This thesis provides an overview of work conducted to investigate human spatial hearing in situations involving multiple concurrent sound sources. Much is known about spatial hearing with single sound sources, including the acoustic cues to source location and the accuracy of localisation under different conditions. However, more recently interest has grown in the behaviour of listeners in more complex environments. Concurrent sound sources pose a particularly difficult problem for the auditory system, as their identities and locations must be extracted from a common set of sensory receptors and shared computational machinery. It is clear that humans have a rich perception of their auditory world, but just how concurrent sounds are processed, and how accurately, are issues that are poorly understood. This work attempts to fill a gap in our understanding by systematically examining spatial resolution with multiple sound sources. A series of psychophysical experiments was conducted on listeners with normal hearing to measure performance in spatial localisation and discrimination tasks involving more than one source. The general approach was to present sources that overlapped in both frequency and time in order to observe performance in the most challenging of situations. Furthermore, the role of two primary sets of location cues in concurrent source listening was probed by examining performance in different spatial dimensions. The binaural cues arise due to the separation of the two ears, and provide information about the lateral position of sound sources. The spectral cues result from location-dependent filtering by the head and pinnae, and allow vertical and front-rear auditory discrimination. Two sets of experiments are described that employed relatively simple broadband noise stimuli. In the first of these, two-point discrimination thresholds were measured using simultaneous noise bursts. It was found that the pair could be resolved only if a binaural difference was present; spectral cues did not appear to be sufficient. In the second set of experiments, the two stimuli were made distinguishable on the basis of their temporal envelopes, and the localisation of a designated target source was directly examined. Remarkably robust localisation was observed, despite the simultaneous masker, and both binaural and spectral cues appeared to be of use in this case. Small but persistent errors were observed, which in the lateral dimension represented a systematic shift away from the location of the masker. The errors can be explained by interference in the processing of the different location cues. Overall these experiments demonstrated that the spatial perception of concurrent sound sources is highly dependent on stimulus characteristics and configurations. This suggests that the underlying spatial representations are limited by the accuracy with which acoustic spatial cues can be extracted from a mixed signal. Three sets of experiments are then described that examined spatial performance with speech, a complex natural sound. The first measured how well speech is localised in isolation. This work demonstrated that speech contains high-frequency energy that is essential for accurate three-dimensional localisation. In the second set of experiments, spatial resolution for concurrent monosyllabic words was examined using similar approaches to those used for the concurrent noise experiments. It was found that resolution for concurrent speech stimuli was similar to resolution for concurrent noise stimuli. Importantly, listeners were limited in their ability to concurrently process the location-dependent spectral cues associated with two brief speech sources. In the final set of experiments, the role of spatial hearing was examined in a more relevant setting containing concurrent streams of sentence speech. It has long been known that binaural differences can aid segregation and enhance selective attention in such situations. The results presented here confirmed this finding and extended it to show that the spectral cues associated with different locations can also contribute. As a whole, this work provides an in-depth examination of spatial performance in concurrent source situations and delineates some of the limitations of this process. In general, spatial accuracy with concurrent sources is poorer than with single sound sources, as both binaural and spectral cues are subject to interference. Nonetheless, binaural cues are quite robust for representing concurrent source locations, and spectral cues can enhance spatial listening in many situations. The findings also highlight the intricate relationship that exists between spatial hearing, auditory object processing, and the allocation of attention in complex environments.

