Égalisation adaptative et non invasive de la réponse temps-fréquence d'une petite salle

RÉSUMÉ Dans le cadre de cette recherche, on s’intéresse au son, à l’environnement dans lequel il se propage, à l’interaction entre l’onde de son et son canal de transmission ainsi qu’aux transformations induites par les composantes d’une chaine audio. Le contexte précis étudié est celui de l’écoute musicale sur haut-parleurs.Pour le milieu dans lequel l’onde se propage, comme pour tout canal de transmission, il existe des fonctions mathématiques permettant de caractériser les transformations induites par le canal sur un signal qui le traverse. Un signal électrique sert de signal d’excitation pour ce canal constitué en l’occurrence d’un amplificateur, d’un haut-parleur et de la salle dans laquelle a lieu l’écoute, qui selon ses caractéristiques, retourne en sortie à la position d’écoute une onde de son altérée. Réponse en fréquence, réponse à l’impulsion, fonction de transfert ; les mathématiques utilisées ne diffèrent en rien de celles servant communément à la caractérisation d’un canal de transmission ou à l’expression des fonctions liant les sorties d’un système linéaire à ses entrées. Naturellement, il y a un but à cet exercice de modélisation : l’obtention de la réponse de la chaine amplificateur/salle/haut-parleur rend possible sa correction. Il est commun dans bien des contextes d’écoute, qu’un filtre soit inséré dans la chaine audio entre la source (exemple : lecteur CD) et le haut-parleur qui transforme le signal électrique en signal acoustique propagé dans la salle. Ce filtre, dit « égalisateur », a pour but de compenser en fréquences l’effet des composantes de la chaine audio et de la salle sur le signal sonore y étant transmis. Ses propriétés découlent de celles de l’amplificateur, du haut-parleur et de la salle. Bien qu’analytiquement rigoureuse, l’approche physique, centrée sur la modélisation physique du haut-parleur et sur l’équation de propagation de l’onde acoustique, est mal adaptée aux salles à géométrie complexe ou changeante au fil du temps. La seconde approche, la modélisation expérimentale, abordée dans ce travail, fait abstraction des propriétés physiques. La chaine amplificateur/haut-parleur/salle y est plutôt vue comme une « boite noire » comprenant entrées et sorties. Le problème étudié est celui de la caractérisation d’un système électro-acoustique ayant comme unique entrée un signal émis à travers un haut-parleur dans une salle, et comme unique sortie le signal capté par un microphone placé à la position d’écoute. L’originalité de ce travail réside non seulement dans la technique développée pour en arriver à cette caractérisation, mais surtout dans les contraintes imposées dans la manière d’y arriver. La majorité des techniques documentées à ce jour font appel à des signaux d’excitation dédiés à la mesure ; des signaux dotés de caractéristiques favorables à la simplification du calcul de réponse impulsionnelle qui en découle. Des signaux connus sont émis à travers un haut-parleur et la réponse à leur excitation est captée à l’aide d’un microphone à la position d’écoute. L’exercice de mesure lui-même pose problème, notamment, lorsqu’un auditoire est présent dans la salle. Aussi, la réponse de la salle peut changer entre le moment de la prise de mesure et l’écoute si la salle est reconfigurée, par exemple un rideau est tiré ou une estrade déplacée. Dans le cas d’une salle de spectacle, le haut-parleur utilisé peut varier selon le contexte. Un recensement des travaux dans lesquels des solutions à ce problème sont suggérées fut effectué. Le principal objectif est de développer une méthode innovatrice permettant de capturer la réponse impulsionnelle de la chaine audio à l’insu de l’auditoire. Pour ce faire, aucun signal dédié à la mesure ne doit être utilisé. La méthode développée permet la capture de la réponse impulsionnelle électro-acoustique en n’exploitant que les signaux musicaux. Le résultat est, un algorithme permettant la modélisation dynamique et en continu de la réponse d’une salle. Un filtre égalisateur numérique à réponse impulsionnelle finie doit être conçu, lui aussi capable de s’adapter dynamiquement au comportement de la salle, même lorsque celui-ci varie au fil du temps. La familiarisation avec des concepts plus avancés de programmation C++ orienté objet étant de mise, une technique permettant d’exploiter des signaux musicaux afin d’obtenir la réponse impulsionnelle et la réponse en fréquence du système fut testée expérimentalement sous forme d’un module VST. L’excitation est procurée par les signaux musicaux émis sur haut-parleurs durant l’écoute. Une moyenne mobile pondérée reconstruit statistiquement, au fil du temps, la réponse de la salle sur toute la plage de fréquences audibles. Dans le but d’en quantifier la performance, la réponse en fréquence obtenue est comparée à celle obtenue par une méthode standard servant de référence. L’erreur quadratique moyenne sert de métrique d’erreur et montre que plus la musique défile, plus la réponse en fréquence obtenue s’apparente à la référence pour un même point d’écoute. Une approche à résolution spectrale variable est utilisée pour construire, par bandes de fréquences, la réponse du filtre découlant de celle de la chaine audio. La réponse en fréquence du système corrigée par le filtre égalisateur est plus plane que celle du système initial. Des techniques explorées dans le cadre de ce travail de recherche ont mené à la publication d’un article scientifique dans une revue à comité de lecture et un article de conférence dans lesquels des méthodes similaires furent exploitées en génie des mines.