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Sound and Noise around us
Marion Anne Burgess
[Abstract]
Sound is fundamental to our social structure from the basic need for communication through to the expression of our emotions in music. Sound at too high a level can cause damage to hearing and it is the role of society, workplace management and governments to ensure that the community is aware of the risk and that this risk is minimised by implementing noise control and noise management. Extensive research has shown that sound levels that commonly occur in our cities and rural areas and that may be below the damage risk level can create annoyance and lead to various health effects. Sometimes we may desire silence. But generally, there is a level of noise that we consider acceptable; when the noise from outside does not interfere with what we are doing at that time. So limits and control measures need to be implemented to keep community noise within the acceptable range while still allowing for the necessary business, industry, transportation and entertainment to continue. Engineering noise control at source is a solution for some problems. For others an acceptable environment may be achieved by coupling a deep understanding of the effects of noise with the application of some of many tools available using modern technology. Promotion of best practices in the wide community is one of the goals for the activities that are part of the ICA organised International Year of Sound in 2020.
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Session chair(s):
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M. Kob, Erich Thienhaus Institute, Detmold University of Music |
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Sounds of the human vocal instrument
Shrikanth Narayanan
[Abstract]
The vocal tract is the universal human instrument played with great dexterity to produce the elegant acoustic structuring of speech, song and other sounds to communicate intent and emotions. The sounds produced by the vocal instrument also carry crucial information about individual identity and the state of health and wellbeing. A longstanding research challenge has been in improving the understanding of how vocal tract structure and function interact, and notably in illuminating the variant and invariant aspects of speech (and beyond) within and across individuals. The first part of the talk will highlight engineering advances that allow us to perform investigations on the human vocal tract in action– from capturing the dynamics of vocal production using novel real-time magnetic resonance imaging to machine learning based articulatory-audio modeling–to offer insights about how we produce sounds with the vocal instrument. The second part of the talk will highlight some scientific, technological and clinical applications using such multimodal data driven approaches in the study of the human vocal instrument.
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Session chair(s):
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J. Fels, Teaching and Research Area of Medical Acoustics |
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Astroparticle Immersive Synthesizer³ or how cosmic “ghost particles” inspire a novel concept of spatialisation of sound
Christopher Wiebusch, Tim Otto Roth
[Abstract]
AIS3 or [aiskju:b] is a contemporary response to the ancient idea of a music of the spheres connecting cosmology and music in novel way. The sound laboratory with its 444 illuminated spherical speakers, which is presented in late Summer 2019 at the Ludwig Forum für Internationale Kunst in Aachen, adopts the form and arrangement of the weirdest telescope in the world: IceCube consists of 5160 light sensors that are frozen deep into the Antarctic ice shield at the South Pole, covering a full cubic kilometre of ice. The sensors register the tiny flashes of light that occur in the rare case of an interaction of a very special type of elementary particle: a neutrino. In our joint presentation we show the way how these ”ghost particles” made their way from astro-particle physics to become the agents of an unique fundamental experiment in psychoacoustics, turning space into a sound generator in which the visitor becomes immersed. www.imachination.net/ais3
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Session chair(s):
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T. Dau, Technical University of Denmark |
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The ear at the age of IoT
Jeremie Voix
[Abstract]
In this age of the Internet of Things (IoT), wearables are now everywhere, sometimes even in your earcanal. The research team from the NSERC-EERS Industrial Research Chair in In-Ear Technologies (CRITIAS) has been actively developing various in-ear technologies designed to complement the human ear, from ”smart” hearing protection against industrial noises, to advanced inter-individual communication systems, to hearing health monitoring devices using otoacoustic emission (OAE), to in-ear EEG Brain Computer Interface (BCI). More fundamental research has also been conducted, particularly on the micro-harvesting of electrical power from inside the earcanal to power future auditory wearables. The lastest research developments will be presented in this keynote presentation, together with some of the recent developments achieved through the CRITIAS industrial research chair in in-ear technologies or other research groups.
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Session chair(s):
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M. Ochmann, Beuth Hochschule für Technik Berlin |
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Thermoacoustic instabilities – physical mechanisms and mathematical modelling
Maria Heckl
[Abstract]
If a flame is put into an acoustic resonator, an escalating interaction between the flame’s heat release and the acoustic field can occur, giving rise to intense pressure oscillations. This phenomenon is termed ”thermoacoustic instability”. It occurs in combustion systems that have a continuously burning flame, such as gas turbines, jet aeroengines, boiler and heating systems, furnaces and rockets. Thermoacoustic instabilities are a serious problem because they can lead to excessive structural vibrations, fatigue, and even catastrophic hardware damage. The escalating flame-sound interaction occurs in tandem with other physical mechanisms, leading to a complex web of interactions, most of which are nonlinear. They include flame-vortex interactions, flame response to fluctuations in fuel concentration, entropy waves, flame-structure interactions, and others. The development of low-pollution combustion systems is very important for our environment. However, such combustion systems are particularly prone to thermoacoustic instabilities. Progress with developing combustion systems that are immune to these is hampered by insufficient physical insight. Efforts to gain further insight are going on by researchers worldwide using experimental, analytical and numerical tools. This talk aims to give an overview of the key physical mechanisms involved in thermoacoustic instabilities and will pay particular attention to mathematical modelling approaches.
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Organization:
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J. Cheer, Institute of Sound and Vibration Research |
C. Shi, UESTC, Chengdu, China |
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Effectiveness of ANC Partition with Film Speaker
Yusuke Makiyama, Shun Hirose, Kohei Oto, Yusuke Komoto, Yoshinobu Kajikawa
[Abstract]
In this paper, we design a ANC partition with film speaker and
demonstrate the effectiveness through experimental results. In a
cubicle of public areas, like a reception desk inside bank,
surrounding sound including conversation and environment
noise avoids smooth conversation and the privacy may not be
protected due to leaking to the adjacent cubicles. In such a
case, the partition should have not only passive sound control
function but also active sound control function. However, the
ordinary loudspeaker (electrodynamic loudspeaker) may not be
implemented into the partition effectively and may not control
enough region in the cubicle. In this paper, we implement a film
speaker into the partition and realize active noise control (ANC)
system. The proposed ANC partition can achieve more than 10
dB noise reduction and wider zone of quiet compared with the
ANC partition using the ordinary loudspeaker.
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Feedforward Control of Fan Noise in Ducts using Multichannel Order-reduced Inverse Filters
Mingsian R. Bai, Shin-Cheng Huang
[Abstract]
Single-channel feedforward control is a commonly used
approach for active noise control (ANC). In this paper, a time-
domain underdetermined multichannel inverse filtering
technique is proposed for the feedforward active control (ANC)
of duct noise. In the commonly used filtered-x least-mean-
square (FXLMS) algorithm, the feedforward control problem is
formulated as an overdetermined inverse filtering problem
which generally results in non-zero residual noise. By introducing
multiple secondary sources, the problem can be reformulated
into an underdetermined system, which admits infinite number
of exact solutions with zero residual noise. Linearly constrained
minimum variance (LCMV) method is employed in the controller
design. However, as a major shortcoming of the time-domain
approach, the length of the filters tends to be too long to admit
digital signal processor (DSP) implementation. To address the
problem, the least absolute shrinkage and selection operator
(LASSO) algorithm, is exploited to effectively reduce the
controller orders. A two-channel system is implemented to
suppress the fan noise of an air-cleaner. Simulation and
experiment results have demonstrated that the proposed
approach has achieved significantly higher noise reduction
than the conventional FXLMS algorithm.
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Integrated Simulation of Active Noise Cancellation using a Computational Fluid Dynamics Approach
Arun Cherkkil, Vinod Narayanan, Nithin George
[Abstract]
An attempt has been made in this paper to simulate active
noise control (ANC) in a three dimensional space using
computational fluid dynamics, pressure acoustics and
solid mechanics. We have considered a rectangular box, with
primary and secondary sources at the two opposite sides of the
box. A model of ANC has been developed, which considers a
cardboard surface and air as the fluid. A control algorithm,
which interacts with the fluid dynamic model has also been
developed, which tries to achieve active noise control. The
proposed approach can be used as an initial feasibility study to
predict acoustic pressure field inside the control space.
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Limits of Noise Control over Space
Thushara Abhayapala, Jihui Zhang, Prasanga Samarasinghe, Wen Zhang
[Abstract]
This paper investigates the best possible performance of
noise cancellation over a region of space for a fixed set of
secondary sources within a given acoustic environment. We
first formulate the spatial active noise control (ANC) problem
in a 3-D acoustic enclosure (room). Secondly, we derive a
wave-domain least square method by matching the
secondary sound field to the primary sound field in wave
domain. Thirdly, we propose a subspace method by
matching the secondary field coefficients to the projection of
primary sound field in the subspace. Simulation results
compare between the wave-domain least square method and
the subspace method, in terms of energy of the loudspeaker
driving signals, noise reduction inside the region, and
residual noise field outside the region. The subspace
matching method provides the limits of the ANC system for
the given constraints and the acoustic characteristics of the
room.
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Formation of local quiet zones using the length-limited parametric array loudspeaker
Yue Wang, Ruicong Li, Chuang Shi, Youxin Li
[Abstract]
This paper presents the numerical study of forming local quiet zones using the length-limited parametric array loudspeaker (PAL). The PAL is a directional sound device using the nonlinear acoustic effect to generate an audio beam in air. The length-limited PAL targets to control the length of the audio beam by using two different ultrasonic carriers. The sound propagation curve and radiation pattern of the length-limited PAL are theoretically distinct from those of the conventional loudspeaker. Therefore, the shape of the local quiet zones formed by the length-limited PAL is of sufficient interest before any practical deployment. For comparison, the local quiet zones formed by the conventional loudspeaker and the normal PAL are also obtained in the numerical simulation.
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Robust stability and performance of local active control systems using virtual sensing
Stephen Elliott, C K Lai, Thibault Vergez, Jordan Cheer
[Abstract]
Local active noise control systems often use virtual sensing techniques, to estimate the pressure to be controlled at the listener’s ears from an array of remote monitoring microphones for example. The virtual sensing process inevitably makes assumptions about the acoustic environment, although the environment is generally subject to uncertainties under different operating conditions. This paper considers the performance and stability of the remote microphone technique, as an example of a virtual sensing method, in a vehicle when subject to uncertainties due to changes in the internal acoustics and road conditions. It is shown, using data from models and experiments, that both the robust stability and the robust performance are improved by using regularisation in the design of the filter used to estimate the pressure at the listener’s ears from the outputs of the array of monitoring microphones.
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Tonal active control of the power scattered by locally-reacting spheres using a small number of radiators near the surface
Mihai Orita, Stephen Elliott, Jordan Cheer
[Abstract]
A theoretical investigation is performed where spherical harmonic series expansions are used to describe the primary and secondary pressure fields in the problem of actively controlling sound scattering around spheres. The study focuses on the scenario of point-monopole radiators in the vicinity of a sphere with a uniform, locally-reacting and real-valued surface impedance. A control method based on minimizing the contributions of the spherical harmonic components to the scattered sound power is proposed. Under the established circumstances, any radiated sound power is found to be dominated by a few spherical harmonic terms of low degree in the region of large relative wavelength. As this wavelength decreases, numerous terms of higher degree become significant, thus increasingly more sources are required to achieve any suppression. A study case is performed where one point-monopole is placed at infinite distance to form an incident plane-wave. The scattering due to this is controlled with up to four point-monopoles on or near the surface of the sphere. Progressively better attenuation is achieved at large relative wavelength when increasing the number of control sources and the behavior can be described using asymptotes of the spherical harmonic coefficients.
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Synthesis of the Ineter by Direct Acceleration Feedback
Neven Alujević, Ivan Ćatipović, Marko Jokić, Hinko Wolf
[Abstract]
Physical realisations of the inerter are often such that they must be either large scale, i.e. rack and pinion inerters, or they inherently include additional elements in parallel or in series with the inerter, i.e. shunted electromechanical transducers or active force feedback inerter realisations, or alternatively they bring along a large parasitic damping, i.e. fluid-based inerters. In this study inerter is realised by feeding back the subtracted outputs of two accelerometers attached to a reactive force actuator terminals. Although in theory such feedback loop is unconditionally stable due to the collocated sensor-actuator arrangement, in practice it may not exhibit good stability properties due to the lack of duality between the sensors and the actuator in conjunction with the internal dynamics of the transducers. Therefore in this paper dynamics of seismic accelerometer sensors and an electrodynamic actuator are fully incorporated into a theoretical model of such an inerter. The inerter model is coupled into a two degree of freedom mechanical system in order to study the stability of the feedback loop and the maximum possible synthesisable inertance. The results indicate that it is crucial to have a highly damped accelerometer resonance in order to achieve good stability and large synthesised inertance.
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Organization:
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S. Herold, Fraunhofer LBF |
E. Rustighi, University of Southampton |
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Theoretical and Experimental Analysis of a Skyhook Damper for Active Control of Sound Transmission
Neven Alujević, Steven Claes, Martina Šimag, Paul Sas
[Abstract]
In this paper active control of sound transmission through a double panel is considered. The active control system consists of one accelerometer mounted on the radiating panel whose time-integrated and amplified output is used as the error signal to drive a voice-coil actuator that reacts between the source and the radiating panel. Effects of one such active damper in the centre of the double panel system are considered, although, in principle, it could be repeated in the x-y plane and then used in the centralised, distributed or a decentralised manner. Theoretical and an experimental results are presented and compared. In the theoretical part, the mobility matrix model is used to investigate the stability of the feedback loop and to calculate the sound transmission ratio with and without control. It is shown that large reductions of the sound transmission ratio are possible, however only if the natural frequency of the fundamental source panel mode is higher than the natural frequency of the fundamental radiating panel mode. This is also discussed and shown analytically on a reduced order model. The experimental part of the study is carried out on the prototype active panel system and the measured result fully corroborate theoretical predictions.
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Hybrid Mass Damper Using Electromagnetic Resonator: Application to a Helicopter
Simon Chesne, Guillaume Inquieté, Paul Cranga
[Abstract]
This work aims to illustrate how to increase the efficiency of
an initial passive damped electromagnetic resonator by
controlling the mass response actively. In aerospace and
aeronautical engineering a fail-safe design for active systems
is usually required. Hybrid Mass Dampers (HMD) tend to
realize these objectives. Design rules and limitations of the so-
called Hybrid Mass Damper (HMD) are discussed and
analyzed. The main idea is to modify a DVA (Dynamic
Vibration Absorber) to use it as an AMD (Active Mass
Damper). Objectives are to take the best of the two
technologies and increasing the performance of the passive
device and/or decreasing the control effort of the actuator
in comparison with a fully active system. A specific
electromagnetic resonator is then designed and an
appropriate control law is proposed. The methodology is
illustrated and validated with an application to a helicopter.
Theoretical and numerical results are completed by
experimental validation.
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Active Engine Mount
Stefan Loheide
[Abstract]
Conventional consumption and weight reduction measures aimed at an optimization of modern engine suspension systems, usually lead to an increase in vibrations acting on the powertrain suspension mounts. By application of active engine mounts these vibrations which are limited to special engine orders are decoupled from the subframe. The design and operating principle of the active engine mount are explained. Particular focus is on the selection criteria for the actuator as well as the interaction and coordination with the characteristic mount parameters. Another focal point of this representation is the description of the control system including the necessary boundary conditions. The operating principle and effect of active engine mounts in the complete system is demonstrated by taking results obtained from measurements in the vehicle into account.
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Simulation-Based Multi-Objective Optimization of a Fuzzy Controller for Semi-Active Suspension
Louis Balzer, Valentin Mees, Jonathan Millitzer, Giovanni Lapiccirella
[Abstract]
This paper presents a simulation-based design process for a fuzzy-controlled semi-active suspension system applied to a real-time vehicle simulation. Firstly, a control architecture for the fuzzy-logic controller is defined. Secondly, different Genetic Algorithms (GAs) are configured for the optimization of the control parameters. A multi-objective optimization is performed in order to simultaneously improve safety and comfort of the driving vehicle, defined using two independent cost functions. The contact forces at the wheels estimate the safety and the vibration in the car body estimates the comfort.
The performance of the controller is compared for each set of parameters obtained by the different GAs adopted. A reduced order real-time simulation environment has been set for a holistic vehicle simulation, which includes a Finite Element Model (FEM) of the chassis, non-linear suspensions, multi-body physics and the designed digital controller. Finally, the real-time environment is integrated into an overall optimization process and is used for fitness evaluation.
The semi-active system under test allows to shift the Pareto front beyond the limit of passive systems, achieving simultaneously better safety and comfort. The different GAs come up with various near-optimal solutions, which are compared using their Pareto fronts.
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Design and applications of lean active resonator silencer cassettes
Jens Rohlfing, Karlheinz Bay, Peter Brandstätt
[Abstract]
Active resonator silencer cassettes (ASCs) are composed of a housed loudspeaker coupled with a microphone sensor via an analog electronic control loop. ASC-units are very efficient in attenuating low-frequency noise, especially in applications with tight constraints on the available installation space. At the Fraunhofer IBP the design of ASC-units, ASC-arrays and their practical application has been investigated for more than two decades. This paper gives a brief overview on the characteristics, the performance and some practical applications of classical ASC-units and arrays. Classical ASC-units have dimensions in the range of decimetres. More resent work on ASCs at the Fraunhofer IBP has been focused on the design of much leaner units with dimensions in the range of centimetres. Scaling the dimensions of the ASC-units down by an order of magnitude widens the range of possible applications, but also comes with a number of design challenges. In this paper, these design challenges and the characteristics of the lean ASC-units and arrays are discussed. The paper also presents results of studies where the lean ASC-units are used to build up a compact active noise control system for a partly opened sliding window, and a compact active silencer system for a domestic ventilation duct.
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A boundary virtual sound barrier system for sound radiation through openings with double-layer secondary sources and error microphones
Shuping Wang, Xiaojun Qiu, Jiancheng Tao
[Abstract]
Apertures, like vents of air-conditioning systems and slits of
partially opened windows or doors are very common in the
buildings, but the transmission loss performances are strongly
influenced by the presence of these apertures. This paper
proposes an analytical method to calculate the sound
radiation from inside an enclosure through an aperture of
finite thickness to the free field based on the modal
superposition method and Kirchhoff-Helmholtz equation. The
active noise control technique is then applied on the
aperture to reduce noise transmitted through it to the
outside. Different configurations of secondary sources and
error microphones are investigated. It is found that sound
radiation can be effectively reduced with an active noise
system at the edge of the aperture, and the upper limit
frequencies of effective control are discussed.
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Organization:
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H. Lissek, Signal Processing Laboratory, École polytechnique fédérale de Lausanne, Switzerland |
S.A. Cummer, Electrical and Computer Engineering Department, Duke University, Durham |
E. Ballestero, London South Bank University, The Acoustics Group |
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Active Poro-elastic Acoustic Meta Materials
Christopher Fuller
[Abstract]
Previous work has demonstrated the potential of poro-elastic acoustic meta materials for providing increased sound absorption at low frequencies[1]. This class of acoustic meta materials consist of a foam poro-elastic matrix supporting periodic arrays of microporous polymer sheets or small spherical masses. The increase in sound absorption and/or transmission loss occurs either by resonant wave scattering between the microporous sheets or the resonant dynamic motion of the embedded masses. In this paper we report preliminary work on extending the poro -elastic acoustic meta material to including active elements. The intention is to increase the bandwidth of the AMM attenuation, which is usually narrow, due it being based upon resonance conditions. In the active poro-elastic meta material some of the embedded masses are replaced with small inertial active vibrators with the same weight as the static mass. The control approach was a standard digital feedforward approach. Tests were carried out with narrowband and broadband plane wave signals in a standing wave tube. The results indicate there is potential in using active elements to broaden the frequency response of poro-elastic acoustic meta materials. 1. Fuller, C.R and Mathur, G.P., ”Poro-elastic Acoustic Meta Materials”, Proceedings of Inter Noise 2017, Hong Kong, August, 2017.
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Analysis and Optimisation of an Active Noise Control System as a Potential Acoustic Metamaterial Building Block
Joe Tan, Jordan Cheer, Stephen Daley
[Abstract]
Active Noise Control (ANC) systems have been employed in a
variety of applications to provide high performance noise
control at low frequencies, within a lightweight and compact
package. Recently, acoustic metamaterials (AMM) have been
proposed and demonstrated as an alternative approach to
achieving high levels of noise control. AMM are engineered
structures that consist of an array of subwavelength unit
cells, which exhibit behaviour not seen in conventional
materials. For example, passive resonators have been
designed to achieve negative effective material properties
and manipulate wave propagation. These AMMs, however,
typically have narrow band gaps, where wave propagation is
forbidden. When acting as an AMM, ANC systems have been
shown to improve performance, tuneability and adaptability,
but physical insights have generally been limited. Therefore,
this paper investigates the effects of traditional ANC on the
effective material properties and shows physical insight
through an analysis of the wave propagation within a one-
dimensional duct system. The active unit cell has then been
optimised to directly minimise the effective material
properties and a corresponding physical analysis has been
presented.
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Design of Acoustic Partitions with Thin Plate-like Acoustic Metamaterials
Felix Langfeldt, Wolfgang Gleine
[Abstract]
Acoustic metamaterials have emerged as new means for sound control with extraordinary properties, such as negative effective density and/or bulk modulus. Amongst the variety of different realizations of acoustic metamaterials, thin plate-like metamaterials (e.g. membrane-type acoustic metamaterials or inhomogeneous plates) have a high potential for improving the sound transmission loss of conventional partitions, especially in the challenging low-frequency regime. These types of metamaterials have been previously shown to achieve very high sound transmission loss values which can exceed the corresponding mass-law values considerably. However, further investigations have shown that when these metamaterials are integrated into a double wall partition, the beneficial effect of the metamaterials can be greatly diminished if an improper design is chosen. This contribution aims at providing a more detailed understanding of the important parameters in the design of acoustic partitions with thin plate-like acoustic metamaterials. The metamaterial is represented by a simple black-box model in order to reduce the parameter space. Analytical calculations of combinations of the metamaterial with single and double walls are used to identify the most relevant design parameters and possible limitations in the acoustic performance. General design guidelines for acoustic partitions are derived from these results and applied to laboratory test samples.
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Non-linear Metamaterial Structures: Array of Particle Dampers
Sifa Gul Demiryurek, Anton Krynkin, Jem Rongong
[Abstract]
Passive dampers to treat the excessive structural vibration has
long been researched and used in industry. Multiple particles
placed in a container can be used to dissipate excessive
vibration through the inelastic collisions between particles and
cavity of the damper has been shown. However, their
application is usually limited to treating certain modes of
structural vibration and their performance is highly dependent
on location. Another limitation of the particle dampers is
nonlinear character caused by discontinuity and randomness of
the collisions and velocity of the particles. In this paper, it is
proposed to modify a particle damper into a metamaterial type
structure in order to expand the applicability range.
Metamaterials are known to exhibit subwavelength
performance offering superior vibro-acoustic properties over a
wide range of frequencies. To maintain metamaterial
properties, the casing of the particle damper is designed to
resonate near selected modal frequencies. The Bloch-Floquet
theory is applied in studying the doubly and singly periodic
arrangement of the resonating damper shells with and without
particles. Finally, the nonlinear effects observed in the
metamaterial structure made of particle dampers are modelled
numerically to predict their vibro-acoustic effects in finite
structures. The theoretical, numerical predictions are compared
with the experimental results.
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Organization:
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M. Pawelczyk, Institute of Automatic Control, Silesian University of Technology, Gliwice, Poland |
Y. Kajikawa, Kansai University |
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Comparisons of Two Virtual Sensing Methods for Broadband Noise
Reo Maeda, Yoshinobu Kajikawa
[Abstract]
Conventional Active Noise Control(ANC) reduces the
target noise
around the error microphone. Therefore, the noise
reduction may not
be enough at the desired position unless the error
microphone is
located there.
To solve this problem, two virtual sensing techniques have
been
already proposed, which can obtain noise reduction at
the desired
position. One is called the remote microphone (RM)
method; it’s valid
especially for reducing the narrow band noise. However,
there are
two main issues in the RM method. When estimating the
transfer
function between microphones, some dips are generated
at particular
frequencies. Furthermore, the transfer functions cannot be
estimated
on some directions of noise arrival. On the other hand, the
other
method, which is called the auxiliary filter based virtual
sensing (AF-
VS), is more effective on reducing the broadband noise
from all
directions.
In this paper, we compare the noise reduction
performances of the
two methods for broadband noise to clarify the
effectiveness through
simulations in MATLAB.
As a result, both methods can shift the maximum noise
reduction to
the desired position, respectively. Comparing the two
methods, AF-
VS method can reduce the target noise better than RM
one at the
desired position on various arrangement.
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Virtual Sensing Technique for a Multi-Reference and Multi-Error Active Noise Control System
Rong Xie, Chuang Shi, Huiyong Li
[Abstract]
This paper investigates the virtual sensing technique for the multi-reference and multi-error active noise control (ANC) system. Our previous study has validated the virtual sensing technique for a multi-channel feedforward ANC system where only one reference signal was available. In this paper, two reference signals are considered so that the numbers of control filters and auxiliary filters are doubled accordingly. A comparison is carried out between the case when 4 physical error microphones are placed to form 8 virtual error microphones and the other case when 8 physical error microphones are used to form 4 virtual error microphones. The limitation of the virtual sensing technique is thus concluded.
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A Method for Detecting Convergence Completion of Adaptive Filter Cancelling Feedback Path Appeared in Active Noise Control Systems
Kensaku Fujii, Mitsuji Muneyasu
[Abstract]
We present a method for detecting the convergence completion of the feedback control filter working as an adaptive filter. Generally, the feedback path is manually estimated before starting active noise control. This estimation, however, is desired to be automatically completed. We first introduce a step size control method capable of completing the estimation with a prescribed error even where the power of disturbance fluctuates. Then, we propose a method for detecting the convergence completion of the adaptive filter. In practical use, since the impulse response of the feedback path and its power gain are unknown, we cannot directly monitor the difference between the impulse response and the coefficients of the adaptive filter. For solving this problem, we apply two adaptive filters with two different step sizes to estimating the feedback path. Actually, we have already presented a recursive filter expression illustrating the structure of the normalized least mean square algorithm. According to the expression, the square error between the coefficients of the two adaptive filters converges on the prescribed error multiplied by the power of the difference between the two step sizes. We finally verify using computer simulations that the proposed method can successfully detect the convergence.
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Practical consideration and implementation for avoiding Saturation of large amplitude Active Noise Control
Shi Dongyuan, Woon-Seng Gan, Bhan Lam, Shulin Wen
[Abstract]
The saturation distortion of the power amplifier is a common and intractable issue for most audio devices, let alone for the active noise control (ANC) system. The effect of amplitude and phase distortion, caused by the saturated audio amplifier, on the adaptive algorithm in ANC is elaborated in this paper. To overcome this issue, the previously proposed the two-gradient FxLMS algorithm is applied, which efficiently suppresses the amplifier saturation and has as the same computational complexity as the FxLMS algorithm. The performance of the 2GD-FxLMS algorithm is experimentally validated for the first time in the control of both periodic and broadband signals.
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Empirical Study of Decentralized Multi-Channel Active Noise Control Based on the Genetic Algorithm
Guoqiang Zhang, Jiancheng Tao, Xiaojun Qiu
[Abstract]
In an active noise control (ANC) system, computational complexity is
one major concern when designing practical control algorithms. One
approach to reducing computational complexity is to apply a
decentralized control scheme rather than the centralized scheme. A
decentralized scheme attempts to control a number of ANC subsystems
independently, where for simplicity, one subsystem consists of one
loudspeaker and one error microphone. Our recent published article has
shown theoretically that decentralized two-channel ANC can achieve
the same noise reduction performance as the centralized controller with
guaranteed convergence in the frequency domain. In this work, we
attempt to extend the results from two-channel case to N (N>1) channel
case. The challenge sits in finding N complex numbers that could
properly shape the eigenvalues of an N * N matrix for each frequency
bin towards guaranteed convergence. Due to the problem complexity,
we conduct empirical study by using the genetic algorithm (GA).
Simulated results on the channel numbers of 2, 4, 6, and 12 demonstrate
that the resulting decentralized ANC controller is also able to achieve
the same noise reduction performance as the centralized controller.
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Design and Construction of Loudspeakers with Low-Bl Drivers for Low-frequency Active Noise Control Applications
Marios Giouvanakis, Konstantinos Kasidakis, Christos Sevastiadis, George Papanikolaou
[Abstract]
The concept of the so-called low-Bl drivers is exploited, through
the design and construction of small vented loudspeakers. Their
high compliance and low-frequency resonance provide high-
efficiency response in a limited frequency region and allow the
implementation of small cabinet volumes. Their purpose is to be
the elementary building components of a compound source for
low-frequency noise control applications. The combination of
such sources can lead to dipole or multipole set-ups, in which
the radiation directivity pattern in open spaces and modal
coupling in closed spaces can be controlled via the parameters
of the individual driving signals. The construction constraints of the cabinet are accounted for,
considering its design optimization. A low-volume cabinet with ports tuned close to the driver’s resonance frequency is constructed to extend the loudspeaker’s output at low frequencies. The impact of the position, the dimension and the number of the enclosure’s tubes to the loudspeaker’s frequency response are examined through measurements. The design analysis and experiments show the direction to their proper construction. The implementation of such loudspeakers for low-frequency noise control applications is advantageous in small rooms, where the available space is of great concern.
|
Organization:
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E. Fernandez-Grande, Technical University of Denmark |
N. Xiang, Arch. Acoust. RPI, USA |
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Acquisition of Bi-Directional Reflectance Functions by Nearfield Acoustical Holography – a preliminary study
Jonathan Hargreaves
[Abstract]
It is well known that material absorption and scattering
is dependent on incidence and observation angle.
Despite this the corresponding standardised
coefficients, which are used to represent these
mechanisms within computational acoustic models,
aggregate all such dependency into single random-
incidence parameters. This limits the accuracy that can
be achieved with computational acoustic models -
even if these algorithms were to capture the wave
physics perfectly, which they often do not, the results
would not match physical reality because the input
data is too low resolution. Bi-Directional Reflectance
Functions are an established way of describing
boundary absorption and scattering in computer
graphics that have been suggested for use in acoustics.
To date, a number of algorithms have been published
that use these in simulation, but no measurement
methods are available to acquire them. There is also
ambiguity over some aspects of their definition e.g.
whether finite panel size is included as a scattering
mechanism. This paper adopts a definition suitable for
high-frequency Boundary Element Method algorithms
that use oscillatory basis functions to capture wave
directions. It then proposes an acquisition method
based on double-layer Near-Field Acoustical
Holography and assesses it accuracy using 2D
simulations.
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Improving in-situ sound absorption measurements using sparse multichannel blind deconvolution
Bruno Masiero, Stelamaris Bertoli, Alvaro Pais
[Abstract]
The sound absorption of materials is traditionally measured in
laboratory condition with one of two methods: random
incidence in a reverberant chamber (ISO 354) or normal
incidence in an impedance tube (ISO 10534).
Nevertheless, there are some materials that cannot be
measured in the lab, e.g., road surfaces, which are
recommended to be measured in-situ with the use of a single
microphone (ISO 13472).
The latter method is based on time-windowing the measured
impulse response to compare the incident and reflected wave
components.
Depending on the measurement setup, the size of the window
may result in degraded measurement quality, specially at low
frequencies.
To alleviate this effect we use a sparse multichannel blind
deconvolution approach. This methods blindly estimates the
response of the sound source, resulting in a cleaner (in this case,
more sparse) measurement that can than be used to estimate
sound absorption.
We compare the results of the proposed method with the results
obtained with the single microphone method for a wall covered
with PET-wool. The sound absorption coefficient was analyzed in
the octave bands between 250Hz and 4000Hz.
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An application of multi-scale directional dictionaries to RIR interpolation
Elias Zea
[Abstract]
The spatio-temporal sparsity of room impulse responses (RIRs)
degrades in the late part due to the stronger wave interference
compared with the early part. Meanwhile, such an interference
decays in amplitude as time progresses due to absorption in the
room, resulting in a decreasing dynamic range of the measurement.
Together these aspects pose challenging conditions for
compressive sensing applications, such as the interpolation of
RIRs measured at sparse microphone positions. In the search of
sparse transformation spaces, this paper examines the application
of a multi-scale directional dictionary (known as shearlets) to
interpolate RIR measurements. These redundant dictionaries
consist of multiple curved elementary functions, which offer a
decomposition of the acoustic wavefronts into various
wavelengths, propagation directions, and times of arrival.
Results reported in this paper demonstrate the potential these
dictionaries have to interpolate RIRs in both convex and
nonconvex rooms, motivating further examination under
experimental conditions and in broader frequency ranges.
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Higher-order processing of spatial impulse responses
Leo McCormack, Archontis Politis, Oliver Scheuregger, Ville Pulkki
[Abstract]
This article primarily concerns parametric methods for rendering
multi-channel impulse responses for multi-channel loudspeaker
playback of anechoic recordings. These methods analyse a set of
spatial parameters, divide the response into components according to
the assumed sound-field model, and render the components to the
loudspeaker setup using methods informed by the analysis results.
For a direct/diffuse model, the sound is divided into non-diffuse and
diffuse components, which are reproduced over directional and
surrounding reproduction methods, respectively. In many cases, the
inputs are first divided into frequency bands for the analysis and
reproduction. In this article, some acoustic conditions, which are
especially critical of parametric methods, are identified, and the effect
of time-frequency processing and diffuse-stream processing is
investigated. In addition, a method that utilises higher-order
spherical microphone components is described, which, according to
the analysis results, overcomes the issues highlighted/emphasised by
these problematic acoustic conditions. The method divides higher-
order signals into multiple directionally biased first-order spherical
harmonic signals, which are then analysed independently. The non-
diffuse parts are reproduced individually for each sector, and the
diffuse parts are encoded back into a higher-order signals, and
subsequently reproduced using a matrixing method, followed by
phase decorrelation.
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Deep Learning Applied to Dereverberation and Sound Event Classification in Reverberant Environments
Mingsian R. Bai, Wen-Chuan Chen
[Abstract]
This paper investigates dereverberation and sound event
detection techniques, with the aid of deep learning. The
system consists of two units: a microphone array front-
end and a deep learning back-end. The system is examined in
the context of two important acoustic signal processing:
dereverberation and sound event classification (SED). For the
dereverberation problem, a neural network-based approach is
compared with other state-of-the-art methods such as
beamforming, the multi-channel inverse filtering (MINT), the
multi-channel Wiener filter (MWF), and the variance-normalized
delayed linear prediction (NDLP). For the SEC problem, two
approaches are also compared. Approach 1 is based on the
signals enhanced by the microphone array front-end, which
serves as the input to the back-end deep neural network (DNN)
classifier. The DNN is implemented by using the VGGNet®.
Approach 2 is a direct approach that uses reverberant data
directly to train the classifier without the front-end
enhancement. The audio features are extracted as MFSC from
the AudioSet from Google®. A room response simulator based
on the image source method is employed to create reverberant
signals for numerous RT60 conditions in the training phase.
Perceptual evaluation of speech quality (PESQ) and F1-Score
are adopted to assess the audio quality and classification
performance.
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Binaural Dereverberation Based on Delayed Widely Linear Prediction in the Time Domain
Xin Leng, Jingdong Chen, Jacob Benesty
[Abstract]
Sound spatial information benefits human listeners in
reverberant environments. In this paper, we deal with the
problem of binaural dereverberation, which reduces
reverberation and meanwhile preserves the spatial
information of the sound source at the binaural outputs.
Specifically, a widely linear (WL) filtering framework is
adopted where the multiple real microphone signals are
merged into complex signals. By doing so, the desired
binaural outputs are also converted into complex signals with
one channel being its real part, and the other channel being
its imaginary part. In such a framework, the complex late
reverberation is modeled using the multichannel delayed WL
prediction by fully taking advantage of the noncircularity of
the complex signals. The maximum likelihood method is then
used to estimate the optimal prediction filter with the speech
signal of interest being modeled by a complex normal
distribution. The relationship between the proposed method
and the weighted prediction error (WPE) method is also
discussed. Finally, simulation results are provided to justify
the effectiveness of the developed method.
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Dereverberation Based on Deep Neural Networks with Directional Feature from Spherical Microphone Array Recordings
Jeongmin Liu, Byeongho Jo, Jung-Woo Choi
[Abstract]
The dereverberation of a reverberant audio signal can be
accomplished through the deconvolution of a room impulse
response (RIR) from the reverberant signal. However, the room
impulse response is unknown in most practical situations, which
makes the dereverberation as a challenging problem. In this
work, we tackle the dereverberation problem by incorporating
a learning-based algorithm and directional input features
measured by spherical microphones. The previous
dereverberation studies utilizing the deep neural network (DNN)
have shown that the performance strongly depends on RIRs
used for the training data, because the trained network cannot
generalize various room conditions. To build a more generalized
model, we use directional features recorded by a spherical
microphone array as the input to the DNN. Since the
reverberation includes sound waves propagating to many
directions, the use of directional cues can provide useful
information on the room condition, thereby helping to construct
a more generalized dereverberation model. When the DNN is
trained by 500 speech utterances convolved with 20 RIRs and
tested with 20 unseen RIRs, the use of directional features
greatly improves short-time objective intelligibility (STOI) and
perceptual evaluation of speech quality (PESQ) as compared
to the same model trained by single channel features.
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Acousto-optic sensing – spatial reconstruction of the sound field enclosed in a room
Samuel Arturo Verburg, Efren Fernandez-Grande
[Abstract]
Measuring the sound pressure field over medium and large volumes, such as rooms, can be cumbersome. Conventional sensing methods acquire the sound pressure (or other field quantity) at single a point. Therefore, to acquire the three-dimensional sound field in a room, a large number of measurement points must be distributed throughout. In this study, we propose a method to capture the sound field in a volume based on the acousto-optic effect. The acoustic pressure field is captured along beams of laser light. As the light (the sensing element of this method) travels through the sound field, it captures information of the sound pressure along its travel path. In this study, the spatial reconstruction of the sound field in a room from acousto-optic measurements is presented and compared with conventional point-wise measurements. The results show that the proposed acousto-optic sensing method is well-suited for the spatial characterization of sound fields over large volumes.
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A High Order Rigid Spherical Microphone Array Design Using MEMS Microphones
Marco Berzborn, Michael Vorländer
[Abstract]
In recent years spherical microphone arrays have gained much prominence in the field of room acoustics. They provide a powerful framework for the analysis of the directional properties of sound fields in rooms. However, their angular resolution is limited by the number of microphones in the array, resulting in a
poor performance for low order arrays when used in highly reverberant sound fields found in reverberation rooms. However, higher order microphone arrays require a large quantity of microphones potentially rendering the design economically unfeasible.
We present a high order spherical microphone array design using consumer grade MEMS microphones combined with an equalization kernel implemented on an FPGA. A performance analysis of the prototype regarding the usable frequency range
and the achievable angular resolution is given.
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Robust localization of early reflections in a room using semi real-valued EB-ESPRIT with three recurrence relations and Laplacian constraint
Byeongho Jo, Jung-Woo Choi
[Abstract]
To localize early reflections in a reverberant room, the
direction-
of-arrival (DOA) estimation technique has been evolved in
various ways. Among many techniques, the subspace-
based
localization techniques such as eigenbeam estimation of
signal
parameters via rotational invariance techniques (EB-
ESPRIT)
have been extensively researched for their ability to
directly
estimate DOA parameters from spherical microphone
recordings. The EB-ESPRIT estimates DOAs by computing
the
directional parameters from a single recurrence relation of
spherical harmonics. For estimating DOAs of early
reflections,
however, the EB-ESPRIT is inappropriate due to the
insufficient
number of detectable echoes and its vulnerability to noise.
In
this study, we attempt to overcome these issues by using
three
different recurrence relations of spherical harmonics and
extra
constraints. The proposed technique can compute
the directional parameters more robustly by imposing
Laplacian constraint derived from the Helmholtz wave
equation.
We show that constraint imposed on the joint
eigenvalue
decomposition (JEVD) algorithms can further enhance the
accuracy of DOA estimation as compared to all EB-ESPRIT
based techniques. In simulations, we analyze the number
of
detectable echoes and validate the robustness of the
proposed
technique in terms of the estimation error under different
signal-
to-noise ratios (SNRs) and direct-to-reverberant energy
ratios
(DRRs).
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Double Tetrahedral Intensity Probes for Reducing the Spatial Bias Error of Source Localization
In-Jee Jung, Jeong-Guon Ih
[Abstract]
A compact tetrahedral probe used for measuring the three-dimensional acoustic intensity vector can be used for the source localization. Although such 3D intensimetry is advantageous in miniaturizing the sensing system size, it has not been popularly used due to its large spectral and spatial bias errors, which are additional to the bias errors at low and high frequency ranges. Compensation methods for spectral bias errors are recently proposed, but the spatial bias error, which is related to the probe orientation, is not easy to compensate. In this work, the idea is to adopt two probes together, thus effectively arranging the microphones to reduce the spatial sparseness of sensors and the irregularity in directivity. The number of microphones is minimized by sharing 1 or 3 microphones in the double module probes, while permitting the calculation of two intensity vectors. Two different types of probe configurations are used: twisted double probes, double tetrahedral probe symmetric to a face of the tetrahedron. A numerical simulation is conducted to compare the proposed probe systems with single probe. The result shows that the residual spatial bias error is less than 7.5∘ for 2.5<kd<4.1 range.
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Sound field reconstruction in a room from spatially distributed measurements
Efren Fernandez-Grande
[Abstract]
The full characterization of the sound field in a room from experimental data is a challenging problem, as the direct measurement of the three-dimensional sound field in the volume requires considerable experimental effort. In this study, the sound field in a room at low frequencies is characterized using a representation in wavenumber domain that infers the modal structure of the sound field. The modal structure is used to synthesize frequency response functions, to characterize the spatio-temporal properties of the wave field. The proposed method makes it possible to infer frequency response functions between source-receiver pairs that have not been measured, exploiting the representation of the sound field as a modal superposition.
|
Organization:
|
M. Cobos, Universitat de Valencia |
D. Kolossa, Ruhr University Bochum |
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Probabilistic Modeling for Learning-based Distance Estimation
Andreas Brendel, Andy Regensky, Walter Kellermann
[Abstract]
Estimating the position of an acoustic source is an essential step for many signal processing applications. Hence, many approaches for acoustic source localization have been proposed in recent years, whereby most of them are based on the estimation of the direction of arrival of the acoustic source relative to an observing microphone array. In contrast, the estimation of the source-microphone distance is much less investigated. Especially for complex sound fields in acoustic enclosures, a promising approach is to apply machine learning approaches to learn the mapping of a distance-related feature to the corresponding source-microphone distance. Here, we focus on a recently proposed method for distance estimation based on the coherent-to-diffuse power ratio and Gaussian process regression. We investigate the influence of different probabilistic models reflected by the choice of kernel and mean functions of the Gaussian process employed for regression. The influence of these choices is quantified by experiments.
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Speaker Distance Estimation using Binaural Hearing Aids and Deep Neural Networks
Mehdi Zohourian, Jakob Stinner, Rainer Martin
[Abstract]
In this paper, we address the problem of speaker distance estimation using binaural hearing aid microphones. The proposed approach is based on deep neural networks which aim to classify discrete distances at different levels of precision. We investigate two types of networks, namely, feedforward and convolutional neural networks which are trained on the binaurally estimated direct-to-reverberant energy ratio. The performance of the proposed algorithm is assessed on speech signals convolved with both synthetic and real binaural room impulse responses for several distances and in different acoustical scenarios. On the one hand, the proposed approach offers a reliable classification of coarse
distances classes which is also robust against variations of the reverberation time. On the other hand, when a more accurate distance estimation is required, the proposed approach is robust only against small variations of the reverberation time.
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Binaural Direction-of-Arrival Estimation in Reverberant Environments Using the Direct-path Dominance test
Hanan Beit-On, Boaz Rafaely
[Abstract]
Speaker localization in reverberant environments is an
important and current challenge in audio signal
processing with several recently methods proposing
solutions to this challenge. One such family of methods is
based on the direct-path dominance (DPD) test and was
developed specifically for spherical arrays. In this paper,
the DPD test based method is theoretically extended to
arbitrary arrays and is specifically studied for a binaural
array, showing good performance under diverse
reverberation conditions. This extension of the DPD test
based approach uses a spherical harmonics
decomposition of the steering matrix to remove the
frequency dependence of this matrix within frequency
bands using focusing. This facilitates frequency smoothing
which is necessary for decorrelating coherent reflections.
The success of this focusing process may be affected by
the frequency bandwidth as well as the array
configuration. This paper presents an analysis of the
performance of this focusing process and investigates its
impact on the DPD test performance for binaural arrays. It
is shown that the frequency bandwidth necessary for
frequency smoothing may be sufficiently small such that
the focusing stage may be skipped when applying the
DPD test for binaural arrays. A simulation study that
validates this result is presented.
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Data-driven Threshold Selection for Direct Path Dominance Test
Orhun Olgun, Huseyin Hacihabiboglu
[Abstract]
Direction-of-arrival estimation methods, when used with
recordings made in enclosures using rigid spherical
microphone arrays (RSMAs), is negatively influenced by the
reflections and reverberation in that enclosure. Direct path
dominance (DPD) test was proposed as a pre-processing
stage which can provide better DOA estimates by selecting
the time-frequency bins with a single dominant sound source
component thereby reducing the total computational cost.
DPD test involves selecting bins for which the ratio of the two
largest singular values of the local spatial correlation matrix
is above a threshold. The selection of this threshold is
typically carried out in an ad hoc manner, which hinders the
generalisation of this approach, potentially increasing the
total computational cost or reducing the accuracy of DOA
estimation. We propose a DPD test threshold selection
method based on a data-driven statistical model of singular
value ratios. The model is based on the approximation of the
singular value ratios as a generalised Pareto distribution and
allows selecting bins based on their probability of occurrence.
We demonstrate the application of this threshold selection
method using numerical simulations, emulations using
measured acoustic impulse responses and a real recording
using an RSMA.
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Source Localization using a Spatial Kernel based Covariance Model and Supervised Complex Nonnegative Matrix Factorization
Antonio Jesús Muñoz Montoro, Violeta Montiel-Zafra, Julio José Carabias-Orti, Juan Torre-Cruz, Francisco Jesús Canadas-Quesada, Pedro Vera-Candeas
[Abstract]
This paper presents an algorithm for source localization
using a
beamforming-inspired spatial covariance model (SCM)
and complex
non-negative matrix factorization (CNMF). The spatial
properties are
modeled as the weighted sum of spatial kernels which
encode the
phase differences between microphones for every
possible source
location in a grid. The actual localization for each
individual source in
the multichannel mixture is estimated using complex-
valued non-
negative matrix factorization (CNMF) where each source
spectrogram
is modeled using a dictionary of spectral patterns learned
a priori
from training material. Localization performance of the proposed system is
evaluated using a
multi-channel dataset with configurations (number of
simultaneous
sources, reverberation time, microphones spacing, source
types and
spatial locations of the sources). Finally, a comparison to
other state-
of-the-art localization methods is performed, showing
competitive
localization performance.
|
Organization:
|
M. Cobos, Universitat de Valencia |
D. Kolossa, Ruhr University Bochum |
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Towards low cost acoustic cameras for the Internet of Things
Jose Javier Lopez, Maximilian Becker, Carlos Hernandez
[Abstract]
In the context of Internet of Thing (IoT) accurate and low-cost sound
source localization methods are of great interest, alone or combined
with video images. In this work, we present an implementation of the
Steered-Response Power Phase Transform (SPR-PHAT) method
employing a small circular array of 10 cm consisting of 8 MEMS
microphones all commanded by a small single-board computer
(Rasperry Pi). The localization is performed both in azimuth and
elevation in one side of the array. A camera emplaced in the center of
the array can be also employed for multimodal localization. The
evolution of the localization accuracy as more microphone pairs
autocorrelacions are added in the estimation of likelihood is
compared. An optimal number of autocorrelation pairs in the sense of
accuracy/computational cost tradeoff is obtained.
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Inference in Nonlinear Dynamical Systems with dynamic Stream Weights for Audiovisual Speaker Tracking
Christopher Schymura, Dorothea Kolossa
[Abstract]
Audiovisual speaker tracking is a prominent example of a multimodal signal processing framework with a large variety of technical applications. Whereas acoustic speaker tracking suffers from decreased performance under adverse acoustic conditions, visual sensors are not affected by purely acoustic disturbances. However, the optimal combination of acoustic and visual information remains a challenging task due to the highly dynamic nature of audiovisual scenes. Recently, a framework for audiovisual speaker localization and tracking has been proposed, which extended the notion of dynamical systems with adaptive stream weights. It essentially implements a weighting for acoustic and visual observations that can be changed at each time step and allows probabilistic inference based on the Gaussian filtering paradigm. This study presents a detailed analysis of the inference procedure and proposes an extension for nonlinear dynamical systems inspired by the unscented Kalman filter. Under the constraint that the process and observation noise terms follow a zero-mean Gaussian distribution, the inference framework presented in this study allows an efficient estimation of the speaker trajectory from audiovisual observations. A systematic experimental evaluation assesses the performance of the proposed method using recorded audiovisual scenes.
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Audiovisual active speaker localization and enhancement for multirotor micro aerial vehicles
Daniele Salvati, Carlo Drioli, Andrea Gulli, Gian Luca Foresti, Federico Fontana, Giovanni Ferrin
[Abstract]
We address the problem of localizing a speaker and enhancing
his voice using audio-visual sensors installed on a multirotor
micro aerial vehicle (MAV). Acoustic-only localization and
signal enhancement through beamforming techniques is
especially challenging in this conditions, due to the nature and
intensity of disturbances originated by the electrical engines
and the propellers. We propose a solution in which an efficient
beamforming-based algorithm for both localization and
enhancement of the source is paired to a video-based human
detection. The video processing front-end detects the human
silhouettes and provides an estimation of direction of arrivals
(DOAs) on the array. When the acoustic localization front-end
detects a speech activity originating from one of the possible
directions estimated by the visual components, the acoustic
source localization is refined and the recorded signal is
enhanced through acoustic beamforming. The proposed
algorithm was tested on a MAV equipped with a compact
uniform linear array (ULA) of four microphones. A set of scenes
featuring two human subjects lying in the field of view and
speaking one at a time is analyzed through this method. The
experimental results conducted in stable hovering conditions
are illustrated, and the localization and signal enhancing
performances are analized.
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Study on Large Scale Projectile Impact Point Positioning Method Based on Trajectory Shock Wave
Difeng Sun, Xubin Liang, Tianqing Zhao, Houlin Fang, Cheng Zhang, Hui Zheng, Liangyong Zhang, Fang Zhang, Deyu Sun, Yang Liu
[Abstract]
Abstract: With multiple nodes of single acoustic sensor layed
around the target, the ballistic shock wave signal of the
projectile generated before hitting the ground can be
detected at different locations. The trajectory can be
estimated by measuring the time difference between the
signal of different nodes, and the impact point can be
coordinated by crossing the trajectory with the target. In this
article, the projectile impact point positioning model is
parameterised and fomulated by nonlinear equations with six
patameters, and it can be solved by means of nonlinear
least
square(NLS) method. By means of Cramer-Rao lower bound
(CRLB) analysis, measurement error distribution by different
incident angle and different impact location is calculated,
and
also, parameter of exterior ballistics model and wind vector is
under consideration.
|
Organization:
|
S. Van De Par, University of Oldenburg |
S. Doclo, Dept. Medical Physics and Acoustics, University of Oldenburg |
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Binding of speech syllables when segregation occurs
Marion David, Mathieu Lavandier, Nicolas Grimault, Andrew, J. Oxenham
[Abstract]
Two previous studies [David et al., 2017, Hear. Res. 344, 235-243;
David et al., 2017, J. Acoust. Soc. Am. 142(3), 1674-1685] have
investigated the segregation of speech syllables made of a
fricative consonant and a voiced vowel, referred to as CV
tokens. The first study explored the segregation of such syllables
based on fundamental frequency differences. In the second
study, the authors assessed the segregation of CV tokens based
on localization cues, especially the spectral cues in the median
plane. In both studies they found that segregation can be
observed based on F0 and on spectral cues. More interestingly,
they found that the whole CV token stay grouped even when
segregation occurs based on cues that affect only one part of
the CV: F0 differences affect mostly the vowel part, whereas
coloration in the median plane is effective mostly at high
frequencies, which selectively affects the consonant part. The
mechanisms that allow the CV to remain grouped under such
circumstances remain unclear. The present manuscript reviews
the results of these two studies and provides some suggestions
as to how such binding might occur.
