During the design and commissioning of combustion equipment, combustion associated instabilities are commonly encountered. These thermo-acoustic instabilities can cause undesirable noise, vibrations, local thermal and mechanical stresses in the combustor and are prominent in lean combustion. An often used mathematical tool to predict the instability in combustors is the so called network model where the system under study is subdivided in several subsystems and the acoustic state variables are regarded as the input/output of these subsystems. Solving this system of equations gives rise to the complex Eigen-frequencies of the system which tell if the complete system will have an unstable/stable response for specific operating conditions. In such a model it is critical to know what the influence of each subpart is on the acoustic wave propagation to correctly predict the unstable frequencies of the system. The area expansion is a common element found in combustors and the acoustic properties of the area expansion under quiescent conditions are well known, however in the presence of flow, acoustic flow interactions may occur. These interactions change the acoustic properties and are challenging to model and accurate experimental data is needed to validate the modelling. In this study, measurements of the aero-acoustic properties of an area expansion are presented, however the focus is on the experimental techniques and methods used to obtain accurate and precise measurement data in the plane wave frequency regime. The measurement accuracy of the setup used to determine the passive aero-acoustic properties of the area expansion is assessed by measuring a known impedance. Several sources of errors are identified and methods to account for these error sources are given. It is shown that the microphone impedance affects the measurement results and the upper limit of the measurement accuracy for quiescent measurements is governed by this error. The measurement precision of the setup is assessed using a multi-variate analysis and compared with results obtained from a Monte-Carlo simulation. Also the problem to determine the uncertainty of the measured complex pressures receives attention. Using a framework based on the Hilbert-transform, expressions are derived which estimate the uncertainty on the measured complex value from the background signal spectrum. The obtained knowledge is used to determine the scattering matrix of the area expansion. For the quiescent case, the measured results agree within 1.5% of the absolute values and within 1 degree in comparison with the analytical models. In the case with flow, the errors are slightly larger due to the increased flow-noise but a good correspondence with analytical models is found. Also a sudden sound absorption at high flow speeds and low frequencies is observed. / Vid konstruktion och användning av förbränningsutrustning observeras ofta instabiliteter i förbränningsprocessen. Dessa så kallade termo-akustiska instabiliteter kan orsaka buller och vibrationer samt lokala termiska och mekaniska belastningar i förbränningskammaren och de uppstår ofta vid mager förbränningen. Nätverksmodeller används ofta att förutsäga när instabiliteter uppträder och i dessa modeller är hela systemet indelat i olika delsystem och de akustiska ljudfältsstorheterna används som ingående och utgående parametrar. Löser man hela modellen, så får man de komplexa egenfrekvenser som ger information om när systemet är instabilt för vissa operativa förhållanden. I de här modellerna är det kritiskt att veta hur varje delsystem påverkar det akustiska ljudfältet om man vill få korrekta förutsägelser av de frekvenser där systemet kommer att vara instabilt. Ett areasprång är ett element som man ofta hittar i förbrännings-utrustning. Dess akustiska egenskaper är välkända utan strömning, men med strömning kan växelverkan mellan strömning och akustisk tillkomma och påverka de akustiska egenskaperna. Denna inverkan är komplex att modellera och det finns behov av noggranna mätdata för att validera sådana modeller. Denna avhandling presenterar experimentellt bestämda akustiska egenskaper för ett areasprång, men fokus är på förbättringar av de experimentella tekniker och metoder som används för att få precisa och noggranna mätdata för akustiska delelement i planvågsområdet. Den experimentella noggrannheten hos mätuppställningen bedöms genom att mäta upp en bekant impedans. Flera felkällor identifieras och metoder att utvärdera felen presenteras. Dessutom visar det sig att mikrofonimpedansen påverkar mätresultaten och impedansen begränsar mätnoggrannheten i fallet med mätningar utan strömning. Precisionen av mätuppställningen bestäms med hjälp av en multivariat analys som jämförs med en Monte-Carlo analys. Det visar sig att det är lämpligt att använda en multivariat analys för att bestämma noggrannhets-intervaller för de uppmätta spridningskoefficienterna för areasprånget. Problemet att bestämma noggrannhet för det uppmätta komplexa ljud trycket behandlas också. Med hjälp av ett ramverk baserat på Hilbert-transformen, härleds uttryck för uppskattning av noggrannheten i det uppmätta ljud trycket baserat på bakgrundsljudets spektrum. De erhållna kunskaperna används sedan för att bestämma de akustiska egenskaperna för ett areasprång. De uppmäta resultaten är inom 1.5% för absolutvärdet och 1 grad för fasen, jämfört med analytiska modeller i fallet utan strömning. Med strömning är noggrannheten något sämre, beroende på strömningsgenererat brus, men resultaten stämmer fortfarande bra överens med modellen. Dessutom observeras en plötslig ljudsabsorption för högre strömhastigheter och låga frekvenser. / <p>QC 20150522</p>

