Dissertations / Theses: 'Sound engineering'

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DelGizzi, Jesse D. "Zydeco Aesthetics| Instrumentation, Performance Practice, and Sound Engineering." Thesis, University of Louisiana at Lafayette, 2019. http://pqdtopen.proquest.com/#viewpdf?dispub=10816360.

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This thesis examines aesthetics, sonic characteristics, and performance practices of zydeco music as heard in south Louisiana today. The first chapter describes the roles of instruments in a zydeco band, focusing specifically on the importance of the kick drum and the snare drum. It also details the evolution of the modern zydeco sound and how certain instruments, their modifications, and their timbres came to characterize the style especially prevalent among a group of artists who play for zydeco trail rides. The second chapter examines the tempo of modern zydeco music through quantitative analysis of musical recordings. This chapter also elucidates the use of beat patterns and drumming techniques within the genre, providing evidence for a current preference for the boogaloo beat over the on-the-one and the double beats. The third chapter discusses sonic goals and values of the sound engineer in zydeco music in live performance. This chapter also includes analysis of the frequency spectrum profiles of live zydeco recordings which depict how sound reinforcement practices, instrument modifications, and playing techniques discussed in the thesis are manifested in these performances. Research methods employed for this thesis include interviews with zydeco musicians, empirical analysis of live musical recordings, and examination of spectrograms.

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Enoksson, Karl, and Bohan Zhou. "Sound following robot." Thesis, KTH, Maskinkonstruktion (Inst.), 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-226665.

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There are many different areas of use for sound localization. This concept is not only used to localize a person that is talking but can also be applied for finding a person in need One method of localizing the position of a sound source is to use several microphones to register the difference of time in which each microphone detects the same sound. Using this information and trigonometry, the direction of the sound source can be calculated. The objective of this thesis is to investigate how precisely the position of a sound source can be determined using the aforementioned technique whilst varying the distance and angle of the sound source. In order to explore the capabilities of TDOA and test the obtainable accuracy, a demonstrator was built. On a complete car chassis, four microphones were mounted and used to determine the direction towards the sound source. Thereafter the robot rotated towards the sound source with an IMU keeping track of how much it had rotated. After this movement a comparison was made between the robots direction and the actual direction of the sound source. Lastly an ultrasonic sensor was placed on the robot for obstacle detection whilst tracking the sound. The vehicle traveled straight forward until the ultrasonic sensor deemed that an object was too close. The results show that an increased distance yields a more accurate sound localization and that there are some angles in which the sound localization functioned better.
Idag finns det många olika användningsområden för ljudlokalisering. Konceptet används inte enbart till att lokalisera en person som pratar men kan också appliceras för att hitta en person i nöd. En metod för att lokalisera en ljudkällas position innebär kortfattat att med flera mikrofoner registrera de olika tiderna då ljudet når de olika mikrofonerna. Utifrån denna information kan riktningen till ljudkällans position beräknas med hjälp av trigonometri. Målet med denna rapport är att undersöka hur precist en ljudkällas position kan beräknas med den ovannämnda teknik genom att variera avståndet och vinkeln till ljudkällan. I syfte att genomföra tester byggdes en prototyp. På ett färdigbyggt chassi monterades fyra mikrofoner som användes för att bestämma riktningen till ljudkällan. Därefter roterade roboten mot ljudkällan med hjälp av en IMU som håller reda på hur mycket den har roterat. Efter denna rörelse utfördes en jämförelse mellan robotens riktning och ljudets faktiska riktning. Slutligen placerades en ultraljudssensor på roboten för att detektera objekt när den spårade ljudet. Fordonet färdades rakt fram tills ett objekt låg för nära ultraljudssensorn. Resultaten visar att ett ökat avstånd ger en mer nogrann ljudlokalisering samt att för vissa vinklar fungerade ljudlokaliseringen bättre.

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Pogaku, Sindhuja. "SOUND MODE APPLICATION." CSUSB ScholarWorks, 2017. https://scholarworks.lib.csusb.edu/etd/445.

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Currently there are millions of Android cell phone users. Whenever a user changes location, he/she should manually modify the sound mode (ring, vibrate, silent). So, it’s slightly inconvenient to constantly monitor whether the phone is in general or silent mode. Sometimes user might forget to switch the mobile mode and may create a disturbance in the classroom or in the work area. To overcome this problem “Sound Mode Application” is an Android application that allows a user to automatically change the sound mode depending on his/her GPS location. Additionally, the user may activate or deactivate the application whenever need be, and user can add as many locations as required based on their daily life.

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Meng, Helen M. "Phonological parsing for bi-directional letter-to-sound/sound-to-letter generation." Thesis, Massachusetts Institute of Technology, 1995. http://hdl.handle.net/1721.1/11413.

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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1995.
Includes bibliographical references (leaves 185-195).
by Helen Mei-Ling Meng.
Ph.D.

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Villareal, Steven G. (Steven Gregory). "Sound enhancements for graphical simulations." Thesis, Massachusetts Institute of Technology, 1997. http://hdl.handle.net/1721.1/43411.

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Putra, Azma. "Sound radiation from perforated plates." Thesis, University of Southampton, 2008. https://eprints.soton.ac.uk/63161/.

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Perforated plates are quite often used as a means of engineering noise control to reduce the sound radiated by structures. However, there appears to be a lack of representative models to determine the sound radiation from a perforated plate. The aim of this thesis is to develop such a model that can be used to give quantitative guidance corresponding to the design and effectiveness of this noise control measure. Following an assessment of various models for the radiation efficiency of an unbaffled plate, Laulagnet’s model is implemented. Results are calculated and compared with those for baffled plates. From this, simple empirical formulae are developed and give a very good agreement with the analytical result. Laulagnet’s model is then modified to include the effect of perforation in terms of a continuously distributed surface impedance to represent the holes. This produces a model for the sound radiation from a perforated unbaffled plate. It is found that the radiation efficiency reduces as the perforation ratio increases or as the hole size reduces. An approximate formula for the effect of perforation is proposed which shows a good agreement with the analytical calculation up to half the critical frequency. This could be used for an engineering application to predict the noise reduction due to perforation. The calculation for guided-guided boundary conditions shows that the radiation efficiency of an unbaffled plate is not sensitive to the edge conditions. It is also shown that perforation changes the plate bending stiffness and mass and hence increases the plate vibration. The situation is also considered in which a perforated unbaffled plate is located close to a reflecting rigid surface. This is established by modifying the Green’s function in the perforated unbaffled model to include an imaginary source to represent the reflected sound. The result shows that the presence of the rigid surface reduces the radiation efficiency at low frequencies. The limitation of the assumption of a continuous acoustic impedance is investigated using a model of discrete sources. The perforated plate is discretised into elementary sources representing the plate and also the holes. It is found that the uniform surface impedance is only valid if the hole distance is less than an acoustic wavelength for a vibrating rectangular piston and less than half an acoustic wavelength for a rectangular plate in bending vibration. Otherwise, the array of holes is no longer effective to reduce the sound radiation. Experimental validation is conducted using a reciprocity technique. A good agreement is achieved between the measured results and the theoretical calculation for both the unbaffled perforated plate and the perforated plate near a rigid surface.

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Strayer, Jayson D. (Jayson Dee). "Underwater sound puluse generator." Thesis, Massachusetts Institute of Technology, 1996. http://hdl.handle.net/1721.1/40200.

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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1996.
Includes bibliographical references (leaves 52-53).
by Jayson D. Strayer.
M.Eng.

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Jonsson, Kaj, and Dennis Lioubartsev. "Sound Localization in Robotic Application." Thesis, KTH, Maskinkonstruktion (Inst.), 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-226682.

