1,721,097 research outputs found

    Advances in sound field analysis and control based on cylindrical coordinates

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    This Ph.D. thesis concerns advances in acoustic transducer array technology for improved sound field analysis and control performance. Four principal investigations are presented, which address specific performance limitations of microphone arrays and loudspeaker arrays. The basic model, on which these investigations are founded, is the general solution of the Helmholtz equation in cylindrical coordinates. The individual acoustical investigations draw on the analysis of the eigenvalues and eigenfunctions of the forward operator, providing information on the robustness of the inverse solutions against non-uniqueness, ill-conditioning and spatial aliasing. A circular microphone array design based on tangentially aligned pressure gradient sensors is studied. The theoretical analysis is complemented by a simulation study, comparing the new design to conventional arrays built from pressure sensors. It is shown that the proposed design can provide an improved performance at low frequencies, while performing worse at high frequencies due to spatial aliasing. The effects of the latter can be compensated if the Direction-of-Arrival (DOA) of the incoming waves is known. A novel DOA estimation method for sound fields measured with circular microphone arrays is proposed to address this. Using analytical expressions to model the sound fields of point sources and plane waves, it is studied for which sound fields the method is applicable and how robust it is against model imperfections. The estimation accuracy for different numbers of sources and different levels of background noise is investigated in a simulation study and the method is tested against real data, obtained through acoustic measurements. The estimation results achieved in simulations and with experimental data compare well. The general solution to the Helmholtz equation is then applied as a model for acoustic radiation in wedge-shaped spaces. This investigation aims to improve the performance of loudspeaker arrays in restricted propagation spaces, e.g. rooms. By introducing boundary conditions to the general model, different sets of basis functions are implemented in the solution and it is shown that the model enables Nearfield Acoustical Holography (NAH). Using the same propagation model, a technique for sound field control with arrays in wedge spaces is developed. The inverse problem is solved by means of a mode-matching approach, leading to an expression for the driving signals ased on a target beam pattern. Both simulations and experiments with a hemi-cylindrical loudspeaker array prototype confirm the applicability of the model for both NAH and beamforming with loudspeaker arrays in wedge spaces. Different beam patterns are considered and the model is tested through simulations and experiments. The implications of the findings, how they are linked and what future developments they may lead to is discussed

    3D audio in rooms with compact wedge arrays

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    Three dimensional audio from compact loudspeaker arrays has been explored in different forms. A number of techniques intentionally steer beams towards the walls in a room and generate reflections that give the impression of a source at the mirrored position behind the wall. This enables the creation of a system with virtual sources around the listener position. Wedge arrays are particularly suitable for this kind of application because their beamforming technique uses a model that includes the reflections from walls, floor and ceiling of the encompassing wedge. Inside a box-shaped room, the wedge can be either a corner but also a wall, in a wider sense. A method for three-dimensional audio from one or more wedge arrays is proposed. Using beam patterns whose main lobes can be steered both horizontally and vertically, the creation of virtual sources around a listening position is achieved. It is shown that some virtual sources can be created through direct beams and beam reflections, while some can only be synthesised as phantom sources, depending on the respective position of the arrays and the listener in the room. Parametric expressions for the control filters creating the beam patterns are derived and the expected beam pattern obtained from a given wedge array is predicted through simulation

    Sound field reproduction with a Cylindrical Loudspeaker Array using first order wall reflections

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    Sound field reproduction is a promising technology for reproducing a desired sound field over a specific area. However, several practical challenges exist, such as the requirement for a large number of loudspeakers and the degradation of spatial reproduction accuracy due to room reflections. This work proposes a sound field reproduction method based on a cylindrical loudspeaker array that is used to both radiate sound directly towards the listener and take advantage of controlled first-order reflections from the room’s walls. The sound field corresponding to one or more virtual sound sources is created by the combination of the direct and reflected sound at the listener position. As long as the lateral walls generate welldefined specular reflections, the simple model of the reverberant field with the proposed formulation enables this method to work without elaborate acoustical measurements. Two approaches with different concepts of controlling the array are compared and their mathematical equivalence is shown. To evaluate the performance, both simulations and experiments were conducted, showing a good agreement with theory

    DOA estimation performance with circular arrays in sound fields with finite rate of innovation