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Musical scene analysis of hearing-impaired and normal- hearing listeners: a melody and instrument matching task
Kai Siedenburg, Saskia Röttges, Kirsten Wagener, Volker Hohmann
[Abstract]
It is well known that scene analysis is at the core of hearing-
impaired (HI) listeners’ difficulties of understanding speech in
noise. Surprisingly little is known about scene analysis of HI
listeners in musical contexts. Here we tested 24 moderately
HI listeners and 28 normal-hearing (NH) listeners in their
ability to identify instruments and melodies in the presence
of a musical accompaniment that acts as a masker
(cello/piano dyads or spectrally matched noise). Target
signals consisted of four-note melodies. In each trial, a
signal-masker mixture was presented followed by two
different versions of the signal without masker. Listeners
judged which signal version was part of the mixture. Signal
versions either differed in terms of timbre (flute vs. trumpet)
or in terms of melody. Signal-to-masker thresholds (71%
correct response rate) were measured by varying the signal
presentation level in a 2down-1up procedure. The masker
level was set to 65 dB SPL for NH listeners and to medium
loudness for HI listeners. We observed drastically elevated
thresholds for HI listeners (>+10dB) compared to NH
listeners. Participants with musical training generally
exhibited significantly lower thresholds. These results for the
first time demonstrate drastic effects of hearing impairment
on musical scene analysis.
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Comparative Study of Single-Channel Algorithms for Blind Reverberation Time Estimation
Heinrich Löllmann, Andreas Brendel, Walter Kellermann
[Abstract]
Knowledge about the reverberation time (T60) is exploited by numerous algorithms, e.g., for automatic speech recognition, speech dereverberation, or as a feature for acoustic scene classification.
This contribution provides a comprehensive survey about various state-of-the-art methods to estimate the reverberation time blindly from a noisy and reverberant speech signal and compares their estimation performance for different acoustic scenarios. The evaluation considers different acoustic conditions regarding the signal-to-noise ratio (SNR), direct-to-reverberation ratio (DRR) and T60, and uses databases with room impulse responses measured in different acoustic enclosures. The findings of the Acoustic Characterisation of Environments (ACE) Challenge are extended by taking into account algorithms and acoustic scenarios which are not considered in this benchmarking campaign, and by using a much larger set of reverberant speech signals.
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The effect of room acoustics on audio event classification
Dimitra Emmanouilidou, Hannes Gamper
[Abstract]
The increasing availability of large-scale annotated databases,
together with advances in data-driven learning and deep neural networks,
have pushed the state of the art for computer-aided detection problems like audio scene analysis and event classification.
However, the large variety of acoustic environments and their acoustic properties
encountered in practice can pose a great challenge for such tasks
and compromise the robustness of general-purpose classifiers
when tested in unseen conditions or real-life applications.
In this work we perform a quantitative analysis of the effect of room acoustics on general audio event detection scenarios.
We study the impact of mismatches between training and testing conditions in terms of acoustical parameters, including the reverberation time (T60) and the direct-to-reverberant ratio (DRR), on audio classification accuracy and class separability.
The results of this study may serve as guidance for practitioners to build more
robust frameworks for audio event classification tasks.
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Deep network source localization and the influence of sensor geometry
Jörn Anemüller, Hendrik Schoof
[Abstract]
Learning-based localization approaches cast the acoustic speaker
localization problem as a machine learning task where a classifier is
trained on example data of acoustic feature vectors in order to
predict likelihood of speech presence as a spatio-temporal
distribution. We investigate the impact that fundamental acoustic
parameters of the auditory scene (e.g. SNR, acoustic scene
complexity, sensor geometry) exert on the ability to faithfully extract
spatio-temporal activity maps for concurrent speakers. Our results
indicate that to some degree shortcomings in the acoustic conditions
can be compensated by increased complexity in the applied
classification techniques. To this end, we systematically investigate
localization performance for a set of deep neural network localizers of
varying complexity, and for six different sensor configurations in a
bilateral hearing aid setup. Deep networks result in improved
performance compared to linear localizers, and their performance
benefits more from an increase in the number of sensor channels. In
specific configurations, deep networks with a smaller number of
microphones perform better than a linear baseline network with a
larger number of microphones. Thus, location-specific information in
source-interference scenarios appears to be encoded non-linearly in
the soundfield, requiring non-linear approaches for optimal decoding.
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Virtual auditory scenes created by time reversal mirror technique
Georgina Alejandra Lizaso, Jorge Petrosino
[Abstract]
The present paper describes the process of generating virtual
auditory scenes exhibiting multiple virtual sources in different
locations, which is accomplished through the application of the
time reversal mirror (TRM) method. This technique, developed by
Mathias Fink, can be used to focus an acoustic signal at a
particular point in space. Time-reversing the transfer function
between a TRM array and an acoustic source generates an
acoustic spatio-temporal focus at said source’s original location.
Thus, this time-reversed focus behaves as a ”virtual” source in
the outbound direction with respect to the TRM. Provided that
an acoustic impulse is previously registered by the TRM device, a
”virtual” audio source can be generated at the impulse’s
location by convolving the TRM impulse response with an audio
signal. Since the system is linear, it allows the addition of impulse
responses belonging to different locations, which can be
convolved with audio signals in order to shape the sound field of
the auditory scene. The numerical simulations implemented to
explore this method, located arbitrary audio signals in selected
positions of an auditory scene. Their results were evaluated via
comparison of the data from spatially localized sources against
the virtual sources generated by the TRM technique.
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Organization:
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J. Ahrens, Chalmers University of Technology |
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Audio Spotlight using Subdivided AM Sideband Wave Delivery from Separate Ultrasonic Array Speakers
Kazuhiro Kondo, Shinichiro Osanai
[Abstract]
Parametric array speakers can deliver an audible sound
beam by modulating sound using an ultrasonic
carrier wave and playing this modulated signal from an
ultrasonic transducer array at high levels. Audible
sound is demodulated in a narrow beam due to the
nonlinear aerial transfer characteristics. However, if sound
can be delivered within a small spotlight, sound can be
delivered only to the intended subject, and avoid
leakage to others. This is partially possible by generating the
carrier wave and the sideband signal of the
modulated signal from separate arrays, resulting in the full
demodulation of an audible sound at the
intersection of the two beams. However, some audible noise
was generated through demodulation of the
sideband signal. The level of this noise is proportional to the
relative frequency difference of the sideband
signal. Thus, we divided the sideband signal into several sub-
bands, limiting the frequency range of each sub-
band, and generated them from separate arrays,
intersecting only at the spotlight. By using 5 sub-bands, it
was possible to limit the spotlight to an effective area of
0.2m by 0.2m, while the audible noise of the
surrounding area was reduced to 20 dB lower levels.
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Evaluation of Thermal Runaway Control Based on Frequency Modulated Carrier Wave in Parametric Array Loudspeaker
Kairi Mori, Masato Nakayama, Takanobu Nishiura
[Abstract]
A parametric array loudspeaker (PAL) can realize a sharper directivity by utilizing an ultrasound wave (carrier wave). The conventional PAL has a problem that ultrasonic transducers cause thermal runaway by strongly emitting an amplitude modulated (AM) wave through a long period of time. Therefore, we have proposed a thermal runaway control method with a frequency modulated carrier wave. In the proposed method, we generate the AM wave with the frequency modulated carrier wave that the frequency of the carrier wave changes with time. By changing the frequency of the carrier wave with time, we can minimize the effects of thermal runaway in ultrasonic transducers. In the paper, to evaluate the effects of thermal runaway, we perform evaluation experiments for temperature and the frequency response of the PAL with the conventional and the proposed methods. Specifically, we compare the change in the frequency response of the PAL before and after emitting the carrier wave of pure tone and the frequency modulated carrier wave for a long period of time, and the change in temperature of the PAL at that time. As a result, we confirmed the effectiveness of the proposed method.
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Three-dimensional Resonance Control Based on Spatial Wave Synthesis with Parametric Array Loudspeaker
Shiori Sayama, Masato Nakayama, Takanobu Nishiura
[Abstract]
A parametric array loudspeaker (PAL) realizes a sharper
directivity utilizing the demodulation of intense amplitude
modulated (AM) wave. The demodulation is modeled as spatial
sound sources generated with the demodulated audible sound
in the acoustic beam. Spatial sound sources form resonance
and anti-resonance positions in the acoustic beam because of
the interference between spatial sound sources. Thus, the sound
pressure level (SPL) of the PAL is reduced at the anti-resonance
position. However, it suggests that resonance positions might be
reformed by spatial wave synthesis with phased array
processing. Therefore, we propose three-dimensional resonance
control based on spatial wave synthesis with the PAL. In this
paper, we employ the PAL consists of line-type PALs
corresponding to column. The PAL can apply different delay
filters to each column. The proposed method designs inverse
filters with a multichannel adaptive algorithm as delay filters for
line-type PALs under the constraint condition that the SPL is
maximized by spatial wave synthesis at the target position. The
proposed method can improve the SPL at the target position by
applying the designed filters to each column. Finally, we
confirmed the effectiveness of the proposed method through
evaluation experiments.
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Demodulation Distance Control Based on Analytic Model between Film Gas-lens Depth and Demodulation Distance for Parametric Array Loudspeaker
Toshihiro Fujii, Masato Nakayama, Takanobu Nishiura
[Abstract]
A parametric array loudspeaker (PAL) can transmit an audible sound to only a particular area by utilizing an ultrasound. We have
previously proposed a demodulation distance control using a film gas-lens with the gas different to the air to control a maximum
demodulation distance (MDD), which is a distance with maximum demodulation of the audible sound. In the previously proposed
method, the MDD has a large numerical error because it employs a simple linear approximation. In this paper, we therefore propose
a new MDD control based on an analytic model between film gas-lens depth and the MDD. To design the analytic model, we
investigate the relationship between film gas-lens depth and the MDD through the preliminary experiments. Also, we design a
regression curve of an exponential function into relationship between input film gas-lens depth and output MDD. Next, to calculate
the relationship between input MDD and output film gas-lens depth, the analytic model is calculated as inverse function of the
regression curve. The analytic model can accurately adjust the film gas-lens depth to control the MDD. Finally, we confirmed the
effectiveness of the proposed method through the evaluation experiments.
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Development of Multi-way Parametric Array Loudspeaker Using Multiplexed Double Sideband Modulation
Yuting Geng, Yusei Nakano, Masato Nakayama, Takanobu Nishiura
[Abstract]
Parametric array loudspeaker (PAL) can achieve a sharper
directivity by utilizing ultrasonic wave than conventional
dynamic loudspeakers. However, PAL has difficulty
reproducing low-frequency sound because it utilizes
demodulation of ultrasound in the air. We have proposed the
multiplexed double sideband (M-DSB) modulation for PAL,
which can enhance sound pressure level (SPL) by utilizing
harmonic distortion in demodulation. However, in lower
frequency, sound quality by PAL with M-DSB is still
insufficient compared with one by dynamic loudspeakers.
Therefore, in this paper, we develop a new multi-way PAL
with combining software and hardware approaches, which
can achieve both sharper directivity and powerful bass
sound. Multi-way PAL consists of a tweeter PAL and woofer
PALs, and each part is an aggregate of multiple ultrasonic
transducers. Tweeter PAL employs single sideband (SSB)
modulation, which gives low-distortion but low-SPL. Woofer
PAL employs M-DSB modulation, which gives high-SPL but
high-distortion. In order to achieve a flatter frequency
response, we conducted preliminary experiments to
determine both a suitable number of ultrasonic transducers
and an optimized arrangement for tweeter PAL and woofer
PALs. Finally, we confirmed the effectiveness of the proposed
method through evaluation experiments.
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Design of a directive source using the radiation mode method and a sound zone algorithm
Manuel Melon, Maryna Sanalatii, Philippe Herzog, Régine Guillermin, Nicolas Poulain, Jean Christophe Le Roux
[Abstract]
A common mean to achieve narrow directive patterns consists in using loudspeaker
arrays. Generally, the position of the sources in the array is arbitrarily fixed, for instance
by using a regular spacing. The filters for each speaker are then calculated to get the
desired acoustic field. In some cases, the predefined speaker positions might not be the
optimal ones. Moreover, the diffraction on the source body may not be negligible for
medium range applications. In this paper, we propose a two-step approach to overcome
these problems. First, the radiation modes of the enclosure are computed. They are used
in an inverse scheme, to find the most efficient vibration patterns over the enclosure,
able to generate a target field at some chosen frequencies. These vibration patterns are
then used to define the positions and sizes of actual loudspeakers on the enclosure. In a
second step, a sound zone algorithm is used to calculate the signals driving the speakers.
This yields a very narrow directivity pattern with reduced side lobe amplitudes. The
proposed method is tested both numerically and experimentally and results are reported.
A discussion about the pros and cons of the proposed approach is then provided.
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Models of spherical and rectangular cuboid loudspeaker arrays
Franz Zotter, Hannes Pomberger
[Abstract]
Compact spherical/cubical loudspeaker arrays are used as beamformers to radiate focused sound with adjustable direction. A common enclosure housing all the transducers is easier to manufacture but acoustically couples their motion, making their independent control more challenging Nevertheless, recetly analytic IIR filters to controll cubical arrays were proposed, which motivates that such could be defined for more general arrays in the future. As a first step, this contribution describes the analytic model of the interior crosstalk in (i) rectangular cuboid loudspeaker arrays using piston transducers, and in (ii) sperical loudspeaker arrays using spherical cap transducers. Both models are based on interior modes and their accurate integrals to evaluate both the contribution of the transducers’ velocity boundary conditions and the forces by which the produced pressure loads on active/passiive transducers. A fullly causal model is obtained by suitable regularization of the resonant frequencies associated the interior modes. Both models are compared to laser velocity measurements at the transducers’ centers of physical prototypes. This verifies the model up to frequencies where transducers break up into partial vibrations. Moreover, higher-order interior modes often become irrelevant, as at high frequencies the transducer mass starts to dominate.
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Reproduction of Multichannel Audio by Frame Loudspeaker Array
Yuta Kakuzaki, Akio Ando
[Abstract]
Multichannel audio reproduces a sound field by loudspeakers
surrounding a listener, which necessarily occupies the listening
space. The purpose of this study is to reproduce the sound by
loudspeakers placed around the image display, which is
referred as ”frame loudspeaker array” in this paper. As the first
step, we focused on the reproduction of the frontal channels of
22.2 multichannel by the array. Each channel signal of the multichannel audio was distributed
into loudspeakers with the weighting coefficients that minimized
the square error between the original sound field by the
channel and the reproduced sound field by the array. The
sound fields were represented by the tenth order spherical
functions. The signal was passed through the auditory band
pass filter and changed its amplitude and delay according to
the weighting coefficient calculated for the center frequency
of the filter. The band pass signals were then summed up to
generate the full band input signal for loudspeakers. The informal listening test showed that this method could
preserve the spatial feature of the original multichannel audio.
The reproduction of channels other than the frontal channels of
22.2 multichannel will be studied in the next step.
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Three-Dimensional Spatial Active Noise Control Based on Kernel-Induced Sound Field Interpolation
Hayato Ito, Shoichi Koyama, Natsuki Ueno, Hiroshi Saruwatari
[Abstract]
Active noise control (ANC) over a three-dimensional (3D)
spatial region is proposed.
Conventional multipoint ANC does not guarantee to
reduce the noise between multiple discrete control points.
Several attempts have been made to reduce the noise
over the continuous target region.
Most methods for spatial ANC found their basis on the
spherical/cylindrical harmonic expansion of the sound
field.
Therefore, they can be applied only for arrays of simple
geometries such as sphere and cylinder.
We propose a 3D spatial ANC method based on sound
field interpolation,
which can be applied to arbitrary array geometries.
The mathematical principle of the proposed method is
based on our recent studies on kernel ridge regression for
sound field interpolation.
The difference in the resulting algorithm from the standard
adaptive control appears only in the weighting matrix for
an error signal,
which can be computed in advance only by using the
relative position of the microphones and the target region.
Numerical simulation results indicated that a large
regional noise reduction is achieved by the proposed
method compared with conventional multipoint ANC.
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On in situ beamforming in an automotive cabin using a planar loudspeaker array
Martin Bo Møller, Martin Olsen
[Abstract]
Compact loudspeaker arrays are often used in applications
where control of the sound dispersion is required, e.g. in audio
reproduction systems and aspects of sound field control. An
example of the latter is sound zoning where the goal is to deliver
individual audio content to one listener without disturbing
another. Typical approaches are based on filter-and-sum
beamforming applied to control the resulting directivity of the
array. However, directivity is defined in free field while sound
zones are generally of interest in listening spaces with reflective
boundaries such as domestic rooms and automotive cabins. In
order to address this discrepancy, in situ zone control can be
utilized. This relies on measured transfer functions from each
array element to the spatial control points defining the zones.
Such methods increase the solution specificity and can lead to
non-causal control filters. In the present paper, methods based
on controlling the flow of acoustic energy are applied to a
planar loudspeaker array located on the dashboard inside a
car cabin. A comparison study including directivity and in situ
zone control is presented, and the solutions are evaluated in
terms of spatial robustness of the resulting sound zoning
performance.
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Minimum Trap Separation for Acoustical Levitation Using Phased Ultrasonic Transducer Arrays
Carl Andersson, Jens Ahrens
[Abstract]
Acoustic levitation for interactive visualizations is an emerging field
which uses ultrasonic transducer arrays to induce radiation forces on
small beads.
To fully utilize the interactive and responsive aspect of the system
each bead must be controllable freely in the interaction space without
interfering with the other beads.
When beads are placed far enough apart it is possible to design a
sound field that traps all the beads at their desired positions, taking
any potential limitations of the hardware into account.
The underlying physics limit how close in space two taps can be
without interfering with each other. In this paper, we investigate the minimum spacing required between
two beads for them to successfully and independently levitate in
acoustical traps.
Multiple methods for the sound field design are considered and
compared with regards to the overall separation required as well as
the gracefulness in the breakdown region.
Simulated results of the sound fields are compared and validated
against measurements of real sound fields and real levitating beads.
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Spatial sound field reproduction using deep neural networks
Thushara Abhayapala, Hanchi Chen
[Abstract]
Sparsity-based sound field reproduction algorithms often
result in improved localization and larger reproduction
region, but also lead to high computational cost. In this
work, we present a novel approach for sparse reproduction,
where a deep neural network (DNN) is trained to determine
the optimal driving signals for a loudspeaker array, given the
desired sound field coefficients as input. We show that when
trained using the proposed method, the DNN-based
algorithm can outperform existing Lasso-based algorithms in
terms of noise sensitivity and computation speed.
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2.5D Local Wave Field Synthesis of a Virtual Plane Wave Using a Time Domain Representation of Spherical Harmonics Expansion
Nara Hahn, Fieter Winter, Sascha Spors
[Abstract]
Wave field synthesis is a well established sound reproduction technique, where the
wavefront of a desired sound field is physically recreated within an extended target area
by using a large number of individually driven loudspeakers. For a non-continuous
loudspeaker distribution, the synthesized sound field typically exhibits spatial aliasing
artifacts which occur above an upper frequency limit. If the synthesis in a relatively
small area is concerned, called local wave field synthesis, a substantial improvement of
the physical accuracy can be achieved. In a recent local wave field synthesis method,
proposed by the authors, the spatial bandwidth of the driving function is limited in order
to reduce the spatial aliasing artifacts around a properly chosen expansion center. The
present paper deals with the time domain realization of this method for a 2.5-
dimensional setup. The driving function for a virtual plane wave is derived based on the
time domain version of the spherical harmonics expansion, by exploiting the connection
with the plane wave expansion. Due to the low computational cost, the proposed
approach is well suited for real-time implementation. The spatial, temporal, and spectral
properties of the synthesized sound fields are evaluated by numerical simulations.
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How the distance and radius of two circular loudspeaker arrays affect sound field reproductions and directivity controls
Yi Ren, Yoichi Haneda
[Abstract]
We have proposed methods for sound field reproductions and
directivity controls by using two parallel circular loudspeaker
arrays (2CLA) so far. Results showed that 2CLAs have a higher
potential than a circular loudspeaker array when reproducing a
virtual sound source outside the array or producing a high
directivity beam. Also, we found that the performance of the
2CLA varies with the distance between the two arrays and the
radii of the two arrays. In this paper, we discuss how the two
factors affect the performances of 2CLAs. We investigate the
influences of either factor with the other fixed, via computer
simulations. Results of sound field reproductions showed that the
reproducing accuracy falls while the distance of the two array
increases, while the radii of the 2CLA do not affect the results if
no constraint has been set to filter gain. On the other hand, if
filter gain is constrained to 0 dB, the reproducing gains while the
radii increase. For directivity controls with filter gain constraints,
the performance gets better when larger radii have been
selected, while a large distance may make the beam wide.
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The EMPAC High-Resolution Modular Loudspeaker Array for Wave Field Synthesis
Johannes Goebel
[Abstract]
In order to verify the perceptual potential and limitations of
Wave Field Synthesis (WFS), a large system was built with
loudspeakers placed as closely as possible. Design criteria
focused on the frequencies dominant in the consonants found
in human speech, which yield important localization cues, also
in transients of instrumental sounds. The built system allows for
undistorted wave field generation up to an aliasing frequency
of 3.3 kHz, with experimental configurations allowing for
observations of perceptual effects at aliasing frequencies
between 700 Hz and around 6 kHz. Critical listening shows that
raising the aliasing frequency to 6 kHz improves the spatial
resolution substantially. A second generation with a
loudspeaker distanced at less than 3 cm is under construction.
Focused sound sources can be placed up to 10 m in front of the
system. The current system’s 558 individually addressable
loudspeakers are housed in 18 modules, each with 31 channels
and an external subwoofer. The design allows for a high degree
of flexibility in deployment, including vertical, overhead, and
circular configurations, enabling the simulation of complex
”natural” acoustic environments and a diversity of applications
in artistic, commercial and research-driven projects.
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A Field-Matching Method for Sound Field Synthesis for Large Scale Sound Reinforcement Systems
Elena Shabalina
[Abstract]
There have been ongoing attempts to adapt existing Sound Field Synthesis methods for large scale sound reinforcement systems. In comparison to smaller indoor setups large scale systems have significant distances between secondary sources and often irregular loudspeaker layouts due to rigging conditions. These restrictions make the use of WFS-related unfeasible. In addition, prominent and frequency-dependent directivity patterns of concert loudspeakers obstruct the use of usual level and delay algorithms as they assume omnidirectional secondary sources.
The paper proposes an alternative method based on a Field Matching technique that bypasses the above-mentioned restrictions. Despite the typically low aliasing frequency of a concert sound system layout the Field Matching approach was shown to provide an accurate localisation and a stable virtual source across the audience area including front seats. The paper presents a theoretical background and a Matlab simulation along with listening tests results. Limitation and challenges of the method for live sound applications are discussed.
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Adjoint-based sound reinforcement in the time domain
Lewin Stein, Florian Straube, Jörn Sesterhenn, Stefan Weinzierl, Mathias Lemke
[Abstract]
Sound Reinforcement aims at synthesizing homogeneous
sound fields
for the entire audio bandwidth using Line Source Arrays.
These arrays
consist of electronically controlled loudspeaker cabinets with
different tilt angles. The determination of the optimal
geometric arrangement and electronic drive is an ill-posed
inverse problem. Recently, the authors presented a novel
adjoint-based approach which lives in the time domain as
an alternative solution method to corresponding problems.
We optimize acoustic sources to match a target sound field
subject to an objective function. For that, we solve the Euler
equations and the corresponding adjoint by means of
computational aeroacoustic techniques. The method can
determine appropriate driving functions and appropriate
positions of the sources for the synthesis of the desired sound
field. We will present validation examples including base
flows, e.g. wind, and thermal stratification for a
representative loudspeaker setup. It will be shown that the
method provides reasonable driving functions also for these
conditions.
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Full scale outdoor concert adaptive sound field control
Jonas Brunskog, Franz Maria Heuchel, Diego Caviedes Nozal, Minho Song, Finn T. Agerkvist, Efren Fernandez-Grande, Enrico Gallo
[Abstract]
Outdoor musical events in urban environments is of large cultural importance, but is also a source of noise, especially for non-participating neighbors. In an attempt to solve this, an Adaptive Sound Field Control (ASFC) system is under development. The purpose is to mitigate low frequency noise annoyance and simultaneously enhance the musical experience of the audience. The system uses a secondary array of loudspeakers, canceling out the sound in a dark zone. To do this, accurate estimates of the transfer function between the loudspeakers and the dark zone is necessary. A static version of the system has been tested in three events: a controlled pre-test, the KappaFutur Festival in Turino, and Tivoli in Copenhagen. These deployments are reported and discussed, with special focus on the KappaFutur Festival, where noise measurements where conducted in the neighborhood, supplementing the insertion loss measurement the ASFC. The performance of the ASFC system is shown to strongly depend on the complexity of the setting. In the final stage, the system needs to be adaptive in order to adjust for changing climate conditions. The adaptive parts of the system is under development, and will be tested in a series of events in 2019.
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Adapting transfer functions to changes in atmospheric conditions for outdoor sound field control
Franz Maria Heuchel, Diego Caviedes Nozal, Efren Fernandez-Grande, Jonas Brunskog, Finn T. Agerkvist, Peter Gerstoft
[Abstract]
The performance of active sound field control solutions is directly dependent on the accuracy of the measured acoustic transfer functions between the control loudspeakers and control areas. Outdoors, these transfer functions are affected by atmospheric conditions and their variation over time. In this work, we investigate strategies for transforming measured transfer functions to changing atmospheric conditions with the goal of adapting outdoor sound field control systems to such changes. Compared to active control methods based on adaptive filtering, such weather model based approaches do not rely on continuous sound pressure recordings inside the control area. We investigate different adaption strategies, one based on explicit delay compensation with effective speed of sound and an alternative based on machine learning. We train and test the adaption strategies against results of a large set of outdoor transfer function measurements in different atmospheric conditions.
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Organization:
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G. Elko, mh acoustics LLC |
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Bayesian inference in direction of arrival analysis using spherical microphone arrays
Ning Xiang, Stephen Weikel, Christopher Landschoot
[Abstract]
One challenging problem in acoustical applications using microphone arrays is determining the directions of arrival (DoAs) of multiple sound events. This work applies a unified Bayesian framework to address this problem in situations involving potentially multiple concurrent sound sources using spherical microphone arrays. It presents a two-level inferential problem of sound source enumeration and direction of arrival estimation. To this end, analytical models based on spherical harmonics are used to predict experimental data derived from spherical harmonics microphone arrays. The prediction is evaluated against the measured data in order to select the simplest such model that can adequately match the experimental data, thereby estimating first the number of sources, then their DoA information. This paper presents the analytic models, the two levels of Bayesian formulation, and analysis results to demonstrate the potential usefulness of this model-based Bayesian analysis for complex sound environments with potentially multiple concurrent sources. This paper discusses the impact of microphone array configurations upon DoA resolution. It additionally estimates performance of the method based on experimental results for simultaneous multiple sound sources, particularly in noisy or reverberant environments.
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Microphone array beamformer processing to reduce noise and reverberation
Gary Elko, Jens Meyer, Eric Diethorn, Steven Backer, Tomas Gaensler
[Abstract]
Hands-free audio communication systems that are designed to allow audio and speech
communication between remote parties are known to be sensitive to room reverberation
and noise, especially when the source is distant from the microphone. One solution
is to use an array of microphones to spatially filter the acoustic field so that
the direct sound field from the talker is emphasized while reverberation and background
noise is attenuated. It is well known that the maximum directional gain for an array of
microphones is N^2 where N is the number of microphones. The maximum
microphone array gain is only attainable with specific microphone geometries.
Typical gains obtained with realizable microphone arrays are lower than this maximum
and the loss can be significant. To increase the direct-to-reverberant ratio in a diffuse
field by 20 dB a classical delay-sum beamformer (maximum directional gain of N)
would need on the order of 100 microphones. Two algorithms are described that attempt
to address the slow growth in directional gain as the number of microphones increase by
replacing the standard linear microphone array beamformer with a parametric nonlinear
multiplicative processing beamformer that uses multiple overlapping and non-
overlapping beamformers to suppress noise and reverberation.
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The max-norm minimization in non-synchronous measurements
Liang Yu, Yi Fan, Jerome Antoni, Haijun Wu, Quentin Leclere, Baihua Yuan, Wei Kang Jiang
[Abstract]
Acoustic imaging from non-synchronous measurements of
microphone array can increase virtually the array aperture and
microphone density by sequentially moving an arbitrary
prototype array. It had attracted great interests recently, since it
beyond the fundamental limitation of Nyquist-Shannon sampling
theorem in the spatial domain (the working frequency that is
determined by the size and microphone density of an array).
The problem of archiving the non-synchronous measurements of
microphone array boils down to a matrix completion of a block
diagonal spectral matrix. In this paper, the max-norm is
investigated for the low complexity modeling of the spectral
matrix, and the corresponding Proximal-Point method for Max-
norm based Spectral matrix completion (PPMSMC) is proposed
for the matrix completion; Second, the rank, nuclear-norm and
max-norm has been investigated as complexity measures in the
context of spectral matrix completion, and their performances
are compared at different setups of frequency range, number
of sources and the increased virtual aperture size/density of
microphone array. An experiment of a High-Speed Blender
noise imaging is presented for validating the proposed methods.
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Microphone Array Method for Determining Noise Angular Energy Distribution on Building Envelopes
Miodrag Stanojević, Miloš Bjelić, Dragana Šumarac Pavlović, Miomir Mijić, Tatjana Miljković
[Abstract]
A method is proposed for determining traffic noise angular
energy distribution on building envelopes using the
microphone array as a source localization tool.
Measurement procedure is presented with the results
obtained by a nonuniform planar array developed for this
purpose with microphone positions optimized for traffic
noise measurements. The results are obtained in two
characteristic urban configurations - canyon and non-
canyon streets. Differences are commented and their
implications on the in-situ sound reduction index of the
façade element are explained when the element is
located in these distinct situations. The algorithms used for
sound source localization are CB in the frequency domain,
along with deconvolution algorithms DAMAS2 and CLEAN-
SC, and their performance is compared. The paper
addresses the problem of the computational complexity
of the algorithms which results in long execution times and
inhibits measurements on a larger scale. Various methods
for algorithm optimization are explored by varying
different parameters such as angular resolution, signal
length, various algorithm specific parameters, etc. The
sensitivity of output results on the variation of these
parameters is analyzed. The goal is to create a time
efficient measurement procedure which would enable
the acquisition of a larger data set covering various urban
terrain configurations.
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Modal Analysis for Damage Detection in Structures by Non-contact Measurements with a Commercial Microphone Array
Olaf Bölke, Jan Heimann, Joquin Garcia
[Abstract]
When monitoring critical structures, fatigue fracture, deformations, holes and much more are cases of failure which must be detected at an early stage. Changes in the modal parameters (eigenfrequencies, damping ratios, and mode shapes) of the structure allow conclusions to be drawn about the extent and location of the deterioration, subject to appropriate preliminary examination. Conventional measurement methods (i.e. acceleration sensors, laser vibrometers, etc.) for vibration analysis have the disadvantage that they can either ”detune” the vibration modes due to their own weight and/or require a long measurement time due to their punctual measurement. In contrast, the use of a suitable microphone array allows the high-resolution acquisition of the entire surface vibration covered by the array. Thus, the modal parameters of interest are determined by measuring the pressure fluctuations in the near field of the structure. A commercial acoustic camera with 120 microphones (Fibonacci120 by gfai tech GmbH) is used for this purpose. On the basis of artificially generated failure cases (load fracture, inhomogeneities, etc.) on application-oriented, large-area structures, a method for the detection of failure cases using a microphone array is demonstrated. As a reference, simulated vibration analyses, as well as vibration analyses measured by laser vibrometers, are used.
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Analysis of the Sound Field in a Room Using Dictionary Learning
Manuel Hahmann, Samuel Arturo Verburg, Efren Fernandez-Grande
[Abstract]
The sound field in a room is often modeled as a superposition of elementary waves, such as plane or spherical waves. These wave expansions provide a powerful means to interpolate or extrapolate the sound field within (and outside) the measurement domain. However, projecting the sound field of a large domain in a room into a planar or spherical wave base yields a high number of very elementary components. We examine the use of dictionary learning to find a set of more complex basis functions that are suitable to represent the sound field enclosed in a room. The resulting dictionary should be able to capture the dominant features of the sound field, and represent it using only a sparse set of functions. In this study, extensive measurements of the sound pressure in a room are obtained and used as a training set to learn a dictionary. We analyze the spatial properties of the learned dictionary, and compare it to simple elementary basis functions such as plane and spherical waves.
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Acoustic transfer admittance of cylindrical cavities in infrasonic frequency range
Paul Vincent, Dominique Rodrigues, Franck Larsonnier, Cécile Guianvarc'H, Stéphane Durand
[Abstract]
Demand for calibration at infrasonic frequencies has emerged in response to earth monitoring problems. The primary standard for sound pressure is defined through the reciprocity calibration method specified in the International Electrotechnical Commission (IEC) Standard 61094-2:2009. This method is based on the use of closed couplers and is routinely applied by the National Metrology Institutes for a large frequency range; however, infrasonic frequencies below 2 Hz have not been explored until recently. The acoustic transfer admittance of the coupler, including the heat conduction effects of the fluid, must be modelled precisely to obtain accurate microphone sensitivity. IEC 61094-2:2009 provides two standardised solutions for the correction of heat conduction. However, researchers have noted significant deviations between these corrections at low frequencies in plane wave couplers, indicating that one or both techniques incorrectly calculate the influence of heat conduction. In this paper, the limitations of the standardised formulations at infrasonic frequencies are identified and two alternative solutions are proposed. An experiment is also reported, which highlights the discussed limitations of the standardised formulations for acoustic transfer admittance, while also demonstrating the validity of the proposed alternative formulations at frequencies down to 0.04 Hz.
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Alternating Least Squares-Based Joint Estimation of RETFs and PSDs for Multi-Channel Speech Enhancement
Simon Doclo, Marvin Tammen, Ina Kodrasi
[Abstract]
The multi-channel Wiener Filter (MWF) is a well-known speech enhancement technique that can be used to improve speech quality and intelligibility of microphone signals recorded in noisy and reverberant environments. It is commonly assumed that (i) late reverberation and ambient noise can be modeled as a spatially diffuse sound field and (ii) the spatial coherence of the remaining noise is known a-priori. Using these assumptions, the MWF requires estimates of the relative early transfer functions (RETFs) of the target speaker and the power spectral densities (PSDs) of the target speaker, the diffuse and the remaining noise. Recently, we proposed a technique to jointly estimate these quantities by minimizing a model-based error matrix via an alternating least squares (ALS) method. In this contribution, we present extensive simulation results comparing the ALS method with a state-of-the-art reference method based on covariance whitening. We demonstrate the effectiveness of the ALS method in both stationary and dynamic acoustic scenarios by using the estimates in an MWF and evaluating its noise reduction and dereverberation performance with respect to the improvement in speech quality. Results show that the ALS method yields more accurate estimates than the reference method, especially in the presence of strong uncorrelated noise.
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Organization:
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K. Brandenburg, TU Ilmenau / Fraunhofer IDMT |
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Perceptual Aspects in Spatial Audio Processing
Karlheinz Brandenburg, Bernhard Fiedler, Georg Fischer, Florian Klein, Annika Neidhardt, Christian Schneiderwind, Ulrike Sloma, Claudia Stirnat, Stephan Werner
[Abstract]
Spatial audio processing includes recording, modification and rendering of multichannel audio. In all these fields there is the choice of either a physical representation or of perceptual approaches trying to achieve a target perceived audio quality.
Classical microphone techniques on one hand and wave field synthesis, higher order ambisonics or certain methods of binaural rendering for headphone reproduction on the other hand target a good physical representation of sound. As it is known today, especially in the case of sound reproduction a faithful physical recreation of the sound wave forms (”correct signal at the ear drums”) is neither necessary nor does it allow a fully authentic or even plausible reproduction of sound.
20 years ago, MPEG-4 standardized different modes for perception based versus physics based reproduction (called ”Perceptual approach to modify natural source” and ”Acoustic properties for physical based audio rendering”). In spatial rendering today, more and more the perceptual approach is used in state of the art systems. We give some examples of such rendering.
The same distinction of physics based versus psychoacoustics (including cognitive effects) based rendering is used today for room simulation or artificial reverb systems.
Perceptual aspects are at the heart of audio signal processing today.
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Perceived Quality and Plausibility of Room Reverberation in VR Reproduction from Measured Images and Acoustics
Luca Remaggi, Hansung Kim, Adrian Hilton, Philip Jackson
[Abstract]
Virtual reality (VR) systems have emerged as platforms for personal
interactive audio-visual media experiences. In order to have a real
world reference against which to evaluate the room acoustics
reproduced within VR, methods are needed to compare the virtual
experience with that of a real room. In this work, two pipelines were
developed for evaluating room reverberation over headphones within
VR, acoustically and visually. The acoustical pipeline involves
parameterisation of B-format room impulse responses via the
reverberant spatial audio object (RSAO) and subsequent binaural
rendering. The optical pipeline uses stereo 360 images to reconstruct
the room geometry and materials that are then translated into
binaural via two popular VR software development kits. Formal audio-
visual subjective tests were conducted to obtain ratings of perceived
quality and plausibility, using hidden anchors, for six captured rooms.
The results show a trade off between quality and interactivity,
meanwhile the virtual acoustics were confirmed as plausible
reproductions of the original rooms. Further work will examine
methods to enhance the interactivity with realistic spatial room
reverberation.
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Organization:
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F. Zotter, Inst. f. Elektr. Musik u. Akustik, KUG |
G. Tauböck, Österr. Akademie d. Wissenschaften, Inst. f. Schallforschung |
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On the Stability of Gabor Phase Retrieval
Matthias Claus Wellershoff, Rima Alaifari
[Abstract]
Phase retrieval refers to the problem of recovering a signal from phaseless measurements. Gabor phase retrieval, in particular, is concerned with reconstruction from the absolute value of the Gabor transform (be it discrete or not) and has applications in the time-frequency analysis of audio signals. From a mathematical point of view, phase retrieval (from frame coefficients) is a challenging problem as it has been shown to be unstable in infinite dimensional Hilbert spaces and severely ill-conditioned in finite dimensional spaces. However, it has also been shown that one can relax the classical stability regime through a so-called semi-global phase reconstruction and obtain a stability result for phase retrieval from the continuous Gabor transform in this setting. Recently, we were able to adapt the semi-global regime to the discrete case and to prove a promising stability result from the discrete Gabor transform. In this contribution, we survey selected highlights from recent research on phase retrieval from frame coefficients with emphasis on phase retrieval from Gabor measurements. In particular, we review results on semi-global stability of phase retrieval in the infinite dimensional case and present our result on the finite dimensional setting.
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Phase-Magnitude Relations and Phaseless Reconstruction for Time-Frequency and Time-Scale representations
Nicki Holighaus, Zdenek Prusa, Günther Koliander
[Abstract]
We discuss the phase-magnitude (PM) relations for the short-time Fourier and wavelet transforms which are derived from analyticity of the respective transform, when an appropriate analysis window or wavelet is used. For the short-time Fourier transform, the PM relations were previously studied by Portnoff and later rediscovered by Auger et al. Recently, it was shown that these relations can be combined with an adaptive integration scheme to enable signal reconstruction from magnitude-only short-time Fourier measurements that often performs on par or better than more expensive, iterative algorithms for phaseless reconstruction. For wavelet transforms, similar relations hold that can be employed for phaseless reconstruction as well. We recall the phase-magnitude relations and the resulting algorithms in both settings and demonstrate their application in the audio domain.
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Organization:
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M. Cobos, Universitat de Valencia |
W. Kellermann, Friedrich-Alexander University Erlangen-Nürnberg |
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Source Localization in Reverberant Rooms using Deep Learning and Microphone Arrays
Hadrien Pujol, Éric Bavu, Alexandre Garcia
[Abstract]
Acoustic source localization is a well-studied topic in array signal
processing, which
could benefit from the emergence of data inference tools. We
present our recent
developments on the use of a Deep neural network fed with
raw multichannel audio for
sound source localization in reverberating environments.
This paradigm allows to avoid the simplifying assumptions that
most traditional
localization methods incorporate. However, for an efficient
training process, supervised
machine learning algorithms rely on large sized, precisely
labelled datasets. There is
therefore a critical need to generate a large number of audio
data recorded by
microphone arrays in various environments.When the dataset is
simulated either with
numerical models or 3D sound-field synthesis, the physical
validity is also critical.
We therefore present an efficient tensor GPU-based
computation of synthetic room
impulse responses using fractional delays for image source
models. We also present
the use of physical 3D sound field synthesis for the learning
process on
microphone arrays. We discuss the advantages of this
reproducible and semi-automated
process, which allows to deal with arbitrary array geometries.
We also analyze the
localization performances of the proposed neural network fed
with this dataset, which
allows a precision as high as 1 degree in a reverberant room.
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Sound Quality Improvement for Speech Acquisition Based on Deep Learning and Harmonic Reconstruction with Laser Microphone
Shoji Ueda, Kenta Iwai, Takahiro Fukumori, Takanobu Nishiura
[Abstract]
A laser microphone has recently been focused on acquiring the
distant target speech without unnecessary sounds. It measures
the vibration of the object near the target sound source by
irradiating the object surface with the laser beam. However, the
speech acquired with this microphone degrades the sound
quality. For instance, the speech components at higher
frequencies are attenuated by vibration characteristics of the
object, and the speech is collapsed by stationary noise due to
lower power of the laser beam from the object. To improve the
sound quality of the degraded speech, deep neural network
(DNN) has recently been proposed. It is trained by using a set of
acoustic features extracted from the degraded speech and the
clean speech. However, the speech components at higher
frequencies are still attenuated after processed by DNN.
Therefore, we propose the method to reconstruct the harmonic
structure of the speech after processed by DNN. The notice
point is that the harmonic structure composes the higher
frequencies of the speech. The proposed method complements
the spectral amplitudes on positive integer multiples of the
fundamental frequency which are the components at
harmonics. We evaluated the effectiveness of the proposed
method though perceptual evaluation of speech quality.
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Performance Analysis of Audio Event Classification Using Deep Features under Adverse Acoustic Conditions
Irene Martin-Morato, Maximo Cobos, Francesc Ferri, Javier Naranjo-Alcázar
[Abstract]
Audio event classification has been traditionally performed by extracting standard features based on human perception, such as Mel-frequency cepstral coefficients (MFCCs). However, the trend followed in the last years is primarily based on the information provided by deep features, which are extracted from the responses to complex input patterns learned within deep neural networks. These have been shown to obtain, in general, better performance than the hand-crafted ones. In fact, deep features are known to provide good generalization properties to classify events not seen during training, and can even be extracted from raw audio data. Since the captured audio data is highly dependent on the acoustic properties of the auditory scene, it is important to assess the impact that adverse acoustic conditions have in the final classification performance. In this paper, we analyze the robustness of deep features under controlled acoustic conditions by simulating different degrees of background noise, reverberation and segmentation errors, as well as in a real-life scenario where more than one audio event can be present at the same time. Results show an acute degradation in the performance given the background noise and segmentation errors, which suggest room for improvement in terms of robustness to different scenarios.
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Joint Separation, Dereverberation and Classification of multiple Sources Using Multichannel Variational Autoencoder with Auxiliary Classifier
Shota Inoue, Hirokazu Kameoka, Li Li, Shoji Makino
[Abstract]
This paper proposes a unified approach for jointly solving
separation, dereverberation, and classification of mixed
sound
sources in microphone array observations. The proposed
method uses a frequency-wise convolutive mixture model
to
express the mixing process under highly reverberant
environments and the auxiliary classifier conditional
variational
autoencoder (ACVAE) to model the complex
spectrograms of
underlying sources. The parameters of these models can
be
simultaneously estimated based on the same criterion by
iteratively updating the separation matrix, the
dereverberation
filters, source-class probabilities and the power
spectrogram of
each source. Through experiments, we confirmed that the
proposed method outperformed conventional methods in
terms
of the separation performance and the source
classification
rate.
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A Neural Network Approach to Broadband Beamforming
Yugo Kuno, Bruno Masiero, Nilesh Madhu
[Abstract]
Beamforming techniques are commonly applied to signals captured by sensor arrays to enhance signals received from desired directions while reducing background noise and localised interference. Where the directions of the desired and interfering sources are known, this knowledge, combined with assumptions on the background noise characteristics, is used to derive the beamformer coefficients for each sensor. This derivation is usually obtained from an optimisation of the second-order statistics of the beamformer response, e.g., minimising the output energy while preserving the signals from desired directions. For broadband signals, like speech, the optimisation is often posed in the frequency domain. Beamformer coefficients are then independently derived for each discrete frequency, as an approximation of the true broadband response. Hereby, the complex inter- frequency interactions, e.g., due to windowing and spectral aliasing, are not modelled, leading to sub-optimal filter characteristics. These shortcomings can be overcome by data-driven approaches. By exploiting recent developments in deep learning, we propose a neural-network approach to generate a desired spatial response while implicitly accounting for the inter-frequency effects in the narrowband model. We test the proposed approach on synthetic and real-world data and compare the performance to standard (second-order) narrowband and broadband beamformers.
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Organization:
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B. Rafaely, Ben-Gurion University of the Negev |
G. Elko, mh acoustics LLC |
J. Meyer, mh acoustics LLC |
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The uncertainty of room impulse response measurements
Ingo Witew, Michael Vorländer
[Abstract]
The measurement of room impulse responses using two channel FFT correlation techniques is a standard approach to determine the acoustic properties of auditoria. Evaluating the uncertainty that is associated with this measurement method is challenging because the ”Guide to the Expression of Uncertainties in Measurements” originally requires establishing an algebraic relation between the system’s stationary in- and output quantities. 2-CH-FFT methods, however, are dynamic measurements as the impulse response is calculated from a microphone’s time domain signal, whose value at one time instant depends on the excitation’s time domain signal at previous time instants. Such measurement strategies don’t quite match the basic GUM approach. To reduce the complexity the dynamic in- and outputs of transfer function measurements are decomposed into quasi-stationary energy components. This allows investigating the uncertainty in line with the classical framework. On these grounds, general strategies are derived to determine the influence of background noise and system non-linearities in dynamic measurements. Second, these tools are used to consider the elements of the measurement chain and how they contribute to the uncertainty in measurements. The discussion concludes with an uncertainty budget for correlation technique measurements in rooms.
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Measurement of the four pole matrix of a sample in a transmission tube
Rene Boonen
[Abstract]
A transmission tube has been developed to measure the four pole matrix of
a sample of absorbing material. The electrical analogy of a sample in a
transmission tube can be considered as a T-equivalent circuit of three
impedances, two in series and one in parallel. As consequence, three
measurements with three different closing impedances needs to be carried
out. From these measurements, a set of equations will result from which
the three impedances of the T-equivalent circuit can be determined in
real and imaginary parts in terms of frequency. Once these impedances are
known, any four pole matrix, such as the transfer or the scattering
matrix, of the sample can be set up. The method has been validated by
measurements of several sample combinations in the impedance tube.
A good agreement has been established between transmission tube and
impedance tube measurements in a range of 40Hz-4kHz.
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Ultrasonic Remote Sensing for Precision Agriculture
Stuart Bradley, Mathew Legg
[Abstract]
Precision Agriculture makes use of quantitative measurements
as input to sophistcated farm management software. For
management of grazing animals, such as dairy cows, a key
input is pasture biomass, so that pasture is not over-grazed nor
have too much or too little application of fertilizer. Aside from
destructively cutting, drying, and weighing the pasture, current
measurement methods are slow and/or unreliable.
We describe a new ultrasonic sensor which is compact and low-
power, and which senses pasture properties remotely from a
moving farm bike or UAV. The sensor comprises co-located log-
spiral arrays of transmitters and receivers, giving high spatial
resolution transverse to the propagation direction without
excessive component counts. A pulsed, radar-like linear-FM
chirp and matched filter gives high along-axis spatial resolution.
With CLEAN image deconvolution, mm resolution is obtained
through the pasture layer. This methodology allows detailed
profiles of pasture density to be obtained at rates of 100 profiles
per second.
We present results retrieved via modelling of the complex
acoustic scattering occurring in the pasture layer.
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PyTTa: Open Source Toolbox for Acoustic Measurements and Signal Processing
William D'Andrea Fonseca, João Vitor Paes, Matheus Lazarin, Marcos Vinicius Reis, Paulo Mareze, Eric Brandão
[Abstract]
Python is a high-level programming language that has gained
strength in the international community. This mainly occurs
because it is open-source nature, that is, any user is entitled to its
use, whether for scientific, commercial or hobby purposes.
Another important feature is that Python is OS-independent. This
means that no matter what OS the code was written, it can be
run by any operating system. In addition, access to AD/DA
acquisition interfaces allows interaction with real systems for their
estimation and control. These features enable its use as a
powerful signal processing tool, whether for acoustics or general
purpose. The Federal University of Santa Maria (UFSM), in Brazil,
has created a group of developers, who are working
cooperatively on a toolbox for acoustic, audio and vibration
signal processing. The project is named PyTTa, standing for
Python in Technical Acoustics. It is object-oriented programmed,
enabling the easy use of variables, classes, and functions. The
collection of codes inside the toolbox creates an environment
that facilitates the acquisition/measurement, pre/post-
processing, I/O operations and plotting/documenting. This is a
continuing work that relies on the collaborative effort to provide
free access to information and work/research tools.
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A Theory for Stethoscope Acoustics
Maximilian Nussbaumer, Leyre Troyas Martinez, Anurag Agarwal
[Abstract]
Maximising the signal to noise ratio while considering
ergonomics and aesthetics is the key design challenge for
modern stethoscopes. In order to optimise the design, there is a
need for a well-validated model for the transfer function from a
source within the chest to the output signal obtained. Most variants of the stethoscope are air-coupled sensors. Here
we propose a new theory for the acoustics of this type of sensor,
which takes into account the coupling between the sensor and
the human chest. We have conducted a rigorous experimental campaign to
characterise the transfer function of the chest-stethoscope
system and have investigated the effects of key design
parameters. Our data confirms traditional findings on the
effects of bell geometry and diaphragm usage, but also
highlights the importance of the coupling between the sensor
and the chest, and reveals features of the transfer function that
are not captured by existing models. Our model employs a transmission matrix formulation and
discretises the system into lumped element components. It can
be used to inform design choices for acoustic, electronic and
dual-mode stethoscopes, opening up the possibility of an
optimum design that maximises the signal to noise ratio for a
desired application.
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Noise Intensity Prediction from Video Frames using Deep Convolutional Neural Networks
Leonardo Oliveira Mazza, José Gabriel Rodríguez Carneiro Gomes, Julio Cesar Boscher Torres
[Abstract]
Some closed-circuit television (CCTV) systems do not have microphones. As a result, sound intensity information is not available in such systems. We present a method to generate traffic noise level estimates using solely video frames as input data. To that end, we trained a fully connected layer on top of VGG16 (pretrained with imagenet) using a dataset that was automatically generated by a single camera with a mono microphone pointing at a busy traffic crossroad with cars, trucks, and motorbikes. For neural network training from that dataset, color images are used as neural network inputs, and true average noise levels are used as neural network targets. The trained neural network successfully tracked trending noise levels with correlation 0.597 despite their blindness to the data temporal properties. These results suggest that average noise level targets are sufficient for convolutional neural networks to detect noise generating sources within a traffic scene.
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Organization:
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H. Führ, RWTH Aachen |
P. Balazs, Acoustics Research Institute, Austrian Academy of Sciences |
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Frame Theory for Psychoacoustics
Peter Balazs, Nicki Holighaus, Thibaud Necciari, Diana Stoeva
[Abstract]
We will give a survey on the link between frame theory and signal processing tasks in psychoacoustics and biology. On the one side, the basic concepts of frame theory are presented. The goal is to reveal to scientists from psychoacoustics and biology how this mathematical theory can be a particularly useful conceptual background for them. Frame theory is a very active field in mathematics. It provides perfect reconstruction, stability, redundancy, and a signal-independent, linear inversion procedure, while still allowing a lot of flexibility. Those properties are in particular important to analyze any filter bank, thereby providing useful insight into its structure and properties. On the other side, basic psychoacoustic concepts are presented to stimulate mathematicians to apply their knowledge in this field. We will show motivating applications where the connection of theory with applications are particularly promising: perceptual irrelevance (i.e. how can we exclude those signal components that are not audible); as well as perceptually-motivated filter bank (i.e. how can we find the best time-frequency transform best adapted to human (or animal) hearing while still keep the frame properties mentioned above. This is a summary of the book chapter ’Frame Theory for Signal Processing in Psychoacoustics’.
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A Localization Algorithm based on Head-Related Transfer Functions
Maike Gerhard, Patrick Schillberg, Hermann Wagner, Hartmut Führ
[Abstract]
A localization algorithm based on head-related transfer functions (HRFTs) is introduced. The algorithm minimizes the distance to direction-dependent binaural signal spaces, in order to find an estimator of the actual sound source position. Numerical experiments with different target sounds (clicks, white noise, speech, rustling of leaves) and several signal-to-noise ratios, where the masker is unspatialized white noise, show that this projection algorithm outperforms template matching in nearly all experimental conditions. In particular, the algorithm is robust to changes of the emitted signal’s phase spectrum, unlike template matching. For white noise maskers, it is possible to compute SNR dependent estimates for the error probability in the task of discriminating two directions, based on the associated HRTFs. We present simulations that demonstrate the precision of the estimates. We show how these probabilities can be employed as a means to mathematically analyze HRTFs, in particular with the aim of predicting localization performance from the HRTF data set.
