Advances in Architectural Acoustics

Satisfactory acoustics is crucial for the ability of spaces such as auditoriums and lecture rooms to perform their primary function. The acoustics of dwellings and offices greatly affects the quality of our life, since we are all consciously or subconsciously aware of the sounds to which we are daily subjected. Architectural acoustics, which encompasses room and building acoustics, is the scientific field that deals with these topics and can be defined as the study of generation, propagation, and effects of sound in enclosures. Modeling techniques, as well as related acoustic theories for accurately calculating the sound field, have been the center of many major new developments. In addition, the image conveyed by a purely physical description of sound would be incomplete without regarding human perception; hence, the interrelation between objective stimuli and subjective sensations is a field of important investigations. A holistic approach in terms of research and practice is the optimum way for solving the perplexing problems which arise in the design or refurbishment of spaces, since current trends in contemporary architecture, such as transparency, openness, and preference for bare sound-reflecting surfaces are continuing pushing the very limits of functional acoustics. All the advances in architectural acoustics gathered in this Special Issue, we hope that inspire researchers and acousticians to explore new directions in this age of scientific convergence

Measurement-Based Automatic Parameterization of a Virtual Acoustic Room Model

Modernien auralisaatiotekniikoiden ansiosta kuulokkeilla voidaan tuottaa kuuntelukokemus, joka muistuttaa useimpien äänitteiden tuotannossa oletettua kaiutinkuuntelua. Huoneakustinen mallinnus on tärkeä osa toimivaa auralisaatiojärjestelmää. Huonemallinnuksen parametrien määrittäminen vaatii kuitenkin ammattitaitoa ja aikaa. Tässä työssä kehitetään järjestelmä parametrien automaattiseksi määrittämiseksi huoneakustisten mittausten perusteella. Parametrisaatio perustuu mikrofoniryhmällä mitattuihin huoneen impulssivasteisiin ja voidaan jakaa kahteen osaan: suoran äänen ja aikaisten heijastusten analyysiin sekä jälkikaiunnan analyysiin. Suorat äänet erotellaan impulssivasteista erilaisia signaalinkäsittelytekniikoita käyttäen ja niitä hyödynnetään heijastuksia etsivässä algoritmissa. Äänilähteet ja heijastuksia vastaavat kuvalähteet paikannetaan saapumisaikaeroon perustuvalla paikannusmenetelmällä ja taajuusriippuvat etenemistien vaikutukset arvioidaan kuvalähdemallissa käyttöä varten. Auralisaation jälkikaiunta on toteutettu takaisinkytkevällä viiveverkostomallilla. Sen parametrisointi vaatii taajuusriippuvan jälkikaiunta-ajan ja jälkikaiunnan taajuusvasteen määrittämistä. Normalisoitua kaikutiheyttä käytetään jälkikaiunnan alkamisajan löytämiseen mittauksista ja simuloidun jälkikaiunnan alkamisajan asettamiseen. Jälkikaiunta-aikojen määrittämisessä hyödynnetään energy decay relief -metodia. Kuuntelukokeiden perusteella automaattinen parametrisaatiojärjestelmä tuottaa parempia tuloksia kuin parametrien asettaminen manuaalisesti huoneen summittaisten geometriatietojen pohjalta. Järjestelmässä on ongelmia erityisesti jälkikaiunnan ekvalisoinnissa, mutta käytettyihin suhteellisen yksinkertaisiin tekniikoihin nähden järjestelmä toimii hyvin.Modern auralization techniques enable making the headphone listening experience similar to the experience of listening with loudspeakers, which is the reproduction method most content is made to be listened with. Room acoustic modeling is an essential part of a plausible auralization system. Specifying the parameters for room modeling requires expertise and time. In this thesis, a system is developed for automatic analysis of the parameters from room acoustic measurements. The parameterization is based on room impulse responses measured with a microphone array and can be divided into two parts: the analysis of the direct sound and early reflections, and the analysis of the late reverberation. The direct sounds are separated from the impulse responses using various signal processing techniques and used in the matching pursuit algorithm to find the reflections in the impulse responses. The sound sources and their reflection images are localized using time difference of arrival -based localization and frequency-dependent propagation path effects are estimated for use in an image source model. The late reverberation of the auralization is implemented using a feedback delay network. Its parameterization requires the analysis of the frequency-dependent reverberation time and frequency response of the late reverberation. Normalized echo density is used to determine the beginning of the late reverberation in the measurements and to set the starting point of the modeled late field. The reverberation times are analyzed using the energy decay relief. A formal listening test shows that the automatic parameterization system outperforms parameters set manually based on approximate geometrical data. Problems remain especially in the precision of the late reverberation equalization but the system works well considering the relative simplicity of the processing methods used

