scispace - formally typeset
Search or ask a question

Showing papers in "Acta Acustica United With Acustica in 2007"


Journal Article
TL;DR: In this paper, the authors present the Harmonoise source model for road vehicles and discuss some aspects of it taking into account some new investigations carried out in the Nordic countries, where the sound power levels of tyre/road noise and propulsion noise are given as equations as a function of frequency, speed and vehicle category.
Abstract: In the European Harmonoise project aiming at calculating the yearly day/evening/night weighted sound pressure levels from traffic noise the source model is completely separated from the propagation model. This paper presents the source model for road vehicles and discusses some aspects of it taking into account some new investigations carried out in the Nordic countries. The sound power levels of tyre/road noise and propulsion noise are given as equations as a function of frequency, speed and vehicle category. There are 5 different main vehicle categories each of which is subdivided into several sub categories. The sound power level is then distributed between point sources on different heights, each having a given vertical and horizontal directivity. These data refer to a reference condition defined by a constant speed, a specified road surface and a specified temperature. For conditions different from the reference conditions, corrections are given for air temperature, road surface, acceleration/deceleration and road surface wetness. In addition further corrections are possible, such as regional corrections to take into account systematic deviations from the conditions on which the reference equations are based. It is shown that the Harmonoise source model works quite well assuming that some regional corrections are taken into account.

71 citations


Journal Article
TL;DR: In this paper, the Harmonoise reference model has been developed in order to predict long-term average sound levels in road and railway situations that are geometrically relatively simple but physically complex.
Abstract: The Harmonoise reference model has been developed in order to predict long-term average sound levels in road and railway situations that are geometrically relatively simple but physically complex. The present paper describes all steps of calculations with this powerful model which includes several advanced numerical propagation methods to calculate the coupled effects of atmosphere, ground surface and obstacles on sound waves. The reference model employs a statistical description of the atmosphere, based on local meteorological data, as well as various impedance models for ground surfaces and other absorbing surfaces. Validations against in situ longterm measurements have been achieved in several sites; agreement between reference model and experimental results ranges from excellent in flat terrain situations down to fairly good in more complex configurations (hilly, viaduct).

56 citations


Journal Article
TL;DR: In this article, a high-speed, digital imaging of the superior and medial surfaces of the vocal fold was performed using an excised human hemilarynx setup, where surface dynamics were characterized and differentiated across a variety of phonatory conditions.
Abstract: The objective of this research was to investigate mucosal wave propagation in laboratory experiments, across a variety of phonatory conditions. In particular, the focus was on the medial and superior surface dynamics of the vocal fold, which quantify mucosal wave propagation, but have been relatively little studied. High-speed, digital imaging of the superior and medial surfaces of the vocal fold was performed using an excised human hemilarynx setup. Surface dynamics were characterized and differentiated across a variety of phonatory conditions. During sustained, flow-induced oscillation, the local maxima of vocal fold mucosal displacements, velocities and acceleration and their particular phase delays in the glottal cycle were investigated. Statistical analysis was performed, examining the influence of induced flow, adductory stimulation, and length of the vocal fold. To give an overview, the grand average values were computed and discussed for the complete series of 24 recordings. Increasing the applied airflow, yielded higher values for lateral displacements as well as higher velocity and acceleration values. Elongating the vocal fold resulted in decreased lateral displacements. The mucosal wave propagation apparently increased for higher flow, elongated folds, and higher adduction forces. Performing grand averages revealed the three-dimensional dynamical behavior over the superior and medial surface of the vocal fold. A significant increase in the amplitudes of the dynamical quantities between inferior and superior regions was detected. The data showed a nearly 180 degree phase delay between inferior and superior regions of the vocal fold with respect to lateral displacements, velocities, and accelerations. Phase delays for the vertical displacements were also present, but less pronounced. Using a hemilarynx methodology, mucosal wave propagation was characterized and differentiated over the superior and medial surfaces of the vocal fold surface across a range of phonatory conditions. While an understanding of the correlation between vocal fold dynamics and phonatory physiology/pathology is still in its infancy, the data presented here help to establish such connections. The data are also useful for the development and evaluation of physical and numerical models of vocal fold vibration. However, since only one larynx has been investigated the results have to be seen as preliminary.