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Organization:
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G. Manley, Carl von Ossietzky University Oldenburg |
J. Christensen-Dalsgaard, University of Southern Denmark |
V. Singh, National Physical Laboratory, New Delhi |
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Evolution of Hearing in the Early tetrapods and Emergence of the Tympanic Middle Ear
Jakob Christensen-Dalsgaard
[Abstract]
During the 120 my from the origin of tetrapods to the origin of tympanic ears in the Triassic, auditory sensitivity must have been based on non-tympanic, bone conduction mechanisms. However, bone conduction is a loose term for several different modes of stimulation of the inner ear. To understand hearing with a non-tympanic ear, I focus on the simplest: that sound translates the head region and that this vibration is transduced by the inner ear. The efficiency of translation of an object by sound is determined by ka (the product of wave number and radius) Comparison of the sensitivity of animals without middle ear (snakes, salamanders, earless frogs, lungfish) to sound and to vibrations of the skull shows that most sensitivity can be explained by translation. Interestingly, simple translation is also the mode of human low-frequency bone conduction sensitivity (for frequencies resulting in a ka>1). Translation by underwater sound is the mode of stimulation of most fishes. It is therefore a straightforward assumption that this was the mechanism of hearing also in the early tetrapods, and assessment of the sensitivity can inform us of the gradual changes leading to the tympanic ear, especially in relation to sensitivity and directionality.
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Why did Solid Otoliths evolve in the Ears of Modern Bony Fishes?
Tanja Schulz-Mirbach, Martin Plath, Friedrich Ladich, Martin Heß
[Abstract]
Modern bony fishes (Teleostei), which comprise almost one-half of all extant vertebrate species, show a remarkable diversity in ear morphology, including otolith shape. Fish bioacousticians are still puzzling over the question of why solid calcareous otoliths with species-specific shapes evolved, while most ”non-teleost” vertebrates possess numerous tiny otoconia. This question is linked to (i) when and how often a switch from loosely aggregated material towards a solid structure-and from apatite to calcium carbonate-occurred during the vertebrate radiation and (ii) whether those character shifts are related to altered hearing. We constructed a hypothetical framework for otolith evolution by compiling the available information on the structure of otoliths and otoconia in > 160 species covering all major vertebrate groups. Solid teleost otoliths may have initially evolved as a selectively neutral by-product of other key innovations, and the teleost-specific genome duplication event may have facilitated their subsequent diversification. Differences in otolith mass and shape might have enabled the perception of different ranges of acoustic information and may have evolved during the diversification of auditory abilities. In some teleost groups, otolith morphology may have co-evolved with ancillary hearing structures, especially if ears are closely connected to a swim/gas bladder.
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Organization:
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M. Wahlberg, University of Southern Denmark |
V. Singh, National Physical Laboratory, New Delhi |
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Comparing pressure and intensity units for in-air and underwater hearing thresholds
Magnus Wahlberg
[Abstract]
Hearing thresholds are usually reported in decibel units, which gives conveniently compressed numbers with biological implications. However, the decibel scale causes confusion when in-air thresholds are compared to underwater ones. Not only are the reference pressures of the in-air and underwater scales different, but the interpretation of the thresholds also depends on whether we assume the ear is detecting the integrated sound intensity or squared pressure (that is, if the acoustic impedance difference between the two media should be compensated for). Here, in-air and underwater hearing thresholds from the literature on toothed whales, seals, marine birds and turtles are compared, both using decibel scales and linear pressure and sound intensity units. The interpretation of how sensitive an animal is to sound in air and underwater critically depends on the choice of units used to report hearing thresholds. Therefore, great care must be taken choosing the adequate units in hearing studies.
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Evolution of Cetacean Underwater Hearing Mechanisms: Similarities and Differences between Archaeocetes, Mysticetes, and Odontocetes
Sirpa Nummela
[Abstract]
Modern cetaceans have their origins in terrestrial mammals that started their aquatic life over 50 million years ago. The history of the earliest whales, the archaeocetes, is fairly well known, and in their skull anatomy one can follow their adaptation to life and hearing underwater, while the earliest ones of them were still able to live and hear on land, too. Mysticetes and odontocetes both stem from archaeocetes, but today these two groups live in very different acoustic worlds, having specialized in low and high frequencies, respectively. It is also apparent that they use different kinds of peripheral auditory mechanisms. It is currently under lively debate whether the ability to hear high frequencies was present already in late archaeocetes, with mysticetes then later specializing in low frequencies, or if the ability to hear high frequencies first appeared with the odontocetes. Sounds emitted by modern whales can be compared to their hearing characteristics, when these are available. Morphological data on the peripheral auditory structures can be used to predict hearing ranges of these animals. These questions will be reviewed and discussed in this presentation.
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The Tympanic Ear as an Efficient Underwater Sound Transducer
Jakob Christensen-Dalsgaard
[Abstract]
All groups of tetrapods have members that adopt aquatic lifestyles with adaptations also of their auditory system. Water is a high pressure, low particle motion medium, and the consequence is that an efficient underwater ear is sensitive to sound pressure. It is often stated that underwater hearing can work efficiently without a middle ear apparatus by bone conduction, but the sensitivity of such an ear is limited by the very low particle motion in water. A comparison of tetrapods ranging from totally aquatic (the clawed frog Xenopus laevis) and mostly aquatic (the red-eared slider Trachemys scripta) to mostly terrestrial (the cormorant Phalacrocorax carbo sinensis) show similar features. All have tympanic middle ears with an air-filled middle ear cavity. The eardrum vibration peaks at the resonance frequency of the middle ear cavity air volume and the eardrum is modified (plate-like). In all three species, the lowest threshold to underwater sound is at this peak frequency and is around 80 dB re 1 µPa. Sensitivity to sound pressure is slightly lower in water than in air, making underwater hearing much more efficient in terms of sound energy. Consequently, the slightly modified tympanic ears of these species are efficient aquatic sound receivers.
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Late Evoked Potentials in Porpoises Speaks Against a Per- Click View of Echolocation, but Suggest Potential for Electrophysiological Audiograms at Low Frequencies
Kristian Beedholm, Peter Teglberg Madsen, Michael Ladegaard, Peter Lloyd Tyack
[Abstract]
In echolocating animals, an open question is whether
returning echoes are evaluated on a click-by-click basis or
whether there is a running evaluation, incorporating
several clicks in shaping the animal’s perception of the
echo scene. During the buzz phase of small odontocetes,
such as porpoises, inter click intervals (ICI) may be as low
as 2ms, which would seem to preclude individual
processing of clicks. We studied the low-frequency
auditory evoked potentials (AEP) in a stationary porpoise
presented with artificial clicks at ICIs of 0.5-1s. There is a
persistent component in the AEPs occurring at latencies
out to at least 150 ms, and it seems highly likely that this
represents remnants of cortical processes, whereby 150ms
should form a lower limit to the ICIs at which click-by-click
processing of echoes could take place. Toothed whales
do therefore not process echo information before
emission of the next click. As an added bonus, we believe
that the relatively slow components of the AEPs
constituting the cortical contributions hold potential for
assessing the audiogram of odontocetes
electrophysiologically down to frequencies well below
what is usually thought possible with traditional amplitude
modulated tonal stimuli, often taken to be limited to
frequencies above several kHz.
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The Cormorant Ear – Adapted to Underwater Hearing?
Ole Næsbye Larsen, Magnus Wahlberg, Jakob Christensen-Dalsgaard
[Abstract]
Diving birds may spend several minutes underwater during foraging dives. However, surprisingly little is known about avian underwater hearing. We do not know their sensitivity or even if they respond to underwater sound. To help filling this gap we measured the audiograms of cormorants (Phalacrocorax carbo sinensis) and studied their ear anatomy. Wild-caught fledglings were anesthetized and their auditory brainstem response (ABR) to clicks and tone bursts was measured, first in an anechoic box in air and then in a large water-filled tank with their head and neck submerged 10 cm below the surface. The overall shape of their air-audiograms was like that reported for birds of the same size in air. The bandwidth and slopes of their audiograms were similar in air and water. However, in air the highest sensitivity was found at 2 kHz, whereas it was displaced towards lower frequencies underwater. These results suggest that cormorants have rather poor in-air hearing compared to similar-sized birds. Their underwater hearing sensitivity, however, is higher than what would have been expected for purely air-adapted ears. A possible reason for the poor in-air sensitivity is the special ear anatomy with the central eardrum shaped as a rigid piston like in turtles.
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Underwater Sound Localization using Internally Coupled Ears (ICE)
Leo Van Hemmen, Anupam Prasad Vedurmudi
[Abstract]
Internally coupled ears (ICE), where an interaural cavity acoustically couples the eardrums, are an anatomical trait present in more than half of all terrestrial vertebrates. The superposition of outside and internal pressure on the two eardrums results in internal instead of interaural time and level differences, which are keys to sound localization. Although ICE is primarily a low-frequency terrestrial adaptation, the African clawed frog Xenopus laevis is a fully aquatic species with a distinct air-filled canal between the ears. In water, the speed of sound is four times that in air. Unlike terrestrial animals with ICE, the Xenopus interaural cavity is also medially connected to the lungs. By modeling the inflated lungs as a Helmholtz resonator, we demonstrate their effect in improving hearing in a low-frequency regime, while simultaneously enhancing sound localization in a disjoint high-frequency regime, corresponding to the frequency ranges of male advertisement calls. In conjunction with its unique plate-like eardrums, we show how Xenopus uses its ICE-like interaural coupling to generate considerable internal level differences between eardrum vibrations and thus overcomes the challenges of underwater sound-localization. Taken together, the two arguments of Helmholtz resonator and plate-like eardrums show the potency of ICE and are interpreted accordingly.
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Organization:
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H. Wagner |
J. Hildebrand |
J. Christensen-Dalsgaard, University of Southern Denmark |
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Effects of traffic noise, land use types and ecotones on the distribution of resident birds in a natural reserve in the Ruhr area, Germany
Philipp Antoniou, Bryce Timothy Lawrence
[Abstract]
While being one of the most densely populated and formerly heavily industrialized regions of Germany, the Ruhr area still offers space for many diverse ecosystems whose preservation is regulated by law. Nevertheless auditory impairments, especially by traffic noise and due to spatial restrictions, are likely. This study examines the acoustic effects of traffic noise on the spatial distribution of birds in the natural reserve ëIm Siesack,’ Dortmund, guided by the research question: How does land use, road noise, and ecotones alter soundscapes and affect avian distribution?. To answer the research questions soundscape measurements were taken throughout the case study area (n=204) and analyzed utilizing five acoustic indices from the Soundecology package in R Studio. The results of the indices were used in a statistical analysis to investigate (1) correlations between the distance to roads and the amount of biophony and (2) the differences between biophonic soundscape indices in various land uses and ecotones. Significant correlations between distance to roads and the amount of biophony and significant differences between soundscapes in different land use types and ecotones were found. Consideration of the results as related to soundscapes and nature reserve policy in Germany are discussed.
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Adaption to Bristish Standards to identify Construction Site Activity noise sources with the Potential to Cause Stress to Giant Pandas
Murray Snaith, Alessandro Rodriguez, Brett Marmo
[Abstract]
Giant Pandas have a broader hearing range than humans and
have higher sensitivity in different frequency bands. Giant
Pandas are known to become stressed when subjected to
undesirable noise, therefore noise which humans may deem
unobtrusive has the potential to cause stress in Giant Pandas. As
Giant Panda breeding programmes are taking place in zoos
throughout the world it is important to understand the impact
that anthropogenic noise and vibration has on these mammals
and how it affects their wellbeing. This paper discusses the
adaptation from international standards on noise and vibration
to appropriately address the impact from construction site noise
on Giant Pandas; as well as inform the acoustic considerations
in the design of a new Giant Panda enclosure at Edinburgh Zoo
in Scotland.
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Organization:
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J. Davy, RMIT University |
C. Guigou-Carter, Centre Scientifique et Technique du Bâtiment |
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Comparison between prediction and measurement of sound attenuation associated to ventilation network elements
Catherine Guigou-Carter, François Bessac, Simon Bailhache
[Abstract]
The AcouReVe Project (2015-2018) aimed to improve the knowledge and the quality of acoustic calculation in ventilation ductworks. Such calculations are based on simplified models and the main issue is the input data. For each component of the ductwork, acoustic insertion loss and/or sound generation due to air velocity has to be known. The present work concentrates on sound attenuation across different ventilation network elements, such as bends, branches and manifold. First a prediction method is described; it involves a multi-modal model pour elements associated to guided waves propagation and a finite elements model for complex elements. Measurements have been performed at CETIAT laboratory on different elements; the experimental setup is described along with the different evaluated configurations. The comparison between the measured results, the predicted results as well as the results from simple standard formulas applicable for the considered elements is presented and discussed in details.
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Sound Transmission through Double-glazed Window: Numerical and Experimental Analyses
Chaima Soussi, Mathieu Aucejo, Walid Larbi, Jean-François Deü
[Abstract]
The domestic windows in the building facade has a significant
role in sound insulation against outdoor airborne noise.
According to the standards, the prediction of their acoustic
performances is carried out in laboratory using two reverberant
rooms. It’s well known that below the Schroeder frequency, the
presence of these rooms can affect the measurements due to
their modal behavior. In this work, various 3D vibroacoustic
numerical configurations are studied to discuss the methods
proposed by acoustic standards. In particular, we analyze the
influence of (i) the presence of the acoustic cavities and (ii) the
diffuseness of the acoustic field, on the sound reduction index of
building elements. Moreover, due to the complexity of the
problem (geometry, materials and boundary conditions), an
accurate numerical model of the window should be
developed. This model, based on the Finite Element Method, is
built through a confrontation with experimental tests. For this, a
modal analysis is carried out to characterize the vibrational
behavior of the structure (eigenfrequencies, modal shapes and
modal damping). Those parameters will be used to design an
equivalent simplified model of the window, which will enable to
carry out parametric analyses.
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Sound Transmission through Aluminum Framings of Window, Door and Façade Systems
Yihe Huang, Fangliang Chen, Tejav Deganyar, Anselm Boehm, Rebeccah Grotemeyer
[Abstract]
Recent environmental regulations bring up higher
standards on acoustic comfort. Mitigating the noise
transmission from façade exterior to interior is rising as an
important issue in building envelop designs. Modern
façade systems consist of aluminum frames and high-
performance multiple layer of insulating or laminated
glass. Thus, the overall sound insulation of such systems
depends not merely on glasses but might more on frames
and the coupling between them. Particularly when the
STC of glass reaches 40 or higher, sound path through the
framing elements becomes more critical. However, this
crucial issue has not received enough investigation and
no satisfactory approach has been developed in the
literature yet. To address this issue, the coupling between
aluminum frames and glazing infill was carefully studied,
and an extended model has been accordingly
developed. A series of laboratory tests with different
frames were conducted to validate the predictions. As an
application of this research, a design-orientated
simulating platform has been developed to provide quick
estimations of the sound insulation performance of
window and façade systems.
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Investigation of rebuild repeatability issues in laboratory impact testing
John Loverde, Wayland Dong, Erik Holmgreen, Scott Bergquist
[Abstract]
Materials such as floor coverings, screeds, and resilient
matting are commonly applied on top of a floor-ceiling
assembly to improve the impact noise insulation. Since the
test measures the insulation of the entire assembly, it can
be difficult to isolate the effect of these top-side
components from the base structural assembly. The
authors previously reported on a laboratory test program
to evaluate the repeatability in the laboratory when
rebuilding assemblies [”Investigation of Extended
Repeatability Conditions in Laboratories” ICSV25 (2018)].
As a continuation of that program, assemblies and
products from historical tests were retested, and the
resulting repeatability was poor. Investigation determined
that the cause of the discrepancies was variation in the
base assemblies, not the top-side materials. The
techniques and results are discussed, and the effect on
the interpretation of laboratory testing is evaluated.
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Uncertainty quantification of the diffuse sound field assumption in structure-borne sound radiation predictions
Edwin Reynders, Pengchao Wang, Geert Lombaert, Cédric Van Hoorickx
[Abstract]
When predicting the radiation of structure-borne sound into a room, it is often assumed that the generated sound field is diffuse. A diffuse field is by definition a random field: it represents a conceptual ensemble of rooms with the same volume and total absorption, but otherwise any possible arrangement of boundaries and small objects that scatter incoming sound waves. Adopting a diffuse sound field model therefore inherently introduces uncertainty on the computed results. This uncertainty can be important, especially at the lower frequency end of the spectrum. In this work, practical formulas are derived for predicting not only the mean, but also the variance of energetic level quantities, such as the band-integrated spatially averaged sound pressure level, in a diffuse sound field caused by a mechanically excited structure. The formulas are first validated in a simulation study, and then applied for predicting the sound pressure level in a room caused by impact excitation of plate structures. Both the average sound pressure level and its standard deviation can be well predicted. This information is essential e.g. for judging at which frequencies the mean diffuse field estimate is sufficiently accurate for the application at hand.
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Numerical Investigation of the Reverberation Method for Measuring the Total Loss Factor of Plate-Like Structures
Yasutomo Yamasaki, Naohisa Inoue, Tetsuya Sakuma
[Abstract]
The total loss factor measurement by the reverberation method
is often performed to estimate the boundary conditions of
vibration fields of plate-like structures. In order to clarify how
measured values involve discrepancies from the true values, this
paper presents a numerical model that simulates the
reverberation method to determine the loss factor in a bending
vibration system of a glass plate supported with an elastic
material. As a reference, the total loss factor of a finite plate
system is theoretically calculated based on the diffuse vibration
field assumption, where the random-incidence vibration
absorption coefficient on the support edge is given for the semi-
infinite plate terminated by the mechanical and moment
impedances. Subsequently, the finite element analysis is
performed for a rectangular plate with the impedance
boundary, and impulse responses are calculated with different
arrangements of excitation and receiving points. Comparing
the theoretical and numerical results, the validity of the
measurement procedures is generally confirmed, however, it is
demonstrated that the measured loss factors tend to become
slightly higher than theoretical ones. Additionally, measured
results on real window systems are discussed briefly.
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Simplified prediction of the vibration reduction indices of double wall junctions
Arne Dijckmans, Lieven De Geetere, Charlotte Crispin
[Abstract]
Heavyweight cavity walls are commonly used as party wall between row houses and between dwellings in apartment buildings in Belgium. The double wall junctions complicate the prediction of flanking transmission. In this paper, empirical rules for the vibration reduction indices of rigid double wall T- and X-junctions are proposed. They are determined from Monte Carlo simulations carried out with a statistical model that was previously validated with laboratory measurements. In general, a double wall junction can be replaced by an equivalent single wall junction in flanking transmission models. The EN ISO 12354 approximation of a double wall T-junction by an equivalent single wall X-junction proves justified. For double wall X-junctions, the thickness of the equivalent single walls should be either the thickness of a single leaf or the total thickness of both double wall leafs, depending on the transmission path of interest.
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Sound Insulation Prediction of Single and Double CLT Panels
Fredrik Ljunggren
[Abstract]
Cross laminated timber (CLT) as building material has
rapidly gained in popularity in recent years, and thereby
also the demand of engineering tools to predict the sound
insulation through CLT walls. The sound reduction through
walls of single and double panels is predicted and the
results are compared with those found in the literature as
well as with a series of measurements conducted in a test
mock-up of two rooms. Various configurations are
considered where the plate thickness, addition of plaster
boards and cavity distance (in case of a double panel)
are altered. The prediction of single panel walls is based
upon an established model (by Sharp) for homogenous
materials, based upon theory and empirical data. For
optimized application to CLT constructions specifically, the
model is modified to some extent. The double panel walls
are analytically modelled under the presumptions of no
mechanical connection between the two panels and a
cavity filled with mineral wool. The model make use of the
results from the developed model for individual panels.
The accuracy of the predicted sound insulation is
estimated to be within a couple of dB’s concerning the
weighted sound reduction indexes from 100 and 50 Hz.
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Determining the airborne sound insulation improvement of thermal cladding systems in combination with heavyweight exterior walls
Claire Churchill, Maximilian Neusser, Simon Hinterseer
[Abstract]
Prediction of the sound reduction index (SRI) of modern thermal insulation systems for exterior walls is considered. These products all have in common a lightweight insulating layer of primarily thermal function (which may consist of a porous or non-porous material), and a weatherproof finish; the systems measured in the laboratory included a grout and skim finish or a prefabricated cladding system. In all cases, a typical heavyweight exterior wall was cladded. Modern examples of such systems are WDVS (Wärmedämmverbundsystem) and curtain walls (Vorhangfassaden). In their simplest form, such cladding systems can be modelled as a spring-mounted mass. However, Austrian standards demand that the spring-mounted layer be combined with structural fixings. In the examined cases sparse distributions (up to 12 connections/m^2) of point connections were used. The SRI of simplified and typical thermal cladding systems (i.e. with and without structural fixings) were measured in an accredited laboratory facility and these results were compared with a calculation method to determine SRI. The possibility to use established methods to determine the coupling loss factors in such combined systems was considered. The structural wall is of heavyweight construction, therefore the SEA problem can be simplified, in many cases, to a primary path analysis.
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Determination and optimization of sound insulation capabilities of geometrically complex walls
Elias Perras, Chuanzeng Zhang, Jihao Chen, Zhijiang Ji
[Abstract]
Several methods have been developed in the past to calculate the sound transmission in building acoustics. However, efficient determination and optimization of the sound insulation capabilities of geometrically complex walls like brick walls or layered wall structures remain a challenging task. In this work, a fully coupled fluid-structure interaction model, based on the frequency-domain spectral element method (FDSEM), is proposed to determine the sound insulation of arbitrary wall structures. A virtual sound measurement laboratory following the guidelines of ISO10140 is designed and used. The calculated frequency responses agree well with the experimental results. Therefore, this method can be used to evaluate the performance of the wall materials and structures in a cost-efficient way. Furthermore, contrary to other methods based on the plate theory, this method is more flexible and avoids many essential simplifications, which could restrict the applicability of the methods in the wall structure optimization. The present method is implemented in a multi-objective optimization procedure based on a genetic algorithm to optimize various wall structures (e.g. novel brick walls, multi-functional laminates), and the results are compared with that by other numerical and analytical methods.
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Prediction methods and evaluation of high sound insulations
Maarten Luykx
[Abstract]
When consulting pop venues or cinema’s with critical spaces
like dwellings nearby, high sound insulations will have to be
realized. In an early stage of the design the related structural
consequences will have to be known very stringently and
reliable, because adaptations afterwards are usually not
possible. In a recent situation with dwellings on top of a pop
venue sound reductions of D’nT,A,house>95 dB had to be
realized. Within the Peutz office several prediction methods for
sound insulation are being used during the design, such as ISO
12354-2 and several own practical methods developed within
Peutz, based on more than 65 years of experience. For this
project specific high pressure vibration insulated mountings had
to be designed and implemented. In order to evaluate the
prediction methods used and to prove that the high sound
insulations that were predicted have actually been realized,
measurements after delivery have to be performed. In this case
a special measurement setup was designed using 18
subwoofers and a noise level of 125 dB(A) house spectrum
inside the pop venue, in order to measure and prove that
extremely high sound insulations of more than D’nT,A,house=99
dB towards the dwellings had been realized.
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Comparison of transmission loss prediction using condensed equivalent plate models
Arasan Uthayasuriyan, Fabien Chevillotte, Luc Jaouen, Dimitrios Chronopoulos, Emmanuel Gourdon
[Abstract]
Equivalent plate models are commonly used in order to
simulate multilayer system as one single layer. This kind of
approach enables for instance to reduce the mesh size
and thus the number of degrees of freedom in finite
element models. Usual condensed model rely on thin
plate theory and are obviously limited when considering
thick multilayers. In this work we present a comparison of the transmission
loss prediction using different analytical models of solid
layers: thin plate, thick plate (first order shear deformation)
and 3D modelling (3D linear elasticity) . Doing so,
analytical limits for the validity of thin and thick plate
theories are derived from wavenumber analyses.
Applications to real world cases are presented.
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Organization:
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J. Davy, RMIT University |
C. Guigou-Carter, Centre Scientifique et Technique du Bâtiment |
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Components of Variation in Reverberation Time Measurement – Part 2: Field Testing Rooms of Heavyweight Construction
Bill Whitfield
[Abstract]
This paper uses analysis of variance (ANOVA) and a specific design of experiment (DOE) and construction choice to isolate the component of variance associated with the part (or room) being measured along with individual contributions of measurement uncertainty from the measurement system and the person making the measurement.
It demonstrates how the gauge repeatability and reproducibility (GRR) technique can be used to identify the variability in reverberation time measurement over the frequency range 100-3150Hz as and individual part of the process associated with field sound insulation testing in the UK.
The experiment uses the measurement of reverberation time associated with the ëinterrupted source method’ from EN ISO 354: 2003.
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Airborne Sound Transmission Modeling of Walls based on Random Point Process Theory
Cédric Van Hoorickx, Edwin Reynders
[Abstract]
A new method is presented to compute the diffuse sound transmission through walls that are either modeled deterministically or are taken to carry a diffuse vibration field. The fact that the natural frequencies can be considered to form a sequence of random points on the frequency axis makes this problem amenable to analysis by random point process theory. Furthermore, the method makes use of the fact that for any diffuse subsystem the statistics of its undamped eigenvalues and mode shapes saturate to universal distributions. The mode shapes saturate to Gaussian random fields, with a spatial covariance function depending only on wavelength and distance. The statistics of the local spacings between the eigenvalues saturate to those of a Gaussian orthogonal ensemble (GOE) matrix. This stochastic modelling approach is computationally very efficient as illustrated for a calcium silicate wall and a double glazing.
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Prediction of the sound transmission loss of two-dimensional periodic structures with a hybrid framework
Carolina Decraene, Edwin Reynders, Geert Lombaert
[Abstract]
When considering a room-wall-room system to simulate a
transmission suite in an acoustic laboratory, the direct sound
transmission in the mid-frequency range can be obtained using
a hybrid framework. The wall is then modelled deterministically
as finite size effects and modal behaviour are still important,
while the rooms carry a diffuse field and are modelled as
stochastic subsystems. The finite element method is usually
employed to compute the modal behaviour of the wall. At
higher frequencies the computational cost then increases
significantly as a finer mesh is required. For this reason, an
alternative approach was recently developed, which allows to
replace the finite element model by an analytical model for
finite-sized thick and layered walls. However, the application of
this approach is limited to layered structures such as sandwich
panels or double glazing. In the present work, the efficient
modelling of more complicated building elements which exhibit
spatial periodicity, will be considered by invoking periodic
structure theory. As a first application, the sound transmission loss
of a brick wall is examined and compared to the predictions of
a thick wall model with equivalent material properties and to
experimental data.
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Acoustic topology optimization of the material distribution on a simply supported plate
Jan Van Den Wyngaert, Mattias Schevenels, Edwin Reynders
[Abstract]
Lightweight plates typically have a low transmission loss due to
their low mass. Since coincidence typically plays a role at high
frequency, the TL can be increased with roughly 6 dB by
doubling the thickness of the plate. By distributing the additional
material in a non-uniform manner, a higher increase in TL can
be achieved in a limited frequency band due to modal
behavior. A good material distribution strategy is needed to
reduce the transmitted sound in a certain frequency range.
Here, the optimal material distribution at a single frequency and
in a frequency band is investigated for a small simply-supported
plate. The TL is modelled within the hybrid Finite-Elements-
Statistical-Energy-Analysis framework to achieve a low
computational cost while maintaining a high prediction
accuracy. Using mass distribution, the TL is increased by shifting
eigenfrequencies of the plate and creating stop bands. This is
achieved by locally adding mass that disturbs the mode shapes
by altering the bending stiffness of the plate. For a 1x1m2
PMMA-plate, an increase up to 33 dB in the TL is achieved for
the single frequency case and between 3 and 30 dB for a 50 Hz
frequency band.
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Adaptation of the Equivalent-Fluid Model to the Additively Manufactured Acoustic Porous Materials
Kamil C. Opiela, Tomasz G. Zielinski
[Abstract]
Recent investigations show that the normal incidence sound absorption in 3D-printed rigid porous materials is eminently underestimated by numerical calculations using standard models. In this paper a universal amendment to the existing mathematical description of thermal dispersion and fluid flow inside rigid foams is proposed which takes account of the impact of the additive manufacturing technology on the acoustic properties of produced samples. The porous material with a motionless skeleton is conceptually substituted by an equivalent fluid with effective properties evaluated from the Johnson-Champoux-Allard-Pride-Lafarge model. The required macroscopic transport parameters are computed from the microstructural solutions using the hybrid approach. A cross-functional examination of the quality (shape consistency, representative surface roughness, etc.) of various periodic specimens obtained from additive manufacturing processes is additionally performed in order to link it to the results of the multiscale acoustic modelling. Based on this study, some of the transport parameters are changed depending on certain quantities reflecting the actual quality of a fabricated material. The developed correction has a considerable influence on the predicted value of the sound absorption coefficient such that the original discrepancies between experimental and numerical curves are significantly diminished.
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Sound Insulation of Monolithic or Laminated Single- and Double-Glazing Panels
Fangliang Chen, Yihe Huang, Tejav Deganyar
[Abstract]
Sound transmission characteristics of single or multiple panels as sound insulation components in buildings have been studied extensively, and numerous models have been developed over the past few decades. Though most of them have been validated individually with specific test results in their developing phases, evidences showed that considerable discrepancy exit from one model to another. By applying many existing models onto glazing panels such as windows, doors or building facades, the authors found that most of them unable to accurately predict their sound insulation performance. Apparent discrepancy up to 3-5 dB in terms of OITC/STC index were even found between the predictions by commercial software and those well recognized test results. Obviously, more accurate theoretical models are highly demanded in industry to guide the design of real practice. To address this urgent need, revisits of most existing theoretical models were made by the authors, and extended models are accordingly developed and presented in this study in order to provide more accurate predictions on the acoustic performance of monolithic or laminated single- and double-glazing panels. Test results collected from literature as well as conducted by the authors are further provided to validate the proposed models.
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Organization:
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M. Schneider, Hochschule für Technik Stuttgart |
F. Morandi, University of Bologna |
B. Zeitler, Hochschule für Technik Stuttgart |
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Floor Impact Sound Insulation of the Six-story Wood-frame Model Building
Atsuo Hiramitsu, Ryuta Tomita, Hirakawa Susumu, Masayoshi Sato
[Abstract]
In Japan, the Act on the Promotion of the use of Wood for Public Buildings etc. was enforced in 2010. As a result, the number of constructions using the timber structure became more popular for the buildings. However, the performance of the floor impact sound insulation in the timber construction buildings is often concerned as worse than the concrete constructions. In this paper, an experiment was carried out on a newly built wood-frame six stories building to investigate the performance of the floor impact sound insulation on five different floors; CLT panel floor, stressed skin panel floor, LVL, I-joist floor and chord truss floor. The result showed that the stressed skins panel floor has the highest performance. The paper also reports the floor vibration performances and sound insulation of separation wall in the timber construction building, and proposes an improvement plan for the better floor impact insulation performance in the timber construction buildings.
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Hybrid joist floor constructions. Evaluation of measurement results.
Anders Homb
[Abstract]
Lightweight building systems in general suffer from limited sound insulation, especially at low frequencies. Still, theoretical models have severe limitations regarding prediction of the impact sound insulation and the design is to a high extent based upon experience and measurements. Recommendations in Norway is including the frequency range down to 50 Hz, which means to include the spectrum adaptation term CI,50-2500 and C50-5000. A possible solution to improve the properties at low frequencies is to add mass and stiffness. Lightweight building systems combining wood and cement-based materials, we refer as hybrid constructions.
The paper will present analysis of relevant measurement results of hybrid joist floor constructions from laboratory and field objects. The paper will discuss possible improvements of the floor, but also take the flanking transmission into consideration. From the ”Silent Timber Build project”, some relevant laboratory measurement data exist. Due to the impact sound insulation focus, some apartments with hybrid joist floors have been built and data collected. This paper is a part of a project at SINTEF Building & Infrastructure aiming to develop robust solutions also involving HVAC components inside the partition structure. Further progress in this project will include new laboratory measurements and a fire test.
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Application of elastic interlayers at junctions in massive timber buildings
Stefan Schoenwald, Niko Kumer, Sebastian Wiederin, Norbert Bleicher, Bernhard Furrer
[Abstract]
The application of resilient interlayers at junctions in massive timber buildings was investigated in a parametric study in the Laboratory for Acoustics at Empa. Flanking sound transmission across a junction of a floor with an exterior wall was investigated for different construction details with and without interlayers, with rigid and decoupled connectors as well as with and without additional load. The performance of the interlayers is discussed by comparing the apparent airborne and impact sound insulation between two rooms one on top of the other. The benefit of the elastic interlayers and connection variants is hereby gauged against other additional measures, for example wall linings, suspended ceilings, floating floors required to achieve equal sound insulation and to fulfil acoustic requirements.
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Numerical simulation of CLT floors and comparison with empirical predictive models
Marco Caniato, Federica Bettarello, Andrea Gasparella
[Abstract]
Wooden constructions are becoming more and more present in the global building market. They are built very quickly, combining structural and thermal performance in a single layering. They are industrially produced and then assembled on site to minimize building yard problems and increase construction speed. In this scenario, acoustics is often not addressed or is delegated to additional layers that are often inserted without a studied ratio. Many studies have been carried out to predict the acoustic behaviour of walls and floors using numerical or empirical methods. Recently, by means of validation with in situ measurements, the acoustic behaviour of raw wooden partitions has been defined with empirical or semi-empirical laws. In this work, a new method for the determination of the floor noise of raw CLT floors comes through the use of numerical simulation with transfer matrices. Then, the results are compared to the available empirical forecasting models and real measurements.
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Development of a quick and non-invasive measurement method for the extraction of the dispersion relation in CLT plates for the evaluation of the elastic parameters
Arved Thies, Federica Morandi, Luca Barbaresi, Massimo Garai, Jörn Hübelt, Niko Kumer
[Abstract]
This contribution discusses a quick and non-invasive method for the extraction of dispersion relations in CLT panels, with the aim of providing an estimate of the elastic parameters of the plates starting from the fitting of the experimental dispersion curve to the theoretical models.
Measurements were performed at the Stora Enso CLT mill on panels that have different thickness and orthotropic ratio. An instrumented hammer was used to excite the plate and accelerometers were used to acquire the impulse responses. Thanks to the reciprocity principle, the measurement points were kept fixed and the hitting point was moved; this allowed to reduce the measurement time and to decrease the number of accelerometers that must be attached.
The dispersion curves were evaluated using the Phase Shift method and the experimental results were fitted with the theoretical model for acoustically thick plates; finally, the elastic parameters were estimated.
The results give slightly low estimates of the Young’s modulus and a fairly good estimate of the shear modulus, while Poisson ratio that optimize the fitting are ”effective” values and not realistic values, as confirmed by recent literature. The precision of the hits is also analysed, while a reproducibility study is left for further developments.
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Organization:
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B. Rasmussen, SBi, Aalborg University (AAU-CPH) |
J. Jeongho, Fire Insurers Laboratories of Korea |
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Survey method for rubber ball impact sound
Jeong Jeongho
[Abstract]
Rubber ball impact sound was standardized in ISO standard as for precision and engineering method in ISO 10140 series and ISO 16283-2. Also, single number quantities for rubber ball impact sound is standardizing. In order to improve low-frequency impact sound isolation performance, quality control in a construction site is one of a necessary point to get reliable isolation performance. A survey method for airborne and light-weight impact sound was regulated in ISO 10052. However, a survey method for rubber ball impact needs to be proposed. To proposed survey method for rubber ball impact sound, the proposed method should be correlated well with measurement results which were measured with precision or engineering method. In this study, field measurements were conducted using KS F 2810-2 with five impact positions and five receiving points in Korean apartment buildings and the results were compared with survey method results which are extracted from the field measurement results.
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Rain Noise
Brian Donohue, John Pearse
[Abstract]
In this paper we discuss the requirements of the generic
international standard (ISO, BS EN ISO 10140) for testing of sound
transmission through sample roofs exposed to simulated rainfall
and of lessons learned during a recent test program. The test
data forms the basis for calculating in-situ sound levels in rooms
beneath the roof and we discuss the differences in sound
produced by simulated rain to that of natural rain. The
differences in impact velocity and raindrop distribution between
simulated and natural rain are key factors that are not
addressed by the Standard. In addition, an optional
normalisation test using a pane of glass is included, for the
explicit comparison of products tested and as quality control for
test laboratories. As this test returns a lower sound intensity value
it has wrongly been used by manufacturers as the basis for
calculating room sound levels. The Standard does not specify
whether the normalisation test should be carried out as a
skylight or as glazing but the two tests have different
requirements. Being optional and intended for inter-lab
comparison suggests that the normalisation data should not be
released to clients as it is misleading and thus should be
excluded from reporting.
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Ratings and classifications for high-frequency impact noise isolation
Wayland Dong, John Loverde, Jochen Scheck
[Abstract]
A two-rating method of evaluating impact noise isolation has been proposed by two of the authors [LoVerde and Dong, J. Acoust. Soc. Am. 141, 428 (2017)], in which low and high-frequency components are evaluated independently. Numerous other researchers have investigated the issue, and although the details differ, there has been some convergence of ratings and classifications for low-frequency impact noise. For the high frequencies, most have assumed that the existing IIC/Ln,w ratings are appropriate, while the authors have proposed high-frequency ratings which are based on the existing ratings but with a limited frequency range of 400-3150 Hz. Analysis of impact data from floor assemblies common to North America shows that the proposed ratings evaluate high-frequency impact sources and mitigation measures better than the existing ratings. The analysis is extended to include data from German floor systems and heavyweight and lightweight stairs. The ratings are compared and possible classifications are suggested.
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Airborne sound insulation in dwellings – single numbers weighted from 50-3150 Hz correlated to Swedish questionaire surveys
Christian Simmons, Fredrik Ljunggren
[Abstract]
In course of work with the draft standard for sound classification of dwelling (ISO/DIS 19488), it was vibrantly debated whether 50 or 100 Hz is the proper lower frequency limit when evaluating airborne sound insulation between dwellings. Although 100 Hz is used in regulations in most countries, there is an ongoing interest in including lower frequencies. In Sweden, evaluation from 50 Hz became mandatory already in 1999 wherefore unique long term experiences from the field are available. In this paper, field measurements are compared to subjective ratings given by the residents. The questionnaire surveys were distributed in total 46 building objects with light weight or heavy concrete walls and floors. Several single number quantities in ISO 717-1, were correlated against the resident’s ratings. The statistical evidence for a 50 Hz limit was found to be small considering all buildings, but analyzing lightweight buildings only show an importance of using 50 Hz. We recommend to measure from 50 Hz when goodsound protection against a broad variety of sound sources is at stake, including music at low frequencies. If protection against disturbing speech is the only concern, measuring from 100 Hz is enough.
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Is noise annoyance from neighbours in multi-storey housing associated with fatigue and sleeping problems?
Birgit Rasmussen, Ola Ekholm
[Abstract]
Danish Health and Morbidity Surveys including questions about neighbour noise and traffic noise have been carried out each 3-5 years since year 2000, the latest one in 2017. In all five surveys, the neighbour noise annoyance was higher than annoyance from traffic noise. In 2017, 14,022 respondents (response rate 56%) completed the self-administered questionnaire, out of which 3,893 adult Danes lived in multi-storey houses. Noise annoyance was assessed by asking the respondent about noise annoyance from neighbours and traffic, respectively, in their home during the past two weeks. Fatigue and sleeping problems during the past two weeks were assessed with a checklist that included eight different health outcomes (including the two mentioned). In total, 36% reported having been very/slightly bothered by neighbour noise and 22% by traffic noise. Noise annoyance from neighbours was strongly associated with both health outcomes. For example, individuals being very annoyed by noise from neighbours had 2.91 times higher odds of being very bothered by fatigue. Similar associations were also observed for traffic noise annoyance. Although, causality cannot be established in the present cross-sectional study, it is concluded that noise annoyance from neighbours is strongly associated with both fatigue and sleeping problems.
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Perceived noises in your residence: which one annoys the most?
Talita Pozzer, Rodrigo Silva Motta, Elaine Lemos, Débora Leite, Iara Cunha
[Abstract]
The Brazilian standard NBR 15575 establishes performance
criteria for residential buildings, including acoustic performance
requirements for airborne sound insulation of internal walls,
facades and floors, as well as for impact sound level of floors.
The standard also specifies non-mandatory sound pressure
levels in the dwelling from service equipment for informative
purposes. This study aims to investigate which type of noise
source is the most uncomfortable for Brazilian dwellings
occupants. In order to reach the proposed objective, a
preliminary questionnaire for collecting sound source descriptors
was carried out. This step aims to gather free-form descriptive
words of the sound sources from subjects that are not
familiarized with acoustic terms. After the annotation analysis, a
questionnaire was applied using a ranking scale. The selected
descriptive words were also correlated to the noise sources
mentioned in the standard. It is expected that this work may
give input to the ABNT NBR 15575 systematic revision, that kicked
off in 2018.
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Subjective Evaluation of Acoustic Quality Classes in Dwelling
Seda Kulak, Nurgun Bayazit
[Abstract]
The aim of the study was to evaluate the acoustic quality
classes subjectively and compare the results with objective
metrics. Interior noise level in a dwelling is described in the
Regulation on Protection of Buildings Against Noise in Turkey
with five classes, three of them (B, C, D classes) was
simulated via laboratory listening test experiment. 30 normal
hearing participants rated the most annoying neighbor
noise is determined with previous study and recorded in
selected household. The subjects also rated three classes in
terms of loudness and annoyance characteristics of the
sounds transmitted through 3 different walls (masonry wall,
aerated concrete wall, dry wall) or 2 different floors
(reinforced concrete slab, hollow tile floor slab) types which
were simulated by audio filters. Computer based interface
was developed to perform the experiment. The sound level
in a headphone was calibrated by using head&torso
simulator. The results were statistically analyzed and
subjective evaluation was correlated with physical
measurement of sound metrics.
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Comparison of Acoustical Performance and Subjective Evaluation in Residential Buildings
Ayca Sentop, Nurgun Bayazit
[Abstract]
The ”Regulation on Protection of Buildings against Noise” was issued by the Turkish Ministry of Environment and Urbanization in May 2017. The regulation was put into force partially in May 2018 obliging all new buildings to follow acoustical requirements. In May 2019, acoustical project by an acoustical expert will be obligatory for large scale or multifunctional buildings. The regulation aims to introduce a scheme for acoustical performance evaluation, ranging from A to F following the EU/COST TU 0901 project outcomes and ISO-CD19488. This study aims to demonstrate the application of the regulatory requirements with field measurements and a survey. Airborne and impact sound insulation, indoor noise level, mechanical noise level and reverberation time were measured in 6 residential buildings and their performance were analysed according to the classification scheme. A survey was conducted with the residents of the buildings. The results were compared and the applicability of the regulation was discussed.
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Sound insulation between dwellings – Comparison of national requirements in Europe and interaction with acoustic classification schemes
Birgit Rasmussen
[Abstract]
Most European countries specify limit values for airborne and impact sound insulation in housing. The requirements are expressed by descriptors defined in ISO standards implemented also as European and national standards. A comparative study investigating the regulatory requirements for airborne and impact sound insulation between dwellings in 35 countries in Europe has been carried out. The comparison shows considerable differences in terms of descriptors, frequency range and level of requirements.
Sufficient sound insulation between dwellings is important to protect against noise from neighbours and to provide privacy and possibilities for activities without causing annoyance. In addition to acoustic regulations, about one-third of the countries have developed acoustic classification schemes with classes reflecting different levels of acoustical comfort/protection. In some countries regulations point to a specific acoustic class, but are in other countries independent. The interaction will be described. All schemes have minimum two classes above regulations, thus enabling specification of higher acoustic quality than regulations, but typically only one class below regulations, implying that major parts of older housing cannot be classified. It is concluded that acoustic classification schemes should preferably include classes covering all existing housing, independent from construction year, thus allowing acoustic labeling in analogy to energy labeling.
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Acoustic regulation in hospitals – Interior acoustics improving the recovery of patients
Georg Schöner
[Abstract]
Busy environments are bound to be noisy and hospitals are no exception. The rampant noise found in modern hospitals are caused by a variety of sources such as patients, staff, and medical alerts, making hospitals unable to meet the desirable acoustic levels. Evidence-Based Design has the ability to reduce treatment times, the consumption of medication by patients, as well as create an environment that supports better sleeping patterns and calmness. For example, a study of 416 patients has shown that quieter environments can reduce hospitalization periods. Major European countries, such as NL, IT, UK, still miss any acoustic regulation for the healthcare sector. Other countries, such as FR, GER, ES, PL, FI and the Scandinavian countries have specific regulation for hospitals in place. This contribution will analyze the details for the requirements in 8 European countries. In addition the discussion will get initiated on what a meaningful ”common denominator” could look like for countries with neither interior acoustic guidance nor regulation in place.
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Organization:
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B. Rasmussen, SBi, Aalborg University (AAU-CPH) |
J. Jeongho, Fire Insurers Laboratories of Korea |
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The New Italian standard UNI 11532 on acoustics for schools
Arianna Astolfi, Linda Parati, Dario D'Orazio, Massimo Garai
[Abstract]
In Italy, the new UNI 11532 standard introduces requirements for acoustic comfort in
different building typologies such as schools, offices and hospitals. In March 2018 the
UNI 11532 - Part 1 ”General requirements” was published, which includes the main
descriptors related to noise, room acoustics and speech intelligibility to be considered
for the design and verification phases. The drafting of Part 2, dedicated to schools, is
now nearing completion. This standard is voluntary for private constructions, but
mandatory in the public sphere following the issue of the Ministerial Decree ”Minimum
Environmental Criteria for the awarding of design and construction services for new
construction, renovation and maintenance of public buildings” (Criteri Ambientali
Minimi or ”CAM”), dated 11 October 2017. The Decree explicitly refers to the UNI
11532 standard for the requirements to be guaranteed in public procurement.
The underlying principles and criteria of the new standard, as well as examples of
application and comparisons with other standards in Europe will be presented and
discussed in this work. Good acoustics is one of the main requirements from
kindergarten to university, where information is transmitted from teachers to students
mainly orally, and where learning is strictly related to the quality of this communication.
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New Finnish building acoustic regulation
Mikko Kylliäinen, Ari Saarinen
[Abstract]
New building acoustic regulation and related guidelines took effect in Finland recently. The statutes include provisions concerning sound insulation, noise and vibration abatement and acoustic conditions necessary for the construction of a new building, repair and alteration of a building and the changing of the intended use of a building. The essential technical characteristics enacted concern health and wellbeing in the manner required by the intended use of the building. The purpose of the guidelines is to clarify and facilitate compliance with the requirements as the statute directs. The guidelines provide instructions on the procedures for design and verification. The statutes apply to buildings with dwellings or accommodation or patient rooms, facilities for teaching, meeting, eating, treatment, recreation or physical exercise, or office spaces.
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Acoustic requirements vs experienced sound in wood structures
Klas Hagberg
[Abstract]
Wood, in particular CLT, is increasing as structural material
in buildings, improving the conditions for circular economy
within the construction sector. However, structural wood is
still creating doubts amongst many developers and one
issue is the difficulty to easily fulfil equal sound insulation
requirements as we are used to from the history with
concrete. The difficulty increases when the buildings are
high. To secure the Swedish sound insulation class B, either
the walls have to be very thick or elastomers have to be
installed between the different storeys. Often high design
margins are needed since the real efficiency of the layers
is unknown when the wall elements are tightly connected
due to static reasons. Therefore, it is the duty of
acousticians to ensure that the requirements we are
stating in the design process really correspond to the
annoyance of the habitants for different building
categories. We should not lower the overall quality but
instead refine the requirements to be more spot on in
each single situation. Therefore, increased knowledge is
necessary, in particular regarding annoyance in different
housing categories. This paper gives examples of the
challenges that we need to take into account for the
future development of wood technology.
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The increment in cost and other parameters to upgrade quality classes in sound insulation
Bilge San Ozbilen, Nurgun Bayazit
[Abstract]
Providing comfortable environments for users is the main objective for architects at the design process today. Physical performance parameters of building elements such as sound insulation, thermal insulation, resistance to fire and moisture are evaluated primarily for user comfort. However, in design phase elements’ costs, thicknesses and weights are considered as well. To ensure sound insulation, according to Acoustic Regulation of Turkey, minimum quality class of C should be provided. The practices like adding layer or increasing weight to upgrade quality class from C to A or B, generally are in confict with the cost criterion. In this study, 366 number of internal, 252 number of external masonry wall alternatives made of brick and autoclaved aerated concrete blocks are evaluated. The optimal wall types for each classes are selected by TOPSIS (Technique for Order Preference by Similarity to Ideal Solution) method. The weighing coefficients, representing importance, are given as sound insulation > cost > weight=thickness= thermal transmittance. This paper presents results of the calculation of required increments as percentages in cost, thickness and weight to upgrade quality classes.
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Organization:
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A. Mayr, Technical University of Applied Sciences Rosenheim |
S. Schoenwald, Empa Materials and Technology |
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Single rig two-stage method for characterizing structure-borne sound sources in buildings
Barry Marshall Gibbs
[Abstract]
Two source quantities are required to predict the structure-
borne power into lightweight buildings: source mobility and free
velocity or blocked force. The average source quantities can be
measured using the reception plate method (RPM). A thin
reception plate gives an estimate of the sum-square free
velocity. A thick reception plate gives an estimate of the sum-
square blocked force. The ratio of these quantities
approximates the square of the average source point mobilities.
In this proposal, only one reception plate is required. The transfer
mobilities from several remote points to a single measurement
centre are recorded. The source is attached about the
measurement centre, and the transfer mobilities re-recorded.
The ratio of the transfer mobilities gives an estimate of the
source mobility. The source is set into operation and the
velocities at the remote points recorded. For each remote point,
the recorded velocity and previously estimated source mobility
gives an estimate of the source free velocity. In the proposed
method: there is no need to change the installation conditions;
knowledge of source-receiver mobility ratio is not required; the
reception plate can be relatively thin/light; the method
automatically collapses the source data to equivalent single
quantities.
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Experimental investigation of a single reception plate method to obtain two source quantities required to predict structure-borne sound transmission in buildings
Fabian Schöpfer, Andreas Mayr, Ulrich Schanda
[Abstract]
To predict the structure-borne sound power into lightweight buildings and other structures, two source quantities are required: the mobility and the free velocity or the blocked force. Using the reception plate method (RPM), the average source quantities can be measured. The combination of measurements on a thick and thin reception plate, the so-called two-stage method, allows the determination of all required quantities using the RPM. The previous paper introduced a proposal that requires only one reception plate for this task. This paper shows an experimental case study that applies this approach. As reference, the free velocity and the source mobility are determined using measurements on the source approximating free conditions.
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Prediction of Maximum Fast Time-Weighted Sound Pressure Levels from Time-Varying Structure-Borne Sound Sources in Heavyweight Buildings
Steffi Reinhold, Carl Hopkins, Gary Seiffert
[Abstract]
At present the prediction of the sound pressure response in a nearby room due to structure-borne sound excitation from building machinery is limited to sources running under steady-state conditions according to the method described in EN 12354-5. As machinery noise also needs to be assessed under running conditions which are not steady-state, some regulations in European countries on installation noise also set requirements in the receiving room based on maximum Fast time-weighted sound pressure levels. In this paper investigations were carried out on a laboratory heavyweight reception plate to quantify the structure-borne sound power of different idealised time-varying signals using equivalent continuous levels over short time periods for input into the prediction model. Empirical corrections to previous tests are then applied to estimate maximum Fast time-weighted sound pressure levels from predicted short equivalent continuous vibration levels in heavyweight buildings. The maximum Fast time-weighted sound pressure level prediction results are validated against in-situ measurements in the building-like situation with close agreement between predictions and measurements.
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In-plane excitation of reception plates according to DIN EN 15657:2017
Ulrich Schanda, Michael Hoßfeld, Fabian Schöpfer, Andreas Mayr
[Abstract]
DIN EN 15657:2017-10 describes methods for the determination of characteristic structure-borne sound power of building service equipment. The so-called indirect method is based on the use of reception plates. The method is robust for the case when the mobility of the source (building service equipment) is significantly higher than the mobility of the receiving plate, which was shown in earlier investigations . In those studies the building sources were all mounted on the plane of the reception plate. Therefore the excitation of the reception plate happens out of plane. In practice however there are situations which require mounting in wall openings, for example roller shutters in window openings or doors. In these cases the excitation happens in the soffit of the structures, the excitation takes place in-plane. The assumption then is that primarily longitudinal waves rather than bending waves are excited in the receiving building elements. This mounting situation was experimentally investigated on a specific reception plate in the lab. In this paper first results will be presented and discussed with the focus on the determination of the characteristic structure-borne sound power of those sources which could be used as input data for prediction models.
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Extraction of high contributing sound of Air Handling Unit and noise reduction using transfer path analysis
Takuya Hirao, Yosuke Miyamoto, Junji Yoshida
[Abstract]
Air Handling Unit (AHU) is generally built in a machine room of
a
building and controls the temperature in multiple office
spaces.
