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Dissertations / Theses on the topic 'Sound location'

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Published: 4 June 2021

Last updated: 6 September 2023

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Wood, K. C. "How is sound location represented in auditory cortex?" Thesis, University College London (University of London), 2015. http://discovery.ucl.ac.uk/1473329/.

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The present work tested two competing hypotheses about how the location of sounds in space is encoded by auditory cortex. The labelled-line hypothesis says that each azimuthal location is encoded by maximal firing of a specific small and sharply tuned population of neurons. The two-channel hypothesis says that a sound location is encoded by the relative activity of two populations of neurons with broad tuning and maximal activity at ± 90. To test these hypotheses a new behavioural task was developed in which subjects had to report the location of a target sound relative to a preceding reference. Models of the two-channel hypothesis and a modified version of the labelled-line hypothesis that accounted for better sound localisation precision at the midline, predicted best performance in the task around the midline with performance decreasing in the periphery whereas the labelled-line hypothesis predicted equal performance throughout space. Consistent with both the two-channel and modified labelled-line model, both ferret and human performance was best at the midline, highlighting the need for neural recordings in auditory cortex to distinguish between these models. The peaks of spatial receptive fields of neurons recorded from auditory cortex of ferrets performing the relative localisation task were distributed across the contralateral hemisphere, rather than clustered at 90 as predicted by the two channel model. Decoding of location from populations of neurons using two-channel or labelled-line maximum-likelihood decoders indicated that both decoders performed as well as ferrets localising sounds in the same testing chamber but that the labelled-line decoder out-performed the two-channel decoder. Finally, the necessity for an intact auditory cortex for sound localisation was confirmed after developing cortical cooling in the ferret as a method to reversibly silence areas of cortex during behaviour.

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Yao, Norikazu. "Auditory localisation : contributions of sound location and semantic spatial cues." Thesis, Queensland University of Technology, 2007. https://eprints.qut.edu.au/16504/1/Norikazu_Yao_Thesis.pdf.

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In open skill sports and other tasks, decision-making can be as important as physical performance. Whereas many studies have investigated visual perception there is little research on auditory perception as one aspect of decision making. Auditory localisation studies have almost exclusively focussed on underlying processes, such as interaural time difference and interaural level difference. It is not known, however, whether semantic spatial information contained in the sound is actually used, and whether it assists pure auditory localisation. The aim of this study was to investigate the effect on auditory localisation of spatial semantic information. In Experiment One, this was explored by measuring whole body orientation to the words &quotLeft", &quotRight", &quotBack", &quotFront" and &quotYes", as well as a tone, each presented from left right, front and back locations. Experiment Two explored the effect of the four spatial semantic words presented either from their matching locations, or from a position rotated 20 degrees anticlockwise. In both experiments there were two conditions, with subjects required to face the position indicated by the sound location, or the meaning of the word. Movements of the head were recorded in three dimensions with a Polhemus Fastrak system, and were analysed with a custom program. Ten young adult volunteers participated in each experiment. Reaction time, movement time, initial rotation direction, rotation direction at peak velocity, and the accuracy of the final position were the dependent measures. The results confirmed previous reports of confusions between front and back locations, that is, errors about the interaural axis. Unlike previous studies, many more back-to-front than front-toback errors was made. The experiments provided some evidence for a spatial Stroop interference effect, that is, an effect on performance of conflicting information provided by the irrelevant dimension of the stimulus, but only for reaction time and initial movement direction, and only in the Word condition. The results are interpreted using a model of the processes needed to respond to the stimulus and produce an orienting movement. They suggest that there is an asymmetric interference effect in which auditory localisation can interfere with localisation based on semantic content of words, but not the reverse. In addition, final accuracy was unaffected by any interference, suggesting that these effects are restricted to the initial stages of response selection.

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Yao, Norikazu. "Auditory localisation : contributions of sound location and semantic spatial cues." Queensland University of Technology, 2007. http://eprints.qut.edu.au/16504/.

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Tollin, Daniel Joshua. "Some aspects of the lateralization of echoed sound in man." Thesis, University of Oxford, 1997. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.363729.

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Matzumoto, Andres Esteban Perez. "A study of microphone arrays for the location of vibrational sound sources." Thesis, University of Southampton, 1991. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.305576.

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Kim, Hyungtai. "Modeling of micro-spatial employment location patterns and its application to the Puget Sound Region : count and choice approaches /." Thesis, Connect to this title online; UW restricted, 2006. http://hdl.handle.net/1773/10826.

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Lopez, Poveda Enrique Alejandro. "The physical origin and physiological coding of pinna-based spectral cues." Thesis, Loughborough University, 1996. https://dspace.lboro.ac.uk/2134/32318.

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This thesis investigates both the physical origin and the physiological coding of pinna-based spectral features observed in experimental Head-Related Transfer Functions (HRTFs). An experimental method for measuring HRTFs using a Knowles Electronics Manikin for Acoustic Research (KEMAR) is presented. The method includes a technique for moulding individualised pinnae to be fitted on to the KEMAR's head. Experimental HRTF data obtained with this method are shown and analysed. The most remarkable pinna-based spectral features are identified and their dependency on source location is characterised, particularly for elevation-dependent spectral notches.

