Acoustic Sound Source localization using signal processing is required in order to estimate the direction from where a particular acoustic source signal is coming and it is also important in order to find a soluation for hands free communication. Video conferencing, hand free communications are different applications requiring acoustic sound source localization. This applications need a robust algorithm which can reliably localize and position the acoustic sound sources. The Steered Response Power Phase Transform (SRP-PHAT) is an important and roubst algorithm to localilze acoustic sound sources. However, the algorithm has a high computational complexity thus making the algorithm unsuitable for real time applications. This thesis focuses on describe the implementation of the SRP-PHAT algorithm as a function of source type, reverberation levels and ambient noise. The main objective of this thesis is to present different approaches of the SRP-PHAT to verify the algorithm in terms of acoustic enviroment, microphone array configuration, acoustic source position and levels of reverberation and noise.

L'apparition récente de microphones numériques MEMS a ouvert de nouvelles perspectives pour le développement de systèmes d'acquisition acoustiques massivement multi-canaux de grande envergure. De tels systèmes permettent de localiser des sources acoustiques avec de bonnes performances. En revanche, de nouvelles contraintes se posent. La première est le flux élevé de données issues de l'antenne, devant être traitées en un temps raisonnable. La deuxième contrainte est de connaître la position des nombreux microphones déployés in situ. Ce manuscrit propose des méthodes répondant à ces deux contraintes. Premièrement, une étude du système d'acquisition est présentée. On montre que les microphones MEMS sont adaptés pour des applications d'antennerie. Ensuite, un traitement en temps réel des signaux acquis via une implémentation parallèle sur GPU est proposé. Cette stratégie répond au problème de flux de données. On dispose ainsi d'un outil d'imagerie temps réel de sources large bande, permettant d'établir un diagnostic dynamique de la scène sonore.Deuxièmement, différentes méthodes de calibration géométrique pour la détermination de la position des microphones sont exposées. Dans des conditions réelles d'utilisation, les méthodes actuelles sont inefficaces pour des antennes étendues et à grand nombre de microphones. Ce manuscrit propose des techniques privilégiant la robustesse du processus de calibration. Les méthodes proposées couvrent différents environnements acoustiques réels, du champ libre au champ réverbérant. Leur efficacité est prouvée par différentes campagnes expérimentales. / Recently, digital MEMS microphones came out and have opened new perspectives. One of them is the design of large-aperture and massively multichannel acoustical acquisition systems. Such systems meet good requirements for efficient source localization. However, new problems arise. First, an important data flow comes from the array, and must be processed fast enough. Second, if the large array is set up in situ, retrieving the position of numerous microphones becomes a challenging task. This thesis proposes methods addressing these two problems. The first part exhibits the description of the acquisition system, which has been developed during the thesis. First, we show that MEMS microphone characteristics are suitable for array processing applications. Then, real-time processing of channel signals is achieved by a parallel GPU implementation. This strategy is one solution to the heavy data flow processing issue. In this way, a real-time acoustic imaging tool was developed, and enables a dynamic wide-band diagnosis, for an arbitrary duration.The second part presents several robust geometric calibration methods: they retrieve microphone positions, based only on the array acoustic signals. Indeed, in real-life conditions, the state of the art methods are inefficient with large arrays. This thesis proposes techniques that guarantee the robustness of the calibration process. The proposed methods allow calibration in the different existing soundscapes, from free field to reverberant field. Various experimental scenarios prove the efficiency of the methods.

This diploma thesis deals with a basic description of dynamic and condenser microphone, its performance regarding polar patterns and a type of mechanical construction. It also describes a principle of operation of a triode. The thesis deals with selecting of a suitable type of microprocessor to control microphone preamplifier functions. Last part contains a power supply and microphone preamplifier circuits design using a semiconductor and vacuum tube technology.