51 citations


Journal Article
TL;DR: The Harmonoise project as discussed by the authors developed two propagation models: a reference model based on numerical simulations and an analytical and heuristic solution based on heuristic solutions, which were validated with respect to typical situations occurring in the mapping of road and railway noise.
Abstract: The Harmonoise project started in august 2001 and ran for three years. The main objective of the project was to provide new predictions methods for the mapping of environmental noise as required by the European Noise Directive COM 2000/49. Aside new predictions schemes for noise emission from road and railway vehicles, the project developed two propagation models: a "reference" model based on numerical simulations and an "engineering" model based on analytical and heuristic solutions. In this paper, the development and validation of the analytical propagation model will be presented. Various analytical and heuristic models have been proposed in the past; the Nord2000 project being the most recent attempt to provide a comprehensive prediction scheme combining all important propagation effects due to ground reflections, diffraction and meteorological refraction. All along its development, the Harmonoise project has taken advantage of the work carried out under the Nord2000 project. Similarities and differences between both models will be pointed out throughout this paper. Within the Harmonoise project, the prediction scheme was build and validated with respect to typical situations occurring in the mapping of road and railway noise; specifically, the project focused on sources close to the ground, propagation distances up to 1 km and under moderate atmospheric refraction conditions. Extensions of the propagation model to higher sources and larger distances are studied in a follow-up project, named IMAGINE, with application to aircraft and industrial noise sources.

46 citations


Journal Article
TL;DR: In this article, a two-part study of the quality of car horn sounds is presented, focusing on the perception of the timbre of the sounds, grounded in a psychoacoustical framework.
Abstract: This paper is the first of a two-part study of the quality of car horn sounds. It aims to provide insights into the design of new sounds: how do we create new sounds that still warn road users against a danger related to a car? In this first part, we study the perception of the timbre of the sounds, grounded in a psychoacoustical framework. We begin to review the approaches to sound quality and choose a framework for our study. Then, after the perceptual validation of the choice of the set of sounds, we ask listeners to categorize all the recorded sounds into main families. Then we model the perception of the timbre of these sounds as three elementary sensations, each correlated with an acoustical descriptor. The results of these experiments suggest that listeners used shared perceptual dimensions along which the sounds were rated. These different dimensions, and the correlated acoustical descriptors rely on the differences between the slight variations of car horn mechanisms. They form the basis for the following study of the invariants that let a sound convey the warning message.

45 citations


Journal Article
TL;DR: In this article, the acoustic performance and mechanical properties of different types of resilient underlayer, used in floating floors, are investigated on the basis of apparent dynamic stiffness measurement, compressibility and compressive creep and of laboratory measurement of the reduction of transmitted impact noise.
Abstract: Acoustical performance and mechanical properties of different types of resilient underlayer, used in floating floors, are investigated on the basis of apparent dynamic stiffness measurement, compressibility and compressive creep and of laboratory measurement of the reduction of transmitted impact noise. Determining and quantifying resilient underlayers properties with time, on the basis of evaluation of thickness under load and behaviour under compression, is the aim of this study. The measurement and analysis methodologies of the macroscopical mechanical behaviour, although in first approximation, allow useful knowledge of acoustical performances of resilient materials used under floating floors.

44 citations


Journal Article
TL;DR: The difference in perceived noise annoyance caused by train and highway noise at the same averaged noise level, has led to the introduction of the "railway bonus" as mentioned in this paper, which has found its way to t
Abstract: The difference in perceived noise annoyance caused by train and highway noise at the same averaged noise level, has led to the introduction of the ’railway bonus’. This bonus has found its way to t ...

43 citations


Journal Article
TL;DR: In this article, the authors show how the basic parameters of an open office can affect the speech privacy and speech level between two neighboring workstations, and demonstrate the importance of masking sound as a basic precondition for good speech privacy if normal voice levels are used in the office.
Abstract: The aim of this study was to show how the basic parameters of an open office affect the speech privacy and speech level between two neighboring workstations. The investigation was carried out in laboratory conditions where two adjacent workstations were located. The parameters studied were screen height, room height, ceiling absorption, floor absorption, screen absorption, and masking sound level. Altogether 50 different combinations were studied at three different masking sound levels and normal voice levels. The horizontal sound field was damped by wall absorbers so that the arrangement resembled a pair of workstations in the middle of a large room. The speech privacy improved with increasing masking sound level, ceiling absorption, screen height, and room height, in the order of partial significance. It was possible to reduce speech levels at most by approximately 15 dBA by using the best combination of ceiling absorber and screen height, compared to the situation without absorbers and screens. However, if the masking sound level was low, below 40 dBA, sufficient speech privacy, i.e. low speech intelligibility, could not be reached. The study emphasizes the importance of masking sound as a basic precondition for good speech privacy if normal voice levels are used in the office. The study also gives preliminary evidence that insufficient attenuation of the horizontal sound field in an open office can seriously undermine the attenuation gained from ceiling absorbers and screens. The acoustical design requires simultaneous solutions for masking sound, and the absorption of horizontal and vertical sound fields.