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Ramamurthy, Anand. "EXPERIMENTAL EVALUATION OF MODIFIED PHASE TRANSFORM FOR SOUND SOURCE DETECTION." UKnowledge, 2007. http://uknowledge.uky.edu/gradschool_theses/478.

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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.

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McConaghy, Nicholas Ralph. "Exploring environmental space through sound – compositional relationships across external location, internal structure and environmentally mediated spaces." Thesis, The University of Sydney, 2021. https://hdl.handle.net/2123/25026.

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This thesis demonstrates an approach to composing with environmental sound that uses spatial concepts and techniques to bridge the aesthetic divide between reductive and relative compositional philosophies. It responds to the complex issues of contextual integration and separation through a detailed compositional approach, which uses the spatial medium of acousmatic music to translate aspects of acoustic ecology, biomimetics, soundscape and ecoacoustics into the domain of contemporary composition. This research makes claims to new knowledge by examining how compositional activities inform the development of novel techniques for environmental sound composition. It takes a practice-based approach to research, which uses the creative practices of field recording, fixed-media composition and software programming to highlight and respond to the issues implicit to the production of environmentally mediated spaces through sound. Furthermore, it provides new theoretical perspectives on the relations between musical form and the external environment. Central to this research is a body of creative work, which presents a portfolio of compositions and the custom software tools integral to its production as research outcomes. As the activities of practice and the insights gained through practice are as crucial to the practice-based research paradigm as its outcomes, this thesis uses a self-reflective approach to document how the knowledge generated during the composition process shaped the outcomes of the creative artefacts. An engagement with spatial concepts characterises the reflective discourse. The non-linear and iterative ways of shaping and applying these ideas in practice underpin the discussion of each composition and the algorithmic realisation of these concepts using the SuperCollider programming language.

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Meyer, Michaela. "Peripheral neural coding strategies for spectral analysis and sound source location in the non-teleost bony fish, Acipenser fulvescens." College Park, Md.: University of Maryland, 2008. http://hdl.handle.net/1903/8147.

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Thesis (Ph. D.) -- University of Maryland, College Park, 2008. Thesis research directed by: Dept. of Biology. Title from t.p. of PDF. Includes bibliographical references. Published by UMI Dissertation Services, Ann Arbor, Mich. Also available in paper.

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Kim, Sungyop. "Neighborhood residential location choice of the elderly : a study of the elderly in the Puget Sound region of Washington /." Thesis, Connect to this title online; UW restricted, 2004. http://hdl.handle.net/1773/10818.

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Bassam, Abdulhamid Ansam, and Gill Namra Jamshaid. "A proof-of-concept of the audio tour guide application, SoundTracker, aimed at friends and family." Thesis, Malmö universitet, Fakulteten för teknik och samhälle (TS), 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:mau:diva-20890.

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Kontextmedvetenhet kan användas i turistguide applikationer för att bidra användare med information och tjänster. Majoriteten av turistguide applikationer utvecklas vanligtvis för utbildnings eller historiska ändamål. Den här studien presenterar en konceptvalidering av den opublicerade ljuvandrings applikationen "SoundTracker", i syfte att förbättra den. Den nya "SoundTracker" prototypen inriktar sig inte på utbildnings eller historiska ändamål, utan den är inriktad på inspelning, uppspelning, och delning av personliga ljudvandringar med vänner och familj. Konceptvalideringen är begränsad av de definierade forskningsfrågorna i den här uppsatsen. Det handlar i korta drag om en förbättring av noggranheten på de inlästa GPS koordinaterna från en användares position, för att förse användaren med en trevlig användarupplevelse. Dessutom är grafiska användargränssnittet av den originala prototypen förbättrat, och testat genom webb enkäter. Vad gäller delnings funktionaliteten, en test applikation är skapad för att hitta essentiella aspekter som behöver tas hänsyn till i den nya "SoundTracker" prototypen när en användare vill dela ett ljudspår med vänner och familj. Test applikationen testas baserat på samlade svar genom en semi strukturerad interview på människor i åldersgruppen 20-30 år.De erhållna resultaten indikerar på att genom användning av Kalman filter, förbättras noggrannheten på användarens position, vilket resulterar i en ljudvandring med mindre avvikelser jämfört med en GPS-sensor. Vad gäller det förbättrade grafiska användargränssnittet, det var enklare för användarna att förstå den förbättrade prototypen såväl som navigera igenom den, än fallet med den originala prototypen. Användarna finner det även intressant när en delningsfunktion implementeras med designen som diskuteras i det här arbetet. Context-awareness can be used in tour guide applications in order to provide users with information and services. The majority of tour guide applications are usually developed for educational or historical review purposes. This paper presents a proof-of-concept of the unpublished audio tour guide application, “SoundTracker”, with the aim of improving it. The new “SoundTracker” prototype does not aim for educational or historical review purposes, instead it is aimed for recording, listening and sharing personal audio tour guides with friends and family. The proof-of-concept is limited by the defined research questions found in this paper. In summary, the accuracy of the read-in GPS coordinates of a user’s position is enhanced with Kalman filter, to give an enjoyable user experience. Additionally, the graphical user interface of the original prototype is improved and tested through web-surveys. As for the sharing functionality, a test application is created in order to find what necessary aspects need to be considered in the new "SoundTracker" prototype when a user wants to share a sound-walk with friends and family. The test application is tested based on responses obtained through a semi-structured interview from people in the age of 20-30 years.The obtained results indicate that, with use of Kalman filter, the location accuracy of the user is enhanced, which results in a sound-walk with less deviations compared to location accuracy of only GPS-sensor. Regarding the improved graphical user interface, users found the new "SoundTracker" prototype easier to understand, as well as to navigate through it, than the case in the original prototype. Users also find it interesting when a sharing function is implemented with the design that is discussed in this work.