----------ABSTRACT In this research, we are interested in sound, environment wherein it propagates, the interaction between the sound wave and a transmission channel, and the changes induced by the components of an audio chain. The specific context studied is that of listening to music on loudspeakers. For the environment in which sound wave propagates, like for any transmission channel, there are mathematical functions used to characterize the changes induced by a channel on the signal therethrough. An electric signal serves as a input for a system, in this case consisting of an amplifier, a loudspeaker, and the room where the listening takes place, which according to its characteristics, returns as an output at the listening position, an altered sound wave. Frequency response, impulse response, transfer function, the mathematics used are no different from those used commonly for the characterization of a transmission channel or the expression of the outputs of a linear system to its inputs. Naturally, there is a purpose to this modeling exercise: getting the frequency response of the amplifier/loundspeaker/room chain makes possible its equalization. It is common in many contexts of listening to find a filter inserted into the audio chain between the source (Eg CD player) and the amplifier/loudspeaker that converts the electrical signal to an acoustic signal propagated in the room. This filter, called “equalizer” is intended to compensate the frequency effect of the components of the audio chain and the room on the sound signal that will be transmitted. Properties for designing this filter are derived from those of the audio chain. Although analytically rigorous, physical approach, focusing on physical modeling of the loudspeaker and the propagation equation of the acoustic wave is ill-suited to rooms with complex geometry and changing over time. The second approach, experimental modeling, and therefore that addressed in this work, ignores physical properties. The system audio chain is rather seen as a “black box” including inputs and outputs. The problem studied is the characterization of an electro-acoustic system as having a single input signal transmitted through a speaker in a room, and a single output signal picked up by a microphone at the listening position. The originality of this work lies not only in the technique developed to arrive at this characterization, but especially in the constraints imposed in order to get there. The majority of technics documented to this date involve using excitation signals dedicated the measure; signals with favorable characteristics to simplify the calculation of the impulse response of the audio chain. Known signals are played through a loudspeaker and the room’s response to excitation is captured with a microphone at the listening position. The measurement exercise itself poses problem, especially when there is an audience in the room. Also, the response of the room may change between the time of the measurement and time of listening. If the room is reconfigured for example, a curtain is pulled or the stage moved. In the case of a theater, the speaker used may vary depending on the context. A survey of work in which solutions to this problem are suggested was made. The main objective is to develop an innovative method to capture the impulse response of an audio chain without the knowledge of the audience. To do this, no signal dedicated to the measurement should be used. The developed method allows the capture of the electro-acoustic impulse response exploiting only the music signals when it comes to a concert hall or using a movie sound track when a movie is a movie theater. As a result, an algorithm for modeling dynamicly and continuously the response of a room. A finite impulse response filter acting as a digital equalizer must be designed and also able to dynamically adapt the behavior of the room, even when it varies over time. Familiarization with more advanced programming concepts of object-oriented C++ being put, a technique to exploit music signals to obtain the impulse response and frequency response of the audio chain was implemented as a VST module and tested experimentally. The excitation is provided by music signals played through speakers. Using a weighted moving average reconstructed statistically over time, the response of the room on the entire audible frequency range is obtained. In order to quantify the performance the frequency response obtained is compared with that obtained by using a standard reference method. The mean square error is used as an error metric and shows that more music scrolls, more the frequency response obtained is similar to the reference one for the same listening position. A multi spectral resolution method is used to build, for diffrent frequency bands, the filter response arising from the inversion of the room/speaker frequency response. The resulting dynamically adapting filter has properties similar to those of the human ear, a significant spectral-resolution in lower frequencies, and high time-resolution at high frequencies. The response corrected by the filter system tends approaching to a pure pulse. Techniques explored in the context of this research led to the publication of a scientific article in a peer reviewed journal and one conference paper in which similar methods were used for mining engineering applications