41 citations


Journal Article
TL;DR: The CUEX algorithm as mentioned in this paper uses a combination of traditional methods to segment the audio stream into tones based on fundamental frequency contour and sound level envelope, from the resulting onset and offset positions, the different tone parameters are computed.
Abstract: CUEX is an algorithm that from recordings of solo music performances extracts the tone parameters for tempo, sound level, articulation, onset velocity, spectrum, vibrato rate, and vibrato extent. The aim is to capture the expressive variations in a music performance, rather than to identify the musical notes played. The CUEX algorithm uses a combination of traditional methods to segment the audio stream into tones based on fundamental frequency contour and sound level envelope. From the resulting onset and offset positions, the different tone parameters are computed. CUEX has been evaluated using both synthesized performances and recordings of human performances. For the synthesized performances, tone recognition of 98.7% was obtained in average. The onset and offset precision was 8 ms and 20 ms, respectively, and the sound level precision about 1 dB. Various applications of the CUEX algorithm are discussed. For human performances, the recognition was 91.8% in average.

40 citations


Journal Article
TL;DR: A case study is presented, consisting of a large set of microscopic traffic simulations and associated noise emission calculations, which provides some insight into the specific dynamics of the noise emission near different types of intersections.
Abstract: Urban noise mapping traditionally involves the use of a traffic simulation model, which is often based on the estimation of macroscopic traffic flows. However, intersections and other local traffic management measures are not always modeled correctly. It is well known that the specific deceleration and acceleration dynamics of traffic at junctions can influence local noise emission. Finding the best strategy for using traffic modeling results in noise mapping is a current topic of research in the IMAGINE project. In this paper, a case study is presented, consisting of a large set of microscopic traffic simulations and associated noise emission calculations, which provides some insight into the specific dynamics of the noise emission near different types of intersections. It will be shown that it is possible to refine current traffic noise prediction models, based on macroscopic traffic simulation, using a correction on the average vehicle emission, aggregated in lane segments. A spatial approach should be used, in which inbound and outbound lanes are divided into deceleration, queuing, stopline and acceleration zones. Results from regression analysis on the numerical simulations indicate that meaningful relations between noise corrections and traffic flow parameters such as traffic intensity and composition can be deduced.

37 citations


Journal Article
TL;DR: In this article, the uncertainty of the single number rating from third-octave band sound insulations and associated uncertainties was investigated by analyzing a large amount of round robin results.
Abstract: Single-number quantities are widely used to characterise acoustic properties of building products and buildings. The uncertainty of these values is now investigated by analysing a large amount of round robin results. Methods are developed to calculate the uncertainty of the single number rating from third-octave band sound insulations and the associated uncertainties. The Monte-Carlo method is used to verify the calculation methods. It turns out that correlation effects between third-octave bands significantly influence the uncertainty of the single number rating. Whereas an upper limit for the uncertainty can be calculated by assuming a full, positive correlation between all third-octave bands, the uncorrelated case does not provide a lower limit due to the occurrence of negative correlation. Since correlation effects cannot be predicted it is recommended to use the averaged uncertainties as derived from round robin tests until a deeper understanding of the governing mechanisms enables a calculation of the uncertainties.

Journal Article
TL;DR: In this article, a reception plate method is proposed which yields the source activity in the form of the sum of the squared free velocities, over the contact points, which can be used to estimate the installed power for the range of receiver mobilities likely to be encountered in buildings.
Abstract: This paper considers a practical structure-borne sound source characterisation for mechanical installations in buildings. Such machines nearly always are installed in contact with heavyweight homogeneous structural floors and walls, or floating floor systems, or stiffened cavity constructions. Manufacturers require a laboratory-based measurement procedure, which will yield single values of source strength in a form transferable to a prediction of the sound power generated in the installed condition, and thence the sound pressure in rooms removed from the source. A novel reception plate method is proposed which yields the source activity in the form of the sum of the squared free velocities, over the contact points. In addition, the source mobility is obtained separately as the average of the magnitude of the effective mobility, over the contact points. Both quantities can be used to estimate the installed power for the range of receiver mobilities likely to be encountered in buildings. It is demonstrated that the installed power can be estimated by reference to a high source mobility condition, a low source mobility condition, or to a matched mobility condition.