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This thesis is focused on implementing sound localizationin robotics to explore how a computer can interpret it’ssurroundings, specifically using ”off the shelf components”.Sound localization gives robots depth to it’s hearing. Tostudy this, a robot is constructed with four microphones.To show the found location, a shooting mechanism is constructedand implemented to shoot a candy to the targetpoint. The core in this project is to accurately measurethe time of the incoming sound in form of impulse sound.This data is analyzed to find the source and then, build aturret that turns to the location and accurately deliver aprojectile to that position.The time difference from four microphones is analyzedto find the angle and distance to the sound source usingtriangulation. Once the location is found, the robot turnstowards the location and calculates the shooting trajectory.Finally the robot shoots a candy to the sound source.The accuracy of these methods are tested.In this thesis a full construction of the device and therelevant theory is presented. The final product is able findthe angle to the sound source and accurately shoot a projectileto a specific point. The distance triangulation deemedto be ineffective, due to sensitivity to errors being to great.
För att undersöka hur en dator kan tolka sin omgivninghar denna rapport fokuserat på att implementera ljudlokalisationi robotik och att lokalisera vinkel samt distans tillen ljudkälla. För att visa det funna värdet har en skjutmekanismkonstruerats och implementerats för att skjutaen projektil till positionen. Kärnan i projektet har varit attså exakt som möjligt mäta tiden för det inkommna ljudeti form av ett impulsljud med hjälp av mikrofoner, analyseradenna data för att hitta ljudkällan och att bygga enkonstruktion som vrider sig mot källan och levererar enprojektil till positionen.Tidsdifferansen från fyra mikrofoner analyseras för atthitta vinkel och distans till ljudkällan med triangulering.Exaktheten av dessa metoder testas.I denna rapport så presenteras den fullständiga konstruktionenoch den relevanta teorin. Den färdiga produktenkan hitta vinkel till ljudkällan och skjuta en projektiltill en bestämt punkt. Trianguleringen ansågs vara ineffektivpå grund av känslighet mot fel.

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Chatterley, James J. "Sound Quality Analysis of Sewing Machines." BYU ScholarsArchive, 2005. https://scholarsarchive.byu.edu/etd/424.

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Sound quality analysis is a tool designed to help determine customer preferences, which can be used to help the designer improve product quality. Many industries desire to know how the consuming public perceives their product, as this affects the product life and success. This research investigates which of the six sewing machines provided by Viking Sewing Machine Group (VSM group) consumers find most acoustically appealing. The sound quality analysis methods used include both jury based listening tests and quantitative sound quality metrics from empirical equations. The results from both methods are completely independent and are shown to have a very strong correlation. The procedures and results of both methods, jury listening tests and mathematical metrics, are presented. Near field sound intensity scans identified acoustic hot spots and give direction for possible design modifications to improve the acoustic signature of the two top tier machines, the Designer 1 and Creative 2144 (Husqvarna Viking and Pfaff respectively). This research determined that the entry level Pfaff Select 1530 has the most acoustically appealing sound of the six machines evaluated. In addition, it was also determined that a reduction in the higher frequency sounds produced by the machines is preferred over a reduction in the lower frequency sounds. Further investigations, including an evaluation of machine isolation and startup sounds, were also performed. The machine isolation results are highly dependant on the individual machine being evaluated and would require independent evaluation. In the machine startup sound assessment, it was discovered that again the Pfaff Select 1530 has the preferred sound. Near field acoustic intensity scans provide additional information on locations of strong acoustic radiation. The near field scans provided valuable design information. The acoustic "hot" spots were discovered to exist in the lower portions of the machines near the main stepper motor in the Designer 1, and radiating from the bottom plate of the machine in the Pfaff Creative 2144. This analysis has led to various design modifications that could be implemented to improve the sound quality of the machines, specifically the Designer 1 and the Creative 2144.

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Sedighian, Pouye. "Pediatric heart sound segmentation." Thesis, California State University, Long Beach, 2014. http://pqdtopen.proquest.com/#viewpdf?dispub=1526952.

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Recent advances in technology have facilitated the prospect of automatic cardiac auscultation by using digital stethoscopes. This in turn creates the need for development of algorithms capable of automatic segmentation of the heart sound. Pediatric heart sound segmentation is a challenging task due to various factors including the significant influence of respiration on the heart sound. This project studies the application of homomorphic filtering and Hidden Markov Model for the purpose of pediatric heart sound segmentation. The efficacy of the proposed method is evaluated on a publicly available dataset and its performance is compared with those of three other existing methods. The results show that our proposed method achieves accuracy of 92.4% ±1.1% and 93.5% ±1.1% in identification of first and second heart sound components, and is superior to four other existing methods in term of accuracy or time complexity.

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Akanji, Omololu. "Sound bullets from nonlinear granular chains." Thesis, University of Warwick, 2015. http://wrap.warwick.ac.uk/80024/.

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The propagation of ultrasound along chains of granular particles has some interesting characteristics. These have the potential to dramatically improve the performance of HIFU (High Intensity Focussed Ultrasound) for the use in therapeutic ultrasound treatments and medical imaging. This thesis has investigated a novel approach for the creation of ultrasonic focussed energy in chains composed of spheres. Within these highly sensitive chains, non-linear propagation is possible which leads to the formation of highly robust localised pulses known as sound bullets. Subject to the right conditions, the chain of spheres become a dynamically tunable system where slight changes to the nature of the Herzian contact between the spheres produce drastic changes in the propagation velocity of the solitary wave. The nature and resulting characteristics of the system to variations such as input excitation frequency, effect of loading, changes in length and diameter of the chain were studied. It was observed that the system was highly dependent of each of these factors, with each situation altering the behaviour of the chain of spheres.

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Yaseen, Ehab Ahmed A. "High intensity sound propagation in flow ducts." Thesis, University of Southampton, 1987. https://eprints.soton.ac.uk/52288/.

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A theoretical and experimental study of the propagation of finite amplitude pressure waves, or high intensity sound, in flow ducts was undertaken. A quantitative appreciation is presented describing the propagation of such waves including reflections. Observations showed that time domain descriptions are appropriate for the analysis of such waves, for which the well established method of characteristics proved ideal, providing that both the particle velocity and pressure time histories were established simultaneously. Existing techniques for predicting wave propagation in ducts based on the method of characteristics were reviewed and found to be inefficient for noise emission predictions when the velocity time history was not known. The approach described here gave reliable estimates of the velocity-time history at a reference plane in the duct using a pair of pressure-time histories obtained from closely spaced transducers flush mounted in the duct wall. The essential precautions and conditions required for the measurement procedures necessary to maintain confidence in the estimates of the velocity have been identified. Estimates of velocity and pressure time histories at two positions downstream and one upstream of the reference plane were obtained and found to compare well with observations. The appropriate boundary conditions for describing wave reflections at an open duct termination have been established and evaluated for a systematic set of practical excitations.

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Richards, Simon. "Aeroacoustic computation of sound radiation from ducts." Thesis, University of Southampton, 2005. https://eprints.soton.ac.uk/47100/.