The generative, analytic and instructional capacities of sound in architecture : fundamentals, tools and evaluation of a design methodology

Premi extraordinari doctorat UPC curs 2017-2018, Àmbit d’Arquitectura, Urbanisme i EdificacióThe disciplines of space and time form two domains to which it is daring to compare, since it is obvious that they are of a different nature. Music happens in time, while architecture happens in space. However, from the first treatises on both architecture and music, repeated calls for comparison, complementarity and influence of both disciplines can be read, at least to the observation of certain common orders between the two domains. In this doctoral thesis we do not question this whole theoretical corpus that has been enriching the relationship between both disciplines. We received it and joined that stream of knowledge. What we do notice, however, is the almost impertinent question that follows: can sound help the architect in his daily tasks? And, therefore, what are the contributions of sound to the architect? To do this we must seek the connection in the principles of both arts, where we can detach ourselves from time and space, and approach the most universal of art forms. The architect, in his daily work, is faced with three particular tasks: the architectural project, the architectural analysis and the teaching of architecture. Each of the three tasks is connected with the other two tasks: the project is carried out again with the analysis and transmitted to the new architect; the analysis supports the project decisions and gives tools to the disciple; and the teaching has the project as its purpose and the analysis as its method. The thesis presented here shows what sound offers to the task of the project, to that of analysis and to that of teaching. These three tasks are approached from three premises: theoretical foundations, tools and evaluation. The interaction of the three tasks with the three premises gives rise to nine lines of work that articulate the chapters of the thesis. The first, fourth and seventh chapters approach the three tasks from the premise of theoretical foundations, foundations that perhaps because they are obvious, have been ignored or overlooked but which constitute the nature of both disciplines. The first shows, by the hand of two 20th century authors - the architect Dom Hans van der Laan and the composer Olivier Messiaen - that creation in both disciplines is of a systematic nature. The fourth one revaluates the analytical systems of representation of form both in architecture and in music which, starting with the basic characteristics of its elements, lead to a symbolic notation and a tool for the analysis of the work: the plan and the score. The seventh introduces the student of architecture to the growing separation between music and architecture that has been accentuated to this day. The second, fifth and eighth chapters approach the three particular tasks from the premise of tools, working tools that help to understand more directly the influence of architecture on sound. The second places virtual reality and auralization techniques at the service of the architectural and urban planning project, enhancing the sound experience in these projects. The fifth deals with the acoustic analysis of exterior spaces and their relationship with the urban configuration of these spaces. The eighth section presents the study of acoustic heritage as an educational tool. The third, sixth and ninth chapters deal with the three tasks from the premise of evaluation, a check that ensures the influence of sound on them through teaching experiments. The third argues and exemplifies that a sound landscape can be the engine and generator of an architectural design. The sixth one reviews the methods for evaluating the subjective and objective parameters of architectural acoustics. The ninth shows that in teaching sound to architects, "learning by listening" should be given priority over "passive learning".Las