Journal Article
TL;DR: In this article, a diffusion-based approach is proposed to predict the sound-level distribution, as well as the sound decay, in a room with specularly-reflecting surfaces.
Abstract: This study focuses on the modeling of sound fields in fitted rooms - i.e. rooms containing many obstacles which scatter sound. A novel approach is proposed to predict the sound-level distribution, as well as the sound decay, in such rooms. The model is derived by coupling two approaches based on a diffusion analogy. The first was initially derived for modeling the scattering of sound by obstacles in a room with specularly-reflecting surfaces. The second was proposed for modeling the reverberant field in empty rooms with diffuse surface reflections. The approach is first applied to a cubic room and to a flat enclosure, and compared to a ray-tracing model (the Rayscat model), with very good agreement. A comparison with experimental data obtained in a real fitted room is also presented, and gives very good agreement, both with the diffusion model and with the Rayscat model. In order to include mixed specular-diffuse reflections in the diffusion model, an empirical approach is also presented. The main advantage of such a technique is that it gives accurate predictions with much lower calculation times than Monte-Carlo-based numerical techniques such as ray-tracing.

Journal Article
TL;DR: Examination of loudness asymmetry between increasing and decreasing levels for 1-kHz tones over the range 60-80 dB SPL and over four ramp durations using two additional loudness ratings using continuous ratings and global ratings reveals an asymmetry depending on the direction of change, the range of levels, and on the loudness rating method involved.
Abstract: Loudness change has been studied for tones with linearly varying levels using different loudness rating methods, such as direct estimation or indirect estimation based on the starting and ending levels. The published results reveal an asymmetry depending on the direction of change (increasing vs. decreasing), the range of levels (high vs. low), and on the loudness rating method involved. The present study examines loudness asymmetry between increasing and decreasing levels for 1-kHz tones over the range 60-80 dB SPL and over four ramp durations (2, 5, 10 and 20 s) using two additional loudness ratings: continuous ratings and global ratings. A continuous analogical/categorical (A/C) rating scale was used, which consisted of an analog scale subdivided into seven discrete categories labeled from very, very loud to very, very soft. Two measures are obtained, examined and analyzed separately: indirect and direct loudness measures that correspond to the loudness change extracted from continuous ratings and the overall loudness impression, respectively. Loudness changes do not reveal any significant perceptual asymmetry between an increasing and a decreasing ramp. In addition, results do not reveal any "decruitment" effect, i.e. the loudness of a continuously decreasing tone changes more rapidly as a function of sound pressure level, which is in agreement with previous results for this range of levels. On the other hand, direct estimation of the global loudness, i.e. an overall loudness rating of the stimulus, is higher for an increasing ramp than for a decreasing ramp. This result is in agreement with previous studies and can be described by a memory process dominated by the ending level.

Journal Article
TL;DR: In this paper, a comparison between the prediction techniques commented above in the context of sound diffusers, paying special attention to the Finite Difference Time Domain method (FDTD) is presented.
Abstract: Since the invention of sound diffusers three decades ago a substantial effort has been made to predict the acoustic behaviour of these structures, for auralisation and prediction purposes as well as in response to the large costs inherent in anechoic measurements. Volumetric methods such as Finite Element Methods (FEM) or the Finite Difference Time Domain method (FDTD) are not often used, due to their large computational cost. However Near Field to Far Field Transformations (NFFFT) can overcome that problem. The main advantages of the FDTD method are that a single calculation is sufficient to study a wide frequency band, and that the time domain behaviour of the reflected sound can be directly inspected. In this paper we present a comparison between the prediction techniques commented above in the context of sound diffusers, paying special attention to the FDTD method. Having demonstrated that the FDTD method can generate results comparable to more established techniques, early results concerning the modelled performance of diffusers in the time domain (‘time spreading’) are reported, opening a new field of research.