This thesis provides an overview of work conducted to investigate human spatial hearing in situations involving multiple concurrent sound sources. Much is known about spatial hearing with single sound sources, including the acoustic cues to source location and the accuracy of localisation under different conditions. However, more recently interest has grown in the behaviour of listeners in more complex environments. Concurrent sound sources pose a particularly difficult problem for the auditory system, as their identities and locations must be extracted from a common set of sensory receptors and shared computational machinery. It is clear that humans have a rich perception of their auditory world, but just how concurrent sounds are processed, and how accurately, are issues that are poorly understood. This work attempts to fill a gap in our understanding by systematically examining spatial resolution with multiple sound sources. A series of psychophysical experiments was conducted on listeners with normal hearing to measure performance in spatial localisation and discrimination tasks involving more than one source. The general approach was to present sources that overlapped in both frequency and time in order to observe performance in the most challenging of situations. Furthermore, the role of two primary sets of location cues in concurrent source listening was probed by examining performance in different spatial dimensions. The binaural cues arise due to the separation of the two ears, and provide information about the lateral position of sound sources. The spectral cues result from location-dependent filtering by the head and pinnae, and allow vertical and front-rear auditory discrimination. Two sets of experiments are described that employed relatively simple broadband noise stimuli. In the first of these, two-point discrimination thresholds were measured using simultaneous noise bursts. It was found that the pair could be resolved only if a binaural difference was present; spectral cues did not appear to be sufficient. In the second set of experiments, the two stimuli were made distinguishable on the basis of their temporal envelopes, and the localisation of a designated target source was directly examined. Remarkably robust localisation was observed, despite the simultaneous masker, and both binaural and spectral cues appeared to be of use in this case. Small but persistent errors were observed, which in the lateral dimension represented a systematic shift away from the location of the masker. The errors can be explained by interference in the processing of the different location cues. Overall these experiments demonstrated that the spatial perception of concurrent sound sources is highly dependent on stimulus characteristics and configurations. This suggests that the underlying spatial representations are limited by the accuracy with which acoustic spatial cues can be extracted from a mixed signal. Three sets of experiments are then described that examined spatial performance with speech, a complex natural sound. The first measured how well speech is localised in isolation. This work demonstrated that speech contains high-frequency energy that is essential for accurate three-dimensional localisation. In the second set of experiments, spatial resolution for concurrent monosyllabic words was examined using similar approaches to those used for the concurrent noise experiments. It was found that resolution for concurrent speech stimuli was similar to resolution for concurrent noise stimuli. Importantly, listeners were limited in their ability to concurrently process the location-dependent spectral cues associated with two brief speech sources. In the final set of experiments, the role of spatial hearing was examined in a more relevant setting containing concurrent streams of sentence speech. It has long been known that binaural differences can aid segregation and enhance selective attention in such situations. The results presented here confirmed this finding and extended it to show that the spectral cues associated with different locations can also contribute. As a whole, this work provides an in-depth examination of spatial performance in concurrent source situations and delineates some of the limitations of this process. In general, spatial accuracy with concurrent sources is poorer than with single sound sources, as both binaural and spectral cues are subject to interference. Nonetheless, binaural cues are quite robust for representing concurrent source locations, and spectral cues can enhance spatial listening in many situations. The findings also highlight the intricate relationship that exists between spatial hearing, auditory object processing, and the allocation of attention in complex environments.

Este trabalho tem por objetivo desenvolver ferramentas que permitam gerar em fones-de-ouvido o efeito psicoacústico de fontes sonoras locomovendo-se no espaço, por meio da auralização do sinal monofônico original. Embora a auralização binaural possa ser feita empregando variações de atraso (chamadas ITD interaural time difference, ou diferença de tempo interaural) e de intensidade (chamadas ILD interaural level difference, ou diferença de nível interaural) entre os canais, melhores resultados psicoacústicos podem ser obtidos ao se utilizar filtros digitais conhecidos como HRTFs (head related transfer functions, ou funções de transferência relativas à cabeça). Uma HRTF insere no sinal monofônico informações que possibilitam ao sistema auditivo identificá-lo como proveniente de uma direção específica, direção esta que é única para cada HRTF. Para posicionar uma fonte estática em uma direção específica, bastaria, então, filtrar o sinal original pela HRTF da direção desejada. Se, no entanto, for desejável que a fonte se locomova em uma trajetória contínua, um