To date electric pianos and samplers tend to concentrate on authenticity in terms of temporal and spectral aspects of sound. However, they barely recreate the original sound radiation characteristics, which contribute to the perception of width and depth, vividness and voice separation, especially for instrumentalists, who are located near the instrument. To achieve this, a number of sound field measurement and synthesis techniques need to be applied and adequately combined. In this paper we present the theoretic foundation to combine so far isolated and fragmented sound field analysis and synthesis methods to realize a radiation keyboard, an electric harpsichord that approximates the sound of a real harpsichord precisely in time, frequency, and space domain. Potential applications for such a radiation keyboard are conservation of historic musical instruments, music performance, and psychoacoustic measurements for instrument and synthesizer building and for studies of music perception, cognition, and embodiment.

No description available.

This doctoral thesis deals with the diagnostics of combustion engines, which have clearly suffered a mechanical damage. The text describes a basic principle of a diagnostics based on an observation of a noise of a running engine. There are mentioned the basic methods of a noise recording with direct conection to the specifics of measurement a running combustion engine noise. The introduction describes the spread of a vibration in solid structures. For the combustion engine vibration and source of noise is dedicated an independent chapter, which focuses on all noise generators, not only mechanical and thermodynamic but also noise caused by fluid and gas flowing. An integral part of the thesis is also a list and examples of signal processing, serving for efficient localization of the noise root cause. The noise statistics are compared with event diagram of engine investigation. Considering the information of the crank shaft position, we are then able to localize a possible root cause of the mechanical problem. For this purpose unique computer software for MS Windows operation system was developed. This system is able to generate event timelines of the piston engines. The inputs are basic engine data, such as the number of cylinders, ignition sequence, valve timing diagrams and information regarding two stroke or four stroke engine types. The program also allows us to detect a potential damage by direct comparison of the engine timeline and noise signal input. The thesis is also supplemented by measurements serving for key-parameter-diagnostics, i.e. the selection of the most suitable running engine operation. Comparison of different technical diagnostic approaches shows the suitability of mentioned diagnostic process.

京都大学 / 0048 / 新制・課程博士 / 博士(情報学) / 甲第18412号 / 情博第527号 / 新制||情||93(附属図書館) / 31270 / 京都大学大学院情報学研究科知能情報学専攻 / (主査)教授 奥乃 博, 教授 河原 達也, 准教授 CUTURI CAMETO Marco, 講師 吉井 和佳 / 学位規則第4条第1項該当 / Doctor of Informatics / Kyoto University / DFAM

No description available.

