Controlling the effective surface mass density of membrane-type acoustic metamaterials using dynamic actuators.

Membrane-type acoustic metamaterials (MAM) are thin and lightweight structures that offer exceptional low-frequency sound transmission loss (STL) values, which can exceed the corresponding mass-law significantly. Typically, the high STL of MAM is confined to a narrow frequency band, which is associated with the so-called anti-resonance. This narrow bandwidth reduces the range of potential noise control applications for MAM. To potentially overcome this challenge, this paper presents an investigation into actively controlling the effective surface mass density of MAM by actuating the MAM with a force that is correlated to the acoustic pressure difference acting on the MAM. In particular, it is shown using theoretical and numerical methods that the anti-resonance frequency of MAM can be adjusted over a wide frequency range by passing the incident sound pressure through an adjustable gain. A simple analytical model to predict the frequency shifting, depending on the gain value, is derived. A realization of this concept is further studied, consisting of a circular MAM with a small electrodynamic actuator (to apply a force to the MAM) and a microphone in front of the MAM (to estimate the pressure difference). Finally, experimental results from impedance tube measurements are used to validate the proposed analytical model.

Realisation of nonreciprocal transmission and absorption using wave-based active noise control.

Nonreciprocal acoustic devices typically break reciprocity by introducing nonlinearities or directional biasing. However, these devices are generally not fully adaptable and often use resonant cavities, which only exhibit nonreciprocal behaviour over a narrow bandwidth. Therefore, to overcome these challenges, this paper investigates how wave-based active control can be used to achieve broadband nonreciprocal behaviour in a one-dimensional environment. Wave-based controller architectures are described for both transmission and absorption control and, through simulation and experimental implementations, it is shown that they can achieve broadband nonreciprocal behaviour. Importantly, the direction of nonreciprocal behaviour can be straightforwardly reversed.

Weighted pressure matching with windowed targets for personal sound zones.

Personal sound zones (PSZ) systems use an array of loudspeakers to render independent audio signals to multiple listeners within a room. The performance of a PSZ system, designed using weighted pressure matching, depends on the selected target responses for the bright zone. In reverberant environments, the target responses are generally chosen to be the room impulse responses from one of the loudspeakers to the control points in the selected bright zone. This approach synthesizes the direct propagation component and all the reverberant components in the bright zone, while minimizing the energy in the dark zone. We present a theoretical analysis to show that high energy differences cannot be achieved for the diffuse reverberant components in the bright and dark zones, and so trying to synthesize these components in the bright zone does not lead to the best performance. It is then shown that the performance can be improved by using windowed versions of these measured impulse responses as target signals, in order to control which reverberant components are synthesized in the bright zone and which are not. This observation is supported by experimental measurements in two scenarios with different levels of reverberation.

Superposition of the uncertainties in acoustic responses and the robust design of active control systems.

The use of virtual sensing allows the frequency range of a local active noise control system located close to a listener's ears to be extended beyond what is possible when only controlling from remote physical sensors, particularly if head tracking is also used to determine the position of the virtual sensors. As the frequency range is extended, however, the uncertainties in the acoustic responses become more significant, and the design of multichannel adaptive controllers that are robustly stable to these uncertainties becomes more important. In order to fully characterise the uncertainties, a very large number of measurements would, in principle, need to be taken, due to the combination of all the possible changes in the acoustic environment. For uncertainties due to the simultaneous change in position of several objects within the acoustic environment, however, it is shown that the uncertainties can be accurately predicted by the superposition of these uncertainties, due to the change in position of the objects individually. This allows the uncertainty that is due to the change in position of a number of objects to be rapidly evaluated from a limited number of experiments and considerably simplifies the controller design process, which is illustrated here for an active headrest system using two different virtual sensing techniques.

Investigation of a directional warning sound system for electric vehicles based on structural vibration.

Warning sound systems for electric vehicles with advanced beamforming capabilities have been investigated in the past. Despite showing promising performance, such technologies have yet to be adopted by the industry, as implementation costs are generally too high and the components too fragile for implementation. A lower cost solution with higher durability could be achieved by using an array of inertial actuators instead of loudspeakers. These actuators can be attached directly to the body of the vehicle and thus require minimal design modifications. A directional sound field can then be radiated by controlling the vibration of the panel via adjustments to the relative magnitude and phase of the signals driving the array. This paper presents an experimental investigation of an inertial actuator-based warning sound system. A vehicle placed in a semi-anechoic environment is used to investigate different array configurations in terms of the resulting sound field directivity and the leakage of sound into the cabin. Results indicate that the most efficient configuration investigated has the actuators attached to the front bumper of the vehicle. Using this arrangement, real-time measurements for different beamformer settings are performed to obtain a thorough picture of the performance of the system across frequency and steering angle.

