H παρούσα διατριβή εξετάζει το πρόβλημα της διαρροής μικροφώνου, δηλαδή την αλληλεπίδραση και παρεμβολή μεταξύ ταυτόχρονα ενεργών ηχητικών πηγών σε πολυκαναλικές ηχητικές διατάξεις. Παρ' όλο που είναι ένα πολύ συχνό φαινόμενο με το οποίο οι μηχανικοί ήχου έρχονται αντιμέτωποι καθημερινά, δεν έχουν προταθεί μέθοδοι επεξεργασίας σήματος για την επίλυση του προβλήματος. Εδώ, το πρόβλημα διατυπώνεται για πρώτη φορά στο πλαίσιο της επεξεργασίας σήματος. Αρχικά, διατυπώνεται στο πλαίσιο του τυφλού διαχωρισμού πηγών (blind source separation) και αναλύονται οι περιορισμοί αυτής της προσέγγισης. Στην συνέχεια, το πρόβλημα επαναδιατυπώνεται σαν πρόβλημα σήματος υπό θόρυβο στα πλαίσια της καταστολής θορύβου. Ένα πρωτότυπο γενικευμένο πλαίσιο καταστολής διαρροής μικροφώνου εξάγεται βασιζόμενο σε ένα φίλτρο Wiener με πολυκαναλικό όρο θορύβο, καθώς και την ευρέως χρησιμοποιούμενη τεχνική «κοντινού μικροφώνου». Το ακουστικό σύστημα που μοντελοποιεί την διαδικασία μίξης και αλληλεπίδρασης των πηγών αναλύεται και γίνεται διαχωρισμός των σχετικών κρουστικών αποκρίσεων χώρου (room impulse responses) σε απ' ευθείας ακουστικά μονοπάτια και ακουστικά μονοπάτια διαρροής. Οι ιδιότητες του απ' ευθείας ακουστικού μονο- πατιού, δηλαδή της απόκρισης «κοντινού μικροφώνου» αναλύονται για πρώτη φορά από την προσέγγιση της επεξεργασίας σήματος και της ακουστικής κλειστών χώρων για πρώτη φορά. Οι ιδιότητες του ακουστικού μονοπατιού διαρροής αναλύονται επίσης για πρώτη φορά με την χρήση ακουστικών παραμέτρων. Έχοντας καθορίσει τις βασικές ιδιότητες του ακουστικού συστήματος, μια μέθοδος για την καταστολή διαρροής μικροφώνου αναπτύσσεται για μια διάταξη δύο καναλιών, βασισμένη σε ένα φίλτρο Wiener και μια άμεση εκτίμηση των σχετικών πυκνοτήτων φασματικής ενέργεiας (power spectral density). Η απόδοση της μεθόδου για ηχογραφήσεις σε πραγματικούς χώρους είναι πολύ ικανοποιητική και με βάση αυτά τα αποτελέσματα, η μέθοδος επεκτείνεται για περισσότερες από δύο πηγές και μικρόφωνα σε αυθαίρετες διατάξεις. Η ολοκληρωμένη μέθοδος είναι τυφλή και αυτόματη, καθώς δεν απαιτεί την επέμβαση του χρήση. Δεν κάνει χρήση πρότερης γνώσης ούτε απαιτεί εκπαίδευση και είναι υπολογιστικά απλή. Προτείνεται επίσης μια πρωτότυπη μέθοδος ανίχνευσης χρονικών διαστημάτων όπου μόνο μια πηγή είναι ενεργή (χρονικά διαστήματα «σόλο»), η οποία επιτρέπει την εκτίμηση συντελεστών στάθμισης οι οποίοι αντιστοιχούν στην σχετική μείωση της ηχητικής στάθμης που υφίσταται κάθε ηχητική πηγή καθώς το σήμα διαδίδεται προς τα μικρόφωνα. Αυτή η μέθοδος σε συνδυασμό με μια νεά, πρωτότυπη τεχνική εκτίμησης των πυκνοτήτων φασματικής ενέργεαις, η οποία βασίζεται στην αναγνώριση των κυρίαρχων διακριτών συχνοτήτων, επιτρέπει την εκτίμηση όλων των σχετικών ποσοτήτων σε μια πολυκαναλική ηχητική διάταξη. Από αυτές υπολογίζεται ένα πολυκαναλικό φίλτρο Wiener για κάθε σήμα μικροφώνου, το οποίο δίνει την εκτίμηση του αντίστοιχου σήματος πηγής. / This thesis examines the problem of microphone leakage, that is the interference between simultaneously active sound sources in multichannel audio applications. Despite being a common problem with which sound engineers are confronted every day, almost no signal processing methods have been proposed to address this issue. In this work, the problem is formulated for the first time in a signal processing framework. First, it formulated inside the blind source separation (BSS) context and the limitations of related methods are analysed and reported. Since, BSS methods seem to be inappropriate for this specific problem, it is reformulated as a signal in noise problem inside the well-known noise suppression framework. Based on the widely