Measurement-Based Automatic Parameterization of a Virtual Acoustic Room Model

Modernien auralisaatiotekniikoiden ansiosta kuulokkeilla voidaan tuottaa kuuntelukokemus, joka muistuttaa useimpien äänitteiden tuotannossa oletettua kaiutinkuuntelua. Huoneakustinen mallinnus on tärkeä osa toimivaa auralisaatiojärjestelmää. Huonemallinnuksen parametrien määrittäminen vaatii kuitenkin ammattitaitoa ja aikaa. Tässä työssä kehitetään järjestelmä parametrien automaattiseksi määrittämiseksi huoneakustisten mittausten perusteella. Parametrisaatio perustuu mikrofoniryhmällä mitattuihin huoneen impulssivasteisiin ja voidaan jakaa kahteen osaan: suoran äänen ja aikaisten heijastusten analyysiin sekä jälkikaiunnan analyysiin. Suorat äänet erotellaan impulssivasteista erilaisia signaalinkäsittelytekniikoita käyttäen ja niitä hyödynnetään heijastuksia etsivässä algoritmissa. Äänilähteet ja heijastuksia vastaavat kuvalähteet paikannetaan saapumisaikaeroon perustuvalla paikannusmenetelmällä ja taajuusriippuvat etenemistien vaikutukset arvioidaan kuvalähdemallissa käyttöä varten. Auralisaation jälkikaiunta on toteutettu takaisinkytkevällä viiveverkostomallilla. Sen parametrisointi vaatii taajuusriippuvan jälkikaiunta-ajan ja jälkikaiunnan taajuusvasteen määrittämistä. Normalisoitua kaikutiheyttä käytetään jälkikaiunnan alkamisajan löytämiseen mittauksista ja simuloidun jälkikaiunnan alkamisajan asettamiseen. Jälkikaiunta-aikojen määrittämisessä hyödynnetään energy decay relief -metodia. Kuuntelukokeiden perusteella automaattinen parametrisaatiojärjestelmä tuottaa parempia tuloksia kuin parametrien asettaminen manuaalisesti huoneen summittaisten geometriatietojen pohjalta. Järjestelmässä on ongelmia erityisesti jälkikaiunnan ekvalisoinnissa, mutta käytettyihin suhteellisen yksinkertaisiin tekniikoihin nähden järjestelmä toimii hyvin.Modern auralization techniques enable making the headphone listening experience similar to the experience of listening with loudspeakers, which is the reproduction method most content is made to be listened with. Room acoustic modeling is an essential part of a plausible auralization system. Specifying the parameters for room modeling requires expertise and time. In this thesis, a system is developed for automatic analysis of the parameters from room acoustic measurements. The parameterization is based on room impulse responses measured with a microphone array and can be divided into two parts: the analysis of the direct sound and early reflections, and the analysis of the late reverberation. The direct sounds are separated from the impulse responses using various signal processing techniques and used in the matching pursuit algorithm to find the reflections in the impulse responses. The sound sources and their reflection images are localized using time difference of arrival -based localization and frequency-dependent propagation path effects are estimated for use in an image source model. The late reverberation of the auralization is implemented using a feedback delay network. Its parameterization requires the analysis of the frequency-dependent reverberation time and frequency response of the late reverberation. Normalized echo density is used to determine the beginning of the late reverberation in the measurements and to set the starting point of the modeled late field. The reverberation times are analyzed using the energy decay relief. A formal listening test shows that the automatic parameterization system outperforms parameters set manually based on approximate geometrical data. Problems remain especially in the precision of the late reverberation equalization but the system works well considering the relative simplicity of the processing methods used

Effects of errorless learning on the acquisition of velopharyngeal movement control