Design and evaluation of personal audio systems based on speech privacy constraints.

Personal audio refers to the generation of spatially distinct sound zones that allow individuals within a shared space to listen to their own audio material without affecting, or being affected, by others. Recent interest in such systems has focussed on their performance in public spaces where speech privacy is desirable. To achieve this goal, speech is focussed towards the target listener and a masking signal is focussed into the area where the target speech signal could otherwise be overheard. An effective masking signal must substantially reduce the intelligibility in this region without becoming an annoyance to those nearby. To assess these perceptual requirements, listening tests were carried out using two examples of loudspeaker arrays with different spatial aliasing characteristics, to determine the impacts of different masking signal spectra on speech intelligibility and subjective preference. The results of these tests were used, alongside objective and subjective metrics, to form a design specification for private personal audio systems.

Active control of the sound power scattered by a locally-reacting sphere.

Active control of the sound power scattered by a sphere is theoretically investigated using spherical harmonic expansions of the primary and secondary fields. The sphere has a surface impedance that is uniform, real, and locally reacting while being subjected to an incident monochromatic plane wave. The scattered power is controlled with a number of monopole sources, initially on the surface of the sphere, and is expressed as the sum of squared amplitudes of the spherical harmonics due to both the scattered and control fields. This quadratic function is minimized to identify the optimal strengths for different numbers of control sources. At low frequencies, the scattered field is dominated by the first few spherical harmonic terms, and its power can be significantly reduced with a single controlling monopole for a soft or absorbent sphere and with two monopoles for a hard sphere. The number of secondary sources required to significantly attenuate the scattered field at higher frequencies is found to be proportional to the square of the frequency, and the attenuation also falls off rapidly if the secondary sources are moved away from the surface of the sphere, no matter what its surface impedance.

Active structural acoustic control using an experimentally identified radiation resistance matrix.

Active structural acoustic control (ASAC) is a widely used active noise control technique that provides control of structurally radiated noise through actuation of the radiating structure. Typically, ASAC drives structural actuators to minimise a real-time measurement of the radiated sound field. However, it is often not practical to position error microphones in the radiated sound field. To overcome this limitation, a number of methods have previously been proposed. One such method utilises the radiation resistance matrix to map structural response measurements to the acoustic response and, thus, enable an estimate of the structurally radiated sound power from structural measurements alone. This has previously relied upon precise modelling of the radiating structure which, for practical structures, can lead to limitations in the accuracy of the estimate. In this paper, an ASAC strategy that utilises an experimentally identified radiation resistance matrix is presented. The robustness of both the sound power estimation and the ASAC controller to system uncertainties is investigated, and it has been shown that the proposed ASAC strategy is able to achieve effective control of the radiated sound power.

A system for controlling the directivity of sound radiated from a structure.

Directional sound fields can be generated by arrays of multiple sound sources such as loudspeaker drivers. These systems, though potentially capable of high levels of directivity control over a broad bandwidth, may prove prohibitively expensive, fragile, or impracticable in certain applications. To overcome these limitations, this paper presents an investigation into the design and limitations of a directional structural-actuator-based array. This provides an affordable and robust alternative to conventional loudspeakers, particularly when the actuators can be used to radiate via a pre-existing structure and where the required audio quality is lower, or the bandwidth somewhat limited. In the first instance, an analytical model is formulated and used to perform a simulation-based parametric study, which provides insights into the design trade-offs. Based on this study, a physical prototype is constructed using six actuators and a flat panel, which enables the model to be experimentally validated and an evaluation of the directional radiation capabilities of the proposed system to be carried out. Experiments show that a simple analytical model is an effective tool in designing such arrays, predicting the trends in the behaviour of the prototype and that the structural actuator-based system is capable of controlling directivity within its intended operational bandwidth.

Experimental identification of the radiation resistance matrix.