adopted close-microphone technique a novel, generalized framework for leakage suppression is derived based on a multichannel Wiener filter. The acoustic system that models the mixing process is analysed and the related room impulse responses are discerned in direct and leakage acoustic paths. The properties of the direct acoustic path, that is the close-microphone response are investigated for the first time, from a signal processing point of view as well as a room acoustics perspective. The properties of the leakage acoustic path are also analysed for the first time using room acoustic parameters. After key properties of the acoustic paths have been identified, a method for the suppression of microphone leakage in a two channel audio setup is developed based on aWiener filter and a crude approximation of the related power spectral densities (PSDs). The performance of this method for actual recordings in real reverberant environments is more than adequate and based on these results, the method is extended for more than two sources and microphones in arbitrary arrangements. The complete method is blind and automatic, since it does not require any user input. It does not assume any prior knowledge or require training and is computationally efficient. A novel solo detection method has been developed that allows the estimation of weighting coefficients that correspond to the relative attenuation experienced by sound sources as they travel to each microphone. Combined with a new and advanced PSD estimation method based on the identification of dominant frequency bins, the related PSDs in a multichannel audio application can be identified. From these an appropriate multichannel Wiener filter for each microphone signal can be calculated, which will provide the estimated source signal at its output.

Contact microphones, especially throat microphones, have been developed to be used in environments with high background noises to improve the speech intelligibility in communication. They pick up vibrations from the surface they are attached to and are therefore less sensitive to sound and noise from the air. Comparison of the speech intelligibility with other types of microphones have previously been done by letting test persons examine the communication devices in question. This study examines the possibility to make use of the STI-method instead and therefore make a comparison faster, more cost-efficient and customizable. The thought is that if the relationship between speech signals and vibrations were known, it could be used to transform the STI test signal to vibrations and then use the STImethod to estimate the speech intelligibility for the chosen contact microphone. This study, containing 22 men and women, evaluated the vibrations at the most suitable locations on the head for contact microphones and compared it with the speech signal of the same test person. Frequency responses were calculated for all locations of the head and a more detailed study showed that the frequency response of the neck may be approximated as a second order lowpass filter with a cut-off frequency of about 300 Hz that attenuates speech signals with higher frequencies. Experiments were also done to measure the STI value of a throat microphone with the known relationship. However, the results pointed out several problems that needs to be addressed before a STI method can be performed successfully. The results from this study may also be used to deeper study the relationships between different vibrations resulting from speech signals and suggestions on how the performance of contact microphones may be improved are given.

