Both corrections perform well at processing flow-on acoustic measurements, and the Green's function correction shows an additional benefit. The improvement in beamforming spatial resolution leads to an increase of the signal to noise ratio.

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Since 2020, all commercial aircraft have been mandated to be equipped with ADS-B Out
transponders. Despite the many advantages of locating an aircraft with openly available and
accessible data, it also has some limitations. Firstly, not all aircraft, such as general aviation, are required to transmit their locations; secondly, due to obstacles such as buildings,
a location is not always transmitted at lower altitudes (75-130 m); thirdly, it is vulnerable to cyberattacks. Therefore, while it is convenient to have ADS-B Out data, creating a
computationally efficient alternative methodology for determining the aircraft location is
advisable. This paper investigates the accuracy, efficiency, and computational cost of two
methods of source localization using data taken by an array of microphones: a global optimization (GO) method called the differential evolution (DE) and the conventional beam-
forming approach (CBF). The real-world data required as input for both methods is obtained
with a 64-microphone phased array placed at a distance of 1.14 km from Rotterdam The
Hague Airport (RTHA). The 2-dimensional flight trajectories, i.e., azimuth, and elevation
relative to the array, obtained from the GO and CBF methods, are compared with the ADS-
B Out data for approaching and departing flyovers. Furthermore, the smallest size of an
array required for satisfactory localization accuracy is investigated.@en

The fan-stator stage of turbofan engines is one of the main regions of broadband noise generation. The trend of increasing bypass ratios will make fan-stator broadband noise even more significant, as jet noise will decrease and nacelles will become shorter, thus leaving less space for liners. Within the fan-stator stage there are several aerodynamic phenomena that can cause broadband noise. However, techniques for experimental identification and quantification (and thus classification) of these broadband noise sources are still immature. Beamforming using in-duct microphones is feasible, but a major challenge is the strong rotational component of the flow that needs to be accounted for in the steering vectors. This paper proposes a solution for that challenge, based on a fast ray tracing approach. With synthesized microphone array data, calculated with an approximate method for the Green’s function in a ducted swirling flow, it is demonstrated that detection of acoustic sources is possible.@en

Aeroacoustic tests in closed wind tunnels are affected by reflections in the tunnel circuit and background noise. Reflections can be mitigated by lining the tunnel circuit. The present study investigates if lining exclusively the most accessible segment of a closed wind tunnel circuit, in particular the test section, is an approach which improves acoustic measurements. Literature shows that a wind tunnel lining material should have high acoustic absorption, low inertial resistivity and low surface roughness. Therefore, the test section of TU Delft's closed Low Turbulence Tunnel is lined with melamine foam wall liners. A total of 4 test section configurations were tested: baseline; test section with lining on the floor and ceiling; test section with lined side--panels; and test section lined at all surfaces (floor, ceiling and side--panels). An omnidirectional speaker is used for evaluating the wind tunnel's acoustic performance. A geometric modelling algorithm, based on the mirror-source method, is used to predict the effect of lining on primary reflections in the test section. In addition, reflections in the test section and in the tunnel circuit are characterized experimentally. The results show that the closed loop of the tunnel circuit is responsible for a long reverberation time in the test section. However, reflections inside the test section itself are the dominant source of acoustic interference at the microphone array location. The low fidelity geometric modelling algorithm is shown to be a valuable approach for an initial estimation of the acoustic benefit of lining, for both flow--off and --on conditions. Lining of the test section walls significantly reduces reflections from the reference source, as well as the aerodynamic background noise that reaches the array.@en