The radiation resistance matrix allows for the calculation of structurally radiated sound power using a series of measured structural responses. Currently, estimating the radiation resistance matrix requires precise modelling of the structure which, for practical structures, can lead to estimation errors. This paper presents two methods for identifying the radiation resistance matrix for a structure using measurable structural and acoustic responses and the solution of an inverse problem. Although well suited to practical, complex structures, to allow the accuracy of the proposed methods of identifying the radiation resistance matrix to be reliably validated, they are compared with the theoretical radiation resistance matrix for a flat plate in an infinite baffle. It is shown through a simulation-based study that the accuracy of the proposed identification methods depends on the number of structural and acoustic sensors and structural forces used in the identification process. The proposed identification methods are then implemented experimentally to identify the radiation resistance matrix for a flat plate. The results demonstrate that an accurate estimate of the sound power can be obtained using the experimentally identified radiation resistance matrix using the two proposed methods, and the limits on the two methods are discussed.

The application of a multi-reference control strategy to noise cancelling headphones.

Active noise cancelling (ANC) headphones have seen significant commercial success and a number of control strategies have been proposed, including feedforward, feedback, and hybrid configurations, using both analogue and digital implementations. Irrespective of the configuration or implementation approach, the strategies proposed in the open-literature have focused on implementations where the control system for each ear of the headphones operates independently. In this paper, a multi-reference ANC strategy is proposed and investigated for noise cancelling headphones. As with standard feedforward ANC headphones, the system utilises a single error microphone and single reference microphone on each cup; however, in the proposed configuration, the left and right reference microphones are used to achieve control at both the left and right ear cups. The performance of this controller design is compared to a standard single reference feedforward controller implementation under a variety of different sound field conditions. Although the proposed strategy requires an increased computational demand, it is shown that there is a significant control advantage for noise sources originating from the side of the user, whilst the performance for front and rear sources is maintained.

Acoustic source localization with microphone arrays for remote noise monitoring in an Intensive Care Unit.

An approach is described to apply spatial filtering with microphone arrays to localize acoustic sources in an Intensive Care Unit (ICU). This is done to obtain more detailed information about disturbing noise sources in the ICU with the ultimate goal of facilitating the reduction of the overall background noise level, which could potentially improve the patients' experience and reduce the time needed for recovery. This paper gives a practical description of the system, including the audio hardware setup as well as the design choices for the microphone arrays. Additionally, the necessary signal processing steps required to produce meaningful data are explained, focusing on a novel clustering approach that enables an automatic evaluation of the spatial filtering results. This approach allows the data to be presented to the nursing staff in a way that enables them to act on the results produced by the system.

Estimation of the pressure at a listener's ears in an active headrest system using the remote microphone technique.

The remote microphone technique is considered in this paper as a way of estimating the error signals at a listener's ears in an active headrest system using remotely installed monitoring microphones. A least-squares formulation for the optimal observation filter is presented, including a regularization factor that is chosen to satisfy both the estimation accuracy and robustness to uncertainties. The accuracy of the nearfield estimation is first investigated for a diffuse field via simulations. Additionally, simulations of a free field are also used to investigate the effect of the spatial directivity of the primary field. Finally, experiments in an anechoic chamber are conducted with 24 monitoring microphones around a dummy head positioned in an active headrest system. When six loudspeakers driven by uncorrelated random disturbances are used to generate the primary field, the best arrangement of monitoring microphones is considered, taking into account both accuracy and robustness.

Application of the remote microphone method to active noise control in a mobile phone.

Mobile phones are used in a variety of situations where environmental noise may interfere with the ability of the near-end user to communicate with the far-end user. To overcome this problem, it might be possible to use active noise control technology to reduce the noise experienced by the near-end user. This paper initially demonstrates that when an active noise control system is used in a practical mobile phone configuration to minimise the noise measured by an error microphone mounted on the mobile phone, the attenuation achieved at the user's ear depends strongly on the position of the source generating the acoustic interference. To help overcome this problem, a remote microphone processing strategy is investigated that estimates the pressure at the user's ear from the pressure measured by the microphone on the mobile phone. Through an experimental implementation, it is demonstrated that this arrangement achieves a significant improvement in the attenuation measured at the ear of the user, compared to the standard active control strategy. The robustness of the active control system to changes in both the interfering sound field and the position of the mobile device relative to the ear of the user is also investigated experimentally.

Head tracking extends local active control of broadband sound to higher frequencies.

Local active sound control systems provide useful reductions in noise within a zone of quiet which only extends to about one tenth of an acoustic wavelength. If active control is required above a few hundred hertz, this generally limits the movement of a listener to unrealistically small changes in head position. We describe a local active sound control system using a fixed array of monitoring microphones, in which the pressures at the ear positions are estimated from these microphone signals using head position information from an optical head tracker. These signals are then actively controlled to give robust attenuation at the ear positions, even as the listener moves their head. Feedforward control provides selective attenuation of noise and broadband attenuation of around 20 dB is measured up to excitation frequencies of 1 kHz under favourable conditions, with head tracking achieved in a few seconds. The active control performance is thus comparable with that achieved with active headphones, but without the listener having anything attached to their head.

