Measuring acoustic pressure gradients is critical in many applications such as directional microphones for hearing aids and sound intensity probes. This measurement is especially challenging with decreasing microphone size, which reduces the sensitivity due to small spacing between the pressure ports. Novel, micromachined biomimetic microphone diaphragms are shown to provide high sensitivity to pressure gradients on one side of the diaphragm with low thermal mechanical noise. These structures have a dominant mode shape with see-saw like motion in the audio band, responding to pressure gradients as well as spurious higher order modes sensitive to pressure. In this dissertation, integration of a diffraction based optical detection method with these novel diaphragm structures to implement a low noise optical pressure gradient microphone is described and experimental characterization results are presented, showing 36 dBA noise level with 1mm port spacing, nearly an order of magnitude better than the current gradient microphones. The optical detection scheme also provides electrostatic actuation capability from both sides of the diaphragm separately which can be used for active force feedback. A 4-port electromechanical equivalent circuit model of this microphone with optical readout is developed to predict the overall response of the device to different acoustic and electrostatic excitations. The model includes the damping due to complex motion of air around the microphone diaphragm, and it calculates the detected optical signal on each side of the diaphragm as a combination of two separate dominant vibration modes. This equivalent circuit model is verified by experiments and used to predict the microphone response with different force feedback schemes. Single sided force feedback is used for active damping to improve the linearity and the frequency response of the microphone. Furthermore, it is shown that using two sided force feedback one can significantly suppress or enhance the desired vibration modes of the diaphragm. This approach provides an electronic means to tailor the directional response of the microphones, with significant implications in device performance for various applications. As an example, the use of this device as a particle velocity sensor for sound intensity and sound power measurements is investigated. Without force feedback, the gradient microphone provides accurate particle velocity measurement for frequencies below 2 kHz, after which the pressure response of the second order mode becomes significant. With two-sided force feedback, the calculations show that this upper frequency limit may be increased to 10 kHz. This improves the pressure residual intensity index by more than 15 dB in the 50 Hz-10 kHz range, matching the Class I requirements of IEC 1043 standards for intensity probes without any need for multiple spacers.

Auditory stream denotes the abstract effect a source creates in the mind of the listener. An auditory scene consists of many streams, which the listener uses to analyze and understand the environment. Computer analyses that attempt to mimic human analysis of a scene must first perform Audio Scene Segmentation (ASS). ASS find applications in surveillance, automatic speech recognition and human computer interfaces. Microphone arrays can be employed for extracting streams corresponding to spatially separated sources. However, when a source moves to a new location during a period of silence, such a system loses track of the source. This results in multiple spatially localized streams for the same source. This thesis proposes to identify local streams associated with the same source using auditory features extracted from the beamformed signal. ASS using the spatial cues is first performed. Then auditory features are extracted and segments are linked together based on similarity of the feature vector. An experiment was carried out with two simultaneous speakers. A classifier is used to classify the localized streams as belonging to one speaker or the other. The best performance was achieved when pitch appended with Gammatone Frequency Cepstral Coefficeints (GFCC) was used as the feature vector. An accuracy of 96.2% was achieved.

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. Les microphones avec une bande passante de plusieurs centaines de kHz et une plage dynamique couvrant de 40Pa à 4 kPa sont nécessaires 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 0.4μV/V/Pa, la sensibilité dynamique est 0.033μV/V/Pa et la plage fréquentielle couvre à partir de 100 kHz avec une fréquence du premier mode de résonance à 400kHz. Pour l'échantillon fabriqué par le micro-usinage de volume, la sensibilité statique mesurée est 0.28μV/V/Pa, la sensibilité dynamique est 0.33μV/V/Pa et la plage fréquentielle couvre à partir de 6 kHz avec une fréquence du premier mode de résonance à 715kHz.