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Modern high-bypass turbofan engines produce high levels of nuisance noise that has a significant impact on the environment near airports as well as the crew and passengers inside the aircraft. Significant research is being undertaken to understand the aeroacoustic noise source mechanisms and to accurately predict engine noise levels. High-performance computers and advanced numerical techniques are now taking an active role in this research area. In this work, a numerical solver is developed to accurately and efficiently predict noise radiation from ducts. The solver is based upon a hybrid methodology whereby only the acoustic near-field is solved using the developed numerical solver, with the resultant far-field directivity determined from an integral solution of the Ffowcs Williams - Hawking equation. Particular emphasis has been placed on the radiation of duct modes from a realistic bypass engine intake geometry. The performance of the numerical schemes employed in the solver is analysed, with particular attention to the dispersion and dissipation qualities. A study into the determination of a suitable non-reflecting boundary condition for duct acoustics is also undertaken. Using a novel formulation of the linearised Euler equations, the solver is applied to noise radiation from a realistic engine intake geometry with background mean flow. The accuracy of the scheme is validated by comparison with analytic solutions for the unflanged duct case. For the unflanged duct case the effect of an acoustic liner is modelled using a time-domain impedance boundary condition. The effect of a locally supersonic inflow on radiation from the engine intake is examined. Finally, the solver is extended to determine multimode radiation from generic engine intakes, with the possibility to incorporate swirling mean flows and asymmetric duct geometries.

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Juhlin, Rasmus. "Anovel: Sound Reading." Thesis, Malmö universitet, Fakulteten för teknik och samhälle (TS), 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:mau:diva-20550.

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Över en trestegsprocess brukar studien research through design för att utforska narrativ presenterade genom ett multimodalt medium bestående av komponenterna text och stream (ljud). En existerande applikation, Booktrack, har blivit undersökt under studien. Därefter utveckades tre prototyped för att identifiera och förstå hur narrativ design kan formas i förhållande till de två komponenterna. Studien har fokuserat på stream design och på uppfattningen av noveller och fiktiva texter som presenteras genom både text och ljud. Med sitt ursprung i sound studies identifierar studien fem generella och tre specifika stream design principer vid arbete av ljud i relation till text. Utöver dessa indikerar studien att en streams närvaro påverkar hur en läsare visualiserar narrativet. Vidare har studien utforskat narrativ design där båda komponenterna är beaktade. Därigenom markeras potentialen hos den multimodala presentationen att kunna expandera narrativet, genom medveten användningen av både text och stream. Den multimodala presentationen antyder dock att likheten med traditionella noveller och liknande ”tysta” texter både skapade intresse och oro. Sammanfattningsvis förstärkte närvaron av en stream inlevelsen samtidigt som det inhöll risken och förmågan att tvingat vägleda en läsares tolkning och föreställning av narrativet. I ett samtid där multimodala presentationer är tillgängliga genom diverse smart-devices, är det troligt att presentationer i likhet med de som undersökts i studien kan utgöra ett nästa steg i alldagliga presentationer. Ta nyhetsartiklar, annonsering, och informationsbrochyrer som ett par områden utanför noveller och fiktion där liknande multimodala presentationer kan komma att utvecklas och användas i samhället inom en snar framtid.
Through a three-part process employing research through design, this study has explored narratives being presented through a multimodal technical medium consisting of both textual and stream (sound) components. It has examined an existing application, Booktrack, and through developing three separate prototypes, has sought to identify and understand how one might approach a narrative when constructed using the aforementioned components. Specifically, it has explored the stream (sound) design and the perception of novels and fiction texts when presented through both text and sound. Taking on a perspective with its origin in sound studies, the study has identified five general and three specific stream design guidelines for working with sound in relation to text. Moreover, it has indicated that contextually appropriate streams’ presences alongside written text affect how a reader visualizes the narrative. Further, it has explored narrative design with both the textual and stream components in mind. Thereby, it posits a venue where the multimodality of the presentation might be used to expand the narrative presentation, using both the text and the stream as tools to further the narrative. However, it also identifies the similarities of the narrative presentation with the traditional novel and similar silent texts as being an indictment of concern. Namely, participants of the study expressed the stream’s intrusion and impact upon their immersion and visualization of written stories as both immersion enhancing and as forcibly guiding their imagination. In a society where multimodal presentations are available through phones, tablets, and other devices, it seems plausible a multimodal presentation, such as the one explored, might constitute the next step in everyday presentations. Take news articles, advertisements, and information brochures as a few tangible areas where this kind of presentation might be employed in the close future.

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Sikora, Joseph J. III. "Sound propagation around underwater seamounts." Thesis, Massachusetts Institute of Technology, 2009. http://hdl.handle.net/1721.1/58863.

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Thesis (Ph. D.)--Joint Program in Applied Ocean Physics and Engineering (Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science; and the Woods Hole Oceanographic Institution), 2009.
Includes bibliographical references (leaves 184-189).
In the ocean, low frequency acoustic waves propagate with low attenuation and cylindrical spreading loss over long-ranges, making them an effective tool for underwater source localization, tomography, and communications. Underwater mountains, or seamounts, are ubiquitous throughout the world's oceans and can absorb and scatter acoustic energy, offering many interesting acoustic modeling challenges. The goal of the research performed in support of this thesis is to measure the acoustic scattered field of a large, conical seamount at long-range, and reconcile observations with 2-D range-dependent acoustic models, for the purpose of understanding the effects of highly range-dependent bathymetry. The Basin Acoustic Seamount Scattering Experiment (BASSEX) was conducted to measure the scattered fields of the two seamounts which form the Kermit-Roosevelt Seamount Complex in the Northeast Pacific Ocean during September and October of 2004. The experiment used fixed and ship-deployed acoustic sources transmitting m-sequence signals at 68.2 and 250 Hz carrier frequencies, with 35 and 83 Hz bandwidth, respectively. The receiver was a towed hydrophone array with 3 m sensor spacing, cut for 250 Hz. BASSEX is the first experiment to measure acoustic arrival patterns in the scattered field of a seamount at many locations at sound path ranges of order 500 km, utilizing a rich bathymetry and sound velocity database. Convergence zones in the forward-scattered field of seamounts at long-range are observed, created by higher order mode coupling and blockage. Acoustic ray arrival angles, travel times, and amplitudes show good agreement with parabolic equation (PE) acoustic modeling results inside the forward-scattered fields; in particular, simulated results are fairly accurate for weak surface-reflected-bottom-reflected acoustic rays. The width of the forward-scattered field is shown to span the projected width of a seamount.
(cont.) Temporal coherence of ray amplitude inside a seamount scattered field could not be determined due to array movement issues, and should be the focus of future research to determine the stability of scattered acoustic rays for applications such as acoustic tomography. Robust adaptive beamforming methods are used to process hydrophone array data gathered in the BASSEX experiment. Non-stationarity in the observed noise field caused by array fluctuations and data acquisition system malfunctions motivate the use of a time varying Capon adaptive beam former, and strong acoustic harmonics from ship operations motivate the use of a frequency and steering angle dependent white noise gain constraint. In an effort to process snap-shot deficient data sets, the novel physically constrained maximum likelihood (PCML) beamformer was further developed and applied. By using orthonormal trigonometric eigenvector bases to determine the maximum likelihood spectral covariance matrix, the PCML beamformer computational efficiency is significantly increased.
by Joseph J. Sikora, III.
Ph.D.

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Ubellacker, Wyatt. "Underwater communication via compact mechanical sound generation." Thesis, Massachusetts Institute of Technology, 2013. http://hdl.handle.net/1721.1/83751.

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Thesis (S.B.)--Massachusetts Institute of Technology, Dept. of Mechanical Engineering, 2013.
Cataloged from PDF version of thesis.
Includes bibliographical references (page 55).
Effective communication with underwater remotely operated vehicles (UROV) can be difficult to accomplish. In water, simple radio communication is quickly dissipated at higher frequencies and lower frequencies require a large antenna, which may not be practical in all applications. Light can also be used to communicate with the vehicles, but requires line of sight between the source and detector. Sound can also be used as a communication method, and has many advantages. It can propagate long distances underwater and does not require line of sight to work effectively. However, generating sound electronically underwater requires a large power speaker to produce tones loud enough to travel far distances. Generating sound mechanically can take advantage of physical resonance and produce high intensity tones in a compact device with a relatively low power input. This can allow for a compact, high intensity method to communicate with remotely operated underwater vehicles.
by Wyatt Ubellacker.
S.B.