número infinitamente grande de filtros seria necessário. Como eles são, normalmente, obtidos empiricamente, um número arbitrariamente alto deles não está disponível. Disso surge a necessidade de técnicas de interpolação de HRTFs, que possibilitem gerar os filtros intermediários não disponíveis. Este trabalho apresenta três novas técnicas de interpolação de HRTFs, para assim alcançar o objetivo de auralizar fontes sonoras móveis: a interpolação triangular, que é uma técnica de interpolação linear baseada na técnica de panorama sonoro VBAP (vector-based amplitude panning, ou panorama sonoro baseado em vetores); o método das movimentações discretas, que busca explorar o limiar de percepção do nosso sistema auditivo para, com isso, gerar uma técnica extremamente barata computacionalmente; e a interpolação espectral, que altera continuamente as estruturas das HRTFs para gerar filtros interpolados. São apresentadas também as implementações feitas dessas novas técnicas desenvolvidas, bem como os testes numéricos realizados para medir sua eficácia. / The goal of this work is the development of tools that allow simulating through headphones the psychoacoustic effect of sound sources moving in space, by the auralisation of the original monophonic signals. Although binaural auralisation can be implemented using variations in delays (called ITD interaural time difference) and in intensities (called ILD interaural level difference) among channels, better psychoacoustic results can be achieved using digital filters known as HRTFs (head related transfer functions). A HRTF inserts in the monophonic signal information that allow the auditory system to perceive this signal to be as if coming from a specific direction, which is unique for each single HRTF. Thus, to position a static sound source at a specific direction, filtering the original signal with the HRTF from the desired direction would be enough. Nevertheless, if it is desired that the sound source moves in a continuous trajectory, an infinitely large amount of filters would be necessary. Since they are usually obtained by measurements, such an arbitrarily large amount of them is not available. In this case, HRTF interpolation techniques that generate intermediary filters must be used. This work presents three new HRTF interpolation techniques in order to auralise moving sound sources: the triangular interpolation, a linear interpolation technique based on the VBAP amplitude panning technique; the discrete movements method, an extremely efficient technique that exploits the auditory systems limitations in perceiving very small changes in direction; and the spectral interpolation, that alters continuously the structures of the HRTFs to generate interpolated filters. Implementations of these techniques are discussed and numerical tests are also presented.

Wind turbines are an environmentally friendly and sustainable power source. Unfortunately, the noise impact can cause deteriorated living conditions for nearby residents. The audibility of wind turbine sound is influenced by ambient sound. This thesis deals with some aspects of noise from wind turbines. Ambient sounds influence the audibility of wind turbine noise. Models for assessing two commonly occurring natural ambient sounds namely vegetation sound and sound from breaking waves are presented in paper A and B. A sound propagation algorithm has been compared to long range measurementsof sound propagation in paper C. Psycho-acoustic tests evaluating the threshold and partial loudness of wind turbine noise when mixed with natural ambient sounds have been performed. These are accounted for in paper D. The main scientific contributions are the following.Paper A: A semi-empiric prediction model for vegetation sound is proposed. This model uses up-to-date simulations of wind profiles and turbulent wind fields to estimate sound from vegetation. The fluctuations due to turbulence are satisfactory estimated by the model. Predictions of vegetation sound also show good agreement to measured spectra. Paper B: A set of measurements of air-borne sound from breaking waves are reported. From these measurements a prediction method of sound from breaking waves is proposed. Third octave spectra from breaking waves are shown to depend on breaker type. Satisfactory agreement between predictions and measurements has been achieved. Paper C: Long range sound propagation over a sea surface was investigated. Measurements of sound transmission were coordinated with local meteorological measurements. A sound propagation algorithm has been compared to the measured sound transmission. Satisfactory agreement between measurements and predictions were achieved when turbulence were taken into consideration in the computations. Paper D: The paper investigates the interaction between wind turbine noise and natural ambient noise. Two loudness models overestimate the masking from two psychoacoustic tests. The wind turbine noise is completely concealed when the ambient sound level (A-weighed) is around 10 dB higher than the wind turbine noise level. Wind turbine noise and ambient noise were presented simultaneously at the same A-weighed sound level. The subjects then perceived the loudness of the wind turbine noise as 5 dB lower than if heard alone. Keywords: Wind turbine noise, masking, ambient noise, long range sound propagation / QC 20100705