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Catling, Aaron. "The Ending Needs Work AKA the Good, the Bad and the Ugly of being an independent filmmaker in Australia." Thesis, Queensland University of Technology, 2005. https://eprints.qut.edu.au/16091/1/Aaron_Catling_Thesis.pdf.

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Over the period of candidature, write and direct a feature film to completion. Furthermore, undertake a thorough reflective phase which involves the analysis of each aspect relating to those key components, writing and directing. Through this form of creative practice and utilising state of the art digital filmmaking techniques it is hoped that an addition to knowledge will be achieved.

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Catling, Aaron. "The Ending Needs Work AKA the Good, the Bad and the Ugly of being an independent filmmaker in Australia." Queensland University of Technology, 2005. http://eprints.qut.edu.au/16091/.

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Wei, Wei. "Underwater measurement of the sound-intensity vector : its use in locating sound sources, and in measuring the sound power of stationary and moving sources /." Full text available from ProQuest UM Digital Dissertations, 1994. http://0-proquest.umi.com.umiss.lib.olemiss.edu/pqdweb?index=0&did=1296083321&SrchMode=1&sid=4&Fmt=2&VInst=PROD&VType=PQD&RQT=309&VName=PQD&TS=1268675088&clientId=22256.

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Thesis (Ph.D.)--University of Mississippi, 1994. Typescript. "May 1994 ." Dissertation director: Dr. Robert Hickling Committee chair: Dr. Richard Raspet Includes bibliographical references (leaves 115-118). Also available online via ProQuest to authorized users.

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Blomgren, Johannes. "SOUNDSTAGE : Miljöförstärkt ljud i smartphone." Thesis, Linköpings universitet, Institutionen för datavetenskap, 2014. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-105081.

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Det här arbetet syftar till att genom design av en telefonapplikation undersöka hur ljud kan förstärkas av den omgivande miljön. Arbetet visar på hur miljöförstärkt ljud skiljer sig från ljudförstärkt miljö samt vilka begränsningar det innebär att använda en vanlig smartphone som plattform. Ett långsiktigt mål är att genom en produkt som låter oetablerade ljudproducenter distribuera sina alster i den offentliga miljön främja fri kultur och kreativitet. Arbetet har följt principen för RtD (research through design), där designprocessen utgör det fall som studeras och dokumentationen av den utgör den bas av data som analyseras och leder till slutsatser kring ämnet. Designprocessen i detta arbete innebär en mindre omfattande användarstudie i enkätform för att identifiera målgrupp, krav och mål. Följd av en andra studie där en prototyp framtagen med grund i resultaten från användarstudien utvärderas av deltagare genom observation, intervju och reaktionskort. De största utmaningarna för designen grundar sig i att ta fram en applikation som inte stör upplevelsen, men samtidigt kan instruera användaren i hur man använder applikationen. Utifrån arbetet dras slutsatsen att användarinstruktioner, såväl visuella som auditiva, löper större risk att störa upplevelsen i miljöförstärkt ljud än i ljudförstärkt miljö. Vidare konstateras att den största bristen i en lösning som bygger på standardutrustning i jämförelse med en lösning som nyttjar dedikerad utrustning utgörs av begränsningar när det kommer till att ange riktning med hjälp av ljudets rumslighet.

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Schwartz, Andrew H. "Effect of dynamic range compression on attending to sounds based on spatial location." Thesis, Massachusetts Institute of Technology, 2013. http://hdl.handle.net/1721.1/84411.