MEMS microphones have been a research topic for the last two and half decades. The state-of-the-art comprises surface mount MEMS microphones in laptops, mobile phones and tablets, etc. The popularity and the commercial success of MEMS microphones is largely due to the steep cost reduction in manufacturing afforded by the mass scale production with microfabrication technology. The current MEMS microphones are de-signed along the lines of traditional microphones that use capacitive transduction with or without permanent charge (electret type microphones use permanent charge of their sensor element). These microphones offer high sensitivity, stability and reasonably at frequency response while reducing the overall size and energy consumption by exploiting MEMS technology. Conceptually, microphones are simple transducers that use a membrane or diaphragm as a mechanical structure which deflects elastically in response to the incident acoustic pressure. This dynamic deflection is converted into an electrical signal using an appropriate transduction technique. The most popular transduction technique used for this application is capacitive, where an elastic diaphragm forms one of the two parallel plates of a capacitor, the fixed substrate or the base plate being the other one. Thus, there are basically two main elements in a microphone { the elastic membrane as a mechanical element, and the transduction technique as the electrical element. In this thesis, we propose and study novel design for both these elements. In the mechanical element, we propose a simple topological change by introducing slits in the membrane along its periphery to enhance the mechanical sensitivity. This simple change, however, has significant impact on the microphone design, performance and its eventual cost. Introduction of slits in the membrane makes the geometry of the structural element non-trivial for response analysis. We devote considerable effort in devising appropriate modeling techniques for deriving lumped parameters that are then used for simulating the system response. For transduction, we propose and study an FET (Field Effect Transistor) coupled micro-phone design where the elastic diaphragm is used as the moving (suspended) gate of an FET and the gate deflection modulated drain current is used in the subthreshold regime of operation as the output signal of the microphone. This design is explored in detail with respect to various design parameters in order to enhance the electrical sensitivity. Both proposed changes in the microphone design are motivated by the possibilities that the microfabrication technology offers. In fact, the design proposed here requires further developments in MEMS technology for reliably creating gaps of 50-100 nm between the substrate and a large 2D structure of the order of a few hundred microns in diameter. In the First part of the thesis, we present detailed simulations of acoustic and squeeze lm domain to understand the effect slits could bring upon the behaviour of the device as a microphone. Since the geometry is nontrivial, we resort to Finite element simulations using commercial packages such as COMSOL Multiphysics and ANSYS in the structural, acoustic and Fluid-structure domains to analyze the behaviour of a microphone which has top plate with nontrivial geometry. On the simulated Finite element data, we conduct low and high frequency limit analysis to extract expressions for the lumped parameters. This technique is well known in acoustics. We borrow this technique of curve Fitting from the acoustics domain and apply it in modified form into the squeeze lm domain. The dynamic behaviour of the entire device is then simulated using the extracted parameters. This helps to simulate the microphone behaviour either as a receiver or as a transmitter. The designed device is fabricated using MEMSCAP PolyMUMPS process (a foundry Polysilicon surface micromachining process). We conduct vibrometer (electrostatic ex-citation) and acoustic characterization. We also study the feasibility of a microphone with slits and the issues involved. The effect of the two dissipation modes (acoustic and squeeze lm ) are quantified with the experimentally determined quality factor. The experimentally measured values are: Resonance is 488 kHz (experimentally determined), low frequency roll-off is 796 Hz (theoretical value) and is 780 Hz as obtained by electrical characterization. The first part of this thesis focusses on developing a comprehensive understanding of the effect of slits on the performance of a MEMS microphone. The presence of slits near the circumference of the clamped plate cause reduction in its rigidity. This leads to an increase in the sensitivity of the device. Slits also cause pressure equalization between the top and bottom of the diaphragm if the incoming sound is at relatively low frequencies. At this frequency, also known as the lower cutoff frequency, the microphone's response starts dropping. The presence of slits also changes the radiation impedance of the plate as well as the squeeze lm damping below the plate. The useful bandwidth of the microphone changes as a consequence. The cavity formed between the top plate and the bottom fixed substrate increases the stiffness of the device significantly due to compression of the trapped air. This effect is more pronounced here because unlike the existing capacitive MEMS microphones, there is no backchamber in the device fabricated here. In the second part of the thesis, we present a novel subthreshold biased FET based MEMS microphone. This biasing of the transistor in the subthreshold region (also called as the OFF-region) offers higher sensitivity as compared to the above threshold region (also called as the ON-region) biasing. This is due to the exponentially varying current with change in the bias voltage in the OFF-region as compared to the quadratic variation in the ON-region. Detailed simulations are done to predict the behaviour of the device. A lumped parameter model of the mechanical domain is coupled with the drain current equations to predict the device behaviour in response to the deflection of the moving gate. From the simulations, we predict that the proposed biasing offers a device sensitive to even sub-nanometer deflection of the flexible gate. As a proof of concept, we fabricate fixed-fixed beams which utilize CMOS-MEMS fabrication. The process involves six lithography steps which involve two CMOS and the remaining MEMS fabrication. The fabricated beams are mechanically characterized for resonance. Further, we carry out electrical characterization for I-V (current-voltage) characteristics. The second part of the thesis focusses on a novel biasing method which circumvents the need of signal conditioning circuitry needed in a capacitive based transduction due to inbuilt amplification. Extensive simulations with equivalent circuit has been carried out to determine the increased sensitivity and the role of various design variables.

