People are usually directed towards each other in conversations, to make it easier to hear what is being said. Algorithms for voice and speech recognition works in a similar way, regarding the microphone direction towards the sound source. In this thesis in electronics has therefore a servo control with binaural sound localization been implemented on a microcontroller connected to two microphones. When people perceive sound, the brain can estimate the sound source direction by comparing the time taken by the sound reaching one ear to the other [1]. The difference in time is called the interaural time difference, and can be calculated using various techniques. An exploratory comparison between the techniques cross-correlation and cross-spectrum analysis was carried out before implementation. Advantages and disadvantages of each technique were evaluated at the same time. The result is a functioning servo control, that uses a cross correlation algorithm to calculate the interaural time difference, and controls a servo motor towards the sound source with a P-regulated error reduction method. The project was implemented on the microcontroller ATmega328P from Atmel without using floating point calculations. The thesis was carried out on behalf of the company Jetspark Robotics.

Constant False Alarm Rate (CFAR) Processors are important for applications where thousands of detection tests are made per second, such as in radar. This thesis introduces a new method for CFAR threshold estimation that is particularly applicable to sound source detection with distributed microphone systems. The novel CFAR Processor exploits the near symmetry about 0 for the acoustic pixel values created by steered-response coherent power in conjunction with a partial whitening preprocessor to estimate thresholds for positive values, which represent potential targets. To remove the low frequency components responsible for degrading CFAR performance, fixed and adaptive high-pass filters are applied. A relation is proposed and it tested the minimum high-pass cut-off frequency and the microphone geometry. Experimental results for linear, perimeter and planar arrays illustrate that for desired false alarm (FA) probabilities ranging from 10-1 and 10-6, a good CFAR performance can be achieved by modeling the coherent power with Chi-square and Weibull distributions and the ratio of desired over experimental FA probabilities can be limited within an order of magnitude.

The detection of sound sources with microphone arrays can be enhanced through processing individual microphone signals prior to the delay and sum operation. One method in particular, the Phase Transform (PHAT) has demonstrated improvement in sound source location images, especially in reverberant and noisy environments. Recent work proposed a modification to the PHAT transform that allows varying degrees of spectral whitening through a single parameter, andamp;acirc;, which has shown positive improvement in target detection in simulation results. This work focuses on experimental evaluation of the modified SRP-PHAT algorithm. Performance results are computed from actual experimental setup of an 8-element perimeter array with a receiver operating characteristic (ROC) analysis for detecting sound sources. The results verified simulation results of PHAT- andamp;acirc; in improving target detection probabilities. The ROC analysis demonstrated the relationships between various target types (narrowband and broadband), room reverberation levels (high and low) and noise levels (different SNR) with respect to optimal andamp;acirc;. Results from experiment strongly agree with those of simulations on the effect of PHAT in significantly improving detection performance for narrowband and broadband signals especially at low SNR and in the presence of high levels of reverberation.

abstract: This study consisted of several related projects on dynamic spatial hearing by both human and robot listeners. The first experiment investigated the maximum number of sound sources that human listeners could localize at the same time. Speech stimuli were presented simultaneously from different loudspeakers at multiple time intervals. The maximum of perceived sound sources was close to four. The second experiment asked whether the amplitude modulation of multiple static sound sources could lead to the perception of auditory motion. On the horizontal and vertical planes, four independent noise sound sources with 60° spacing were amplitude modulated with consecutively larger phase delay. At lower modulation rates, motion could be perceived by human listeners in both cases. The third experiment asked whether several sources at static positions could serve as "acoustic landmarks" to improve the localization of other sources. Four continuous speech sound sources were placed on the horizontal plane with 90° spacing and served as the landmarks. The task was to localize a noise that was played for only three seconds when the listener was passively rotated in a chair in the middle of the loudspeaker array. The human listeners were better able to localize the sound sources with landmarks than without. The other experiments were with the aid of an acoustic manikin in an attempt to fuse binaural recording and motion data to localize sounds sources. A dummy head with recording devices was mounted on top of a rotating chair and motion data was collected. The fourth experiment showed that an Extended Kalman Filter could be used to localize sound sources in a recursive manner. The fifth experiment demonstrated the use of a fitting method for separating multiple sounds sources. / Dissertation/Thesis / Doctoral Dissertation Speech and Hearing Science 2015