disciplinas del espacio y del tiempo forman dos dominios a los que resulta atrevido comparar, pues es obvio que son de naturaleza distinta. La música ocurre en el tiempo, mientras que la arquitectura en el espacio. No obstante, desde los primeros tratados tanto de arquitectura como de música, se pueden leer repetidas llamadas a la comparación, al complemento y a la influencia de ambas disciplinas, cuanto menos a la constatación de ciertos órdenes comunes entre ambos dominios. En esta tesis doctoral no ponemos en cuestión todo este corpus teórico que ha venido enriqueciendo la relación entre ambas disciplinas. La recibimos y nos unimos a esa corriente de conocimiento. En lo que sí reparamos, en cambio, es en la pregunta casi impertinente que surge seguidamente: ¿puede el sonido ayudar al arquitecto en sus tareas diarias? Y, por tanto, ¿cuáles son las contribuciones del sonido para el arquitecto? Para ello debemos buscar la conexión en los principios de ambas artes, allí donde podemos despegarnos del tiempo y del espacio, y acercarnos a la más universal de las formas de arte. El arquitecto, en su tarea diaria, se enfrenta a tres tareas particulares: el proyecto arquitectónico, el análisis arquitectónico y la enseñanza de la arquitectura. Cada una de las tres tareas está conectada con las otras dos: el proyecto se reconduce con el análisis y se transmite al nuevo arquitecto; el análisis soporta las decisiones de proyecto y da herramientas al discípulo; y la enseñanza tiene como fin el proyecto y como método el análisis. La tesis aquí presentada pone de manifiesto lo que el sonido ofrece a la tarea del proyecto, a la del análisis y a la de la enseñanza. Estas tres tareas son abordadas desde tres premisas: los fundamentos teóricos, las herramientas y la evaluación. La interacción de las tres tareas con las tres premisas da lugar a nueve líneas de trabajo que articulan los capítulos de la tesis. Los capítulos primero, cuarto y séptimo abordan las tres tareas desde la premisa de los fundamentos teóricos, fundamentos que quizá por ser obvios, se han obviado o pasado por alto pero que constituyen la naturaleza de ambas disciplinas. El primero muestra, de la mano de dos autores del siglo XX -el arquitecto Dom Hans van der Laan y el compositor Olivier Messiaen- que la creación en ambas disciplinas es de naturaleza sistemática. El cuarto revaloriza los sistemas analíticos de representación de la forma tanto en arquitectura como en música que, empezando por las características básicas de sus elementos, conducen a una notación simbólica y una herramienta de análisis de la obra: el plano y la partitura. El séptimo presenta al estudiante de arquitectura la creciente separación entre la música y la arquitectura que se ha venido acentuando hasta nuestros días. Los capítulos segundo, quinto y octavo abordan las tres tareas particulares desde la premisa de las herramientas, útiles de trabajo que ayudan a comprender de modo más directo la influencia de la arquitectura en el sonido. El segundo sitúa la realidad virtual y las técnicas de auralización al servicio del proyecto de arquitectura y urbanismo, potenciando la experiencia sonora en estos proyectos. El quinto aborda el análisis acústico de espacios exteriores y su relación con la configuración urbana de estos espacios. El octavo presenta el estudio del patrimonio acústico como herramienta pedagógica. Los capítulos tercero, sexto y noveno abordan las tres tareas desde la premisa de la evaluación, comprobación que asegura mediante experimentos docentes la influencia del sonido en ellas. El tercero argumenta y ejemplifica que un paisaje sonoro puede ser el motor y generador de un diseño arquitectónico. El sexto realiza una revisión de los métodos de evaluación de los parámetros subjetivos y objetivos de la acústica arquitectónica. El noveno muestra que en la enseñanza del sonido para los arquitectos debe priorizarse "aprender escuchando" antes que el "aprendizaje pasivo".Award-winningPostprint (published version