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    A novel Direction-of-Arrival (DOA) estimation method based on a plane wave sound field model was recently proposed for circular microphone arrays [1] . This article presents a detailed theoretical analysis of the method that relies on a Finite Rate of Innovation (FRI) assumption, investigating the impact of different theoretical acoustic source models. The method's estimation accuracy, and robustness against noisy measurement data and deviations from the model are investigated. The estimation performance is validated and assessed on the basis of results obtained from both simulations and experimental data

    Circular microphone array with tangential pressure gradient sensors

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    Microphone arrays have already been successfully applied to record sound fields. They are typically composed of pressure sensors and different designs have been suggested, each trying to overcome practical difficulties, such as transducer noise, spatial aliasing and non-uniqueness of the inverse solution. Typical designs are of spherical (3D) or circular form (2D) and use pressure sensors. The array corpus is usually either solid or as acoustically transparent as possible.In this paper, the theoretical model of a circular microphone array, observing the tangential component of the pressure gradient on its boundary is presented

    Sound field control with hemi-cylindrical loudspeaker arrays

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    An acoustical model for the sound field generated by hemi-cylindrical loudspeaker arrays is presented and a method for beamforming with said arrays is derived. The sound field model is obtained by introducing two independent boundary conditions for the sound field of a single impinging plane wave. The model for the radiation from a single loudspeaker in the array is then obtained from the reciprocity principle. Various beam patterns are presented and the theoretically predicted sound field is evaluated as a function of frequency. The results are discussed and an experimental array prototype is presented

    Theoretical study of acoustic circular arrays with tangential pressure gradient sensors

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    Microphone arrays as a means of sound field acquisition have been the topic of extensive research for more than eight decades now. A number of designs have been suggested, each trying to overcome difficulties that are inherent to either the decomposition of the sound field, the transducers in use or the presence of the array itself. This work presents a theoretical analysis of circular microphone arrays that do not measure the sound pressure but the component of its gradient that is tangential to a given boundary. Its performance is compared to that of a conventional pressure sensor array as a benchmark. The focus of the analysis and subsequent assessment lies on spatial aliasing and performance in the presence of noise

    Plane wave identification with circular arrays by means of a finite rate of innovation approach

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    Many problems in the field of acoustic measurements depend on the direction of incoming wave fronts w.r.t. a measurement device or aperture. This knowledge can be useful for signal processing purposes such as noise reduction, source separation, de-aliasing, and super-resolution strategies among others. This paper presents a signal processing technique for the identification of the directions of travel for the principal plane wave components in a sound field measured with a circular microphone array. The technique is derived from a finite rate of innovation data model and the performance is evaluated by means of a simulation study for different numbers of plane waves in the sound field

    Optimisation of a controllable Helmholtz resonator to achieve high frequency reconfigurability with large absorption values

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    Helmholtz resonators (HR) are effective acoustic absorbers in the low frequency regime, where the performance of conventional porous absorbers is limited. This is evident by their widespread applications, from controlling room modes to reducing noise transmission in ducts and fan noise radiated by aero-engines. However, HR-based sound absorbers only achieve significant sound absorption within a narrow frequency range. Thus, HR can become ineffective as soon as the acoustic environment changes (e.g., when increasing the occupancy of a room) and the problematic frequency falls outside the HR’s target frequency range. In response to the dynamically changing acoustic environment, a controllable HR with a large frequency reconfigurability range would be required. Recent research efforts have focused on reconfigurable HR that enable resonance frequency tuning over a large frequency range, but the sound absorption levels of the HR were not considered. In this contribution, a numerical optimisation study of a controllable HR to achieve a high frequency reconfigurability with simultaneously large sound absorption values throughout the reconfiguration range is presented. To achieve reconfigurability, a cylindrical HR with a controllable neck radius is considered and its geometry as well as added layers of porous material in the HR cavity are optimised

    Interactive real-time implementations of Higher Order Ambisonics to binaural rendering using VISR

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    This work presents the implementation and experimental validation of an interactive binaural renderer that uses spherical microphone array recordings. The plane wave density function is used to represent the sound field. One implementation using a complete head-related transfer function dataset and one using a spatially re-sampled set are considered. System’s performance is measured based on interaural time and level differences. Static performance validation is given by comparison to an established database. For the dynamic case, a real-time implementation using a head tracker is done. Good agreement is seen for interaural time differences. Significant errors for interaural level differences are found above the spatial aliasing frequency. The spatially re-sampled set implementation improves high-frequency content without affecting interaural time and level differences
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