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Hirst, J. M. "Spatial impression in multichannel surround sound systems." Thesis, University of Salford, 2006. http://usir.salford.ac.uk/2226/.

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Spatial impression in both concert halls and reproduced sound has been identified as an important attribute of the listening experience. In this study, the synthesis and objective measurement of spatial impression in reproduced sound is examined. A novel, multichannel spatializing technique for musical synthesis has been developed that entailed the separation of the individual harmonics of a musical note that were spatially distributed over multichannel surround systems. Subjective testing of the techniques revealed that the perceived degree of spatial impression significantly increased as the angular spread of harmonics increased, however, extending the spatial spread beyond 90° did not significantly increase the perception of spatial impression. The concert hall measure of spatial impression, the interaural cross correlation coefficient (IACC) was used to objectively measure the effects of the spatializing techniques. The IACC measurements displayed a strong correlation to the subjective results. Further examination of the IACC measurement indicated the possibility of it’s adaptation to multichannel surround sound in general. A method of adapting IACC to reproduced sound was further developed that involved comparing IACC measurements taken in a concert hall to IACC measurements taken in reproduced versions of the same concert hall. The method was first conducted as a simulation using basic auralisation techniques. Real concert hall measurements and reproduction systems were then employed. Results showed that the method was able to discriminate between the spatial capabilities of a number of different surround sound systems and rank them in a predictable order. The results were further validated by means of a subjective test. In an attempt to sensitise the IACC measurement, the frequency dependency of IACC was investigated by means of a subjective test. The results indicated that a perceptually more accurate indication of spatial impression may be gained by applying a frequency-dependent weighting to IACC measurements. This may be useful in the spatial measurement of both reproduced sound and concert halls.

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Sikora, Joseph J. III. "Sound propagation around underwater seamounts." Thesis, Massachusetts Institute of Technology, 2005. http://hdl.handle.net/1721.1/39196.

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Thesis (S.M.)--Joint Program in Applied Ocean Science and Engineering (Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science; and the Woods Hole Oceanographic Institution), 2005.
Includes bibliographical references (leaves 120-121).
This thesis develops and utilizes a method for analyzing data from the North Pacific Acoustic Laboratory's (NPAL) Basin Acoustic Seamount Scattering Experiment (BASSEX). BASSEX was designed to provide data to support the development of analytical techniques and methods which improve the understanding of sound propagation around underwater seamounts. The depth-dependent sound velocity profile of typical ocean waveguides force sound to travel in convergence zones about a minimum sound speed depth. This ducted nature of the ocean makes modeling the acoustic field around seamounts particularly challenging, compared to an isovelocity medium. The conical shape of seamounts also adds to the complexity of the scatter field. It is important to the U.S. Navy to understand how sound is diffracted around this type of topographic feature. Underwater seamounts can be used to conceal submarines by absorbing and scattering the sound they emit. BASSEX measurements have characterized the size and shape of the forward scatter field around the Kermit-Roosevelt Seamount in the Pacific Ocean. Kermit-Roosevelt is a large, conical seamount which shoals close to the minimum sound speed depth, making it ideal for study. Acoustic sources, including M-sequence and linear frequency-modulated sources, were stationed around the seamount at megameter ranges. A hydrophone array was towed around the seamount to locations which allowed measurement of the perturbation zone. Results from the method developed in this thesis show that the size and shape of the perturbation zone measured coincides with theoretical and experimental results derived in previous work.
by Joseph J. Sikora, III.
S.M.

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Wang, Xing. "Structure-borne sound transmission on frameworks of beams." Thesis, University of Liverpool, 2015. http://livrepository.liverpool.ac.uk/2020839/.

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Many engineering structures are built from frameworks of beams, particularly lightweight structures. For the purpose of noise control from airborne and structure-borne sources, it is useful to be able to predict vibration transmission across these frameworks. This thesis investigates the potential use of Advanced Statistical Energy Analysis (ASEA) to predict structure-borne sound transmission when the beams support multiple wave types due to wave conversion at the junction. In contrast to Statistical Energy Analysis (SEA), ASEA is able to account for high propagation losses and indirect coupling through the use of ray tracing. SEA and ASEA were validated through comparison with measurements and numerical experiments with Finite Element Methods (FEM). When each beam supports at least two local modes for each wave type in the frequency band of interest and the modal overlap factor is at least 0.1, FEM and measurement data tend to have average values which form smooth curves such as those predicted by SEA and ASEA. It was shown that SEA and ASEA models could incorporate Euler-Bernoulli and Timoshenko theory by changing over from Euler-Bernoulli to Timoshenko group velocity when calculating the coupling loss factors. However, comparisons with measurements were not conclusive although there were indications that a suitable crossover frequency could be when Timoshenko and Euler-Bernoulli group velocities differ by at least 26%. Agreement between FEM and ASEA indicates that it is appropriate to ignore phase effects in the ray tracing approach used with ASEA. This was particularly noteworthy for the three-bay and five-bay truss beams as these were perfectly periodic for which phase effects could be important. Results for an L-junction, a rectangular beam frame and a five-bay truss with relatively long beams and relatively high internal loss factors demonstrated that ASEA was able to incorporate high propagation losses. This was not possible with SEA. For a three-bay truss beam with relatively short beams ASEA showed close agreement with FEM and measurements confirming that there was significant indirect coupling rather than high propagation losses. There are indications from the five-bay truss beams that ASEA may no longer be accurate in predicting the response on beams that are at least three structural junctions away from the source beam, particularly when ASEA predicts high propagation losses on the receiving beam.

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Hummersone, Christopher. "A psychoacoustic engineering approach to machine sound source separation in reverberant environments." Thesis, University of Surrey, 2011. http://epubs.surrey.ac.uk/2923/.

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Reverberation continues to present a major problem for sound source separation algorithms, due to its corruption of many of the acoustical cues on which these algorithms rely. However, humans demonstrate a remarkable robustness to reverberation and many psychophysical and perceptual mechanisms are well documented. This thesis therefore considers the research question; can the reverberation-performance of existing psychoacoustic engineering approaches to machine source separation be improved. The precedence effect is a perceptual mechanism that aids our ability to localise sounds in reverberant environments. Despite this, relatively little work has been done on incorporating the precedence effect into automated sound source separation. Consequently, a study was conducted that compared several computational precedence models and their impact on the performance of a baseline separation algorithm. The algorithm included a precedence model, which was replaced with the other precedence models during the investigation. The models were tested using a novel metric in a range of reverberant rooms and with a range of other mixture parameters. The metric, termed Ideal Binary Mask Ratio, is shown to be robust to the effects of reverberation and facilitates meaningful and direct comparison between algorithms across different acoustic conditions. Large differences between the performances of the models were observed. The results showed that a separation algorithm incorporating a model based on interaural coherence produces the greatest performance gain over the baseline algorithm. The results from the study also indicated that it may be necessary to adapt the precedence model to the acoustic conditions in which the model is utilised. This effect is analogous to the perceptual Clifton effect, which is a dynamic component of the precedence effect that appears to adapt precedence to a given acoustic environment in order to maximise its effectiveness. However, no work has been carried out on adapting a precedence model to the acoustic conditions under test. Specifically, although the necessity for such a component has been suggested in the literature, neither its necessity nor benefit has been formally validated. Consequently, a further study was conducted in which parameters of each of the previously compared precedence models were varied in each room in order to identify if, and to what extent, the separation performance varied with these parameters. The results showed that the reverberation-performance of existing psychoacoustic engineering approaches to machine source separation can be improved and can yield significant gains in separation performance.