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Thesis (Ph. D. in Speech and Hearing Bioscience and Technology)--Harvard-MIT Program in Health Sciences and Technology, 2013. Cataloged from PDF version of thesis. Includes bibliographical references (pages 118-125). Many hearing aids introduce nonlinear compressive gain to accommodate the reduced dynamic range that often accompanies hearing loss. Unfortunately, when applied independently at either ear, this gain can introduce fluctuations in interaural level difference (ILD), which is an important cue for spatial perception and attending to sounds in an acoustic mixture. Moreover, natural sounds produce complicated interactions between different sounds in a mixture, as a compressor's gain is driven by whichever source dominates the mixture within a specified temporal window. While independent compression can interfere with spatial perception of sound, it does not always interfere with localization accuracy or speech identification. This thesis investigates the role of dynamic range compression on the ability to attend to target speech in the presence of interfering speech. First, the fundamental concepts behind dynamic range compression and its use are introduced, and used to develop a framework to understand some of the possible effects on ILD and spatial perception. This framework is applied toward the interpretation of the existing literature regarding dynamic range compression and spatial perception, bringing together a seemingly contradictory range of results. In particular, the framework presented here predicts that dynamic range compression will only affect performance in tasks for which relatively small spatial separations are tested, whereas many existing studies compare only large spatial separations to no spatial separation. We describe and analyze the results of an experiment designed to test this prediction by systematically varying the spatial separation between different speech sources that normal-hearing listeners attended to. We found a robust but modest detrimental effect of dynamic range compression on listeners' performance. Linking the left and right compressors so that ILD was unaltered restored performance. Lastly, we develop a model to describe the utility of ILD for such tasks. The results of this model provide insight into the reported behavioral results, and generate predictions for how hearing impairment may alter the observed pattern of results. by Andrew H. Schwartz. Ph.D.in Speech and Hearing Bioscience and Technology

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Reed, Chelsea Clarke. "Locating Philadelphia Jazz: The Intersections of Place, Sound, and Story in the Classroom." Master's thesis, Temple University Libraries, 2018. http://cdm16002.contentdm.oclc.org/cdm/ref/collection/p245801coll10/id/493556.

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History M.A. This study explores a place based pedagogy of Philadelphia jazz history for K-12 students. While many intersections exist between place based programming and jazz public history both nationally and locally, the Philadelphia jazz public history community does not focus on educational programming. Though centered in Philadelphia, this study includes educational materials and field research for both formal and informal educators to increase critical, interdisciplinary African American musical history content in the classroom. The lesson plans found within exemplify a cross section of social studies educational literature, the history of African American narratives in Philadelphia schools, and place based jazz history in the city. Temple University--Theses

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Bradley, Simon. "Archaeology of the voice : exploring oral history, locative media, audio walks, and sound art as site-specific displacement activities." Thesis, University of Huddersfield, 2016. http://eprints.hud.ac.uk/id/eprint/28316/.

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This thesis develops a notion of an archaeology of the voice that is situated between three principal areas of research and practice: oral history, locative media, and sound art. The research takes place in the context of contested urban space in Holbeck, Leeds one of the most deprived neighbourhoods in the U.K. Through a reiterative and reflexive process of extensive interviewing, soundwalking and field recording the area is deep mapped and material gathered in order to produce a percipient led sitespecific presentation of oral history I term 'phonoscape'. Although the technology exists to connect oral history to place via locative media within a database aesthetic, a practical and conceptual gap is identified between these technologies for those working with audio interview material. In this context a purpose-built app is developed to enable oral history audio archives to be distributed geospatially, becoming navigable aurally on foot. In order to distribute a polyvocal sampling of an archive in time-space, techniques and principles from contemporary sound art are introduced, in particular a form of field composition involving an understanding of constitutive silence, soundscape, and voice editing techniques. Research into contemporary audio walk and memoryscape practice confirms that non-linear, fragmented narrative forms are used the construction of polyvocal understandings of place, and this is taken forward within a conception of the embodied hypertextual affordance of locative technology. The findings are then brought together in a transdisciplinary manoeuvre that introduces Displacement Activities, a translocational form of site-specific participatory performance art, providing a public vehicle that draws attention to phonoscape, its oral history content, and the archive itself. As an open work that is generative and reflexive, Displacement Activities extend the notion of site-specificity, finding global analogues before returning to the original site to begin the work again.

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阮筱喬. "The Sound Location Tracking System with Self-Adjusting." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/53000283374762823353.

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Li, Tsung-Hsien, and 李宗憲. "Studies of Resolution for Sound Source Location Using a Microphone array." Thesis, 2003. http://ndltd.ncl.edu.tw/handle/51242969441979773924.

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碩士 大同大學 資訊工程研究所 91 In this thesis, we discuss the correlation between frequency, and position of sound signal and interval between microphones. For a given requirement and signal characteristics, we try to provide some guide to line to make optimum specification then get the best resolution. For the sound source localization, the delay of arrival (DOA) technique has been proposed few decades. The technique detects the signal wave delay in microphone pairs and use the delay time to compute the sound source direction.The cross-power spectrum phase (CSP) analysis and first wavefront detected method are usually used in DOA technique. The CSP analysis use the signal's phase different of microphone pair to calculate the sound source direction. The wavefront detected method is obtained the results by detecting the first wavefront. In the experiment, we adjust the sound's signal frequency, angle and interval between two microphones. The results would exhibit the interactive among these factors. From these results, some guide lines to decide some parameters of sound source location are given.

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Chang, Jun-wei, and 張竣惟. "Vibration and Sound Radiation of Rectangangular Composite Panels stiffened at Different Location." Thesis, 2011. http://ndltd.ncl.edu.tw/handle/31092538227289440062.