The motors of AHU generate sound and vibration and have a
possibility to cause noise subject in office. Accordingly, it is
necessary to extract high contributing part of the AHU and
apply intensive countermeasures to the part quickly. In this
study, we attempted to reduce the noise by applying the
operational TPA to the simple AHU model. In the test, the
radiated sound in front of the model was measured and the
frequency band which increases the over all noise was
extracted. In addition, operational TPA was applied to find
out
the main contributor to the radiated noise. As the result, the
noise at 200 Hz band was large and the motor radiated noise
was found to be main contributor. Furthermore, the noise was
increased in the unit case by the cavity resonance. Then, the
inside volume of the AHU was separated using a plastic plate
for
the reduction of the motor noise in the unit. The result shows
that
the unit radiated noise at the frequency band could be
decreased well.
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Experimental Validation of a Finite Element Model for a Heavy Impact from the Standard Rubber Ball on a Timber Floor
Xiaoxue Shen, Carl Hopkins
[Abstract]
In lightweight buildings there can be problems due to heavy
impacts, such as from footsteps in bare feet or children jumping.
For this reason the standard rubber ball is used in measurement
standards to assess heavy impacts. In this paper, experimental
work on a mock-up timber floor is used to provide benchmark data
to validate a model using Finite Element Methods (FEM) of the
standard rubber ball exciting a timber floor. Experimental Modal
Analysis was initially used to extract damping information and
the Modal Assurance Criterion was used to confirm that the FEM
model of the timber floor was reasonable. A FEM model was then
used to predict the time-domain response of the timber floor when
excited by a single impact from the standard rubber ball which
was included in the FEM model. Comparison of these FEM
predictions with experimental data was assessed using the
Frequency Domain Assurance Criterion which indicated reasonable
agreement with the experimental results.
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Structure-borne sound sources in timber buildings – prediction of machinery noise using measured transmission functions
Fabian Schöpfer, Andreas Mayr, Ulrich Schanda
[Abstract]
Prediction of machinery noise in buildings involves two stages: (a) the emission of structure-borne sound and (b) the transmission (i.e. the propagation) across the building structure. This paper concerns the latter in timber buildings. In previous work a practical approach to predict the equipment sound pressure level using measured transmission functions was presented. A research project conducted at the Technical University of Applied Sciences Rosenheim pursued this approach by measuring a range of transmission functions in various timber buildings. In total a data set of 120 transmission functions measured in 19 timber buildings is now available. This paper gives a survey of this data by grouping similar constructions and transmission paths. The approach can be used to derive average transmission functions for similar situations. Those could potentially be provided in a catalogue giving data in one-third-octave bands. The application of this empirical prediction method is presented in case studies using fictive and real structure-borne sound sources in timber buildings.
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Organization:
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A. Santoni, Engineering Department, University of Ferrara |
C. Hoeller, OTH Regensburg University of Applied Sciences |
P. Fausti, Engineering Department, University of Ferrara |
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Measurement of the bending stiffness of masonry walls by using time-frequency analysis: Practical considerations and validation
Charlotte Crispin, Christian Mertens
[Abstract]
The bending stiffness is one of the parameters that determines the mechanical behaviour of a plate and therefore influences the acoustic performance of building elements. Besides, it appears in the prediction models for calculating the sound reduction index, R, or recently the vibration reduction index, Kij. Its accurate determination is thus essential in the field of acoustics. The Atomic Energy Research Institute of Korea has proposed an interesting measurement technique to determine this parameter in-situ. This technique, which uses the time-frequency analysis, has been successfully applied to thin plates. However, this paper shows that corrections should be used for thick walls. Some practical considerations are given and some measurements results for masonry walls are presented.
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Benchmarking of Methods for the Identification of Flexural Wavenumbers in Wooden Plates
Federica Morandi, Olivier Robin, Luca Barbaresi, Massimo Garai, Noureddine Atalla, Nicolas Quaegebeur, Patrice Masson
[Abstract]
This paper investigates and compares three methods for the identification of the flexural wavenumbers in thin laminated wood panels with the aim of verifying under which conditions each setup can provide accurate results. Laboratory measurements are performed on a plywood panel, an inhomogeneous and orthotropic plate, and the propagation characteristics of flexural waves are characterized along five radial directions. The methods considered for the analysis are the Inhomogeneous Wave Correlation method, the Prony method and the time-of-flight method. The first two methods have been implemented on the same data set by exploiting measurement along a dense mesh of scanning points in the target direction with a 3D-Laser Doppler Vibrometer and using a piezoelectric transducer for generating waves. The third method requires a different data set, gathered using an instrumented hammer as a source and four accelerometers as receivers and uses the cross-spectral density function for calcula. The results show a remarkable match between the first two methods, while the third method requires an accurate choice of the time window to provide accurate results. Nevertheless, the third method provides a quick and low-cost alternative to the use of the laser vibrometer that can be more suitable for in situ measurements.
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Using near-field acoustic measurements to characterise mechanical and acoustic properties of lightweight building structures
Andrea Santoni, Patrizio Fausti, Paolo Bonfiglio
[Abstract]
The evaluation of the vibrational field on a certain surface
has great importance in noise control engineering
applications. It can be helpful, both for noise reduction
purposes, and diagnostic purposes, in product
optimisation, or in order to characterise the mechanical
properties of the vibrating structure. Accelerometers are
still the most used sensors to measure vibration. However,
contactless transducers, such as laser scanning
vibrometers, have been widely used recently, presenting
several advantages. Near-field acoustic holography may
represent a valid alternative to LSV, in order to reconstruct
the vibrational field on a surface from sound pressure
measurements performed with an array of microphones.
In this study, the vibrational field induced by an electro-
dynamic shaker on a plywood panel was reconstructed
from the sound pressure measured close to the vibrating
surface. The test rig used allows to scan the sound pressure
over the vibrating surface on a grid of points, using an
array of microphones. The reconstructed panel’s dynamic
response was used to evaluate its elastic and acoustic
properties. The accuracy of this experimental approach
was assessed by comparing the results with the ones
obtained using miniature accelerometers. The
advantages of the method are discussed together with
drawbacks and limitations.
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Structural reverberation time measurements on WOODSOL prototype
Simone Conta, Ulrich Simon, Anders Homb
[Abstract]
The WOODSOL research project aims at developing urban
timber buildings based on moment resisting frames. One of
the key aspects when dealing with the sound insulation in the
building system is the flanking sound transmission across the
floor elements and through the special connecting elements
of the floor to the columns. We build a system prototype to
be able to investigate experimentally the vibroacoustic
behavior of the floor elements and the vibration transmission
to the columns considering in particular impact excitation.
In this paper, we focus on the structural reverberation time,
which is a key parameter to determine calculation quantities
such as e.g. Kij required by the ISO12354. The Kij-index is
necessary to estimate flanking sound transmission and
therefore a key input to SEA based calculation models.
We present the measurement setup used at the Woodsol
prototype and the measurement results obtained following
the ISO 10848-1. Results will be obtained from both impact
hammer excitation and shaker excitation with swept sine. We
discuss the challenges we encountered and present the
further steps in our analysis strategy.
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Bending wave based characterization of viscoelastic materials
Max Miller, Sadeq Malakooti, Tahereh Taghvaee, Ning Xiang, Hongbing Lu, Nicholas Leventis
[Abstract]
Obtaining dynamic mechanical properties of viscoelastic
solid materials can be challenging at acoustic
frequencies. Commercially available characterization
equipment is typically limited to about 500 Hz. Time-
temperature superposition extrapolation to higher
frequencies is indirect and time consuming. Resonance
methods do not yield broadband data and fail for
materials with high loss moduli. A new relatively simple
characterization method, relying on bending wave
excitation and laser Doppler vibrometer measurement, is
proposed. Complex elastic moduli and phase speeds are
now attainable to frequencies beyond 10 kHz with high
reproducibility. The materials under test are an array of
polyurea aerogels with fixed chemical composition but
different nanomorphologies. Low cost and ease of
synthesis add appeal to materials already noted for their
sound transmission loss performance.
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An indirect method for the measurement of impact sound insulation
Simon Bailhache, Stéphanie Colin, Michel Villot
[Abstract]
This paper describes a way of measuring impact sound
insulation for heavy structures without using the standard
tapping machine. Instead, another source of known contact
force - e.g. an impact hammer or electrodynamic shaker - is
connected to the building element under test. The sound
pressure level is measured in the receiving room and then
corrected from the difference in force level between the
standard tapping machine and the substitution source to obtain
the impact sound pressure level. An application of this indirect
method is the laboratory determination of the impact sound
performance of walls, which can be used as an input for
prediction of impact noise of isolated heavy stairs as well as of
structure-borne noise from service equipment in situ; the
standard tapping machine is indeed not suitable for exciting
vertical building elements. In this paper, the indirect method is
described, then applied to a floor element and the results are
compared to the impact sound pressure level directly
determined using the standard tapping machine on the same
floor. Several measurements are performed using different
numbers of source and microphone positions. The
corresponding results are presented and discussed.
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Continuing Prediction of Heavy/Hard Impacts on Resilient Sports Floors in Existing Buildings
Matthew Golden, Faiz Musafere
[Abstract]
Previously the authors have presented a method to
accurately predict the vibration performance of resilient
sports flooring in existing buildings. The method includes the
measurement of the force pulse injected into a building by a
heavy weight drop onto resilient sports flooring. This force
pulse is combined with an in-situ transfer function to predict
in-situ performance. Extensions to the model include
predictions over greater distances and multiple structure
types. Further work extends the prediction model to include
one-third-octave-band sound pressure levels. Finally, an
alternative one-third-octave-band methodology will be shown
and compared to one-third-octave-band prediction methods
suggested by others.
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Structure-Borne Sound Isolation of Acoustic Test Chambers: In-Situ Validation
Abhay Rajmane
[Abstract]
Acoustic test chambers are controlled environment for
diverse acoustic and vibration measurements. They are
broadly classified as Anechoic Chambers and
Reverberation chambers. The background noise level inside an acoustic test
chamber is important factor during measurements of
silent specimens and sensitive equipment. Often the
acoustic test chambers are built in noisy industrial halls.
To achieve low background noise level, it is inevitable
to structurally isolate the acoustic test chamber from
surrounding structure. This is performed using vibration
isolators e.g. G+H MAFUND Rubber Sheets,
VIBREX Spring Strips or Metallic Springs. These System
minimize the structure-borne sound entering in the floor
of acoustic test chamber, thereby minimize the
radiated airborne sound in the chamber. The elastic
isolation is designed in such a way that the natural
frequency is sufficiently below the cut-off frequency of
acoustic test chamber. To Validate the elastic isolation, the natural frequencies
are experimentally measured on floor of test chamber.
Using the experimental modal analysis, the natural
frequencies and mode shapes of the test chambers are
extracted. It is validated that the measured natural
frequency practically coincides with designed
fundamental frequency. The experimental results from
several acoustic test chambers are presented in this
work.
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Qualification of an Anechoic Chamber
Anthony Nash
[Abstract]
An anechoic chamber is typically used for acoustical experiments requiring a free-field environment. The free-field properties of the chamber can be determined by using a so-called ”divergence loss” (i.e., a draw-a-way) test to demonstrate that sound pressure decreases with increasing distance in accordance with the inverse square law. For the qualification of free-field environments, the principal governing standard is ISO 26101:2012. This standard prescribes protocols suitable for qualifying both hemi- and fully anechoic chambers including tolerances for deviations from the inverse square law. Either random noise or pure tones can be used for the ISO divergence loss test method, depending on the intended application of the chamber. This paper describes an ISO qualification test for a medium-sized anechoic chamber with an internal volume of 50 cubic meters. The chamber was equipped with a partly-open floor grate and furnished with foam wedges having questionable sound-absorptive performance. The chamber could satisfy the ISO tolerances when using random noise but failed to qualify when using pure tones. The paper discusses the measurements and the likely causes of the discrepancy.
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Sound Absorption Measurement Method using Ensemble Averaging Technique: A Robust Method for Surface Impedance Including in-situ Applications
Otsuru Toru, Reiji Tomiku, Siwat Lawanwadeekul, Daiki Matsuoka, Ryota Yoshimoto
[Abstract]
Sound absorption characteristics of materials are of importance for controlling the
quality of sound fields as well as of materials. In our previous paper, we proposed the
concept and measurement method for surface normal impedance and sound absorption
coefficient at random incidence condition, namely, ensemble averaged surface normal
impedance and sound absorption coefficient. We named the measurement method as EA
method, for short. We also showed the theoretical validation using the boundary
element method and experimental confirmations on the effectiveness of EA method in
successive several papers. In this paper, the outline of EA method is summarized, first.
Then, reproducibility of the results are examined on measurements conducted both in
reverberation rooms and in-situ. All the measurements were conducted by EA method
with a pressure-velocity sensor (Microflown PU-sensor) which was calibrated on site in
an acoustic tube. In-situ measurements were conducted at office room, living room,
cafeteria and outside terrace. The results reveal that rather robust absorption coefficient
values with the uncertainties less than 0.04 ∼ 0.05 are obtained regardless of
measurement places and that EA method has broad adaptability to sound absorption
measurements at various conditions.
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Measurement of Oblique-Incidence Absorption Coefficients of Various Types of Absorbing Materials in a Thin Chamber
Tetsuya Sakuma, Naohisa Inoue, Yuta Sakayoshi
[Abstract]
A novel measurement system for oblique-incidence absorption
coefficients has been developed using a thin rectangular
chamber and two arrays of microphones. In this method, a
series of incident and reflected plane waves at corresponding
angles are decomposed based on a propagation mode
expansion technique. Accordingly, absorption coefficients are
determined at discrete incidence angles, of which number
increases with frequency. First, a test measurement for glasswool
was done in comparison with theoretical estimates. As a result,
the measurement system was basically validated, but abnormal
values were occasionally observed at singular combinations of
frequency and incidence angle. The faults theoretically occur
under two conditions: the interval of microphone arrays
coincides with half wavelength in incidence angle, and some
modes are not generated due to loudspeaker position
mismatch. Therefore, a post-process was additionally
introduced to omit abnormal values systematically. Then,
measurements were done for various types of absorbing
materials, such as urethane foam, rockwool board, perforated
panel and honeycomb resonator, and their incidence angle
dependence of absorption coefficients was examined.
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Advanced Investigation Using The EApu Method on the Effect of Quantitation and Particle Size of Charcoal in Clay Bricks on Sound Absorption Coefficient
Siwat Lawanwadeekul, Otsuru Toru, Reiji Tomiku, Hiroyasu Nishiguchi
[Abstract]
Due to rapid urbanization, noise pollution becomes the
environmental problems in Asia and their harmful effects
on people’s health are undeniable. Reducing noise by
absorption material is a good alternative. Bricks are one of
the most common construction supplies in Asia because
of the cost. Therefore, this study aims to apply the authors’
EApu method (sound absorption characteristics
measurement method using ensemble averaging
technique and a pressure-velocity sensor) to clay bricks to
clarify the relationship between the sound absorption
coefficient and the amount and size of an additive. The
measurement was conducted in a reverberation room
following the procedure presented in our previous papers.
Measured bricks have four types of porous that were
designed by selecting from different sizes of charcoal that
was added into the specimens in the amounts of 0-30%wt.
Next, the samples were fired at 900-1100 C and tested for
sound absorption coefficient. Lastly, the microstructure
analysis was conducted. The results indicated that the
larger size and higher percentage of charcoal added in
fired clay bricks presented the excellent absorption
coefficient. Such a detailed investigation can be
performed by the EApu method, and the EApu method
helps material designing process become more
accessible to control sound absorption.
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A Measurement Technique of Sound Absorption Coefficient and Impedance Using an Impedance Tube and Two Cardioid Microphones
Kazuma Hoshi, Toshiki Hanyu
[Abstract]
Sound pressure and velocity can be measured using two
cardioid microphones which are located at same position and
pointed to opposite direction each other. This technique is
called C-C method. Applying this technique, we tried to
measure acoustic impedance of glass-wool 32 kg bulk
density 50 mm thickness using an impedance tube and two
cardioid microphones. The result is compared with a
measurement result of two (omnidirectional) microphones
technique. As results, it is clarified that surface impedance of
material can be measured by two cardioid microphones over
all frequency range in limitation of the impedance tube.
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A new technique for measuring sound particle velocity and sound pressure using face-to-face cardioid microphones
Toshiki Hanyu, Kazuma Hoshi
[Abstract]
Generally sound particle velocity can be measured by using a particle velocity sensor directly or by two sound pressure type microphones. Impedances and absorption coefficients of materials can be measured by these sensors. We have proposed a new technique for measuring sound particle velocity and sound pressure using face-to-face cardioid microphones, which is called the C-C method. A cardioid microphone responds to both sound pressure and particle velocity. Directivity of the cardioid microphone is formed by a combination of both sound pressure and particle velocity. In our new technique, the sound pressure and particle velocity can be obtained simply by summation and subtraction of responses of the face-to-face cardioid microphones. However, errors of measured sound pressure and sound particle velocity occur due to differences of directivities of the cardioid microphones. Therefor we have developed a calibration method of the cardioid microphones. By this calibration, accurate sound pressure and particle velocity can be measured even by using any kinds of cardioid microphones. We have conducted measurements in order to verify the new technique. Results of the measurements supported that the new technique can be used for measuring sound pressure and sound particle velocity accurately. Advantages of this technique are also discussed.
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Airflow Resistance Measurements between Room Temperature and 800 C
Thamasha Samarasinghe, Carl Hopkins, Gary Seiffert, Jilly Knox
[Abstract]
The acoustic properties of porous materials are usually measured at room temperature but in many industries, porous materials are required to provide sound absorption in high temperature applications such as in Heat Recovery Steam Generators (HRSG) applications, automobile silencers and aero engine liners. In this paper, measurements have been made of the airflow resistance of three different densities of Alkaline Earth Silicate (AES) fibrous materials. Measurements at room temperature to ISO 9053 and in a bespoke test rig at temperatures up to 800∘C were made using a kiln to achieve the high temperatures. The high temperature tests on these particular materials indicated that it might be possible to assume no significant effect of temperature on specific airflow resistance between 20 and 100∘C but not at higher temperatures. Above 600∘C, the effect of crystallisation and the change in material thickness meant that specific airflow resistance (rather than airflow resistivity) was used in regression analysis to identify empirical relationships for the three different density materials at different temperatures.
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Organization:
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D. Bard-Hagberg, Lund University |
A. Homb, SINTEF Building & Infrastructure |
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Reducing ground-borne noise due to railways. Part I: assessing the problem
Catherine Guigou-Carter, Guillaume Coquel, Philippe Jean, Alexandre Jolibois
[Abstract]
The vibration field along a railway line was measured and modeled to predict its impact on a future real estate project. Indeed, vibration levels due to rolling stocks evaluated near the tracks are usually evaluated on the vertical axis and at one or a few positions, supposed to be representative of the entire vibration field around the tracks. When the project size is rather extended along the tracks, this assumption might be incorrect, especially if the project site is located near a train station, where stresses in the tracks can be very different depending on the direction of the train due to acceleration or deceleration. This paper presents vibration measurements performed in the different directions before buildings construction along the tracks. Furthermore, the attenuation of ground vibration as a function of distance from track is used to evaluate the loss factor in the ground based on a comparison with a 2.5D BEM model of the ground including the different layers. The model is then used to identify equivalent forces in the different directions associated to train passage to obtain similar vibration transmission behavior in the ground. The mitigation measures are discussed in a companion paper.
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Reducing ground borne noise due to railways. Part II : mitigation measures
Philippe Jean, Catherine Guigou-Carter, Alexandre Jolibois
[Abstract]
The construction of new buildings very close to existing tracks with heavy commuting traffic requires careful design. Such projects must be accompanied with noise reduction measures. The practical case here considered has been analysed by first assessing the existing situation prior to construction (type of soil and traffic: results exposed in a companion paper). Here we focus on the mitigation measures which have been tested and the expected corresponding vibration and ground-borne noise levels inside the buildings. A 2.5D FEM/BEM commercial software (Mefissto), developed by CSTB, has been employed in what is a very challenging numerical context: multi-layered ground, large multi-levels buildings, large frequency range (10-200 Hz). In every situation the use of an original 2.75D approach leads to the noise levels inside the building. As the tracks cannot be modified the solutions tested consist in inserting different types of barriers in the ground or in inserting spring-like materials in the lower part of the structure. Results are compared to target values in terms of vibration and ground-borne noise levels selected. In order to limit dwellings occupants’ annoyance. Satisfactory results can only be achieved with the use of the spring-like solution.
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Correction of Sound Pressure Levels Calculated for Small Rooms Using Diffuse-Field Theory
Kiyoshi Masuda, Hikari Tanaka
[Abstract]
The most frequently used method for predicting sound pressure levels arising in a room as a result of transmitted sound or floor impact sound from adjacent rooms is an acoustic energy calculation assuming a diffuse sound field. If the dimensions of the room under investigation are sufficiently large, sound pressure levels can be predicted accurately by this method. However, especially at low frequencies, the wavelength of the sound may be nearly equal to or exceed the room dimensions, leading to reduced predictive accuracy. Although the Waterhouse correction is typically used to correct for this error, its reliance on statistical energy considerations means that it cannot take into account the response to the frequency range below the first-order mode and to frequencies between adjacent lower-order modes. In this work, a new correction method is derived from the relationship between the sound pressure level calculated using wave theory and the sound pressure level using diffuse-field theory. By comparing calculated results obtained using the new method with both actual measurements and FEM calculations, its accuracy is examined.
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Using realistic test signals to evaluate existing structures for low frequency sound transmission from clubs, live music venues, discos, and exercise facilities
David Sage Woolworth
[Abstract]
Low frequency sound used in entertainment (and sometimes exercise) facilities is difficult to reproduce in standard testing procedures for existing facilities such as ANSI E336 and ISO 10848-1, as the test cutoff frequency is well above sounds generated in-situ for these facilities, and some methods or loudspeakers cannot reproduce the sound level of low frequency that would be typical for everyday operation for these facilities This paper proposes using alternate test signals and methods that are appropriate to facility programming that must be isolated from neighboring structures. It addresses ”worst case” low frequency source characterization, sound levels that incite vibration of finish materials that are transferred through the structure, and metrics used to evaluate receiving spaces.
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Experimental Study on Low-Frequency Averaging of Indoor Sound Pressure Level in Façade Sound Insulation Measurement
Jinyu Liu, Naohisa Inoue, Tetsuya Sakuma
[Abstract]
In the ISO 16283 series for field measurement of sound insulation, a low-frequency procedure is specified for determining indoor average sound pressure level, which is the so-called corner method. In the procedure, additional measurements are required in the corners in addition to the default measurements in the central zone, and the indoor average level is corrected with the highest level in the corners. However, this procedure is empirically proposed, and its validity is not fully examined for various cases. In this paper, façade sound insulation measurements are performed in the low frequency range for a mock wooden house and a 1/4-scale model house. In the rooms with the volume of about 20 m3, sound pressure levels are measured at multiple grid points including corners. Regarding the average level for all grid points as a reference, the ISO procedure and other possible averaging schemes are examined in the aspects of bias and deviation due to measurement point selection.
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Additional sound insulation panels ZIPS-experience of 20 years of application
Alexander Boganik, Anatoly Livshits
[Abstract]
In apartment buildings and offices is often the task of increasing
the sound insulation of walls and ceilings from airborne noise.
Traditional methods of solving this problem, gypsum plating on
metal or wood frame - have low efficiency. This is, in particular,
due to transmission sound vibrations through the frame and the
presence of a large number of wall lining elements that must be
properly performed during installation. The frameless panels of
additional sound insulation (ZIPS) were developed in 1999 in
Russia. They have high rates of additional sound insulation (9...18
dB) in depending from thickness of 43...120 mm. This system was
patented and is still widely used. In addition, the system provides
the necessary strength and ease of installation. Over 20 years,
the system has undergone several upgrades. The article
describes the physical principle of the ZIPS-system, the data of
laboratory and full-scale acoustic measurements, latest
modification of the system with adjustable mounts.
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Noise from plants systems and Building Information Modeling: The Code Checking.
Costantino Carlo Mastino, Roberto Baccoli, Andrea Frattolillo, Martino Marini
[Abstract]
The acoustic performance of buildings is increasingly a performance requirement required both in the construction of private buildings and public buildings. At international level, in recent years we have witnessed the revision and issue of new standards relating to the project phase (forecast calculation) of passive requirements based on the ISO 12354 series of standards. At the plant design level, the reference standard is represented by EN 12354-5 and ISO 16032 while in the post-operation verification phase the ISO 16032 and ISO 10052 standards are the reference for the measurements. One of the fundamental aspects to achieve high performance is to reach a level of design study that is increasingly detailed and shareable. Building information modeling, in addition to being an obligation for some types of work, is increasingly used by designers and construction companies, to have a better knowledge of the case on particular and methods of execution. The present work presents an overview of procedures that exploit the information of the BIM model for the predictive calculation of the acoustic performances of the systems, the verification of the legislative limits and the integration of the instrumental data recorded post-operam. The procedures were finally tested on a case study.
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Organization:
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F. Martellotta, DICAR-Politecnico di Bari |
V. Wittstock, Physikalisch-Technische Bundesanstalt |
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Experimental characterisation of absorbing materials made from renewables
Volker Wittstock, Daniel Sgriess
[Abstract]
The amount of ecological implications caused by the whole building process is gaining more and more attention. One measure to reduce ecological implications may be the use of building material made from renewables. To investigate this possibility, a joint research project is under way in Germany, which is focused on insulating material made from renewables. Besides fire protection, heat insulation, moisture proofing, emissions and sustainability assessment, also sound insulation is investigated. PTB contributes to this project by experimentally characterising insulating materials made from renewables. For this, existing measurement methods have been reviewed and adapted to the special needs of the project. The contribution is focused on the measurement of airflow resistivity by the alternating flow method and measurements in a transmission tube. Special attention is given to the influence of the material orientation and to the uncertainty of the measurement results.
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Investigations regarding the Influence of Static Load and Airflow Resistance on the Measurement of Dynamic Stiffness
Heinrich Bietz, Martin Schmelzer, Volker Wittstock, Spyros Brezas
[Abstract]
Within the scope of a research project regarding the physical properties of insulation materials made from renewables, one of PTB’s tasks is to determine the dynamic stiffness of such materials used as elastic layers. The common measurement procedure is described in DIN EN 29052-1. As a closer examination of this standard bears some open questions, several investigations have been conducted recently. This presentation focuses on two aspects: The first one is the influence of the static preload. This might be relevant for insulation materials used for thermal insulation composite systems, where, compared to floating floors, the static preload is significantly smaller. As the standardised measurement procedure is not suited for small static preloads, an alternative measurement set-up was designed to investigate this topic. The second aspect is the effect of airflow resistance on the resulting dynamic stiffness. The measurement setup as devised in the standard does not necessarily reflect the boundary conditions the material is exposed to when it is mounted in situ. The standard describes a procedure to handle the influence of airflow resistance, but this is partially incomplete and not applicable for some materials. Some theoretical considerations and measurement results regarding this topic will also be presented.
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Insulation materials made of renewable raw materials for the sound insulation prognosis of building components
Simon Mecking, Andreas Rabold, Anton Huber
[Abstract]
In order to increase the application possibilities in the building sector for insulating materials made from renewable raw materials, a project was initiated by the Fachagentur Nachwachsende Rohstoffe e.V. which shows the different requirements for these insulating materials and provides planning and verification options. In the field of sound insulation, building component catalogues for these insulating materials are expanded, test methods for the material properties of the insulating materials are defined and calculation models are further developed. The airborne and impact sound transmissions of the components are predicted in order to investigate the influence of the different insulating materials from renewable raw materials in comparison to conventional insulating materials. Component measurements indicate that a characterization of the cavity-insulation alone by the flow resistance is insufficient. Using a semianalytical calculation approach with an equivalent fluid, further insulation material properties can be taken into account in the prognosis of double wall elements. The input data for the prognosis calculation are determined by measurements of the transmission properties and compared with results of other calculation models based on the flow resistance. By validating the first prediction results with measurements, it should be shown to what extent the conventional measured variables are sufficient.
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Organization:
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F. Martellotta, DICAR-Politecnico di Bari |
V. Wittstock, Physikalisch-Technische Bundesanstalt |
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Preliminary investigation on the acoustic properties of absorbers made of recycled textile fibers
Chiara Rubino, María Angeles Bonet-Aracil, Stefania Liuzzi, Francesco Martellotta
[Abstract]
In recent years, wastes reduction and their conversion into
new resources are becoming a major requirement for our
society. A considerable interest is focused on the use of
recycled textile materials in building products and on their
potential application as absorbing acoustic material. In
developing innovative products it is not only important to
recycle, but also to manufacture devices with recycled
content. The challenge is the use of textile waste fibers
and nontoxic binder to obtain composite materials with
less social, economic and environmental impact. In this
paper 100% merino wool wastes bounded with chitosan
are investigated to obtain sound absorbing materials,
particularly for the acoustic refurbishment of the open-
office spaces. The use of chitosan instead of synthetic and
plastic elements represents a valid solution to obtain bio-
materials for optimizing buildings environmental
sustainability. The acoustic performance of innovative
composite materials were investigated. Measurements of
normal incidence sound absorption coefficient were
carried out by means of a standing wave tube, according
to ISO standard 10534-2. Experimental results were very
promising, showing absorption coefficient higher than 0.5
at frequencies above 500 Hz. This proved that it could be
possible to produce sustainable alternative to traditional
commercial synthetic products.
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Characterization of woven fabrics for development of micro perforated panel absorbers
. Gunawan, Iwan Prasetiyo, B Yuliarto, K Anwar, D R Andhika
[Abstract]
Woven fabrics have a potential to be developed as sound absorber system because the presence of micro perforation. By introducing air cavity backing the woven fabrics up, such system can undergo Helmholtz resonator mechanism as found in micro-perforated panel absorber. The micro perforations in a woven fabric are formed by yarn in x and y direction. The perforations can create a viscos-inertial effect when interact with sound fields that is useful as a basis for sound absorption mechanism. In this study, we are focused to investigate the relationship of woven fabric material and manufacturing technique to sound absorption characteristics. Natural and synthetic fibers-based yarn are used to produce woven fabrics. For investigation, the geometrical and physical properties of the yarns and fabrics are characterized. The geometrical properties are characterized using Scanning Electron Microscope (SEM) while the physical properties are measured using textiles testing devices. The sound absorption coefficients are evaluated by impedance tube using transfer function method. From this study, it can be concluded that the sound absorption coefficients are affected by material and manufacturing technique properties. It is expected that the results can be beneficial for developing design procedure of development woven fabric based micro perforated panel absorber system.
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Sound Absorption Characteristics of a Back-Perforated Honeycomb Panel with an Air-Layer
Kyungsoo Kong, Tetsuya Sakuma, Naohisa Inoue
[Abstract]
A honeycomb panel has a high potential in noise control
engineering as well as in any other engineering fields. This paper
demonstrates that a back-perforated honeycomb panel (BPHP)
placed with a back air-layer exhibits unique absorption
characteristics composed of the combination of panel- and
resonator-type mechanisms. A BPHP is theoretically modeled as
a two-mass system composed of panel and resonator’s neck
masses, and multiple air stiffnesses. First, resonance mechanisms
are clarified from the model. It is revealed that the panel-type
resonance can arise from the panel mass and air stiffness
corresponding to the total thickness of the panel and air-layer,
even if a BPHP is placed considerably closely to the back wall.
Next, normal- and random-incidence absorption coefficients
are calculated with changing several parameters such as
perforation diameters, hole pitches, panel thickness, back air-
layer thickness and so on. Furthermore, a supplementary
numerical calculation confirms the validity of the proposed
theoretical model. Through the case studies, several charts are
arranged for effective design of BPHPs as low-frequency sound
absorber. Finally, absorption performances of BPHPs are
measured by the reverberation room method, and the results
fairly agree with those calculated by the proposed model.
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Architectural Acoustic Design using absorption materials: the case study 'Snooze Panel'
Livio Mazzarella, Maria Cairoli
[Abstract]
The Snooze® sound-absorbing panel is the result of the
necessity to characterize an architectural space such as
offices, corridors, classrooms, hospitals while guaranteeing
the required acoustic characteristics using renewable
materials. An experimental case study is proposed showing
how different chooses in panel assembly arrangement are
affecting its acoustic performance.
The proposed procedure allows evaluating the acoustic
properties of a panel assembly, specifically the rating of
sound absorption according to the EN ISO 11654 i.e. the
weighted sound absorption coefficient, αw, considering the
intrinsic panel characteristics and the different types of
installation with variable distance from the wall surface
behind the panel.
While the material characteristics can be assessed in
laboratory using appropriate instruments as explained, the
acoustic behavior of the panel due to the geometry and
installation solutions are evaluated in actual assembly in
reverberation chamber according to the standard EN ISO
354.
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Analysis of environmental and seasonal effects on sound absorption by green wall systems
Emmanuel Attal, Nicolas Côté, Takafumi Shimizu, Bertrand Dubus
[Abstract]
Buildings and roads are made of acoustically rigid materials which reflect sound emitted from many sound sources and contribute strongly to acoustic environment In that context, green roofs and/or walls used for building envelope greening were found to be efficient solutions for acoustic protection in cities when applied in a significant urban scale. Green systems (roofs and walls) are usually composed of plants, soils and/or air gaps. Below 1000 Hz, sound attenuation is low in plants and air but much higher in soils. However, sound absorption by green systems vary strongly with the arrangement of plants, air and soil and efficient broadband absorption is only obtained when the arrangement is optimized.
In this work, the effect of green wall arrangment as well as environmental (moisture content) and seasonal (leaf density) effects on absorption coefficient are studied. Effective properties of plants and soils are measured in an impedance tube using three-microphones-two loads method between 100 and 1000 Hz under controlled moisture content for soils and leaf density for plants. Sound absorption coefficient at normal incidence is then calculated using matrix manipulation. Variation of absorption coefficient with moisture and leaf density is provided for various green wall systems.
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Organization:
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K. Larsson, RISE Research Institutes of Sweden |
L. Barbaresi, University of Bologna |
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Facade Sound Insulation as Protection to Outdoor Noise
Chiara Scrosati, Fabio Scamoni, Michele Depalma, Matteo Ghellere
[Abstract]
In recent years the acoustic qualification of buildings has
gained more and more importance at international level.
One of the important elements for determining the sound
quality of houses is the insulation against external noise. In
fact, many issues related to excessive noise in dwellings
arise from their inclusion in noisy environments, or from their
original surrounding environment, which has grown from
silent to very noisy over the years.
The main strategy to limit the noise inside the buildings is
the façade insulation improvement. Regulatory
requirements and classification schemes in Europe present
a high degree of diversity. In particular, the approach to
the façade insulation acoustic requirement adopted in
the various countries is very diversified also because
closely related to the external noise climate. The main
issues related to the façade sound insulation are
investigated with particular regard to the choice of
descriptors, to the evaluation methods and their
reproducibility, which is the less easy to evaluate.
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Measurements and prediction of sound insulation of innovative ventilated façade solutions
Francesca Di Nocco, Luca Barbaresi, Federica Morandi, Massimo Garai
[Abstract]
This work is part of ”Progetto Involucro”, a research project funded by the Emilia-Romagna Region in the frame of POR-FESR 2014-2020. It aims at testing innovative ventilated façade solutions for improving buildings’ energy and acoustic performance. Two identical test buildings (prototypes) are built in real scale on the same site and different ventilated façade solutions are installed on the South facing walls. In order to evaluate the performance of different alternatives, a benchmark solution is installed on the first prototype, while all the other innovative solutions are installed on the second one and they all are tested under the same environmental conditions. Sound insulation measurements are performed on site with the aim to investigate the contribution of different rainscreens in relation to their fastening system and individual component elements such as natural ventilation grills, open joints and potential openings on the external side. Furthermore, a measurement campaign focused on flanking transmission is conducted to evaluate sound transmission across the façade junctions for each innovative solution installed. This paper shows the preliminary results of the measurement campaign. Data obtained from on-site measurements are commented and discussed in relation to the values calculated using the model proposed in the ISO 12354-3 standard.
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Auralization of the Coincidence Effect on Façade Sound Insulation
Dag Glebe, Krister Larsson
[Abstract]
If a façade contains windows, their sound insulating properties will often govern the insulating performance of the façade. When presenting the façade’s insulating effect on the in-door sound environment, a natural approach is to auralize a realistic example of traffic noise that has been acoustically filtered by a window. One approach is to use results from measurements of the window’s sound insulating properties according to ISO 10140-1. The third octave band sound reduction indices can then be used to filter a traffic noise sound file. However, some important properties are not covered by this approach but should be considered. One example is the coincidence effect, which is only represented as an integration of noise incident at all angles, due to the diffuse field, and thus only contribute to the amplitude. However, the relationship between incidence angle and frequency may be important for the perception of traffic noise, particularly at single vehicle passages. Therefore, two auralization approaches were evaluated in a listening test with respect to how they were perceived: the simple amplitude-based approach with data from standard measurements, and an approach employing the transfer function of different incidence angles.
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Designing a Better Plenum Window or Balcony for Higher Noise Reduction Against Outdoor Noise
Liu Yee Louisa Cheung, David Bk Yeung, Ching Chan, Calvin Chiu
[Abstract]
Plenum windows have been used as innovative
measures to reduce traffic noise in the past decade.
Nowadays, they are also used in mitigating more serve
traffic noise, railway noise or even fixed noise which
require different noise reduction characteristic and
enhanced performance.
To enhance the noise reduction performance of the
plenum windows and balconies, different kinds of
absorption treatment like micro-perforated absorbers
and perforated absorption panels had already been
used and tested. They could provide around 1dB of
improvement. Meanwhile, window designers are finding
alternative ways to further enhance the performance,
such as by changing the physical design and
dimensions. The most common changes are to increase
the overlapping length of the plenum of the window or
by adding a solid parapet in front of the plenum
balcony door.
Numerous designs of windows and balconies with
different dimensions have been tested, with different
overlapping length, different acoustic treatment and
even outer window or balcony door design. The tests
were done in full scaled mock up using linear array of
loud speakers mincing traffic noise source. The effects in
noise reduction of implementing these design changes
to plenum windows/balconies are being studied by
comparing the measurement results in this paper.
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Sound Insulation of Fenestration Systems: A Comprehensive Web-Based Simulation Program and Validation
Tejav Deganyar, Fangliang Chen, Yihe Huang, Rebeccah Grotemeyer
[Abstract]
Aluminum framed fenestration systems are becoming
leading choice for modern buildings, as they provide a
better aesthetic appearance, higher degree of
weathering reliability, and quicker installation while
maintain more transparency and environmental
efficiency. Because of the introduced complexity of the
systems, however, estimation of sound insulation of such
systems for preliminary design or analysis become much
more time consuming and difficult. Currently, no
straightforward solution to fulfil such purpose is available,
and a convenient and reliable tool is highly desirable to
provide a quick estimation of the sound insulation
performance of fenestration systems. Bearing this burden,
the engineer team led by the authors conducted
extensive studies on various fenestration systems consisting
of different glazing and aluminum frames. Based on our
research outcomes, a comprehensive web-based
simulation program, so-called Digital Acoustic Lab (DAL),
has been developed recently. This program is able to
provide quick analysis and accurate prediction of sound
insulation of window, door and façade systems consists of
various aluminum frames and glazing infills. A series of
laboratory tests were conducted to validate the
predictions by the DAL. The goal of this program is to
provide designers, engineers and architects an effective
and economic tool to facilitate the design.
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Adaptive acoustic comfort: Assessing noise with provisions for ventilation and overheating in dwellings
Jack Harvie-Clark, Anthony Chilton, Nick Conlan, David Trew
[Abstract]
In the design of residential developments in the UK, it has
been common practice for façade sound insulation (to
protect against outdoor noise) to be considered separately
from the provisions for ventilation and for mitigating
overheating. This fragmented approach has led to different
designers making different, incompatible assumptions about
the internal environmental quality conditions: the acoustic
designer assumes that windows are closed to control
external noise ingress, while the mechanical designer
assumes that windows are open for ventilation or mitigating
overheating, for example. This leaves occupants with a
choice between reasonable noise levels or thermal comfort,
but not both. This problem is exacerbated by increased
overheating risk in modern buildings and future climate
scenarios.
In response to this issue, the Association of Noise Consultants
has produced the Acoustics Ventilation and Overheating
Residential Design Guide - the ”AVO Guide”. The AVO
guide recommends an approach to acoustic assessment
that takes regard of the interdependence of provisions for
external noise ingress, ventilation and overheating. This
paper describes the context, background and content of
the AVO Guide. There are references to attenuated options
for ventilative cooling to help designers avoid simply
specifying mechanical cooling.
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Requirements for the façade sound insulation for different types of outdoor noise
Steffen Körper
[Abstract]
Buildings are exposed to a wide variety of external noise situations. The time course of the sound pressure level and the frequency response of external noise can vary greatly between the vicinity of roads compared to the vicinity of airports or railways.
The sound insulation of facades protects the indoor space and the residents against external noise day and night. The German standard DIN 4109 ”Sound insulation in building construction” contains the minimum requirements for sound insulation for all new buildings in Germany. All requirements are based on a single-number-value, the equivalent sound pressure level, without adequate respecting to the particularities of the frequency response and time structure of a sound source.
The presentation shows the German standard values in building acoustics and the subject of external noise. The results of measurements of different sound sources will be shown. A new approach and an outlook for further possibilities to protect inhabitants appropriately to the noise situation will be discussed.
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Ventilative cooling in noisy environments: practical options for the UK
Nick Conlan, Jack Harvie-Clark
[Abstract]
It is only very recently that a proper consideration of the
internal acoustic environment when mitigating overheating
has started to be taken seriously in the UK. Previously this
issue has not been addressed by the various regulatory
systems, and developers have been reluctant to pay for
what they have seen as an unnecessary consideration. The
Association of Noise Consultants has produced the criteria
and guidance in its Acoustics, Ventilation & Overheating:
Residential Design Guide.
At this crucial stage in the establishment of new approaches
for the construction industry, we are setting out ”rules of
thumb” with typical options for ventilative cooling in varying
external noise environments. We are developing solutions in
collaboration with mechanical engineers and product
manufacturers that are suitable for UK building typologies.
Vents in the external facade are found to present the
easiest adoption into current building designs. As acoustic
attenuation typically involves airflow resistance, mechanical
fans are required to ensure ventilation rates are achieved in
noisier environments or where the apartment has single-
sided ventilation opportunities.
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Innovative Noise Mitigation Measure- Baffle Type Acoustic Window
Ching Chan, Liu Yee Louisa Cheung, David Bk Yeung, Calvin Chiu, Billy Fan
[Abstract]
Hong Kong is one of the densely populated area in the world
with residential sites surrounded by traffic roads. Road traffic
noise pollution problem becomes a major environmental issue
for residents in Hong Kong.
In an attempt to tackle road traffic noise impact on residents
and ensure good quality of individual’s health and well-being,
an innovative measure called Baffle Type Acoustic Window has
been proposed, which designs to benefit residents from
environmental noise planning and building ventilation’s aspect.
It consists of two glazing: (i) an outer openable window and (ii)
an inner sliding panel with air gap in between. The inner sliding
panel is situated right behind the outer openable window.
To investigate the acoustic performance of Baffle Type Acoustic
Windows with different design parameters, a series of acoustic
laboratory tests have been carried out in accordance to ISO
Standard to discover the relationship between the key design
parameters of the window (e.g. width of overlapping between
sliding panel and the window system, size of outer window
opening and acoustic material used etc.) and its corresponding
acoustic performance.
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Organization:
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J.Y. Jeon, Hanyang University |
C. Hopkins, Acoustics Research Unit, University of Liverpool |
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Development of a modified impact testing method for simultaneously evaluating multiple floor toppings
Wayland Dong, John Loverde
[Abstract]
Materials such as floor coverings, screeds, and resilient matting are commonly applied on top of a floor-ceiling assembly to improve the impact noise insulation. Theory and experience show that the benefit of such materials increases with frequency, with minimal effect below the natural frequency of the resilient materials and the largest effect at high frequencies. This improvement in impact insulation is added to the base performance of the structural assembly. Floor design can involve multiple parameters (finish floor, screed thickness, resilient mat thickness) even from the same manufacturer or product line, and the number of permutations rises quickly. These permutations must also be tested on multiple structural assemblies. The authors are investigating a modified method where multiple top-side assemblies could be installed on the same structural assembly, allowing for a more efficient testing process. The expectation is that airborne and low-frequency impact insulation would be consistent since the base assembly is unchanged, but high-frequency impact insulation would provide the same result as a full-scale test. Data from the investigation is presented and analyzed to validate the model and document the uncertainties.
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Round robin for testing tapping machines
Sylvia Stange-Kölling, Volker Wittstock
[Abstract]
From a metrological point of view, tapping machines are working standards for a defined vibrational excitation of building elements. To ensure a uniform excitation, tapping machines have to comply with several requirements according to the relevant ISO-standards. Whether a tapping machine is in conformity with these requirements is tested by different laboratories in Europe. Since the requirements are well defined but not the way the conformity has to be tested it is questionable whether different laboratories come to the same results for the same tapping machine.
On that occasion, a round robin on tapping machine testing was organized by PTB. Two different tapping machines were tested in 5 different laboratories in Europe. The main quantities hammer curvature, time between hammer impacts, velocity of the hammer at the impact and hammer mass were measured by all laboratories. Results of these measurements are presented and possible future developments are discussed.
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New single-number quantities for evaluation of impact sound insulation
Mikko Kylliäinen, Petra Karoliina Virjonen, Valtteri Hongisto
[Abstract]
The standardized single-number quantities for evaluation of impact sound insulation do not correlate especially well with the subjective judgment of living impact sound sources directed to the floors. The aim of this study was to find new single-number quantities better associated with the subjective annoyance caused by different impact sounds. The new single-number quantities were developed on the bases of experimental data of laboratory measurements of impact sound insulation of floors and a psychoacoustic experiment. Five studied impact sound types were walking with hard shoes, socks, and soft shoes, super ball bouncing, and chair moving. Basic requirement was that the new single-number quantities could be expressed as the sum of L’n,w or L’nT,w and new spectrum adaptation terms. Reference spectra for calculation of new spectrum adaptation terms for each sound type were derived by the means of a mathematical optimization method. In addition, an optimized reference spectrum based on all five sound types was derived.
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Efficient numerical prediction and experimental validation of impact sound radiation by timber joist floors
Pengchao Wang, Cédric Van Hoorickx, Geert Lombaert, Edwin Reynders
[Abstract]
Achieving good impact sound insulation with a timber
floor is
challenging, given its low weight. Therefore, the acoustic
performance for a timber floor should be considered at
the
planning stage, and a prediction tool is required to
evaluate the
impact sound insulation design. This work investigates the
prediction of the impact sound radiated by a timber joist
floor.
The floor properties are determined by modal tests of the
separate floor components. The impact sound radiated by
the
floor is subsequently computed in two steps. In the first
step, the
impact force of an ISO tapping machine is computed by
considering both the global and the local mobilities of the
floor.
In the second step, the response of the point-excited floor
is
computed by the finite element method, and the sound
pressure level in the underlying room is computed by a
diffuse
room model. Special attention is paid to the influence of
boundary conditions on the computed eigenmodes of the
floor;
these are also compared against identified eigenmodes of
the
floors. Finally, the sound pressure levels and the single
number
rating results are computed, for excitation by a
instrumented
hammer and the ISO tapping machine. The computed
results
are experimentally validated.
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The preliminary study on subjective rating of different floors characterised by Ln,w+CI,50-2500
Vojtech Chmelík, Jakub Benklewski, Monika Rychtarikova, Dominik Kisić, Kristian Jambrošić, Marko Horvat, Herbert Muellner
[Abstract]
The problem of noise in dwellings is topic of large
discussions
nowadays. Not only airborne noise but also impact
noise are
responsible for decreased comfort of living. The
subjective
evaluation of impact noise sounds in dwellings are
described
in this paper. Listening tests based on method of
adjustment
(Scharf,1961) with statistically sufficient sample of
tested
subjects were performed in listening room in the
laboratory
of TGM where subjects compared performance of
presented floors. Impact noise stimuli were filtered
with
various floor constructions with similar Ln,W values,
sorted into
two main groups: (i) light-weight (LW) and (ii) heavy-
weight
(HW). Comparisons of listening tests performance
with
calculated single number quantities for each
construction
were prepared in order to understand the perception
of
presented sounds by dwellers. The results from
preliminary
study are meant as a basis for adjusting of currently
applied
reference curve also with particular focus on the low
frequency range.
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Perception of Combined Indoor Noise Sources in Lightweight Buildings
Alessia Frescura, Pyoung Jik Lee
[Abstract]
Contrary to community noise field, the combined effects of
multiple noise sources on annoyance have not been
investigated in building acoustics. Therefore, the present study
aims to quantify the total annoyance caused by indoor noise
from the upstairs and next-door neighbours. Footstep sounds
were recorded in a laboratory equipped with a timber joists floor
in different configurations. Listening tests were then performed
to assess the relationship between annoyance ratings and
sound pressure levels in several scenarios. In the first part, the
annoyance caused by individual impact and airborne noise
was evaluated considering the sound insulation performances
of different wall and floor structures. The participants were also
asked to evaluate the total annoyance in the second part
when impact and airborne sound sources were presented at
the same time. This paper will show models of noise annoyance
when impact and airborne sources are heard separately and
simultaneously. The findings of this study would be helpful to
understand how we perceive the combination of noise sources
we are usually exposed in timber houses.
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Influence of suspended ceiling type of residential building on heavyweight floor impact sound
Song Hansol, Jongkwan Ryu, Inho Kim, Yong Hee Kim
[Abstract]
This study investigated the influence of suspended ceiling type
of residential building on heavyweight floor impact sound
through floor impact sound measurements in test building using
rubber ball. In general, gypsum board is suspended using
hangers for the ceiling finishing in the Korean residential building,
and air space between slab and the suspended ceiling thus
occurs. Three types of the suspended ceiling (perforated, sound
absorbent, and resonator ceiling) were tested to explore
reduction of floor impact sound level in comparison with
general gypsum board. Results showed that three suspended
ceilings significantly reduced by 1∼4 dB in single number
quantity for rating heavyweight floor impact sound insulation. It
was also found that dominant frequency band of the reduction
depended on the suspended ceiling types.
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Organization:
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A. Kohlrausch, Eindhoven University of Technology |
K. Bijsterveld |
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MATLAB-based simulation software as teaching aid for physical acoustics
Jorge Petrosino, Lucas Fernando Landini, Georgina Alejandra Lizaso, Antonio Ian Kuri, Ianina Canalis
[Abstract]
This paper presents examples of simulated experiments that can be
run in MATLAB or Octave as a demonstration for students, allowing
them to explore and understand the basics of wave propagation. The
use of the k-Wave toolbox is proposed. This open source add-on is
capable of performing simple and efficient simulation of wave
propagation in the time domain, as well as providing comprehensible
real-time visualization of the process. It is mainly used to simulate
ultrasonic waves in biological media; however, its reliability isn’t
diminished in applications where lower frequencies are involved. In
this work, 2D acoustic signals in the audible range are used to explore
a variety of phenomena related to physical acoustics, such as
reflection, diffusion, diffraction, absorption, resonance. The placement
of the acoustic elements in the medium for each numerical
experiment is extracted from a BMP image file depicting sources,
sensors and solid materials, encoded in different colors. The proposed
method allows users possessing elementary knowledge of MATLAB /
Octave code to interact with the k-Wave toolbox in a beginner-friendly
fashion. Vast control over the simulation conditions can be exercised
by simply drawing, cutting, pasting, moving or recoloring elements of
the image file.
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Information Retrieval and Topic Modeling based on the ICA Proceedings
Thore Oltersdorf
[Abstract]
Information retrieval is a key feature to get an idea on topics that are new to someone. It is a general purpose technique that any scientist is being educated in during his career. But times and techniques have changed rapidly in the past few years. Multiple approaches are possible now due to massively available open data. It is also a recursive process since often the linguistics and/or programming capabilities are missing but needs to be known before its usage can be started. Based on open source software tools approaches from NLP (natural language processing) will be demonstrated on a certain aspect of technical acoustics. This paper will introduce how ICA proceedings from 1953 up to 2016 are used to get an idea of acoustical aspects of technical systems used for HVACR. This is not the same as understanding and having the knowledge of each detail but it is key to gain it.
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Organization:
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J. Bruyninckx, Maastricht University |
V. Tkaczyk, Max Planck Institute for the History of Science / Humboldt-Universität zu Berlin |
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From Noise Control to Intelligibility: a history of office acoustics
Joeri Bruyninckx
[Abstract]
As historians have shown, at least since the early twentieth
century, architectural acoustics has been concerned with
dampening reverberant spaces into a clean, echoless
acoustical ambience that came to represent the so-called
ësoundscape of modernity’ (Thompson 1995). In this paper, I
show how architectural acousticians actually developed a
completely different approach to noise control, shifting from
acoustic noise control to managing (psycho-)acoustic
intelligibility. I will discuss this shift against the context of office
planning. From the 1950s onwards, new office concepts (such
as the open office) were embraced for their planning flexibility,
efficiency and advantages to inter-employee communication.