Combining the remote microphone technique with head-tracking for local active sound control.

This paper describes practical integration of the remote microphone technique with a head-tracking device in a local active noise control system. The formulation is first reviewed for the optimized observation filter and nearfield pressure estimation. The attenuation performance and stability of an adaptive active headrest system combined with the remote microphone technique are then studied. The accuracy of the nearfield estimation and the effect of the head-tracking on the control performance are investigated in real-time experiments. The regularization factor of the observation filter is selected as a trade-off between its accuracy and its robustness. The integrated active headrest system is used to estimate and attenuate disturbance signals at a listener's ears from a single tonal primary source, while a commercial head-tracking device detects and provides the real-time head position to the active headrest system whose responses are updated accordingly.

Active control of scattered acoustic fields: Cancellation, reproduction and cloaking.

The active control of sound fields has been widely applied in both active noise control and sound field reproduction, however, relatively few studies have focused on active acoustic cloaking. In order to build upon the knowledge and understanding in the areas of active noise control and sound field reproduction, this paper investigates the corresponding physical limitations and compares them to the active cloaking problem when the three strategies are employed in the presence of an acoustic scatterer. The three sound field control strategies have been formulated within a consistent framework, and this has enabled insight into the physical control mechanisms. Two different three-dimensional scattering problems have then been simulated and used to investigate the performance limitations of the three strategies. The influence of the number and proximity of control sources to the scattering object have been investigated, and it has been shown that the requirements for active cloaking differ from those for active noise control and sound field reproduction. Specifically, it has been shown that there is a clear distinction between controlling the internal and external sound fields in the three cases.

Modeling local active sound control with remote sensors in spatially random pressure fields.

A general formulation is presented for the optimum controller in an active system for local sound control in a spatially random primary field. The sound field in a control region is selectively attenuated using secondary sources, driven by reference sensors, all of which are potentially remote from this control region. It is shown that the optimal controller is formed of the combination of a least-squares estimation of the primary source signals from the reference signals, and a least-squares controller driven by the primary source signals themselves. The optimum controller is also calculated using the remote microphone technique, in both the frequency and the time domains. The sound field under control is assumed to be stationary and generated by an array of primary sources, whose source strengths are specified using a spectral density matrix. This can easily be used to synthesize a diffuse primary field, if the primary sources are uncorrelated and far from the control region, but can also generate primary fields dominated by contributions from a particular direction, for example, which is shown to significantly affect the shape of the resulting zone of quiet.

The effect of reverberation on personal audio devices.

Personal audio refers to the creation of a listening zone within which a person, or a group of people, hears a given sound program, without being annoyed by other sound programs being reproduced in the same space. Generally, these different sound zones are created by arrays of loudspeakers. Although these devices have the capacity to achieve different sound zones in an anechoic environment, they are ultimately used in normal rooms, which are reverberant environments. At high frequencies, reflections from the room surfaces create a diffuse pressure component which is uniform throughout the room volume and thus decreases the directional characteristics of the device. This paper shows how the reverberant performance of an array can be modeled, knowing the anechoic performance of the radiator and the acoustic characteristics of the room. A formulation is presented whose results are compared to practical measurements in reverberant environments. Due to reflections from the room surfaces, pressure variations are introduced in the transfer responses of the array. This aspect is assessed by means of simulations where random noise is added to create uncertainties, and by performing measurements in a real environment. These results show how the robustness of an array is increased when it is designed for use in a reverberant environment.

A superdirective array of phase shift sources.

A superdirective array of audio drivers is described, which is compact compared with the acoustic wavelength over some of its frequency range. In order to minimize the overall sound power output, and hence reduce the excitation of the reverberant field when used in an enclosed space, the individual drivers are made directional by using phase shift enclosures. The motivating application for the array is the enhancement of sound from a television, in a particular region of space, to aid hearing impaired listeners. The design is initially investigated, using free-field simulations, by comparing the performance of 8 monopoles, 8 phase shift loudspeakers, and a double array of 16 monopoles, with a contrast maximization formulation. The construction and testing of an array of 8 drivers is then discussed, together with its measured response in an anechoic environment. The result of using acoustic contrast maximization is then compared with a least squares formulation, which demonstrates that the performance of the least squares solution can be made similar to that given by acoustic contrast maximization in this application, with a suitable choice of the target field.