Journal Article
TL;DR: In this article, the authors demonstrate the concept of time-reversal acoustic sink (TRAS) in audible frequency regime, in order to overcome the diffraction limit imposed by the TRM focusing.
Abstract: Time-reversal mirrors (TRM) have been developed since 1986 in order to focus ultrasonic transient waves in complex media. In the last few years, the properties of TR of acoustic fields have been studied in many different areas. Nevertheless, few applications of TR have been developed in audible range acoustics. The aim of this paper is to demonstrate the concept of time-reversal acoustic sink (TRAS) in audible frequency regime, in order to overcome the diffraction limit imposed by the TRM focusing. The major difference between the TRAS and TRM experiments in ultrasonics and audible range is the ratio between the wavelength and the size of the transducers and objects on which the focusing is achieved. The audible range experiment are lead in Fresnel field (near field), whereas the ultrasonic experiments are lead in Fraunhoffer field (far field). We present the first experimental results with a TRAS in this frequency range. The focusing behaviour in a reverberation room using different transient sounds and frequency domains are investigated and discussed, showing that one can take advantage of reverberation in order to achieve subwavelength sound focusing using a single-element TRM. We report that a focal spot of a seventh of a wavelength has been recorded using the TRAS techniques in audible range, compared to the half wavelength obtained with normal TRM processing. A promising application of a numerical TRAS-method in acoustic imaging and localization of acoustic and vibrational sources is presented.

Journal Article
TL;DR: In this paper, a non-linear lumped model of the reed-mouthpiece-lip system of a clarinet is formulated, in which the lumped parameters are derived from numerical experiments with a finite-difference simulation based on a distributed reed model.
Abstract: A non-linear lumped model of the reed-mouthpiece-lip system of a clarinet is formulated, in which the lumped parameters are derived from numerical experiments with a finite-difference simulation based on a distributed reed model. The effective stiffness per unit area is formulated as a function of the pressure signal driving the reed, in order to simulate the effects of the reed bending against the lay, and mass and damping terms are added as a first approximation to the dynamic behaviour of the reed. A discrete-time formulation is presented, and its response is compared to that of the distributed model. In addition, the lumped model is applied in the simulation of clarinet tones, enabling the analysis of the effects of using a pressure-dependent stiffness per unit area on sustained oscillations. The analysed effects and features are in qualitative agreement with players' experiences and experimental results obtained in prior studies.

Journal Article
TL;DR: How the open phase of the glottal cycle may affect the acoustic response of the vocal tract is investigated and has implications for realistic modelling of modal voice, especially where there is a permanentglottal leak, and breathy voice including pathological cases.
Abstract: In the classical theory of vowel production it is standard to assume linear separability of the voicing source, located at the glottis, from the vocal tract downstream. In this paper we consider an effect of relaxing this assumption and investigate how the open phase of the glottal cycle may affect the acoustic response of the vocal tract. A mechanical model of the larynx and vocal tract is used to make measurements of the formant frequencies with both a static glottis and a time-varying glottal area. For the static glottis the vocal tract was excited by an external sound source. The first and second formant frequencies increased with increasing glottal width. For the widest glottis investigated the upward shift in the first formant was 13% of the value found with a closed glottis. The direction of the shift is well-modelled by a theoretical transmission line model of the vocal tract for which the value of the glottal impedance can be varied, although the F1 values are underestimated by approximately 5%. For the time-varying glottis the acoustic excitation came from the periodic interruption of a steady air-flow. The first formant frequency increased with both increasing glottal width and increasing glottal open quotient. Our findings have implications for realistic modelling of modal voice, especially where there is a permanent glottal leak, and breathy voice including pathological cases.

Journal Article
TL;DR: In this paper, the effect of the source orientation on the measured values of acoustic parameters and both the experimental standard deviation (STD exp ) and the subjective just noticeable difference (jnd) of the corresponding parameters are analyzed by means of a set of measurements carried out with two "omni-directional" sources according to ISO 3382 specifications.
Abstract: The acoustic assessment of a receiver position in a room is evaluated from the measurement of different acoustic parameters derived from the room impulse response (RIR). The definition of such parameters as well as their measurement procedures are described in the ISO 3382 standard. With regard to the sound source, the loudspeaker should be as omni-directional as possible. Most of the commercial dodecahedron loudspeakers comply with the maximum allowed directional deviations of the source specified in the standard. While this requirement is adequate for the derivation of reverberation times, for the detailed investigation of the RIR's time structure, however, the directivity of the source becomes more important. Presently it is undetermined whether changes in the sources orientation alter the results of the acoustic parameters to a greater extent than the subjective just noticeable difference (jnd) of the corresponding parameter. In this work, the effect of the source orientation is analysed by the means of a set of measurements carried out with two "omni-directional" sources according to ISO 3382 specifications. The comparison of newly obtained deviations of the measured values of acoustic parameters and both the experimental standard deviation (STD exp ) - characterizing the dispersion under repeatability conditions - and the jnds of the corresponding parameters shows that directionality of the source has a significant influence at high frequencies. Several parameters, such as C 50 , C 80 , G, and IACC E , show deviations caused by the source orientation that are larger than the respective jnds, while the overall uncertainty of the measurement is insignificant. For acoustic parameters that are calculated with longer integration times, e.g. T 30 or EDT, the deviations are lower. The results obtained with the different sources vary on a coincidental basis. The evaluation of the acoustic quality on the basis of the different measurements may yield to vague results even if the same source-receiver position is characterized.