Este trabalho tem por objetivo desenvolver ferramentas que permitam gerar em fones-de-ouvido o efeito psicoacústico de fontes sonoras locomovendo-se no espaço, por meio da auralização do sinal monofônico original. Embora a auralização binaural possa ser feita empregando variações de atraso (chamadas ITD interaural time difference, ou diferença de tempo interaural) e de intensidade (chamadas ILD interaural level difference, ou diferença de nível interaural) entre os canais, melhores resultados psicoacústicos podem ser obtidos ao se utilizar filtros digitais conhecidos como HRTFs (head related transfer functions, ou funções de transferência relativas à cabeça). Uma HRTF insere no sinal monofônico informações que possibilitam ao sistema auditivo identificá-lo como proveniente de uma direção específica, direção esta que é única para cada HRTF. Para posicionar uma fonte estática em uma direção específica, bastaria, então, filtrar o sinal original pela HRTF da direção desejada. Se, no entanto, for desejável que a fonte se locomova em uma trajetória contínua, um número infinitamente grande de filtros seria necessário. Como eles são, normalmente, obtidos empiricamente, um número arbitrariamente alto deles não está disponível. Disso surge a necessidade de técnicas de interpolação de HRTFs, que possibilitem gerar os filtros intermediários não disponíveis. Este trabalho apresenta três novas técnicas de interpolação de HRTFs, para assim alcançar o objetivo de auralizar fontes sonoras móveis: a interpolação triangular, que é uma técnica de interpolação linear baseada na técnica de panorama sonoro VBAP (vector-based amplitude panning, ou panorama sonoro baseado em vetores); o método das movimentações discretas, que busca explorar o limiar de percepção do nosso sistema auditivo para, com isso, gerar uma técnica extremamente barata computacionalmente; e a interpolação espectral, que altera continuamente as estruturas das HRTFs para gerar filtros interpolados. São apresentadas também as implementações feitas dessas novas técnicas desenvolvidas, bem como os testes numéricos realizados para medir sua eficácia. / The goal of this work is the development of tools that allow simulating through headphones the psychoacoustic effect of sound sources moving in space, by the auralisation of the original monophonic signals. Although binaural auralisation can be implemented using variations in delays (called ITD interaural time difference) and in intensities (called ILD interaural level difference) among channels, better psychoacoustic results can be achieved using digital filters known as HRTFs (head related transfer functions). A HRTF inserts in the monophonic signal information that allow the auditory system to perceive this signal to be as if coming from a specific direction, which is unique for each single HRTF. Thus, to position a static sound source at a specific direction, filtering the original signal with the HRTF from the desired direction would be enough. Nevertheless, if it is desired that the sound source moves in a continuous trajectory, an infinitely large amount of filters would be necessary. Since they are usually obtained by measurements, such an arbitrarily large amount of them is not available. In this case, HRTF interpolation techniques that generate intermediary filters must be used. This work presents three new HRTF interpolation techniques in order to auralise moving sound sources: the triangular interpolation, a linear interpolation technique based on the VBAP amplitude panning technique; the discrete movements method, an extremely efficient technique that exploits the auditory systems limitations in perceiving very small changes in direction; and the spectral interpolation, that alters continuously the structures of the HRTFs to generate interpolated filters. Implementations of these techniques are discussed and numerical tests are also presented.

A new technique to de-Dopplerize microphone signals from moving sources of sound is derived. Currently available time domain de-Dopplerization techniques require oversampling and interpolation of the microphone time data. In contrast, the technique presented in this dissertation performs the de-Dopplerization entirely in the frequency domain eliminating the need for oversampling and interpolation of the microphone data. As a consequence, the new de-Dopplerization technique is computationally more efficient. The new de-Dopplerization technique is then implemented into a frequency domain beamforming algorithm to locate moving sources of sound. The mathematical formulation for the implementation of the new de-Dopplerization technique is presented for sources moving along a linear trajectory and for sources moving along a circular trajectory, i.e. rotating sources. The resulting frequency domain beamforming method to locate moving sound sources is then validated using numerical simulations for various source configurations (e.g. emission angle, emission frequency, and source velocity), and different processing parameters (e.g. time window length). Numerical datasets for sources with linear motion as well as for rotating sources were simulated. For comparison purposes, selected datasets were also processed using traditional time domain beamforming. The results from the numerical simulations show that the frequency domain beamforming method is at least 10 times faster than the traditional time domain beamforming method with the same performance. Furthermore, the results show that as the number of microphones and/or grid points increase, the processing time for the traditional time domain beamforming method increases at a rate 20 times larger than the rate of increase in processing time of the new frequency domain beamforming method. / Ph. D.

Disertační práce se zabývá lokalizací zdrojů zvuku a akustickým zoomem. Hlavním cílem této práce je navrhnout systém s akustickým zoomem, který přiblíží zvuk jednoho mluvčího mezi skupinou mluvčích, a to i když mluví současně. Tento systém je kompatibilní s technikou prostorového zvuku. Hlavní přínosy disertační práce jsou následující: 1. Návrh metody pro odhad více směrů přicházejícího zvuku. 2. Návrh metody pro akustické zoomování pomocí DirAC. 3. Návrh kombinovaného systému pomocí předchozích kroků, který může být použit v telekonferencích.

