京都大学 / 0048 / 新制・課程博士 / 博士(情報学) / 甲第18412号 / 情博第527号 / 新制||情||93(附属図書館) / 31270 / 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, 准教授 CUTURI CAMETO Marco, 講師 吉井 和佳 / 学位規則第4条第1項該当 / Doctor of Informatics / Kyoto University / DFAM

This thesis is aimed at simulation of auditory space. It describes the most frequently used panning method: Vector Base Amplitude Panning. Also is focused on image source method, which allows computing the parameters of direct sound wave and reflections in rectangular room. This method is compared with ray–tracing method, which is also often used. It deals with the matter of the frequency – dependent absorption of materials in reflection of the sound wave against the wall. On the basis of these information two applications were designed in MATLAB development environment. The first one allows the simulation of auditory space. The other one is the application for sound source panning by its azimuth and distance.

In teleconferencing the audio quality is key to conducting successful meetings. The conference room setting imposes various challenges on the speech signal processing, such as noise and interfering signals, reverberation, or participants positioned far from the telephone unit. This work aims at improving the received speech signal of a conference telephone by implementing sound source localization and beamforming. The implemented microphone array signal processing techniques are compared to the performance of an existing multi-microphone solution and evaluated under various conditions using a planar uniform circular array. Recordings of test-sequences for the evaluation were performed using a custom-built array mockup. The implemented algorithms did not show good enough performance to motivate the increased computational complexity compared to the existing solution. Moreover, an increase in number of microphones used was concluded to have little or no effect on the performance of the methods. The type of microphone used was, however, concluded to have impact on the performance and a subjective listening evaluation indicated a preference for omnidirectional microphones which is recommended to investigate further. / God ljudkvalitet är en grundsten för lyckade telefonmöten. Miljön i ett konferens-rum medför ett flertal olika utmaningar för behandlingen av mikrofonsignalerna: det kan t.ex. vara brus och störningar, eller att den som talar befinner sig långt från telefonen. Målet med detta arbete är att förbättra den talsignal som tas upp av en konferenstelefon genom att implementera lösningar för lokalisering av talaren och riktad ljudupptagning med hjälp av ett flertal mikrofoner. De implementerade metoderna jämförs med en befintlig lösning och utvärderas under olika brusscenarion för en likformig cirkulär mikrofonkonstellation. För utvärderingen användes testsignaler som spelades in med en specialbyggd enhet. De implementerade algoritmerna kunde inte uppvisa en tillräcklig förbättring i jämförelse med den befintliga lösningen för att motivera den ökade beräkningskomplexitet de skulle medföra. Dessutom konstaterades att en fördubbling av antalet mikrofoner gav liten eller ingen förbättring på metoderna. Vilken typ av mikrofon som användes konstaterades däremot påverka resultatet och en subjektiv utvärdering indikerade en preferens för de rundupptagande mikrofonerna, en skillnad som föreslås undersökas vidare.

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.

Time Delay of Arrival (TDOA) based algorithms and Steered Response Power (SRP) based algorithms are two most commonly used methods for sound source detection and localization. SRP is more robust under high reverberation and multi-target conditions, while TDOA is less computationally intensive. This thesis introduces a modified TDOA algorithm, TDOA delay table search (TDOA-DTS), that has more stable performance than the original TDOA, and requires only 4% of the SRP computation load for a 3-dimensional space of a typical room. A 2-step adaptive thresholding procedure based on a Weibull noise peak distributions for the cross-correlations and a binomial distribution for combing potential peaks over all microphone pairs for the final detection. The first threshold limits the potential target peaks in the microphone pair cross-correlations with a user-defined false-alarm (FA) rates. The initial false-positive peak rate can be set to a higher level than desired for the final FA target rate so that high accuracy is not required of the probability distribution model (where model errors do not impact FA rates as they work for threshold set deep into the tail of the curve). The final FA rate can be lowered to the actual desired value using an M out of N (MON) rule on significant correlation peaks from different microphone pairs associated is a point in the space of interest. The algorithm is tested with simulated and real recorded data to verify resulting FA rates are consistent with the user-defined rates down to 10-6.