The generative, analytic and instructional capacities of sound in architecture : fundamentals, tools and evaluation of a design methodology

The disciplines of space and time form two domains to which it is daring to compare, since it is obvious that they are of a different nature. Music happens in time, while architecture happens in space. However, from the first treatises on both architecture and music, repeated calls for comparison, complementarity and influence of both disciplines can be read, at least to the observation of certain common orders between the two domains. In this doctoral thesis we do not question this whole theoretical corpus that has been enriching the relationship between both disciplines. We received it and joined that stream of knowledge. What we do notice, however, is the almost impertinent question that follows: can sound help the architect in his daily tasks? And, therefore, what are the contributions of sound to the architect? To do this we must seek the connection in the principles of both arts, where we can detach ourselves from time and space, and approach the most universal of art forms. The architect, in his daily work, is faced with three particular tasks: the architectural project, the architectural analysis and the teaching of architecture. Each of the three tasks is connected with the other two tasks: the project is carried out again with the analysis and transmitted to the new architect; the analysis supports the project decisions and gives tools to the disciple; and the teaching has the project as its purpose and the analysis as its method. The thesis presented here shows what sound offers to the task of the project, to that of analysis and to that of teaching. These three tasks are approached from three premises: theoretical foundations, tools and evaluation. The interaction of the three tasks with the three premises gives rise to nine lines of work that articulate the chapters of the thesis. The first, fourth and seventh chapters approach the three tasks from the premise of theoretical foundations, foundations that perhaps because they are obvious, have been ignored or overlooked but which constitute the nature of both disciplines. The first shows, by the hand of two 20th century authors - the architect Dom Hans van der Laan and the composer Olivier Messiaen - that creation in both disciplines is of a systematic nature. The fourth one revaluates the analytical systems of representation of form both in architecture and in music which, starting with the basic characteristics of its elements, lead to a symbolic notation and a tool for the analysis of the work: the plan and the score. The seventh introduces the student of architecture to the growing separation between music and architecture that has been accentuated to this day. The second, fifth and eighth chapters approach the three particular tasks from the premise of tools, working tools that help to understand more directly the influence of architecture on sound. The second places virtual reality and auralization techniques at the service of the architectural and urban planning project, enhancing the sound experience in these projects. The fifth deals with the acoustic analysis of exterior spaces and their relationship with the urban configuration of these spaces. The eighth section presents the study of acoustic heritage as an educational tool. The third, sixth and ninth chapters deal with the three tasks from the premise of evaluation, a check that ensures the influence of sound on them through teaching experiments. The third argues and exemplifies that a sound landscape can be the engine and generator of an architectural design. The sixth one reviews the methods for evaluating the subjective and objective parameters of architectural acoustics. The ninth shows that in teaching sound to architects, "learning by listening" should be given priority over "passive learning".Las disciplinas del espacio y del tiempo forman dos dominios a los que resulta atrevido comparar, pues es obvio que son de naturaleza distinta. La música ocurre en el tiempo, mientras que la arquitectura en el espacio. No obstante, desde los primeros tratados tanto de arquitectura como de música, se pueden leer repetidas llamadas a la comparación, al complemento y a la influencia de ambas disciplinas, cuanto menos a la constatación de ciertos órdenes comunes entre ambos dominios. En esta tesis doctoral no ponemos en cuestión todo este corpus teórico que ha venido enriqueciendo la relación entre ambas disciplinas. La recibimos y nos unimos a esa corriente de conocimiento. En lo que sí reparamos, en cambio, es en la pregunta casi impertinente que surge seguidamente: ¿puede el sonido ayudar al arquitecto en sus tareas diarias? Y, por tanto, ¿cuáles son las contribuciones del sonido para el arquitecto? Para ello debemos buscar la conexión en los principios de ambas artes, allí donde podemos despegarnos del tiempo y del espacio, y acercarnos a la más universal de las formas de arte. El arquitecto, en su tarea diaria, se enfrenta a tres tareas particulares: el proyecto arquitectónico, el análisis arquitectónico y la enseñanza de la arquitectura. Cada una de las tres tareas está conectada con las otras dos: el proyecto se reconduce con el análisis y se transmite al nuevo arquitecto; el análisis soporta las decisiones de proyecto y da herramientas al discípulo; y la enseñanza tiene como fin el proyecto y como método el análisis. La tesis aquí presentada pone de manifiesto lo que el sonido ofrece a la tarea del proyecto, a la del análisis y a la de la enseñanza. Estas tres tareas son abordadas desde tres premisas: los fundamentos teóricos, las herramientas y la evaluación. La interacción de las tres tareas con las tres premisas da lugar a nueve líneas de trabajo que articulan los capítulos de la tesis. Los capítulos primero, cuarto y séptimo abordan las tres tareas desde la premisa de los fundamentos teóricos, fundamentos que quizá por ser obvios, se han obviado o pasado por alto pero que constituyen la naturaleza de ambas disciplinas. El primero muestra, de la mano de dos autores del siglo XX ?el arquitecto Dom Hans van der Laan y el compositor Olivier Messiaen- que la creación en ambas disciplinas es de naturaleza sistemática. El cuarto revaloriza los sistemas analíticos de representación de la forma tanto en arquitectura como en música que, empezando por las características básicas de sus elementos, conducen a una notación simbólica y una herramienta de análisis de la obra: el plano y la partitura. El séptimo presenta al estudiante de arquitectura la creciente separación entre la música y la arquitectura que se ha venido acentuando hasta nuestros días. Los capítulos segundo, quinto y octavo abordan las tres tareas particulares desde la premisa de las herramientas, útiles de trabajo que ayudan a comprender de modo más directo la influencia de la arquitectura en el sonido. El segundo sitúa la realidad virtual y las técnicas de auralización al servicio del proyecto de arquitectura y urbanismo, potenciando la experiencia sonora en estos proyectos. El quinto aborda el análisis acústico de espacios exteriores y su relación con la configuración urbana de estos espacios. El octavo presenta el estudio del patrimonio acústico como herramienta pedagógica. Los capítulos tercero, sexto y noveno abordan las tres tareas desde la premisa de la evaluación, comprobación que asegura mediante experimentos docentes la influencia del sonido en ellas. El tercero argumenta y ejemplifica que un paisaje sonoro puede ser el motor y generador de un diseño arquitectónico. El sexto realiza una revisión de los métodos de evaluación de los parámetros subjetivos y objetivos de la acústica arquitectónica. El noveno muestra que en la enseñanza del sonido para los arquitectos debe priorizarse "aprender escuchando" antes que el "aprendizaje pasivo"

