Exploiting partial reconfiguration through PCIe for a microphone array network emulator

The current Microelectromechanical Systems (MEMS) technology enables the deployment of relatively low-cost wireless sensor networks composed of MEMS microphone arrays for accurate sound source localization. However, the evaluation and the selection of the most accurate and power-efficient network’s topology are not trivial when considering dynamic MEMS microphone arrays. Although software simulators are usually considered, they consist of high-computational intensive tasks, which require hours to days to be completed. In this paper, we present an FPGA-based platform to emulate a network of microphone arrays. Our platform provides a controlled simulated acoustic environment, able to evaluate the impact of different network configurations such as the number of microphones per array, the network’s topology, or the used detection method. Data fusion techniques, combining the data collected by each node, are used in this platform. The platform is designed to exploit the FPGA’s partial reconfiguration feature to increase the flexibility of the network emulator as well as to increase performance thanks to the use of the PCI-express high-bandwidth interface. On the one hand, the network emulator presents a higher flexibility by partially reconfiguring the nodes’ architecture in runtime. On the other hand, a set of strategies and heuristics to properly use partial reconfiguration allows the acceleration of the emulation by exploiting the execution parallelism. Several experiments are presented to demonstrate some of the capabilities of our platform and the benefits of using partial reconfiguration

A Low-Cost Robust Distributed Linearly Constrained Beamformer for Wireless Acoustic Sensor Networks with Arbitrary Topology

We propose a new robust distributed linearly constrained beamformer which utilizes a set of linear equality constraints to reduce the cross power spectral density matrix to a block-diagonal form. The proposed beamformer has a convenient objective function for use in arbitrary distributed network topologies while having identical performance to a centralized implementation. Moreover, the new optimization problem is robust to relative acoustic transfer function (RATF) estimation errors and to target activity detection (TAD) errors. Two variants of the proposed beamformer are presented and evaluated in the context of multi-microphone speech enhancement in a wireless acoustic sensor network, and are compared with other state-of-the-art distributed beamformers in terms of communication costs and robustness to RATF estimation errors and TAD errors

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

Source ambiguity resolution of overlapped sounds in a multi-microphone room environment

When several acoustic sources are simultaneously active in a meeting room scenario, and both the position of the sources and the identity of the time-overlapped sound classes have been estimated, the problem of assigning each source position to one of the sound classes still remains. This problem is found in the real-time system implemented in our smart-room, where it is assumed that up to two acoustic events may overlap in time and the source positions are relatively well separated in space. The position assignment system proposed in this work is based on fusion of model-based log-likelihood ratios obtained after carrying out several different partial source separations in parallel. To perform the separation, frequency-invariant null-steering beamformers, which can work with a small number of microphones, are used. The experimental results using all the six microphone arrays deployed in the room show a high assignment rate in our particular scenario.Peer ReviewedPostprint (published version

A directionally tunable but frequency-invariant beamformer on an acoustic velocity-sensor triad to enhance speech perception

Herein investigated are computationally simple microphone-array beamformers that are independent of the frequency-spectra of all signals, all interference, and all noises. These beamformers allow the listener to tune the desired azimuth-elevation “look direction.” No prior information is needed of the interference. These beamformers deploy a physically compact triad of three collocated but orthogonally oriented velocity sensors. These proposed schemes’ efficacy is verified by a jury test, using simulated data constructed with Mandarin Chinese (a.k.a. Putonghua) speech samples. For example, a desired speech signal, originally at a very adverse signal-to-interference-and-noise power ratio (SINR) of -30 dB, may be processed to become fully intelligible to the jury

EXPERIMENTAL EVALUATION OF MODIFIED PHASE TRANSFORM FOR SOUND SOURCE DETECTION

The detection of sound sources with microphone arrays can be enhanced through processing individual microphone signals prior to the delay and sum operation. One method in particular, the Phase Transform (PHAT) has demonstrated improvement in sound source location images, especially in reverberant and noisy environments. Recent work proposed a modification to the PHAT transform that allows varying degrees of spectral whitening through a single parameter, andamp;acirc;, which has shown positive improvement in target detection in simulation results. This work focuses on experimental evaluation of the modified SRP-PHAT algorithm. Performance results are computed from actual experimental setup of an 8-element perimeter array with a receiver operating characteristic (ROC) analysis for detecting sound sources. The results verified simulation results of PHAT- andamp;acirc; in improving target detection probabilities. The ROC analysis demonstrated the relationships between various target types (narrowband and broadband), room reverberation levels (high and low) and noise levels (different SNR) with respect to optimal andamp;acirc;. Results from experiment strongly agree with those of simulations on the effect of PHAT in significantly improving detection performance for narrowband and broadband signals especially at low SNR and in the presence of high levels of reverberation