At a cocktail party, one can choose to scan the room for conversations of interest, attend to a specific conversation partner, switch between conversation partners, or not attend to anything at all. The ability of the normal-functioning auditory system to flexibly listen in complex acoustic scenes plays a central role in solving the cocktail party problem (CPP). In contrast, certain demographics (e.g., individuals with hearing impairment or older adults) are unable to solve the CPP, leading to psychological ailments and reduced quality of life. Since the normal auditory system still outperforms machines in solving the CPP, an effective solution may be found by mimicking the normal-functioning auditory system. Spatial hearing likely plays an important role in CPP-processing in the auditory system. This thesis details the development of a biologically based approach to the CPP by modeling specific neural mechanisms underlying spatial tuning in the auditory cortex. First, we modeled bottom-up, stimulus-driven mechanisms using a multi-layer network model of the auditory system. To convert spike trains from the model output into audible waveforms, we designed a novel reconstruction method based on the estimation of time-frequency masks. We showed that our reconstruction method produced sounds with significantly higher intelligibility and quality than previous reconstruction methods. We also evaluated the algorithm's performance using a psychoacoustic study, and found that it provided the same amount of benefit to normal-hearing listeners as a current state-of-the-art acoustic beamforming algorithm. Finally, we modeled top-down, attention driven mechanisms that allowed the network to flexibly operate in different regimes, e.g., monitor the acoustic scene, attend to a specific target, and switch between attended targets. The model explains previous experimental observations, and proposes candidate neural mechanisms underlying flexible listening in cocktail-party scenarios. The strategies proposed here would benefit hearing-assistive devices for CPP processing (e.g., hearing aids), where users would benefit from switching between various modes of listening in different social situations. / 2022-05-19T00:00:00Z

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.

Penser les traitements du son spatialisés en ambisonie permet de mettre en valeur le potentiel musical de la décomposition du champ sonore en harmoniques sphériques, et amène à redéfinir la représentation de l’espace sonore. Cette thèse défend que les représentations abstraites et intermédiaires de l’espace sonore permettent d’élaborer de nouvelles approches originales de la mise en espace du son. Le raisonnement amenant à cette affirmation commence par l’appropriation musicale de l’approche ambisonique. La création de nouveaux traitements de l’espace et du son amène à utiliser de manière originale les signaux associés aux harmoniques sphériques, et à concevoir différemment les relations qui les régissent, ainsi que leur hiérarchisation. La particularité de ces approches expérimentales et les caractéristiques singulières des champs sonores générés, nécessitent de concevoir de nouveaux outils théoriques et pratiques pour leur analyse et leur restitution. Les changements opérés permettent alors de libérer cette approche des enjeux techniques et matériels initiaux en ambisonie. Mais ils permettent surtout de s’émanciper des modèles psychoacoustiques et acoustiques sur lesquels ces techniques reposent originellement. Dans ce contexte, les signaux associés aux harmoniques sphériques ne sont plus nécessairement une représentation rationnelle du champ sonore, mais deviennent une représentation abstraite de l’espace sonore possédant en soi, un potentiel musical. Cette thèse propose alors un nouveau modèle de spatialisation fondé sur une décomposition matricielle de l’espace sonore permettant de valider les hypothèses. / The creation of sound effects in space with Ambisonics highlights the musical potential of sound field decomposition by spherical harmonics, and redefines the representation of the sound space. This thesis defends that the abstract and intermediate representations of the sound space make it possible to develop new original approaches to sound spatialization. The reasoning that leads to this affirmation begins with the musical appropriation of the ambisonic approach. The creation of new space and sound processing patterns in Ambisonics leads to an original way of using signals associated with spherical harmonics, and to a different conception of the relations between them, and their hierarchization. The specificities of these experimental approaches and the singular characteristics of the sound fields generated call for the design of new theoretical and practical tools, for their analysis and restitution. The performed changes make it possible to free this approach from the initial technical and material issues of Ambisonics. But above all, it emancipates this approach from the psychoacoustic and acoustic models on which ambisonic techniques are originally defined. In this context, the signals associated with spherical harmonics are no longer necessarily a rational representation of the sound field but become an abstract representation of the sound space possessing in itself a musical potential. To validate the hypotheses, this thesis then proposes a new spatialization model based on a matrix decomposition of the sound space.

Steered Response Power based algorithms are widely used for finding sound source location using microphone array systems. SRCP-PHAT is one such algorithm that has a robust performance under noisy and reverberant conditions. The algorithm creates a likelihood function over the field of view. This thesis employs image processing methods on SRCP-PHAT images, to exploit the difference in power levels and pixel patterns to discriminate between sound source and background pixels. Hough Transform based ellipse detection is used to identify the sound source locations by finding the centers of elliptical edge pixel regions typical of source patterns. Monte Carlo simulations of an eight microphone perimeter array with single and multiple sound sources are used to simulate the test environment and area under receiver operating characteristic (ROCA) curve is used to analyze the algorithm performance. Performance was compared to a simpler algorithm involving Canny edge detection and image averaging and an algorithms based simply on the magnitude of local maxima in the SRCP image. Analysis shows that Canny edge detection based method performed better in the presence of coherent noise sources.