349 research outputs found
Spatial sampling and beamforming for spherical microphone arrays
Spherical microphone arrays have been recently studied for spatial sound
recording, speech communication, and sound field analysis for room acoustics
and noise control. Complementary theoretical studies presented progress in
spatial sampling and beamforming methods. This paper reviews recent results in
spatial sampling that facilitate a wide range of spherical array
configurations, from a single rigid sphere to free positioning of microphones.
The paper then presents an overview of beamforming methods recently presented
for spherical arrays, from the widely used delay-and-sum and Dolph-Chebyshev,
to the more advanced optimal methods, typically performed in the spherical
harmonics domain
Clustering Inverse Beamforming and multi-domain acoustic imaging approaches for vehicles NVH
Il rumore percepito all’interno della cabina di un veicolo è un aspetto molto rilevante nella valutazione della sua qualità complessiva. Metodi sperimentali di acoustic imaging, quali beamforming e olografia acustica, sono usati per identificare le principali sorgenti che contribuiscono alla rumorosità percepita all’interno del veicolo. L’obiettivo della tesi proposta è di fornire strumenti per effettuare dettagliate analisi quantitative tramite tali tecniche, ad oggi relegate alle fasi di studio preliminare, proponendo un approccio modulare che si avvale di analisi dei fenomeni vibro-acustici nel dominio della frequenza, del tempo e dell’angolo di rotazione degli elementi rotanti tipicamente presenti in un veicolo. Ciò permette di ridurre tempi e costi della progettazione, garantendo, al contempo, una maggiore qualità del pacchetto vibro-acustico. L’innovativo paradigma proposto prevede l’uso combinato di algoritmi di pre- e post- processing con tecniche inverse di acoustic imaging per lo studio di rilevanti problematiche quali l’identificazione di sorgenti sonore esterne o interne all’abitacolo e del rumore prodotto da dispositivi rotanti. Principale elemento innovativo della tesi è la tecnica denominata Clustering Inverse Beamforming. Essa si basa su un approccio statistico che permette di incrementare l’accuratezza (range dinamico, localizzazione e quantificazione) di una immagine acustica tramite la combinazione di soluzioni, del medesimo problema inverso, ottenute considerando diversi sotto-campioni dell’informazione sperimentale disponibile, variando, in questo modo, in maniera casuale la sua formulazione matematica. Tale procedimento garantisce la ricostruzione nel dominio della frequenza e del tempo delle sorgenti sonore identificate. Un metodo innovativo è stato inoltre proposto per la ricostruzione, ove necessario, di sorgenti sonore nel dominio dell’angolo. I metodi proposti sono stati supportati da argomentazioni teoriche e validazioni sperimentali su scala accademica e industriale.The interior sound perceived in vehicle cabins is a very important attribute for the user. Experimental acoustic imaging methods such as beamforming and Near-field Acoustic Holography are used in vehicles noise and vibration studies because they are capable of identifying the noise sources contributing to the overall noise perceived inside the cabin. However these techniques are often relegated to the troubleshooting phase, thus requiring additional experiments for more detailed NVH analyses. It is therefore desirable that such methods evolve towards more refined solutions capable of providing a larger and more detailed information. This thesis proposes a modular and multi-domain approach involving direct and inverse acoustic imaging techniques for providing quantitative and accurate results in frequency, time and angle domain, thus targeting three relevant types of problems in vehicles NVH: identification of exterior sources affecting interior noise, interior noise source identification, analysis of noise sources produced by rotating machines. The core finding of this thesis is represented by a novel inverse acoustic imaging method named Clustering Inverse Beamforming (CIB). The method grounds on a statistical processing based on an Equivalent Source Method formulation. In this way, an accurate localization, a reliable ranking of the identified sources in frequency domain and their separation into uncorrelated phenomena is obtained. CIB is also exploited in this work for allowing the reconstruction of the time evolution of the sources sought. Finally a methodology for decomposing the acoustic image of the sound field generated by a rotating machine as a function of the angular evolution of the machine shaft is proposed. This set of findings aims at contributing to the advent of a new paradigm of acoustic imaging applications in vehicles NVH, supporting all the stages of the vehicle design with time-saving and cost-efficient experimental techniques. The proposed innovative approaches are validated on several simulated and real experiments
Real-time Microphone Array Processing for Sound-field Analysis and Perceptually Motivated Reproduction
This thesis details real-time implementations of sound-field analysis and perceptually motivated reproduction methods for visualisation and auralisation purposes. For the former, various methods for visualising the relative distribution of sound energy from one point in space are investigated and contrasted; including a novel reformulation of the cross-pattern coherence (CroPaC) algorithm, which integrates a new side-lobe suppression technique. Whereas for auralisation applications, listening tests were conducted to compare ambisonics reproduction with a novel headphone formulation of the directional audio coding (DirAC) method. The results indicate that the side-lobe suppressed CroPaC method offers greater spatial selectivity in reverberant conditions compared with other popular approaches, and that the new DirAC formulation yields higher perceived spatial accuracy when compared to the ambisonics method
Beamforming with double-sided, acoustically hard planar arrays