Spectrum Sharing in Wireless Networks via QoS-Aware Secondary Multicast Beamforming

Secondary spectrum usage has the potential to considerably increase spectrum utilization. In this paper, quality-of-service (QoS)-aware spectrum underlay of a secondary multicast network is considered. A multiantenna secondary access point (AP) is used for multicast (common information) transmission to a number of secondary single-antenna receivers. The idea is that beamforming can be used to steer power towards the secondary receivers while limiting sidelobes that cause interference to primary receivers. Various optimal formulations of beamforming are proposed, motivated by different ldquocohabitationrdquo scenarios, including robust designs that are applicable with inaccurate or limited channel state information at the secondary AP. These formulations are NP-hard computational problems; yet it is shown how convex approximation-based multicast beamforming tools (originally developed without regard to primary interference constraints) can be adapted to work in a spectrum underlay context. Extensive simulation results demonstrate the effectiveness of the proposed approaches and provide insights on the tradeoffs between different design criteria

Advanced algorithms for audio and image processing

The objective of the thesis is the development of a set of innovative algorithms around the topic of beamforming in the field of acoustic imaging, audio and image processing, aimed at significantly improving the performance of devices that exploit these computational approaches. Therefore the context is the improvement of devices (ultrasound machines and video/audio devices) already on the market or the development of new ones which, through the proposed studies, can be introduced on new the markets with the launch of innovative high-tech start-ups. This is the motivation and the leitmotiv behind the doctoral work carried out. In fact, in the first part of the work an innovative image reconstruction algorithm in the field of ultrasound biomedical imaging is presented, which is connected to the development of such equipment that exploits the computing opportunities currently offered nowadays at low cost by GPUs (Moore\u2019s law). The proposed target is to obtain a new pipeline of the reconstruction of the image abandoning the architecture of such hardware based In the first part of the thesis I faced the topic of the reconstruction of ultrasound images for applications hypothesized on a software based device through image reconstruction algorithms processed in the frequency domain. An innovative beamforming algorithm based on seismic migration is presented, in which a transformation of the RF data is carried out and the reconstruction algorithm can evaluate a masking of the k-space of the data, speeding up the reconstruction process and reducing the computational burden. The analysis and development of the algorithms responsible for carrying out the thesis has been approached from a feasibility point in an off-line context and on the Matlab platform, processing both synthetic simulated generated data and real RF data: the subsequent development of these algorithms within of the future ultrasound biomedical equipment will exploit an high-performance computing framework capable of processing customized kernel pipelines (henceforth called \u2019filters\u2019) on CPU/GPU. The type of filters implemented involved the topic of Plane Wave Imaging (PWI), an alternative method of acquiring the ultrasound image compared to the state of the art of the traditional standard B-mode which currently exploit sequential sequence of insonification of the sample under examination through focused beams transmitted by the probe channels. The PWI mode is interesting and opens up new scenarios compared to the usual signal acquisition and processing techniques, with the aim of making signal processing in general and image reconstruction in particular faster and more flexible, and increasing importantly the frame rate opens up and improves clinical applications. The innovative idea is to introduce in an offline seismic reconstruction algorithm for ultrasound imaging a further filter, named masking matrix. The masking matrices can be computed offline knowing the system parameters, since they do not depend from acquired data. Moreover, they can be pre-multiplied to propagation matrices, without affecting the overall computational load. Subsequently in the thesis, the topic of beamforming in audio processing on super-direct linear arrays of microphones is addressed. The aim is to make an in depth analysis of two main families of data-independent approaches and algorithms present in the literature by comparing their performances and the trade-off between directivity and frequency invariance, which is not yet known at to the state-of-the-art. The goal is to validate the best algorithm that allows, from the perspective of an implementation, to experimentally verify performance, correlating it with the characteristics and error statistics. Frequency-invariant beam patterns are often required by systems using an array of sensors to process broadband signals. In some experimental conditions, the array spatial aperture is shorter than the involved wavelengths. In these conditions, superdirective beamforming is essential for an efficient system. I present a comparison between two methods that deal with a data-independent beamformer based on a filter-and-sum structure. Both methods (the first one numerical, the second one analytic) formulate a mathematical convex minimization problem, in which the variables to be optimized are the filters coefficients or frequency responses. In the described simulations, I have chosen a geometry and a set-up of parameters that allows us to make a fair comparison between the performances of the two different design methods analyzed. In particular, I addressed a small linear array for audio capture with different purposes (hearing aids, audio surveillance system, video-conference system, multimedia device, etc.). The research activity carried out has been used for the launch of a high-tech device through an innovative start-up in the field of glasses/audio devices (https://acoesis.com/en/). It has been proven that the proposed algorithm gives the possibility of obtaining higher performances than