In teleconferencing the audio quality is key to conducting successful meetings. The conference room setting imposes various challenges on the speech signal processing, such as noise and interfering signals, reverberation, or participants positioned far from the telephone unit. This work aims at improving the received speech signal of a conference telephone by implementing sound source localization and beamforming. The implemented microphone array signal processing techniques are compared to the performance of an existing multi-microphone solution and evaluated under various conditions using a planar uniform circular array. Recordings of test-sequences for the evaluation were performed using a custom-built array mockup. The implemented algorithms did not show good enough performance to motivate the increased computational complexity compared to the existing solution. Moreover, an increase in number of microphones used was concluded to have little or no effect on the performance of the methods. The type of microphone used was, however, concluded to have impact on the performance and a subjective listening evaluation indicated a preference for omnidirectional microphones which is recommended to investigate further. / God ljudkvalitet är en grundsten för lyckade telefonmöten. Miljön i ett konferens-rum medför ett flertal olika utmaningar för behandlingen av mikrofonsignalerna: det kan t.ex. vara brus och störningar, eller att den som talar befinner sig långt från telefonen. Målet med detta arbete är att förbättra den talsignal som tas upp av en konferenstelefon genom att implementera lösningar för lokalisering av talaren och riktad ljudupptagning med hjälp av ett flertal mikrofoner. De implementerade metoderna jämförs med en befintlig lösning och utvärderas under olika brusscenarion för en likformig cirkulär mikrofonkonstellation. För utvärderingen användes testsignaler som spelades in med en specialbyggd enhet. De implementerade algoritmerna kunde inte uppvisa en tillräcklig förbättring i jämförelse med den befintliga lösningen för att motivera den ökade beräkningskomplexitet de skulle medföra. Dessutom konstaterades att en fördubbling av antalet mikrofoner gav liten eller ingen förbättring på metoderna. Vilken typ av mikrofon som användes konstaterades däremot påverka resultatet och en subjektiv utvärdering indikerade en preferens för de rundupptagande mikrofonerna, en skillnad som föreslås undersökas vidare.