International audienceSeveral types of microphone arrays have been proposed for the purpose of capturing spatial audio signals. When the aim is to capture a full 3D sound field, the most obvious geometry is that of a sphere, and this is the array shape which has received the most attention from authors in this field. The wave equation and its solutions can be separated into the spherical coordinates, and for an internal problem like an open microphone array, the solutions are linear combinations of the well-known ambisonic basis functions. These lend themselves well to the analysis and optimization of spherical arrays. One important observation is that the radial part of these solutions is described by the spherical Bessel functions and that these oscillate around zero. This leads to the problem that a single-radius open array (i.e. one which does not perturb the sound field) will not be able to function over a wide frequency band.An open multi-radius array is able to overcome the problem by simultaneously sampling the Bessel functions at different points, chosen such that not all of them are zero at the same time. Another solution is to sample the sound field with a mixture of zeroth- and first-order microphones. Since the radial pressure gradient is maximal where the pressure crosses zero, these different microphone types can complement each other to produce a continuous coverage over a wide frequency range. However, the most popular solution to the problem is to mount zeroth-order microphones on a hard spherical shell. The scattering off the shell is maximal at those frequencies where the pressure of the incident field is zero at the radius of the shell. The resulting frequency response of the total sound field on the shell is a smooth function without zero crossings.Another geometry which has been studied for the same purpose is the planar array. This effectively samples the sound field in the x-y plane. The shape of the ambisonic basis functions is such that about half of them are zero in this plane. As with the spherical arrays, there are three possible solutions to this problem; the microphones can be replaced with a mixture of zeroth- and first-order microphones, the array can be comprised of several parallel layers of microphones, or a scattering surface can be introduced in the x-y plane, with zeroth-order microphones placed on both sides. A paper concerning this last solution has only recently been published. So far, these arrays have only been studied with regards to their suitability for capturing ambisonic signals.Among the studies that have been published concerning spherical arrays, several authors have proposed beamforming methods that perform better than methods that use ambisonic signals as an intermediate representation of the sound field. In particular, it has been shown that effective beamforming can be achieved beyond the frequencies where the ambisonic signals become unusable due to spatial aliasing.The current paper proposes beamforming methods for acoustically hard planar arrays and compares these with methods using intermediate ambisonic signals. The methods are tested numerically and experimentally on a 17 cm diameter disc-shaped array with 84 microphones in a 7-by-6-by-2 configuration (6 rings of 7 microphones on each side). The implications of these techniques on the optimal microphone layout of the array are addressed
Spatial dissection of a soundfield using spherical harmonic decomposition
A real-world soundfield is often contributed by multiple desired and undesired sound sources. The performance of many acoustic systems such as automatic speech recognition, audio surveillance, and teleconference relies on its ability to extract the desired sound components in such a mixed environment. The existing solutions to the above problem are constrained by various fundamental limitations and require to enforce different priors depending on the acoustic condition such as reverberation and spatial distribution of sound sources. With the growing emphasis and integration of audio applications in diverse technologies such as smart home and virtual reality appliances, it is imperative to advance the source separation technology in order to overcome the limitations of the traditional approaches.
To that end, we exploit the harmonic decomposition model to dissect a mixed soundfield into its underlying desired and undesired components based on source and signal characteristics. By analysing the spatial projection of a soundfield, we achieve multiple outcomes such as (i) soundfield separation with respect to distinct source regions, (ii) source separation in a mixed soundfield using modal coherence model, and (iii) direction of arrival (DOA) estimation of multiple overlapping sound sources through pattern recognition of the modal coherence of a soundfield.
We first employ an array of higher order microphones for soundfield separation in order to reduce hardware requirement and implementation complexity. Subsequently, we develop novel mathematical models for modal coherence of noisy and reverberant soundfields that facilitate convenient ways for estimating DOA and power spectral densities leading to robust source separation algorithms. The modal domain approach to the soundfield/source separation allows us to circumvent several practical limitations of the existing techniques and enhance the performance and robustness of the system. The proposed methods are presented with several practical applications and performance evaluations using simulated and real-life dataset
Multiple source localization in the spherical harmonic domain using augmented intensity vectors based on grid search
Multiple source localization is an important task in acoustic signal processing with applications including dereverberation, source separation, source tracking and environment mapping. When using spherical microphone arrays, it has been previously shown that Pseudo-intensity Vectors (PIV), and Augmented Intensity Vectors (AIV), are an effective approach for direction of arrival estimation of a sound source. In this paper, we evaluate AIV-based localization in acoustic scenarios involving multiple sound sources. Simulations are conducted where the number of sources, their angular separation and the reverberation time of the room are varied. The results indicate that AIV outperforms PIV and Steered Response Power (SRP) with an average accuracy between 5 and 10 degrees for sources with angular separation of 30 degrees or more. AIV also shows better robustness to reverberation time than PIV and SRP
Design of time-domain modal beamformer for broadband spherical microphone arrays
ABSTRACT An approach to real-valued time-domain implementation of modal beamformer for spherical microphone arrays is proposed. The advantage of the time-domain implementation is that we can update the beamformer when each new snapshot arrives. Our technique is based on a modified filter-and-sum spherical harmonics domain (SHD) beamforming structure. The time series received at the microphones are converted into SHD data using spherical Fourier transform. The SHD data input to the steering unit and then feed a bank of finite impulse response (FIR) filters. The filter outputs are summed to produce the beamformer output time series. The FIR filters tap weights are optimally designed by making a compromise among multiple conflicting array performance measures such as directivity, mainlobe spatial response variation (MSRV), sidelobe level, and robustness. The design problem is formulated as a multiply constrained problem which is solved using second-order cone programming (SOCP). Results of simulations show good performance of the proposed time-domain SHD beamformer design approach
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