Дисертації з теми "SounBe"
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Gunawan, David Oon Tao Electrical Engineering & Telecommunications Faculty of Engineering UNSW. "Musical instrument sound source separation." Awarded By:University of New South Wales. Electrical Engineering & Telecommunications, 2009. http://handle.unsw.edu.au/1959.4/41751.
Повний текст джерелаAlghassi, Hedayat. "Eye array sound source localization." Thesis, University of British Columbia, 2008. http://hdl.handle.net/2429/5114.
Повний текст джерелаPompei, F. Joseph (Frank Joseph) 1973. "Sound from ultrasound : the parametric array as an audible sound source." Thesis, Massachusetts Institute of Technology, 2002. http://hdl.handle.net/1721.1/7987.
Повний текст джерелаVita.
Includes bibliographical references (leaves 91-94).
A parametric array exploits the nonlinearity of the propagation medium to emit or detect acoustic waves in a spatially versatile manner, permitting concise, narrow directivity patterns otherwise possible only with physically very large transducer geometries. This thesis explores the use of the parametric array as an audible sound source, permitting audible sound to be generated with very high directivity compared to traditional loudspeakers of comparable size. The thesis begins with a review of basic underlying mathematics and relevant approximate solutions of nonlinear acoustic systems. Then, these solutions are used to construct suitable methods of ultrasonic synthesis for low-distortion audio reproduction. Geometrical modelling methods for predicting the acoustic distribution are presented and evaluated, and practical applications are explored experimentally. Issues of risk associated with ultrasonic exposure are presented, and the feasibility of a phased-array system for beam control is explored.
F. Joseph Pompei.
Ph.D.
Olsson, Erik. "Sound source localization from laser vibrometry recordings." Doctoral thesis, Luleå : Division of experimental mechanics, Luleå University of Technology, 2007. http://epubl.ltu.se/1402-1544/2007/23/.
Повний текст джерелаBenichoux, Victor. "Timing cues for azimuthal sound source localization." Phd thesis, Université René Descartes - Paris V, 2013. http://tel.archives-ouvertes.fr/tel-00931645.
Повний текст джерелаShare, C. P. "Real-time simulation of sound source occlusion." Thesis, Queen's University Belfast, 2011. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.546425.
Повний текст джерелаLam, Alice. "3D sound-source localization using triangulation-based methods." Thesis, University of British Columbia, 2017. http://hdl.handle.net/2429/63551.
Повний текст джерелаApplied Science, Faculty of
Mechanical Engineering, Department of
Graduate
Cavalieri, André Valdetaro Gomes. "Wavepackets as sound-source mechanisms in subsonic jets." Thesis, Poitiers, 2012. http://www.theses.fr/2012POIT2253/document.
Повний текст джерелаHydrodynamic wavepackets are studied as a sound-source mechanism in subsonic jets. We first analyse numerical simulations to discern properties of acoustic sources such as compactness, intermittency and azimuthal structure. The simulations include a DNS of a two-dimensional mixing layer (Wei and Freund 2006) and an LES of a Mach 0.9 jet (Daviller 2010). In both cases we identify intermittent radiation, which is associated with changes in coherent structures in the flows. A wave-packet model that includes temporal changes in amplitude and axial extension is proposed to represent the identified phenomena using Lighthill's analogy. These parameters are obtained from velocity data of two subsonic jet simulations, and an agreement to within 1.5dB between the model and the acoustic field of the simulations confirms its pertinence. The proposed mechanism is then investigatedexperimentally, with measurements of acoustic pressure and velocity of turbulent subsonic jets, allowing the decomposition of the fields into azimuthal Fourier modes. We find close agreement of the directivities of modes 0, 1 and 2 of the acoustic field with wave-packet radiation. Modes 0 and 1 of the velocity field correspond also to wavepackets, modelled as linear instability waves using parabolised stability equations. Finally, correlations of order of 10% between axisymmetric modes of velocity and far-field pressure show the relationship between wavepackets and sound radiated by the jet
Kjellson, Angelica. "Sound Source Localization and Beamforming for Teleconferencing Solutions." Thesis, Umeå universitet, Institutionen för matematik och matematisk statistik, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:umu:diva-89707.