However, they also posed new problems, as employees
complained of noise, reduced concentration and lack of
privacy. Professional acousticians in the 1960s and 70s
addressed these complaints by modelling ëprivacy’ as a
problem of speech intelligibility. They did this, among others, by
repurposing the Articulation Index, a wartime standard
developed to evaluate telephonic communication, into an
inverted measure of privacy. Drawing on archival research, on
acoustical isolation manufacturer Owens-Corning Fiberglas and
office design manufacturer Herman Miller Inc., I trace how
sound masking prompted a series of new technologies,
materials and approaches to ëcondition’ building acoustics–yet
again radically changing the architectural soundscape.
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Sound & Science: Digital Histories: Navigations through the History of Acoustics
Viktoria Tkaczyk
[Abstract]
This paper will present the database ”Sound & Science: Digital
Histories,” an initiative of the Max Planck Research Group
”Epistemes of Modern Acoustics.” (https://acoustics.mpiwg-
berlin.mpg.de) The ”Sound & Science” database, currently in its
beta version, provides valuable access to a diverse range of
sources in the history of acoustics-many of them previously
unavailable in digital format. The resource includes a
multimedia archive of primary source material, documentation
of surviving technology, and historical reenactments of crucial
experiments in acoustics. The database can be browsed using
a full-text search function or via curated collections, or
resources can be viewed on a map. In my paper, I will outline
user-scenarios for the database’s visualization tools, which open
up large-scale and longue durée views on the history of sound
and science, allowing users to explore and assemble material in
unprecedented ways.
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The Consolidation of Engineering Acoustics as an Example of Contextual History of Science
Joachim Scheuren
[Abstract]
Following widespread practice in natural sciences, the history of Acoustics is often understood and depicted as a sequence of scientific ideas and achievements provided by ingenious individual scientists or scientific schools. Even if this approach is accepted because it foregrounds the sequence and relation of ideas it risks to ignore the driving energy of any social and technical context of science. This is particularly true for applied sciences like acoustics which, from a certain level of scientific insight, are much more driven by the needs of technology than by purely scientific interest and progress. The mutual influence and dependence of scientific, technological and social determinants can be illustrated exemplarily for the renaissance of acoustics in the mid of the last century. Being underestimated as conclusively understood and lacking new challenges, the relevance of Acoustics changed significantly with the demands of the post-war era. The respective requirements were able to initiate an iterative process linking technological progress and needs with new scientific insight. With reference to some typical examples it will be shown how these interactive efforts finally established the disciplines of engineering acoustics and noise control which essentially contributed to renew and strengthen acoustics as an indispensable interdisciplinary field.
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Echoes and Guns: Objects, Practices and Aims in Early Modern Acoustics
Leendert Van Der Miesen
[Abstract]
As interest in acoustical experiments grew in the early modern period, the range of acoustic objects and practices greatly expanded. Although many acoustic objects-such as the string or organ pipe-were still related to the field of music, scholars increasingly studied the acoustic properties of other objects, for example guns or the echo. These objects played a large part in the development of notions of sound propagation and indicate a wide variety of uses and aims of acoustical knowledge. Through examples ranging from the French scholar Marin Mersenne in the early seventeenth century to scientific academies such as the Royal Society in London and the Académie des sciences in Paris, my paper considers the different sites of experimentation, and especially the aims and overarching practices of the nascent discipline of acoustics. With these objects, I am able to investigate the different roles of experimentation and the possible implementation of acoustical knowledge.
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The History of Acoustics in Breslau/Wroclaw before and after 1945: A Bridge over Time
Andrzej Dobrucki, Peter Koeltzsch
[Abstract]
The paper presents the history of research and the most important people active in the field of acoustics in Breslau before 1945 and in Wrocław after this date. The first important acoustician to lecture in Breslau and here died in 1827 was Ernst Chladni. Organized research in acoustics originated at the end of the 19th century by Otto Lummer at the University of Breslau. His successor was Professor Clemens Schaefer. In 1910, the TH Breslau was founded, and research in acoustics was also conducted here. The leading figure was Erich Waetzmann. He dealt with room acoustics and electroacoustics. Professor Kurt Schuster was a pioneer in the measurements of acoustic impedance. The book ”Ultrasound” by Prof. Ludwig Bergmann remains important to this day. After 1945, the research was concentrated at the Wrocław University of Technology. The pioneer of this research was Professor Zbigniew Żyszkowski, who dealt with electroacoustics and telephoneometry. His students were: Wojciech Majewski, specialist in speech and Janusz Renowski - psychoacoustician. The professors in next generation are Andrzej Dobrucki, who currently heads the Chair of Acoustics and Tadeusz Gudra, active in ultrasound. The youngest professor, from the further generation is Krzysztof Opieliński.
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The Church building secularization through its sound Sound in a contemporary case study
Maria Cairoli, Livio Mazzarella
[Abstract]
Acoustics in a contemporary liturgical place, particularly in a
catholic context, is far from obvious. This work describes some
main indications for the design of new churches, starting from
the project of Sant’Ambrogio Church in Trezzano, Milan (IT).
The article presents room acoustics procedures, results,
discussions, theoretical interpretations of the sound
propagation in the specified space, and the implementation
of computer simulation techniques, emphasizing the
continuity and breaking elements with the historical
churches.
In the past the sacred sense in a church was linked to
emphatic rites and to the attainment of transcendent states.
This effect had to be greatly reinforced by the surrounding
atmosphere created in very reverberant spaces and possible
echoes that infatuated the perception of God and his
presence. On the contrary, nowadays concentration and
intimacy are preferred, the reverberation time is reduced,
increasing the speech intelligibility of the homily to
understand the message of God, while music or songs steel
accompany the choral word of church goers.
The proposed acoustic design includes a new approach with
an acoustic response suitable for both speech and music,
considering also the recent room acoustics studies in other
typologies of building such as congress centers and
multipurpose auditoria.
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Organization:
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B. Boren, American University |
A. Andreopoulou, Laboratory of Music Acoustics and Technology, University of Athens |
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Acoustic effects at prehistoric landscapes: an archaeoacoustics analysis of rock art sites from Western Mediterranean
Margarita Díaz-Andreu, Angelo Farina, Enrico Armelloni, Laura Coltofean, Mathieu Picas, Tommaso Mattioli
[Abstract]
Many pre-state societies around the world give special
importance to places where special acoustic effects are
generated, and often these places receive special treatment
including the production of rock paintings in them. In the
Western Mediterranean, it seems that outstanding acoustic
effects such as directional echoes, augmented audibility and
long reverberation time are present in some rock art areas with
Neolithic depictions dating between the VII and IV millennia BC
(painted in Macroschematic, Levantine and Schematic rock art
styles). On
the basis of this evidence, it has been argued that there is a
strong probability of acoustics having been used as a method
by Neolithic artists to select the shelters in which to produce
rock art. This paper presents the results of the ongoing
ERC research project in archaeoacoustics (Artsoundscapes -
The sound of special places: exploring rock art soundscapes
and the sacred). This project seeks to explore the
role of sound in the creation and use of these archaeological
sites. In particular, the authors discuss the results of previous
fieldwork in
three countries (Spain, France and Italy) and the development
of an innovative set of research methods that includes 3D
Ambisonic recordings, Transmission Loss measurements, and GIS
soundshed analysis.
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From Methodology to Archaeoacoustics in the Time of Scripture: Complex Dialogue Between Archaeological Evidence, Texts from Scholars and Written Mentions
Jean-Christophe Valiere, Estèle Dupuy, Bénédicte Bertholon
[Abstract]
Nearly 15 years of research on the acoustics of pots from 11th to 18th centuries have led us to crossing data from various disciplines and developing a specific methodology. The presentation analyzes the pitfalls of interpreting this technique through the eyes of the scholars that have addressed the issue since antiquity and of written mentions rather than using the archaeoacoustical data found in France and Europe. An archaeoacoustic research will only be meaningful if it can prove a deliberate intention to produce an object with an acoustic function. However, such a proof necessitates observing regularities, by studying several buildings or objects that attest that an effective choice has been made. A common error is to restrict the hypotheses to those which are scientifically admissible today. The case of pots acoustics is particularly edifying in this respect. Nevertheless, precise analysis of historical texts makes it possible to widen the field of hypotheses. Semantic study of the evolution of the meaning of key words and their associated concepts makes it possible to study the evolution of the acoustical techniques and to observe their progress through the centuries.
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A case of archaeological evidence in favour of acoustical intentions linked with pots in church vaults: Montivilliers Abbey
Jean-Dominique Polack, Jean-Christophe Valiere, Bénédicte Bertholon, Pauline Carvalho
[Abstract]
Montivilliers Abbey near Le Havre in France presents a case
where pots in the vault were obviously employed to act on the
acoustics. The archives of the abbey contain a written mention
that links pots with acoustics. Originally built in the 13th century,
the Romanesque vault of the transept crossing was later
modified at an unknown date by the addition of pots. In 1648
the vault was lowered down, hiding the original pots. A dozen of
new pots were inserted in the new vault. The presentation
focuses on the acoustics of the abbey and documents the
organization of the pots by the builders in the 1st and the 2nd
vaults. Room-acoustical measurements of the church were also
carried out in order to evaluate its acoustical quality. A major
conclusion is that lowering the vault has singled out the church
choir where the nuns chanted and which is characterized by
reverberation times lower than in the nave. Intermediate
reverberation times were measured in the crossing.
Reverberation times in octave 125Hz, which includes the
resonance frequencies of the pots, are marginally lower in the
choir and the transept for source in the nave, which could
indicate an effect of the pots.
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Virtual reconstructions of the Théâtre de l’Athénée for archeoacoustic study
David Poirier-Quinot, Barteld N. J. Postma, Brian F. G. Katz
[Abstract]
The French ECHO project is a multi-disciplinary
collaboration combining the efforts of historians, theater
scientists, and acousticians to study the use of voice in
20th century French theaters. An audio-visual
simulation framework was created which combines
auralizations with visualizations of former
configurations of the Théâtre de l’Athénée, Paris. These
configurations corresponded to a series of
renovations, enabling researchers to perceive realistic
stage performances in foregone rooms. Simulations
include the room, two actors moving on stage, and an
audience. To achieve these simulations, archived
architectural drawings were studied, providing various
details of the different theater configurations from
which the corresponding visual and room acoustic
geometrical acoustics (GA) models were created. The
GA model of the theater’s current state was calibrated
to reference measurements. Subsequent
modifications to the GA model allowed for reversing
several renovations to arrive at the original 1896
configuration of the Athénée. Simulations across
configurations accommodated the investigation of the
impact of these renovations on acoustic conditions,
such as reverberation and clarity. These observations
can then be compared to historical information, such
as theater critics of the time. The final resulting
audio-visual simulations allow for 360∘ presentations at
various positions in the theatre using commercial
standard hardware.
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Acoustic Simulation of Elizabeth I's Speech at Tilbury
Braxton Boren
[Abstract]
The largest gatherings in human history prior to the advent
of electronic amplification were all effectively limited by the
loudness and intelligibility of the human voice. Various
accounts throughout history depict single orators addressing
crowds or armies of many tens of thousands, though some
historians have cast doubts onto the veracity of these claims.
Prior acoustic simulation has shown that reasonable
intelligibility might be achieved for up to 20,000 listeners for
very loud orators and very quiet crowds with high density.
Nearly all such accounts, however, depict a man addressing
the crowd, with one notable exception: in 1588 Elizabeth I of
England addressed a large army of between 8,000-19,000
troops gathered to repel the impending invasion posed by the
Spanish Armada. The acoustic plausibility of this account is
tested via STI simulations, focusing primarily on the density,
topography, and environmental conditions at the time of the
speech, as well as the frequency-dependent differences
between female and male orators’ vocal spectra and a sound
radiation patterns.
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Organization:
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M. Gatt, Technical University of Munich |
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How does what we hear sound? The qualia problem in acoustics
Michael Haverkamp
[Abstract]
Development of quantitative methods for description of
auditory perception has extended the possibilities of detailed
analysis and optimization of sound properties. Psychoacoustic
quantities help to assess audible features like loudness,
roughness, sharpness, tonality and pitch. But why is it still not
possible to precisely describe how an auditory phenomenon
really manifests itself within the subjective perception of an
individual? This would be of essential interest in terms of
definition of targets for sound design, legal determination of
branding sounds and comprehension of annoyance of
auditory incidents. In order to approach this question, the so called qualia-
problem shall be outlined in view of auditory perception. It is
evident that development of effective metrics for sound
evaluation is hindered by this matter of fact. Even under the hypothetic presumption that a complete set
of relevant parameters could be achieved, a definite
description would fail due to the impossibility to include the
perceived subjective content. Moreover, the holism of
perception impedes to achieve this goal. Various approaches will be discussed briefly: 1. Verbalization of sensations and onomatopoeia
2. Inclusion of further modalities as tertium comparationis
3. Parameters related to the human body, like movement
4. Approaches for notation of music and sounds
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Alexa, Siri and more: The impact of speech recognition on social behaviour and our responsibility as its creators
Andrea Ebner
[Abstract]
Digitalization and globalization allow us to create machines that can be used all over the world. But we must consider what people in the target markets want. What do they think of the hidden microphones in their living rooms, why are they afraid of speech-control? Are people ready to give us their trust in our devices and their consent to share their privacy with us? Since the catchphrase Artificial Intelligence can be read in every newspaper and it seems like there is no other technical topic out there to talk about any more, I want to ask some questions. What are the society’s worries? Where may they come from? How can we handle our responsibility as the creators of change? The political debate often seems idealistic. We as scientists and engineers can support the decision-makers and the society with giving them rational facts and provide simple, understandable explanations of what is going on in the tech world. Because it is not just about with which good or evil device we interact. Instead of being afraid of change it is a matter of which principles we want to manifest this change on to transform the world.
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The inner voice
Monika Gatt, Marcus Maeder, Steffen Marburg
[Abstract]
With inner voice, contemporary common sense often labels an individual creative motivation of acting in private life, social media and business.
From a philosophical point of view, the inner voice refers to the ancient Socratic conception of soul and genius. And that is why the secret roots of the profession ”Ingenieur” (german) and Engineer (english) and the understanding of audition might be found deaper: deaper in culture, in science and as well in someone’s becoming a person. On the one hand, reliable results of a research study in Acoustic focus on criteria of objectivity, but every step might also be mesuared by each one’s own personal situation.
This talk deliveres three main topics:
1. a crucial philosophical-logical matrix of objectivity as an example of subjective resonance in research,
2. a phenomenological diagnosis of privacy space in relation to outdoor noise,
3. a free philosophical development of the inner voice as a healthy ethical space of narration.
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Organization:
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S. Luzzi, Vie en.ro.se. Ingegneria |
C. Bartalucci, University of Florence |
A. Di Bella, University of Padova -Department of Industrial Engineering |
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Comparison of the recreational noise regulation in Russia, Europe, and Asia
Sergio Luzzi, Chiara Bartalucci, Iuliia Rassoshenko, Aleksandr Shashurin, Nickolay Ivanov
[Abstract]
In October 2018 the new Environmental Noise Guidelines for the European Region were published by the World Health Organization (WHO), providing strong evidence that noise is one of the top environmental hazards to both physical and mental health and well being in Europe. However, recreational noise has not yet been investigated in detail. After a preface about the more recent outcomes on recreational noise provided by EU Guidelines and projects, the current article presents the requirements for noise levels in recreational areas of various categories (parks, specially protected areas, reserves, etc.) in Russia, Europe and Asia. Moreover, it describes the main existing legal and regulatory acts in the countries, the history of their development and restrictions on their use. Furthermore, the article shows a comparative analysis with the allocation of advantages and disadvantages of each system. Finally, suggestions have been made to improve the system for regulating noise levels in recreational areas.
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Recreational Noise Management in Turkey
Nılgün Akbulut Coban, Sezer Kaya, Güray Doğan, Mustafa Çoban
[Abstract]
Noise is one of the inevitable environmental types of pollution
especially in densely populated urban areas with the design
approach including the recreational places within the
residential areas. It is well-known that there are lots of evidence
on the health effects of environmental noise and to control this,
the improvements have been carrying on in both legislative
basis and scientific basis. In Turkey, during the harmonization of
European Directives into national legislation as a full candidate
country of European Union, 2002/49/EC of Environmental Noise
Directive was one of the first ones transposed with the name of
”Regulation on Assessment and Management of Environmental
Noise (RAMEN)” in 2005. In the scope of RAMEN, the noise
criteria and evaluation procedures are identified for the main
noise sources including also entertainment places.
In this paper, recreational noise management in Turkey will be
explained and discussed through giving the information on what
the main environmental noise criteria for entertainment places
is, which authorities responsible for noise audits are, how the
procedures to take the music allowance are applied, what
requirements of technical staff to conduct the environmental
noise measurements and preparing the noise assessment reports
are needed.
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Long term monitoring of noise pollution in social gatherings places: time analysis and acoustic capacity as supports of management strategies
Enrico Gallo, Louena Shtrepi
[Abstract]
City of Torino (Italy), as many European cities, is facing an increasing noise pollution related to recreational noise of nightlife in open urban areas, such as streets, squares and terraces, where thousands of people meet spending all evening and night time. Noise from people speaking in these spaces can be very loud, due to crowd levels (since communication is possible only with a raised voice level) and people behaviour. This has led administrations to attempt to manage this specific noise problems with different strategies. This study shows the analysis of long term noise data collected in two years by a low-cost IoT network in Torino. It highlighted the regularities on night levels, mainly on a weekly basis and with seasonal trends. Preliminary investigations on time series have been developed, in order to correlate noise levels and number of people involved, coupling environmental data and crowd sensors. Moreover, the concept of ëAcoustic Capacity’ related to the noise levels has been investigated. This approach based on a deeper quantitative knowledge is explored as an help to local administration and stakeholders in planning and implementing mitigation actions.
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Revision of Chinese National Standard GB 50118 - Code for Design of Sound Insulation of Civil Buildings
Guojun Yan, Weibin Wu, Jie Lin, Chun Xu, Qiyuan Zhao
[Abstract]
The national standard GB 50118 Code for Design of Sound Insulation of Civil Buildings, issued by Ministry of Housing and Urban-Rural Development (MOHURD), is the basic code for indoor acoustical quality of civil buildings in China. It stipulates the limits for indoor noise levels and sound insulation performance of building elements, and specifies the design criteria of noise control. The current version was published in 2010, in which requirements had been raised compared to previous version. However, with the rapid development in China during the last decade, current regulations could not meet the growing needs for better sound environment. Traffic noise, HVAC noise and structure-borne noise from service equipment have become increasingly annoying, and inadequate sound insulation aggravates the situation. Therefore, MOHURD commissioned China Academy of Building Research to undertake the revision of current standard in 2018. Based on the correlation between field measurement results and occupants’ perception, and referring to guidelines and regulations in developed regions, we’d like to put forward stricter requirements for indoor noise and sound insulation performance of building elements. Besides, limits for structure-borne noise caused by vibrations of building service equipment or any other noise sources will be introduced to assess the dominant low-frequency noise.
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Organization:
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G. Brambilla, IDASC-CNR |
G. Licitra, ARPAT – Environmental Protection Agency of Region of Tuscany, Area Vasta Costa |
C. Asensio, I2A2-Universidad Politécnica de Madrid |
B. Schulte-Fortkamp, Technische Universität Berlin |
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Acoustic noise maps - exchange / integration of data within various database systems - problems, needs, new possibilities, interopracy
Tomasz Malec
[Abstract]
When creating acoustic maps, it is necessary to use many data sources such as Numeric Model of Terrain (NMT), Building Data Base (BDT), Environmental Data Base (cover, bridge, etc.) and others. For these data it is necessary to apply a layer of acoustic data such as traffic parameters and type of pavement for roads/rails, sound power level for industrial source, noise barriers, etc. The sources of these data may vary, including those based on open licenses(OpenStreetMap, Inspire), paid government data (NMT, BDT, etc.).
Each of the above data can be presented in several generally used data formats.
All data was integrating in the noise map calculation programs. This approach blurs the transparency of data. It becomes legitimate to ask if it can be standardized so that it is possible to pass data between programs to calculate noise maps or generally available GIS tools.
The work presents an overview of data, tools for their usage, methods of their storage. Based on the review, the directions of changes necessary to increase the transparency of these data and their interoperability were proposed.
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Noise Low Emission Zone implementation in urban planning: results of monitoring activities in pilot area of LIFE MONZA project
Rosalba Silvaggio, Salvatore Curcuruto, Manlio Maggi, Antonio Fasanella, Giorgio Cattani, Alessandro Di Menno Di Bucchianico, Alessandra Gaeta, Gianluca Leone, Enrico Mazzocchi, Raffaella Bellomini, Sergio Luzzi, Francesco Borchi, Chiara Bartalucci, Monica Carfagni, Giulio Arcangeli, Nicola Mucci, Carlo Casati, Giulia Pessina
[Abstract]
The main objective of LIFE MONZA (Methodologies fOr Noise low emission Zones introduction And management), started in September 2016, concerns an easy-replicable method, and related guidelines for regulatory purposes, for the identification and the management of the Noise Low Emission Zones, urban areas subject to traffic restrictions, whose impacts and benefits regarding noise issues are testing in the pilot area of the city of Monza, located in North Italy.
Noise LEZ has been introduced in Libertà district, by top-down actions (traffic restriction and vehicles speed regulations, lanes-width reduction and pedestrian crossing introduction, substitution of the asphalt with a silent road surface) and encouraging an active involvement of the people, in the definition of a more sustainable lifestyle (bottom-up actions).
Monitoring activities in pilot area, regarding noise, air quality improvement and the wellbeing conditions of the citizens, have been carried out, and they are still ongoing, before and after the Noise LEZ introduction and the available results are described in this paper, in order to discuss and to evaluate, in an unitary way, the main effects of Noise LEZ introduction in urban planning and in lifestyle of inhabitants.
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Soundscape Planning as a Tool for Urban Planning
Moritz Lippold, Bryce Timothy Lawrence
[Abstract]
Dealing with noise is one of the major challenges cities are facing. Spatial planning and noise legislation in Germany is commonly based on the idea of noise avoidance and noise control, whereby noise is treated as a waste product. At the same time, there is consensus that quantitative noise reduction is not enough to address the acoustic environment. Therefore a readjustment of the instruments and methods of spatial planning is necessary. As a suitable alternative in thinking about noise the soundscape approach might open up new, more holistic perspectives by considering the acoustic environment as a resource. However, there is currently a lack of practical approaches to include the idea of Soundscapes in spatial planning. This study asks the question: How can a soundscape planning approach be integrated as an instrument for spatial planning in Germany? To answer the research question, a systematic review of existing literature and a summary of German noise-related policy instruments will provide a framework of how sound and noise is treated in Germanys legal system and give insight into how a soundscape approach could be integrated into the German spatial planning system.
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Noise complaints and its relation to socio-economic factors at city/region scale in England
Huan Tong, Jian Kang
[Abstract]
Reporting noise complaints is an important part of informing
noise legislation. While the effects of various socio-economic
factors on noise perception have been investigated in
previous studies, the aim of this study is to examine
relationships between the rate of noise complaints and socio-
economic factors. The noise complaints and socio-economic
datasets from the government open data source at district
and unitary authority levels across the England are used. The
socio-economic dataset is categorised into four groups in this
study, namely demographic, working, property and
deprivation factors. Correlation analysis is conducted
between noise complaint rate and socio-economic factors,
and the results suggest that the correlations are generally
significant. Cities/regions with a higher proportion of young
and single residents are likely to receive more noise
complaints, and so are city/regions with a higher
unemployment rate and higher proportion of residents living
in flats. The deprived city/regions, measured by the English
Index of Multiple Deprivation, tend to have higher noise
complaint rate.
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Introduction to a new approach in urban acoustics
Alexander Lee, Jochen Schaal, Berndt Zeitler
[Abstract]
Statutory rules describe how to generate noise maps by defining methods and the choice of input data for calculating sound propagation. In highly reflective and diffractive environments, as exist in urban surroundings, those rule based methods have their limits. The currently used geographical input data for buildings consists of generalized blocks, which disregard detailed structures and materials in the cities’ streetscape. However, for urban environments a new simulation approach, applied to an urban model with more structural details, could be useful for planning and predicting noise. Luckily, building data with a higher level of detail (LOD) is available through geographic information systems, though cannot be utilized by conventional noise map calculations. So, as a first step, a conventional noise map of such a low LOD input is compared to a map created by soundPLANnoise’s sound particle diffraction (SPD) model with the same data set. Since the SPD model can incorporate high LOD input, in a second step, the effect of model details on noise maps is compared. Stöckach will serve as showcase area, a quarter in Eurocities member city, Stuttgart.
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Organization:
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F. Schelle, Institute for Occupational Safety and Health (IFA) |
Y. Demiral, DOKUZ EYLUL UNIVERSITY |
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ASR A3.7 “Noise”: Technical Rules for the German Ordinance on Workplaces
Florian Schelle, Jan Selzer
[Abstract]
The most fundamental national requirements regarding noise at workplaces in Germany are the ”Ordinance on Noise and Vibration Protection” for workplaces characterized by an LEX,8h ≥ 80 dB, as well as the ”Ordinance on Workplaces”, which applies to workplaces with LEX,8h < 80 dB. Both are concretised by means of ”Technical Rules”.
Important aspects of both Technical Rules will be presented. An emphasis will be put on the relatively new ASR A3.7 ”Noise”, which came into force in May 2018. It was designed especially for the assessment and rating of noise that is not hazardous to the inner ear, but can potentially cause extra-aural effects. Therefor it relies on reverberation time along with a so-called ”Rating Level” Lr, which will be presented in more detail. The given parameters, limit values and recommendations will be discussed as well as practical implications based on the experience of actual workplace measurements.
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Noise emission data as a prerequisite for Buy Quiet: Challenges related to the European Machinery and Outdoor Noise Directive
Fabian Heisterkamp
[Abstract]
Article 5 of the OSH Directive 2003/10/EC on noise aims at ”avoiding or reducing the exposure” of workers to noise, so that (Article 5 1.) ”...the risks arising from exposure to noise shall be eliminated at source or reduced to a minimum”. The Buy Quiet-concept, i.e. the selection of machinery with a special focus on low noise emissions, can represent an effective means to comply with this legal requirement. However, reliable and useful noise emission data are not always available. Especially, machinery designed to be used outdoors can represent a challenge. This kind of machinery is covered by both the ”Outdoor Noise”-Directive 2000/14/EC (OND), a Global Approach directive, and the Machinery Directive 2006/42/EC (MD), a New Approach directive. The interplay of these two directives results in noise emission declarations that differ from those of other machinery and can represent a challenge for manufacturers as well as employers. The arising problems are discussed and explained at observations in practice. Equipment covered by the OND is almost always provided with emission data. The challenges when trying to select quieter machines based on these data are discussed and possible strategies to identify quieter machines, but also to improve the OND, are presented.
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Admissible Values and Methods of Measurement of Noise, Ultrasonic Noise and Infrasonic Noise at Workplaces in Poland
Dariusz Pleban
[Abstract]
The adverse effects of noise, ultrasonic noise and infrasonic noise on human are associated mainly with the hearing organ. Nevertheless, non-auditory effects can be also observed in the entire human body. Therefore, the admissible values of noise and ultrasonic noise at workstations are specified in the Regulation of the Minister of Family, Labour and Social Policy. According to this Regulation, measurement methods should be specified in Polish Standards, and, consequently, the method for the measurement of noise at workplaces is defined in the Polish Standard PN-EN ISO 9612. In the case of ultrasonic noise the draft Polish Standard prPN-Z-01399 ”Ultrasonic noise. Requirements for measurements in the work environment” was developed on the basis of the results of studies carried out in CIOP-PIB.
Moreover, as regards the workplaces occupied by juveniles and pregnant women, the admissible values of noise, ultrasonic noise and infrasonic noise are defined in regulations issued by the Council of Ministers. In turn, values that constitute the criteria of annoyance of noise and infrasonic noise are defined in Polish Standards PN-N-01307 and PN-N-01338.
The discussion of the admissible values and the characterization of the draft Polish Standard prPN-Z-01339 are presented in the paper.
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Ultrasound Noise Policy and Assessment: Canada Safety Code – 24, a Canadian Perspective
Lixue Wu
[Abstract]
Ultrasound is extensively used nowadays in work and home. Scientific evidence on the health effects of exposure to ultrasound recommends that caution should be taken when using ultrasound. This leads to safety regulations established and enforced by government agencies in many countries. In Canada, early regulations respecting occupational safety and health made under Part IV of the Canada Labour Code, were substituted by Canada Occupational Health and Safety Regulations (SOR/86-304), effective March 31, 1986. Where a device that is capable of producing and emitting energy in the form of non-ionizing radiation is used in the work place, the employer shall implement the applicable document, in respect of ultrasound equipment, Safety Code - 24, dated 1990. The National Research Council Canada (NRC) is one of the government agencies who provided reviews and comments in the preparation of the Code. In this paper, the Canadian recommended human exposure limits for airborne ultrasound and its measurement techniques are discussed with the focus on challenges. To implement the safety code, a measurement system for ultrasound pressure level measurements was developed recently at NRC. This paper presents the details with the focus on measurement uncertainties. An onsite noise assessment example is given using the system.
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Noise exposure of employees in retail trade
Jan Selzer, Florian Schelle, Andrea Wolff, Frank Rokosch, Anne Gehrke
[Abstract]
Noise can lead to extra-aural effects, which can diminish the
health of employees in the long term. To protect the employees
from extra-aural noise effects, a new national regulation for
measurement and rating of noise at workplaces became active
in Germany by means of a technical rule. In order to assess the situation of employees in retail trade,
measurements of noise exposure were carried out following the
mentioned technical rule and ISO 9612. Personal sound
exposure meters were used to determine the physical exposure
at the ear of employees. Stationary measurements next to the
cashier’s area were performed additionally. For psychological
evaluation of working-condition-perception, a validated
questionnaire (BASA II) was used. Furthermore, room acoustic
measurements were performed in representative retail stores as
well as binaural recordings for psychoacoustic analysis
purposes. Outcomes of this project will be presented and discussed for
personal and stationary sound pressure levels. Measurement
method and rating of results in German legal requirements
turned out to be difficult to apply for these types of workplaces,
as will be discussed in detail. In addition, the results of the
measurements and the questionnaire will be reviewed to show
the relationship between perceptual and analytical variables.
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Organization:
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H. Boegli, Federal Office for the Environment |
R. Weinandy, Umweltbundesamt |
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The EPA-Network Interest Group of Noise Abatement (IGNA)
Urs Walker, René Weinandy, Hans Boegli, Nina Mahler
[Abstract]
Noise abatement in Europe has made considerable progress in recent decades. Starting with the Green Paper on Noise Policy in 1996, followed by the adoption of the Environmental Noise Directive in 2002, many noise maps and action plans to reduce noise exposure of the population have been developed. However, the problem of noise has not yet been solved and that is why the EPA Network - the informal group of ”Heads of European environment protection agencies” - decided in 2010 to boost this topic by creating an ”Interest Group on Noise Abatement” (IGNA).
IGNA’s first mandate from 2011 to 2016 focused on best abatement measures at traffic noise, decision criteria for applying noise abatement measures, limit values and monitoring methods. IGNA organised eight workshops and issued four reports covering road, railway and aircraft noise as well as various letters to EU-institutions asking for tighter noise regulations. In its second mandate from 2017 to 2022 IGNA concentrates on all noise sources, limit values in Europe and quiet areas.
The activities of IGNA are leading to a common understanding of how to tackle and solve noise problems and to disseminate the information to players and stakeholders in other countries.
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Recommendations for Traffic Noise Abatement
René Weinandy, Hans Boegli, Urs Walker
[Abstract]
Since 2010 the EPA-Network Interest Group on Noise Abatement has developed various reports with best practice recommendations covering road, railway and aircraft noise as well as decision criteria for applying noise abatement measures. The most important recommendations are as follows:
Road traffic: Further tighten limit values for road vehicles and tyres, extent the knowledge of the public (both users, suppliers and other stake holders) about the noisiness class of products like vehicles and tyres, continue development and application of low noise surfaces.
Railway traffic: substitute cast-iron block brakes with even synthetic material or disk brake and maintain high quality wheel and rail smoothness, introduce stricter noise requirements for all existing freight wagons.
Aircraft traffic: Introduce tighter limit values of noise certification levels, harmonize noise categories of aircraft, establish permanent incentives to optimize flight procedures regarding noise.
Criteria for applying abatement measures: Develop harmonized methods to determine and internalize the costs and benefits of measures to reduce traffic noise.
Despite the best practice list, one has to be aware that the classic noise abatement procedure has to be complemented with new innovative measures in order to face the challenge of future trends and developments. IGNA is working on that.
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Problems of Road Traffic Noise Annoyance and Sleep Disturbance in Some Slovakian Cities
Ladislav Mihalcik, Stanislav Sekretar, Jana Jurkovicova, Jan Simonovic, Lubica Argalasova
[Abstract]
The paper presents the results of a pilot cross-sectional study
focused on subjective traffic noise annoyance and sleep
disturbance among the residents living close to major inner city
corridors in several Slovakian towns including the capital
Bratislava. The subjective adaptation to traffic noise, subjective
evaluation of health status were assessed as well.
The questionnaire survey was distributed to the pilot sample of
543 respondents (average age 45,3 (73,1% respondents from 35
to 65 years), 57% females, 71% living in houses more than six
years). The inhabitants with bedroom windows facing noisy
streets or quite streets represented the exposed and the control
group. Road traffic noise annoys significantly more daily and
night activities of respondents in the exposed group, who are
unable to adapt to it (OR=5,28; 95 % CI=3,56−7,86 for sleep
disturbance).
The questionnaire was supplemented by traffic noise-
measurements in Bratislava on noisy facades (LAeq,day = from
57 dB to 72 dB, LAeq,night= from 52 dB to 62 dB) close to major
streets with intensity transport from about 16000 to 38000
vehicles per day.
After completion of the results, we plan to propose interim
measures to new apartment buildings as well as intervention
procedures.
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Overview of environmental noise limits in the European Region
Rosan Nusselder, Bert Peeters
[Abstract]
In October 2018, the World Health Organization (WHO) published environmental noise guidelines based on dose-response curves for relevant noise-related health impacts. The guidelines provide recommended noise limits, but do not describe what consequences of exceedance should be. Also, it is helpful to know how the currently adopted limit values in European countries relate to the recommended limit values. Our research aims to address these questions. To do so, a survey has been sent out to national experts to collect data regarding current critical noise limits. From the output, an overview has been created which allows a comparison of limit values between countries. Also, a comparison is made with respect to the new WHO recommendations, revealing the current state of noise legislation in the European region. In addition, information has been gathered on the scope and basis of these values, as well as detailed information on their assessment, exceptions and legal consequences. From it, general trends in noise level policy and enforcement methods are extracted. The results provide insight in the broad spectrum of noise legislation within the European region, and form a basis from which to build best practices for countries that aim to implement the WHO guidelines.
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Assessment of Quiet Areas in Europe
Eulalia Peris, Nuria Blanes Guardia, Jaume Fons-Esteve, Miquel Sáinz De La Maza
[Abstract]
Living in a quiet area provides health benefits such as increased
quality of life and well-being. Nearby access to quiet areas can
also offer psychological restoration and can help in reducing
noise annoyance reactions. Quiet areas are not only beneficial
for human health but also help to protect areas of valuable
habitat. The Environmental Noise Directive (END) recognises the
need to preserve areas of good acoustic quality, referred as
”quiet areas”. However, the data reported as part of the END
contains little information on how the countries, regions and
cities define and protect quiet areas in their territories and
whether there has been a significant improvement in
designation and protection of these areas over the past years.
The aim of this paper is to present an overview of how countries
define and protect quiet areas in Europe. This is achieved using
data from case studies comprised of online questionnaires,
which were completed by noise representatives of different
countries, regions and cities. In addition to the analysis of the
questionnaires, a combined spatial assessment of noise
exposure, land use and land cover data for areas potentially
unaffected by noise pollution in European cities is presented for
2012 and 2017.
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Noise Advocacy
Tony Paul Dolan
[Abstract]
Following the completion of the Round 3 noise maps and action plans, the Irish Environmental Protection Agency is focussed on the provision of public information and advocacy around transport noise. The publication of the noise maps was tweeted by our Communications Unit. The noise maps are being used by the various Local Authorities to communicate to the public about the modelled noise levels in their area and how they may be addressed in their noise action plan. It has given the public improved access and a better understanding of what the noise maps are all about. We are also developing a noise briefing paper to give an update on environmental noise, which we will add to the noise mapping information on the EPA website. In addition, we are producing an environmental noise Factsheet/Infographic to help provide better public information on noise. We are also liaising with our Communication team to publish a noise feature article, and some other noise related features during 2019. The intention is to increase public awareness around environmental noise with a view to putting a greater focus on the various LA noise maps and action plans, and their implementation over the next five years.
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New trends in noise abatement
Hans Boegli, Urs Walker, René Weinandy
[Abstract]
The classic noise abatement approach consisted in evaluating the problem by establishing noise maps, evaluating noise abatement measures in form of action plans and keeping the public informed of the progress with a regular monitoring. This was at least the procedure undertaken in Switzerland over the last forty years and the EU follows basically the same procedure. This approach is driven by the belief that there are some efficient technical or constructive measures available to reduce noise emissions. Moreover, one expected that low-noise innovations would inevitably solve the problem and noise as an environmental burden would eventually disappear from the agenda.
However, new trends and developments have made noise abatement aiming at moving targets and facing new battlefields of conflicts. These changes can be attributed to evolution of attitude of human beings, social-demographic changes and progresses in the techno-economical environment. In answer to these developments, the strategy of noise abatement has to be adapted in order to cope with the new challenges. This adaptation should ensure that type of noise abatement measures are less ”end-of-pipe” such as noise barriers or protection windows, but measures at source based on holistic concepts that integrate all relevant aspects of modern life.
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Organization:
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B. Peeters, M+P |
A. Kok, RIVM |
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Flaws in the Cnossos Calculation Method and Proposed Solutions
Arnaud Kok
[Abstract]
January 1 2019 all member states were required to have transposed the Cnossos calculation method (EU directive 2015/996) in their legislation. In preparation a study was conducted in 2017 to evaluate the new calculation method. Part if this evaluation was to see if the method could be used as a replacement for the Dutch national method. In this study we found errors within the method that, if implemented, would lead to implausible results. The results of this study were presented to the EU commission and the Noise Regulatory Committee. As a result an EU working group, chaired by the Netherlands, was established to study and propose amendments to the method. This EU working group has found numerous issues. Some issues are about unclear text, that could lead to different interpretations of the method. Other issues are more fundamental. They are clear errors in the method. One example is a problem that occurs with multiple diffractions in favourable conditions. For almost all the issues a solution was drafted and a report was finalized where these issues and proposed solutions are presented. In this paper we will address the problems with the method and proposed solutions.
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Establishing noise emission for an electric road vehicle category in CNOSSOS-EU
Bert Peeters, Erik De Graaff, Hilary Notley, Simon Shilton, Matthew Muirhead
[Abstract]
The CNOSSOS-EU road noise model described
in EU Directive 2015/996 makes use of model
coefficients that represent the average noise emission
spectrum and speed dependence for each particular
vehicle category. Separate coefficients are provided
for the road surface influence and for
accelerating/decelerating at intersections. Appendix F
provides default values for cars, trucks and two-
wheelers with combustion engines. This study describes a method to establish noise
emission coefficients and road surface corrections for a
new vehicle category. The primary aim is to include
(hybrid) electric vehicles in the noise assessments, to
enable evaluation of EV/HV proliferation scenarios. To
separate rolling and propulsion noise, a combination of
pass-by (CPB) and close proximity (CPX) methods on an
appropriate selection of instrumented vehicles is
proposed. Acceleration and deceleration coefficients
may also be derived from instrumented vehicle
measurements. The total noise emission is calibrated
using roadside measurements on vehicles in regular
traffic. Acoustic transfer functions are calculated to
derive the sound power coefficients from the
measurement results. The methodology provides an approach to adding any
vehicle categories to CNOSSOS-EU, and for a potential
future BSI standard to replace the existing British CRTN
assessment method. Recommendations for further
development of such a standard are given.
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The determination of road surface corrections for CNOSSOS-EU model for the emission of road traffic noise
Fabienne Anfosso Ledee, Luc Goubert
[Abstract]
In the CNOSSOS-EU noise emission model for road traffic noise, the calculation of sound power emission of vehicles introduces a road surface correction term, which can be calculated from a set of values provided for non-standard types of road surfaces. However, the data may be of limited use for the authorities in charge of strategic noise mapping if they do not match with the national or local types of surfaces or if newly developed types of surfaces are laid on their networks. A procedure for the acoustic characterization of road surface properties is currently discussed within the European standardization group CEN/TC227/WG5. This procedure, based on CPX tyre-road noise measurements, could serve in two ways for the evaluation of road surface corrections as input parameters in CNOSSOS-EU. The first is a direct snapshot measurement on an existing network, providing a rapid and up to date evaluation of the road surface correction term, without considering the nature of the surface. The second is a reference measurement to collect reference data to serve for calculations at later stage. The paper will present the proposed method and its compatibility with the existing noise emission model.
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Transposition of CNOSSOS-EU into German Law
René Weinandy, Thomas Myck
[Abstract]
One of the most important environmental issues in densely populated areas is the problem of noise. Traffic noise from cars, railway vehicles and airports located in close proximity to the city is not only annoying for residents; it also leads to serious health issues and has an enormous negative economic impact. In 2002, to improve the noise situation in Europe, the EU issued the Environmental Noise Directive (2002/49/EC). The aim of the Directive is to reduce environmental noise and to prevent an increase in noise in areas which are traditionally quiet. For this purpose, noise calculations are conducted for major roads, major railway lines, and various major airports with a high volume of traffic as well as large agglomerations. Until 2018, for this task a national interim method has been used. In 2019, it is replaced by a new calculation procedure, because the European Commission has developed ”Common Noise Assessment Methods in Europe, CNOSSOS-EU”. These methods have been implemented in Annex II to Directive 2002/49/EC in 2015. The transposition of this calculation procedure into German law will be explained in detail.
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Implementation of CNOSSOS-EU method for road noise in Italy
Gonzalo De Leon, Francesco Fidecaro, Mauro Cerchiai, Marco Reggiani, Elena Ascari, Gaetano Licitra
[Abstract]
The European Directive 2015/996 describes the common
methods (CNOSSOS-EU) for the determination and
management of environmental noise at European level,
providing guidelines for a harmonized approach for Member
States.
CNOSSOS-EU includes different emission data for road, railway
and industrial noise and provides a common model for sound
propagation. The emission model was derived to be valid under
reference conditions in terms of meteorology and traffic as
detailed in the reference report by the Joint Research Centre
(JRC) of the European Commission. In order to take into
account situations that differ from the reference one,
appropriate corrections were also offered.
For a correct application on a local scale, the default values
suggested by CNOSSOS-EU are required to be redefined in
order to more adequately describe the Italian reality (i.e. car
fleet, surfaces, climate conditions, etc.). Regional Agency for
Environmental Protection of Tuscany (ARPAT) carried out
specific measurements campaigns to acquire data useful to
validate the methods for Italian conditions.
At first, regression coefficients used in the emission model, were
recalculated from campaigns in 10 different sites carefully
chosen to resemble the reference conditions. Finally, different
surfaces typologies were studied, with the purpose of obtaining
their proper correction factor.
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Matching noise emission from French medium-heavy vehicles and CNOSSOS models
Marie-Agnès Pallas, Adrien Le Bellec, David Ecotiere
[Abstract]
The CNOSSOS-EU prediction method, mandatory since 2019 for noise mapping in compliance with the European Noise Directive, specifies the road vehicle noise equations to be used per vehicle category. French noise emission data differ significantly from the European model and a national decree was published in June 2018 to provide CNOSSOS corrective coefficients in accordance with the national NMPB2008 model. However, the European semi-heavy vehicle category is non-existent in both French emission model and databases, thus lacking to support the proposed coefficients at this stage.
To provide missing information, detailed measurements under controlled conditions have been performed on a sample group of semi-heavy vehicles. The contribution of propulsion noise and rolling noise were separated and average emission equations were proposed by subcategory. For two traffic composition scenarios, mainly differing by the ratio of public transport vehicles, a noise emission model has been proposed to represent an average medium-heavy vehicle. The paper presents the approach and the models obtained. The representativeness of the European prediction model in its initial version published in 2015, its French adaptation of June 2018 and the European correction discussed in 2018, is considered for both scenarios. Comparisons of frequency characteristics are emphasized.
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The First Agglomeration Noise Maps in Niš, Serbia, Using CNOSSOS-EU:2015
Simon Shilton, Rafdouglas Tommasi, Jiri Michalik, Álvaro Grilo, Kristina Peric, Ivana Krstic
[Abstract]
Strategic noise mapping across cities has been undertaken
under EC Directive 2002/49/EC since 2006 using a range of
national and EC recommended Interim Methods. Under EU
Directive 2015/996 the new common methods were introduced
which have been the required methods of assessment since 31
December 2018. Within the EuropeAid funded project for
strategic noise mapping in City of Nis agglomeration, CNOSSOS-
EU:2015 has been used to develop the first city-wide results in
the Republic of Serbia. The project has included: the collection
of input data from a wide range of National, Regional and City
authorities; development of the noise models and noise
calculations under CNOSSOS-EU and presentation of the results
to the public using a hosted SaaS solution; and verification of
the results using long term noise measurement surveys. Based
upon the experience gained in Nis, the project is developing
guidelines for strategic noise mapping in Serbia using the
CNOSSOS-EU:2015 methodology, which will be used as the basis
for conducting training of state and municipal authorities in
noise mapping in connection with future membership of the EU.
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Organization:
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S. Luzzi, Vie en.ro.se. Ingegneria |
M.A. Burgess, UNSW Australia |
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Model for total noise assessment under consideration of source specific exposure-response ralationships
Manfred Liepert, Johannes Lang, Ulrich Möhler, Dirk Schreckenberg
[Abstract]
Although most of the German population is exposed to multiple
noise sources, noise legislation in Germany is subdivided in
regulations for each noise source. In order to improve the quality
of life there is a lack of a total noise assessment. The German
Federal Environmental Agency therefor initiated the research
project ”Model for total noise assessment”. As a part of this
project the method of the German VDI standard 3722-2 on the
assessment of the impact of noise from multiple sources was
further developed. With the help of exposure - response
relationships
for individual source types a total noise assessment for road
traffic, railway, aircraft and industrial noise is possible. For
this purpose the exposure-response relationships of the new
WHO Environmental Noise Guidelines are used. The possibility of
considering health
effects and combining annoyance ratings and health effects
are discussed.
In a second step of this project noise mitigation in case of
multiple noise sources are investigated. Different methods for
the cost distribution were examined regarding their advantages
and disadvantages in the application. The proposed method of
cost distribution was finally tested in case studies using the
further development of the VDI standard.
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Infrasound and low-frequency noise immission. Structural Vibrations Induced by industrial noise. Improving the ISO 1996- 2:2017 in order to propose a plausible standardized procedure for using in Legal noise assessment
Walter Alfredo Montano, Elena Gushiken
[Abstract]
It is well known that dBA measurements don’t are the best tool
to evaluate an alleged noise/vibration disturbance, mainly
when impacted dwellings are in a mixed acoustical zones. The
results of the measurements, made in Peru inside and outside
dwellings subjected to structural vibrations, are presented here.
The issue is that there is not always a simple way to distinguish a
specific sound from the total sound, and methods
recommended in the ISO 1996-2 Standard are very general. An
empirical criterion had to be created to find the level of specific
sound contained within the total sound. These acoustic
determinations consist of removing the outlier values from a
time-history filtering of the total sound. In this work, the author
proposes a new method of employing statistical tools, adapting
Pearson’s criterion, using discrete mathematics, inference, and
acoustics criteria in order to develop a computer program to
remove spurious noise with not complex algorithms. The purpose
of this Paper is to share to the acoustician community that it is
plausible to achieve a standardized method, in the sense that
an algorithm could be agreed among specialists for having an
”universal computer program” to determine the specific sound
contained in the total sound.
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A New Approach for Road Traffic Noise Mapping Using Big Data
Gaetano Licitra, Antonino Moro, Luca Teti, Alessandro Del Pizzo, Francesco Bianco
[Abstract]
The present paper describes a new approach developed
within the BEEP project (Big data for Environmental
and occupational EPidemiology) aimed to improve road
traffic noise mapping in epidemiological studied.
The BEEP project, funded by the National Institute for
Insurance against Accidents at Work (INAIL), aims
at using big data for the evaluation of negative health
effects due to air and noise pollution on the Italian
general population and the risk of occupational injuries in
sub-populations of workers.
Noise maps provide noise emissions that are usually
calculated by traffic flows measured or derived by a
model. The evaluation of traffic flows can determine
significant uncertainty in noise estimate: accurate
traffic data can significantly improve the meaningfulness
of noise models.
In this paper, a new method based on Google API and Big
Data treatment was developed to estimate traffic
flow and produce noise maps of Rome agglomeration.
Correlation between travel time and traffic flow
considering road characteristics was found using street
clustering. Noise maps obtained were compared to those
produced by conventional means; results show
that the use of big data could improve traffic estimates, in
particular during the night period, which is well
known to be strongly related to health issues.
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Organization:
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D. Schreckenberg, ZEUS GmbH |
T. Yano, Graduate school of Science and Technolody, Kumamoto University |
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Noise Annoyance – What does it mean?
Paul David Hooper, Ian H. Flindell
[Abstract]
Current uncertainty regarding to what extent chronic noise annoyance contributes to long term adverse health effects (or not) is not helped by general confusion about what is actually meant by ëchronic noise annoyance’ and the best ways to measure it. The problem arises because the term ënoise annoyance’ means different things to different people in different situations and contexts. The solution to this problem depends on understanding the different situations and contexts in which these terms may be used; understanding the often different requirements of people who use these terms; and reflecting these differences both in the measurement instruments which are actually used and in the constraints and limitations which should be placed on the interpretation of those measurements in different situations and contexts. In the paper we provide practical examples to illustrate how this problem arises and how it can be solved in real-life situations and contexts.
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Measuring Noise Annoyance with Multiple Question Scales - an Overview
Jördis Wothge
[Abstract]
Noise annoyance is a diverse concept. This is reflected by the great variety of different definitions and understandings that exist in the noise impact research community. Noise annoyance can be defined as an emotion, a rational decision, knowledge, an attitude or the result of a disturbance. Noise annoyance can also be regarded as a multi-dimensional construct that entails some or all of the aspects mentioned above. Currently, most studies on noise annoyance make use of the widely accepted single-item ICBEN question to measure noise annoyance. There is, however, an ongoing discussion in the research field whether a single-item question can comprehensively cover all facets of noise annoyance. This lead to the postulation of multi-item questionnaires to measure noise annoyance in order to reflect the multi-dimensionality of noise annoyance more adequately. The current contribution aims to give an overview about different multiple question scale approaches to measure noise annoyance.
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Assessment of the impact of changes in noise exposure at an expanding airport by means of the multiple item aircraft noise annoyance scale (MIAS)
Dirk Schreckenberg, Rainer Guski, Julia Haubrich, Jan Spilski
[Abstract]
The Multiple Item Noise Annoyance Scale (MIAS) measures noise annoyance as a multidimensional concept including 1) the experience of an often repeated noise-related disturbance and the behavioural response to it, (2) an emotional/attitudinal response to the sound and its disturbing impact, (3) the perception of control of the noise situation. MIAS consists of seven items including the 5-point ICBEN noise annoyance scale and the sub-dimensions ënoise disturbances’ and ëlack of coping capacity’, each consisting of three items. Its psychometric properties are investigated for aircraft, road, and railway noise annoyance. In this study, we analyse changes in values of the sub-dimension of MIAS for aircraft noise annoyance before and after the opening of a new runway and the implementation of a night curfew in communities around Frankfurt Airport. Previous studies have shown evidence of a shift in exposure-response relationships for annoyance in situations of changes in noise exposure (change effect). With the assessment of MIAS and its components it can be identified, whether the change effect is more pronounced in the ëdisturbance’ component of annoyance or in the lack of capacity to cope with the (new) noise situation. This improves the understanding of annoyance in situations of changes in noise exposure.
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Forty-five years of surveys on annoyance from road traffic noise
Truls Gjestland
[Abstract]
The Miedema & Vos curve for predicting road traffic noise annoyance was based on 18 different surveys conducted over a period of 9 years. The curve was developed using a statistical regression technique. An extended analysis comprising 46 surveys conducted between 1969 and 2013 yields a very similar dose-response relationship. This analysis was based on the CTL method (Community Tolerance Level). There are no obvious temporal trends, and contrary to claims by other researcher the prevalence of highly annoyed residents exposed to road traffic noise today is the same as 45 years ago.
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Laboratory Studies of Annoyance
Sonoko Kuwano, Seiichiro Namba
[Abstract]
It is said that in the concept of annoyance non-acoustic factors are included as well as physical factors of sounds. Therefore, it is difficult to investigate annoyance in laboratory studies where the situation is somewhat different from daily life situations.
Annoyance is usually asked in social surveys. Laboratory studies have merits that various parameters are precisely controlled The authors tried various laboratory studies of annoyance by developing and controlling the experimental methods. One of them is the method called ”subject-interrupted noise source method”. The study using this new method and other laboratory studies of annoyance will be introduced in this paper.