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Alkacir, Hakan. "Sound in motion : Ljud, musik och aktivitet." Thesis, Konstfack, Industridesign, 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:konstfack:diva-5845.

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DiPerna, Daniel T. "Sound scattering by cylinders of noncircular cross section." Thesis, Massachusetts Institute of Technology, 1993. http://hdl.handle.net/1721.1/35370.

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Suh, In-Soo 1964. "An investigation of sound quality of I.C. engines." Thesis, Massachusetts Institute of Technology, 1998. http://hdl.handle.net/1721.1/10070.

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Kernen, Ulrica. "Airborne sound insulation of floating floors." Licentiate thesis, KTH, Byggnader och installationer, 2000. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-1036.

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Clifford, Alice. "Reducing microphone artefacts in live sound." Thesis, Queen Mary, University of London, 2013. http://qmro.qmul.ac.uk/xmlui/handle/123456789/8383.

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This thesis presents research into reducing microphone artefacts in live sound with no prior knowledge of the sources or microphones. Microphone artefacts are defined as additional sounds or distortions that occur on a microphone signal that are often undesired. We focus on the proximity effect, comb filtering and microphone bleed. In each case we present a method that either automatically implements human sound engineering techniques or we present a novel method that makes use of audio signal processing techniques that goes beyond the skills of a sound engineer. By doing this we can show that a higher quality mix of a live performance can be achieved. Firstly we investigate the proximity effect which occurs on directional microphones. We present a method for detecting the proximity effect with no prior knowledge of the source to microphone distance. This then leads to a method for reducing the proximity effect which employs a dynamic filter informed by audio analysis. Comb filtering occurs when the output of microphones reproducing the same source are mixed together. We present a novel analysis of how the accuracy of a technique to automatically estimate the correct delay of the source between each microphone is affected by source bandwidth and the windowing function applied to the data. We then present a method for reducing microphone bleed in the multiple source, multiple microphone case, both in determined and overdetermined configurations. The proposed method is extended from prior research in noise cancellation, which has not previously been applied to musical sound sources. We then present a method for simulating microphone bleed in synthesised drum recordings, where bleed enhances the realism of the output. Through subjective listening tests and objective measures each proposed method is shown to succeed at reducing the microphone artefacts while preserving the original sound source.

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Gabovich, Vladislav Y. (Vladislav Yurievich) 1980. "A multi-stage sound-to-letter recognizer." Thesis, Massachusetts Institute of Technology, 2002. http://hdl.handle.net/1721.1/87225.

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Thesis (M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2002.
Includes bibliographical references (p. 87-88).
by Vladislav Y. Gabovich.
M.Eng.

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Colobong, Genee Lyn O. (Genee Lyn Ollero) 1976. "Sound devices for the Cricket Bus System." Thesis, Massachusetts Institute of Technology, 1999. http://hdl.handle.net/1721.1/80529.

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Thesis (S.B. and M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1999.
Includes bibliographical references (leaves 78-79).
by Genee Lyn O. Colobong.
S.B.and M.Eng.

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Reed, Darrin Kiyoshi. "Virtual audio localization with simulated early-reflections and generalized head-related transfer functions." Thesis, Montana State University, 2009. http://etd.lib.montana.edu/etd/2009/reed/ReedD1209.pdf.

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In a natural sonic environment a listener is accustomed to hearing reflections and reverberation. It is conceived that early reflections could reduce front-back confusion in synthetic 3-D audio. This thesis describes experiments which seek to determine whether or not simulated reflections can reduce front-back confusions for audio presented with non-individualized head-related transfer functions (HRTFs) via headphones. To measure the contribution of the reflections, 13 human subjects participated in localization experiments which compared their localization ability with anechoic HRTF processing versus HRTF processing with a single early-reflection. The results were highly subject dependent; some showed improvement while others seemed to be inhibited by the reflections. Statistical analysis of the overall results concluded that a single reflection does not provide a significant difference in localization ability. Although this data rejects the hypothesis of this investigation, some suspicion regarding the contribution of lateral reflections in an auditory environment remains.

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Fisher, Andrew N. "Efficient, sound formal verification for analog/mixed-signal circuits." Thesis, The University of Utah, 2016. http://pqdtopen.proquest.com/#viewpdf?dispub=10003590.

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The increasing demand for smaller, more efficient circuits has created a need for both digital and analog designs to scale down. Digital technologies have been successful in meeting this challenge, but analog circuits have lagged behind due to smaller transistor sizes having a disproportionate negative affect. Since many applications require small, low-power analog circuits, the trend has been to take advantage of digital's ability to scale by replacing as much of the analog circuitry as possible with digital counterparts. The results are known as \emph{digitally-intensive analog/mixed-signal} (AMS) circuits. Though such circuits have helped the scaling problem, they have further complicated verification. This dissertation improves on techniques for AMS property specifications, as well as, develops sound, efficient extensions to formal AMS verification methods. With the \emph{language for analog/mixed-signal properties} (LAMP), one has a simple intuitive language for specifying AMS properties. LAMP provides a more procedural method for describing properties that is more straightforward than temporal logic-like languages. However, LAMP is still a nascent language and is limited in the types of properties it is capable of describing. This dissertation extends LAMP by adding statements to ignore transient periods and be able to reset the property check when the environment conditions change. After specifying a property, one needs to verify that the circuit satisfies the property. An efficient method for formally verifying AMS circuits is to use the restricted polyhedral class of \emph{zones}. Zones have simple operations for exploring the reachable state space, but they are only applicable to circuit models that utilize constant rates. To extend zones to more general models, this dissertation provides the theory and implementation needed to soundly handle models with ranges of rates. As a second improvement to the state representation, this dissertation describes how octagons can be adapted to model checking AMS circuit models. Though zones have efficient algorithms, it comes at a cost of over-approximating the reachable state space. Octagons have similarly efficient algorithms while adding additional flexibility to reduce the necessary over-approximations. Finally, the full methodology described in this dissertation is demonstrated on two examples. The first example is a switched capacitor integrator that has been studied in the context of transforming the original formal model to use only single rate assignments. Th property of not saturating is written in LAMP, the circuit is learned, and the property is checked against a faulty and correct circuit. In addition, it is shown that the zone extension, and its implementation with octagons, recovers all previous conclusions with the switched capacitor integrator without the need to translate the model. In particular, the method applies generally to all the models produced and does not require the soundness check needed by the translational approach to accept positive verification results. As a second example, the full tool flow is demonstrated on a digital C-element that is driven by a pair of RC networks, creating an AMS circuit. The RC networks are chosen so that the inputs to the C-element are ordered. LAMP is used to codify this behavior and it is verified that the input signals change in the correct order for the provided SPICE simulation traces.

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Park, Sewon. "Sound wave scattering by cyclindrical shells with internal structures." Thesis, Massachusetts Institute of Technology, 1995. http://hdl.handle.net/1721.1/36061.

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Eskenazi, Jérémie 1976. "A computer model for sound propagation around conical seamounts." Thesis, Massachusetts Institute of Technology, 2001. http://hdl.handle.net/1721.1/8778.