Session 1pSC - Speech Communication: Cross-Linguistic Studies of Speech Sound Learning of the Languages of Hong Kong (Poster Session)The implicit motor learning literature suggests a benefit for learning if errors are minimized during practice. This study investigated whether the same principle holds for learning velopharyngeal movement control. Normal speaking participants learned to produce hypernasal speech in either an errorless learning condition (in which the possibility for errors was limited) or an errorful learning condition (in which the possibility for errors was not limited). Nasality level of the participants’ speech was measured by nasometer and reflected by nasalance scores (in %). Errorless learners practiced producing hypernasal speech with a threshold nasalance score of 10% at the beginning, which gradually increased to a threshold of 50% at the end. The same set of threshold targets were presented to errorful learners but in a reversed order. Errors were defined by the proportion of speech with a nasalance score below the threshold. The results showed that, relative to errorful learners, errorless learners displayed fewer errors (50.7% vs. 17.7%) and a higher mean nasalance score (31.3% vs. 46.7%) during the acquisition phase. Furthermore, errorless learners outperformed errorful learners in both retention and novel transfer tests. Acknowledgment: Supported by The University of Hong Kong Strategic Research Theme for Sciences of Learning © 2012 Acoustical Society of Americapublished_or_final_versio

Focusing high-power electromagnetic waves using time-reversal

A main aspect of this work has been to develop analytical and statistical models of the power efficiency of a time-reversal amplification system (TRAS).It is also important to evaluate the efficiency of a reverberation chamber. This allows quantifying the power received by one or more antenna when the reverberation chamber is excited. This factor is important when considering construction of the most efficient chamber for time-reversal amplification.Measurements assessing the loading effect of antennas in reverberation chambers when the field can be considered diffused were also undertaken. The study focuses on the evaluation of the varying quality factor when adding loaded antennas in the chamber.Another focus of this work is to evaluate the ratios between signals during calibration and focusing phase. An important aspect of the studies presented in this work thus concerns evaluation of the maximum value of the impulse response in a complex propagation system.We also present the power gain of time-reversal techniques and its statistical advantages compared to a classic use of a reverberation chamber.The development of a prototype required the design and implementation of each of the branches of the complete systems.The first measurement campaigns allowed the complete validation of the models.L'objectif de la thèse a été de mettre en place dans un premier temps des modèles analytiques et statistiques permettant d'évaluer les performances d'un système à retournement temporel de fortes-puissances puis de les vérifier grâce à des mesures.Des campagnes de mesures ont alors permis de vérifier les modèles. Des simulations numériques ont aussi montrées les possibilités offertes par un tel système.En parallèle, des travaux sur l'impact des antennes dans une chambre réverbérantes ont été menés afin d'évaluer les performances d'un système ayant plusieurs sorties.Les résultats de thèses ont permis l'élaboration de nouvelles métriques des performances du système.Le développement d'un prototype a nécessité la conception et la réalisation de chacune des branches du système complet.Les premières campagnes de mesures ont permis la validation complète des modèles

Simulation of Acoustic Wall Reflections Using the Finite-Difference Time-Domain Method