L'objectif de ce travail de thèse est la conception d'un système de haut-parleurs transportable, capable de générer un champ sonore prédéfini et focalisé avec un contraste spatial élevé. Ce système doit permettre à terme d'effectuer différents types d'études, par exemple des essais de transparence acoustique ou encore des essais vibratoires en conditions non-anéchoïques. La minimisation du nombre de canaux à piloter ainsi que du nombre des transducteurs est l'un des enjeux principaux du travail. Le choix du nombre de sources et la sélection de leurs positions optimales afin de générer un champ acoustique cible n'a pas de solution triviale. Pour répondre à cette question, la méthode proposée se base sur la décomposition du rayonnement d’une source en série de fonctions orthogonales indépendantes (les"modes de rayonnement"), construits numériquement via une décomposition en valeurs singulières de la matrice d'impédance. En filtrant les termes évanescents, le champ lointain peut être reconstruit à l'aide d'un faible nombre de termes. De plus, la méthode permet d'estimer une distribution de débit efficace pour générer le champ cible. La méthode proposée étant relativement peu étudiée dans la littérature, la première partie de la thèse a été consacrée au problème de la validation expérimentale de la méthode directe et à l'étude des principaux paramètres en influençant le résultat. La problématique du positionnement des sources permettant de synthétiser un champ sonore prédéfini et focalisé est abordée dans la deuxième partie du travail. / The goal of this thesis is the design of a transportable speaker system, able to generate a predefined and focused sound field with a high spatial contrast. This system has eventually to allow carrying out different types of studies, for example acoustic transmission loss tests or vibration tests in non-anechoic conditions. The minimization of the number of driven channels and the number of transducers is one of the main goals of the work. The choice of the number of sources and the selection of their optimal positions in order to generate a target acoustic field has no trivial solution. To answer this question, the proposed method is based on the decomposition of the source radiation into a series of independent orthogonal functions (the "radiation modes"), constructed numerically via a singular value decomposition of the impedance matrix. By filtering the evanescent terms, the far field can be reconstructed using a small number of terms. In addition, the method allows the estimation of an efficient flow distribution to generate the target field. With the proposed method having been scarcely studied in the literature, the first part of the thesis is devoted to the problem of the experimental validation of the direct method and the study of the main parameters that are influencing the result. The problem of sources positioning in order to synthesize a predefined and focused sound field is discussed in the second part of the thesis.

Dans les systèmes d'échappement automobile, les sources de bruit d'origine aéro-acoustique représentent une partie importante du contenu fréquentiel, objectivement et subjectivement identifiable. De robustes procédures de tests ont été mises en place mais la simulation du contenu du bruit n'a pas encore fait ses preuves dans les processus de développement au quotidien. Cette thèse montre que le bruit aéro-acoustique provenant de sources type dipôle est dominant pour ce qui concerne les systèmes automobiles. La simulation des écoulements à l'origine de ces bruits spécifiques combinée avec les outils de calculs acoustiques classiques est très lourde voir tout simplement impossible. Le but de cette thèse est d'analyser la loi d'échelle pour des modèles de sources compactes, permettant de déterminer l'émission de la puissance acoustique selon différentes configurations géométriques "simples" et généralement répandues (par ex. tube perforé, diaphragme placé dans un tube…) basées sur des données empiriques. Il est démontré à l'aide de simulations que son utilisation est simple et que la précision de ces modèles de sources est satisfaisante si l'on ne s'écarte pas trop des géométries déjà analysées. / On automotive exhaust systems aero acoustic noise is a dominant and critical noise content, which is clearly objectively and subjectively detectable. Robust test procedures are established but the simulation of this noise content has not gained ground in the real life development processes. This thesis shows that the dominating characteristic of the aero acoustic noise of automotive systems is dipole noise. The simulation of these specific noise sources with classical computational areo acoustics is very cumbersome or even just impossible. The aim of the thesis is a review of the scaling law approach for compact source models, enabling the determination of the sound power emission of discret configurations based on empirical data. Application simulations show that the use of these source models is simple and that the accuracy is acceptable within the geometry limits analysed.

