Tato diplomová práce se zabývá problematikou implementace Hands-Free komunikačních systémů v automobilovém průmyslu. První kapitola je zaměřena na teoretické aspekty zpracování řeči v embedded aplikacích, jako je potlačení šumu, potlačení akustické zpětné vazby a další faktory ovlivňující kvalitu Hands-Free systémů. Druhá kapitola obsahuje návrh kompaktního flexibilního mobilního testovacího zařízení pro bezdrátové komunikační Hands-Free moduly.

Disertační práce se zabývá bezkontaktní analýzou vibrací pomocí metod akustické holografie v blízkém poli. Akustická holografie v blízkém poli je experimentální metoda, která rekonstruuje akustické pole v těsné blízkosti povrchu vibrujícího předmětu na základě měření akustického tlaku nebo akustické rychlosti v určité vzdálenosti od zkoumaného předmětu. Konkrétní realizace této metody závisí na použitém výpočetním algoritmu. Vlastní práce je zaměřena zejména na rozbor algoritmů, které využívají k rekonstrukci zvukového pole v blízkosti vibrujícího objektu transformaci do domény vlnových čísel (prostorová transformace), kde probíhá vlastní výpočet. V úvodu práce je vysvětlena základní teorie metody akustické holografie v blízkém poli s popisem základních vlastností a dále rozborem konkrétních nejčastěji používaných algoritmům pro lokalizaci a charakterizaci zdroje zvuku a pro následnou vibrační analýzu. Stěžejní část práce se věnuje pokročilým metodám zpracování, které se snaží určitým způsobem optimalizovat přesnost predice zvukového pole v blízkosti vibrujícího předmětu v reálných podmínkách. Jde zejména o problematiku použitého měřicího systému s akustickými snímači, které nejsou ideální, a dále o možnost měření v prostorách s difúzním charakterem zvukového pole. Pro tento případ byla na základě literárního průzkumu optimalizována a ověřena metoda využívající dvouvrstvé mikrofonní pole, které umožňuje oddělení zvukových polí přicházejících z různých stran a tedy úspěšné měření v uzavřených prostorách např. kabin automobilů a letadel. Součástí práce byla také optimalizace, rozšíření a následné ověření algoritmů publikovaných v posledních letech pro měření v reálných podmínkách za použití běžně dostupných akustických snímačů.

Intake and Exhaust system noise makes a huge contribution to the interior and exterior noise of automobiles. There are a number of linear acoustic tools developed by institutions and industries to predict the acoustic properties of intake and exhaust systems. The present project discusses and validates, through measurements, the proper modelling of these systems using BOOST-SID and discusses the ideas to properly convert a geometrical model of an exhaust muffler to an acoustic model. The various elements and their properties are also discussed. When it comes to Acoustic properties there are several parameters that describe the performance of a muffler, the Transmission Loss (TL) can be useful to check the validity of a mathematical model but when we want to predict the actual acoustic behavior of a component after it is installed in a system and subjected to operating conditions then we have to determine other properties like Attenuation, Insertion loss etc,. Zero flow and Mean flow (M=0.12) measurements of these properties were carried out for mufflers ranging from simple expansion chambers to complex geometry using two approaches 1) Two Load technique 2) Two Source location technique. For both these cases, the measured transmission losses were compared to those obtained from BOOST-SID models. The measured acoustic properties compared well with the simulated model for almost all the cases.

No description available.