Tässä diplomityössä tutkitaan kerrosteisten seinärakenteiden akustisia ominaisuuksia simuloimalla ääniaallon etenemistä SRL FDTD -menetelmällä. Tutkittavat rakenteet koostuvat säleisestä paneelista ja takaseinästä, joiden väliin muodostuu ilmatila. Ulkoisesti samankaltaiset rakenteet tunnetaan alan kirjallisuudessa yleisesti resonoivina vaimentimina. Tässä opinnäytetyössä tutkitut rakenteet ovat kuitenkin mitoiltaan eri luokassa, eivätkä sisällä vaimentavaa materiaalia. Rakenteita löytyy esim. Helsingin Musiikkitalon konserttisalista. Aiheen kannalta olennainen taustatieto, liittyen ääniaallon ominaisuuksiin, mittausmenetelmiin ja mallinnusmenetelmiin, käsitellään lyhyesti. SRL FDTD -simulaatiomenetelmän perusteet esitellään 3-D menetelmän muodossa. Alustavat simulaatiot toimivat johdantona 2-D ja 3-D menetelmien ominaisuuksiin ja lisäksi käsitellään simulaatioiden toteutukseen liittyviä asioita. 2-D ja 3-D menetelmiä vertaileva tapaus osoittaa, että saatavat tulokset ovat laadullisesti vastaavia, kun tutkittavassa rakenteessa on yksi symmetria-akseli ja toteutukset vastaavat toisiaan. Näin 2-D menetelmä osoittautuu sopivammaksi tätä opinnäytetyötä varten. Tutkimus toteutettiin taajuus- ja aika-tason analyysillä, diffuusiomittauksilla ja visualisaatioilla. Rakenteiden todettiin aiheuttavan kampasuodatusta sekä taajuudesta riippuvaa äänienergian paikallista ja ajallista leviämistä.In this thesis, the reflection characteristics of layered wall structures are studied using the standard rectilinear (SRL) finite-difference time-domain (FDTD) method for modeling sound wave propagation. The structures consist of a panel with slats combined with a back wall, forming a cavity in between. Outwardly similar structures are known to have resonant properties and are generally known as resonant absorbers or distributed Helmholtz resonators. The structures studied in this thesis, however, are in a different parameter range and do not include absorptive material. They are found in practical use in, e.g., the Helsinki Music Centre Concert Hall. The various features of the structure are varied to establish how the reflection responses change with the features. The relevant background in acoustics - including wave phenomena, measurement methods and acoustic modeling methods - is briefly reviewed. The basics of the SRL FDTD method is introduced in the context of the 3-D method. Introductory simulations are done to explore the nature of the 2-D and 3-D methods, and issues regarding the simulation setup are addressed. A comparison case of the 2-D and 3-D methods shows that the results can be qualitatively equivalent for structures having one axis of symmetry. The 2-D method is deemed a better suited choice for the work in this thesis. The investigation was conducted through frequency- and time-domain analysis, diffusion measurements and visualizations. The studied structures were found to exhibit various degrees of comb filtering effects and frequency-dependent spatial and temporal spreading

The acoustics of concentric sources and receivers – human voice and hearing applications

One of the most common ways in which we experience environments acoustically is by listening to the reflections of our own voice in a space. By listening to our own voice we adjust its characteristics to suit the task and audience. This is of particular importance in critical voice tasks such as actors or singers on a stage with no additional electroacoustic or other amplification (e.g. in ear monitors, loudspeakers, etc.). Despite the usualness of this situation, there are very few acoustic measurements aimed to quantify it and even fewer that address the problem of having a source and receiver that are very closely located. The aim of this thesis is to introduce new measurement transducers and methods that quantify correctly this situation. This is achieved by analysing the characteristics of the human as a source, a receiver and their interaction in close proximity when placed in acoustical environments. The characteristics of the human voice and human ear are analysed in this thesis in a similar manner as a loudspeaker or microphone would be analysed. This provides the basis for further analysis by making them analogous to measurement transducers. These results are then used to explore the consequences of having a source and receiver very closely located using acoustic room simulation. Different techniques for processing data using directional transducers in real rooms are introduced. The majority of the data used in this thesis was obtained in rooms used for performance. The final chapters of this thesis include details of the design and construction of a concentric directional transducer, where an array of microphones and loudspeakers occupy the same structure. Finally, sample measurements with this transducer are presented

Noise and Vibration Control in the Built Environment

With global urbanization rapidly rising and the increasing need of high-quality built environments, the requirement of achieving improved sound quality, both for outdoor and indoor environments, has received a lot of attention. This Special Issue collection reflects the current state of the art, with 12 papers covering environmental acoustics; the influence of soundscapes on people’s behavior; soundscape pleasantness estimation; tranquility; perceived quality of sonic environments; sound and vibration-related health complaints concerning tramways; and the disturbance of construction machines; as well as in building and room acoustics, including natural ventilation-enabling façade noise control devices; the effect of external shading devices; rating method of airborne sound insulation; water supply and drainage noise, and the effect of diffusive surfaces in auditoria. Some papers cover research on the engineering aspects of sound and vibration, such as sound propagation and noise control techniques, as well as perception aspects of sound, such as indoor acoustic comfort and environmental soundscapes. The book is prefaced by Prof. S Wu, entitled ‘Sustainable Urban Sound Environment’