Aeroacoustic measurements performed by flush-mounted microphone arrays on the walls of closed-section wind tunnels are contaminated by the hydrodynamic pressure fluctuations of the wall's boundary layer. This study evaluates three different microphone cavity geometries for mitigating this issue. Their improvement to the signal-to-noise ratio (SNR) and the accuracy of their acoustic imaging results are compared to a flush-mounted microphone array. The four geometries include: (1) an array of flush-mounted microphones as the baseline, (2) a cylindrical hard-plastic cavity with a countersink, (3) a conical cavity made of melamine acoustic absorbing foam, and (4) a conical cavity with star-shaped protrusions, also made of melamine. The three arrays with cavities were covered with a steel-wire cloth to reduce the boundary layer fluctuations at the microphone while the baseline array was uncovered. Two sound sources were tested in an aeroacoustic wind tunnel for assessing the performance of the different cavities: a speaker placed outside the flow and a distributed sound source generated by a flat plate inside of the flow. When using conventional frequency domain beamforming, both cavities made of melamine offer up to a 30 dB increase in SNR with respect to the flush-mounted case, followed by the hard-walled cavity with up to a 20 dB increase. This is a 20 dB improvement when compared to the single microphone cases. The melamine cavities also provide cleaner acoustic source maps and accurate spectral estimations for a wider frequency range. The effect of cavity placement and geometry on the coherence, which affects the beamforming analysis of the acoustic signal was negligible for all cases. Distributed sound source measurements using the three arrays agreed with predictions using the Brooks, Pope, and Marcolini (BPM) model, showing that the cavities could detect vortex shedding that was undetectable by the flush array.

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In this paper, the performance of four acoustic imaging methods: conventional frequency domain beamforming (CFDBF), functional beamforming (FUNBF), enhanced high resolution CLEAN–SC (EHR–CLEAN–SC) and generalized inverse beamforming (GIBF), is investigated in terms of accuracy and variability. Three experimental test cases are considered: 1) a single speaker emitting synthetic broadband noise, 2) two speakers emitting incoherent synthetic broadband noise, and 3) trailing–edge noise generated by a tripped NACA 0018 airfoil. All the measurements were performed in the anechoic wind tunnel of Delft University of Technology. Overall, GIBF and EHR–CLEAN–SC offer the most accurate results when point sources (speakers) are present. They even achieve super–resolution by separating sound sources beyond the Rayleigh resolution limit. Repeating the measurements indicates a standard deviation in the results of less than 1 dB. When analyzing distributed sound sources, such as trailing–edge noise, CFDBF and FUNBF provide the best performance. This indicates that the acoustic imaging method needs to be selected based on the expected sound source configuration.

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In this paper an accuracy criterion is derived for acoustic array measurements that are disturbed by incoherent noise, for example turbulent boundary layer noise in a wind tunnel. The accuracy criterion is a minimum signal-to-noise ratio at which the output of the array processing method has at least 95% likelihood of being within 1 dB error. Array processing is done with source power integration methods applied to cross-spectral matrices without diagonal. The integrated results are scaled to average microphone data and can therefore be considered as denoised array data. This paper derives the accuracy criterion for broadband noise signals consisting of multiple plane waves and for a line source of zero coherence length. For both cases, the validity of the criterion is confirmed with synthesized broadband noise array data.@en

The recently introduced high-resolution (HR)-CLEAN-SC algorithm for acoustic imaging provides ‘super-resolution’, i.e. the ability to discern sound sources located closer than the Rayleigh resolution limit. This is achieved by allowing the source markers to be relocated from the actual source locations within a certain constraint to avoid the combined influence of the other sound sources. The freedom to relocate the source markers to increase the performance of the algorithm depends on the maximum sidelobe level of the acoustic array used. This paper presents an ‘enhanced’ version of the HR-CLEAN-SC algorithm which benefits from low maximum sidelobe level array design. The source marker constraint μ is adapted to the maximum sidelobe level at each frequency. Application to up to four synthetic sound sources shows that the sources can be resolved at half the frequency associated with the Rayleigh resolution limit, when an acoustic array optimized for low maximum sidelobe level is used in combination with Enhanced HR-CLEAN-SC. This improves source discrimination compared to when the HR-CLEAN-SC algorithm is used with a benchmark acoustic array design. The results are confirmed by experimental validation in which up to four loudspeakers and the same array configurations as in the synthesized data case are used.