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碩士 國立交通大學 機械工程學系 100 This paper is focused on the sound radiation behavior of stiffened sound radiation panels for flat-panel loudspeaker. If the sound radiation panel is not properly design, it is easy for a flat-panel speaker to have a significant sound pressure drop in the mid-frequency range, which may affect the sound quality of the speaker. To suppress or even eliminate the sound pressure drop, this thesis attempts to enhance the rigidity of the plate by using an appropriate stiffening method. There are two parts in this thesis. In the first part, a Rayleigh-Ritz Method is constructed to analyze the sound radiation behavior of stiffened sound radiation panels. The results obtained using the proposed Rayleigh-Ritz Method are compared with those obtained from the finite element software ANSYS. It has been shown that the natural frequencies and the sound pressure level (SPL) curves produced by the proposed method are in good agreement with those produced by ANSYS or experiments. In the second part, ANSYS is used to find the appropriate stiffening pattern for designing the sound radiation panel and the ideal material constants of the panel so that the major sound pressure drop in the mid-frequency range can be suppressed.

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Chun-Fu, Chang, and 張春富. "Development of a Sound Source Location and Identification Technique for Oil Immerged Power Transformers." Thesis, 2004. http://ndltd.ncl.edu.tw/handle/09756315345316253531.

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碩士 國立臺灣大學 工程科學與海洋工程學系 92 In consideration of the imperious requirement in promoting efficiency and in-serve detection on load tap changers (OLTC) in large scale oil-immerged power transformers, an acoustic based on-line detection, failure source location and patterns recognition technique is developed in this article. To determine the wave mode change, velocity and propagating path of acoustic wave which propagate from the source to the outer wall of transformers and the Lamb wave is dispersive phenomena have to confirm before, to take for source location technique a basis. In this article, further to confer on load tape changers action the circumstances, to correct the propagating paths to use the point detection to match up the wavelet transform based time-frequency analysis, source location is applied. In part of source patterns recognition, first, using a hydrophone to measure different source waveform, to proceed the waveform autocorrelation coefficients, reflection coefficients, then to train and recognition based the Back Propagation Network (BPN), and to establish the database on the model. Second to reconstitute the source waveform, using a sensor and the source geometry relationship, to appoint the concept of Leaky Lamb wave to compute the dispersive curve, the relationship between the incident angle, wave velocity and frequency was found, then it’s easy to know the attenuation and time delay could be computed, and to invert waveform. Finally, use the inversion source waveform to compare the species with a hydrophone measure waveform database, to accomplish the source recognition module in transformers. From the result of experiments that input a hydrophone waveform, the recognition result to reach satisfied, AE signal measured outside the transformer wall to inversion source waveform to compare the database, wolfram peak to peak, a mechanism action have effect from inversion Source, and for classifying the source species in large scale oil-immerged power transformers. The study concludes that source location and patterns recognition is to confirm and verify from experiment simultaneously.

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Leonardo, Ricardo Miguel Pontes. "Contextual Information based on Pervasive Sound Analysis." Master's thesis, 2018. http://hdl.handle.net/10362/36897.

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In recent times, there has been a continuous increase in the ubiquity, processing power and sensing capabilities of modern smartphones. This has made possible the emergence of new technologies that allows users to keep track of information regarding their health, activities and location, even in indoor places were GPS signal is not available. These technologies typically rely on fusing and processing information coming from multiple sensors, such as the accelerometer or the magnetometer. This thesis proposes a framework for indoor location and activity recognition from new source of information: the sound perceived through the device’s microphone. It does so by extracting information relative to the user’s position and activities through machine learning and audio processing techniques. In the context of indoor location, the proposed SoundSignature algorithm allows the device to learn from labeled data and predict the location it is in. These locations may be different rooms or distinct regions of large places, such as open spaces. Another proposed algorithm, SoundSimilarity, further refines this positioning by comparing the sound signals from two or more devices in real time. A novel audio similarity metric identifies if the devices are close to one another, mitigating the potential errors of the SoundSignature algorithm. This also has many other use cases, such as detecting proximity between the user and devices. Finally, the Activity Monitoring algorithm allows the device to learn from labeled data to recognize the activity the user is performing. This information may be also used to further refine the location algorithm by recognizing location-dependent activities such as the closing of doors or watching television.

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Lee, Chen-hsi, and 李承熹. "The impact of Acoustic and Sound Quality caused by location of impeller rib in blower." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/80188223186410441768.

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碩士 國立中山大學 機械與機電工程學系研究所 98 Abstract The advances of modern technology , creating a CPU and hardware speed greatly enhance the relative increase in number of work- ing temperature , so as not to make parts for high temperature, shortened life expectancy , we must take active cooling can be resolved , is gener- ated by the use of forced convection fan the heat away , but the fan noise when running , although not damage the hearing , have the potential to affect users of emotions ; as people increasingly rely on the computer, the fan gradually pay attention to the noise problem , so fans of R & D in recent years , in addition to increasing thermal efficiency , the increasing demands for noise. This article will examine the main centrifugal fan to AIO computer system for the experiment; AIO computer is the host and screen into one of the computer system, characterized by the internal space is small, heat is not easy, so the general would choose as the main centrifugal fan cooling fan, but the disadvantage of centrifugal fan noise is high, so this will do for the improvement of centrifugal fan noise research; to change the location of support ribs as the experimental setting, experimental method consists of three steps; first: Analog and down air inlet differences in velocity distribution, the second: the actual measurement centrifugal fan, is divided into single and measurement noise into the system and the sound quality of the judge, the third: performance testing; study showed that support rib in the middle of the design, the noise about 10% area reduction, sound quality also improved the effectiveness of nearly 50%, performance is also about 10% of the upgrade, the support rib in the middle of the design for double inlet centrifugal fans, for the better design.