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Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Ocean Engineering, 2001.
Includes bibliographical references (leaves 67-69).
This paper demonstrates a technique for computing the long-range sound pressure field around a penetrable conical seamount. The pressure field is generated by a harmonic point source. The seamount is positioned in a vertically stratified ocean. It is modeled as an outgrowth of the sediment layer covering the ocean bottom. First, the seamount is decomposed into superposed rings of diameters increasing with the depth. Thus the problem reduces to a cylindrically layered system. Then, the method of normal modes is used to compute the sound pressure field in each layer. In order to maintain numerical stability, the Direct Global Matrix approach is used. The radial eigenfunctions are expressed as functions of normalized Hankel and Bessel functions, and the linear system that arise is organized in an unconditionally stable matrix. The results show a perturbation zone behind the seamount. It is bounded by two lines going from the source and tangent to the ring that is at the depth of the source. The values of the sound pressure inside the perturbation zone can be higher or lower than the values outside of it, according to the dimensions of the seamount.
by Jérémie Eskenazi.
S.M.

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Tavakoli, Nia Hadi. "Acoustic function of sound hole design in musical instruments." Thesis, Massachusetts Institute of Technology, 2010. http://hdl.handle.net/1721.1/61924.

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Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Mechanical Engineering, 2010.
Cataloged from PDF version of thesis.
Includes bibliographical references (p. 69-70).
Sound-hole, an essential component of stringed musical instruments, enhances the sound radiation in the lower octave by introducing a natural vibration mode called air resonance. Many musical instruments, including those from the violin, lute and oud families have evolved complex sound-hole geometries through centuries of trail and error. However, due to the inability of current theories to analyze complex sound-holes, the design knowledge in such sound-holes accumulated by time is still uncovered. Here we present the potential physical principles behind the historical development of complex sound-holes such as rosettes in lute, f-hole in violin and multiple sound-holes in oud families based on a newly developed unified approach to analyze general sound-holes. We showed that the majority of the air flow passes through the near-the-edge area of the opening, which has potentially led to the emergence of rosettes in lute family. Consequently, we showed that the variation in resonance frequency and bandwidth of different traditional rosettes with fixed outer diameter is less than a semitone, while the methods based on the total void area predicts variations of many semitones. Investigating the evolution of sound-holes in violin family from circular geometry in at least 10th century to the present-day f-hole geometry, we found that the evolution is consistent with a drive toward decreasing the void area and increasing the resonance bandwidth for a fixed resonance frequency. We anticipate this approach to be a starting point in discovering the concepts behind the geometrical design of the existing sound-hole geometries, and helping the musicians, instrument makers and scientists utilize this knowledge to design consistently better instruments.
by Hadi Tavakoli Nia.
S.M.

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Mugagga, Pius Kavuma Basajjabaka. "A binaural sound sources localisation application for smart phones." Master's thesis, University of Cape Town, 2015. http://hdl.handle.net/11427/24295.

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The ability to estimate positions of sound sources is one that gives animals a 360° awareness of their acoustic environment. This helps compliment the visual scene which is restricted to 180° in humans. Unfortunately, deaf people are left out on this ability. Smart phones are rapidly becoming a common tool amongst mobile users in developed and emerging markets. Their processing ability has more than doubled since their introduction to mass consumer markets by Apple in 2007. Top-end smart phones such as the Samsung Galaxy Series; 3, 4, and 5 models, have two microphones with which one can acquire stereo recordings. The purpose of this research project was to establish a feasible Sound source localization algorithm for current top-end smart phones, and to recommend hardware improvements for future smart phones, to pave way for the use of smart phones as advanced auditory sensory devices capable of acting as avatars for intelligent remote systems to learn about different acoustic scenes with help of human users. The GCC-PHAT algorithm was chosen as the underlying core DOA algorithm due to its suitability for pair-wise localization as highlighted in literature. A stochastic power accumulation algorithm was designed and implemented to improve estimation outcomes by GCC-PHAT. This algorithm was based on inspiration from W-disjoint orthogonality assumption in literature and was extended to perform sound source counting and time domain source separation. The system yielded satisfactory azimuth estimates of sound source directions in real time with pin-point DOA estimation accuracy rates of 64%, and 90.67% accuracy rate when a tolerance of ± 1 correlation sample is considered. An effort to resolve front back ambiguity using phone orientation data from the MEMs sensors yielded un-satisfactory results prompting a recommendation that an extra microphone would be needed to achieve 360° localization in a more user friendly way. The dissertation concludes with plans for further work on the topic and provision of a further refined API and optimised libraries to facilitate development of customised solutions using this system.

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Dunne, Gerard T. "The introduction of a Sound Quality Engineering Process to Jaguar Cars : executive summary." Thesis, University of Warwick, 2003. http://wrap.warwick.ac.uk/3976/.

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The control of the noise and vibration generated by an automobile is referred to as Noise, Vibration and Harshness (NVH) engineering. It involves identifying the design detail required to reduce the noise and vibration inside the passenger compartment of the vehicle to levels that are acceptable to the customer. It also involves delivering an engine or a powertrain sound character that is both pleasing to the customer and that suits the character of the vehicle. Tuning the sound generated by a vehicle to deliver a particular character is referred to as Sound Quality Engineering. This document summarizes the work of the EngD research programme that was aimed at developing a structured process for engineering the Powertrain Sound Quality of an automobile. The need for developing a Sound Quality Engineering Process at Jaguar Cars was identified through a review of customer evaluations of the sound in Jaguar's vehicles and those of its competitors. This review established that Jaguar's existing vehicles were trailing the leading competition in terms of the delivery of Powertrain Sound Quality. The reason for this shortfall was that the NVH Department at Jaguar did not have a focus on delivering the customer requirements. Without this focus there was no means of using the customer level requirements, for Sound Quality to drive the vehicle design process. The EngD research programme resulted in the formulation and implementation of a Sound Quality Engineering Process at Jaguar Cars that addressed this need. The first part of the research programme involved developing a means of quantifying the differences in the subjective Sound Quality character perceived by the customer. It was established that the subjective nature of the Powertrain Sound Quality could be represented by two underlying dimensions; a measure of the degree of Refinement and a measure of degree of Powerfulness. An assessment technique was developed that enabled the subjective Sound Quality character for a given vehicle to be quantified through its location within a 2-Dimensional Sound Quality Space, the axes of which were defined by each of the two underlying dimensions of Sound Quality. This 2- Dimensional Sound Quality Space provided the means of quantifying the differences in the Sound Quality characters for all of the vehicles competing in the luxury vehicle sectors. It was applied to define subjective Sound Quality targets for all of the new vehicle programmes at Jaguar Cars. These targets identified the required improvements to each of the two underlying dimensions of Sound Quality needed to address the shortfalls in Jaguar Cars' existing vehicles. The second part of the research programme involved identifying the key acoustic features within the sound signatures of Jaguar's vehicles that were responsible for determining the differences in subjective perception between these vehicles and their competitors. The changes to these key acoustic features were related to the required improvements to each of the two dimensions of Sound Quality that were established from the subjective target setting process. The final part of the research programme involved developing techniques that linked these key acoustic features to the noise sources and paths that were responsible for generating them. Through this link it was possible to establish the changes to these noise sources and paths that were necessary to deliver the required changes to the key acoustic features. In this way the required improvements to each of the two underlying dimensions of Sound Quality were used to define the vehicle design specification at the concept stage of the vehicle development programme and consequently drive the vehicle design process. The ability to link the subjective customer level requirements for Sound Quality to the design detail specification has overcome the previously identified shortfall within the NVH development process at Jaguar Cars. The techniques developed during the EngD research programme were formulated into a Sound Quality Engineering Process. Although the process was developed for Jaguar Cars the findings from the research and the techniques developed have since been applied by the different brands within the Ford Motor Company. Within Jaguar Cars the process has been implemented across all of the new vehicle programmes. It has directly resulted in significantly improved Sound Quality characters in the new vehicles that have been recently introduced to the luxury vehicle market.