Повний текст джерелаGod ljudkvalitet är en grundsten för lyckade telefonmöten. Miljön i ett konferens-rum medför ett flertal olika utmaningar för behandlingen av mikrofonsignalerna: det kan t.ex. vara brus och störningar, eller att den som talar befinner sig långt från telefonen. Målet med detta arbete är att förbättra den talsignal som tas upp av en konferenstelefon genom att implementera lösningar för lokalisering av talaren och riktad ljudupptagning med hjälp av ett flertal mikrofoner. De implementerade metoderna jämförs med en befintlig lösning och utvärderas under olika brusscenarion för en likformig cirkulär mikrofonkonstellation. För utvärderingen användes testsignaler som spelades in med en specialbyggd enhet. De implementerade algoritmerna kunde inte uppvisa en tillräcklig förbättring i jämförelse med den befintliga lösningen för att motivera den ökade beräkningskomplexitet de skulle medföra. Dessutom konstaterades att en fördubbling av antalet mikrofoner gav liten eller ingen förbättring på metoderna. Vilken typ av mikrofon som användes konstaterades däremot påverka resultatet och en subjektiv utvärdering indikerade en preferens för de rundupptagande mikrofonerna, en skillnad som föreslås undersökas vidare.
Martin, Keith Dana. "Sound-source recognition : a theory and computational model." Thesis, Massachusetts Institute of Technology, 1999. http://hdl.handle.net/1721.1/9468.
Повний текст джерелаIncludes bibliographical references (p. 159-172).
The ability of a normal human listener to recognize objects in the environment from only the sounds they produce is extraordinarily robust with regard to characteristics of the acoustic environment and of other competing sound sources. In contrast, computer systems designed to recognize sound sources function precariously, breaking down whenever the target sound is degraded by reverberation, noise, or competing sounds. Robust listening requires extensive contextual knowledge, but the potential contribution of sound-source recognition to the process of auditory scene analysis has largely been neglected by researchers building computational models of the scene analysis process. This thesis proposes a theory of sound-source recognition, casting recognition as a process of gathering information to enable the listener to make inferences about objects in the environment or to predict their behavior. In order to explore the process, attention is restricted to isolated sounds produced by a small class of sound sources, the non-percussive orchestral musical instruments. Previous research on the perception and production of orchestral instrument sounds is reviewed from a vantage point based on the excitation and resonance structure of the sound-production process, revealing a set of perceptually salient acoustic features. A computer model of the recognition process is developed that is capable of "listening" to a recording of a musical instrument and classifying the instrument as one of 25 possibilities. The model is based on current models of signal processing in the human auditory system. It explicitly extracts salient acoustic features and uses a novel improvisational taxonomic architecture (based on simple statistical pattern-recognition techniques) to classify the sound source. The performance of the model is compared directly to that of skilled human listeners, using both isolated musical tones and excerpts from compact disc recordings as test stimuli. The computer model's performance is robust with regard to the variations of reverberation and ambient noise (although not with regard to competing sound sources) in commercial compact disc recordings, and the system performs better than three out of fourteen skilled human listeners on a forced-choice classification task. This work has implications for research in musical timbre, automatic media annotation, human talker identification, and computational auditory scene analysis.
by Keith Dana Martin.
Ph.D.
Beauvois, Michael W. "A computer model of auditory stream segregation." Thesis, Loughborough University, 1991. https://dspace.lboro.ac.uk/2134/33091.
Повний текст джерелаMak, Lin Chi Mechanical & Manufacturing Engineering Faculty of Engineering UNSW. "Non-Line-of-Sight localisation of a sound source." Awarded by:University of New South Wales. Mechanical & Manufacturing Engineering, 2009. http://handle.unsw.edu.au/1959.4/44702.
Повний текст джерелаJohansson, Anders M. "Acoustic sound source localisation and tracking : in indoor environments /." Karlskrona : Department of Signal Processing, Blekinge Institute of Technology, 2008. http://www.bth.se/fou/Forskinfo.nsf/allfirst2/beeb3e73884ff408c125744f00370a08?OpenDocument.
Повний текст джерелаJohansson, Anders. "Acoustic Sound Source Localisation and Tracking : in Indoor Environments." Doctoral thesis, Blekinge Tekniska Högskola [bth.se], School of Engineering - Dept. of Signal Processing, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-00401.