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Perception of low-level Sound Sources in Everyday Situations
Siegbert Versümer, Jörg Becker-Schweitzer
[Abstract]
Noise research has focused on urban sounds like traffic or machine noise. The present paper, in contrast, investigates low-level sound sources that can affect human wellbeing and health and that are only perceived in specific situations.
To address our research questions, we conducted an online survey with around 2000 participants representative for the German population. In the course of the survey, participants reported on situations in which they typically noticed low-level sound sources. They described the characteristics of the sounds as well as their frequency of occurrence. In addition, they reported on typical activities in these specific situations and personal strategies to cope with negative effects. Finally, they filled out questionnaires measuring the Big Five personality traits extraversion and neuroticism as well as noise sensitivity.
The results include a ranked list of sound sources that people classify as being ”low-noise” and that show a potential to affect humans in a negative way. They further constitute the basis for a multi-level prediction model on the effects of low-level sound sources including situational and person-related factors. Therefore, the findings contribute to a deeper understanding and to an improvement of low-level acoustic environments.
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Community response to high-speed railway noise in Tianjin, China
Lan Zhang, Hui Ma
[Abstract]
Adverse effects of noise from high-speed railway have
become more
and more serious and widespread with the rapid
development of
high-speed rail in China. A social survey on community
response
to high-speed railway noise was carried out in Tianjin,
China
from March to July, 2018 along the Beijing-Tianjin intercity
high-speed rail line and Beijing-Shanghai high-speed rail
line.
674 residents were interviewed by questionnaires to
explore their
noise annoyance, activity interference caused by high-
speed
railway noise, sensitivity to noise, and satisfaction with living
environment. The exposure of high-speed railway noise
was
measured for 24h at each site, and standardized questions
and
scales suggested by ICBEN Team 6 were used in the social
survey.
As a result, the dose-response relationship curve for high-
speed
railway noise of Chinese residents was established, and it
showed
that with the increasing of noise level (Ldn), annoyance
increased exponentially. Compared with Japanese and
French
people, Chinese residents showed more annoyed to high-
speed
railway noise. It also suggested background noise had
important
impact on noise annoyance evaluation to high-speed
railway.
Furthermore, annoyance caused by high-speed railway
noise was
also greatly influenced by subjective characteristics such
as
noise sensitivity, satisfaction of living environment and
attitudes.
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Using mobile application to assess quality of acoustic and visual environment in relationship with aircraft noise
Ferenc Marki, Catherine Lavandier, Dirk Schreckenberg, Stephan Grossarth
[Abstract]
Aircraft noise annoyance has been studied for more than 50
years, asking people to rate their annoyance on a
unidimensional scale. The importance of non-acoustic factors
shows that a more global approach could help to understand
how people perceive their acoustic environment. Inspired by
the soundscape studies, and in the frame of the ANIMA
European project on the impact of aviation noise, an
Experience Sampling Method with signal-contingent sampling
is
chosen to question people in their daily live about the quality
of
their acoustic and visual environment, and not on the
annoyance. A dedicated mobile application is developed,
which collects data each hour of a day (night periods are
excluded).The experiment lasts two or three weeks
depending
on the number of notifications per day the participant found
acceptable. After the notification, participants are asked to
record acoustic measurements (third octave bands, each
second during one minute). After the questionnaire,
participants
are asked to take a picture of the environment. Summary
questionnaires are filled at the end of each week, and a
global
questionnaire is filled at the end of the experiment. This
paper
focuses on a pilot study whose aim is to evaluate the
acceptability of the method by participants.
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Assessment of Short-term Annoyance due to Shooting Noise Using the Experience Sampling Method
Stephan Grossarth, Dirk Schreckenberg
[Abstract]
Experience Sampling or Ecological Momentary Assessment allows researchers to assess behavioural and psychological phenomena directly in the situation of occurrence/in a given situation. In a longitudinal study, we aim to model short term annoyance due to shooting noise as a function of intra- and inter-individual differences in perceived annoyance by using a hierarchical data modelling approach. The MovisensXS app is used to collect data from three different study sites in Germany from 80 Participants. They are asked to participate within two weeks for a period of five consecutive days per week and, during this time, to fill out a survey, consisting of 19 items, five times a day at set intervals. This will add up to approximately 4,000 single measurements, depending on the response rate and the compliance. We describe the methodology of the study, the planning and executional steps taken to ensure a proper data quality. Furthermore, we will explain what has been done to prepare for the data analysis. Finally, we will give a forecast into data analysis and will exemplarily talk about longitudinal data analysis and linear mixed effect modeling.
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Organization:
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A.L. Brown, Griffith School of Environment, Griffith University, Australia |
T. Van Renterghem, Ghent University, Department of Information Technology, WAVES research group |
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Comparing noise policies of 8 European cities using a noise intervention classification scheme
Timothy Van Renterghem, Hannelore Hernalsteen, Alan Lex Brown
[Abstract]
Over the last decades, cities have been developing noise policies. Comparing such efforts, however, is not a simple task. For this purpose, a previously proposed noise intervention classification scheme is used as a framework. This framework discriminates between source interventions (type A), path interventions (B), closing/opening new infrastructure (C), other physical interventions (D, e.g. achieving a quiet side) and communication/education measures (E). Responsible officers dealing with noise at environmental departments of 8 large European cities were interviewed. The invitations for an interview were sent out taking into account the geographical distribution of cities over Europe while a reasonably high position on a sustainable cities ranking was required. The interviews focused on the noise problem definition and what is in the noise management toolbox of the city. In addition, it is assessed how noise policies are evaluated, and to what extent public participation in the noise policy and planning process occurs. The interviews show highly contrasting opinions and practices among the interviewed cities at almost any aspect of the noise management process.
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Are perceived noise control and its value related to behavioural determinants of residents’ civic engagement? A cross-sectional study among older adults
Natalie Riedel, Adrian Loerbroks, Joachim Scheiner, Susanne Moebus, Irene Van Kamp, Thomas Claßen, Heike Koeckler, Gabriele Bolte
[Abstract]
We had suggested that residents’ perception and value of being able to control one’s personal traffic noise exposure at home are relevant for the readiness to participate in noise policies. Further, we had framed residents’ participation readiness by specifying behavioural determinants (positive attitude towards and subjective norm of taking action for a quiet environment as well as behavioural control perceptions, i.e. noise-related institutional knowledge, self-efficacy, and communal mastery). Using cross-sectional questionnaire data on about 1,600 older adults in Germany in 2016 and logistic regression analyses, we modelled the probability of agreement to each behavioural determinant in relation to residents’ perceived noise control (PNC) and its value as measured by six point Likert-scales. Besides other covariates, knowledge of public consultation in noise action planning, internet access, and social network were added to the models. Agreement varied from 36% for a positive subjective norm to 72% for a positive attitude. Overall, results confirmed positive associations between PNC, its value and behavioural determinants (e.g. ORPNC= 1.21, 95%-CI 1.09-1.35, ORvalue=1.20, 95%-CI 1.13 1.28 for behaviour-specific self-efficacy), indicating a link between perceived exposure-specific characteristics and participation readiness. Next, we will investigate whether our selected behavioural determinants statistically predict intended and performed civic engagement.
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Soundscape approaches in urban planning: implications for an intervention framework
Irene Van Kamp, Alan Lex Brown, Dirk Schreckenberg
[Abstract]
There is increasing demand for innovative approaches to decrease the negative impact of noise by sound-conscious design. While the health effects of environmental noise are well mapped, little is known about the effectivity of interventions on health. This is one of our conclusions in the WHO-review on health impacts of noise interventions. More and better interventions studies are needed, in particular for sources other than road traffic, and for health outcomes other than annoyance. Soundscape interventions were in this review classified under the category” Other physical interventions”. Three out of four studies in this domain demonstrated potential efficacy. Although soundscapes seem essential for urban planning, guidelines on how to implement this knowledge are missing. Against this background an inventory of architectural soundscape projects in the Netherlands and elsewhere was made, combining interviews with experts and planners with a literature review. This led to the conclusion that well documented approaches all combined changes in the physical environment, with stakeholder involvement. In that way a sense of ownership was created by the interventions, forming an additional category in the typology of interventions as canvassed earlier. This paper suggests some extensions the typology of interventions.
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Road Traffic Noise Interventions: Development of a Method to Quantify their Effects on Annoyance and Sleep Disturbance on a Small Urban Scale
Maud Dohmen, Maarten Hornikx, Irene Van Kamp
[Abstract]
Road traffic is the main source contributing to noise exposure in
urbanized areas. With the rapid urbanization and the increasing
attention on the way sound can affect health and well-being,
the question is how to shield ourselves from harmful effects. The
most common noise intervention research studies regard only
large-scale interventions, like noise barriers along a highway, the
effect on health and well-being are often based on rough
estimations of sound reduction. In this study, the effect of a small
scale intervention on annoyance and sleep disturbance will be
investigated, considering relative changes in dose-response
relationships to account to local effects. It will give more insight
into the subtle changes due to interventions on a local scale,
providing more detail on top of existing conventional studies. To develop this methodology an intervention study in Rotterdam
is taken as a test case. The location and its intervention are
modeled using 3D-geometrical acoustic software (Olive Tree
Lab). The predicted noise reduction due to the interventions, as
well as the predicted health effects are compared with
measured data and local surveys. The results shed a light on the
suitability of this approach to predict health effects due to
interventions at a local scale.
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Organization:
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S. Yokoyama, Kobayasi Institute of Physical Research |
F. Van Den Berg, University of Groningen |
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How do audio and visual characteristics of wind turbines contribute to noise annoyance?
Beat Schäffer, Reto Pieren, Ulrike Wissen Hayek, Nadine Biver, Adrienne Grêt-Regamey
[Abstract]
Wind farms often evoke strong annoyance reactions in residents. Literature suggests that both acoustical characteristics and the visibility of wind turbines may contribute to noise annoyance. However, studies on the mutual audio-visual effects on annoyance are still rare. The objective of this study was to investigate the short-term noise annoyance reactions to different wind turbine noise situations in a controlled laboratory experiment. A set of 24 audio-visual rural scenarios containing a single wind turbine was synthesized, i.e., visualized and auralized. Combined with the full factorial design of the experiment, this allowed separating the individual contributions of the following variables to noise annoyance: distance to the wind turbine, periodic amplitude modulation of the sound (with, without) and visual setting (landscape with visible wind turbine, landscape only, grey background). The experiment revealed that both visual and acoustical characteristics strongly affect noise annoyance. In particular, annoyance increased with periodic amplitude modulation and/or when the wind turbine was visible, but decreased when only the landscape was visible. While the acoustical effects could be reliably assessed, the visual effects were less straightforward to reveal, as they are afflicted with carryover effects. The presentation order of audio-visual stimuli may therefore be crucial for study outcomes.
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Implementation Of The Issue Of Noise From Wind Turbines At Low Frequencies
Martino Marini, Costantino Carlo Mastino, Roberto Baccoli, Andrea Frattolillo, Antonino Di Bella
[Abstract]
The enduring energy scenario leads to further promote the development of the exploitation of renewable energy sources. Recent European standards, have defining a path to reach in 2050 a level of decarbonization lower of 80% compared to 1990. Wind farms have been growing quickly for last decade with individual wind turbines getting larger and larger. In addition to the above benefits drawbacks has arisen, due to noise generation from wind turbines and adverse reaction of some nearby residents. The noise generated by wind turbines has a broad spectrum character but the low frequency noise causes special problems.It is a fact that in different European countries special laws have been adopted to impose noise limits and evaluation methods for the assessment of environmental low frequency noise from this kind of sound sources. Other countries are still lacking specific rules but in the authorization procedure such analysis is required by environmental control agencies. The purpose of this study consists of comparing the assessment procedures currently used in different European countries for the prediction of low frequency noise from wind turbines and its propagation. The comparison of procedures gives a chance to put forward progressions in low frequency noise emission and reception.
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Pilot study on perceived sleep acceptability of low-frequency, amplitude modulated tonal noise
Kristy Lee Hansen, Phuc Nguyen, Branko Zajamsek, Gorica Micic, Peter Catcheside
[Abstract]
The presence of amplitude modulation (AM) in wind farm noise has been shown to result in increased annoyance. An equivalent unmodulated noise would have to be louder by several decibels to be as equally annoying as modulated noise. Despite this finding, many standards and guidelines do not specify a penalty for the presence of AM. This study investigates the prevalence of AM over a 1 year period at four different residences located between approximately 1 km and 3.5 km from three different wind farms. The analysis is based on continuous measurements of raw time data that were collected using remote monitoring systems. Preliminary results will be presented based on the data that has been collected and analysed to date. As well as considering the percentage of time that wind farm AM is present, this investigation considers other factors such as the AM depth and wind farm power output. The results provide insight into the prevalence and characteristics of AM at various distances from a wind farm over a significant period of time.
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Epidemiological study on long-term health effects of low-frequency noise produced by wind power stations in Japan
Tatsuya Ishitake, Yoshitaka Morimatsu, Kunio Hara
[Abstract]
We investigated whether long-term exposure to wind turbine noise (WTN) including
low-frequency noise generated by wind power facilities is a risk factor of sleep
disorders. We performed an epidemiological study of living environment and health
effects, surveying 9,000 residents (≥20 years) living in areas with operational wind
power facilities. Sleep disorders were assessed using the Athens Insomnia Scale. To
assess environmental noise in residential areas near the wind turbines, low-frequency
sound exposure levels were measured at 50 community centers of the town. Multiple
logistic regression analysis was used for evaluation of a risk factor for several noise
exposure indices. Significant relationships between the distance from the nearest WT to
dwellings and hearing, annoyance, sleep disorders were observed. By multiple logistic
analysis the prevalence rate of sleep disorders was significantly higher for residents who
reported subjectively hearing noise being than for those who did not. Moreover, the
reported prevalence rate of sleep disorders was significantly higher in residents living at
a distance of ≤1,500 m from the nearest wind turbine compared to that for residents
living at a distance ≥2,000 m. The attitudes of residents towards wind power facilities
and sensitivity to noise strongly affected their responses regarding sleep disorder
prevalence.
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Perception of low-frequency components contained in general environmental noises including wind turbines
Sakae Yokoyama, Tomohiro Kobayashi
[Abstract]
The influence of low frequency components in noises including wind turbine noise has become a serious problem and various researches are being made in many countries. Also in Japan, synthetic study programs mainly targeted for wind turbine noise have been performed, in which field measurements, social survey on the response of nearby residents and laboratory experiments on audibility of low frequency sounds were conducted. In parallel to these studies, field measurements on general environmental noises including transportation noises inside and outside of various vehicles were performed by paying attention to low frequency components. As a result, it has been found that low frequency components are included not only in wind turbine noise but also in general environmental noises. The environmental sounds recorded on sites were reproduced by a test facility which can reproduce low frequency components including infrasound and auditory tests were performed. As a result, it has been found that the A-weighted sound pressure level is robustly applicable to the assessment of loudness for such kinds of environmental sounds and the low frequency components contained in wind turbine noise around residential area is not audible.
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How to present pure-tone infrasound to the ear
Holger Joost, Björn Friedrich, Jesko Verhey, Thomas Fedtke
[Abstract]
An increasing number of individuals are being exposed to infrasound. It is, however, still not clear, how infrasound is perceived by human beings. Regarding auditory perception, one hypothesis is that non-linear processes within the ear generate distortion products that make infrasound audible. To investigate this hypothesis, it is crucial to rule out that such distortions were produced technically, i.e., by the stimulus presentation system rather than the auditory system. This paper introduces a compact, low-distortion infrasound system, based on a commercially available audiometric earphone, for simultaneous presentation of pure-tone infrasound and audio sound to the ear via an ear insert. This system also allows for the recording of sound in the ear canal during signal presentation. Performance characteristics of the system for the stimulus presentation and the sound recording units are presented. It is shown that the interaction of infrasound and audio sound, such as modulation of the audio sound by the infrasound, is negligible with respect to auditory perception.
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Frequency characteristics of oppressive and vibratory feeling to low-frequency sound
Makoto Morinaga, Ippei Yamamoto, Tomohiro Kobayashi, Koichi Makino, Hiroaki Ochiai, Hideki Tachibana
[Abstract]
Although the threshold for oppressive and vibratory feeling for
low frequency sound are widely known, the experiments based
thereon were conducted about 40 years ago in Japan, and
there are some questions to be examined. For example,
although the term ”oppressive and vibratory feeling” is generally
unfamiliar words, it is unknown how the instruction was provided
to the participants about the meaning of the term. In the
present study, an experiment using similar method to the
previous study was conducted, and the threshold of the feeling
was reviewed. Using one-third octave band noises with the
center frequency in the range from 10 Hz to 630 Hz, laboratory
experiment was conducted to investigate the value of sound
pressure level where oppressive and vibratory feeling
preferentially appeared for each frequency. Participants were
divided into two groups; one is the group provided instruction
for the meaning of oppressive and vibratory feeling and the
other is without such detailed instruction, and the results were
compared. It was suggested that the sound pressure level
where the feeling preferentially occurs is slightly different
between the two groups. Furthermore, it was found that the
feeling appeared even in the higher frequency band
compared with the previous study.
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The effect of brown and black noise on persons suffering from a low frequency sound
Frits Van Den Berg, Karin De Boer
[Abstract]
Some people suffer from hearing an often untraceable low-pitched sound. To help them two types of low frequency noise (brown and black noise) were put on CD. Sufferers could obtain this free CD on request. Over the course of more than 10 years about 200 CD’s were sent out. In these 10 years 26 applicants have given a spontaneous reaction by e-mail after having used the CD for a shorter or longer period of time. 23 of these comments were (very) positive, 2 comments stated the CD did not help. In 2018 a retrospective survey was designed for those that had requested a CD before November 2018. The main goal of this study is to investigate how the complaints have changed over time in relation to the use of the CD. Further questions examine which other measures participants have taken and how they use(d) the CD in practical terms (brown and/or black noise, playing time, frequency of use, volume). The study will start in February 2019 and a first analysis will be given in the paper. This may inspire others to help sufferers by using the same method. This could be an opportunity to include more participants in different areas.
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Equal annoyance contours at frequencies 4 - 1000 Hz
Ville Rajala, Jarkko Hakala, Valtteri Hongisto
[Abstract]
BACKGROUND. Knowledge of human perception of infrasound is very limited. Only few publications deal with the hearing threshold and subjective loudness of infrasound. Our aim is to create equal loudness contours extending to infrasound frequencies and to examine hearing threshold of infrasound by using sine tones in frequency range starting from 4 Hz. METHODS. A psychoacoustical experiment involved 20 participants in the 19-26 years age group. The experimental sounds were sinusoidal tones in frequency region 4-8000 Hz. The participant was located in a pressure chamber. Low frequency tones were played by using a special loudspeaker system. High frequency tones were played with headphones. The hearing threshold was determined by using staircase method. Loudness levels 20, 40 and 60 phon were defined by using randomized maximum likelihood sequential procedure. The equal loudness contours will be calculated by using methods described in ISO 226. RESULTS. The experiment is ongoing and the results will be available until the deadline of the full paper (1 June, 2019).
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Portable Infrasound Monitoring Device with Multiple MEMS Pressure Sensors
Ryouichi Nishimura, Zhenglie Chi, Yôiti Suzuki
[Abstract]
MEMS sensors are mass-product and therefore expected to be available to produce a low cost device for infrasound monitoring. A low cost device is useful in constructing a sensor network with various scales from a small to large number of sensors. To test its feasibility, we developed a device consisting of 32 MEMS pressure sensors. A laboratory experiment showed that internal noise increases as signal frequency goes high but the noise can be reduced by simple averaging. Based on this fact, we stacked eight MEMS pressure sensors on a single board to assemble an infrasound monitoring device which enable to detect a signal with amplitude of more than approximately 5 Pa. A field test was carried out, deploying them in the surrounding area of an active volcano, Sakurajima in Japan. Signals associated with explosive eruptions were clearly observed in the frequency region between 0.1 Hz to 5 Hz. The source position was also estimated using the signals observed at four different points. The estimated source position showed in general a good agreement with that of the crater of the volcano. It was also implied that signals around the acoustic cut-off frequency can also be observed.
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Infrasound research activities in Kochi University of Technology - Infrasound observation network by using multiple comprehensive infrasound sensors and its application for disaster mitigation as well as safe human life -
Masa-Yuki Yamamoto
[Abstract]
Infrasound research activities in Kochi University of Technology will be presented. Infrasound observation network by using multiple comprehensive-type infrasound sensors was established recently in Kochi, Japan.
Main purpose of the sensing network is for disaster mitigation/prevention mainly for tsunami, indicating a good application of infrasound sensing.
If we can detect infrasonic waves before coming real tsunami waves to sea shore, it can effectively be used for saving human lives from the flood of sea water.
However, the sensors can also detect many types of natural and artificial low-frequency audible sound and infrasound. It means the network sensing system can be applied to safe human life by means of emergency level alert to local citizens like weather forecasts.
Moreover, we recently tried to evaluate wind turbine low-frequency audible/infrasonic noise affection for human health. In a vicinity of two wind turbines we measured sound field environment by using audible/infrasonic sensors together with biological information sensing system with multiple methods with respect to more than 30 human subjects/testers. With recent progress of biological information sensor technology, we can use them freely outside but many kinds of artifacts can also affect real signals of heart beat and brain wave measurements like our noise hearings.
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Organization:
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R. Guski, Ruhr-University Bochum |
M. Morinaga, DFEIA |
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Short-term annoyance from nocturnal aircraft noise in children: The influence of acoustical and non-acoustical factors
Julia Quehl, Susanne Bartels, Rolf Fimmers, Daniel Aeschbach
[Abstract]
Acute psychological effects of aircraft noise have mostly been investigated in healthy adults so far. Recent noise effects research is increasingly focusing on individuals particularly sensitive to harmful noise effects. These vulnerable groups include especially children. Almost no studies exist on effects of nocturnal aircraft noise on short-term annoyance response in children. We aimed at closing this research gap by means of a field study conducted around Cologne-Bonn Airport in 51 children aged 8 to 10 years. Children were examined during four consecutive nights. During all nights, aircraft noise exposure was recorded inside the children‘s bedroom. Aircraft noise metrics were calculated based on these measurements. Participants retrospectively rated short-term annoyance each morning via a five-point scale. Information on relevant moderating psychological factors (e.g., perception and evaluation of air traffic in the residential area, personal variables including sensitivity to noise) was obtained in personal interviews on the first study day. Logistic regression analysis was used to develop new exposure-response curves as a function of different acoustical and non-acoustical variables. Unlike adults, children’s short-term annoyance was not influenced by noise metrics such as number of nocturnal aircraft noise events. Noise sensitivity and attitudes towards air traffic, however, had a significant effect.
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Noise-induced Annoyance due to Nocturnal Road Traffic: Results of a Field Study
Sarah Weidenfeld, Eva-Maria Elmenhorst, Sarah Sanok, Uwe Müller, Daniel Aeschbach
[Abstract]
Since many traffic noise studies are based on laboratory surveys or exhibit shortcomings in noise measurement, there is a lack of valid exposure-response relationships between traffic noise and annoyance. To fill this gap for road traffic, precise measurements of noise parameters (e.g. energy equivalent sound pressure level [LAeq]) are needed. In this field study 40 healthy participants (mean age = 29.1; SD = 11.7; 26 females) rated their annoyance due to moderate road traffic noise exposure during the previous night on four consecutive mornings. Further questionnaires were used to measure psychological, sleep-related and subject-related variables. Measurements of acoustic parameters were undertaken inside the residents’ bedroom. A generalized estimating equation model (GEE) revealed a statistically significant rise in the annoyance with increasing LAeq of the traffic noise (p = .023). The extent of the subjectively perceived noise exposure had a reinforcing effect (p < .001), whereas a high subjective sleep quality had a reducing effect (p = .007) on annoyance. The present study emphasizes the importance of sleep in annoyance responses and delivers a prediction model for noise-induced nocturnal short-term annoyance in areas with moderate road traffic. Prospective investigations should be extended to residential areas with dense road traffic.
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Effects of The Number of Noise Events from SHINKANSEN Railway on Annoyance
Keiji Yagawa, Ayumi Shiotani, Toshihiko Matsui, Takashi Yano
[Abstract]
This paper describes the relation between the number of noise events of Shinkansen and the increasing noise annoyance. In Japan, the maximum noise level (LA,Smax) is adopted as the noise index for the environmental quality standards for Shinkansen railway noise since 1975. As the number of trains has increased with the development of Shinkansen railway network, it has been debated which of LA,Smax or energy averaged indices such as LAeq,24h and Lden is more appropriate for the index. The Ministry of Environment initiated nation-wide social surveys on Shinkansen noise in 2014 and 2017 in the areas along Shinkansen railways which had 40 to 340 Shinkansen service a day. The total sample size was 1455 and LA,Smax ranged from 53 to 84 dB. We hypothesized that annoyance for Shinkansen railway noise increased according to the increase of noise events even if maximum noise levels are equal. It was confirmed that the number of noise events increased annoyance for Shinkansen railway noise up to around 200/day, but the effect saturated in the range more than 300/day. Comparing logistic regression models between high annoyance and LA,Smax/Lden, either of LA,Smax or Lden is not always superior to the other.
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Leq + X: Re-Assessment of exposure-response relationships for aircraft noise annoyance and disturbances to improve explained variance
Julia Haubrich, Mark Brink, Rainer Guski, Ullrich Isermann, Beat Schäffer, Rainer Schmid, Dirk Schreckenberg, Jean Marc Wunderli
[Abstract]
There is evidence from literature that an increased fraction of people is highly annoyed by aircraft noise by given average noise levels than in previous decades. Among potential reasons are changes of the air traffic over the last decades such as the increased number of movements along with a less noisy fleet mix that might be relevant for noise responses but are not adequately reflected by average noise metrics. In the Leq+X project, data of two German and two Swiss field studies comprising aircraft noise exposure and annoyance data of about 37’700 residents living around altogether seven airports are re-analysed. Among others, the analyses include logistic regression models of the percentage (highly) annoyed, (highly) sleep disturbed and (highly) disturbed in communication on the Leq (day/night), on the Leq combined with either maximum sound level (Lmax) or number of events above threshold (NAT), and on the Lmax combined with NAT for different thresholds. Further models include the Leqk with non-energy-equivalent parameters k = 20 or 30, which weight the number of events more strongly than the energy-equivalent Leq (k = 10). In this paper, the methodological approach and first results will be presented.
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Do we need different aircraft noise metrics to predict annoyance for different groups of people?
Jan Spilski, Kirstin Bergström, Ulrich Möhler, Thomas Lachmann, Maria Klatte
[Abstract]
In the literature, LAeq and Lden are the dominating noise
exposure metrics for calculating the relationship between
noise and annoyance. However, alternative metrics such
as the number of events above a threshold (NAT), LAmax
or Emergence are rarely considered, although they may
have the potential to explain additional variance. This
seems necessary because the average relationship
between aircraft noise levels and annoyance raw scores is
.44 (Guski et al., 2017) which represents an explained
variance of only 19.36% (R2 = .1936).
Here, we present a post hoc analysis of the NORAH data
set (Klatte et al., 2016) and aim to evaluate the adequacy
of alternative noise metrics. We calculated multilevel
analyses with robust estimation algorithms (MLR) and using
different noise metrics (LAeq, LAmax, Emergence, NAT).
The analyses were carried out for different groups of
people (children, parents, teachers) and different
contexts (school: workplace or learning environment,
residential environment).
The calculations with the NAT60 criterion showed an
increase of explained variance (R2) in peopleës
annoyance. The effect was robust for different contexts
and groups of people.
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Organization:
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S. Yokoshima, Kanagawa Environmental Research Center |
O. Breugelmans, Netherlands National Institute for Public Health and the Environment |
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Combined effect of vibrations on railway noise annoyance
Shigenori Yokoshima, Takashi Morihara, Yasunao Matsumoto
[Abstract]
In Japan, noise annoyance associated with the Shinkansen
railway is more serious than that associated with conventional
railway, as Yokoshima at al. reported in the previous study. The
reasons for the difference have been explored in psychological
and physical aspects. Tamura pointed out in a comparison
survey along the Shinkansen and conventional railways that
people living along the Shinkansen railways had negative
attitudes to noise source. Therefore, no recognition of necessity
of the Shinkansen railway noise brought about higher
annoyance than that of conventional railway noise. Following
this study, Yokoshima and Tamura clarified the combined effect
of vibrations on noise annoyance associated with the
Shinkansen railway trains. However, the combined effect of
vibrations on noise annoyance from conventional railway trains,
including the comparison of the Shinkansen railway trains, has
not quantified. We performed a secondary analysis using
individual datasets equipped with noise and vibration exposure
as well as noise annoyance associated with the railway trains.
Applying a multiple logistic regression analysis to the datasets,
we clarified individual combined effect of vibrations on noise
annoyance associated with the Shinkansen and conventional
railways and discussed the impact of combined effect on the
prevalence of noise annoyance.
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Experimental Investigation of Evaluation Method of Horizontal Vibration in Building Caused by External Vibration Sources
Kentaro Hayashi, Yasunao Matsumoto, Toyohiko Higashida
[Abstract]
The purpose of this study was to examine the evaluation
method based on subjective responses to vibration caused
by
external vibration source including road traffic and railroads
for
building habitability.
The target buildings were small-scale buildings, such as
detached houses in Japan whose main structure was
wooden
or steel frame. The first natural frequency of vibration mode of
those building which has dominant horizontal vibration has
been
known to be in the range from 4Hz to 8Hz. Horizontal
vibrations
at the ground in that frequency range can be amplified due
to
the dynamic characteristics of buildings.
An experiment involving human subjects were conducted to
investigate relationship between subjective responses and
evaluation value computed from vibration acceleration time
histories. The input stimuli consisted of sinusoidal vibrations at
4Hz
and 8Hz for several different durations and vibrations based
on
building vibrations measured in actual residential buildings. 7-
step evaluation scales composed of expressions to provide
equal psychological intensity intervals about ”Magnitude”,
”Concern” and ”Discomfort” were used to measure subjective
responses in the experiment. It was found that Vibration Dose
Value(VDV), which evaluated the effect of vibration duration
by
the fourth power, was most appropriate among the
evaluation
methods used in this study.
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Case examples of the measurement of vibration and annoyance response in residential buildings beside roads and railways
Yasunao Matsumoto, Shigenori Yokoshima, Kentaro Hayashi
[Abstract]
Vibrations induced by road traffic and trains often degrade the
habitability of surrounding buildings. The Vibration Regulation
Law was enforced in 1976 in Japan. Road traffic vibration is a
target of regulation, although there have been many cases in
which the assessment based on the law is inconsistent with
adverse comments on vibration raised from building occupant.
An investigation was conducted to understand the relationship
between vibrations inside and outside of building and
annoyance response of building residents. Vibration
measurements were made at several residential buildings
located beside roads and railways. Those buildings were single-
family houses, which were two- or three-story wooden building.
Acceleration time histories were recorded at three locations
inside and outside of buildings to understand the dynamic
characteristics of those buildings. The buildings for measurement
were selected from houses of respondents in a separate
questionnaire survey on vibration annoyance. Annoyance
responses estimated from measured vibration records with the
application of different metrics available in current standards
and guidelines were compared with the annoyance responses
determined in the questionnaire survey. Subjective responses
that were estimated based on the results of separate laboratory
experimental study were also compared with the annoyance
responses in the questionnaire survey.
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Annoyance reactions due to noise and vibrations caused by different train types in Sweden: results from the EpiVib study
Elise Van Kempen, Mikael Ögren, Laura Maclachlan, Laith Hussain-Alkatheeb, Kerstin Persson Waye
[Abstract]
The few existing exposure-response relationships describing the association between exposure to vibrations from rail traffic and annoyance were not always able to distinguish between the different types of trains. However, indications were found that exposure to vibrations caused by freight trains are stronger related to annoyance than exposure to vibrations caused by passenger trains. Although the results of the studies looking into the difference in exposure-response functions describing the association between exposure to noise from trains and annoyance between the different train types are not consistent, there are also concerns that noise from freight trains may be more annoying than noise from passenger trains.
This paper reports on a study investigating annoyance reactions due to vibrations and noise from different types of trains. To this end, we used data from the Swedish EpiVib project, collected by means of a survey carried out in 2017 among more than 7,200 people aged 18-80 years living in the Västa Götaland, Värmland and Örebro regions in Sweden close to the railway track.
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Effects of changes in operational and residential factors on public health and reactions at the vicinity of Noi Bai International Airport
Thulan Nguyen, Bach Lien Trieu, Takashi Yano, Takashi Morihara, Yasuhiro Hiraguri, Makoto Morinaga
[Abstract]
The rapid development of air transport, together with the urbanization of the area around the airports, resulted in increased levels and wider influential scale of noise due to the airport operation. The number of flights operated and the population density at residential areas around Noi Bai International Airport has increased significantly in the past nine years. The exposure noise levels (Lden) at surveyed sites ranged between 48 and 61 dB in 2009, but increased to range between 54 and 76 dB in 2017. General health indicators such as Body Mass Index (BMI), blood pressure, etc. were investigated in two surveys conducted in November 2017 and August 2018. Exposure-response relationships established in the five surveys conducted from 2014 to 2018 were found to be higher than that established in 2009 survey. Comparisons of respondents with high blood pressure and insomnia ratios at different noise exposure level ranges showed that there is no significant association between ratios of hypertension and noise exposure levels (Lden) but a significant exposure-response relationship was found between night-time noise exposure levels and insomnia. This result suggested that a restriction on nighttime flight operation should be considered to protect health of resident living around airports in Vietnam.
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Organization:
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M. Basner, University of Pennsylvania |
M. Smith, University of Pennsylvania Perelman School of Medicine |
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Pilot Field Study on the Effects of Aircraft Noise on Sleep Around Atlanta International Airport
Mathias Basner, Michael Smith, Sarah Rocha, Maryam Witte
[Abstract]
Aircraft noise can disrupt sleep and impair recuperation. The
most recent US investigation on the effects of aircraft noise on
sleep was conducted more than 20 years ago. Since then,
traffic patterns and noise levels produced by single aircraft
have changed substantially. It is therefore important that field
studies be conducted in the US. to acquire current data on
sleep disturbance relative to varying degrees of noise exposure.
To inform the design of a larger national study, a pilot field study
was conducted around Atlanta Airport. The
primary goal was to assess feasibility of acquiring acoustical and
physiological data with no investigator on site and equipment
mailed to participants who then applied electrodes and started
and stopped measurements each night on their own. We
mailed 4080 recruitment questionnaires to randomly selected
households exposed to aircraft noise (>35 dB LAEq,23-
07,outdoor). Among the 407 respondents, 34 participated in the
unattended five night in-home study. Indoor sound levels were
determined via measurement. Arousal from sleep was recorded
via a device measuring body movement and
electrophysiological heart rate. Self-reported sleep was assessed
with morning questionnaires. Several ways to improve data
quantity and quality were identified. Overall, the approach was
found to be feasible.
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Effects of nocturnal aircraft noise on objective and subjective sleep quality in primary school children
Susanne Bartels, Julia Quehl, Daniel Aeschbach
[Abstract]
Little is known about childhood sleep exposed to transportation noise. We investigated the effect of aircraft noise on sleep in a field study in 51 children aged 8 to 10 years who resided near a 24-h operating airport. Sleep was measured polysomnographically during four nights. Besides, subjects rated their sleep quality and current fatigue each morning. Aircraft noise exposure was operationalized by the number of aircraft noise events above 30 dB(A) per night derived from continuous acoustic recordings at the children’s ear. For the prediction of objective and subjective sleep quality, we applied mixed models with random intercept and the number of noise events as dichotomous factor (median split at 37 events). Mixed models were also used to analyze the relationship between objective and subjective measurements. Noise affected the macrostructure of sleep. A higher number of noise events was associated with a reduction of slow wave sleep and an increase of wake time after sleep onset. In contrast, aircraft noise exposure did not affect subjective sleep quality and fatigue. Except for wake time, objective sleep measures were not related to subjective ratings. The findings underline the importance of objective sleep measurements for the examination of transportation noise effects in children.
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The effect of road traffic noise spectrum on sleep
Valtteri Hongisto, Saana Myllyntausta
[Abstract]
Two facades having nominally equal sound insulation declaration may have very different frequency behaviors. Our purpose was to determine, how the spectrum of RTN affects sleep when the A-weighted level is constant. Twenty one participants slept three nights in a sleep laboratory in three different sound conditions: L (high-frequency prominence, 37 dB), H (low-frequency prominence, 37 dB), and Q (control, quiet, 19 dB). Conditions L and H were created by modifying a 8-hour recording of RTN using two different frequency filters corresponding to the sound insulation spectra of two constructions. Sleep quality was measured both objectively (polysomnography) and subjectively (morning and evening questionnaires). According to the subjective measures, sleep quality was worse in conditions L and H than in Q but L and H did not differ from each other. SWS duration was longer in conditions L and H than in Q. Polysomnography revealed no differences between the conditions regarding sleep duration, sleep latency, WASO, or REM, N1, or N2 durations. However, retrospective questionnaire revealed that 81% of participants assessed condition H to be the most disturbing for sleep. High-frequency RTN may be more adverse than low-frequency RTN. More research is needed on the effects of spectrum on sleep.
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Traffic Noise and its Impact on Sleep Depth Measured by the Odds Ratio Product
Michael Smith, Magdy Younes, Daniel Aeschbach, Uwe Müller, Mathias Basner
[Abstract]
Traffic noise can lead to cortical and autonomic activation,
disrupt sleep and impair physical and mental restoration. We
used the odds ratio product (ORP), a validated continuous
measure of sleep depth and quality based on automatic
analysis of physiologic sleep data in 3s epochs, to investigate
temporal changes of sleep in response to nocturnal noise
events. Seventy two healthy participants slept for 11 nights in a
laboratory, during which we measured sleep with
polysomnography. In 8 nights, participants were exposed to 40,
80 or 120 road, rail and/or aircraft noise events at 45-65 dB
LAS,max. Event-related maximum change of ORP in a 90s
window relative to pre-event baseline was analysed with linear
mixed models. ORP increased monotonically with LAS,max for all
traffic modes, reflecting reduced sleep depth and quality, and
there was a greater response to road and rail noise than to
aircraft noise. There were no statistically significant effects of
noise duration, sex or sleep spindle rate of occurrence on event-
related ORP change. Average whole-night ORP increased with
age, but was not affected by nighttime noise level (LAEq) or
number of noise events. The clinical relevance of event-related
elevations of ORP is currently unknown, and warrants further
investigation.
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Do characteristics of short term transportation noise exposure fluctuation better predict self-reported sleep disturbances than Leq-based average noise metrics?
Mark Brink
[Abstract]
We report on a study that aimed at establishing exposure-response relationships reflecting the per-centage highly sleep disturbed (%HSD) as functions of road traffic, railway, and aircraft noise at night time, measured by various night time noise exposure indicators at both the loudest and faintest fa-cade point of dwellings. We therefore conducted a mixed-mode representative population survey in a stratified random sample of 5592 residents exposed to transportation noise all over Switzerland, to-gether with high temporal resolution noise exposure calculations. We hypothesized that in addition to average exposure metrics (e.g. Lnight), the effects of noise on self-reported sleep disturbance can be better explained when also considering the intensity of short-term variations of noise level over time, parametrized in the Intermittency Ratio (IR) metric, which expresses the ”eventfulness” of noise expo-sure situations as the energetic contribution of individual noise events relative to the total sound en-ergy in a given measurement period. The study also explored the role of a range of coping strategies taken against noise exposure and the potential benefit of the availability of a quiet side of the dwell-ing as well as the effect of different sleep-timing patterns on exposure-response relationships.
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Organization:
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M. Klatte, Technische Universität Kaiserslautern |
C. Clark, Queen Mary University of London |
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Comparing intentional switching of auditory selective attention in children and adults in an experiment suited for children
Karin Loh, Edina Fintor, Sophie Nolden, Janina Fels
[Abstract]
A remarkable part of children’s development and educations
happens in educational institutions. Acoustic environments in
these institutions are usually highly complex and noisy, hence it
is
demanding to identify relevant target speakers and to ignore
irrelevant sounds. Previous research has thoroughly analyzed
auditory selective attention in adults, both in a dichotic and a
binaural listening environment. Until now, there is little
knowledge of auditory selective attention in children. In the
present work, the original paradigm was adapted for children
by using a task suited for children which included child-oriented
elements. Further, the subject’s anthropometric sizes were
considered for an aurally-accurate reproduction of the acoustic
scene. An experiment with 24 adults and 24 children was
conducted analyzing noise and noise-free conditions and
various target-distractor distributions in the room among others.
The result of this experiment revealed significant differences
between adults and children, especially in the way auditory
attention was influenced by noise.
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Effects of irrelevant background speech on verbal working memory tasks
Maria Klatte, Kirstin Bergström, Thomas Lachmann
[Abstract]
Short-term memory for visually presented verbal items is impaired by task-irrelevant background speech that the participants are instructed to ignore. This so-called ”irrelevant speech effect” (ISE) has been attributed to interference with phonological representations, interference with representations of serial order, or attentional capture.
In order to learn more about the task characteristics necessary for ISE evocation, we designed 3 verbal working memory tasks that were comparable with respect to task structure, but differed in the cognitive processes involved. In each task, sequences of 7 German nouns were presented pictorially. In Exp. 1, participants had to reconstruct the serial order of the items. In Exp. 2, participants had to decide which of the 7 words represented by the pictures had no ”partner” with the same initial phoneme. In Exp. 3, participants had to decide which of the 7 items had no ”partner” with respect to semantic category. Serial order reconstruction and phonological categorization were significantly impaired by irrelevant speech, whereas semantic categorization was unaffected. The results indicate that serial order retention is not a necessary precondition for ISE evocation. Implications for theoretical accounts of the ISE and practical implications are discussed.
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Remembering landmarks in a virtual maze: Does the disturbance impact of background speech depend on the spatial information inherent in the speech signal?
Sabine Schlittmeier, Edina Fintor
[Abstract]
Whether a particular background sound impairs cognitive task performance depends on both sound and task characteristics. Although this so-called interference-by-process principle (e.g. Marsh et al., 2009) is not questioned in principle, it has not yet been fully empirically tested. In recent years, research on cognitive noise effects has focused on the detrimental impact of background speech’s intelligibility, semantics and temporal-spectral variability on performance in verbal tasks (e.g. verbal short-term memory, reading, writing). For visual-spatial tasks, however, the spatial characteristics of background noise should be decisive for inducing a performance decrement. We present a series of experiments in which short-term memory for visual-spatial information was explored during background speech conditions in which talkersë locations either varied or not. Considering wayfinding and orientation as everyday visual-spatial tasks, we set up virtual mazes including landmarks (e.g. a bakery, a park). Participants had to find their way through these virtual mazes and recall afterwards the spatial position of the landmarks. The effects of the different background speech conditions on visual-spatial memory will be discussed in terms of cognitive psychological implications as well as applied contexts.
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Organization:
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E. Van Kempen, Netherlands National Institute for Public Health and the Environment |
N. Riedel, University of Bremen |
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Aircraft noise exposure and saliva cortisol in a pooled-analysis from seven European countries
Clémence Baudin, Marie Lefevre, Jenny Selander, Wolfgang Babisch, Ennio Cadum, Marie-Christine Carlier, Patricia Champelovier, Konstantina Dimakopoulou, Danny Huithuijs, Jacques Lambert, Bernard Laumon, Goran Pershagen, Töres Theorell, Venetia Velonaki, Anna Hansell, Anne-Sophie Evrard
[Abstract]
Introduction
HYENA (HYpertension and Exposure to Noise near Airports), a well-known study on the health effects of aircraft noise, involved 4,861 participants from six European countries (except France). DEBATS (Discussion on the health effects of aircraft noise), a study with a similar protocol, included 1,244 participants from France.
Saliva samples were obtained for 439 and 954 participants in HYENA and DEBATS respectively, to determine cortisol concentration as a possible marker of noise-induced stress. Associations were found between aircraft noise and cortisol levels in each study separately, but they were not consistent between the two studies. The objective of the present study was to combine datasets to improve statistical power. Methods
We investigated the associations between aircraft noise levels and, the log-cortisol levels in the morning and in the evening, and the log-relative variation per hour, adjusted for relevant confounders. Results
We found evidence in women for an increase in the log-evening cortisol level β=0.09 (95%CI 0.01;0.17), and for flattening of the log-relative variation per hour β=-0.09 (95%CI -0.16;-0.02), with a LDEN 10-dB(A) increase, but no significant effects in men. Conclusions
These findings provide some support for psychological stress induced by aircraft noise exposure, resulting in disruption of hormonal rhythms.
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Association Between Transportation Noise and Cardio-metabolic Diseases: an Update of the WHO Meta-analysis
Danielle Vienneau, Ikenna Eze, Nicole Probst-Hensch, Martin Röösli
[Abstract]
For the development of the WHO Noise Guidelines for the European Region, meta-analyses for various cardiometabolic outcomes were conducted to derive exposure-response associations for aircraft, railway and road traffic noise. Papers published until August 2015 were considered. Since then, several new studies have been published. In the framework of revision of regulatory noise limits in Switzerland, up to date evidence for the health effects of noise is needed. Thus, the aim is to update the WHO meta-analysis for ischemic heart disease and diabetes with studies published until February 2019. The same protocol is applied as done by the WHO Environmental Noise Guideline group. Systematic literature search follows the search strategy of the WHO. Risk of bias for each study is evaluated for various design aspects. The quality of evidence is assessed using GRADE. Pooled exposure-response associations are calculated for road, railway and aircraft noise based on fixed or random-effects model depending on the outcome of the Cochran’s Q-test and the I2-statistic, which reflects the percentage of between-study heterogeneity. We will present the updated evidence and exposure-response analysis for Lden and discuss whether and how evidence for ischemic heart disease and diabetes has changed in the last few years.
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Organization:
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I. Van Kamp, Netherlands National Institute for Public Health and the Environment |
K. Kawai, Kumamoto University |
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Noise and health in children with autism spectrum disorder
Hidetoshi Takahashi
[Abstract]
Autism Spectrum Disorder (ASD) is associated with persistent impairments in adaptive functioning across multiple domains including social, communicative, occupational, and other important areas of daily life. Sensory abnormalities have often been reported as symptoms of ASD, and auditory over-responsiveness (AOR) is the most common sensory-perceptual abnormality in individuals with ASD.
Recently, we reported that a greater acoustic startle reflex magnitude in response to weak stimuli of 65-85 dB was found in children with ASD compared to those with typical development, and that this index was related to various clinical features, including sensory problems, autistic traits, emotional/behavioral difficulties, and adaptive/maladaptive behaviors in ASD children. This intensity level of acoustic stimuli is frequently experienced in everyday situations including the classroom environment. Noise affects several aspects of both physical and mental health and those with AOR might be affected more by their daily environment. This would apply not only to children but also to adults working in schools, who suffer from AOR. Thus, for the healthy development of ASD children with AOR it is important to create an auditory-friendly environment including in the classroom.
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Effects of aircraft noise and living environment on children´s well-being and health
Jan Spilski, Martin Rumberg, Martin Berchtold, Kirstin Bergström, Ulrich Möhler, Thomas Lachmann, Maria Klatte
[Abstract]
There are few and inconsistent results suggesting that
chronic exposure to aircraft noise is associated with poorer
quality of life or health in children. In their WHO review,
Clark & Paunovic (2018) pointed out, that the current
evidence is insufficient. Previous publications mainly
analyzed the isolated relationships between single stressors
(e. g. aircraft noise, pollution) and health in children. A
combined consideration of environmental stressors and
residential environment factors, such as household
crowding, access to green/open space urbanity, and
other urban planning data did not occur.
In the NORAH study (Klatte et al., 2016), health, noise
exposure, and a variety of potential moderating variables
(residential environments factors) were assessed from
second-graders who live in the vicinity of the
Frankfurt/Main airport, Germany. We present a
theoretically motivated secondary analysis of the NORAH
data set. Post hoc, we linked this data with spatial and
urban planning data, to model the impact of aircraft
noise and children’s real life circumstances. We
calculated moderated and mediated regression models
and found empirical evidence for moderation effects in
the relationship between noise, residential environment
and well-being or health.
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Restorative soundscapes and stress recovery in Children
Shan Shu, Hui Ma
[Abstract]
Emerging studies have shown evidence-based effects of
restorative soundscapes on adults’ health. Little is known,
though, about children’s physiological and emotional responses
to restorative soundscapes. Therefore, the present study
examined the restorative effects of soundscapes on children’s
health in a simulated urban park setting. In the experiment, 53
children aged 8-12 were exposed to different soundscapes after
a stressful arithmetic task. Physiological responses and emotional
evaluations were measured respectively during the soundscape
exposure. The results showed that all the soundscapes could
decrease children’s skin conductance level (SCL) at the
beginning of the recovery period, but showed significant
divergent influence afterwards. In addition, different
soundscapes showed marginally different effects on children’s
heart rate variability (HRV) which is indicative of
parasympathetic activation. Moreover, children gave
soundscapes significantly different emotional ratings on
pleasure and arousal, but not on dominance. Those results
suggest that soundscapes such as music and stream sounds do
facilitate physiological and emotional recovery on children to
some extent after a psychological stressor. However, more
evidence was needed to be proven in future studies.
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Associations of traffic noise and air pollution with birth outcomes in Alpine areas: Results from the UIT and BBT surveys
Angel Mario Dzhambov, Iana Markevych, Peter Lercher
[Abstract]
The present study investigated associations between traffic noise and air pollution and birth outcomes in several Alpine areas. We used data from two representative cross-sectional studies (UIT, n = 573 and BBT, n = 518) in the Tyrol Region (Austria and Italy). Only mothers who had lived in their current residence during the whole pregnancy were included. They completed a questionnaire, and medical records were used to draw data on birth weight, low birth weight (LBW), preterm birth, and small for gestational age (SGA). Road/railway traffic noise (Ldn, Lden) and air pollution (NO2) were determined using a combination of sophisticated modelling and field measurements. In the UIT survey, higher Ldn was associated with higher odds of LBW. No associations were observed for NO2. In the BBT survey, results were not straightforward. Lden was associated with lower birth weight when adjusted for NO2 and among preterm babies, while its association with LBW was inconsistent. NO2 was associated with lower birth weight and higher odds of LBW in preterm babies. Our preliminary findings suggest that traffic emissions might increase risk of poor birth outcomes in the Alpine region.
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Effect of Sound Absorption on Children’s Behavior in Daycare rooms: a Field Experiment with Installation of Temporary Sound Absorption.
Keiji Kawai, Shota Masumoto, Ryuichiro Yamane
[Abstract]
We have been investigating the effect of sound absorption for children in child daycare rooms and the results demonstrated a mitigating effect of bustling atmosphere in the rooms and an improvement of word intelligibility of children. In this study, following these previous findings, we attempt to find behavioral change of children in acoustically absorbed daycare rooms by a field experiment in two daycare centers with installation of temporary sound absorption. The experiment consisted of 1) teachers’ observation, where they daily evaluated children’s calmness or mood, 2) interview to teachers, and 3) behavioral observation for the concentration of children by gaze or head direction through video-analysis. As a result, while there were little difference in the teachers’ evaluation with/without absorption, the teachers, in the interview, preferred the absorbed conditions mainly because of the calmness in the room. The recorded video was divided into 1 s segments and the ratio of the segments in which the eyes or head of a child was directed at the teacher was defined as concentration ratio, and the ratio was obviously increased with sound absorption.
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Organization:
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S. Stansfeld, Wolfson Institute of Preventive Medicine, Queen Mary University of London |
I. Van Kamp, Netherlands National Institute for Public Health and the Environment |
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Study on methodology to perform an environmental noise and health assessment – a guidance document for local authorities in Europe
Irene Van Kamp, Dirk Schreckenberg, Elise Van Kempen, Mathias Basner, Alan Lex Brown, Charlotte Clark, Danny Houthuijs, Oscar Breugelmans, Annemarie Van Beek, Brigit Janssen-Stelder
[Abstract]
The Environmental Noise Directive (END) is for local authorities the most important instrument to determine the levels of noise pollution. The Directive is aimed at the evaluation of effect of measures and can make the effects of alternative measures visible. The Directive stimulates the Member States to take action to reduce the aversive effects of environmental noise. In preparation of the update of Annex III of the Directive RIVM in collaboration with international partners has prepared a guidance document taking the new WHO guidelines as a point of departure. Not only annoyance and sleep disturbance are addressed as health effects, but also cardiovascular effects and cognitive impact in children (comprehensive reading impairment). The guidance document was prepared on request of the EU commission. This paper will describe the steps of a health impact assessment one by one and explain the accompanying decisions and conditions. Next, the actual calculation methods are further explained for two indicators: the number of healthy life years adjusted for disease, disability and death (DALY) and the number of people that experiences adverse effects of noise ((NafP). Finally, as an example, the health impact of noise in Düsseldorf is presented.