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Jerner, Joel. "On Exaggeration of Sound Detail as a Way of Affecting Perceived Realism in Sound Effects and Musical Instruments." Thesis, Luleå tekniska universitet, Institutionen för konst, kommunikation och lärande, 2019. http://urn.kb.se/resolve?urn=urn:nbn:se:ltu:diva-74758.

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Based on previous research into realism in sound effects, this experiment was intended to study the effect of exaggerating minor details as a way of increasing perceived realism in sound effects. To do this, a framework for discussing components and subcomponents of sounds was constructed and a listening test designed using the framework was conducted. A two-way repeated measures ANOVA was used to measure the two main effects exaggeration (yes/no) and sound type (musical/non-musical). There was no effect of exaggeration by itself (p = 0.224). Musical sounds were perceived to be more realistic than non-musical sounds (p = 0.004). There was some evidence (p = 0.102), non-significant but interesting, of an interaction effect but it cannot be declared with confidence.

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Hough, Susan Patricia. "Some implications of causality in the active control of sound." Thesis, University of Southampton, 1988. https://eprints.soton.ac.uk/52274/.

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Active noise control is evaluated for a variety of primary source waveforms in a one-dimensional free field, and white noise primary sources in a one-dimensional enclosure with varying end conditions. A study is made of the consequences of changes in cost function. Frequency and time domain methods of defining optimum causal controllers are reviewed, and the ability of an adaptive controller to approximate the optimum is evaluated. Time and frequency domain models of sound fields are compared, and good agreement shown, for both one-dimensional (duct) and three-dimensional (room) enclosed sound fields. The performance of finite causal secondary source controllers in reverberant rooms is observed for varying microphone locations and wall reflection coefficients. Multi-sensor, single-secondary active control systems for three-dimensional reverberant enclosures are briefly studied, for secondary sources near to and remote from the primary source. Constraints of causality, finite length, and calculation delay are imposed, with consequent effects on controller configuration and resulting sound field with primary and secondary sources in operation. Comparison is made between the sound field due to the primary source alone, the sound field due to primary and optimal secondary source, and the sound field due to primary and realizable secondary source.

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Sun, Renfei. "Cavity-enhanced jetting sound produced by a baffled piston." Thesis, Boston University, 2014. https://hdl.handle.net/2144/21259.

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Thesis (M.Sc.Eng.) PLEASE NOTE: Boston University Libraries did not receive an Authorization To Manage form for this thesis or dissertation. It is therefore not openly accessible, though it may be available by request. If you are the author or principal advisor of this work and would like to request open access for it, please contact us at open-help@bu.edu. Thank you.
When a circular piston vibrates within an aperture of a baffle, the flow within the annular gap between the piston and the baffle is opposite in phase, which causes a reduction in the radiated sound. That is, the acoustic power is always less than in the absence of a leaking edge flow. However, when the piston is backed by a cavity, the overall acoustic power can be increased. This thesis focuses on approximating the effect of the ‘leakage’ for a piston of radius a in a cylindrical cavity of radius b within a rigid baffle. First we consider the piston within a baffle without a cavity, which leads to a reduced acoustic power, then we examine the two cases of a closed cavity and an open cavity. The acoustic power is found to increase when the piston vibrates close to a resonant frequency of the closed or open cavity. The smallest resonant frequency fmin depends on the cavity depth and end-correction. The maximum ‘gain’ in acoustic power ~ 10 dB which depends on the nonlinear edge flow and also depends on the ratio a/b, the aspect ratio b/L and ζ0/b, where L is the cavity depth and ζ0 is the amplitude of piston displacement.
2031-01-01

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38

Yang, Qin. "Computational study of sound generation by surface roughness in turbulent boundary layers." Thesis, University of Notre Dame, 2014. http://pqdtopen.proquest.com/#viewpdf?dispub=3578996.

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Noise generated by flow over rough surfaces is an important issue in naval applications and in aeronautical engineering. This work numerically investigates roughness-induced noise from low-Mach-number turbulent boundary layers. The computational approach is based on Lighthill's acoustic analogy with acoustic sources obtained from large-eddy simulation. An acoustic formulation is derived, which shows that each roughness element acts as an individual in-plane dipole source strengthened by its image in the wall. Flow configurations investigated include boundary-layer flows over a single hemispherical roughness element, a pair of streamwisely aligned hemispherical elements and three roughness fetches consisting of 10 × 4 hemispherical, cuboidal and cylindrical roughness elements, respectively.

Results for a single hemispherical roughness element and a pair of hemispherical elements show that the spanwise dipole, which has been overlooked before, is of larger or similar strength compared to the streamwise dipole. The viscous contribution to the dipoles is negligible compared to the pressure contribution. The main sound sources arise from the impingement of incoming turbulence and the unsteady horse-shoe vortices generated around the element. The roughness-induced unsteady wake motions are unimportant as a source of self noise. However, they significantly enhance sound radiation from a downstream hemisphere.

The effects of multi-element interactions and the roughness shape are investigated with arrays of 10 × 4 sparsely distributed hemispheres, cuboids and short cylinders. The dipole strength, orientation and spatial distribution show strong dependence on the roughness shape. Correlations between dipole sources associated with neighboring elements are found to be small for these sparsely distributed roughness arrays. Correlations and coherence between roughness dipoles and surface pressure fluctuations are analyzed, which reveals the importance of the impingement of upstream turbulence and surrounding vortical structures to dipole sound radiation, especially in the streamwise direction. For roughness shapes with sharp frontal edges, the edge-induced unsteady separation and reattachment also play important roles in sound generation. Large-scale turbulent structures in the boundary layer have a relatively low influence on roughness dipoles, except for the first row of elements.

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Martin, Keith Dana. "Sound-source recognition : a theory and computational model." Thesis, Massachusetts Institute of Technology, 1999. http://hdl.handle.net/1721.1/9468.

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Thesis (Ph.D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1999.
Includes 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.

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Kim, Frank Seong-Huhn. "Converting EKG signals into 3-D stereo sound." Thesis, Massachusetts Institute of Technology, 1994. http://hdl.handle.net/1721.1/35955.

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Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1994.
Includes bibliographical references (leaves 85-86).
by Frank Seong-Huhn Kim.
M.S.

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41

Tinney, 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.

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Findlay, David A. "Three multi-track recording projects : an analysis of aesthetic and technical engineering considerations." Thesis, McGill University, 1987. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=63956.

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Roemer, Jake. "Practical High-Coverage Sound Predictive Race Detection." The Ohio State University, 2019. http://rave.ohiolink.edu/etdc/view?acc_num=osu1563505463237874.

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Park, Munhum. "Models of binaural hearing for sound lateralisation and localisation." Thesis, University of Southampton, 2007. https://eprints.soton.ac.uk/162297/.

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The current study suggests two models of binaural hearing, which aim to make predictions for inside- and outside-head localisation of a single sound source in the horizontal plane. Both models consider free-field ITDs and ILDs as the memory of sound localisation to which the target interaural disparity is compared. The first model, the characteristic-curve (CC) model acquires the best estimate of a source location by finding the nearest-neighbour of the target ITD and ILD in the characteristic curve of free-field interaural disparities. On the other hand, the second model, the pattern-matching (PM) model, assumes that the excitation-inhibition cell activity pattern suggested by Breebaart et al. [J. Acoust. S. Am., 110(2):1074-1088, 2001] provides the internal representation of the sound localisation cues. Given the uniqueness of EI-patterns, the pattern-matching process operates in each auditory frequency band to give an estimate of the sound source position, which is then frequency-weighted to finally establish the probability function of target location. In the two listening tests presented in the current study, it has been found that both models are capable of predicting many important features of human sound localisation. For example, the inside-head localisation (laterality) of dichotic pure tones has been reasonably well predicted at low source frequencies, 600 Hz and 1200 Hz, by the CC model individualised for each participant. In addition, the prediction of the PM model has been successfully compared to listening test results where the outside-head localisation of the participants was investigated for real and virtual acoustic sources. Given the simplicity and the originality in modelling the central processes of auditory spatial hearing, particularly in handling the ILD information of binaural signals, the predictive scope of the models is regarded as being worthy of further investigation. Furthermore, considering the reasonable predictions made for both lateralisation and localisation of acoustic stimuli, the models developed appear also to be well-suited to the computational evaluation of spatial audio systems.