Im Genehmigungsverfahren für den Bau hochliegender Schallquellen (z.B. Windenergieanlagen) muss der Nachweis geführt werden, dass von den Anlagen keine schädlichen Umwelteinwirkungen ausgehen. Es ist es daher notwendig, die Schallausbreitung derartiger Quellen grundsätzlich zu untersuchen. Eine Schwierigkeit stellt dabei die gekoppelte Wirkung von Temperatur-, Windgeschwindigkeits- und Windrichtungsprofil in Zusammenhang mit dem Bodeneinfluss auf die Schallausbreitung dar. Dieser zeitlich und räumlich variable Atmosphäreneinfluss wird insbesondere bei Langzeituntersuchungen der Schallimmission bisher nur unzureichend in den operationellen Modellen beschrieben. Das Ziel der Studie besteht deshalb darin, die gekoppelte Wirkung von Atmosphäre- und Boden-Einfluss auf die Schallausbreitung in einem Bereich bis zu 2 km Entfernung von der Schallquelle mit dem Modell SMART (Sound propagation model of the atmosphere using ray-tracing ) zu untersuchen. / The licensing procedure for the construction of high-placed sound sources (e.g. wind power stations) demands to proof that no (significant and) harmful impact on environment is outgoing from these systems. Therefore, it is necessary to analyse the sound propagation of such a kind of sources. In this context one central problem has to be managed: the coupled effect of temperature, wind speed and wind direction profiles combined with the influence of surface on sound propagation. The temporally and spatially variable influence of the atmosphere is only insufficiently described by the operational models, especially in relation to long-time investigations of sound immission. Consequently, the aim of this study was to investigate the coupled effect of atmospheric and surface influence on sound propagation up to distances of 2 km away from the sound source using the model SMART (Sound propagation model of the atmosphere using ray-tracing).

In dieser Studie wird ein Verfahren näher erläutert, welches im Rahmen einer Projektbearbeitung für die Bundeswehr entwickelt worden ist. Dieses Verfahren prognostiziert die Ausbreitung von Schießlärm unter besonderer Berücksichtigung der Einflüsse meteorologischer Verhältnisse (Schallwetter) und ermöglicht so die Prognose erhöhter Lärmbelastungen. Weiterhin ist mit diesem Verfahren auch eine regionale Einteilung eines Gesamtgebietes in schallklimatologisch ähnliche Teilgebiete möglich. Eine solche Untersuchung erfolgt durch die Analyse von Schalldruckpegeldämpfungskarten für eine Vielzahl von Atmosphärenstrukturen. Im Wesentlichen stützt sich das Verfahren dabei auf zwei Komponenten, auf das Schallausbreitungsmodell SMART (Sound propagation model of the atmosphere using ray-tracing) und das Anwendungstool MetaVIS (Meteorological attenuation visualization). Mit Hilfe des Modells SMART werden Schalldruckpegeldämpfungen berechnet, welche dann prognostische Aussagen zur Lärmbelastung an einem Ort zulassen. Die Darstellungssoftware MetaVIS bietet schließlich die Möglichkeit, eine aktuelle Schallausbreitungssituation analysieren und bewerten zu können. Die Bewertung kann aufgrund einer umfangreichen Datenbibliothek erfolgen. Nach Beendigung des Projektes kann dieses Verfahren vom Auftraggeber zu einer bewerteten ortsabhängigen Schallausbreitungsprognose unter Einbeziehung meteorologischer Parameter genutzt werden. / In this study a method will be specified, which is developed in line of a project work for the Bundeswehr. This method predicts the propagation of shooting noise in particular consideration of the effects of meteorological conditions (weather of sound) and so allows the prediction of increased noise levels. Furthermore, with this method a regional classification of an area like Germany in sound climatologically similarly areas is possible by analysing maps of sound level attenuation for a multiplicity of structure of the atmosphere. Basically the method bases on two parts, a model of sound propagation SMART (Sound propagation model of the atmosphere using ray-tracing) and an application software Meta-VIS (Meteorological attenuation visualization). By means of SMART the attenuation of the sound level will be calculated which approves statements about noise exposure at a certain place. The visualisation software MetaVIS offers finally the chance to analyse and to evaluate a present situation of sound propagation. The estimation can be carried out as a result of a large library of data. Upon completion of the project, the method will be used by the customer to predict the sound propagation dependent on location, namely with inclusion of meteorological parameters.