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Teófilo, Ana Filipa Frazão de Almeida. "Low-Cost Indoor Localisation Based on Inertial Sensors, Wi-Fi and Sound." Master's thesis, 2021. http://hdl.handle.net/10362/130418.

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The average life expectancy has been increasing in the last decades, creating the need for new technologies to improve the quality of life of the elderly. In the Ambient Assisted Living scope, indoor location systems emerged as a promising technology capable of sup porting the elderly, providing them a safer environment to live in, and promoting their autonomy. Current indoor location technologies are divided into two categories, depend ing on their need for additional infrastructure. Infrastructure-based solutions require expensive deployment and maintenance. On the other hand, most infrastructure-free systems rely on a single source of information, being highly dependent on its availability. Such systems will hardly be deployed in real-life scenarios, as they cannot handle the absence of their source of information. An efficient solution must, thus, guarantee the continuous indoor positioning of the elderly. This work proposes a new room-level low-cost indoor location algorithm. It relies on three information sources: inertial sensors, to reconstruct users’ trajectories; environ mental sound, to exploit the unique characteristics of each home division; and Wi-Fi, to estimate the distance to the Access Point in the neighbourhood. Two data collection protocols were designed to resemble a real living scenario, and a data processing stage was applied to the collected data. Then, each source was used to train individual Ma chine Learning (including Deep Learning) algorithms to identify room-level positions. As each source provides different information to the classification, the data were merged to produce a more robust localization. Three data fusion approaches (input-level, early, and late fusion) were implemented for this goal, providing a final output containing complementary contributions from all data sources. Experimental results show that the performance improved when more than one source was used, attaining a weighted F1-score of 81.8% in the localization between seven home divisions. In conclusion, the evaluation of the developed algorithm shows that it can achieve accurate room-level indoor localization, being, thus, suitable to be applied in Ambient Assisted Living scenarios. O aumento da esperança média de vida nas últimas décadas, criou a necessidade de desenvolvimento de tecnologias que permitam melhorar a qualidade de vida dos idosos. No âmbito da Assistência à Autonomia no Domicílio, sistemas de localização indoor têm emergido como uma tecnologia promissora capaz de acompanhar os idosos e as suas atividades, proporcionando-lhes um ambiente seguro e promovendo a sua autonomia. As tecnologias de localização indoor atuais podem ser divididas em duas categorias, aquelas que necessitam de infrastruturas adicionais e aquelas que não. Sistemas dependentes de infrastrutura necessitam de implementação e manutenção que são muitas vezes dispendiosas. Por outro lado, a maioria das soluções que não requerem infrastrutura, dependem de apenas uma fonte de informação, sendo crucial a sua disponibilidade. Um sistema que não consegue lidar com a falta de informação de um sensor dificilmente será implementado em cenários reais. Uma solução eficiente deverá assim garantir o acompanhamento contínuo dos idosos. A solução proposta consiste no desenvolvimento de um algoritmo de localização indoor de baixo custo, baseando-se nas seguintes fontes de informação: sensores inerciais, capazes de reconstruir a trajetória do utilizador; som, explorando as características dis tintas de cada divisão da casa; e Wi-Fi, responsável pela estimativa da distância entre o ponto de acesso e o smartphone. Cada fonte sensorial, extraída dos sensores incorpora dos no dispositivo, foi, numa primeira abordagem, individualmente otimizada através de algoritmos de Machine Learning (incluindo Deep Learning). Como os dados das diversas fontes contêm informação diferente acerca das mesmas características do sistema, a sua fusão torna a classificação mais informada e robusta. Com este objetivo, foram implementadas três abordagens de fusão de dados (input data, early and late fusion), fornecendo um resultado final derivado de contribuições complementares de todas as fontes de dados. Os resultados experimentais mostram que o desempenho do algoritmo desenvolvido melhorou com a inclusão de informação multi-sensor, alcançando um valor para F1- score de 81.8% na distinção entre sete divisões domésticas. Concluindo, o algoritmo de localização indoor, combinando informações de três fontes diferentes através de métodos de fusão de dados, alcançou uma localização room-level e está apto para ser aplicado num cenário de Assistência à Autonomia no Domicílio.

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Karthik, Girija Ramesan. "Binaural Source Localization using subband reliability and interaural time difference patterns." Thesis, 2018. https://etd.iisc.ac.in/handle/2005/5189.