L'étude expérimentale de la propagation et du rayonnement multimodal en guide large est abordée via des mesures par antennerie microphonique de la réflexion des modes pour différentes extrémités. Le banc expérimental est constitué d'un guide large fermé à une extrémité et débouchant sur différentes terminaisons à l'autre extrémité ; en paroi du guide sont branchées une source acoustique et deux antennes microphoniques. Chaque composant du banc est étudié pour améliorer les résultats de mesure. Une méthode de vérification des performances des haut-parleurs constituant la source et une méthode de pilotage de la source acoustique sont proposées pour favoriser la génération des différents modes de propagation de l'onde. Une méthode de calibration in-situ pour l'antenne est développée pour les différents modes. Un calcul des incertitudes pour l'estimation du coefficient de réflexion est proposé.Enfin les mesures sont effectuées pour différentes extrémités de guide : avec une bride, sans épaisseur, avec un écran infini. Le principe de la méthode de mesure de la réflexion des différents modes consiste à appliquer la méthode du doublet microphonique adaptée aux signaux issus de la décomposition modale obtenue au moyen de deux antennes de microphones. Les résultats de mesure pour le mode plan sont avantageusement comparés aux résultats théoriques issus de la littérature. Les résultats pour les premiers modes supérieurs montrent l'aptitude du système à extraire le coefficient de réflexion en module et en phase suffisamment précisément pour distinguer l'effet de la condition de rayonnement. / The experimental study of multimodal propagation and radiation in a wide guide is proposed via measurements of the reflection of modes for different terminations by using microphone arrays. The experimental bench consists of a wide guide closed at one end and ended with different terminations at the other end; an acoustic source and two microphone arrays are flush-mounted to the wall of the guide. Each component of the bench is first studied to improve the measurement results. A method of verifying the performance of the loudspeakers constituting the acoustic source and a method of controlling the acoustic source are proposed in order to facilitate the generation of the different modes of propagation of the wave. An in-situ calibration method for the microphone array is developed for the different modes. A calculation of the uncertainties for the estimation of the reflection coefficient is proposed.Then, measurements are performed for different guide terminations: with a finite flange, without flange, and with an infinite flange. The measurement of the reflection for the different modes consists of applying the method of two microphones to the signals from the modal decomposition obtained by means of two microphone arrays. Results of measurements for the plane mode are satisfactorily compared with theoretical results from the literature. Results for the first higher modes show the ability of the system to extract the reflection coefficient in modulus and in phase with sufficient precision to distinguish the effect of the radiation condition.

This thesis is related to the field of acoustic signal processing and its applications to emerging communication environments. Acoustic signal processing is a very wide research area covering the design of signal processing algorithms involving one or several acoustic signals to perform a given task, such as locating the sound source that originated the acquired signals, improving their signal to noise ratio, separating signals of interest from a set of interfering sources or recognizing the type of source and the content of the message. Among the above tasks, Sound Source localization (SSL) and Automatic Speech Recognition (ASR) have been specially addressed in this thesis. In fact, the localization of sound sources in a room has received a lot of attention in the last decades. Most real-word microphone array applications require the localization of one or more active sound sources in adverse environments (low signal-to-noise ratio and high reverberation). Some of these applications are teleconferencing systems, video-gaming, autonomous robots, remote surveillance, hands-free speech acquisition, etc. Indeed, performing robust sound source localization under high noise and reverberation is a very challenging task. One of the most well-known algorithms for source localization in noisy and reverberant environments is the Steered Response Power - Phase Transform (SRP-PHAT) algorithm, which constitutes the baseline framework for the contributions proposed in this thesis. Another challenge in the design of SSL algorithms is to achieve real-time performance and high localization accuracy with a reasonable number of microphones and limited computational resources. Although the SRP-PHAT algorithm has been shown to be an effective localization algorithm for real-world environments, its practical implementation is usually based on a costly fine grid-search procedure, making the computational cost of the method a real issue. In this context, several modifications and optimizations have been proposed to improve its performance and applicability. An effective strategy that extends the conventional SRP-PHAT functional is presented in this thesis. This approach performs a full exploration of the sampled space rather than computing the SRP at discrete spatial positions, increasing its robustness and allowing for a coarser spatial grid that reduces the computational cost required in a practical implementation with a small hardware cost (reduced number of microphones). This strategy allows to implement real-time applications based on location information, such as automatic camera steering or the detection of speech/non-speech fragments in advanced videoconferencing systems. As stated before, besides the contributions related to SSL, this thesis is also related to the field of ASR. This technology allows a computer or electronic device to identify the words spoken by a person so that the message can be stored or processed in a useful way. ASR is used on a day-to-day basis in a number of applications and services such as natural human-machine interfaces, dictation systems, electronic translators and automatic information desks. However, there are still some challenges to be solved. A major problem in ASR is to recognize people speaking in a room by using distant microphones. In distant-speech recognition, the microphone does not only receive the direct path signal, but also delayed replicas as a result of multi-path propagation. Moreover, there are multiple situations in teleconferencing meetings when multiple speakers talk simultaneously. In this context, when multiple speaker signals are present, Sound Source Separation (SSS) methods can be successfully employed to improve ASR performance in multi-source scenarios. This is the motivation behind the training method for multiple talk situations proposed in this thesis. This training, which is based on a robust transformed model constructed from separated speech in diverse acoustic environments, makes use of a SSS method as a speech enhancement stage that suppresses the unwanted interferences. The combination of source separation and this specific training has been explored and evaluated under different acoustical conditions, leading to improvements of up to a 35% in ASR performance. / Martí Guerola, A. (2013). Multichannel audio processing for speaker localization, separation and enhancement [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/33101 / TESIS

