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Design and realization of acoustic camera
Koníček, Cyril ; Pristach, Marián (referee) ; Fujcik, Lukáš (advisor)
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.
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Beamforming system with FPGA
Vadinský, Václav ; Vágner, Martin (referee) ; Havránek, Zdeněk (advisor)
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.
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Design and testing of the system for localizing a sound source
Havelka, Martin ; Vechetová, Jana (referee) ; Najman, Jan (advisor)
This bachelor’s thesis deals with possibilities of sound source localization. There are generally mentioned possible hardware and software solutions at the beginning, from which two applicable methods and suitable components of test device are selected. Further there is a practical part of this work, where programmed algorithms are described. An essential part is testing. Accuracy is tested for the selected microphone array, depending on the change in the number and position of the sensors, the effect of ambient noise, and the location of the obstacle in the area. The output of the tests is the evaluation of functionality and accuracy for both methods used.
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Visualization of sound fields
Kovář, Petr ; Klusáček, Stanislav (referee) ; Havránek, Zdeněk (advisor)
This bachelor's thesis deals with problems of acoustical field visualization using near-field acoustical holography. This method is often refered as NAH. Within the thesis the basic acoustical field variables are processed and described. Two algorithms were analysed, namely the Statistically Optimal Algorithm (SONAH) and the Iterative Algorithm using the Fourier Transform (STFT) with a K-filter (Wave-Domain Filter). These algorithms are described in detail. The outcome of the practical part of this thesis is the application for sound pressure visualization. The application is created in the LabVIEW environment from the known codes and the calculation of the sound pressure is performed with one of the two algorithms described in this thesis. From depicted acoustical maps the disadvantages of this method and differences between algorithm are shown. These algorithms are then compared according to speed of the calculation.
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Acoustic Detection of Speaker Position Using Microphone Array
Horázný, František ; Mošner, Ladislav (referee) ; Szőke, Igor (advisor)
This thesis describes the problem of determining the approximate position of a sound source in a coordinate system needed using the microphone field. It covers all possible variables influencing the detection of the sound source and explains the basic methods which can be used to determine the origin of the sound. The solution proposed in this thesis is to use synchronized static recordings and further modifications for running the program in real-time on the provided ARM/SHARC system, which has limited performance. This thesis contains also tests of the individual components and their parameters. The effect of changing these parameters on the behavior of the system is also shown in this thesis. Additionally, the developed application is used to perform the experiments demonstrating the shift of results during computation without any limitations and when running on the sound system. It also shows experiments with the resulting application, how the results change when calculating without performance limitation and when running on a sound card. Finally, this thesis gives several recommendations and assumptions on how to improve the results when using the program and how to eliminate several system limitations in unfavorable conditions.
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Microphone Arrays for Speaker Recognition
Mošner, Ladislav ; Plchot, Oldřich (referee) ; Černocký, Jan (advisor)
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.
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Acoustic Detection of Speaker Position Using Microphone Array
Chu, Yen Nhi ; Žmolíková, Kateřina (referee) ; Szőke, Igor (advisor)
The purpose of this thesis is to create a system for detection of speaker position in a closed room equipped with microphone array. It deals with two algorithms for sound source localization, namely beamforming and time difference of arrival. The methods were implemented on the provided hardware, reliability of the methods was tested and results were evaluated. The results have shown that both used methods were relatively accurate if used in small space (approx. 3×3 m). Shortcomings of the solution were analyzed in detail, possible improvements were proposed. This work can be used as a basis for creating embedded systems for sound source localization.
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Far-Field Speech Recognition
Žmolíková, Kateřina ; Malenovský, Vladimír (referee) ; Černocký, Jan (advisor)
Systémy rozpoznávání řeči v dnešní době dosahují poměrně vysoké úspěšnosti. V případě řeči, která je snímána vzdáleným mikrofonem a je tak narušena množstvím šumu a dozvukem (reverberací), je ale přesnost rozpoznávání značně zhoršena. Tento problém je možné zmírnit využitím mikrofonních polí. Tato práce se zabývá technikami, které umožňují kombinovat signály z více mikrofonů tak, aby byla zlepšena kvalita výsledného signálu a tedy i přesnost rozpoznávání. Práce nejprve shrnuje teorii rozpoznávání řeči a uvádí nejpoužívanější algoritmy pro zpracování mikrofonních polí. Následně jsou demonstrovány a analyzovány výsledky použití dvou metod pro beamforming a metody dereverberace vícekanálových signálů. Na závěr je vyzkoušen alternativní způsob beamformingu za použití neuronových sítí.
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Acoustic detection of starlings through artificial intelligence
Zezula, Benjamin ; Vlachová Hutová, Eliška (referee) ; Marcoň, Petr (advisor)
To protect the vineyards from starlings is very costly and ineffective with the available resources over a period of time. In my project, I have designed a detection module that relays information about the flock location in the vineyard to the control module and thus produces scaring. I have designed two kinds of detection modules, which are compared. The first variant consists of a single directional microphone that rotates and sequentially senses the entire area of the vineyard. The second variant consists of four microphones, each one is directed to the cardinal point and based on the intensity of sound, the module will provide the information about the direction where the flock is located. Both detection modules process the signal in a Raspberry Pi 4 single board computer using an artificial neural network algorithm powered by MobileNetV2 architecture.
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Acoustic Detection of Speaker Position Using Microphone Array
Horázný, František ; Mošner, Ladislav (referee) ; Szőke, Igor (advisor)
This thesis describes the problem of determining the approximate position of a sound source in a coordinate system needed using the microphone field. It covers all possible variables influencing the detection of the sound source and explains the basic methods which can be used to determine the origin of the sound. The solution proposed in this thesis is to use synchronized static recordings and further modifications for running the program in real-time on the provided ARM/SHARC system, which has limited performance. This thesis contains also tests of the individual components and their parameters. The effect of changing these parameters on the behavior of the system is also shown in this thesis. Additionally, the developed application is used to perform the experiments demonstrating the shift of results during computation without any limitations and when running on the sound system. It also shows experiments with the resulting application, how the results change when calculating without performance limitation and when running on a sound card. Finally, this thesis gives several recommendations and assumptions on how to improve the results when using the program and how to eliminate several system limitations in unfavorable conditions.
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