Nas últimas décadas, a poluição sonora tornou-se um grande problema para a sociedade. É por esta razão que a indústria tem aumentado seus esforços para reduzir a emissão de ruído. Para fazer isso, é importante localizar quais partes das fontes sonoras são as que emitem maior energia acústica. Conhecer os pontos de emissão é necessário para ter o controle das mesmas e assim poder reduzir o impacto acústico-ambiental. Técnicas como \"beamforming\" e \"Near-Field Acoustic Holography\" (NAH) permitem a obtenção de imagens acústicas. Essas imagens são obtidas usando um arranjo de microfones localizado a uma distância relativa de uma fonte emissora de ruído. Uma vez adquiridos os dados experimentais pode-se obter a localização e magnitude dos principais pontos de emissão de ruído. Do mesmo modo, ajudam a localizar fontes aeroacústicas e vibro acústicas porque são ferramentas de propósito geral. Usualmente, estes tipos de fontes trabalham em diferentes faixas de frequência de emissão. Recentemente, foi desenvolvida a transformada de Kronecker para arranjos de microfones, a qual fornece uma redução significativa do custo computacional quando aplicada a diversos métodos de reconstrução de imagens, desde que os microfones estejam distribuídos em um arranjo separável. Este trabalho de mestrado propõe realizar medições com sinais reais, usando diversos algoritmos desenvolvidos anteriormente em uma tese de doutorado, quanto à qualidade do resultado obtido e à complexidade computacional, e o desenvolvimento de alternativas para tratamento de dados quando alguns microfones do arranjo apresentarem defeito. Para reduzir o impacto de falhas em microfones e manter a condição de que o arranjo seja separável, foi desenvolvida uma alternativa para utilizar os algoritmos rápidos, eliminando-se apenas os microfones com defeito, de maneira que os resultados finais serão obtidos levando-se em conta todos os microfones do arranjo. / In recent decades, noise pollution has become a major problem for society. It is for this reason that the industry has increased its efforts to reduce the emission of noise. To do this, it is important to find out which parts of the sound sources are emitting greater acoustic energy. Knowing the emission points is required to keep track of them and thus be able to reduce acoustic and environmental impact. Techniques such as \"beamforming\" and \"Near-Field Acoustic Holography\" (NAH) allow obtaining acoustic images. These images are obtained using an array of microphones located at a relative distance from a noise source. Once acquired experimental data, one can obtain the location and the magnitude of the main points of noise emission. Similarly, we can find aeroacoustic and vibroacoustic sources, because they are general-purpose tools. Usually, these types of sources work in different frequency bands of the emission. Recently, a Kronecker Array Transform (KAT) was developed to microphone arrays, which provides a significant reduction in computational cost when applied to various image reconstruction methods, provided that the microphones are distributed in a separable array. This thesis proposes perform measurements with real signals using different algorithms previously developed in a dissertation about the quality of the obtained results and computational complexity, and the development of alternatives for data processing when some microphones of the array are defective. To reduce the impact of failures in microphones and maintain the condition that the array is separable, one alternative has been developed to use the fast algorithms by removing only the defective microphones, so that the final results will be obtained taking into account every microphone of the array.

Nas últimas décadas, a poluição sonora tornou-se um grande problema para a sociedade. É por esta razão que a indústria tem aumentado seus esforços para reduzir a emissão de ruído. Para fazer isso, é importante localizar quais partes das fontes sonoras são as que emitem maior energia acústica. Conhecer os pontos de emissão é necessário para ter o controle das mesmas e assim poder reduzir o impacto acústico-ambiental. Técnicas como \"beamforming\" e \"Near-Field Acoustic Holography\" (NAH) permitem a obtenção de imagens acústicas. Essas imagens são obtidas usando um arranjo de microfones localizado a uma distância relativa de uma fonte emissora de ruído. Uma vez adquiridos os dados experimentais pode-se obter a localização e magnitude dos principais pontos de emissão de ruído. Do mesmo modo, ajudam a localizar fontes aeroacústicas e vibro acústicas porque são ferramentas de propósito geral. Usualmente, estes tipos de fontes trabalham em diferentes faixas de frequência de emissão. Recentemente, foi desenvolvida a transformada de Kronecker para arranjos de microfones, a qual fornece uma redução significativa do custo computacional quando aplicada a diversos métodos de reconstrução de imagens, desde que os microfones estejam distribuídos em um arranjo separável. Este trabalho de mestrado propõe realizar medições com sinais reais, usando diversos algoritmos desenvolvidos anteriormente em uma tese de doutorado, quanto à qualidade do resultado obtido e à complexidade computacional, e o desenvolvimento de alternativas para tratamento de dados quando alguns microfones do arranjo apresentarem defeito. Para reduzir o impacto de falhas em microfones e manter a condição de que o arranjo seja separável, foi desenvolvida uma alternativa para utilizar os algoritmos rápidos, eliminando-se apenas os microfones com defeito, de maneira que os resultados finais serão obtidos levando-se em conta todos os microfones do arranjo. / In