Phased microphone arrays are valuable tools for aeroacoustic measurements that can measure the directivity of multiple acoustic sources. However, when deployed in closed test-section wind tunnels, the acoustics suffer due to intense pressure fluctuations contained in the wall-bound turbulent boundary layer. Furthermore, phased microphone arrays require many sensors distributed over a large aperture to ensure good spatial resolution over a wide frequency range. Microphone arrays of such large count are not always feasible due to constraints in space and cost. This thesis describes an alternative approach for measuring single broadband acoustic sources that uses an acoustic metasurface. The metasurface is comprised of a meandering channel of quarter-wave cavities and an array of equally spaced half-wave open through-cavities. A series of tests were conducted in Virginia Tech's Anechoic Wall-Jet Tunnel where combinations of a wall-bound turbulent jet-flow and a single broadband acoustic source were used to excite the metasurface and produce acoustic surface waves. Measurements of the acoustic surface waves were performed using two methods: a pair of traversing microphones scanning the pressure field along the length of the metasurface 0.25 mm beneath its bottom face, and an array of unequally spaced microphones embedded inside the metasurface. Spectral analysis on the measurements revealed that the inclusion of multiple through-cavities leads to constructive reinforcement of select acoustic surface waves as a function of the acoustic source location. In the case of the embedded microphones, acoustic beamforming was applied in order to extract spatial information. This reinforcement was observed during measurements made with both flow and acoustic excitation, up to Wall-Jet Tunnel nozzle exit speeds of 40 m/s beyond which it was no longer seen. A series of quiescent measurements made with a range of speaker locations constituted a calibration for the metasurface which was used to locate an unknown broadband acoustic source within an The Root-Mean-Square (RMS) error of 1.06 degrees. / Master of Science / Phased microphone arrays are valuable tools for aeroacoustic measurements that can measure the directivity of multiple acoustic sources within a sound field. When used in conjunction with signal processing techniques, such as delay-and-sum beamforming, a researcher or engineer can obtain an intuitive view of the sound field and distinguish between multiple sources over a wide frequency range. However, these microphone arrays often utilize dozens of microphones which raises the array's complexity and cost. Furthermore, when a phased microphone array is mounted flush to the wall of a wind tunnel test section, it is submerged under a turbulent boundary layer which imposes intense pressure fluctuations on the microphones making it difficult to identify acoustic sources. Boundary layers form at the interface between a fluid and solid interface. This thesis describes experimentation performed in the Virginia Tech Anechoic Wall-Jet Tunnel on a new type of pressure sensing microphone array that leverage acoustic metamaterial technology. The acoustic metamaterial shields the microphones from the flow, lessening the influence of the turbulent boundary layer on the measurement. The focus in this thesis is on the novel array's ability to locate a single broadband acoustic source using as few as six microphones. The metasurface was installed in the Wall-Jet Tunnel test plate such that an array of evenly spaced through-cavities are flush to the surface. The through-cavities communicate the pressure field on top of the test surface to a meandering channel of interconnected closed cavities below. Near the resonant depth frequencies of the closed cavities, acoustic surface waves form which are evanescent pressure waves that are bound to the surface or structure that support them. The interference between the acoustic surface waves generated at each through-cavity leads to reinforced acoustic surface waves which are sensitive to the direction of a broadband source. In all, an acoustic metamaterial was tested under a variety of conditions such as: Wall-Jet Tunnel flow speed, speaker location, and the number of through-cavities open. The performance of the novel array and future plans are discussed.

Sound radiation from an acoustic source typically exhibits directional behavior, as is the case for the human voice, musical instruments, and loudspeakers, to name just a few. The necessity of directional data for many applications, such as sound source modeling, microphone placement, room acoustical design, and auralization, motivates directivity measurements. However, these measurements require careful understanding and implementation to produce the most meaningful results. Accordingly, this dissertation addresses several topics relevant to directivity theory, measurement, processing, and application. It first expands and amends previously published concepts of an acoustic source center and demonstrates the close relationship between the center and a source's far-field directional response. This relationship subsequently leads to an acoustic centering method that improves source placements within directivity measurement arrays. The dissertation then addresses several measurement considerations, including the required numbers of sampling positions for directivity measurements, quadrature rules applicable to standardized dual-equiangular sampling schemes, and a source's far-field response from arbitrarily shaped microphone arrays. Selected directivity results for the human voice and musical instruments illustrate applications of the developed measurement theories for procuring high-resolution results over a sphere. Compiled voice and musical instrument directivities now appear in an open-source database for use in room acoustical modeling, microphone placements, and other applications. To better elucidate and help predict sound source radiation, this work proposes several theoretical models, including equivalent point-source models, low-frequency radiation from a radially vibrating cap set on a rigid spherical shell with a circular aperture, and radiation from a vibrating cap on a rigid sphere with imposed mode shapes. Finally, this dissertation presents two microphone placement methods for audio and other applications. The first method approximates the measurement of a source's sound power spectrum through a single-channel measurement; the second considers microphone placement for maximum perceived loudness. The work's various developments, results, and conclusions will assist researchers and practitioners in better evaluating, predicting, and applying sound source directivities for many uses.