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Phased microphone arrays have become a well-established tool for performing aeroacoustic measurements in wind tunnels (both open-jet and closed-section), flying aircraft, and engine test beds. This paper provides a review of the most well-known and state-of-the-art acoustic imaging methods and recommendations on when to use them. Several exemplary results showing the performance of most methods in aeroacoustic applications are included. This manuscript provides a general introduction to aeroacoustic measurements for non-experienced microphone-array users as well as a broad overview for general aeroacoustic experts.

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A mathematical model is proposed that evaluates acoustic propagation of turbulent boundary layer waves on top of a cavity enclosing a microphone. The goal is to optimize these cavity geometries to improve the signal-to-noise ratio of acoustic measurements. This model predicts the attenuation of the turbulent boundary layer fluctuations propagating within the cavity for a given wind tunnel speed, porous surface resistance and cavity geometry. Duct acoustics predict that the spatial wave numbers of the turbulent boundary layer pressure fluctuations are shorter than the acoustic wavelength and are therefore evanescent in the cavity. These cavities are also covered with a porous material such as a metallic mesh or kevlar, to attenuate hydrodynamic fluctuations through the cavity. This model supports the investigation of 3D cylindrical cavities and incorporates both soft and hard cavity walls. Good agreement was found with experimental data. @en

Most acoustic imaging methods assume the presence of point sound sources and, hence, may fail to correctly estimate the sound emissions of distributed sound sources, such as trailing-edge noise. In this contribution, three integration techniques are suggested to overcome this issue based on models considering a single point source, a line source, and several line sources, respectively. Two simulated benchmark cases featuring distributed sound sources are employed to compare the performance of these integration techniques with respect to other well-known acoustic imaging methods. The considered integration methods provide the best performance in retrieving the source levels and require short computation times. In addition, the negative effects of the presence of unwanted noise sources, such as corner sources in wind-tunnel measurements, can be eliminated. A sensitivity analysis shows that the integration technique based on a line source is robust with respect to the choice of the integration area (shape, position, and mesh fineness). This technique is applied to a trailing-edge-noise experiment in an open-jet wind tunnel featuring a NACA 0018 airfoil. The location and far-field noise emissions of the trailing-edge line source were calculated.

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This paper proposes the enhancement of a previously developed method for beamforming in engine ducts, which enables phased array beamforming in the interstage section of a turbofan engine. The enhanced method features the use of tailored Green’s functions, to deal with the annular geometry and the non-uniform (swirling) flow in this part of the engine duct. A complicating factor is the occurrence of resonant acoustic modes, when there is no acoustic lining. The paper describes the computation of the tailored Green’s functions and how resonant modes can be subtracted. The use of the method is firstly demonstrated by application to simulated sources, and is subsequently applied to test data. At the moment there is no other information on the sources that could be used to validate the method. The only indication that the method is working well is that it should find the correct number of rotating fan blades in the beamforming plots. This is indeed the case, but only in a small fraction of the results.

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Unequally spaced transducer rings make it possible to extend the range of detectable azimuthal modes. The disadvantage is that the response of the mode detection algorithm to a single mode is distributed over all detectable modes, similarly to the Point Spread Function of Conventional Beamforming with microphone arrays. With multiple modes the response patterns interfere, leading to a relatively high “noise floor” of spurious modes in the detected mode spectrum, in other words, to a low dynamic range. In this paper a deconvolution strategy is proposed for increasing this dynamic range. It starts with separating the measured sound into shaft tones and broadband noise. For broadband noise modes, a standard Non-Negative Least Squares solver appeared to be a perfect deconvolution tool. For shaft tones a Matching Pursuit approach is proposed, taking advantage of the sparsity of dominant modes. The deconvolution methods were applied to mode detection measurements in a fan rig. An increase in dynamic range of typically 10–15 dB was found.

