112 research outputs found

    Localization of sound sources : a systematic review

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    Sound localization is a vast field of research and advancement which is used in many useful applications to facilitate communication, radars, medical aid, and speech enhancement to but name a few. Many different methods are presented in recent times in this field to gain benefits. Various types of microphone arrays serve the purpose of sensing the incoming sound. This paper presents an overview of the importance of using sound localization in different applications along with the use and limitations of ad-hoc microphones over other microphones. In order to overcome these limitations certain approaches are also presented. Detailed explanation of some of the existing methods that are used for sound localization using microphone arrays in the recent literature is given. Existing methods are studied in a comparative fashion along with the factors that influence the choice of one method over the others. This review is done in order to form a basis for choosing the best fit method for our use

    Distributed Microphone Array System for Two-way Audio Communication

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    TÀssÀ työssÀ esitellÀÀn hajautettu mikrofoniryhmÀjÀrjestelmÀ kahdensuuntaisessa ÀÀnikommunikaatiossa. JÀrjestelmÀn tavoitteena on paikallistaa hallitseva puhuja ja tallentaa puhesignaali mahdollisimman korkealaatuisesti. TyössÀ esiteltÀvÀssÀ jÀrjestelmÀssÀ jokainen mikrofoniryhmÀ toimii polynomirakenteella parametrisoituna keilanmuodostajana (PBF), joka mahdollistaa jatkuvan keilanohjauksen. Hallitsevan puhelÀhteen suunta pÀÀtellÀÀn PBF:n jokaisen keilan ulostulotehoista. Lopuksi yhdistÀmÀllÀ jokaisen PBF:n kaikkien keilojen ulostulotehot muodostetaan avaruudellinen todennÀköisyysfunktio (SLF), jonka suurin arvo mÀÀrÀÀ puhujan paikan. Puhesignaali tallennetaan ohjaamalla puhujaa lÀhinnÀ olevan PBF:n keila puhujan suuntaan. TÀssÀ työssÀ esiteltÀvÀn jÀrjestelmÀn toiminta arvioitiin simuloidulla ja mitatulla datalla. Arvionti nÀyttÀÀ, ettÀ toteutettu jÀrjestelmÀ pystyy paikallistamaan puhujan noin 40 cm paikannustarkkuudella ja jÀrjestelmÀ vaimentaa muista suunnista tulevia hÀiriölÀhteitÀ noin 15 dB. Lopuksi jÀrjestelmÀ toteutettiin reaaliakaisena systeeminÀ Pure Data signaalinkÀsittelyympÀristössÀ.In this work a distributed microphone array system for two-way audio communication is presented. The goal of the system is to locate the dominant speaker and capture the speech signal with highest possible quality. In the presented system each microphone array works as a Polynomial Beamformer (PBF) thus enabling continuous beam steering. The output power of each PBF beam is used to determine the direction of the dominant speech source. Finally, a Spatial Likelihood Function (SLF) is formed by combining the output beam powers of each microphone array and the speaker is determined to be in the point that has highest value of SLF. The audio signal capture is done by steering the closest microphone array to the direction of the speaker. The presented audio capture front-end was evaluated with simulated and measured data. The evaluation shows that the implemented system gives approximately 40 cm localization accuracy and 15 dB attenuation of interference sources. Finally the system was implemented to run in real-time in the Pure Data signal processing environment

    GUNSHOT DIRECTION OF ARRIVAL DETERMINATION USING BIO-INSPIRED MEMS SENSORS

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    A key component of battle space awareness is direction of arrival (DoA) determination of gunshots. In the initial stages of an engagement, quick and reliable DoA determination enhances a Marine’s ability to execute the observe-orient-decide-act (OODA) loop, increasing chances of survival and mission success. Naval Postgraduate School (NPS) has developed a novel, biomimetic acoustic sensor modeled after the auditory system of the Ormia Ochracea fly. This microelectromechanical system (MEMS)-based directional sound sensor, which consists of two wings connected to a substrate using two torsional legs in the middle, is well documented in previous NPS theses. Each sensor has a uniform dipole beam pattern. By combining two crossed MEMS sensors (crossed-dipoles) with an omni-directional microphone, 360° DoA determination can be fully resolved. The objective of this thesis is to evaluate, optimize, and develop DoA estimators for gunshots in the time- and frequency-domain, specifically for the crossed-dipoles sensors plus an omni-directional microphone configuration.ONR, Arlington, VA 22203Outstanding ThesisEnsign, United States NavyApproved for public release. Distribution is unlimited

    Towards Secure, Power-Efficient and Location-Aware Mobile Computing

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    In the post-PC era, mobile devices will replace desktops and become the main personal computer for many people. People rely on mobile devices such as smartphones and tablets for everything in their daily lives. A common requirement for mobile computing is wireless communication. It allows mobile devices to fetch remote resources easily. Unfortunately, the increasing demand of the mobility brings many new wireless management challenges such as security, energy-saving and location-awareness. These challenges have already impeded the advancement of mobile systems. In this dissertation we attempt to discover the guidelines of how to mitigate these problems through three general communication patterns in 802.11 wireless networks. We propose a cross-section of a few interesting and important enhancements to manage wireless connectivity. These enhancements provide useful primitives for the design of next-generation mobile systems in the future.;Specifically, we improve the association mechanism for wireless clients to defend against rogue wireless Access Points (APs) in Wireless LANs (WLANs) and vehicular networks. Real-world prototype systems confirm that our scheme can achieve high accuracy to detect even sophisticated rogue APs under various network conditions. We also develop a power-efficient system to reduce the energy consumption for mobile devices working as software-defined APs. Experimental results show that our system allows the Wi-Fi interface to sleep for up to 88% of the total time in several different applications and reduce the system energy by up to 33%. We achieve this while retaining comparable user experiences. Finally, we design a fine-grained scalable group localization algorithm to enable location-aware wireless communication. Our prototype implemented on commercial smartphones proves that our algorithm can quickly locate a group of mobile devices with centimeter-level accuracy