Tato diplomová práce se zabývá problematikou vzdáleného rozpoznávání mluvčích. V případě dat zachycených odlehlým mikrofonem se přesnost standardního rozpoznávání značně snižuje, proto jsem navrhl dva přístupy pro zlepšení výsledků. Prvním z nich je použití mikrofonního pole (záměrně rozestavené sady mikrofonů), které je schopné nasměrovat virtuální "paprsek" na pozici řečníka. Dále jsem prováděl adaptaci komponent systému (PLDA skórování a extraktoru i-vektorů). S využitím simulace pokojových podmínek jsem syntetizoval trénovací a testovací data ze standardní datové sady NIST 2010. Ukázal jsem, že obě techniky a jejich kombinace vedou k výraznému zlepšení výsledků. Dále jsem se zabýval společným určením identity a pozice mluvčího. Zatímco výsledky ve venkovním simulovaném prostředí (bez ozvěn) jsou slibné, výsledky z interiéru (s ozvěnami) jsou smíšené a vyžadují další prozkoumání. Na závěr jsem mohl systémem vyhodnotit omezené množství reálných dat získaných přehráním a záznamem nahrávek ve skutečné místnosti. Zatímco výsledky pro mužské nahrávky odpovídají simulaci, výsledky pro ženské nahrávky nejsou přesvědčivé a vyžadují další analýzu.

Thanks to the development in recent years, the placement of miniaturized sensors such as accelerometers, gyroscopes, magnetometers, global positioning system receivers (GPS), microphones or others to commercially sold smartphones is increasing. Use of these sensors (which are to be found in the smartphone) for human activity recognition with health care improvement in mind is a discussed theme. Advantages of the use of smartphone for human movement monitoring lies in the fact that it is a device that the person measured carries with them and there are no additional costs. The disadvantages are a limited storage and battery. Therefore, only accelerometer, gyroscope, magnetometer, and microphone were chosen because their combination achieves best results. GPS sensor was excluded for its lack of reliability in sampling and for being energy demanding. Features were computed from the measured data and used for learning of the classification model. The highest accuracy was achieved with the use of a machine learning method called Random Forest. The main goal of this work was to create an algorithm for transportation mode recognition using signals sensed by a smartphone. The created algorithm succeeds in classification of walk, car, bus, tram, train, and bike in 97.4 % with 20 % holdout validation. When tested on a new set of data from the tenth volunteer, the resulting accuracy counted as average form classification recall for each transportation mode reached 90.49 %.

Master’s thesis deals with description of issues of reading spatial audio signals using microphone array. Basic method of beamforming (Delay and Sum) is characterized on basis of chosen conception of the microphone array. Specific issues of audio detection and digital signal processing of converted audio signals are characterized and some ways how to solve the issues are adumbrated. Features and limitations of chosen ARM processor in the digital processing of multiple audio signals are described. Especially features and limitations of an internal A/D converter from the perspective of the beamforming are described.

This thesis deals with design and realization of acoustic camera for sound source location. The essence of the device is microphone field listening for acoustic impulses which are processed in real-time in order to locate the source. Found location is displayed on LCD together with images from regular video camera. FPGA is used as a computational unit.

This thesis deals with issue of traffic noise, its measurement and data evaluation. Theoretic part of my work concretes basic information about acoustics and also describes and analyzes traffic noise sources of tire/pavement. Next part deals with quite pavement and with measurement methods of traffic noise. Practical part of my thesis contains description, results and evaluation of realized measurement with methods CPX and SPB.

Recent advances in near-field acoustical holography (NAH) have expanded the theory to interior spaces where multiple sources and/or reflections are present. In 1990, Tamura presented the spatial Fourier transform separation method to measure the reflection coefficient at oblique angles using two measurement planes in the wave number domain. This paper adapts the spatial Fourier transform separation method for application in interior NAH. A practical exploration of important experimental parameters is performed, which include the relative amplitudes of primary and disturbing sources, the measurement plane separation distance, and an acceptable noise floor. This technique is successfully applied in a reverberant environment to reconstruct the velocity of a clamped vibrating plate. NAH methods based on the measurement of pressure and particle velocity have led to the ability to reduce the required measurement locations. Other recent advances in NAH have expanded the theory to interior spaces where multiple sources and/or reflections are present. This paper investigates the use of interpolation techniques to reduce the required measurement locations for interior NAH. Specifically, the benefits of a bi-cubic Hermite surface patch interpolation are discussed and compared to other interpolation routines. Although the required inputs for the Hermite interpolation can be measured using a variety of devices, a scanning six-microphone probe in a tetrahedral configuration is suggested. The six microphones are utilized to simultaneously sample pressure on two parallel planes and estimate the pressure gradients on both of these planes. The two interpolated measurement holograms are used to separate the incoming and outgoing waves using the spatial Fourier-transform method. Analytical simulations of simply supported plates are shown as well as experimental results in a reverberation room to characterize the reduction in measurement locations. Depending on the spatial frequency of the hologram, a measurement location reduction of 20–80% was observed.

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