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Bar-Yehoshua, Gilhad 1970. "Quantifying the effect of dispersion in Continental Shelf sound propagation." Thesis, Massachusetts Institute of Technology, 2002. http://hdl.handle.net/1721.1/91362.

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Gray, Michael Dean. "An experimental investigation of the anomalous behavior of underwater acoustic volume displacement sensors." Thesis, Georgia Institute of Technology, 1992. http://hdl.handle.net/1853/16796.

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Famighetti, Tina Marie. "Investigations into the performance of the reverberation chamber of the integrated acoustics laboratory." Thesis, Available online, Georgia Institute of Technology, 2005, 2005. http://etd.gatech.edu/theses/available/etd-04022005-223652/unrestricted/famighetti%5Ftina%5Fm%5F200505%5Fmast.pdf.

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Thesis (M. S.)--Mechanical Engineering, Georgia Institute of Technology, 2005.
Berthelot, Yves, Committee Member ; Cunefare, Kenneth A, Committee Chair ; Lynch, Christopher, Committee Member. Includes bibliographical references.

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Von, Gossler J. "NVH benchmarking during vehicle development using sound quality metrics." Thesis, Stellenbosch : University of Stellenbosch, 2007. http://hdl.handle.net/10019.1/2955.

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Thesis (MScEng (Mechanical and Mechatronic Engineering))--University of Stellenbosch, 2007.
Noise, Vibrations and Harshness (NVH) characteristics are becoming ever more significant in today’s vehicle manufacturing industry. Similar to good vibration and harshness characteristics, the perception of a vehicle’s quality is enhanced by a well sounding vehicle interior. This study’s main aim was to develop objective equations to directly optimise interior sound quality of light commercial vehicles ( ½ ton LCVs) on the South African market. The effects the noise of the engine, the wind and road/tyre interaction during steady-state conditions have on the interior sound quality of eleven comparable vehicles was investigated with the aid of a binaural head. Steady-state condition in this content refers to the fact that vehicles were tested at constant speeds, no acceleration involved. A strong emphasis was laid on the influence road noise has on the interior sound quality of LCVs. Other objectives for the thesis were, to provide a method to benchmark the interior SQ of LCVs and to develop target values for objective metrics for these vehicles. Establishing a comprehensive literature survey formed another objective of this study. It seeks to provide a summary of the modern SQ analysis procedure and the findings of a number of studies. The survey also presents an opportunity to compare this thesis’s results with previous studies. A last objective was to develop a list of possible hardware modifications that would improve the vehicle interior sound quality, influenced by different noise sources. A strong correlation between vehicle and engine speed and Zwicker loudness as well as Aure sharpness was found, for all test conditions. The road surface roughness was observed to also have a strong influence on the objective metrics of vehicle interior SQ. Loudness was found to be around 25% higher and sharpness around 5.6% lower in vehicles driving on rough tar roads than on smooth roads. Good correlations between a newly developed metric (the SPF), an equation in Zwicker loudness and Aure sharpness, and subjective ratings was obtained for a number of test conditions. Four objective equations, as well as target values for loudness and sharpness have been developed to objectively optimise the sound quality of LCVs. Benchmarking interior sound quality utilising the developed equations, will ensure continuous improvements in the SQ sector for future LCVs.

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49

Wilson, Joshua David. "Quantifying hurricane wind speed with undersea sound." Thesis, Massachusetts Institute of Technology, 2006. http://hdl.handle.net/1721.1/39204.

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Thesis (Ph. D.)--Joint Program in Oceanography/Applied Ocean Science and Engineering (Massachusetts Institute of Technology, Dept. of Mechanical Engineering; and the Woods Hole Oceanographic Institution), 2006.
Includes bibliographical references (p. 155-169).
Hurricanes, powerful storms with wind speeds that can exceed 80 m/s, are one of the most destructive natural disasters known to man. While current satellite technology has made it possible to effectively detect and track hurricanes, expensive 'hurricane-hunting' aircraft are required to accurately classify their destructive power. Here we show that passive undersea acoustic techniques may provide a promising tool for accurately quantifying the destructive power of a hurricane and so may provide a safe and inexpensive alternative to aircraft-based techniques. It is well known that the crashing of wind-driven waves generates underwater noise in the 10 Hz to 10 kHz range. Theoretical and empirical evidence are combined to show that underwater acoustic sensing techniques may be valuable for measuring the wind speed and determining the destructive power of a hurricane. This is done by first developing a model for the acoustic intensity and mutual intensity in an ocean waveguide due to a hurricane and then determining the relationship between local wind speed and underwater acoustic intensity.
(cont.) Acoustic measurements of the underwater noise generated by hurricane Gert are correlated with meteorological data from reconnaissance aircraft and satellites to show that underwater noise intensity between 10 and 50 Hz is approximately proportional to the cube of the local wind speed. From this it is shown that it should be feasible to accurately measure the local wind speed and quantify the destructive power of a hurricane if its eye wall passes directly over a single underwater acoustic sensor. The potential advantages and disadvantages of the proposed acoustic method are weighed against those of currently employed techniques. It has also long been known that hurricanes generate microseisms in the 0.1 to 0.6 Hz frequency range through the non-linear interaction of ocean surface waves. Here we model microseisms generated by the spatially inhomogeneous waves of a hurricane with the non-linear wave equation where a second-order acoustic field is created by first-order ocean surface wave motion. We account for the propagation of microseismic noise through range-dependent waveguide environments from the deep ocean to a receiver on land. We compare estimates based on the ocean surface wave field measured in hurricane Bonnie with seismic measurements from Florida.
by Joshua David Wilson.
Ph.D.

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50

Huang, Zhendong. "On the sound produced by a synthetic jet device." Thesis, Boston University, 2014. https://hdl.handle.net/2144/21179.

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Thesis (M.Sc.Eng.) PLEASE NOTE: Boston University Libraries did not receive an Authorization To Manage form for this thesis or dissertation. It is therefore not openly accessible, though it may be available by request. If you are the author or principal advisor of this work and would like to request open access for it, please contact us at open-help@bu.edu. Thank you.
Synthetic jet is a quasi-steady jet of fluid generated by oscillating pressure drop across an orifice, produced by a piston-like actuator. A unique advantage of the synthetic jet is that it is able to transfer linear momentum without requiring an external fluid source, and has therefore attracted much research within the past decade. Principal applications include aerodynamic flow boundary-layer separation control, heat transfer enhancement, mixing enhancement, and flow-generated sound minimization. In this thesis, the method of deriving the volume flux equation for a duct is first reviewed, combined with this method, a simplified synthetic jet model is presented, based on the principles of aerodynamic sound, the pressure fluctuation in the acoustic far field is predicted. This model is then been used to predict the minimum synthetic jet cavity resonance frequency, acoustic power, acoustic efficiency, root-mean-square jet speed, acoustic spectrum and their dependence on the following independent parameters: the duct length and radius, the aperture radius, the piston vibration frequency, and the maximum piston velocity.
2031-01-01

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Dissertations / Theses on the topic 'Sound engineering'

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