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Beamforming performance can be improved in two ways: optimizing the location of microphones on the acoustic array and applying advanced beamforming algorithms. In this study, the effects of the two approaches are studied. An optimization method is developed to optimize the location of microphones for an acoustic array used in an open-jet anechoic wind tunnel. Then the benefits of using the optimized array with the recently-developed advanced beamforming algorithm, the High-Resolution (HR) CLEAN-SC algorithm are investigated. The microphone locations were optimized to obtain both good resolution and low side lobe levels. By using the optimized array and applying the HR CLEAN-SC algorithm, it was found that two closely-spaced sound sources can be resolved in a broad frequency range below the Rayleigh limit. The findings have also been confirmed through experimental validation.@en

This paper proposes a fast method for calculating the acoustic time delay between an observer and a receiver in a shear flow. This method is applied to an outdoor microphone array measurement on a large-scale wind turbine. In such a set-up, a shear flow represents the actual wind field better than a uniform flow. Steering vectors for beamforming can be obtained by calculating the time delay between each point on a scan grid and each microphone. It is argued that omission in the steering vectors of the decay due to spherical spreading is preferable. Beamforming images show the benefits of ignoring the spherical spreading and of using a shear flow model. It is demonstrated that a shear flow model can also be used in combination with rotating source beamforming, again leading to beamforming improvements.@en

Most acoustic imaging methods assume the presence of point sound sources and, hence, fail to correctly estimate the sound emissions of distributed sound sources (such as trailing– edge noise). In this contribution, three integration techniques are suggested to overcome this issue based on models considering a single point source, a line source and several line sources, respectively. Two simulated benchmark cases featuring distributed sound sources are used to compare the performance of these integration techniques with respect to other well–known methods. The considered integration methods provide the best performance in retrieving the source levels and require short computational times. In addition, the presence of unwanted noise sources, such as corner sources in wind–tunnel measurements, no longer affects the results negatively when using the last method. A sensitivity analysis shows that the integration technique based on a line source is robust with respect to the choice of the integration area (shape, position and mesh fineness). Practical recommendations are provided for the application of these methods to experimental cases. @en

In this article, a high-resolution extension of CLEAN-SC is proposed: high-resolution-CLEAN-SC. Where CLEAN-SC uses peak sources in ‘dirty maps’ to define so-called source components, high-resolution-CLEAN-SC takes advantage of the fact that source components can likewise be derived from points at some distance from the peak, as long as these ‘source markers’ are on the main lobe of the point spread function. This is very useful when sources are closely spaced together, such that their point spread functions interfere. Then, alternative markers can be sought in which the relative influence by point spread functions of other source locations is minimised. For those markers, the source components agree better with the actual sources, which allows for better estimation of their locations and strengths. This article outlines the theory needed to understand this approach and discusses applications to 2D and 3D microphone array simulations with closely spaced sources. An experimental validation was performed with two closely spaced loudspeakers in an anechoic chamber.

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Unequally spaced transducer rings make it possible to extend the range of detectable azimuthal modes. The other side of the coin is that the response of the mode detection algorithm to a single mode is distributed over all detectable modes, similarly to the Point Spread Function of Conventional Beamforming with microphone arrays. With multiple modes the response patterns interfere, leading to a relatively high “noise floor” of spurious modes in the detected mode spectrum, in other words, to a low dynamic range. In this paper a deconvolution strategy is proposed for increasing this dynamic range. It starts with separating the measured sound into shaft tones and broadband noise. For broadband noise modes, a standard Non-Negative Least Squares solver appeared to be a perfect deconvolution tool. For shaft tones a Matching Pursuit approach is proposed, taking advantage of the sparsity of dominant modes. The deconvolution methods were applied to mode detection measurements in an AneCom fan rig. An increase in dynamic range of typically 10 to 15 dB was found.

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