For a robot to succeed at speech recognition, it is advantageous to have a strong and clear signal tointerpret. To facilitate this the robot can steer and aim for the sound source to get a clearer signal, todo this a sound source localization system is required. If the robot turns towards the speaker thisalso gives a more natural feeling when a human interacts with the robot. To determine where thesound source is positioned, an angle relative to the microphone pair is calculated using theinteraural time difference (ITD), which is the difference in time of arrival of the sound between thepair of microphones. To achieve good result the microphone signals needs to be preprocessed andthere are also different algorithms for calculating the time difference which are investigated in thisthesis. The results presented in this work are from tests, with an emphasis on focusing at real-timesystems, involving noisy environment and response time. The results show the complexity of thebalance between computational time and precision. / För att en robot ska lyckas med taleigenkänning, är det fördelaktigt att ha en stark och tydlig signalatt tolka. För att underlätta detta kan roboten styra och rikta in sig mot ljudkällan för att få entydligare signal och för att detta skall vara möjligt krävs ett system för lokalisering av ljudkällan.Om roboten vänder sig mot talaren ger detta även en mer naturlig känsla när en människainteragerar med roboten. För att avgöra var ljudkällan är placerad, beräknas en vinkel i förhållandetill mikrofonparet med hjälp av interaurala tidsskillnaden (ITD), vilket är skillnaden i ankomsttid avljudet mellan mikrofonparet. För att uppnå bra resultat måste mikrofonsignalerna förbehandlas ochdet finns också olika algoritmer för att beräkna tidsskillnaden som undersöks i detta examensarbete.Det resultat som presenteras i detta arbete kommer från tester, med tonvikt på att fokusera pårealtidssystem, som inbegriper bullrig miljö och svarstid. Resultaten visar komplexiteten i balansenmellan beräknings tid och precision.

Sound source localization (SSL) is a broad field, with many important application areas. In outdoor environments SSL systems can, among other things, be used to increase citizens’ safety by detecting and locating abnormal sounds such as gunshots or screams. Localization is a complex field, in the case of an outdoor setting, the sound signal is affected by weather conditions, noise, and objects blocking the propagation path. Furthermore, challenges concerning implementing cost-effective algorithms, robustness, accuracy, and balancing trade-offs, still remain. SSL is a field of intense research, and new studies are continuously published. However, to the best of the authors knowledge, there are no recent reviews of state of the art SSL solutions, applicable in an outdoor urban setting. Hence, this study provides a knowledge base concerning current SSL approaches, intended for the aforesaid environment, and to this end a systematic literature review was performed. The review consisted of a total of 43 studies, published between 2017-2021. From the extracted data, a taxonomy of currently seen design principles was developed. Additionally, both the applied measurement techniques and the positioning methods were defined. It can be seen from the result that classical methods such as direction of arrival and time difference of arrival still are the most used principles in research. However, learning-based approaches have seemingly started to attract more attention. Furthermore, a general description of the SSL approaches has been presented. Thus, the knowledge base provided by this study contains both information on what current state of the art techniques are most commonly adopted as well as the basic ideas behind these principles.

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京都大学 / 0048 / 新制・課程博士 / 博士(情報学) / 甲第17928号 / 情博第510号 / 新制||情||90(附属図書館) / 30748 / 京都大学大学院情報学研究科知能情報学専攻 / (主査)教授 奥乃 博, 教授 河原 達也, 教授 山本 章博 / 学位規則第4条第1項該当 / Doctor of Informatics / Kyoto University / DFAM