recent decades, noise pollution has become a major problem for society. It is for this reason that the industry has increased its efforts to reduce the emission of noise. To do this, it is important to find out which parts of the sound sources are emitting greater acoustic energy. Knowing the emission points is required to keep track of them and thus be able to reduce acoustic and environmental impact. Techniques such as \"beamforming\" and \"Near-Field Acoustic Holography\" (NAH) allow obtaining acoustic images. These images are obtained using an array of microphones located at a relative distance from a noise source. Once acquired experimental data, one can obtain the location and the magnitude of the main points of noise emission. Similarly, we can find aeroacoustic and vibroacoustic sources, because they are general-purpose tools. Usually, these types of sources work in different frequency bands of the emission. Recently, a Kronecker Array Transform (KAT) was developed to microphone arrays, which provides a significant reduction in computational cost when applied to various image reconstruction methods, provided that the microphones are distributed in a separable array. This thesis proposes perform measurements with real signals using different algorithms previously developed in a dissertation about the quality of the obtained results and computational complexity, and the development of alternatives for data processing when some microphones of the array are defective. To reduce the impact of failures in microphones and maintain the condition that the array is separable, one alternative has been developed to use the fast algorithms by removing only the defective microphones, so that the final results will be obtained taking into account every microphone of the array.

<p>In modern cars a lot of effort is put on reducing the background noise level. Despite these efforts it is often difficult for persons in the rear seat(s) to hear the persons in the front seat. This is partly due to the background noise, but also geometry and acoustics properties of the passenger compartment. </p><p>The aim of this thesis was to implement a speech enhancement system to increase the audibility between the driver and the rear passenger(s). The speech enhancement system should not affect the directivity of the speech or increase the background noise level. </p><p>A speech enhancement system has been implemented on a DSP in a test car. A microphone was placed in front of the driver to collect his/her speech. The microphone signal was bandpass filtered to remove the main part of the background noise and to avoid aliasing. The signal was delayed before it was sent out in the rear loudspeaker. The delay made the speech from the driver reaching the rear passenger before the sound the rear loudspeakers. This delay was enough to get the right directivity of the sound, i.e. making speech sounding as if it came from the driver instead of the rear loudspeakers. </p><p>In the thesis other methods to reduce background noise and get directivity of the sound were evaluated, but not implemented in the test car. The evaluations of the system showed that the audibility was increased. At the same time the background noise level was not noticeable increased. The work has been performed at A2 Acoustics AB in Linköping, during spring 2003.</p>

L'aéroacoustique est une filière de l'acoustique qui étudie la génération de bruit par un mouvement fluidique turbulent ou par les forces aérodynamiques qui interagissent avec les surfaces. Ce secteur en pleine croissance a attiré des intérêts récents en raison de l'évolution de la transportation aérienne, terrestre et spatiale. Alors que les tests sur un objet réel sont possibles, leur implantation est généralement compliquée et les résultats sont facilement corrompus par le bruit ambiant. Par conséquent, les tests plus strictement contrôlés au laboratoire utilisant les modèles de dimensions réduites sont préférables. Toutefois, lorsque les dimensions sont réduites par un facteur de M, l'amplitude et la bande passante des ondes acoustiques correspondantes se multiplient respectivement par 10logM en décibels et par M. Les microphones avec une bande passante de plusieurs centaines de kHz et une plage dynamique couvrant de 40Pa à 4 kPa sont ainsi nécessaires pour les mesures aéroacoustiques. Les microphones MEMS ont été étudiés depuis plus de vingt ans, et plus récemment, l'industrie des semiconducteurs se concentre de plus en plus sur ce domaine. Par rapport à tous les autres principes de fonctionnement, grâce à la miniaturisation, les microphones de type piézorésistif peuvent atteindre une bande passante plus élevée et ils sont ainsi bien adaptés pour les mesures aéroacoustiques. Dans cette thèse, deux microphones MEMS de type piézorésistif à base de silicium polycristallin (poly-Si) latéralement cristallisé par l'induction métallique (MILC) sont conçus et fabriqués en utilisant respectivement les techniques de microfabrication de surface et de volume. Ces microphones sont calibrés à l'aide d'une source d'onde de choc (N-wave) générée par une étincelle électrique. Pour l'échantillon fabriqué par le micro-usinage de surface, la sensibilité statique mesurée est de 0.4 μV/V/Pa, la sensibilité dynamique est de 0.033 μV/V/Pa et la plage fréquentielle commence à 100 kHz avec une fréquence du premier mode de résonance à 400 kHz. Pour l'échantillon fabriqué par le micro-usinage de volume, la sensibilité statique mesurée est de 0.28 μV/V/Pa, la sensibilité dynamique est de 0.33 μV/V/Pa et la plage fréquentielle commence à 6 kHz avec une fréquence du premier mode de résonance à 715 kHz.