Повний текст джерелаLiao, Wei-Hsiang. "Modelling and transformation of sound textures and environmental sounds." Thesis, Paris 6, 2015. http://www.theses.fr/2015PA066725/document.
Повний текст джерелаThe processing of environmental sounds has become an important topic in various areas. Environmental sounds are mostly constituted of a kind of sounds called sound textures. Sound textures are usually non-sinusoidal, noisy and stochastic. Several researches have stated that human recognizes sound textures with statistics that characterizing the envelopes of auditory critical bands. Existing synthesis algorithms can impose some statistical properties to a certain extent, but most of them are computational intensive. We propose a new analysis-synthesis framework that contains a statistical description that consists of perceptually important statistics and an efficient mechanism to adapt statistics in the time-frequency domain. The quality of resynthesised sound is at least as good as state-of-the-art but more efficient in terms of computation time. The statistic description is based on the STFT. If certain conditions are met, it can also adapt to other filter bank based time-frequency representations (TFR). The adaptation of statistics is achieved by using the connection between the statistics on TFR and the spectra of time-frequency domain coefficients. It is possible to adapt only a part of cross-correlation functions. This allows the synthesis process to focus on important statistics and ignore the irrelevant parts, which provides extra flexibility. The proposed algorithm has several perspectives. It could possibly be used to generate unseen sound textures from artificially created statistical descriptions. It could also serve as a basis for transformations like stretching or morphing. One could also expect to use the model to explore semantic control of sound textures
Lee, Chung. "Sound texture synthesis using an enhanced overlap-add approach /." View abstract or full-text, 2008. http://library.ust.hk/cgi/db/thesis.pl?CSED%202008%20LEE.
Повний текст джерелаNoshino, Eri. "Sound-Intensity-Dependent Compensation for the Small Interaural Time Difference Cue for Sound Source Localization." Kyoto University, 2008. http://hdl.handle.net/2433/124080.
Повний текст джерелаNalavolu, Praveen Reddy. "PERFORMANCE ANALYSIS OF SRCP IMAGE BASED SOUND SOURCE DETECTION ALGORITHMS." UKnowledge, 2010. http://uknowledge.uky.edu/gradschool_theses/50.
Повний текст джерелаRamamurthy, Anand. "EXPERIMENTAL EVALUATION OF MODIFIED PHASE TRANSFORM FOR SOUND SOURCE DETECTION." UKnowledge, 2007. http://uknowledge.uky.edu/gradschool_theses/478.
Повний текст джерелаShub, Daniel E. (Daniel Eric) 1974. "The role of the precedence effect in sound source lateralization." Thesis, Massachusetts Institute of Technology, 2001. http://hdl.handle.net/1721.1/86768.
Повний текст джерелаKhan, Muhammad Salman. "Informed algorithms for sound source separation in enclosed reverberant environments." Thesis, Loughborough University, 2013. https://dspace.lboro.ac.uk/2134/13350.
Повний текст джерелаZantalis, Dimitrios. "Guided matching pursuit and its application to sound source separation." Thesis, University of York, 2016. http://etheses.whiterose.ac.uk/13204/.
Повний текст джерелаChapman, David P. "Playing with sounds : a spatial solution for computer sound synthesis." Thesis, University of Bath, 1996. https://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.307047.
Повний текст джерелаTinney, Charles E. "Low-dimensional techniques for sound source identification in high speed jets." Related electronic resource: Current Research at SU : database of SU dissertations, recent titles available full text, 2005. http://wwwlib.umi.com/cr/syr/main.
Повний текст джерелаRådsten-Ekman, Maria. "MAY NOISY SOUND ENVIRONMENTS BE IMPROVED BY ADDING PLEASANT WATER SOUNDS?" Thesis, Stockholms universitet, Psykologiska institutionen, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:su:diva-43834.
Повний текст джерелаSheaffer, J. "From source to brain : modelling sound propagation and localisation in rooms." Thesis, University of Salford, 2013. http://usir.salford.ac.uk/29210/.
Повний текст джерелаBraun, Michael E. "Sound source contributions for the prediction of vehicle pass-by noise." Thesis, Loughborough University, 2014. https://dspace.lboro.ac.uk/2134/16322.