Voice commanded systems (VCS) have been proved to be vulnerable to signal in- jections mimicking voice commands and explored security flaws in market avail- able products for the time of each respective study. Signal injection caused with the help of amplitude modulated ultrasonic waves (being known as DolphinAt- tacks - DA) were proved to work on several such devices in 2017. In 2019, another study were also successful in achieving signal injections using modulated laser also known as LightCommands (LC). This thesis has investigated the occurring circumstances which enables such injections. Simulations and laboratory trials have shown a thermoacoustic origin enabling LC to be injected and the response differs with respect to microphones physical size. DA utilizes the non-linearity of microphones and more linear microphones have indeed been shown to withstand DAs better and physical parameters have been shown to indicate how DA may be optimized for successful injections. The results have been used to provide ideas on how a VCS system can be designed to be more resilient. / Röststyrda System har visat sig vara sårbara mot signalinjektioner som härmar röstkommandon och utnyttjar kryphål hos produkter som fanns på marknaden i samtid när studierna som först tog upp kryphålen publicerades. Signalinjek- toner inducerade med hjälp av amplitudmodulerat ultraljud (känt som Dolphi- nAttacks - DA) bevisades fungera på flertalet enheter år 2017. 2019 visade en annan studie framgång med signalinjektion genom modulerad laser, även känt som LightCommands (LC). Detta examensarbete har utrett de bakomliggande faktorer som möjliggör sådana injektioner. Simuleringar och laboratorieexperi- ment har visat att termoakustiska effekter möjliggör LC med resultat som beror på mikrofoners fysiska storlek. DA nyttjar ickelinjäritet hos mikrofoner och linjä- rare mikrofoner har visat sig stå emot DA bättre och det har visat sig att DA kan optimeras för bättre lyckade injektioner. Resultaten har används för att bidra till idéer och resonemang från föregående studier på hur lösningar mot LC och DA skulle kunna implementeras och göra mikrofoner och dess tillhörande system tåligare mot sådana angrepp.

This paper presents the theory, design, and validation of a microphone calibrator used to calibrate multiple microphones simultaneously. This work was done in conjunction with the development of an acoustic energy density probe, which was used to validate the calibrator. The probe uses multiple microphones to acquire the data needed to compute the acoustic energy density. The probe microphones are 0.006 m diameter electret microphones which typically do not have an ideal “flat" response over a wide range of frequencies as compared to precision condenser microphones. The probe microphone characteristics prompted the need for simultaneous, multi-microphone magnitude calibration. The idea behind the calibration process was to simultaneously subject each microphone on the probe to the same known acoustic pressure over the frequency range of the probe (0-2 kHz). This is done using equal-length small-diameter tubes connected to a single source at each microphone. The calibrator was modeled using and equivalent circuit model. The model results are presented and compared to measured results. The calibrator was validated to result in the same pressure along individual paths to each microphone simultaneously. Test results show that the calibrator can calibrate each probe microphone within ± 0.5 dB up to 2000 Hz, and within ± 1 dB up to 4900 Hz with a confidence level of 95%. Directivity tests were performed to further characterize the probe. Using a correction factor the probe is shown to be directionally independent.