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Case study of an environmental noise and health assessment in the City of Düsseldorf, Germany
Dirk Schreckenberg, Irene Van Kamp, Sarah Leona Benz, Stephan Grossarth, Elise Van Kempen, Mathias Basner, Alan Lex Brown, Charlotte Clark, Danny Houthuijs, Oscar Breugelmans, Annemarie Van Beek, Brigit Janssen-Stelder
[Abstract]
In their recent Environmental Noise Guidelines (ENG) the World Health Organization (WHO) identified several critical health outcomes of environmental noise, among them cardiovascular diseases, annoyance, effects on sleep, cognitive impairment, and hearing impairment/ tinnitus. Aiming at protecting the population from adverse noise effects, the Environmental Noise Directive 2002/49/EC (END) is the main instrument guiding the noise mapping and the development of noise action plans in EU Member States. END Annex III describes methods of the assessment of health impacts (HIA) of noise in terms of the number of highly annoyed and sleep disturbed persons in a given area. A new study commissioned by the European Commission describes a HIA methodology for environmental noise which considers the critical health outcomes as identified in the WHO ENG. In summary, two methods, the assessment of number of people affected by different health outcomes and the environmental burden due to environmental noise in terms of the calculation of disability-adjusted live years (DALY) are described. We present a case study where both HIA methods were applied to the City of Duesseldorf, Germany, in order to describe the status quo of the health impact as well as the health impact of simulated transportation noise interventions.
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Environmental Noise Guidelines Implementation in Poland
Anna Preis
[Abstract]
In the European Noise Directive 2002/49/EC (END) in the article 6(3) there is a statement that ”harmful effects may be assessed by means of dose-effect relations referred to in ANNEX III”. The dose-effect relations could be used to assess the effect of noise on populations. In particular, it is possible for a given value of noise index, e.g. Lden of a given noise source, to predict the percentage of people who will be annoyed or highly annoyed by this noise. The WHO environmental noise guidelines document presents the recommended Lden values for five different noise sources, which when applied could prevent people from an adverse health effects. The recommendations presented in the WHO guidelines are based on the dose-effect relations for annoyance and sleep disturbance. It seems natural that these WHO proposals should be implemented in the revised version of ANNNEX III of the END. Right now the Noise Committee is working on the new version of ANNEX III and to my knowledge representatives of EU countries vary in their views. In my talk I will present how WHO guidelines are received in Poland.
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A critical review of the basis for WHO's new recommendations for limiting annoyance due to environmental noise
Truls Gjestland
[Abstract]
The new WHO Environmental Noise Guidelines for the European Region include recommendations for limiting noise exposure to levels without adverse health effects. This paper reviews WHO’s evidence with respect to aircraft and road traffic noise annoyance. It demonstrates that WHO’s new guidelines are based on a non-representative selection of existing studies that comprise data that cannot be generalized to the residential population at large. The new WHO recommendations are unwarranted and unsupported by the reported evidence.
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Valuing impacts of noise on health - exposure response relationships in current UK guidance and the WHO Environmental Noise Guidelines 2018
Benjamin Fenech, Georgia Rodgers
[Abstract]
Following the publication of the Environmental Noise
Guidelines for the European Region (ENG), a lot of debate
has focused on the Guideline Recommendations, in
particular the specific guideline levels and the strength of the
recommendations. Nevertheless it is widely acknowledged
that a sizeable proportion of the population is currently
exposed to noise levels above the WHO recommendations,
and it is important for policy and decision makers to be able
to quantify the associated burden of disease as accurately
as possible. From this perspective, the systematic reviews of
the scientific evidence commissioned by the WHO are an
equally important aspect of the ENG. In the UK the current recommended methodology for valuing
health impacts associated with environmental noise is
outlined in two documents published by the Department for
Environment Food and Rural Affairs (Defra) and the
Interdepartmental Group on Costs and Benefits Noise Subject
Group (IGCBN). [1,2] This paper will look into the studies that
informed the exposure response relationships recommended
by these two documents. A comparison is then made with
the more recent studies that informed the ENG, followed by
a discussion of which aspects of a study would make it more
applicable to a local population.
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Reviewing the Current Guidance in England for the Valuation of Noise Impacts in light of the Publication of the WHO Environmental Noise Guidelines 2018 and other Relevant Evidence
Hilary Notley, Alisha Iyer, Emma Powell
[Abstract]
In England, the current guidance for valuing impacts associated with environmental noise is published by the Department for Environment Food and Rural Affairs (Defra) and the Interdepartmental Group on Costs and Benefits Noise Subject Group, (IGCB(N)) [1]. The development of the World Health Organisation (WHO) Environmental Noise Guidelines [2] followed a rigorous methodology and their recommendations are based on systematic reviews published separately [3]. The systematic reviews considered evidence published between 2000 and 2014 - 2016 (health outcome dependent). There have been several high quality studies published since those considered by the WHO that may further develop the understanding of the link between noise exposure and health. This paper describes the steps that the IGCB(N) are taking to determine whether they should update their current recommendations. The paper describes the commissioned reviews of evidence relating to health effects and efforts to assess a number of potential effects that are outside of the scope of the WHO, such as the benefits of quiet areas/tranquillity and the impact on productivity. The paper further describes how the valuation of any effects will be considered. [1] https://www.gov.uk/guidance/noise-pollution-economic-analysis [2] WHO Regional Office for Europe Environmental Noise Guidelines for the European Region http://www.euro.who.int/__data/assets/pdf_file/0008/383921/noise-guidelines-eng.pdf?ua=1 [3] https://www.mdpi.com/journal/ijerph/special_issues/WHO_reviews
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Organization:
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T. Masayuki, Kyushu University |
M.E. Altinsoy, Technische Universität Dresden, Lehrstuhl für Akustik und Haptik |
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Analysis of vehicle horn use and factors at intersections in an urban area of Taiwan
Takada Masayuki, Shoki Tsunekawa, Kazuma Hashimoto, Tamaki Inada, Yoshinao Oeda, Katsuya Yamauchi, Ki-Hong Kim, Shin-Ichiro Iwamiya
[Abstract]
In the urban areas of many countries, noise pollution is generated by frequent use of car and motorbike horns. The final goal of the present study is to clarify the causes of horn use and to propose a transportation system to suppress horn use. To this end, we have been investigating relationships between horn use and factors such as driver awareness and behavior and vehicular environment. Our previous study investigated causal relationships between horn use and the vehicular environment in Taiwan and found that horn use was frequent when gridlock formed at intersections. The present study analyzed details of gridlock situations during which vehicle horns were used at three intersections. We found many cases where honking began after vehicles entered the intersection to turn left during the green-light phase. In particular, drivers often used their horns when they had to wait for more than 4 s for the vehicles in front of them to start moving at the beginning of the green-light phase for turning left. Logistic regression analysis revealed that the duration between the signal turning green and the vehicle in front of them starting to move significantly affected the probability that a driver would use a vehicle horn.
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Electric and Autonomous Vehicle: from Sound Quality to Innovative Sound Design
Nicolas Misdariis, Andrea Cera, William Rodriguez
[Abstract]
This study follows long-term lines of research and realizations
done within the frame of electric vehicle sound design. It
currently deals with the most up-to-date issue addressing the
near future mobility: Autonomous Driving (AD). It is implemented
in a 2-year collaboration with a french car manufacturer that
deployed a high-tech demo car where the sonic dimension
was explored in a spatial, and a multisensory, approach.
Specific hardware devices (multi-channel diffusion system) and
software protocols (spatialization and vehicle data transmission)
made it possible to study and realize numerous and accurate
internal / external warning sounds and auditory Human-
Machine Interfaces - and finally to contribute to a kind of
”passive branding” of the artifact. The Autonomous Driving
condition was especially considered as it corresponds to an
unprecedented user experience in terms of human-centered
process, where sounds could particularly play a role for
informing and, if necessary, reassuring the driver and
passengers. The paper presents a thorough analysis of the
project: initial requirements and data, multi-disciplinary
framework, technical details, sound concepts and realization. It
also opens onto reflexive and prospective considerations about
the global notion of sound quality that especially look at
possible forms of ”low-intrusiveness” in sound design, and their
related properties.
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Design of Acoustic Vehicle Alerting System Sound Assuming Listening Situation of Pedestrians
Hiroshi Matsuda, Masato Suzuki, Nobuo Machida
[Abstract]
The role of acoustic vehicle alerting system (AVAS) sound is to tell pedestrians the approaching of quiet vehicles such as electric vehicles or hybrid-electric vehicles (HEV). We propose the AVAS sound that satisfies both of comfortability and perception. The comfortability of AVAS sound in this study means pedestrians do not feel noisy. The perception of AVAS sound means pedestrians to notice of the approaching quiet vehicles at safely avoidable distance. The AVAS sound model produced in this study has two peak frequencies. These models are two kinds of the steady sound with constant sound level and the fluctuating sound formed by modulating amplitude of steady sound. We measured impression of AVAS sound and perceptual distance. The impression of AVAS sound was measured by the psychological questionnaire using SD method. The perceptual distance: the distance between subject and HEV when the subject notices the approaching of HEV emitting the AVAS sound was measured in a test road. We also measured the perceptual distance of different listening situation such as standing or smartphone operation during walking. We show the sound characteristic to improve the comfortability and the perception of AVAS sound, and the effect of the listening situation on the perceptual distance.
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Relationship between acoustic characteristics and impression for warning sounds on electric vehicles
Nozomiko Yasui
[Abstract]
The motor sound on electric vehicle is quiet at low speeds. Thus, pedestrians have difficulty detecting those vehicles approaching them. Although those vehicles were designed to play an alert sound to solve this problem, it has not been solved yet. When the sound is designed, it should be concerned not only detectability of approaching quiet vehicles but also impression of the sound. For pedestrians, it’s important to make it easier to recognize quiet vehicle. Also, warning sounds shouldn’t contribute to traffic noise annoyance. Our previous studies found that acoustic characteristics of amplitude fluctuation are effective to make them detect approaching vehicles. Here, this study evaluates impressions of those fluctuated sound, investigates relationship between the impression and those characteristics. The impressions of synthesized warning sounds that have periodic and non-periodic amplitude fluctuations were measured by semantic differential method. The obtained data were analyzed by factor analysis. The results revealed that characteristic of amplitude fluctuation influences the factors of ”familiarity”.
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Pedestrian awareness of the approach of quiet vehicles: Effect of approach informing sound and designing awareness
Takahiro Kosuge, Tomoya Tsujinaga, Toki Kobayashi, Mariko Tsuruta-Hamamura, Hiroshi Hasegawa
[Abstract]
Hybrid and electric vehicles are quieter than combustion-
engine vehicles. To prevent accidents resulting from this
quietness, an acoustic vehicle alerting system generates
an approach informing sound (AIS). However, the
recognition rate of the AIS is not high (∼40%) and it might
be difficult for pedestrians and others to notice the
approach of quiet vehicles via hearing alone. To improve
AIS recognition, we designed an AIS based on
onomatopoeic sounds. A subjective evaluation
experiment using these sounds revealed that people
tended to evaluate noisy and rough sounds, such as those
similar to the sound of a car traveling or an engine, as a
good indication of a car traveling. Another experiment
investigated the sounds that pedestrians actually use to
recognize the approach of quiet vehicles. It was found
most participants recognized the approach of a
quiet vehicle not by the AIS but by the sounds emitted
from the vehicle, such as road and wind noise. This
tendency was also observed after participants correctly
recognized the AIS. However, some participants reported
that their awareness of AIS changed and they became
more aware of AIS in daily life after the experiment.
Designing awareness is effective and necessary for the
design of AIS.
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Effects of localization control of warning sound combined with visual information in vehicle cockpit
Kiichi Naka, Katsuya Yamauchi, Nobuaki Tanoue, Ayumu Kawata
[Abstract]
Recently, many kinds of information are provided to drivers
because of the improvement of ADAS (Advanced Driver
Assistance System). Amount of information in vehicle cockpit
seems to be increased due to the development of automobile
technology such as autonomous driving. It is very important to
consider the design of appropriate auditory information to
avoid overlooking visual information cues. In this paper, we
examined the effect of localization control of warning sounds to
determine the direction of visual information using a driving
simulator. The number of overlooked visual information cues and
subjective task load were significantly decreased by the
localization control in comparison with the no localization and
no sound conditions. A driver could recognize the direction of
visual information without removing their gaze from frontal view
so that a driver also could find the visual information rapidly and
accurately, and could watch the frontal view more long
duration. It led to steady driving performance such as steering
and speed control. In contrast, these effects were diminished if
the timbre of the signal was difficult to localize.
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Organization:
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T. Toi, Chuo University |
W.-H. Cho, KRISS |
S.-H. Shin, Kookmin university |
I. Jung, Hyundai Motor Company |
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Evaluation of Independent Sound Zones in A Car
Ji-Ho Chang, Wan-Ho Cho
[Abstract]
Recently sound field control techniques have been
applied to vehicles that generate different sound zones in
a space with loudspeaker arrays. A typical example in a
car is to provide a driver and a passenger with different
sound such as voice-guided navigation, music, or
telephone voice. Acoustic contrast has been widely used
as a performance index in sound field control that
effectively shows the difference between sound levels in
the acoustically bright zone and the dark zone. However,
these sound levels are usually measured with microphone
arrays that are placed in the sound zones without the
listeners, and averaged. These values can be different
from what the listeners would have at their ears. This study
compares the acoustic contrasts derived using
microphone arrays with those using dummy head
microphones at several positions and those using binaural
microphones mounted in several users’ ears. From this
comparison, this study attempts to propose a simple and
effective evaluation method of the independent sound
zones.
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Sounds for Enhancing Energy Efficient Driving: A Simulator Pre-Study
Arne Nykänen, Johan Fagerlönn, Stefan Lindberg, Roger Johnsson, Sathish Kumar Ramanathan
[Abstract]
Computer game technology was used for rapid prototyping of a sound based interface encouraging truck drivers to drive energy efficiently. The design process was inspired by user-centred agile methods. Interior sounds were made interactive by actively controlling them based on speed, engine speed, torque and acceleration. User feedback was collected at an early stage through contextual enquiry sessions during simulated driving. Based on the feedback the sounds were adjusted and effects on driver behaviour were measured in a simulator experiment. The effects on driver behaviour were small and not statistically significant (p>.05). However, assessments of preference and informativeness showed statistically significant differences between the design concepts (p<.05). The qualitative part of the study showed that the use of game technology for enabling assessment of interactive sounds in early design phases was useful and allowed for getting users into the loop early on. The framework was found to be suitable for designing interactive sounds, and the data collected provides insight into driver responses to using active noise control as a means for providing information to the driver.
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Investigation of Effect on the Acoustic Transfer Function in a Vehicle Cabin According to Change of Configuration
Wan-Ho Cho, Ji-Ho Chang
[Abstract]
Recently, the various audio information are presented in a
vehicle cabin not only for a driver but also other passengers and
various sound field control methods had been applied for
various purpose such as personal audio. Basically, the
measurement of acoustic transfer function in a vehicle cabin is
relatively easier than other living spaces because of its small
volume and movement of objects. However the effect of
configuration cannot be ignored due to the relatively large
volume change, especially related to the change of number of
passengers. Here, the effect on the acoustic transfer function in
a vehicle cabin according to change of configuration is
observed and method to estimating robust solution is
investigated. The solutions estimated by using the transfer matrix
measured by planar arrays, dummy heads, and arrays
surrounding head position with dummy heads are compared by
changing the number of passenger. To observe the robustness
of estimated solution, the solutions are applied to the different
conditions not used to estimate the solution and the
degradations are compared for each case.
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Factors Analysis of Gear Sound by Using Numerical Simulation
Chengfeng Pu, Takafumi Takeda, Toru Hama, Takeshi Toi
[Abstract]
As an approach to identification of a factor that contributes to
gear sound, In the quasi - static loading condition of low speed
operation, estimation of gear transmission error of spur gear was
investigated from combination of vibration measurement and
numerical simulation. Furthermore, build a numerical simulation
approach to analyse the prediction and contribution of
vibration and sound under high speed operation conditions.
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Design of Optimal Car Interior Sound Improving Driving Feeling
Su-Ho Cha, Sung-Hwan Shin, Jae-Ho Yang
[Abstract]
The virtual driving sound has been used as a tool improving the driving feeling of passenger car. The purpose of this study proposes is to propose a strategy generating virtual driving sound that is closely connected with the driving condition. In this end, previous study ascertained that dynamic feeling of car interior sound was related to the change of tone color during the acceleration. The tone color was affected by the strength of the firing order components. And then, it was analyzed that the relation between the dynamic feeling obtained from the subjective listening test and the acceleration condition of passenger car and as its result, it was identified when a virtual sound should be generated to improve the driving feeling such as the starting and stopping points of virtual sound and its reproduction duration. Next, optimal levels of the firing order components are found out by subjective listening tests. Lastly, the effect of the proposed strategy generating virtual driving sound is investigated. It is expected that the method can improve driving feeling related to not only dynamic but also pleasant perception.
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Organization:
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S. Sakamoto, Institute of Industrial Science, The University of Tokyo |
K. Yamauchi, Kyushu University, Faculty of Design |
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Road Traffic Noise Prediction Model “ASJ RTN-Model 2018” Proposed by The Acoustical Society of Japan – Part 1: Outline of the calculation model
Shinichi Sakamoto, Yasuaki Okada, Akinori Fukushima, Toshio Matsumoto, Terutoshi Tajika
[Abstract]
The Acoustical Society of Japan (ASJ) has published a
new version of road traffic noise prediction method ”ASJ
RTN-Model 2018” in this April. The technical committee on
road traffic noise in the Acoustical Society of Japan (ASJ)
had been working for structuring the road traffic noise
prediction models in conformity with the times for
decades, and ”ASJ RTN-Model 2018” is an up-grade
version of the previous model proposed in 2014. In
developing the new version, existing knowledge is widely
taken into account, in particular, specification of sound
power levels of vehicles and calculation methods for
sound propagation are improved in wide range. This new
model is introduced in five papers with the same main title
in this congress. In this paper, the framework of the model,
the calculation principle, the general procedure of
prediction calculation, the outline of the revision, are
described.
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Road Traffic Noise Prediction Model “ASJ RTN-Model 2018” Proposed by The Acoustical Society of Japan – Part 2: Calculation Model of Sound Emission of Road Vehicles
Yasuaki Okada, Akinori Fukushima, Katsuya Yamauchi, Shinichi Sakamoto
[Abstract]
The Acoustical Society of Japan (ASJ) has published a new revised version of road traffic noise prediction method ”ASJ RTN-Model 2018”, in which the calculation formula for the A-weighted sound power level of each type of road vehicle is specified. The sound power level is given simply as a function of the running speed in consideration of practicality and convenience. The Research Committee in ASJ has been accumulating new data about noise emission of vehicles running on the dense and porous asphalt pavements over the last decade. As a results, the sound power level of light vehicle (includes passenger cars, hybrid vehicles and small-sized vehicles) for the dense asphalt pavement was changed to low value. In addition, a improved calculation formula of sound emission on the drainage asphalt pavement were introduced. In this paper, the revised sound power level and spectrum models for road vehicles using the newly acquired data are described.
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Road Traffic Noise Prediction Model “ASJ RTN-Model 2018” Proposed by The Acoustical Society of Japan – Part 3: Calculation model of sound propagation
Akinori Fukushima, Shinichi Sakamoto, Yosuke Yasuda, Takatoshi Yokota
[Abstract]
As the third part of ”ASJ RTN-Model 2018”, calculation model of
sound propagation is presented. The model is an up-grade
version of the previous model that was proposed in 2014. This
model is basically developed as a practical calculation model
based on Geometrical Acoustics. The calculation equations in
this model are defined on the base of experimental or
numerical analysis data. The overall values of the A-weighted
sound pressure level propagated from a vehicle are directly
calculated by considering the frequency characteristics of
vehicle noise. This calculation model consists of shielding effects
by several type barriers, excess attenuation due to ground
surface, atmospheric absorption effect, sound reflection and
meteorological effects. Based on newly obtained knowledge,
several improvements were made on the calculation method to
develop the model. The procedures of application to roads with
special cases such as road tunnel, semi-underground road and
double deck viaduct road are also included.
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Road Traffic Noise Prediction Model “ASJ RTN-Model 2018” Proposed by The Acoustical Society of Japan – Part 4: Accuracy Verification of a Practical Method for Areas behind Buildings in Urban Districts
Ken Anai, Toshio Matsumoto, Takatoshi Yokota, Shinichi Sakamoto
[Abstract]
The purpose of this study is verification of ASJ RTN-Model for
areas behind buildings, which is named in this paper as a
practical method, by comparing predicted noise levels and
measured ones in urban districts. Two kinds of verification are
demonstrated in this paper. In the one kind of verification, max
noise levels in unit patterns by running an inspection car in
steady speed on a flat road or a cut road facing at detached
residential areas are focused on. In the other kind of verification,
equivalent noise levels along two city roads are predicted by
the practical method with road traffic census data. As a
consequence, differences between the predicted levels and
the measured ones are within plus and minus 3 dB in the max
noise levels case, and the differences are from plus and minus 3
dB to 5 dB in the equivalent noise levels case.
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Road Traffic Noise Prediction Model “ASJ RTN-Model 2018” Proposed by The Acoustical Society of Japan – Part 5: Study on Prediction Accuracy
Katsuya Yamauchi, Terutoshi Tajika, Akinori Fukushima, Ken Anai
[Abstract]
This paper introduces the study on prediction accuracy of the
ASJ RTN-Model 2018. The authors provide the report as a
part of the series of five papers which has same main title in
this congress, on behalf of the research committee on road
traffic noise in the Acoustical Society of Japan. It is very
important to examine the prediction accuracy of the new
model, because some assumptions are included in modeling
the source power level of each type of vehicles and in
simplifying the calculation of noise propagation. Some
examination of the correspondence between the predicted
values and measured traffic noise levels are introduced in
this paper. The accuracy of the method of prediction for noise
behind building complex are also examined. The measured
values were obtained recently in Japan on flat and strait
section of general roads and on expressways. This paper also
introduces the examination on assumed causes of errors that
are thought to be important in considering prediction
accuracy, such as the condition setting of hypothetical traffic
lanes, vehicle type classification, and some typical causes of
uncertainty on actual measurement.
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Experimental Modelling of Tyre/Road Noise from Road Texture Spectra on Rubberized Road Surfaces
Alessandro Del Pizzo, Gonzalo De Leon, Luca Teti, Francesco Bianco, Antonino Moro, Luca Fredianelli, Gaetano Licitra
[Abstract]
The present work studies the correlation between road
texture and tyre/road noise from an experimental point of
view, by analysing road texture and CPX measurements on
rubberized road surfaces. Tyre deformation has also been
addressed using the indenter method proposed by Goubert
and Sandberg. The correlation coefficient between texture
and noise bands was calculated using data deriving from
CPX measurements and enveloped texture spectra. Results
showed that tyre/road noise can be separated into low,
middle and high frequency components, which correlate
differently with road texture bands. This frequency separation
was used to develop two different linear models that relate
low and high frequency noise to road texture one-third
octave bands.
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An Australian case study on the estimation of heavy vehicle noise emission on grade
Jeffrey Peng, Daipei Liu, Jeffrey Parnell, Nicole Kessissoglou
[Abstract]
Heavy vehicles are considered the primary determinant of
night-time noise disturbance, particularly in rural areas or for
residents along freight routes. To capture the dynamic
influence of heavy vehicles associated with variation in
speed on noise emission, heavy vehicle kinematic variables
need to be incorporated within a road traffic noise emission
model. These kinematic variables in turn assist with accurate
estimation of engine noise and rolling noise. The American
FHWA TNM road traffic noise model is the only existing
prediction method that considers the driving speed profiles
of articulated trucks. However, it can only consider a fixed
set of speed profiles based on a single heavy vehicle power-
to-weight ratio. As such, the current model is limited and
does not accurately represent longer and heavier vehicle
combinations. In this work, the FHWA TNM method is
modified to include the kinematic equations for a heavy
vehicle on grade. The model is applied to heavy vehicles
commonly utilized in Australia. A case study based on a
principal freight route in New South Wales, Australia, is
presented to illustrate the predicted variation in engine
noise and rolling noise throughout the heavy vehicle’s
journey.
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Vehicular traffic noise in bus station users modeling and prediction, based on the analysis of direct and specular paths
Dayane Cristina Lima Estercio, Paulo Fernando Soares
[Abstract]
Bus stop for public transportation systems are structures where users
are exposed to intense noise levels and are poorly evaluated in
surveys. Therefore, this research aims to develop a mathematical
model of noise level prediction at bus stops, in order to contribute to
the characterization and evaluation of the influence of traffic noise
level in these buses. In order to achieve the purpose of the research,
a prediction model was developed based on the direct and specular
reflections. A standard bus stop model was adopted on a three-lane
road, simulating the users who are standing and sitting The sound
source was positioned on the axis of each strip, every five meters. In
total, 378 source positions were evaluated. The descriptors were the
sound pressure level (SPL) and the sound equivalent level (LAeq).
For the validation, the Anderson-Kurze model and the t-Student test
were applied. The results presented better correspondence between
the developed model and the reference model in the 1000Hz range
and in the 500Hz frequency there was a greater variance between
the applied models. As a preliminary study, the research presented
satisfactory results, which may contribute to future research.
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Organization:
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M. Wegerhoff, HEAD acoustics GmbH |
J.-G. Ih, Acoustics Laboratory, Dept. of Mechanical Engineering, KAIST |
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Transfer Path Analysis of Rumbling Noise in a Passenger Car Based on Measured In-Situ Blocked Force
Sang Kwon Lee, Taejin Shin, Yeonsoo Kim, Kanghyun An
[Abstract]
The control of rumbling noise is one of the major strategic targets of interior sound quality inside the cabin of a passenger car. To effectively control rumbling noise in a passenger car, the transfer path of the rumbling noise should be initially identified. It is known that the major source of this noise the combustion force of an engine. The combustion force excites the engine and induces vibrations of the powertrain. These vibrations are then transferred to the body of the vehicle via its structural transfer path. Moreover, the vibrations of the vehicle’s body emit internal vibra-acoustic noise. This noise is often referred to as the rumbling noise due to the structural borne path. If there are structural resonances among the structural paths such as the engine, transmission, mount bracket, suspension, and the vehicle’s body, the rumbling noise could be amplified. Recently, blocked force transfer path analysis (BF-TPA) was introduced and this approach does not require the decoupling of the substructures. In this study, we identify the structure-borne path of rumbling sound based on blocked force transfer path analysis (BF-TPA) in a passenger car. In addition to identification, the passive control method for rumbling sound is presented.
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Combining Structural Modification with In-Situ Transfer Path Analysis to Solve Noise and Vibration Problems
Andrew Elliott
[Abstract]
In-situ TPA measurements provide blocked forces that can
be used to diagnose structure borne noise problems in
vehicles. One of the main advantages of in-situ TPA, also
known as blocked force TPA, is that all measurements are
made with the source and receiver in a coupled state
making measurements easier and ensuring representative
operation of the vibration source. Perhaps the greatest
advantage however is that the blocked forces obtained,
unlike contact forces, are an intrinsic property of the
vibration source which means they remain valid for modified
or completely different receiver structures. In the paper, real
world practical examples that exploit the advantages of in-
situ TPA in terms of structure borne noise prediction and
structural modification are presented. The first example is of
a complex vibration source coupled at multiple points
through six degrees of freedom per interface to a test bench
that is subsequently built into a new assembly for which
structure borne noise predictions are made. A further
example is then used to demonstrate how structural
modifications can be made to the receiver side of an
assembly for the purposes of troubleshooting noise and
vibration problems using in-situ TPA.
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In-Situ Transfer Path Analysis of Multiple Vibration Sources in a Complex Source- Receiver Assembly
Lucy Susan Barton, Andrew Elliott, Andy Moorhouse, John Smith
[Abstract]
Depending on where the source-receiver boundary is defined within an assembly, an airborne sound source can also be considered as a structure borne sound source. For example, if the internal forcing mechanisms within a machine are regarded as the sources, the rest of the assembly including the surrounding air can be considered the receiver. If the structure borne noise resulting from these internal forcing mechanisms can be predicted in real-time, the radiated noise can potentially be predicted and managed- either actively or passively- using numerous approaches. The primary aim of the work presented in the paper is the development of an accurate method for the real-time prediction of structure borne noise, whilst continually monitoring the reliability of these predictions. To facilitate this process, it is necessary to also monitor the stability of the vibro-acoustic frequency response functions that relate the internal forcing mechanisms to the receiver position, and to eliminate the influence of background noise resulting from noise or vibration sources other than those of interest. Presented in the paper are the results of a laboratory transfer path analysis case study of a source-receiver assembly composed of multiple sources attached to a single receiver structure.
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Analysis of a vibrating structure as an airborne sound source by means of matrix inversion
Serafima Anisovich, Roland Sottek, Matthias Wegerhoff
[Abstract]
The decomposition of a complex airborne sound source into a
number of simple sources has found wide application in transfer
path analysis. One of the most advanced ways to solve this task
is to use the matrix inversion method. This method works well for
describing concentrated sources like monopoles. In practice,
however, it is also desirable to analyze distributed sources such
as a vibrating structure (e.g., the housing of an engine). A large
number of grid points, representing emitting partial surfaces,
and an even larger number of measurement points complicate
the task. The large matrix of transfer functions between source
points and measurement points that need to be inverted is
often ill-conditioned. Mathematical approaches such as
regularization techniques are required for the matrix inversion.
Understanding the ways to apply these approaches and their
physical meaning is one of the most important steps in finding
suitable solutions to the mathematical inversion problem without
losing physical information. In this paper, some of the most
common matrix inversion strategies are analyzed for the
example of a vibrating plate.
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Can I trust my TPA results?
Frank Jürgens, Christian Nettelbeck, Philipp Sellerbeck
[Abstract]
In the automotive development process, Transfer Path Analysis (TPA) is a widely known tool for troubleshooting purposes of existing vehicles or for the prediction of vehicle interior noise based on test bench or CAE data. There are many different approaches, each having its pros and cons. Either determining interface forces while the source is disconnected or equivalent forces for in-situ condition; measured in the vehicle or at the test bench; calculated in frequency domain or in time domain. As different as the approaches are in detail, one question they all have in common: How reliable are their results? Within this paper, the need of a validation and refinement process of the TPA model - independent of the specific TPA method - is discussed and illustrated by typical application examples. New tools like the Mosaic View for the visualization of the crosstalk and useful procedures like the Root Cause Analysis and the Model Robustness Check are exemplified. Furthermore, these technologies are evaluated to show their potential for the identification of errors in the TPA model and how to deal with them in order to increase the liability.
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Physics-Informed Transfer Path Analysis with Parameter Estimation using Gaussian Processes
Christopher Albert
[Abstract]
We apply a variant of Gaussian process regression to augment
experimental data from transfer-path analysis (TPA) by known
information about the underlying physical properties of the system
under investigation. The approach can be used as an alternative to
model updating and is also applicable if no detailed simulation model
of the system exists. For vibro-acoustic systems at least three
features are know. Firstly, observable quantities fulfill a wave
equation or a Helmholtz-like equation in the frequency domain.
Secondly, the relation between pressure/stress and
displacement/velocity/acceleration are known via constitutive
relations involving mass density and elastic constants of the
material. The latter also determine the propagation speed of waves.
Thirdly, the geometry of the system is often known up to a certain
accuracy. Here we demonstrate that taking into account this
information can enhance TPA results and quantify their uncertainties
at the same time. In particular this is the case for noisy
measurement data and if material and geometry parameters are
(partly) unknown. Due to the probabilistic nature of the procedure
unknown parameters can be estimated, making the method also
applicable to material characterization as an inverse problem.
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Obtaining method of high contributing body and frame vibration behavior to road noise using principal component contribution analysis
Hiroki Taguchi, Kanon Nukata, Junji Yoshida
[Abstract]
Recently, principal component (PC) contribution analysis
utilizing operational TPA was developed to find out high
contributing vibration behavior of the target structure to the
interior noise. However, in case a vehicle is composed of
body and frame, identifying the high contributing part
between them is difficult using this method. In this study, we
then attempted to propose an identification method of the
high contributing part between body and frame using the
PC contribution analysis sequentially. In the test, a simple
vehicle model was employed and vibration at multiple points
on the body and frame were measured simultaneously with
the interior noise when random input signals was given under
the tire. Subsequently, PC contribution analysis was applied in
two steps. In the first step, high contributing body vibration
behavior to the interior noise was extracted and then, high
contributing frame vibration behavior to the important body
mode was obtained in the second step. As the result, the
frame vibration behavior was found to be the main part
increasing the interior noise at the target frequency. Finally,
the interior noise at the target frequency could be
decreased by applying intensive countermeasure to the high
contributing frame vibration behavior.
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Organization:
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S. Lutzenberger, Müller-BBM |
A. Kuijpers, M+P |
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The Application of Dither for Suppressing Curve Squeal
Wolfgang Kropp, Arthur Aglat, Jannik Theyssen, Astrid Pieringer
[Abstract]
Curve squeal is a highly disturbing tonal sound generated by vehicles like railways, metros or trams, when negotiating a sharp curve. The probability that squeal occurs increases with reduced curve radius of the track. Curve squeal noise is attributed to self-excited vibrations caused by stick/slip behaviour due to lateral creepage of the wheel tyre on the top of the rail. With respect to the enormous number of the rolling stock units and the long lifetime of waggons there is an urgent need for a cheap and simple retrofitting measure to reduce curve squeal. The main objective of paper pro is therefore to investigate the potential to reduce curve squeal by means of active control in the form of dither in an efficient and robust way. Dither control has been applied in the field of mechanical engineering for systems including non-linear components. There it has been shown to suppress self-excited oscillations very efficient. The control is an open loop control. It consists in adding a forced vibration to the vibrational system. The demand on this additional signal is that it is higher in frequency than the friction-induced response. From a physical point of view, dither control modifies the effective friction characteristic.
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The Effects of Weather Conditions and Wheel Wear on Curve Squeal
Thomas Maly, Florian Biebl, Michael Ostermann
[Abstract]
The mechanisms of curve squeal generation (lateral stick-slip effect on the top of the rail and/or friction of the wheel flange) are well known. Even if various influencing parameters were already investigated mostly on a theoretical level, in real train operation the parameters are in general not measurable in their entirety and their effects due to the complex interdependencies between them are often not predictable. That is why curve squeal generation often seems to have a random character. As a continuation of previous studies the aim of the work presented was to investigate the influence of weather condition changes over a whole year and wheel profile parameters changes due to wear on the occurrence frequency of curve squeal. For this, measurements of the pass-by noise emissions and the weather conditions in a narrow curve (radius 230m) of a Vienna suburban train line were carried out over a period of 11 months. Due to a previously developed algorithm for automatic detection of curve squealing more than 20000 data sets were available for subsequent statistical analysis. In the paper the dependencies of squeal frequency on rain, air humidity, dew point, and rail temperature, as well as wheel parameters are presented in detail.
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Another Halving of Rail Freight Traffic Noise by Wheel Absorber
Martin Fehndrich
[Abstract]
By 2020 the freight traffic noise in Germany and Switzerland should be halved by changing cast iron brake blocks wheels to composite brake blocks. By that levels 93 dB(A) should become 83 dB(A), which is the limit defined in the TSI Noise. A further halving reduction would lead to 73 dB(A), which is part of a SBB Cargo vision for its future fleet.
The fact this reduction is possible is shown by first results of the SBB Cargo’s 5L project, where with the help of a number of suppliers a train with 16 freight wagons with a series of different noise-reducing measures (e.g. disc brakes instead of block brakes) was built, which is tested now in regular traffic.
Since the wheel is still the predominant source of sound emission in railway traffic, the biggest impact is reducing the wheel noise. In SBB Cargo’s 5L project wagons with BVV wheels and BVV wheel absorbers sound levels of 73 to 75 dB(A) and an equivalent of no more than 75 dB TSI level could be achieved.
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Pass-By Noise Source Identification for Railroad Cars using Array Measurements
Hans Rudolf Graf, Christian Czolbe
[Abstract]
Microphone arrays are suitable for acoustic source localization.
For good accuracy, adequate time averaging is required. With
fast moving sources, the time interval during which a source
remains within a focal spot is very short, resulting in poor
accuracy. By tracking and averaging the sources during pass-
by, available averaging time and consequently accuracy can
be substantially improved. The technique is demonstrated by
pass-by measurements of innovative freight cars investigated as
part of the research project www.innovativer-gueterwagen.de
by the Federal Ministry of Transport and Digital Infrastructure
(BMVI). The measurements were carried out in Spring 2018 on
the test track in Wildenrath, Germany. In the post-processing,
the acoustic source intensity maps were integrated for areas of
interest, such as the wheelsets, rail sections and frameworks,
resulting in sound power level of the corresponding
components. Results of selected noise reduction measures will
be presented.
Spatial resolution of arrays is rather limited at low frequencies.
Therefore, noise sources of the rail and wheel cannot be clearly
separated based only on array results. Additional signals of
structure borne noise of the rail may be used to improve the
separation of wheel and rail related sound radiation.
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Rail roughness surveys – a tool for effective environmental noise control
Lisette Mortensen, Stig Junge
[Abstract]
A mobile method developed by Sweco for estimation of rail roughness level on a large scale is presented. The method is suitable for countrywide surveys and the data has proved to be very useful in the planning of track maintenance both in terms of environmental noise and rail quality.
During the last two years roughness measurements have been carried out twice on the Danish railway network and all roughness data from these campaigns have been stored in Banedanmark’s track measurement database. Banedanmark has developed a tool to rank track sections according to rail quality and population density in residential areas to most effectively reduce the amount of environmental noise from the track and to document the effect of noise reduction measures.
The measurement method uses measurement of noise from the contact point between wheel and rail to derive an estimate of the rail roughness level and is based on the transfer function from rail roughness to acoustical noise. The method has been tested at measurement train speeds from 60 - 120 km/h and shows a high degree of reproducibility.
Banedanmark has given these measurements a significant role in planning the large-scale maintenance operations of grinding and milling.
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Organization:
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C. Spehr, DLR |
T. Geyer, BTU Cottbus-Senftenberg |
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Aircraft noise: Conversion of an existing to a desired number of subtracks with identical lateral dispersion to obtain smooth noise contours
Olivier Schwab
[Abstract]
In aircraft noise calculations, the dispersion of flight tracks on a certain air route is often represented by a backbone track and a number of laterally displaced subtracks, where each track is assigned a fraction of the route’s movements to obtain a certain lateral distribution. For routes with a large dispersion in the flight tracks, the number of subtracks should be larger than for routes with small dispersion to obtain smooth footprints (i.e., single event level grids) in both cases. However, if subtracks are drawn by hand using expert knowledge, e.g. because no reference radar data is available, the number of available tracks is usually limited. Here, it can be useful to be flexible in the creation of additional and/or modification (smoothing) of existing subtracks.
In this contribution, we present a method to construct a different (usually larger) number of subtracks based on a set of existing subtracks by geometric matching of the subtracks with the backbone track. The construction assumes identical lateral dispersion of the converted tracks as the original subtracks. We discuss example cases featuring curves with tight turns and non-symmetric subtracks with respect to the backbone track, and compare resulting footprints with those of the original tracks.
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SonicBAT: Some Highlights and Subsequent Developments
Victor W. Sparrow, Trevor Stout, Kevin Bradley, Christopher Hobbs
[Abstract]
The Sonic Booms in Atmospheric Turbulence (SonicBAT) project concluded in 2018. The overall goal of SonicBAT was to develop tools to predict the influence of atmospheric turbulence on shaped sonic booms for newly proposed low-boom supersonic passenger aircraft. There was a substantial experimental component to obtain statistically-useful datasets of recorded sonic booms in both dry and humid environments. The project was carried out by a large team, and the authors of this paper are a small subset of the individuals involved. The purpose of this presentation is to highlight some of the main findings of SonicBAT and to provide pointers to the primary technical report and other publications that are now coming available, both from SonicBAT and from subsequent findings. Overall, SonicBAT was successful in developing multiple ways of predicting the influence of atmospheric turbulence on the propagation of arbitrarily shaped sonic boom waveforms as they propagate through the earth’s planetary boundary layer. The influence of turbulence is profound for N-wave sonic booms but is reduced for low-boom sonic boom signatures. [Work supported by NASA.]
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Calculation of the Fan Rotational Speed Based on Flyover Recordings for Improving Aircraft Noise Prediction Models
Roberto Merino-Martinez, Mirjam Snellen, Dick G. Simons
[Abstract]
Noise control regulations in the vicinity of airports usually rely on the
estimations of aircraft noise levels by noise prediction models. Current
best-practice noise contour calculation methods assume default
engine thrust values depending on the engine type and the altitude of
the aircraft. These prediction tools provide a single noise level for a
certain aircraft in a certain flight phase and at a specific distance from
the observer. In practice, however, changes in the thrust occur and
cause variations in the noise levels of several decibels. In this paper, an
approach is presented to estimate the fan rotational speed N1% (and
hence the thrust) directly from flyover audio recordings. This method
estimates the blade passing frequency (BPF) of the fan by searching its
characteristic tonal peak (and its higher harmonics) and accounting
for the Doppler effect. This method was applied to more than a
thousand measurements of Airbus A330-300 and Boeing 777-200
aircraft. The results show a significant correlation between the
recorded noise levels and the fan rotational speed, explaining up to
45% of the variability in noise levels. Considering the calculated N1%
values in the noise prediction models notably increases the agreement
with the recorded noise levels.
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Numerical and Experimental Investigation of the Turbofan First Booster Stage Tone Noise
Victor Mileshin, Sergey Pankov, Anton Rossikhin
[Abstract]
An important problem of turbofan design is maintenance of low
noise emission. One of the components of turbofan noise is the
tone noise arising due to inter-row interactions in the booster
stages. For modern turbofans this component of noise can be
significant at approach certification point. The investigations of
the tone noise of the first booster stage have been carried out in
CIAM during last several years. They were performed using
frequency domain method for simulation of tone noise of
multistage turbomachines developed in CIAM. The advantage
of the method is the high speed of calculations; however it relies
on some ad hoc assumptions about the process of interactions
between flow field disturbances and blade rows. This work
presents the results of the tone noise calculation of the first
booster stage using time domain approach based on multirate
Runge-Kutta scheme. The result, obtained in the calculation, is
independent from any assumptions about the interactions in the
multistage turbomachine. This allows verifying the results of
previous calculations. In general satisfactory correspondence
were found between the results of computations in time and in
frequency domains and between them and experimental data
obtained in the CIAM C-3A acoustic test facility.
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Organization:
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W. Kropp, Chalmers University of Technology, Division of Applied Acoustics |
C. Hoever, Continental Reifen Deutschland GmbH |
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The influence of tyre cavity resonances on the exterior noise
Wolfgang Kropp
[Abstract]
Rolling noise due to tyre/road interaction is mostly explained by
tyre vibrations. The contribution of air-pumping might be an
important contributor for certain cases. This observation has
often been based on evaluating the speed exponent observed
in measurements of the radiated sound. However simulations
show that in many cases the observed speed exponent can be
explained by tyre vibrations. In addition simulation show that for
rough roads the resonances in the tyre cavity can play an
important role. In the paper the influence of sound generation
in the tyre cavity is discussed in more detailed. For this the
Chalmers Tyre/Road Interaction model is utilised. The simulations
clearly show the contribution of tyre cavity resonances to rolling
noise as function of surface type. The results indicate that
additional damping in he cavity would not only be beneficial
for the interior noise inside the vehicle but in many cases also for
the exterior rolling noise due to tyre/road interaction.
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Prediction of dynamic hub forces as a source of structure-borne tire/coarse road noise using a high-fidelity simulation approach
Daniel De Gregoriis, Frank Naets, Peter Kindt, Wim Desmet
[Abstract]
When optimizing the tire/road structure-borne interior
noise, multiple other coupled tire performances need to
be simultaneously optimized. Typically, different tire
design prototypes are built and tested. However, with
the recent advent in virtual product development and
the Digital Twin concept, high-fidelity numerical
simulation approaches become a viable alternative to
the experimental approach. These numerical
approaches allow for e.g. design space exploration
without the need for building and testing physical tire
prototypes. Therefore, in this work the use of a fully
predictive nonlinear numerical approach for the
prediction of the dynamic contact- and hub forces of a
tire rolling over a coarse road surface is described.
These predicted hub forces can then be further used to
assess the tire/road structure-borne noise performance
of a tire. The influence of incorporating tire model
nonlinearities as well as the air cavity and rim flexibility
on the simulated hub forces is discussed. Furthermore,
nonlinear model order reduction and hyper-reduction
techniques are applied to greatly reduce the numerical
simulation time, being approximately ten times faster
than the experimental approach. Finally, a comparison
between the predicted hub forces and experimental
hub forces is shown, where a sufficient correspondence
between both can be observed for design engineering
purposes.
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Use of waste and marginal materials for silent roads
Lily Poulikakos, Sahand Athari, Peter Mikhailenko, Zhengyin Piao, Muhammad Rafiq Kakar, Moises Bueno, Reto Pieren, Kurt Heutschi
[Abstract]
Inhabitants of the urban environment are exposed to high
levels of traffic noise and increase in amounts of waste. One
way to mitigate their exposure to noise is to take advantage
of the existing public spaces between buildings and
transform the available surfaces of pavements into elements
that foster comfort and health for urban residents.
This work aims to develop low noise pavements by using
certain waste materials. Semi-dense asphalt (SDA) mixtures
(Air Voids = 12-16%) will be used in this study due to their
acoustic properties. Parts of the aggregate matrix will
incorporate i) crumb rubber (CR) and ii) recycled concrete
aggregates (RCA). The various mixtures produced will be
compacted into 100mm dimeter specimens using the
gyratory compactor which will be subject to texture
measurements by 3D laser profilometry, porosity measure-
ments and sound absorption measurements by the
impedance tube method (EN 10534-2). This study will give us
an understanding on how waste materials impact the texture
and acoustic properties of roads and ultimately, to what
degree they can be incorporated in low-noise asphalt
mixtures.
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A LIFE NEREiDE test track with a poro-elastic pavement in Belgium
Luc Goubert
[Abstract]
A poro-elastic road surface (PERS) is an innovative pavement consisting of stone aggregates and rubber granulates, bound with an elastic resin. It was invented around 1978 in Sweden and can yield unequalled reductions of the tyre/road noise: between 7 and 12 dB, whereas the best performing ”conventional” noise reducing pavement, two-layer porous asphalt, reaches maximum 7 dB. Since the invention of PERS, in several countries experiments have been carried out demonstrating the major drawback with this pavement type: its vulnerability and limited lifetime. In the FP7 PERSUADE project (2009-2015) a holistic approach was followed covering all relevant aspects. Several field tests on trafficked roads revealed that the prefab PERS slabs, glued on a rigid under layer is the most promising PERS technology. The LIFE project NEREiDE (2016-2020) focuses on the use of recycled materials in noise reducing pavements and comprises a work package about PERS. In the frame of the LIFE NEREiDE project, a 44 m long test track was constructed on a road in Gent, Belgium with prefab poro-elastic slabs, building on the technology developed in the PERSUADE project. This contribution outlines the construction process and the first test results obtained in the subsequent monitoring program.
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CFD Modeling of Pressure Variation in a Road Cavity with Volume Variation
Marianne Bou Leba Bassil, Julien Cesbron, Philippe Klein
[Abstract]
This paper deals with the variation of air volume at the contact interface between a rolling tyre and a road
surface leading to aerodynamic phenomena such as air-pumping. A numerical model was developed (in 2D
and 3D) based on Computational Fluid Dynamics, using the Fluent solver, coupled with a displacement
function of the bottom of the cavity representing its volume variation. Results showed that the maximum
pressure reached at the bottom of the cavity during its complete closure increased as its volume decreased.
The pressure variation was related to the volume variation by a relationship close to Laplace law for perfect
gases. In addition, pressure oscillations after the opening of the cavity increased in amplitude and frequency
as its volume decreased. Moreover, in the case of INRETS (1990) experimental setup, introducing the volume
variation estimated by Laplace Law in the 3D model has shown better agreement between the calculated
pressure and the measured pressure in comparison with the model neglecting this volume variation. Future
work would therefore be to obtain the actual volume variation caused by the penetration of the tyre tread
inside the cavity. This could be done either experimentally or numerically using a tyre/road contact model.
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Parametric Study of the Estimation of Indoor Trolley Wheel Stiffness for Use in a Rolling Noise Prediction Model
Matt Edwards, Fabien Chevillotte, François-Xavier Bécot, Luc Jaouen, Nicolas Totaro
[Abstract]
In vehicle tyre/road contact modeling, dynamic models are typically used which incorporate the vehicle’s suspension in their estimation: thus relying on a known stiffness to determine the movement of the wheel in response to roughness excitation. For the case of a wheeled device rolling on a floor (such as would be the case for a delivery trolley moving merchandise around inside a commercial building), there is often no suspension, yet the wheel is still too soft to able to be considered mechanically rigid (as is the case in train/rail contact). A model which is aimed at incorporating the dynamic effects of the trolley in predicting the sound generated by rolling needs to provide a robust way of estimating the wheel’s effective stiffness. This work presents an original technique for estimating the stiffness of an indoor trolley wheel. A parametric study was conducted in order to identify the dependence of the wheel stiffness on each of the relevant variables: including the wheel’s radius, axle size, width, applied load, and material properties. The methodology may be used to estimate the stiffness of new wheel types (i.e. different geometries and materials) without needing to solve a finite element model each time.
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Organization:
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W. Kropp, Chalmers University of Technology, Division of Applied Acoustics |
J.S. Bolton, Herrick Laboratories/Purdue University |
T. Hashimoto, Seikii University |
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Asphalt-surface defects detection, based on Tyre/Road noise analysis and geo-processing
Carlos Andrés Ramos Romero, César Asensio
[Abstract]
A new approach to detect different asphalt defectology based on tyre/road signal-noise analysis by Machine Learning algorithms is proposed. The method exploits the acoustic signal generated by the rolling at the tyre-asphalt proximity zone. The audio data is recorded by a sensored vehicle that travelled an established route with supervised surface conditions. The acoustic data have shown to be relevant on automatic asphalt-defect classification by features extraction in frequency domain.
In this study, local and distributed road-surface defects were classified using supervised pattern recognition algorithms. This new approach does not require specific on-board instrumentation, so any vehicle can easily become an asphalt status tester with the installation of a microphone.
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Investigation on Abnormal Gear Rattle Noise in Automated Manual Transmission (AMT) after Gear Upshifting
Abhishek Lakhanlal Vaishya, Archan Sunilbhai Pujara, Devendra Kumar Khare, Kingshuk Satpathy, Vivek Singh
[Abstract]
Customers are getting delighted after receiving best in class vehicle NVH performance. In order to enhance customer NVH comfort, a specific abnormal gear rattle noise phenomenon during product development was investigated. The transmission produces gear rattling noise by the impacting of gears oscillating through their gear backlash. Similar noise was instigated immediately after gear upshifting, sustains for some time and then inaudible. In this case, the hearing frequency band is found higher than the usual gear rattling frequency band. Along with the assessment methodology, root causes of this abnormal gear rattle noise were discussed in this paper.
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Study for Whoop Noise Reduction in passenger vehicle with Clutch Hydraulic System
Devendra Kumar Khare, Anant Saran Pandey, Abhishek Lakhanlal Vaishya
[Abstract]
The noise, vibration and harshness (NVH) is an important
aspect
in driving comfort improvement. Low noise and vibration is
one
of the primary requirements of passenger cars which play an
important role in the customer’s perception. It includes a
silent
cabin with minimum vibrations and quite operation of the
parts.
In a vehicle, engine vibrations are transferred to clutch pedal
through fluid pulsation inside the clutch operating system. In
some cases, vibration gets amplified while operating clutch
and
a noise is generated which is called a whoop noise. Sound
pressure level of whoop noise should be low to enhance
customer comfort. The biggest challenge for NVH engineers
in
present scenario is to strike a balance between cost and
good
NVH. This paper elaborates the study of root cause analysis and
reduction in whoop noise generated during clutch pedal
operation.
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A Study of the Interaction Between Vehicle Exterior Noise Emissions and Vehicle Energy Demands for Different Drive Cycles
Johan Nygren, Susann Boij, Romain Rumpler, Ciarán O'Reilly
[Abstract]
This initial study investigates the interaction between vehicle noise emissions and the energy required to move the vehicles along different drive cycles. There is often a trade-off between reducing noise emissions and at the same time reducing other environmental impacts. A vehicle’s energy demand associated with a specific drive cycle may be affected when a different route is chosen between two locations to reduce the noise exposure at certain observer points. The methodology of the study was to use the existing IMAGINE traffic noise model as a source model, and to evaluate the sound exposure level (SEL) at observer points as a function of instantaneous sound pressure level estimates as the source moved from one location to another via two different routes. A noise impact estimate with a linear dependence on the difference between the SEL and a threshold level was proposed. Also, the energy demand for each route was calculated. The results indicated that there is a difference between the two routes if the aim is to reduce the noise exposure or the energy demand. Suggested future research is to further improve the noise impact evaluations in the context of very short durations of exposure.