Повний текст джерелаPreston, Giles Andrew. "Modelling sound source regions for the prediction of coaxial jet noise." Thesis, University of Southampton, 1995. https://eprints.soton.ac.uk/173779/.
Повний текст джерелаCobos, Serrano Máximo. "Application of sound source separation methods to advanced spatial audio systems." Doctoral thesis, Universitat Politècnica de València, 2010. http://hdl.handle.net/10251/8969.
Повний текст джерелаCobos Serrano, M. (2009). Application of sound source separation methods to advanced spatial audio systems [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/8969
Palancia
Zheng, Haosheng, and Kaichun Zhang. "Noise Analysis of Computer Chassis and Secondary Sound Source Noise Reduction." Thesis, Blekinge Tekniska Högskola, Institutionen för maskinteknik, 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-18547.
Повний текст джерелаZiemann, Astrid, and Kati Balogh. "Gekoppelter Atmosphäre-Boden-Einfluss auf die Schallausbreitung einer höher gelegenen Schallquelle." Universitätsbibliothek Leipzig, 2017. http://nbn-resolving.de/urn:nbn:de:bsz:15-qucosa-221985.
Повний текст джерелаThe licensing procedure for the construction of high-placed sound sources (e.g. wind power stations) demands to proof that no (significant and) harmful impact on environment is outgoing from these systems. Therefore, it is necessary to analyse the sound propagation of such a kind of sources. In this context one central problem has to be managed: the coupled effect of temperature, wind speed and wind direction profiles combined with the influence of surface on sound propagation. The temporally and spatially variable influence of the atmosphere is only insufficiently described by the operational models, especially in relation to long-time investigations of sound immission. Consequently, the aim of this study was to investigate the coupled effect of atmospheric and surface influence on sound propagation up to distances of 2 km away from the sound source using the model SMART (Sound propagation model of the atmosphere using ray-tracing)
Verviers, Claire Juliette. "The Influence Of Sound Properties On The Semantic Associations Of Product Sounds." Master's thesis, METU, 2010. http://etd.lib.metu.edu.tr/upload/12612412/index.pdf.
Повний текст джерелаBekkos, Audun. "Source Direction Determination with Headphones : An Adaptable Model for Binaural Surround Sound." Thesis, Norges teknisk-naturvitenskapelige universitet, Institutt for elektronikk og telekommunikasjon, 2012. http://urn.kb.se/resolve?urn=urn:nbn:no:ntnu:diva-18578.
Повний текст джерелаNakamura, Keisuke. "Robust Sound Source Localization Based on Microphone Array Processing for Robot Audition." 京都大学 (Kyoto University), 2013. http://hdl.handle.net/2433/174833.
Повний текст джерелаHummersone, Christopher. "A psychoacoustic engineering approach to machine sound source separation in reverberant environments." Thesis, University of Surrey, 2011. http://epubs.surrey.ac.uk/2923/.
Повний текст джерелаFulford, Ross Anthony. "Structure-borne sound power and source characterisation in multi-point-connected systems." Thesis, University of Liverpool, 1995. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.284262.
Повний текст джерелаWilkie, Sonia. "The effect of audio cues and sound source stimuli on looming perception." Thesis, Queen Mary, University of London, 2015. http://qmro.qmul.ac.uk/xmlui/handle/123456789/8974.
Повний текст джерелаItoyama, Katsutoshi. "Source Separation of Musical Instrument Sounds in Polyphonic Musical Audio Signal and Its Application." 京都大学 (Kyoto University), 2011. http://hdl.handle.net/2433/142116.
Повний текст джерелаWang, Cheng-Kang, and 汪正剛. "Multiple Sound Source Direction Estimation and Sound Source Number Estimation." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/23799296724314561439.
Повний текст джерела國立交通大學
電機與控制工程系所
96
This work proposes a multiple sound source direction estimation and sound source number estimation method by using distributed microphone array without the information of sound velocity. This work also proposes an eigenstructure-based generalized cross correlation method (ES-GCC) for estimating time delay between microphones. Upon obtaining the time delay information, the sound source direction and velocity can be estimated by least square method. In multiple sound source case, the time delay combination among microphones is arranged such that the estimated sound speed value falls within an acceptable range. By accumulating the estimation result of sound source direction and using adaptive K-means++ algorithm, we can estimate the sound source number and direction for each source. Experimental results are carried out in the real environment to evaluate the performance of the proposed algorithm.