Ce travail de recherche porte sur le problème de l'identification des sources de bruit en espace clos. La motivation principale était de proposer une technique capable de localiser et quantifier les sources de bruit à l'intérieur des véhicules industriels, d'une manière efficace en temps. Dans cette optique, la méthode pourrait être utilisée par les industriels à des fins de réduction de bruit, et donc construire des véhicules plus silencieux. Un modèle simplifié basé sur la formulation par sources équivalentes a été utilisé pour résoudre le problème. Nous montrerons que le problème est mal conditionné, dans le sens où il est très sensible face aux erreurs de mesure, et donc des techniques dites de régularisation sont nécessaires. Une étude détaillée de cette question, en particulier le réglage de ce qu'on appelle de paramètre de régularisation, a été important pour assurer la stabilité de la solution. En particulier, un critère de régularisation basé sur une approche bayésienne s'est montré très robuste pour ajuster le paramètre de régularisation de manière optimale. L'application cible concernant des environnements intérieurs relativement grands, nous a imposé des difficultés supplémentaires, à savoir: (a) le positionnement de l'antenne de capteurs à l'intérieur de l'espace; (b) le nombre d'inconnues (sources potentielles) beaucoup plus important que le nombre de positions de mesure. Une formulation par pondération itérative a ensuite été proposé pour surmonter les problèmes ci-dessus de manière à: (1) corriger pour le positionnement de l'antenne de capteurs dans l'habitacle ; (2) obtenir des résultats corrects en terme de quantification des sources identifiées. Par ailleurs, l'approche itérative nous a conduit à des résultats avec une meilleure résolution spatiale ainsi qu'une meilleure dynamique. Plusieurs études numériques ont été réalisées afin de valider la méthode ainsi que d'évaluer sa sensibilité face aux erreurs de modèle. En particulier, nous avons montré que l'approche est affectée par des conditions non-anéchoïques, dans le sens où les réflexions sont identifiées comme des vraies sources. Une technique de post-traitement qui permet de distinguer entre les chemins directs et réverbérants a été étudiée. La dernière partie de cette thèse porte sur des validations expérimentales et applications pratiques de la méthode. Une antenne sphérique constituée d'une sphère rigide et 31 microphones a été construite pour les tests expérimentaux. Plusieurs validations académiques ont été réalisées dans des environnements semi-anéchoïques, et nous ont illustré les avantages et limites de la méthode. Enfin, l'approche a été testé dans une application pratique, qui a consisté à identifier les sources de bruit ou faiblesses acoustiques à l'intérieur d'un bus. / This thesis is concerned with the problem of noise source identification in closed spaces. The main motivation was to propose a technique which allows to locate and quantify noise sources within industrial vehicles, in a time-effective manner. In turn, the technique might be used by manufacturers for noise abatement purposes such as to provide quieter vehicles. A simplified model based on the equivalent source formulation was used to tackle the problem. It was shown that the problem is ill-conditioned, in the sense that it is very sensitive to errors in measurement data, thus regularization techniques were required. A detailed study of this issue, in particular the tuning of the so-called regularization parameter, was of importance to ensure the stability of the solution. In particular, a Bayesian regularization criterion was shown to be a very robust approach to optimally adjust the regularization parameter in an automated way. The target application concerns very large interior environments, which imposes additional difficulties, namely: (a) the positioning of the measurement array inside the enclosure; (b) a number of unknowns ("candidate" sources) much larger than the number of measurement positions. An iterative weighted formulation was then proposed to overcome the above issues by: first correct for the positioning of the array within the enclosure and second iteratively solve the problem in order to obtain a correct source quantification. In addition, the iterative approach has provided results with an enhanced spatial resolution and dynamic range. Several numerical studies have been carried out to validate the method as well as to evaluate its sensitivity to modeling errors. In particular, it was shown that the approach is affected by non-anechoic conditions, in the sense that reflections are identified as "real" sources. A post-processing technique which helps to distinguish between direct and reverberant paths has been discussed. The last part of the thesis was concerned with experimental validations and practical applications of the method. A custom spherical array consisting of a rigid sphere and 31 microphones has been built for the experimental tests. Several academic experimental validations have been carried out in semi-anechoic environments, which illustrated the advantages and limits of the method. Finally, the approach was tested in a practical application, which consisted in identifying noise sources inside a bus at driving conditions.