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Machine localization of sound sources is necessary for a wide range of appli- cations, including human-robot interaction, surveillance and hearing aids. Robot sound localization algorithms have been proposed using microphone arrays with varied number of microphones. Adding more microphones helps increase the localization performance as more spatial cues can be obtained based on the number and arrangement of the microphones. However, hu- mans have an incredible ability to accurately localize and attend to target sound sources even in adverse noise conditions. The perceptual organi- zation of sounds in complex auditory scenes is done using various cues that help us group/segregate sounds. Among these, two major spatial cues are the Interaural time difference (ITD) and Interaural level/intensity difference(ILD/IID). Popular algorithms, for binaural source localization, model the distributions of ITD and ILD in each frequency subband us- ing Gaussian Mixture Models (GMMs) and perform likelihood integration across the time-frequency plane to estimate the direction of arrival (DoA) of the sources. In this thesis, we use ITDs and show that the localization performance of a GMM based scheme varies across subbands. We pro- pose subband selection and subband weighting schemes in order to exploit the subband reliability for localization. Source localization results demon- strate that the proposed schemes perform better than uniformly weighing all subbands. In particular, the best set of weights closely correspond to the case of selecting only the most reliable subband. We also propose a new binaural localization technique in which templates, that capture the direction-speci c interaural time di erence patterns, are used to localize sources. These templates are obtained using histograms of ITDs in each subband. DoA is estimated using a template matching scheme, which is experimentally found to perform better than the GMM based scheme.

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Jhang, Jyun Jia, and 張峻嘉. "Sound Radiation of Composite Panels Excited at Different Locations." Thesis, 2011. http://ndltd.ncl.edu.tw/handle/61908997894239105857.

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碩士 國立交通大學 機械工程學系 100 This thesis studies the sound radiation of strip-type flat plates excited at different locations. The purpose of exciting the plate at different locations is two-fold: Raise the sound pressure level (SPL) and improve the smoothness of the SPL curve. This study consists of three parts. In the first part, a Rayleigh-Ritz method is constructed to study the vibration of sound radiation plates which comprising a flat plate, a flexible surround, and a number of voice coils. In the mathematical model of the sound radiation plate, the vibration of the plate is formulated on the basis of the classical lamination theory. The flexible surround is treated as spring. The voice coils are modeled by simple beams. The accuracy of the method is verified by the finite element code ANSYS. In the second part, the vibration response of the plate is used in the first Rayleigh integral to construct the SPL curve of the plate. The experimental SPL curves as well as those obtained using ANSYS are used to validate the suitability of the present method for the analysis of sound radiation plates. Finally, the effects of excitation locations on the SPL curves of sound radiation plates are studied. The more appropriate locations of the voice coils are determined to minimize the magnitude of the sound pressure level drop in the mid-frequency range for the plates. The effects of the material constants of the sound radiation plates on the SPL curves are also studied theoretically. The results obtained in this thesis should be useful for the design of flat-panel speakers.

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Wei, Hao-Shiang, and 魏浩翔. "Sound Radiation of Long Rectangular Plates Excited at Multiple Locations." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/25515343849845066476.

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碩士 國立交通大學 機械工程學系 99 This thesis aims to investigate the effects of exciting location on the sound radiation behavior of long flat-panel loudspeakers. In response to the current trend of flat electronic products, the speakers need to be long and narrow. When the speaker becomes narrow, the small width of the radiating plate will limit the size of the conventional circular transducer. The sound pressure level (SPL) of the speaker will become low and a major SPL dip in the mid frequency range may be incurred. In this thesis, the contents are divided into two parts. In the first part, a track-shape exciter is designed. The air-gap magnetic flux densities for different dimensions of transducer components are determined to find the optimal exciter design. Next, establish the magnetic flux database to provide a fast magnetic flux estimation method to reduce the large amount of computer time. The second part is about the use of the track-shape transducer to excite plates with different aspect ratios. The optimum transducer parameters for obtaining maximum SPL or minimum major dip in mid frequency range are determined. A number of loudspeakers are fabricated and tested to verify the proposed design procedure. Finally, a set of design guidelines for improving the SPL curves of different long rectangular flat speakers is proposed.

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Ho, Li-Wen, and 何立文. "Using Generalized Gaussian Mixture Model to Detect Sound Locations of Unknown Number of Sources for Sound Segregation." Thesis, 2012. http://ndltd.ncl.edu.tw/handle/01698459290351194610.