Azarfar, Alireza. "Biological sound source localization." Master's thesis, 2012. http://hdl.handle.net/10400.1/10835.
Повний текст джерелаLin, Chi-Hao, and 林祺豪. "Probabilistic Structure from Sound and Sound Source Localization." Thesis, 2008. http://ndltd.ncl.edu.tw/handle/48001457364720395043.
Повний текст джерела國立臺灣大學
資訊工程學研究所
96
Auditory perception is one of the most important functions for robotics applications. Microphone arrays are widely used for auditory perception in which the spatial structure of microphones is usually known. The thesis first describes the affine Structure from Sound (SFS) algorithm. The structure from sound is a problem to simultaneously localize microphones and sound sources. However, the existing method does not take measurement uncertainty into account and does not provide uncertainty estimates of the SFS results. In this thesis, we propose a probabilistic structure from sound (PSFS) approach using the unscented transform. The PSFS algorithm not only localizes microphones and sound sources but also estimates the uncertainties of the SFS results. In addition, a probabilistic sound source localization (PSSL) approach using the PSFS results is provided to improve sound source localization accuracy. The ample results of simulation and experiments using low cost, off-the-shell microphones demonstrate the feasibility and performance of the proposed PSFS and PSSL approaches.
Lin, Chi-Hao. "Probabilistic Structure from Sound and Sound Source Localization." 2008. http://www.cetd.com.tw/ec/thesisdetail.aspx?etdun=U0001-2807200820013400.
Повний текст джерелаYang, Shan-hsiang, and 楊善翔. "Three-dimensional Sound Source Localization." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/00117977278704040317.
Повний текст джерела國立臺灣科技大學
資訊工程系
97
In this thesis, we study and implement a system to detect the direction of a sound source in three-dimensional space. For the hardware part, an equilateral triangle microphone array composed of only three microphones is used to input the voice signals. For the software part, VAD (Voice Activity Detection), TDOA (Time Delay of Arrival) estimation and direction detection are executed in order. In the processing of VAD, we propose a spectral-entropy plus SNR-verification based method to distinguish speech/non-speech frames. To estimate TDOA, an approximation algorithm is used to compute a generalized cross correlation function. We propose a synchronous phase replication method to solve the problem of unstable phase. In addition, we propose a parabolic interpolation based method to increase the accuracy of estimated TDOA values. Then, the distances between the estimated vector of TDOA values and the vectors of theoretical value are computed in order to find the direction of a sound source. Also, the accuracy is improved by using interpolation. Furthermore, we propose a weighted voting mechanism to determine the final direction angle from the angles obtained in several speech frames. According to the results of on-line experiments, our system can do real-time processing by using small amount of computations. The averaged error of azimuth angle is 3.43 degrees and the averaged error of elevation is 2.08 degrees. Therefore, the overall performance of our sound source localization system is good.
McNeese, Andrew Reed. "An investigation of the combustive sound source." Thesis, 2010. http://hdl.handle.net/2152/ETD-UT-2010-08-1699.
Повний текст джерелаtext
Lee, Po-Lin, and 李柏霖. "The Investigations on the Sound Field Generated by Moving Sound Source." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/83380965990684888265.