In modern cars a lot of effort is put on reducing the background noise level. Despite these efforts it is often difficult for persons in the rear seat(s) to hear the persons in the front seat. This is partly due to the background noise, but also geometry and acoustics properties of the passenger compartment. The aim of this thesis was to implement a speech enhancement system to increase the audibility between the driver and the rear passenger(s). The speech enhancement system should not affect the directivity of the speech or increase the background noise level. A speech enhancement system has been implemented on a DSP in a test car. A microphone was placed in front of the driver to collect his/her speech. The microphone signal was bandpass filtered to remove the main part of the background noise and to avoid aliasing. The signal was delayed before it was sent out in the rear loudspeaker. The delay made the speech from the driver reaching the rear passenger before the sound the rear loudspeakers. This delay was enough to get the right directivity of the sound, i.e. making speech sounding as if it came from the driver instead of the rear loudspeakers. In the thesis other methods to reduce background noise and get directivity of the sound were evaluated, but not implemented in the test car. The evaluations of the system showed that the audibility was increased. At the same time the background noise level was not noticeable increased. The work has been performed at A2 Acoustics AB in Linköping, during spring 2003.

The two-noise source model for predicting jet noise claims that the radiated jet noise is composed of two distinct sources one associated with the small-scale turbulence and another associated with the large-scale turbulence. The former source is claimed to radiate noise predominantly at larger angles with respect to the downstream jet axis, whereas the large-scale turbulence radiates predominantly at the shallower angles. A key objective of this effort is to experimentally validate this model using correlation and coherence measurements. Upon the successful validation of the two-noise source model for jets exhausting from multiple nozzle geometries driven at Mach numbers ranging from subsonic to supersonic, a three-microphone signal enhancement technique is employed to separate the contribution of the small-scale turbulence from that of the large-scale turbulence in the far-field. This is the first-ever quantitative separation of the contributions of the turbulence scales in far-field jet noise measurements. Furthermore, by suitable selection of far-field microphone positions, the separation of the contribution of any internal or core noise from that of the jet-mixing noise is achieved. Using coherence-based techniques to separate the contributions of the small-scale turbulence, large-scale turbulence, and any internal or core noise from far-field exhaust noise measurements forms the backbone of this effort. In the application of coherence-based multiple-microphone signal processing techniques to separate the contributions of the small-scale turbulence, large-scale turbulence, and any internal or core noise in the far-field, research efforts focus on three techniques (1) the coherent output power spectrum using two microphones, (2) an ordinary coherence method using the three-microphone technique, and (3) the partial-coherence method using five microphones. The assumption of jet noise incoherence between correlating microphone is included in each of these methods. In light of the noise radiation mechanisms described within the framework of the two-noise source model and their spatial characteristics as experimentally determined in the far-field, the assumption of jet noise incoherence is evaluated through a series of experiments designed to study jet noise coherence across a variety of nozzle geometries and jet Mach numbers ranging from subsonic to supersonic. Guidelines for the suitable selection of far-field microphone locations are established.