the state of the art of similar algorithms, additionally providing the possibility of connecting directivity or better generalized directivity to the statistics of phase errors and gain of sensors, extremely important in superdirective arrays in the case of real and industrial implementation. Therefore, the method proposed by the comparison is innovative because it quantitatively links the physical construction characteristics of the array to measurable and experimentally verifiable quantities, making the real implementation process controllable. The third topic faced is the reconstruction of the Room Impluse Response (RIR) using audio processing blind methods. Given an unknown audio source, the estimation of time differences-of-arrivals (TDOAs) can be efficiently and robustly solved using blind channel identification and exploiting the cross-correlation identity (CCI). Prior blind works have improved the estimate of TDOAs by means of different algorithmic solutions and optimization strategies, while always sticking to the case N = 2 microphones. But what if we can obtain a direct improvement in performance by just increasing N? In the fourth Chapter I tried to investigate this direction, showing that, despite the arguable simplicity, this is capable of (sharply) improving upon state-of-the-art blind channel identification methods based on CCI, without modifying the computational pipeline. Inspired by our results, we seek to warm up the community and the practitioners by paving the way (with two concrete, yet preliminary, examples) towards joint approaches in which advances in the optimization are combined with an increased number of microphones, in order to achieve further improvements. Sound source localisation applications can be tackled by inferring the time-difference-of-arrivals (TDOAs) between a sound-emitting source and a set of microphones. Among the referred applications, one can surely list room-aware sound reproduction, room geometry\u2019s estimation, speech enhancement. Despite a broad spectrum of prior works estimate TDOAs from a known audio source, even when the signal emitted from the acoustic source is unknown, TDOAs can be inferred by comparing the signals received at two (or more) spatially separated microphones, using the notion of cross-corrlation identity (CCI). This is the key theoretical tool, not only, to make the ordering of microphones irrelevant during the acquisition stage, but also to solve the problem as blind channel identification, robustly and reliably inferring TDOAs from an unknown audio source. However, when dealing with natural environments, such \u201cmutual agreement\u201d between microphones can be tampered by a variety of audio ambiguities such as ambient noise. Furthermore, each observed signal may contain multiple distorted or delayed replicas of the emitting source due to reflections or generic boundary effects related to the (closed) environment. Thus, robustly estimating TDOAs is surely a challenging problem and CCI-based approaches cast it as single-input/multi-output blind channel identification. Such methods promote robustness in the estimate from the methodological standpoint: using either energy-based regularization, sparsity or positivity constraints, while also pre-conditioning the solution space. Last but not least, the Acoustic Imaging is an imaging modality that exploits the propagation of acoustic waves in a medium to recover the spatial distribution and intensity of sound sources in a given region. Well known and widespread acoustic imaging applications are, for example, sonar and ultrasound. There are active and passive imaging devices: in the context of this thesis I consider a passive imaging system called Dual Cam that does not emit any sound but acquires it from the environment. In an acoustic image each pixel corresponds to the sound intensity of the source, the whose position is described by a particular pair of angles and, in the case in which the beamformer can, as in our case, work in near-field, from a distance on which the system is focused. In the last part of this work I propose the use of a new modality characterized by a richer information content, namely acoustic images, for the sake of audio-visual scene understanding. Each pixel in such images is characterized by a spectral signature, associated to a specific direction in space and obtained by processing the audio signals coming from an array of microphones. By coupling such array with a video camera, we obtain spatio-temporal alignment of acoustic images and video frames. This constitutes a powerful source of self-supervision, which can be exploited in the learning pipeline we are proposing, without resorting to expensive data annotations. However, since 2D planar arrays are cumbersome and not as widespread as ordinary microphones, we propose that the richer information content of acoustic images can be distilled, through a self-supervised learning scheme, into more powerful audio and visual feature representations. The learnt feature representations can then be employed for downstream tasks such as classification and cross-modal retrieval, without the need of a microphone array. To prove that, we introduce a novel multimodal dataset consisting in RGB videos, raw audio signals and acoustic images, aligned in space and synchronized in time. Experimental results demonstrate the validity of our hypothesis and the effectiveness of the proposed pipeline, also when tested for tasks and datasets different from those used for training. Chapter 6 closes the thesis, presenting a development activity of a new Dual Cam POC to build-up from it a spin-off, assuming to apply for an innovation project for hi-tech start- ups (such as a SME instrument H2020) for a 50Keuro grant, following the idea of the technology transfer. A deep analysis of the reference market, technologies and commercial competitors, business model and the FTO of intellectual property is then conducted. Finally, following the latest technological trends (https://www.flir.eu/products/si124/) a new version of the device (planar audio array) with reduced dimensions and improved technical characteristics is simulated, simpler and easier to use than the current one, opening up new interesting possibilities of development not only technical and scientific but also in terms of business fallout