Röststyrning som användargränssnitt kan erbjuda flera fördelar jämfört med mer traditionella styrmetoder. Det saknas dock färdiga lösningar för specifika industriella miljöer, vilka ställer särskilda krav på att korta kommandon tolkas korrekt i olika grad av buller och med begränsad eller ingen internetuppkoppling. Detta arbete ämnade undersöka potentialen för röststyrning i industriella miljöer. Ett koncepttest genomfördes där ordfelsfrekvens (på engelska Word Error Rate eller kortare WER) användes för att utvärdera träffsäkerheten för olika kombinationer av taligenkänningsarkitekturer, brusreduceringstekniker samt kommandolängder i verkliga bullriga miljöer. Undersökningen tog dessutom hänsyn till Lombard-effekten.  Resultaten visar att det för samtliga testade miljöer finns god potential för röststyrning med avseende på träffsäkerheten. Framför allt visade DeepSpeech, en djupinlärd taligenkänningsmodell med rekurrent lagerstruktur, kompletterad med domänspecifika språkmodeller och en riktad kardioid-mikrofon en ordfelsfrekvens på noll procent i vissa scenarier och sällan över fem procent. Resultaten visar även att utformningen av kommandon påverkar ordfelsfrekvensen.  För en verklig implementation i industriell miljö behövs ytterligare studier om säkerhetslösningar, inkluderande autentisering och hantering av risker med falskt positivt tolkade kommandon. / Voice command as a user interface can offer several advantages over more traditional control methods. However, there is a lack of ready-made solutions for specific industrial environments, which place particular demands on short commands being interpreted correctly in varying degrees of noise and with limited or no internet connection. This work aimed to investigate the potential for voice command in industrial environments. A proof of concept was conducted where Word Error Rate (WER) was used to evaluate the accuracy of various combinations of speech recognition architectures, noise reduction techniques, and command lengths in authentic noisy environments. The investigation also took into account the Lombard effect.  The results indicate that for all tested environments there is good potential for voice command with regard to accuracy. In particular, DeepSpeech, a deep-learned speech recognition model with recurrent layer structure, complemented with domain-specific language models and a directional cardioid microphone, showed WER values of zero percent in certain scenarios and rarely above five percent. The results also demonstrate that the design of commands influences WER. For a real implementation in an industrial environment, further studies are needed on security solutions, including authentication and management of risks with false positive interpreted commands.

The developments of new types of conductors and increase of voltage level have driven the need to carry out research on evaluating overhead line acoustic noise. The surface potential gradient of a conductor is a critical design parameter for planning overhead lines, as it determines the level of corona loss (CL), radio interference (RI), and audible noise (AN). The majority of existing models for surface gradient calculation are based on analytical methods which restrict their application in simulating complex surface geometries. This thesis proposes a novel method which utilizes both analytical and numerical procedures to predict the surface gradient. Stranding shape, proximity of tower, protrusions and bundle arrangements are considered within this model. One of UK National Grid's transmission line configurations has been selected as an example to compare the results for different methods. The different stranding shapes are a key variable in determining dry surface fields. The dynamic behaviour of water droplets subject to AC electric fields is investigated by experiment and finite element modelling. The motion of a water droplet is considered on the surface of a metallic sphere. To understand the consequences of vibration, the FEA model is introduced to study the dynamics of a single droplet in terms of phase shift between vibration and exciting voltage. Moreover, the evolution of electric field within the whole cycle of vibration is investigated. The profile of the electric field and the characteristics of mechanical vibration are evaluated. Surprisingly the phase shift between these characteristics results in the maximum field occurring when the droplet is in a flattened profile rather than when it is ‘pointed’.Research work on audible noise emitted from overhead line conductors is reviewed, and a unique experimental set up employing a semi-anechoic chamber and corona cage is described. Acoustically, this facility isolates undesirable background noise and provides a free-field test space inside the anechoic chamber. Electrically, the corona cage simulates a 3 m section of 400 kV overhead line conductors by achieving the equivalent surface gradient. UV imaging, acoustic measurements and a partial discharge detection system are employed as instrumentation. The acoustic and electrical performance is demonstrated through a series of experiments. Results are discussed, and the mechanisms for acoustic noise are considered. A strategy for evaluating the noise emission level for overhead line conductors is developed. Comments are made on predicting acoustic noise from overhead lines. The technical achievements of this thesis are summarized in three aspects. First of all, an FEA model is developed to calculate the surface electric field for overhead line conductors and this has been demonstrated as an efficient tool for power utilities in computing surface electric field especially for dry condition. The second achievement is the droplet vibration study which describes the droplets' behaviour under rain conditions, such as the phase shift between the voltage and the vibration magnitude, the ejection phenomena and the electric field enhancement due to the shape change of droplets. The third contribution is the development of a standardized procedure in assessing noise emission level and the characteristics of noise emissions for various types of existing conductors in National Grid.

This thesis describes the design and realization of a mobile robot with autonomous audio-visual control. This robot is able of movement based on sensors consisting of camera and microphone. The mechanical part consists of components made with 3D print technology and omnidirectional Mecanum wheels. Software utilizes OpenCV library for image processing and computes MFCC a DTW for voice command detection.