Blind estimation of room acoustic parameters from speech and music signals

The acoustic character of a space is often quantified using objective room acoustic parameters. The measurement of these parameters is difficult in occupied conditions and thus measurements are usually performed when the space is un-occupied. This is despite the knowledge that occupancy can impact significantly on the measured parameter value. Within this thesis new methods are developed by which naturalistic signals such as speech and music can be used to perform acoustic parameter measurement. Adoption of naturalistic signals enables passive measurement during orchestral performances and spoken announcements, thus facilitating easy in-situ measurement. Two methods are described within this work; (1) a method utilising artificial neural networks where a network is taught to recognise acoustic parameters from received, reverberated signals and (2) a method based on the maximum likelihood estimation of the decay curve of the room from which parameters are then calculated. (1) The development of the neural network method focuses on a new pre-processor for use with music signals. The pre-processor utilises a narrow band filter bank with centre frequencies chosen based on the equal temperament scale. The success of a machine learning method is linked to the quality of the training data and therefore realistic acoustic simulation algorithms were used to generate a large database of room impulse responses. Room models were defined with realistic randomly generated geometries and surface properties; these models were then used to predict the room impulse responses. (2) In the second approach, a statistical model of the decay of sound in a room was further developed. This model uses a maximum likelihood (ML) framework to yield a number of decay curve estimates from a received reverberant signal. The success of the method depends on a number of stages developed for the algorithm; (a) a pre-processor to select appropriate decay phases for estimation purposes, (b) a rigorous optimisation algorithm to ensure the correct maximum likelihood estimate is found and (c) a method to yield a single optimum decay curve estimate from which the parameters are calculated. The ANN and ML methods were tested using orchestral music and speech signals. The ANN method tended to perform well when estimating the early decay time (EDT), for speech and music signals the error was within the subjective difference limens. However, accuracy was reduced for the reverberation time (Rt) and other parameters. By contrast the ML method performed well for Rt with results for both speech and music within the difference limens for reasonable (<4s) reverberation time. In addition reasonable accuracy was found for EDT, Clarity (C80), Centre time (Ts) and Deutichkeit (D). The ML method is also capable of producing accurate estimates of the binaural parameters Early Lateral Energy Fraction (LEF) and the late lateral strength (LG). A number of real world measurements were carried out in concert halls where the ML accuracy was shown to be sufficient for most parameters. The ML method has the advantage over the ANN method due to its truly blind nature (the ANN method requires a period of learning and is therefore semi-blind). The ML method uses gaps of silence between notes or utterances, when these silence regions are not present the method does not produce an estimate. Accurate estimation requires a long recording (hours of music or many minutes of speech) to ensure that at least some silent regions are present. This thesis shows that, given a sufficiently long recording, accurate estimates of many acoustic parameters can be obtained directly from speech and music. Further extensions to the ML method detailed in this thesis combine the ML estimated decay curve with cepstral methods which detect the locations of early reflections. This improves the accuracy of many of the parameter estimates.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