Journal Article
TL;DR: In this paper, the authors proposed a new maximum likelihood approach that allows for the multiple decay rates found in real spaces to be more correctly modelled, and thus improves the accuracy of the estimations.
Abstract: Reverberation time is an important objective parameter for the acoustics of enclosed spaces. Blind estimation of the reverberation time from naturally occurring sound sources alleviates the logistical constraints of traditional measurement methods, and hence enables non-invasive, in-situ measurements in occupied spaces. A maximum likelihood method using a simple exponential decay model was recently developed. It demonstrated the potential of this approach, but the decay model hinders its accuracy and usefulness in the presence of more complex decays. This paper proposes a new maximum likelihood approach that allows for the multiple decay rates found in real spaces to be more correctly modelled, and thus improves the accuracy of the estimations. Envelopes of received sound signals are segmented and suitable decay phases automatically selected. A maximum likelihood estimation is performed using a multi-decay model whose function adapts to yield the most likely reverberation parameters and decay curve estimates. Simulation, validation and real room tests confirm the improved accuracy; the estimation errors are typically below perceptual difference limens. Thus a new method for blind estimation of reverberation parameters from naturally occurring sound sources, including speech and orchestra music, is achieved.

Journal Article
TL;DR: In this article, auralized speech-intelligibility test conditions were modeled using the CATT-Acoustics prediction and auralization system, and the results in the real and virtual classrooms were compared.
Abstract: Speech-intelligibility tests with human subjects give more realistic results than do the measurement or the prediction of objective metrics. However, human testing in real environments has a number of limitations. Auralization offers a solution to the limitations, having unlimited capability to reproduce realistic listening environments. The work presented here aimed to validate the auralization technique for speech-intelligibility testing, in comparison with live listening tests in real classrooms. Results for two real and virtual classrooms were compared. Acoustical parameters were measured in the real classrooms and speech-intelligibility tests performed. The Modified Rhyme Test and speech-babble noise were generated by loudspeakers. Speech-intelligibility tests were performed at three positions with normal-hearing subjects. The classrooms and the speech-intelligibility test conditions were modeled using the CATT-Acoustics prediction and auralization system. Predicted and measured room-acoustical parameters were compared. Speech-intelligibility tests were performed in the virtual classrooms with the same subjects, and the results in the real and virtual classrooms were compared. Auralized speech-intelligibility tests were found to be reliable if the classroom was not very absorptive or noisy.

Journal Article
TL;DR: In this article, an improved numerical method for deducing the phase factors necessary for the synthesis part of the analysis-synthesis system was proposed. But this method gave more accurate results compared to the original proposal in some cases.
Abstract: The article "Frequency analysis and synthesis using a gammatone filterbank" [1] introduces an efficient implementation of all-pole gammatone filters with complex-valued output. It describes the design of an auditory filterbank using these gammatone filters, together with a low-delay synthesis method. This article introduces an improved numerical method for deducing the phase factors necessary for the synthesis part of the analysis-synthesis system. This method gives more accurate results compared to the original proposal in some cases. Also introduced is a numerical optimization procedure to deduce band-specific weight factors needed for the synthesis stage. Finally, a method is introduced to reduce the computational cost of the already efficient gammatone filter implementation by approximately 60%, at the cost of storing additional tables of precomputed parameters.

Journal Article
TL;DR: In this paper, the authors investigated the effectiveness of a window system using finite element method based software FEMLAB and found that the difference between external and internal sound pressure levels (SPL) is typically around 20 dB without absorber within the window in terms of the average value at 125 Hz-1 kHz, whereas with absorbers, including MPA, the effectiveness can be much greater.
Abstract: The objective of this research is to develop a window system which allows natural ventilation while reducing noise transmission. The core idea is to create a ventilation path by staggering two layers of glass. Transparent micro-perforated absorbers (MPA) can be used along the path created to reduce noise. This paper investigates the effectiveness of this window system through using finite element method based software FEMLAB. Starting with a study of the feasibility of using the numerical method for the window system, the paper then presents a parametric study. It has been shown that with such a window system the difference between external and internal sound pressure levels (SPL) is typically around 20 dB without absorber within the window in terms of the average value at 125 Hz-1 kHz, whereas with absorbers, including MPA, the effectiveness can be much greater. With hard boundaries, the external and internal opening sizes and the air gap width affect the acoustic performance considerably at individual frequencies, whereas in terms of the average performance over the frequency range considered the difference is rather small, within about 2dB. Placing acoustically hard louvers within the two sheets of glass does not bring much extra acoustic benefit. but with absorbent surfaces their performance can be significantly improved. A hood hung outside the opening is very effective in increasing the SPL difference and the effectiveness is improved by increased hood length. The ventilation simulation suggests that the studied windows, including the configurations with the external hood and louvers, will provide sufficient background ventilation as well as rapid ventilation for comfort.