Повний текст джерела國立清華大學
動力機械工程學系
98
Abstract The Phenomenon of the sound field generated by a moving sound source has been investigated in the present work with two-part subject. The first part is to establish the mathematical model and the corresponding numerical scheme; the second part is to employ the established numerical scheme in practical acoustic problems. In order to establish the mathematical model, a novel governing equation is derived and it can be viewed as a modified Ffowcs Williams-Hawkings equation (FW-H-equation). The major characteristic of the novel governing equation is to include the interior and the exterior domain. In addition, the acoustic position vector, the quality describes the relations about the distance and the direction between the observer and the sound source, is represented for applying to the condition of the sound source moving with variable speed. As to the solution of the governing equation, it is expressed in the form of the Surface Integral Formula (SIF) by convoluting the free-space Green’s function in time domain. Then, this SIF is used to numerical implementation in the concept of the Boundary Element Method in time domain (BEMTD). After establishing the numerical scheme and verifying the correctness of the calculating results, two acoustic problems were investigated for revealing the ability of the numerical scheme. The first case considers that a moving line source with variable speed. This case reveals that the restriction of the constant speed is released in the established numerical scheme. For the simulation results, it shows that the effect of the variable speed not only influenced the variation rate of the frequency modulation, i.e., Doppler effect, but also the time about the maximum acoustic pressure being observed. In addition, the rate of the amplitude variation is shaper than that in the constant speed case when the line source is approaching to the observer point. The second case investigates the binaural hearing perceived by a moving sound source. For understanding the weighting of the eventful cues about perceiving the direction and the speed of the moving sound source, the sound pressure at the entrance of the external ear canal was calculated by the established numerical scheme. Furthermore, the Hilbert Huang transformation (HHT) is used to find the instantaneous frequencies of acoustic signal. Results show that the Interaural Level Difference (ILD) and the frequencies shifting are eventful than Interaural Time Difference (ITD). The perceived loudness level will be larger in the motional case than that found in the stationary case. These engineering problems are shown that the acoustic properties are different for comparing with the sound field generated by a moving sound source and that by a stationary sound source. As to the analytical methodology developed in the present work, it turns out that it indeed can be used to simulate and analyze these acoustic problems whenever the sound source moves or not. Furthermore, some meaningful phenomenon relating to these problems then can be observed through discussing the calculating results.
Ling, Hung, and 林宏. "Study Sound Source Localization Based on TDOA." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/33588336422359344240.
Повний текст джерела亞洲大學
資訊工程學系碩士班
97
Science and technology change with each passing day in recent years. The interface needs more accurate and convenient between machine and people, one of them used the microphone that respond to location of the sound source. This research used Microphone array to find the TDOA of sound source localization on Cross Correlation function. There are two methods of sound source localization. First kind of method is using TDOA on hyperbola. The other is using TDOA on neural network. The research perform simulation with MATLAB. Based on the simulation results ,the prove the feasibility of the research. And offer a reference of technical research in the future.
Wu, Xiang. "Binaural Sound Source Localisation in Complex Conditions." Phd thesis, 2020. http://hdl.handle.net/1885/205556.
Повний текст джерелаΚεττένης, Χρίστος. "Σύστημα εντοπισμού ηχητικής πηγής". Thesis, 2011. http://nemertes.lis.upatras.gr/jspui/handle/10889/4390.
Повний текст джерелаIn this diploma Thesis a sound source positioning system, was studied, designed and implemented. The sound, in particular, could be produced by "tapping a keyboard" onto a table surface or it is the noise produced while writing with a pencil or a piece of chalk, onto paper or on the surface of a blackboard, respectively. The aim of this application is to transform the surface of a desk or a blackboard into a "cheap" but effective electronic input device, in other, words an electronic tablet. Lastly, this Thesis is focused on the problems causing the reduction of accuracy in estimating the posistion of the acoustic source and also on the design and construction of the hardware and software that support the produced system.
Goel, Priyank. "Harmonic Sound Source Separation in Monaural Music Signals." Thesis, 2013. http://hdl.handle.net/2005/2803.
Повний текст джерелаChung, Yi-Hao, and 鍾毅豪. "A bi-sensor system for sound source estimation." Thesis, 1999. http://ndltd.ncl.edu.tw/handle/90880198860324681031.
Повний текст джерела淡江大學
資訊工程學系
87
Due to the great advance in the technique of microprocessor and the sharp drop of cost, digital signal processing applications are widely spread. In addition, the array signal processing technique has been vibrantly developed, no matter in noise deduction or in signal enhancement we have achieved certain degree of satisfactory result. Another interesting application of array signal processing is to use multi-sensors to estimate the signal source. In this thesis we investigate methods in array signal processing and implement a system for estimating the source location of a sound. Our system is a real-time embedded system implemented with Texas Instrument TMS320C542 StarterKit. Like all animals with two ears, we use two sensors (microphones) to receive input sound and estimate the location of the sound source. We shall apply all benefits from array signal processing in noise deduction, signal enhancement and sound source estimation in our system. We hope our system can be further developed a hearing aid system for human being.