Context: Large numbers of bats are killed by collisions with wind turbines and there is at present no accepted method of reducing or preventing this mortality. We designed a system, which detects and records any bats’ activity in and around the surroundings of wind turbines. The system can help to study bats by identifying the species that are present in that particular locality. Objectives: The main objective of this thesis is to design an ultrasound-based IoT system, which detects the bats to prevent them from clashing with wind turbines. The design is based on a study of bats’ behaviors. Methods: The system has been developed using User-Driven Design, UDD, approach. The required functionalities have been embedded into IoT based system. An ultrasonic technology along with other sensors are used. The sensors are intended to activate monitoring during favorable conditions for bat activity. Results: A model of a system has been developed. The model was implemented into a prototype. Recorded bats’ activities are uploaded to a server by employing a suitable app, which informs the user about the activities of bats' various sub-species. Conclusions: A surveillance for bats approaching the wind farms within 80 m has been developed. The monitoring system is activated when the weather conditions are favorable for bat activities.

This thesis explores problematics of speaker localization using microphone array. Aim of this thesis is implementation of algorithms for speaker localization and experiments with those algorithms. Calculation of TDOA was done using cross-correlation and hyperbolic method was used to calculate position estimation. Finished microphone array is able to locate speaker within certain variance. Results of this thesis allow reader to make assumptions regarding accuracy of localisation using microphone array and ARM kit with limited performance. Precision of position estimation using microphone array reached several decimeters, but this precision is dependent on distance from microphone array.

This master's thesis is focused on the noise source identification using microphone arrays and beamforming as the signal processing method. It describes parts of such a system and provides a comparison with other systems that serve a similar purpose (eg. NAH). Various types of microphone arrays are mentioned with their influence on the resulting ability to identify the noise source. We are further focusing on Delay-And-Sum technology, on which we are explaining the basic principles and constraints of beamforming. The practical part describes the implementation of the DAS method in MATLAB and C language, the specific structures of built microphone arrays and assembly of complete systems capable of identifying sources of noise. These systems were tested by performing a practical experiment. Achievements in the form of distribution maps of acoustic energy in the focused space are interpreted in the last chapter.

This master’s thesis deals with problems of noise source identification using nearfield acoustical holography (NAH). In the beginning there is the summary of basic terms and values of a sound pressure field, which is unnecessary for understanding of the theme. In the next part the thesis continues with more detailed description of the NAH technology and the historical context of its emergence. Measurement equipment which is used for scanning of sound pressure fields is also introduced. In addition, the kinds of NAH (according the shape of the wave front) are showed and the planar NAH is descripted most closely. Because of the NAH algorithms are implemented in the wave number domain (k-space), there is also a chapter focused to this problem in the thesis. There are briefly descripted some similar methods in next chapter, like statistically optimized NAH, (SONAH) and iterative NAH with recursive filtration. The main product of the thesis is the practical part represented by testing application. That is created in the Matlab environment and is able to calculate and display hologram of the scanned array by the planar NAH method using the “k-space” filter. The application supposes a planar sound source and in other cases the accuracy of the reconstruction is not guaranteed. There are also given some holograms calculated with the application.

This thesis deals with processing signals from the microphone arrays for sound source localization. Compares different types of fields, such as cross-field and circular array. It is shown here how to implement Beamforming on FPGA and design of signal processing with a microphone array.