Journal Article
TL;DR: In this paper, the authors describe the determination of vertical profiles of the effective sound speed from meteorological measurements at only one height above ground, which are needed for the prediction of long-term sound levels under the consideration of atmospheric refraction.
Abstract: The paper describes the determination of vertical profiles of the effective sound speed from meteorological measurements at only one height above ground. The vertical profiles are needed for the prediction of long-term sound levels under the consideration of atmospheric refraction. The study is based on the results of field campaigns which were performed during the European project HARMONOISE. In a first step different theoretical-empirical approaches were tested to derive vertical effective sound speed profiles from measured turbulent fluxes of momentum and heat. The results of the profile generation were compared with sound speed profiles derived from profile measurements at a meteorological tower. It was found that a two-parameter logarithmic-linear approach is sufficiently accurate to generate vertical sound speed profiles. In a second step profiles generated on the basis of local flux measurements were compared with those based on conventional data of a nearby routine weather station. If local measurements are not available the representativeness of nearby weather stations has to be carefully checked, above all in topographically structured terrain.

Journal Article
TL;DR: For an isotropic elastic solid, all the rigidity coefficients can be evaluated from the velocities of the shear wave and of the compressional wave as mentioned in this paper, and the second parameter is the velocity of the Rayleigh waves slightly modified by the finite thickness of the samples.
Abstract: For an isotropic elastic solid, all the rigidity coefficients can be evaluated from the velocities of the shear wave and of the compressional wave. For an isotropic porous frame, the rigidity coefficients of the frame in vacuum can be evaluated from the velocities of the frame compressional Biot wave and the shear Biot wave when the porous frame is saturated by air. The velocity of the frame compressional Biot wave is evaluated at audible frequencies from the detection of the quarter compressional wavelength resonance generated with a point source in air for samples bonded on a rigid impervious backing. In a previous work, the second measured parameter was the velocity of the shear wave evaluated from the quarter shear wave resonance. In this work, the second parameter is the velocity of the Rayleigh waves slightly modified by the finite thickness of the samples, the observation of the shear resonances being difficult for the melamine foam.

Journal Article
TL;DR: In this article, a model for simulating a single impact of a ball on a damped plate has been adapted to simulate the sound of the ball rolling over a plate, and the model is validated by means of three different types of simulations.
Abstract: A model previously developed for simulating a single impact of a ball on a damped plate has been adapted to simulate the sound of a ball rolling over a plate. The original model has the advantage of being well tested and showing good agreement between measurements and time-domain simulations of various impacted plates [1]. The main changes for its adaptation to rolling sounds were made in the ball-plate contact. Instead of an impact point that is fixed in space and short in time the model now incorporates an interacting contact point that is continuously moving in space. To allow for the variable position of the contact point, we use a special spatial window that is optimized for this purpose. Furthermore, a model for the surface roughness of the plate was added. The model is validated by means of three different types of simulations. The numerical results are either compared with experiments or with analytical calculations. The first type of simulation is that of a ball that rolls over a surface with some random asperities. The main observation is that the ball looses contact with the plate at some speed. The second simulation is that of a sinusoidally time-varying source that moves over the plate. Here the characteristic Doppler effect is identified. The third set of simulations are of a ball that is dropped on a plate. The ball bounces back to some height that is lower than the original release height. This fraction of height, also called the restitution coefficient, was measured and compared with simulated data. Following the validation procedure, the model is used to simulate rolling objects. It is shown that different kinds of contact exist between ball and plate. Four different types of rolling with different plate/ball contact parameters are identified: amplitude modulations, periodic bouncing, chaotic bouncing and continuous contact. Comparisons are made between measured and simulated accelerations of a fixed point on a aluminum plate with a sinusoidal waviness profile, which is set into vibrations by rolling spheres of various sizes, stiffnesses and densities. © S. Hirzal Verlag.