    Multichannel source separation and tracking with phase differences by random sample consensus

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    Blind audio source separation (BASS) is a fascinating problem that has been tackled from many different angles. The use case of interest in this thesis is that of multiple moving and simultaneously-active speakers in a reverberant room. This is a common situation, for example, in social gatherings. We human beings have the remarkable ability to focus attention on a particular speaker while effectively ignoring the rest. This is referred to as the ``cocktail party effect'' and has been the holy grail of source separation for many decades. Replicating this feat in real-time with a machine is the goal of BASS. Single-channel methods attempt to identify the individual speakers from a single recording. However, with the advent of hand-held consumer electronics, techniques based on microphone array processing are becoming increasingly popular. Multichannel methods record a sound field from various locations to incorporate spatial information. If the speakers move over time, we need an algorithm capable of tracking their positions in the room. For compact arrays with 1-10 cm of separation between the microphones, this can be accomplished by applying a temporal filter on estimates of the directions-of-arrival (DOA) of the speakers. In this thesis, we review recent work on BSS with inter-channel phase difference (IPD) features and provide extensions to the case of moving speakers. It is shown that IPD features compose a noisy circular-linear dataset. This data is clustered with the RANdom SAmple Consensus (RANSAC) algorithm in the presence of strong reverberation to simultaneously localize and separate speakers. The remarkable performance of RANSAC is due to its natural tendency to reject outliers. To handle the case of non-stationary speakers, a factorial wrapped Kalman filter (FWKF) and a factorial von Mises-Fisher particle filter (FvMFPF) are proposed that track source DOAs directly on the unit circle and unit sphere, respectively. These algorithms combine directional statistics, Bayesian filtering theory, and probabilistic data association techniques to track the speakers with mixtures of directional distributions

    Design Strategies for Adaptive Social Composition: Collaborative Sound Environments

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    In order to develop successful collaborative music systems a variety of subtle interactions need to be identified and integrated. Gesture capture, motion tracking, real-time synthesis, environmental parameters and ubiquitous technologies can each be effectively used for developing innovative approaches to instrument design, sound installations, interactive music and generative systems. Current solutions tend to prioritise one or more of these approaches, refining a particular interface technology, software design or compositional approach developed for a specific composition, performer or installation environment. Within this diverse field a group of novel controllers, described as ‘Tangible Interfaces’ have been developed. These are intended for use by novices and in many cases follow a simple model of interaction controlling synthesis parameters through simple user actions. Other approaches offer sophisticated compositional frameworks, but many of these are idiosyncratic and highly personalised. As such they are difficult to engage with and ineffective for groups of novices. The objective of this research is to develop effective design strategies for implementing collaborative sound environments using key terms and vocabulary drawn from the available literature. This is articulated by combining an empathic design process with controlled sound perception and interaction experiments. The identified design strategies have been applied to the development of a new collaborative digital instrument. A range of technical and compositional approaches was considered to define this process, which can be described as Adaptive Social Composition. Dan Livingston

    MICROPHONE ARRAY OPTIMIZATION IN IMMERSIVE ENVIRONMENTS

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    The complex relationship between array gain patterns and microphone distributions limits the application of traditional optimization algorithms on irregular arrays, which show enhanced beamforming performance for human speech capture in immersive environments. This work analyzes the relationship between irregular microphone geometries and spatial filtering performance with statistical methods. Novel geometry descriptors are developed to capture the properties of irregular microphone distributions showing their impact on array performance. General guidelines and optimization methods for regular and irregular array design are proposed in immersive (near-field) environments to obtain superior beamforming ability for speech applications. Optimization times are greatly reduced through the objective function rules using performance-based geometric descriptions of microphone distributions that circumvent direct array gain computations over the space of interest. In addition, probabilistic descriptions of acoustic scenes are introduced to incorporate various levels of prior knowledge for the source distribution. To verify the effectiveness of the proposed optimization methods, simulated gain patterns and real SNR results of the optimized arrays are compared to corresponding traditional regular arrays and arrays obtained from direct exhaustive searching methods. Results show large SNR enhancements for the optimized arrays over arbitrary randomly generated arrays and regular arrays, especially at low microphone densities. The rapid convergence and acceptable processing times observed during the experiments establish the feasibility of proposed optimization methods for array geometry design in immersive environments where rapid deployment is required with limited knowledge of the acoustic scene, such as in mobile platforms and audio surveillance applications

    Beyond input devices : a new conceptual framework for the design of physical-digital objects

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    Thesis (M.S.)--Massachusetts Institute of Technology, Program in Media Arts & Sciences, 1998.Includes bibliographical references (leaves 108-111).Matthew G. Gorbet.M.S
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