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碩士 國立交通大學 電信工程研究所 100 In this thesis, we propose a method to separate speech signals from spectrograms of sound mixtures with unknown number of sources. Recently, many sparse source separation algorithms using time-frequency masking have been proposed. However, most of these algorithms demand a known number of mixed sources in advance, which is not convenient in practice. In our proposed method, we first model the histogram of estimated angles of the direction of arrival (DOA) with a generalized Gaussian mixture model (GGMM) for detecting the number of sources and sound locations. The GGMM parameters are estimated using the expectation-maximization (EM) algorithm. Based on DOA information of each time-frequency (T-F) unit of the mixed spectrogram, a DOA-weight mask is estimated for each speech signal. The spectrogram of each speech signal is then extracted using the corresponding mask. In our simulations, speech signals are given different delays and amplitude at two microphones to produce DOA information for different locations. In addition to the generalized Gaussian distribution, the Gaussian distribution and the Laplace distribution are also investigated in modeling the DOA histogram. Two kinds of masks, the binary mask and the DOA-weight mask, are investigated in segregating signals from the mixture. Simulation results are compared with outputs of NPCM and DOA-NPCM. Results show that all methods perform equivalently in tests of spatial resolution. On the other hand, the NPCM, DOA-NPCM and GGMM have higher accuracy in estimating the DOA. For segregation, DOA-weight mask performs the best in most test conditions.

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Chen, Yu-Ting, and 陳右庭. "Estimation of the Multiple Sound Source Locations Using a Microphone Array." Thesis, 2014. http://ndltd.ncl.edu.tw/handle/39874458413941779293.

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碩士 國立高雄應用科技大學 電機工程系博碩士班 102 In this thesis, we use a microphone array with blind source separation algorithm and time difference of arrival (TDOA) in far-situation to do multiple sound source localization and source number estimation. Most of the researches of multiple sound source localization are focused on the angle of source. However, there is a problem that the user cannot identify the corresponding angle of each sound source, and we name this problem as data association. To solve this problem, we use nonnegative matrix factor 2-D deconvolution (NMF2D) to separate the mixed single and then we can consider the multiple sound source localization as many independent sound sources localization. Therefore, we can calculate the corresponding angle of each sound source. We also use the interaural level difference (ILD) and TDOA to estimate the distance of each sound source. Finally, the multiple sound source localization system is programmed by LabVIEW 2012 software.

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Chen, Yen Sheng, and 陳彥昇. "The correlation between the snore sound pressure and the locations of snoring." Thesis, 2009. http://ndltd.ncl.edu.tw/handle/44163616425334573203.

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碩士 長庚大學 醫療機電工程研究所 97 When people fall asleep, the airway would become narrow due to the reduction of muscle tension, and then the snoring would occur when the air passed through the narrow airway. In recent years, the uvulapalatopharyngoplasty was the most common treatment for reducing snoring. However, the accurate part for surgery and the improvement in snoring for patients after surgery should be considered at pre-op. Thus, the correlation between the BMI value and the snoring sound frequency of snoring part would be discussed in this study. The sleep partner’s perception of sleep affected by snoring sound pressure level with head shadow effect would also be discussed. Based on the results, the mean sound pressure level of snoring was 36.58 dBSPL by palate, 23.49 dBSPL by epiglottic and 16.3 dBSPL by tongue, respectively. Based on the correlation analysis for the BMI value and the snoring sound frequency of snoring part, the p-value was 0.49 for palate, 0.29 for epiglottic and 0.84 for tongue, respectively. There was no significant correlation between the BMI value and the sound pressure level of snoring. Based on the results of head shadow effect for sleeping place measurement, the significant difference of sound pressure level was 3.67 dB at 90 degrees of snoring sound source. As for the pinna shadow effect for the near ear, the difference of sound pressure level was 3.43 dB at 105 degrees of sound source. The difference of sound pressure level was 2.78 dB at 105 degrees of sound source for the far ear. After transferring the sound pressure level of snoring to the hearing level of snoring, the snoring sound pressure level was 16.63 dBHL for palate, 16.24 dBHL for epiglottic and 10.79 dBHL for tongue, respectively.

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Tsai, Ping-Chang, and 蔡秉昌. "Using a Microphone Array to Detect the Locations of Unusual Sound Sources in an Outdoor Area." Thesis, 2010. http://ndltd.ncl.edu.tw/handle/00121520137114982148.

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碩士 國立暨南國際大學 資訊工程學系 98 In this thesis, we study the unusual sound source localization problem in an outdoor area. In an outdoor environment, the accuracy of sound source localization will be influenced by the wind velocity, the air temperature, and background noises. This work aims to develop a sound source localization method which is accuracy and robust against noise and is able to localize multiple sound sources having non-overlapping spectrums in the frequency domain. The probability density function (PDF) of the time delay of arrival (TDOA) between two signals is derived based on the PDF of the phase angle between two signals. The PDF of the phase angle is derived from a noise model of the input signals. According to the derived probabilistic model, possible locations of the sound sources can be computed. The mean-shift algorithm is used to find clusters of possible locations. Too small clusters are discarded and the centers of the remaining clusters represent the estimated locations of the sound sources. To test the proposed method, a uniform linear microphone array consisting four microphones is constructed to collect sound signals in an outdoor area. The experimental results show that, in a strictly single sound source scenario, the accuracy of the proposed method is comparable to a very popular method known as the phase transform (PHAT) technique. Furthermore, when the background noises, such as the sounds from insects and/or birds, are not negligible, the proposed method outperforms the PHAT method. Additionally, the experimental results of estimating two simultaneous sound sources show that the proposed method also can achieve considerable stability for detecting/localizing multiple sound sources provided that the sound sources have non-overlapping spectrums.

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