Journal Article
TL;DR: In this paper, the interaction between the wall vibrations of a stretched elastic cylindrical membrane and the inner acoustic field is considered under plane wave approximation, and a model for the dispersion curves is presented and is experimentally validated.
Abstract: In this paper, the interaction between the wall vibrations of a stretched elastic cylindrical membrane and the inner acoustic field is considered under plane wave approximation. Three waves exist at low frequencies for this coupled system. The first of these, called Korteweg's wave, propagates mainly within the fluid and corresponds to the acoustic plane wave which is closely coupled to the wall vibrations. The two other waves mostly propagate within the structure and correspond to coupled longitudinal/flexural motions: one corresponds to predominant longitudinal motions in the membrane and the other exists only when tension is applied to the membrane and is similar to a string bending wave. A model for the dispersion curves is presented and is experimentally validated. In particular, the model and experiments reveal that three frequency ranges exist for which the propagation of the Korteweg's wave is subsonic, evanescent and supersonic. The experimental validation is achieved using the acoustic impedance measurements for a stretched rubber membrane. Assuming that the vibratory and acoustic fields are dominated by one wave, the latter are described by using only one dispersive wave, in this case, of equivalent wave speed. The input acoustic impedance curve can be fitted using this expression which only requires one equivalent wave.

Journal Article
TL;DR: In this paper, the authors presented a new 'in-vitro' set-up, which overcomes some of the limitations of the previous study by the use of a digital camera synchronised with a light source and of pressure sensors, which allows measurement of the area of the replica opening and imposition of independent initial conditions, such as height of the initial opening and internal pressure in the replica.
Abstract: Insight into vocal fold and lip oscillation mechanisms is important for the understanding of phonation and the sound generation process in brass musical instruments In general, a simplified analysis of the physical 3D fluid-structure interaction process between the living tissues and the airflow is favoured by most workers Several simple models (lumped parameter models) have been proposed and these represent the tissues as a distribution of elastic mass(es) The mass-spring-damper system is acted on by a driving force resulting from the pressure exerted by the airstream The results from these theoretical models have been validated 'in-vitro' using rigid or deformable replicas mounted in a suitable experimental set-up Previous research by the authors focused on the prediction of the pressure threshold and oscillation frequency of an 'in-vitro' replica, in the absence and presence of acoustical feedback In the theoretical model a lip or vocal fold is represented as a simple lumped mass system The model yielded accurate prediction of the oscillation threshold and frequency In this paper a new 'in-vitro' set-up is presented, which overcomes some of the limitations of the previous study By the use of a digital camera synchronised with a light source and of pressure sensors, this set-up allows 1) measurement of the area of the replica opening and 2) imposition of independent initial conditions, such as height of the initial opening and internal pressure in the replica The impact of these findings on physical modelling is discussed

Journal Article
TL;DR: In this paper, the applicability and adequacy of the two-microphone method for ground impedance measurement is investigated, and the proposed implementation in an ANSI standard ("Direct Deduction of Ground Impedance") will contribute to its wide use.
Abstract: The two-microphone method is a convenient and well-known procedure to measure the surface impedance in-situ. Its proposed implementation in an ANSI standard ("Direct Deduction of Ground Impedance") will contribute to it's wide use. The applicability and adequacy of this method for this purpose is investigated in this article. Measurements of the surface impedance of a number of grounds, which are carried out with the three ANSI S1.18 geometries, are presented. It is shown that the two-microphone method does provide reasonable results on not too hard grounds for frequencies above about 400 Hz, while the performance at lower frequencies and for hard grounds is poor. Practical advise is given on the number of measurements needed and the data pre-processing.

Journal Article
TL;DR: In this article, a phase-lock-in technique was used to measure the phase shift caused by increments in the amplitude of an acoustic excitation, and the measured phase shift as a function of dynamic strain amplitude is used to curve fit the non-classic nonlinear equation of state in order to extract the hysteretic nonlinearity parameter.
Abstract: This paper describes a phase shift method for studying the non-linear acoustic behaviour of a soil. The method uses a phase-lock-in technique to measure the phase shift caused by increments in the amplitude of an acoustic excitation. The measured phase shift as a function of dynamic strain amplitude is used to curve fit the non-classic non-linear equation of state in order to extract the hysteretic non-linearity parameter. It is found that the non-linearity parameter obtained by this method is in good agreement with that obtained from the frequency shift method. This study also highlights several non-linear acoustic characteristics of the soil, including resonance frequency shift, high harmonic generation, energy transfer, and phase shift.