Room acoustic parameter extraction from music signals

A new method, employing machine learning techniques and a modified low frequency envelope spectrum estimator, for estimating important room acoustic parameters including reverberation time (RT) and early decay time (EDT) from received music signals has been developed. It overcomes drawbacks found in applying music signals directly to the envelope spectrum detector developed for the estimation of RT from speech signals. The octave band music signal is first separated into sub bands corresponding to notes on the equal temperament scale and the level of each note normalised before applying an envelope spectrum detector. A typical artificial neural network is then trained to map these envelope spectra onto RT or EDT. Significant improvements in estimation accuracy were found and further investigations confirmed that the non-stationary nature of music envelopes is a major technical challenge hindering accurate parameter extraction from music and the proposed method to some extent circumvents the difficulty

Room acoustic parameter extraction from music signals

A new method, employing machine learning techniques and a modified low frequency envelope spectrum estimator, for estimating important room acoustic parameters including Reverberation Time (RT) and Early Decay Time (EDT) from received music signals has been developed. It overcomes drawbacks found in applying music signals directly to the envelope spectrum detector developed for the estimation of RT from speech signals. The octave band music signal is first separated into sub bands corresponding to notes on the equal temperament scale and the level of each note normalised before applying an envelope spectrum detector. A typical artificial neural network is then trained to map these envelope spectra onto RT or EDT. Significant improvements in estimation accuracy were found and further investigations confirmed that the non-stationary nature of music envelopes is a major technical challenge hindering accurate parameter extraction from music and the proposed method to some extent circumvents the difficulty. © 2006 IEEE

Blind estimation of room acoustic parameters from speech and music signals

The acoustic character of a space is often quantified using objective room acoustic parameters. The measurement of these parameters is difficult in occupied conditions and thus measurements are usually performed when the space is un-occupied. This is despite the knowledge that occupancy can impact significantly on the measured parameter value. Within this thesis new methods are developed by which naturalistic signals such as speech and music can be used to perform acoustic parameter measurement. Adoption of naturalistic signals enables passive measurement during orchestral performances and spoken announcements, thus facilitating easy in-situ measurement.Two methods are described within this work; (1) a method utilising artificial neural networks where a network is taught to recognise acoustic parameters from received, reverberated signals and (2) a method based on the maximum likelihood estimation of the decay curve of the room from which parameters are then calculated. (1)The development of the neural network method focuses on a new pre-processor for use with music signals. The pre-processor utilises a narrow band filter bank with centre frequencies chosen based on the equal temperament scale. The success of a machine learning method is linked to the quality of the training data and therefore realistic acoustic simulation algorithms were used to generate a large database of room impulse responses. Room models were defined with realistic randomly generated geometries and surface properties; these models were then used to predict the room impulse responses.(2)In the second approach, a statistical model of the decay of sound in a room was further developed. This model uses a maximum likelihood (ML) framework to yield a number of decay curve estimates from a received reverberant signal. The success of the method depends on a number of stages developed for the algorithm; (a) a pre-processor to select appropriate decay phases for estimation purposes, (b) a rigorous optimisation algorithm to ensure the correct maximum likelihood estimate is found and (c) a method to yield a single optimum decay curve estimate from which the parameters are calculated.The ANN and ML methods were tested using orchestral music and speech signals. The ANN method tended to perform well when estimating the early decay time (EDT), for speech and music signals the error was within the subjective difference limens. However, accuracy was reduced for the reverberation time (Rt) and other parameters. By contrast the ML method performed well for Rt with results for both speech and music within the difference limens for reasonable (<4s) reverberation time. In addition reasonable accuracy was found for EDT, Clarity (C80), Centre time (Ts) and Deutichkeit (D). The ML method is also capable of producing accurate estimates of the binaural parameters Early Lateral Energy Fraction (LEF) and the late lateral strength (LG).A number of real world measurements were carried out in concert halls where the ML accuracy was shown to be sufficient for most parameters. The ML method has the advantage over the ANN method due to its truly blind nature (the ANN method requires a period of learning and is therefore semi-blind). The ML method uses gaps of silence between notes or utterances, when these silence regions are not present the method does not produce an estimate. Accurate estimation requires a long recording (hours of music or many minutes of speech) to ensure that at least some silent regions are present. This thesis shows that, given a sufficiently long recording, accurate estimates of many acoustic parameters can be obtained directly from speech and music.Further extensions to the ML method detailed in this thesis combine the ML estimated decay curve with cepstral methods which detect the locations of early reflections. This improves the accuracy of many of the parameter estimates