The aim of this work is the design and implementation of a sound level meter for emdedded systems. It is designed sound level meter sensor. This sensor includes a microphone, microphone preamplifier and ADC. This sensor is connected to a development kit STM32F4 Discovery, where sound is processed. It is processed filtration, time weighting, calculation of sound leve and loudness weighting. The sound level is then sent via the serial communication USART. This thesis includes a theoretical analysis of noise and its human perceptions. It also describes the development platform used.

This diploma thesis deals with the design and implementation of an electronic module for acoustic detection. The module has the task of detecting a predetermined acoustic signals through them learned classification model. The module is used mainly for security purposes. To identify and classify the proposed model using machine learning techniques. Given the possibility of retraining for a different set of sounds, the module becomes a universal sound detector. With acoustic sound using the digital MEMS microphone, for which it is designed and implemented conversion filter. The resulting system is implemented into firmware microcontroller with real time operating system. The various functions of the system are realized with regard to the possible optimization (less powerful MCU or battery power). The module transmits the detection results of the master station via Ethernet network. In the case of multiple modules connected to the network to create a distributed system, which is designed for precise time synchronization using PTP protocol defined by the IEEE-1588 standard.

Matched frequency responses are a fundamental starting point for a variety ofimplementations for microphone arrays. In this report, two methods for frequencyresponse-calibration of a pre-assembled microphone array are presented andevaluated. This is done by extracting the deviation in frequency responses of themicrophones in relation to a selected reference microphone, using a swept sine asa stimulus signal and an inverse filter. The swept sine includes all frequencieswithin the bandwidth of human speech. This allows for a full frequency responsemeasurements from all microphones using a single recording.Using the swept sine, the deviation in frequency response between the microphonescan be obtained. This deviation represents the scaling factor that all microphonesmust be calibrated with to match the reference microphone. Applying the scalingfactors on the recorded stimulus signal shows an improvement for both implementedmethods, and where one method matches the frequency response of the microphoneswith high accuracy.Once the scaling factors of the various microphones is obtained, it can be usedto calibrate other recorded signals. This leads to an minor improvement formatching the frequency responses, as it has been shown that the differencesin frequency response between the microphones is signal-dependent and variesbetween recordings. The response differences between the microphones dependson the design of the array, speaker, room and the acoustic frequency dispersionthat occurs with sound waves. This makes it difficult to calibrate the frequencyresponses of the microphones without appropriate equipment because the responseof the microphones is noticeably affected by these other factors. Proposals to addressthese problems are discussed in the report as future work. / Matchade frekvenssvar är en grundläggande utgångspunkt för ett flertal implementationer för mikrofonarrayer. I denna rapport presenteras och utvärderas tvåmetoder för frekvenssvarskalibrering för en förmonterad mikrofonarray. Detta görsgenom att extrahera avvikelsen i frekvenssvar hos alla mikrofoner i förhållandetill en vald referensmikrofon. Frekvenssvaren tas fram med hjälp av ettsinussvep som stimulanssignal och ett inverterat filter. Sinussvepet inkluderar helafrekvensbredden för mänskligt tal och möjliggör att mikrofonernas fulla frekvenssvarkan analyseras från en enda inspelning.Med hjälp av sinussvepet kan avvikelsen i frekvenssvar mellan mikrofonerna erhållas.Denna avvikelse representerar den skalningsfaktor alla mikrofoner måste kalibrerasefter för att matcha referensmikrofonen. Genom att applicera faktorerna på deninspelade stimulussignalen ses en förbättring för båda implementerade metoderna,där en metod matchar mikrofonernas frekvenssvar med hög noggrannhet.När skalningsfaktorn för de olika mikrofonerna har erhållits kan den användas föratt kalibrera andra inspelade signaler. Detta leder till en liten förbättring i att matchafrekvenssvaren, då det har visat sig att skillnader mellan mikrofonernas frekvenssvarär signalberoende och varierar mellan inspelningar. Skillnader i frekvenssvar mellanmikrofonerna beror på ljudets utbredning i rummet, utformningen av arrayen,högtalaren och den akustiska frekvensspridningen som uppstår hos ljudvågor. Dettagör det svårt att kalibrera frekvenssvaren hos mikrofonerna utan lämplig utrustningeftersom mikrofonernas respons märkbart påverkas av dessa andra faktorer. Förslagför att kringgå dessa problem diskuteras i rapporten och tas upp som framtidaarbete.