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Development of a Thermally Insulating Vibration Damping Compound
Alexander Rasa
[Abstract]
Attenuating vibration in structures, such as marine vessels and rail carriages, through the use of damping materials can return a wide range of benefits from reducing air-borne and structure-borne noise to improving component life. Thermal insulation is typically also installed alongside vibration damping materials, preventing condensation and the transfer of thermal energy, therefore improving energy efficiency and the comfort of passengers and crew, along with increasing component life through the reduction of moisture. This paper details the development and evaluation of a single material, Decicoat T35, which is both vibration damping and thermally insulating, allowing easier and faster treatment over traditionally installing multiple materials. The compound is water-based, non-toxic and easily sprayable. The material has been evaluated to EN 12664 for thermal conductivity and to an internally developed method for anti-condensation in rail and marine industries. Decicoat T35 was also investigated using a variety of methods for vibration damping, including dynamic mechanical analysis (DMA) to ISO 6721-5 and experimental modal analysis (EMA). Furthermore, vibration damping performance was compared to a similar competitors sprayable thermal compound, highlighting the elevated vibration damping properties of the newly developed material. A thermally insulating vibration damping compound was successfully developed and evaluated.
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Organization:
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T. Van Renterghem, Ghent University, Department of Information Technology, WAVES research group |
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Modelling reflections from single trees and entire forests
Jean Marc Wunderli
[Abstract]
A model for the prediction of diffuse reflections from forested areas is presented. A single tree is thereby modelled as a cylinder of finite height, representing the trunk, and a number of spheres to account for reflections from the crown. The scattering properties of the involved structures are derived from the classical analytical solutions, complemented by a Fresnel zone weighting. For performance reasons not every tree in a forest is modelled individually but representative super trees are distributed over forested areas with a finer local resolution along the forest edges and a coarser representation in the depth of the forest. The model is formulated in one-third octave bands and yields sound exposure level as well as maximum sound pressure level. As the depth of the forest is automatically scaled correctly it can be applied for small-scaled vegetation and is for example capable of predicting levels behind forested barriers or berms. Comparisons with measurements showed a generally good agreement.
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Applying the diffusion equation to urban scenarios: Computational analysis of the diffusion coefficient
Raúl Pagán Muñoz, Juan Miguel Navarro Ruiz, Maarten Hornikx
[Abstract]
Prediction of sound propagation is playing a key role in the planning and design process of urban areas and during the last decades different techniques have been developed for the computation of the sound fields in cities. Among them, the diffusion equation, based on the propagation of sound energy, is a simple and attractive tool for certain scenarios where the diffuse field is predominant, such as inner city environments. The diffusion equation is a well-known and efficient method to compute sound fields in rooms with a low amount of absorption. However, this method, which includes the diffusion coefficient describing the diffusivity of the energy propagation, requires further understanding for urban environments. In this work, the diffusion coefficient has been evaluated using wave-based numerical techniques for simplified two-dimensional urban scenarios. The parameter is computed spatially from the acoustic intensity and sound energy density fields in every direction for different geometries. This paper investigates the directional components of the diffusion coefficient.
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Solution of wide-angle parabolic equations for long-range sound propagation in a moving medium
David Keith Wilson, Michael Muhlestein, Vladimir Ostashev, Michael J. Shaw, Michelle Swearingen, Sarah McComas
[Abstract]
Narrow-angle parabolic equations, which are widely used for
outdoor sound propagation, are suitable for propagation
angles up to 15∘-20∘ off the main axis (generally the horizontal
direction). Wide-angle parabolic equations (WAPEs) are thus
needed to accurately solve long-range problems involving
refraction and scattering from elevated layers, such as audible-
range propagation in stable nighttime conditions, and
infrasound refraction by the stratosphere. However, it is difficult
to derive WAPEs that are numerically feasible to solve while
properly accounting for motion in the propagation medium (i.e.,
wind in the atmosphere). As a starting point, we adopt here a
recently derived extra-wide-angle parabolic equation (EWAPE)
for moving media which accounts for the propagation angles
up to 90∘ with respect to the nominal propagation direction. A
WAPE is then derived from the EWAPE using a Padé (2,2)
approximation, which is valid for angles up to roughly 55∘. The
resulting equation is generally suitable for long-range sound
propagation applications in a moving atmosphere. We consider
here solutions for example problems involving ducting and
shadow zones. The solutions are compared to narrow-angle
approximations and to wide-angle solutions based on the
effective sound-speed approximation.
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Uncertainty Analysis of Environmental Sound : Analysis of a Series of Field Experiments
John Fenlon
[Abstract]
A series of experiments used attended and unattended
instruments at two primary sites: (i) adjacent to the A428 (a
second-tier road), and (ii) adjacent to the West Coast
mainline railway. These are both rural sites, so that the
traffic / rail noise will tend to dominate even at some
distance from the noise source. Various configurations of
monitors at different distances from the noise source, on
different days and for different times were tested, and
sound measurements were integrated over 5-minute
(road) or 1-minute (rail) periods. At each site a series of
measurements were taken during July and August in two
separate years, which included several 4-day sets, using
two different types of noise meter. The underlying aim of the project was to develop a simple
ëuncertainty’ budget for the measurement of
environmental noise. The primary interest was in
determining day-to-day variation, differences between
instruments (at the same position), and differences
between measurements at different distances from the
source. One set of experiments also tested the difference
between measurement positions chosen by different
engineers.
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Model Test Stand for Acoustic Scattering
Wolfram Bartolomaeus
[Abstract]
Scattering, diffraction and reflection on acoustical hard or soft obstacles affect the sound field in its propagation.
The scattering of sound waves on an acoustical hard cylinder was calculated analytically using a harmonic development according to Bessel functions, simulated with the finite element method (FEM) and measured in the Laboratory for Acoustic Modeling on a scaled level. A good agreement between all three results was found.
Similarly, the diffraction of a sound pulse at an edge was analytically calculated using the theory of Biot, Tolstoy and Medwin (BTM) and simulated with an FEM model. These simulations and the room impulse response measured in the Laboratory also agree satisfactorily.
For further measurements, the new Acoustic Scattering Model Test Stand (MPaS) can be used also on objects or sets of objects that no longer can be described analytically in an easy way. Likewise, measurements at diffraction edges and reflection planes are feasible. These can be compared with analytical solutions or at least with FEM simulations.
Thus, the MPaS test stand provides an important tool for modeling the propagation of the sound field over obstacles, which makes it possible in the future to construct e. g. optimized noise barrier attachments for roads.
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On Generation And Propagation Of Acoustic Surface Waves Over Rough, Periodic Surfaces
Shahram Taherzadeh, Alex Stronach, David Berry, Keith Attenborough
[Abstract]
An acoustic surface wave is an evanescent wave that
spreads cylindrically along the surface but decays
exponentially with height above the surface. This wave
travels at a little less than the speed of sound in air and is
primarily related to the pore or surface. Surface waves are
generated also when sound propagates over rough hard
surface. This phenomenon is related to coherent multiple
scattering from roughness elements.
This paper investigates conditions under which surface
waves are generated and propagated and the
relationship between this surface wave frequency and the
geometry are explored by measurements of the sound
field due to a point source above periodically-spaced
rigid strips placed on a rigid surface and by a Boundary
Element Method in time-domain.
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A data bank of outdoor transfer functions
Sylvain Cheinet, Loic Ehrhardt, Matthias Cosnefroy, Adrien Dagallier
[Abstract]
From the viewpoints of sound sensing and noise pollution applications, the propagation channel can be modeled by a transfer function. This transfer function is a sort of Grail, which readily connects any near-field signature to its propagated counterpart. In the scenario of outdoor propagation above a flat ground, the atmospheric conditions vary, and a data bank of transfer functions is needed.
This study addresses the computational and physical challenges of forming such a data bank. It uses a three-dimensional, Finite-Difference, Time-Domain (FDTD) solver for propagating a reference wideband signal at 200 m above a finite-impedance ground. The simulations span over a wide range of meteorological conditions with consistent turbulence realizations.
The formed transfer functions are compared to recent atmospheric-acoustic measurements. Their sensitivity to the wind direction and strength is discussed. The data bank is unique in its extensiveness and physical realism, and can be used for applicative purposes.
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Comparison of ASJ RTN-Model 2013 and the Harmonoise Engineering Model under Thick Barrier Configurations
Takuya Oshima, Azusa Hoshikawa, Yumi Kurosaka
[Abstract]
Engineering models for environmental noise are developed in
many countries under different backgrounds. Such differences
make each model have different characteristics. In European
countries, comparative studies have been carried out to
illustrate the characteristics of models such as Nord2000,
Harmonoise and CNOSSOS-EU. However, comparison of the
Japanese road traffic noise model, ASJ RTN-Model 2013, to
another model has not been made. The ASJ model uses A-
weighted levels throughout the calculation process, unlike
European models that use either octave-band or 1/3-octave-
band levels.
In this study, a methodology to derive consistent quantities
between the ASJ model and the European Harmonoise model,
such as the single-vehicle A-weighted power level, is
established to make strict comparisons between the two
models. The methodology is applied to thick barrier test cases
that are included in a Harmonoise deliverable. At the same
time, finite-difference time-domain simulations are carried out
to obtain reference solutions. A-weighted source strengths
between the two models are found to be larger for a light
vehicle of the ASJ model. Comparison between each model
and the reference solutions weighted by each source spectrum
agree well for both models. The final A-weighted levels
obtained by both models agree within a 2 dB difference.
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Managing the Uncertainty of Long-distance Sound Propagation from a Large Industrial Noise Source
Tim Procter, Deanna Tomerini, Alan Lex Brown
[Abstract]
The attenuation of a sound signal over long distances is
accounted for by atmospheric absorption, atmospheric stability,
ground effects, barriers and spherical spreading. Conversely,
the propagation of sound over long distances can be
enhanced by source to receiver gradient winds and vertical
temperature profiles that create downward refraction of the
sound waves. While the nature of each of these phenomena is
well understood the prediction of the sound propagation over
long distances from a large industrial source such as an open
cut mine is inconsistent with the measured sound pressure level
attributed to the source. The uncertainty around the theoretical
model predictions and the measured source sound pressure
levels in a multi-source environment creates a control problem
when managing day to day mining operations. This paper
investigates the relationship between the measured and
predicted sound pressure levels at three continuous noise
meters along a propagation path at distances of 500m, 1500m
and 2500m from the source. The objective of the analysis was to
reduce the uncertainty in the measurement of the source’s
contribution at the noise monitor located in a multi-source
environment 2500m from source.
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Selection of the correction due to receiver location when performing environmental noise measurements using microphone array
Luka Čurović
[Abstract]
Measurement and assessment of environmental noise is in many cases performed using procedures described in ISO 1996 - 2:2017 standard. As the measurement uncertainty is as important as the measurement result the standard gives guidance on the calculation of measurement uncertainty which is compliance with ISO/IEC Guide 98-3 (GUM). The sound pressure level is normally considered as the free field sound pressure level. But the measurement used in regulations are often performed in front of reflecting surfaces. In such cases one of the input quantities that determines the measured value is an input quantity to allow for any uncertainty due to selection of receiver location (δ_loc) that can take values from 0 to 6 dB. δ_loc has a profound effect on the measurement result. The choice of the right value is in many cases not trivial and only defined for traffic noise. In our study we will present measurements performed with a microphone array that can help in selecting the optimal measurement location when performing the noise measurement due to industrial sources in front of reflecting surfaces.
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Propagation of Bird Vocalizations in the Alpine Environment
Didier Dragna, Loïc Berger, Sébastien Ollivier, Frédéric Sèbe
[Abstract]
Rock ptarmigan is a bird living in the mountains tops, whose population tends to decrease in Europe over the last decade, notably due to climate change, habitat loss and human disturbance. Censusing of number of birds in the Alps is realized via point-count monitoring using birds vocalization. However, detection of bird vocalizations by the human hear is somewhat tricky in these non-homogeneous landforms, due to sound propagation effects. In order to improve the counting method, it is necessary to understand how bird vocalizations propagate in the Alpine environment. With this aim, a propagation model based on a ray-tracing approach is developed and includes topography, ground reflection and absorption and atmospheric refraction and absorption. The model is first validated in both frequency and time domains against reference results obtained with a numerical solution of the linearized Euler equations for simple cases. Propagation of the bird vocalization in the Alpine environment is then illustrated. It is shown that the active space of the bird dramatically depends on the environment. The modeling of active and detection space will provide suitable information for birds monitoring, enabling managers to define more precisely their counting plans.
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Prediction of atmospheric sound propagation subject to parameter variability of atmospheric turbulence
Jasmin Hörmeyer, Clemens Hübler, Tobias Bohne, Raimund Rolfes
[Abstract]
As part of the project ”WEA-Akzeptanz” an atmospheric sound propagation model will be developed to predict the sound immission from a wind turbine. The CNPE- (Crank Nicolson Parabolic Equation) method, based on the Helmholz equation is chosen as the sound propagation model. Complex environmental conditions such as atmospheric turbulence can be implemented with a random number generator in this model. Atmospheric turbulence is characterized by temporal fluctuating wind speed and temperature, which has an impact on the sound propagation and leads to scattered sound waves due to the turbulent eddies. Accordingly, the input of the model contains parameter variability. As a result, the sound pressure level at the point of immission also has a high variability and is not entirely deterministic. In this work, the influence of random input variables due to atmospheric turbulence on the uncertainty of the resulting sound pressure level will be investigated. Using this, the probability of the predicted sound pressure level can be quantified to get a better idea of sound immission under complex atmospheric conditions.
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Organization:
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F. Bertagnolio, DTU Wind Energy |
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In Situ Measured Facade Sound Insulation of Wind Turbine Sound
Pontus Thorsson
[Abstract]
In most countries there are regulations of wind turbine sound level outdoors at dwellings. Often there are also regulations of the sound levels inside the dwelling, however not often directly aiming at wind turbine sound. The sound level indoors from wind turbines has attracted more interest in the latest years, and then especially in the low frequency region (up to 200 Hz). Studies on the in situ sound level difference between outside and inside of dwellings are however scarce. This paper presents the in situ measured sound level difference for two Swedish houses in rural locations, both using a loudspeaker and using the wind turbine sound as exciting signal. This is possible due to a 2 month long measurement series with simultaneous sound recordings outside and inside. The sound pressure level difference from the two methods are shown to differ substantially.
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Prediction of Broadband Noise Generated from Turbulent Boundary Layers of a Horizontal Axis Wind Turbine
Soichi Sasaki, Moe Htet Zaw
[Abstract]
We expand the blade element momentum theory for the prediction of the broadband noise of a horizontal axis wind turbine. For the prediction of the acoustic radiation from the turbulent boundary layers, the prediction theory suggested by B. D. Mugridge is applied. In this prediction methodology, the noise level of the wind turbine can be predicted by the model size of the isolated blade and the main dimensions of the objective wind turbine. At same time, the relative velocity and the attack angle becomes the important parameters. From the results of the wind tunnel test, NACA0018 generates the humped noise in the attached flow condition, whereas the noise spectra in the separated flow condition makes the broadband noise. We point out that the humped noise source in the wind turbine was made from the mid-span to the blade tip on the impeller based on this methodology. In the full length paper, the prediction methodology of the broadband noise, the experimental results of the wind tunnel test of NACA0018, moreover, not only the predicted wind turbine performance but the its broadband noise spectra generated from the turbine with 20m diameter will be presented.
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Organization:
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R.M. Alsina-Pagès, GTM - La Salle (URL) |
J.M. Navarro Ruiz, Universidad Católica San Antonio Murcia. |
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Performance Analysis of the Acoustic Event Detector in the DYNAMAP's Rome suburban area
Rosa Ma Alsina-Pagès, Francesc Alías, Joan Claudi Socoró, Ferran Orga
[Abstract]
Environmental noise is increasing year after year and,
besides annoyance, it causes harmful health effects on
people according to last 2018 WHO report. The
Environmental Noise Directive 2002/49/EC (END) is the main
instrument of the European Union to identify and combat
noise pollution, followed by the CNOSSOS-EU
methodological framework. In order to apply the END
legislation, the EU Member States have to publish noise
maps and action plans every five years. The use of Wireless
Acoustic Sensor Networks (WASNs) changes the paradigm
that addresses the END regulatory requirements as they
enable the dynamic ubiquitous measurement of
environmental noise pollution. Following the END, the
DYNAMAP project develops a WASN-based low-cost noise
mapping system to monitor in real-time the impact of road
infrastructures in two pilot areas: Milan and Rome. To avoid
biasing the noise maps with noise levels unrelated to traffic
noise, an Anomalous Noise Event Detector (ANED) is
included to remove them from the corresponding LAeq. The
paper reflects the adaptation of the ANED algorithm to the
WASN of the suburban area of Rome, which requires a
specific analysis of the particularities of the suburban audio
database, as well as future challenges and research on the
generalization of the WASN.
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The Collation and Use of Data from Continuous Remote Monitoring Systems for the Control of Sound Emissions from a Large Industrial Noise Source
Tim Procter, Stephen Lyons
[Abstract]
This paper shows how the analysis of the sound emissions from
the constantly changing environs of a large open-cut mine
using traditional closed-loop control theory identified the source
of a time lag in the implementation of appropriate control
strategies. As a result, a new control strategy based on ësmart’
technology has been developed to reduce the error signal time
lag and help improve the control of noise emanating from the
mine. Key elements of the new control system discussed in this
paper include: the collation, analysis and reporting of the
continuous real-time noise and meteorological monitoring data;
confirmation of the source’s contribution in a multi-source
environment; and identification of high-risk operational activities
associated with the noise source. The smart technology
presents this information in a format that: enhances the mining
supervisor’s perception of the current environment; improves the
comprehension of the data; reduces the uncertainty associated
with identifying the mine’s contribution to the acoustic
environment; and enables potential future actions and
outcomes to be identified. This paper then shows that
enhancing a user’s perception and awareness of the situation
enables pre-emptive rather than reactive decision-making that
results in reduced noise impacts and improved productivity.
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Estimation of Noise Immission Directivity using Small Microphone Array
Jurij Prezelj, Luka Čurović, Tadej Novaković, Jure Murovec
[Abstract]
Environmental noise monitoring systems extensively use digital
signal processing to perform computation and storage of noise
levels, spectral filtering, narrowband spectral analysis,
evaluation of statistical indices, wave recordings, detection of
noise events based on thresholds, and other similar tasks.
Increase in processing power can be exploited for
development of advanced apparatus, capable of
multichannel processing on hand held devices, as for example
beamforming. A research has been undertaken to upgrade
and implement beamforming for developing three new
measurement options for environmental noise monitoring
systems; 1) automatic exclusion of uncorrelated noise events
from measurements, 2) automatic identification of the
dominant noise source location and 3) automatic classification
of observed noise event. Implementation of these three features
can reduce need for human resources, resulting in reduced
costs and more accurate results. Presented technical solution
provides a synergy between these three options, implemented
in an apparatus, which provides reliable results.
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Travel times in complex environments
Adrien Dagallier, Sylvain Cheinet, Daniel Juvé, Aurélien Ponte, Jonathan Gula
[Abstract]
Times of Arrival (TOAs) of propagated signals are of utmost interest in seismic, underwater as well as aerial acoustics. From TOAs, one may reconstruct the propagation media properties from known sources and sensors positions, or conversely, find sound events locations in a known environment. Modeling the TOAs in realistic environments (wind or current, sound speed gradients, obstacles...) requires a very general physical model, able to factor in the impact of refractive and diffractive processes.
We present an interface-tracking model based on Sethian’s Fast-Marching method for computing TOAs. This method is applicable to complex 3D meshed environments. Examples will be given in an urban area, in the complex atmosphere, as well as in the ocean. The combination of the marching model with the time-matching principle is illustrated to yield a general localization framework applicable to these various environments.
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Organization:
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G. Dutilleux, Norwegian University of Science and Technology |
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Quality assurance methods demonstrated with the calculation of sound propagation with ISO 9613-2 and with CNOSSOS-EU
Wolfgang Probst
[Abstract]
Many methods have been implemented in legally binding rules how sound propagation shall be calculated to support the minimization of environmental noise. With the series of the International Standard ISO 17534 we created a framework to ensure this necessary precision. One of the main pillar of the strategy is to clarify all ambiguous definitions that will cause different results if programmed by different software developers. With the example ISO 9613-2 examples are shown where even existing definitions had to be modified. It can be stated that all calculation methods have their shortcomings that can only be detected if the method is realized in software and then practically applied. The same is true with the harmonized European calculation method - here abbreviated as ”CNOSSOS-EU”. Examples are shown where it was necessary to modify and clarify some definitions before it was possible to create test cases to check the correct implementation of the method. With this work on about 30 Test Cases for ISO 9613-2 and for CNOSSOS-EU we learned a lot about the optimal strategy if a new method shall be implemented or an existing method shall be improved. Some recommendations for future projects of that type will be given.
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Meta-Modeling for urban noise mapping
Antoine Lesieur, Pierre Aumond, Vivien Mallet, Arnaud Can
[Abstract]
Urban noise maps are usually computed by noise propagation models coupled with emission and attenuation rules like CNOSSOS. These models can require several hours to produce a map over a city for a single set of input data. This computational cost makes the models unusable for applications like uncertainty quantification where hundreds of simulations may be required. One solution is to replace the physical model with a metamodel which is extremely fast and yet fairly reproduces the results of the physical model. The strategy is first to reduce the dimension of both inputs and outputs of the physical model, which leads to a reduced model. This reduced model is then replaced by a statistical emulator. The emulator is trained with calls to the reduced model for a set of chosen inputs. The emulator uses linear regression and interpolation between the training output values. We applied this approach to the NoiseModelling software over part of Paris, using either radial basis function or Kriging at the interpolation step. It required a training set of 1000 calls to the physical model. The metamodel is 10000 times faster, and preserves the main behavior of the physical model, with only 1dBA of quadratic error.
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Global sensitivity analysis for urban noise modelling
Pierre Aumond, Arnaud Can, Vivien Mallet, Benoît Gauvreau, Gwenaël Guillaume
[Abstract]
Regulatory noise cartography relies on many input data that can be incomplete, erroneous or inexistent. Thus, operators have to complete/validate/qualify those data by prioritizing some information sources and parameters to the detriment of others. Consequently, the sensitivity of the noise model to such input parameters should be determined to focus on the most influential factors. For this purpose, a global sensitivity analysis was carried out on the open-source software NoiseModelling, which integrates the CNOSSOS regulatory method, using the Morris screening method. As part of the ANR CENSE project (2016-2020), a global sensitivity analysis will be presented on a case study of a 1 km2 area in the city of Lorient in France.
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Uncertainty analysis for the in situ sound power level determination using the substitution method
Spyros Brezas, Volker Wittstock
[Abstract]
Sound power determination at different environments and using different techniques exhibits deviations especially at low frequencies. In previous studies the establishment of traceability has been introduced and the realisation and dissemination of the unit watt in airborne sound has been discussed. For the completion of traceability, the sound power level determination of realistic sources is required. The contribution presents the application of the substitution method for the in situ sound power determination of real sources. The substitution method includes both sound pressure and sound intensity measurements. For collecting experimental data, different measurement methods were applied in various environments. Following the concept of traceability, the sound power level is accompanied by the related uncertainty, which explicitly explains the contributing factors along with their extent of contribution. A comparison is made between the uncertainty levels using sound pressure and sound intensity. The proposed uncertainty covers a wider frequency range compared to the up-to-date standards.
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Organization:
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J. Forssén, Chalmers University of Technology, Architecture and Civil Engineering |
E. Zea, KTH Royal Institute of Technology |
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Future Low-Noise Aircraft Technologies and Procedures – Perception-based Evaluation using Auralised Flyovers
Reto Pieren, Lothar Bertsch, Demian Lauper, Beat Schäffer
[Abstract]
Residents living in the vicinity of airports are exposed to noise from departing and approaching aircraft. Noise may be reduced by introducing novel aircraft technologies and flight procedures. By means of auralisation and listening experiments, such possible future changes can be evaluated by considering human sound perception. In this study, flyovers of different aircraft types and flight procedures were auralised at multiple virtual observer locations, and subsequently evaluated in a psychoacoustic laboratory experiment with respect to short-term noise annoyance. Approaches of an existing reference aircraft, a possible low-noise retrofitted version and a future low-noise design were simulated with standard and tailored procedures. Separate source signals were synthesised for engine broadband, fan tone, airframe broadband, and cavity tone noise. Further, smooth transitions between configurations and operational conditions were modelled to create realistic sounds. To increase plausibility, the propagation simulation considered ground reflection and frequency-dependent amplitude modulation due to propagation through a turbulent atmosphere. The flyover sounds were spatially reproduced by a hemispherical loudspeaker array. The listening experiment revealed significant annoyance reductions for low-noise aircraft types and flight procedures, that maximal benefit is achieved by the combined optimization of aircraft technology and procedure, and that distributed observers need to be considered.
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Auralization of Aircraft Noise in an Urban Environment
Roalt Aalmoes, Maurits Van Der Veen, Henk Lania
[Abstract]
Realistic simulation of aircraft fly-over using auralization and
visualisation techniques help in the research towards more
silent aircraft types and/or operations. Up until recently most
auralizations of aircraft were created in a virtual countryside,
only taking the direct path and ground reflection into
account. In an urban environment, reflections and
absorption of buildings must be considered as well as
different ambient sound levels. This research effort aims at
creating an auralization of a turbofan aircraft flying over an
urban environment. Fly-over recordings were obtained of
aircraft approaching Amsterdam Airport Schiphol. The
spectral content of the aircraft noise is used to synthesize the
source noise of the aircraft. A dynamic 3D model of the
urban environment is constructed using publicly available
city data. This data is combined with recorded video data
to match visual and audio occlusion of the aircraft by
buildings. Reflection surfaces are used to determine
mirrored noise sources, which are spatially added to the 3D
buildings. At the observer the sound from the different paths
is received and presented as binaural (directional) sound
together with a realistic visualisation in Virtual Reality using
the Oculus Rift.
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Physically-Based Auralization of Railway Rolling Noise
Julien Maillard, Abbes Kacem, Nadine Martin, Baldrik Faure
[Abstract]
Railway noise contribute significantly to noise pollution both outside and within cities. In recent years, prediction models have been developed to study exposure levels and evaluate abatement solutions. Going one step further, auralization may provide an effective mean for evaluating perceptually the influence of railway noise to the soundscape near existing or future infrastructures. This paper extends railway noise emission models to propose an auralization approach based on physical parameters. As a first step, the approach focuses on rolling noise radiated by the track and wheels, which represents the dominant noise source over a wide range of speed. The excitation of the wheel/rail system by surface roughness is modeled in the time domain. The rail contribution is synthesized using a set of discrete coherent monopoles, preserving the characteristics of the emitted sound field. The contribution of the wheel uses resonant filters based on its structural response. Finally, the contribution of track sleepers is included following the standard TWINS model. Preliminary validations of the approach compare sound levels with existing models as well as measured data. Results from listening tests evaluating the realism of auralized pass-by noise samples are also presented.
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Recording-based Auralization of Train Noise in a Digital Urban Sound Planning Toolbox
Krister Larsson, Jens Forssén, Peter Lundén, Xuetao Zhang, Andrea Sandberg
[Abstract]
Densification around railway stations has the potential to create more climate-neutral transport and resource-efficient travel, but the noise challenge is a limiting factor for sustainable and effective solutions. Currently, railway traffic increases in Europe and major investments in railway infrastructure are planned in Sweden and other European countries. The idea of this project is to finalise a demonstrator of a digital planning tool that facilitates investment in innovative noise measures on or in connection with railway infrastructure close to stations. The digital toolbox will provide the opportunity, at early planning stage, to virtualize the environment for common train types and traffic situations, to describe the effects of planned noise measures including auralization as well as socioeconomic costs and benefits, impact on property values and climate impact. The auralizations developed in this project are based on binaural and ambisonics recordings, which are modified to simulate several measures including rail dampers and various noise barriers such as thin and tall screens and low-height barriers of different widths with acoustically hard or soft surface materials as well as finite lengths. The paper describes the method of auralization, as well as the co-creation and workshop approaches used for the development of the toolbox.
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A Parametric Method to Synthesize Wind Turbine Sounds
Pontus Thorsson
[Abstract]
The annoyance to wind turbine sound is today well known. There is no doubt that increased sound pressure level causes in general an increased annoyance. However, when it comes to additional descriptors of the wind turbine sound, such as e g equivalent frequency spectrum or amplitude modulation, there is no consensus on what parameters that are most important. In the research project Wind Turbine Noise Effects on Sleep (WiTNES) there was a need for 8 hour long sound signals from wind turbines that were completely free of extraneous sounds. Recordings from reality are difficult to maintain free from disturbances. Instead a parametric synthesis method was developed that is capable of creating wind turbine sounds of arbitrary length. The sounds can be designed in terms of equivalent frequency spectrum, what frequency bands that have amplitude modulation as well as frequency dependent amplitude modulation strength and variation. Recordings of wind turbine sounds at immission distances from different wind turbine types were used to find relevant values for the parameters. This paper describes the synthesis method and some examples of input parameters.
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Analysing the Effectiveness of Approaches to Auralisation for Applications in Environmental Acoustics
Alex Southern, Frank Stevens, Damian Murphy
[Abstract]
Auralisation is becoming more commonplace as a means
of communicating and disseminating plans for changes in
our environmental landscape due to major infrastructure
projects such as road, rail and airport developments,
where noise can be a significant issue. This paper presents
an approach to this challenge in the context of a major
public consultation for a UK road project, and offers
comment on different means for facilitating presentation
of, and engagement with, both audio and visual material.
Although such methods are generally used for information
only, every effort is made to ensure that the final result is
both plausible and comparable to existing on-site
conditions, and calibrated accordingly, with a view to
supporting other more standardised methods and metrics.
However, it is also possible to develop more quantitative
measures as to how effective and successful such
auralisations are in terms of conveying a representation of
a given scene, and any planned for interventions. In this
work the Self Assessment Manikin and soundscape
classification descriptors are used in a series of listening
tests to better understand the approach that has been
adopted for this project.
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Auralisations for Outdoor Noise Sources
Per Finne, Erik Thysell, Christian Weirum Claumarch
[Abstract]
For years auralisations have been used in addition to mandatory
noise maps and calculated number of noise exposed dwellings
in the EAA processes carried out by i.g. the Danish Road
Administration. Auralisations make it possible to get a realistic
impression of the future noise during the planning of new roads
or other important noise sources. Till now, only major road
projects have been published using auralisations. Noise sources
like wind turbines, railways and light rails, can obviously benefit
from auralisations in communicating noise consequences to the
influenced people influenced like neighbours etc. Auralisations makes it possible to aurally explain the mix of noise
combined by two or even more noise sources. Normally, noise
from various noise sources are treated and explained
individually, and they even have different noise limits to meet.
Auralisations are made to represent the total soundscape which
is much more realistic to the noise in the close surroundings as it
will be perceived and experienced by the future neighbours.
This presentation deals with auralisations made for other sources
than road noise and the combined noise generated by various
noise sources. The project is supported by the Danish Agency for
Institutions and Educational Grants.
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Audiovisual simulation inside the residential rooms of roadside buildings
Asakura Takumi, Riku Hashimoto
[Abstract]
This study aims at an audiovisual simulation scheme of
the environment inside such a room inside roadside
residential buildings, which can be applied to the
physical/physiological evaluation of the environments
inside unbuilt buildings. For that purpose, in this study, the
authors performed a basic study for composing a
simulation scheme of both the sound and visual
environment inside rooms. In the study, the sound of the
passing vehicles transmitted into the rooms were
synthesized based on measured dry sources of the pass-by
vehicle sounds, whereas the 3-dimensional visual settings
inside the room were made by using the 3-dimensional
modeling software of Unity. In addition, by using the
method, the subjective evaluation experiment on the
impression of the audiovisual environment inside the virtual
rooms were performed by using the synthesized sound
and visual data. Based on the results of the experiment,
the subjective effects caused by the auditory and visual
stimuli were discussed.
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Traffic Flow Auralisation based on Single Vehicle Pass-by Noise Synthesis
Yang Fu, Damian Murphy, Alex Southern
[Abstract]
Research into the auralisation of road traffic noise has drawn
more attention in the recent years, as it can provide a more
direct and immersive experience with which to communicate
the implication of major urban design decisions with city
planners and the public. Some synthesis models of single vehicle
pass-by noise, and methods for outdoor sound propagation
modelling have been proposed, which may be integrated for
auralisation of traffic flow scenes. In this paper, a framework for
traffic flow auralisation is presented. The synthesis model is
composed of multiple single vehicle pass-by examples, based
on tyre noise and engine noise synthesis. A first-order image
source method is implemented, as a first approximation to
represent sound propagation across a local region of a wider
urban environment. The resulting plausibility of the proposed
traffic flow auralisation model under different traffic flow scenes
is tested using both objective sound quality metrics and
subjective listening tests.
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Progressive region-of-interest filtering for urban sound auralization applications with multiple reflected and diffracted propagation paths
Jonas Stienen, Michael Vorländer
[Abstract]
When auralizing urban environments, two major challenges can be identified for the propagation simulation: the extent of the built environment and the wave-based propagation features, above all diffraction into the shadow region. It is in the nature of open space that sound can propagate long distances. For the auralization of city acoustics without a-priori knowledge on the sound source characteristics huge data must be taken into account. With the increase of freely accessible open format databases providing layers with simplified geometric models of entire landscapes including buildings, a solution to create virtual acoustic environments based on this data is desirable. In this attempt, a progressive broad-band propagation simulation model based on Geometrical Acoustics is proposed that primarily exploits the frequency-independent energy loss subject to geometrical spreading of spherical waves for the spatial filtering of the geometry input. An ellipsoidal hull representing a region of interest is constructed from the source and receiver position as well as sound power and sensitivity, respectively. Subsequently a fast filtering algorithm with linear complexity removes approximately irrelevant geometry for each propagation step releasing the propagation algorithm from executing ray-polygon intersection tests on the entire geometry dataset and speeding up the propagation simulation significantly.
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Sound insulation auralization filters design for outdoor moving sources
Imran Muhammad, Anne Heimes, Michael Vorländer
[Abstract]
Auralization techniques are rapidly attaining popularity,
mostly due to their applications in building acoustics and
virtual reality. As an application, sound insulation auralization
has become a valuable tool to assess the sound insulation of
buildings in terms of investigating how the indoor and
outdoor noise sources are influencing the daily working and
living routine of humans.
The airborne sound insulation prediction standards have
several simplifications that are implicit in the formulation on
which they are based. Therefore, it is necessary to develop a
model to build up an auralization chain of the sound
insulation by taking into account the complex wave fields on
the surfaces of the building elements and the spatial and
temporal variation of the sound pressure field inside arbitrarily
shaped rooms.
We investigate the bending (flexural) wave patterns on the
extended walls of buildings, the directionality of the outdoor
moving sound sources to construct perceptually correct
binaural filters. Additionally, we address the problem of
synthesising the room impulse response from the one-third
octave band values of the reverberation times. Our
proposed approach increases acoustic interaction with the
virtual environments creating a deeper, more realistic and
immersive scene and leading to more accurate perceptual
evaluations of sound insulation.
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Organization:
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K. Attenborough, The Open University |
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A review of natural means for noise abatement
Keith Attenborough
[Abstract]
In addition to wavefront-spreading and air absorption, noise from sources near the ground can be abated during its path to the receiver. ëNatural’ ways of doing this are described. While the reduction in sound levels attributable to ground effect, sometimes called ground absorption, is included in prediction schemes for surface transport noise, it is possible to increase it either by choosing particular types of acoustically-soft ground surface or by deliberately introducing roughness on an acoustically-hard ground. The most effective acoustically-soft grounds are those with low flow resistivity and recent cultivation without compaction. Low parallel walls or lattice arrangements have been shown to be useful forms of roughness. Reflections from façades and barriers can be reduced by vegetative coverings. While hedges offer relatively little noise reduction, 15 m wide belts of trees can be designed to give on the order of 10 dB reduction at a receiver 30 m from a road. Potential noise reductions from combining ënatural’ methods in situations where barriers are not feasible will be explored.
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The influence of organic matter on acoustical properties of soil
Kamrun Suravi, Ho-Chul Shin, Keith Attenborough, Shahram Taherzadeh, Richard Whalley
[Abstract]
To study the effects of organic matter on the physical and
mechanical properties of soils, which may influence the
propagation of sound across cultivated land and acoustic-to-
seismic coupling, soil samples from a long-term agriculture
experiment involving different treatments, have been subjected
to a triaxial test method to measure the normal consolidation
and shear deformation. Also, differential thermal analysis (DTA)
has been used to deduce the organic matter content and its
composition and water release characteristics and soil strength
have been measured with the suction plate method and
indirect tension method, respectively. Data show the main
effect of organic carbon is on the porosity at any consolidation
pressure and that porosity in soil, shear deformed to the critical
state condition, is correlated with internal surface area inferred
from DTA. Indirect tension shows significant variation in the
tensile strength (F= 8.95; P<0.001) of soils from different
management practices. Initial acoustic impedance tube
measurements made on sand samples with various water
content show similar results to those made by Horoshenkov and
Mohamed (2006). Results of further measurements on soils with
different organic matter content will be presented.
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Acoustic absorption of a living green wall - Parametric transducer and XYZ gantry measurement method
Anna Romanova, Kirill Horoshenkov
[Abstract]
This work provides evidence of the importance of considering
urban green structures and in particular the Living Green Walls
(LGW) as sound-scaping instruments. An insight to Living Green
Walls sound absorption properties is provided with particular
interest of sound absorption coefficient and plant morphology.
The work was carried out in-situ in internal and external
environment and makes use of parametric transducer and XYZ
gantry. Novel method of parametric transducer was previously
verified by the use of impedance tube tests bench-marking
against porous media material. The proposed XYZ gantry
method eliminates the reflection from the ground and averages
the discrepancy of the absorbed/reflected sound due to plant
morphology over a given area of interest. The new method is an
alternative to the ISO354-2003 and CEN/TS 1793-5:2016 standard
methods to measure acoustic absorption of materials.
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Sound Propagation in a Forest Based on 3D Multiple Scattering Theories
Vladimir Ostashev, David Keith Wilson, Michael Muhlestein, Keith Attenborough
[Abstract]
This presentation overviews recently developed approaches
for forest acoustics based on three-dimensional (3D) multiple
scattering theories. The mean sound field is calculated with
3D effective wavenumber which represents the scattering by
trunks, branches, and the canopy. The radiative transfer
equation is formulated appropriately to forest acoustics; this
result enables calculations of the mean sound intensities
transmitted and backscattered from a stand of trees. Using a
modified Born approximation, pulse propagation in a forest is
analyzed and compared with results of measurements. In
addition it is demonstrated that the equations for the first two
statistical moments of the sound field in a forest have the
same form as those in a turbulent atmosphere if the scattering
properties of the media are expressed in terms of the cross
sections. This correspondence, and the existing theory of the
interference of the direct and ground reflected waves in a
turbulent atmosphere, enable analysis of a similar effect in a
forest, where these waves are scattered by forest elements.
The results obtained are compared with data.
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Sound Absorption by Tree Bark
Mengmeng Li, Timothy Van Renterghem, Jian Kang, Dick Botteldooren
[Abstract]
Scattering of sound waves by trunks is part of the noise reducing
potential of tree belts, and it has been shown before that the
absorbing properties of the trunks are relevant in this respect.
Information on bark absorption is currently very limited. Therefore,
laboratory experiments were conducted with an impedance tube to
measure the bark sound absorption of various species, including
variation in bark thickness, tree age, bark porosity, density, and
trunk diameter. Preliminary measurements were made to define the
relevant part of the trunk for its acoustic absorption and to come to a
sample handling procedure. From each trunk, 4 cylindrical samples
were were made including 2 centimeters of wood behind the bark.
The measurements show that the absorption (at normal incidence) is
generally below 10 %, and that there is a small variety in between
species.
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Organization:
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B. Xie, South China University of Technology |
P. Majdak, Acoustics Research Institute, Austrian Academy of Sciences |
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Investigating Uncertainties in Fast HRTF Measurements
Shaima'A Doma, Hark Braren, Janina Fels
[Abstract]
In personalized binaural synthesis, the use of individual head-related transfer functions (HRTFs) promises improved auditory immersion and localization performance, compared to generic HRTFs. A direct way of acquiring individual HRTFs is via acoustical measurements. At the Institute of Technical Acoustics of RWTH Aachen University, a measurement arc with 64 loudspeakers has been developed and optimized for this purpose, performing a fast measurement whilst rotating continuously around the subject. Optical tracking allows an investigation into positional uncertainties of the arc, as well as into subject movement. Previous evaluation approaches of the measurement system are expanded in this study. A series of human HRTF measurements is conducted, together with 3D structured light head scans of the subjects, which can be used for Boundary Element Method (BEM) simulation of the HRTF. By means of objective distance measures, differences between HRTFs, as well as repetition errors are quantified and measurement accuracy is assessed.
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Independent modes and dimensionality reduction of head- related transfer functions based on tensor decomposition
Tong Zhao, Bosun Xie
[Abstract]
Head-related transfer functions (HRTFs) are acoustical
transfer functions from a source to two ears in the free-field
and vital to spatial audio. Generally, far-field HRTFs vary
with source direction, frequency and subject. Accordingly,
the dimensionality of a full set of HRTFs is very large. Linear
decomposition has been used to reduce the dimensionality
of HRTFs. Due to the multiple variable-dependent
characteristics of HRTFs, in present work, tensor
decomposition is applied to analyze the independent
modes of HRTF variation so as to achieve a more efficient
representation of them. The measured HRTF magnitudes are
first smoothed with a frequency bandwidth of 1 ERB. Then
the smoothed HRTF magnitudes of 52 Chinese subject, 493
directions for each subjects and 42 discrete frequencies (up
to 14 kHz) are combined as a 3rd-order tensor and then
undergone a Tucker decomposition. The results indicate
that 8 subject-related modes, 13 directional-related modes
and 6 frequency-related modes account for more than 95
% variations of HRTFs at corresponding dimension
respectively, and thus the HRTF dataset can be reduced by
truncating the modes to corresponding orders. Based on
the results, it is also deduced that 8 anthropometric
parameters at least are needed for the anthropometry-
based HRTF customization.
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Objective Differences between Individual HRTF Datasets of Children and Adults
Hark Braren, Janina Fels
[Abstract]
Subjective evaluations of noise by children and its effect
on their learning abilities in classrooms are a current
research topic. Studies in this field often make use of virtual
acoustic scenarios to be able to present a controlled
acoustic environment in which to conduct the
experiments. The presentations of realistic acoustic scenes
requires a set of Head-Related Transfer Functions (HRTFs)
that in the best case is the subject’s individual HRTF, or
matches it as closely as possible. However in most cases
only a generic HRTF from an artificial head is available. This
leads to perceptive deviations in location, coloration and
reduces the immersion in the presented scene.
Previous studies have shown, that anthropometric
parameters, and by inference HRTFs differ especially for
younger children compared to adults. In this study,
individual high-resolution HRTF datasets as well as head
and pinna geometries of 1st and 2nd grade children (ages
6-9 years) have been measured using the fast
measurement system at the Dep. of Medical Acoustics
(Institute of Technical Acoustics, RWTH Aachen University)
and a structured light 3D Scanner. The HRTFs and
anthropometric measurements are compared to those
from adults and commonly available artificial heads to
objectively evaluate their similarities and differences.
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Rendering virtual source at various distances using binaural Ambisonics scheme in dynamic virtual auditory display
Bosun Xie, Jianliang Jiang
[Abstract]
In dynamic virtual auditory display, free-field virtual source at
various directions and distances is conventionally realized by
filtering the input stimulus with near-field HRTFs. However,
this conventional method meets with some problems in
practical use, due to the difficulty in acquiring near-field
HRTFs and requiring a large memory to storage them. In
present work, a scheme for rendering virtual source at
various distances using binaural near-field compensated
Ambisonics with appropriate order is proposed. Based on the
principle of spherical Bessel and harmonics decomposition of
sound field, the directional and distance information of sound
field is first encoded into Ambisonics signals and then
converted to headphone signals by dynamic binaural
synthesis using far-field HRTFs only. It is proved that the 5th
order dynamic binaural Ambisonics with 36-64 pairs of HRTF-
based filters is able to recreate desired binaural pressures for
various source directions and distances up to 2 - 3 kHz,
covering the frequency range for ITD (and its dynamic
variation) as well as low-frequency ILD as directional and
distance localization cues, respectively. A psychoacoustic
experiment validates that the proposed scheme is able to
recreate appropriate distances localization perception outside
the median plane.
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Local Ambisonics panning method for creating virtual source in the median plane
Kailing Yi, Bosun Xie
[Abstract]
With the development of multichannel sound with height, it is desired to create virtual source in different vertical directions, especially in the frontal median plane. For this purpose, a local Ambisonics panning method is proposed. In a case of analysis, three loudspeakers are arranged in the frontal median plane, with one locating in the horizontal plane and other two locating 30∘-45∘ above or below the horizontal plane, respectively. The signal amplitudes of loudspeakers are a linear combination of the zero order and first order harmonics of elevation angle. It is proved that, by properly choosing the weights of linear combination for each loudspeaker signals, the current method is able to create appropriate interaural time difference as well as its dynamic variation caused by head rotation and head tilting. Therefore, based on classical Wallach’s hypothesis and its modern validations, the current method is able to create appropriate dynamic cue for vertical localization in the median plane at low frequency. Results of psychoacoustic experiment indicate that the current method yield reasonable localization performance and is superior to traditional pair-wise amplitude panning. The method can also be extended to create virtual source in other vertical planes.
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The limitation of static transaural reproduction with two frontal loudspeakers
Lulu Liu, Bosun Xie
[Abstract]
Incorporated HRTF-based binaural synthesis and cross-talk
cancellation, transaural reproduction with two frontal
loudspeakers is theoretically able to duplicate the binaural
pressures of an actual sound source, and then is expected to be
able to recreate virtual source at arbitrary direction. However,
many experiments have indicated that for static transaural
reproduction, in which transaural processing is fixed regardless
of head turning, the perceived virtual source is usually limited in
the frontal-horizontal plane. The reason may be that, in static
transaural reproduction, the dynamic cue for front-back and
vertical localization is omitted, while the high-frequency spectral
cue is unstable against head movement. To validate this
hypothesis, a psychoacoustic experiment is conducted in
present work. A dynamic transaural system with two frontal
loudspeakers is used in the experiment. According to the
contemporary orientation of head detected by a head-tracker,
the system changes the transaural processing so that binaural
pressures in reproduction follow the turning of subject’s head.
The experimental results indicate that, by incorporating the
dynamic cue caused by head turning, dynamic tranaural
reproduction is able to recreate virtual source at front, back
and vertical directions. Therefore, the limitation in conventional
static transaural reproduction is indeed caused by the omitting
of dynamic cue.
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Effect of individualized near-field head-related transfer functions on distance perception in dynamic virtual auditory display
Guangzheng Yu, Liliang Wang, Liang Linda, Bosun Xie, Jianliang Jiang, Chengyun Zhang
[Abstract]
Virtual auditory displays (VAD) aim to create virtual source at various directions and distances in headphone reproduction by using HRTF-based binaural synthesis. Incorporating the variation of binaural signals caused by head turning in three degrees of freedom, a dynamic VAD reduce front-back and up down confusion, and improve the externalization of perceived virtual sources. It is well known that individualized far-field HRTFs also improve the direction localization of perceived source. In present work, a virtual source localization experiment is conducted to further examine the effect of individualized near-field HRTFs on direction and distance perception in dynamic virtual auditory display. The results indicate that dynamic binaural synthesis with both individualized and non-individualized near-field HRTFs is able to create perceived virtual source at various target distance within 1.0 m outside the median plane, especially at the lateral directions. Individualized HRTFs influence little to distance perception, but improve accuracy of directional perception in terms of elevation errors.
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3-D Sound Image Localization in Reproduction of 22.2 Multichannel Audio Based on Room Impulse Response Generation with Vector Composition
Kaige Zheng, Misaki Otsuka, Takanobu Nishiura
[Abstract]
To simplify content creation processes and improve the realism
of 22.2 multichannel system, a method to localize sound images
is derived in this paper, which is an improvement of the method
based on vector base amplitude panning. The sound image
localization can be realized by a specific direction and
distance. A sound field is simulated to make spatial impressions
in 22.2 multichannel reproduction, which includes reflection and
reverberation, and the simulation approach is based on room
impulse response generation with vector composition. To control
the sound pressure on the ears of the listener, the amplitude of
the input signal is attenuated in advance by the distance of
sound image. Evaluation experiments were carried out both
subjectively and objectively with binaural recording. The
improvement in the reproduction of sound image’s distance is
realized, while the direction of the sound image stays the same
as the method based on vector base amplitude panning.
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Evaluation of Moving Sound Image Localization for Reproduction of 22.2 Multichannel Audio Using Up-mix Algorithm
Hiromu Suzuki, Takanobu Nishiura
[Abstract]
22.2 multichannel audio had been developed for ultra-high
definition televisions. It can reproduce a three-dimensional (3-D)
sound field by simultaneously driving 22 loudspeakers and two
woofers. However, it is difficult to easily acquire multichannel
signals because of equipment costs and so on. The up-mix
algorithm for generating multichannel signals from the existing
fewer channel signals has been proposed to solve this problem,
Whereas, the conventional up-mix algorithm has been
proposed without the 3-D sound source location. As a result, it
cannot reproduce the moving sound image in vertical and
back-ward directions. In this paper, we propose a novel up-mix
algorithm which extends two channel signals to 22.2
multichannel signals while using the 3-D location of the sound
source. Both objective and subjective evaluations were
performed to comparing with the accuracy of reproduced
sound image localization. Results of the evaluations show that
the proposed up-mix algorithm can reproduce 3-D moving
sound images more accurately than the conventional one.
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[Abstract]
Depending on the spatial audio reproduction method, surrounding spherical loudspeaker arrays enable either physically correct sound field synthesis in a physically constrained sweet spot or rather aim at perceptually correct reproduction. Feasible methods include modal-based approaches such as higher-order Ambisonics with a continuous virtual panning function, facilitating phantom sources with panning-invariant energy and spread by use of sub-band weighting for side lobe suppression and a suitable decoder strategy for the given loudspeaker layout. Instead, multiple-direction amplitude panning is based on three-dimensional amplitude panning with discrete loudspeaker gains and vectors, also aiming at constant spread of phantom sources with reduced coloration, but without claiming physical correctness. Representing a fundamentally different approach, acoustic crosstalk cancellation relies on the concept of binaural technology and enables the creation of physically correct signals at the entrance of the ear canals. Although these methods have been well researched individually, there has been only little comparative measurement-based evaluation to validate simulations considering practical imperfections. Such a comparison is presented based on various error metrics.
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Digital Cinema: Analysis of Multi-Channel Systems to Control Sound Levels
Peter Houtave
[Abstract]
Since 2014, Flemish Authorities have installed an auto-regulatory system for cinemas in order to control the sound pressure levels during screening of different digital source materials. In preparation of this regulation, an acoustic study was necessary to fully understand the existing sound pressure levels and how they are generated in different movie theatres. Therefore, the whole sound system was studied in extensive details to describe the subsequent steps in the motion-picture multi-channel sound reproduction chains. The analysis started at the levels [dBFS] of the DCP 5.1 multichannel digital source signals, passing through the signal processing and amplification, integrating the electro-acoustic and room responses at different listening positions. As a result, it became clear that in modern digital cinemas with proper room calibration, a good prediction of the resulting sound pressure levels can be achieved. This result provides therefore a mean of control of the resulting sound pressure levels even before actual playback. Different sources [film, trailer, commercial announcement], different sizes of venues and different audience positions were investigated. Results of the different analysis steps will be presented and discussed.
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Estimation of Cross-Talk Compensation Filter using Bone Conduction Ear Microphone
Irwansyah Irwansyah, Tsuyoshi Usagawa
[Abstract]
Placing a bone conduction (BC) transducer anywhere on the head will result in a BC sound reaching the cochlea in both ears, unlike air conduction. This “cross-talk” limits a listener’s ability with a pair of BC transducer to sense sound direction. In this paper, we discuss a way to minimize cross-talks in BC sound reproduction using a method called “cross-talk cancellation.” Ideally, the method requires transfer functions (TFs) from each of the BC transducers to each of the listener’s cochleae to accurately synthesize a cross-talk compensation (CTC) filter; however, a direct measurement of the TFs at the cochlea is not possible. Since the closest to the cochlea is the ear canal, we thus hypothesize that the TFs measured at each ear canal may be used to achieve cross-talk cancellation at the cochlea in the inner ear. Therefore we utilized a BC ear microphone to capture the vibration of the bony ear-canal caused by vibrating transducers on the mastoid. The filtered-x least mean square (FxLMS) algorithm was used to estimate the CTC filter. Experiments with and without cancellation were performed to determine the effective frequency range that could achieve cancellation in both the ear canal and inner ear perceived by participants.
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Primary ambient extraction for random sign Hilbert filtering decorrelation
Lu Chen, Chuang Shi, Huiyong Li
[Abstract]
The primary ambient extraction (PAE) is a crucial step in spatial audio analysis and reproduction. The PAE decomposes a stereo mixture into separated primary and ambient components based on different spatial assumptions. The principal component analysis (PCA), although simple to calculate, often suffers from severe extraction error, particularly when the stereo mixture contains relatively strong ambient components. Reviewing the derivation of the PCA, it is found that the unbalanced ambient estimates incurred in the PCA degrades the extraction performance significantly. Therefore, this paper attempts to modify the classic PCA to enable the adjustable ambient components catering for various sound scenes. Comparing with the approximate solution to the ambient spectrum estimation, the proposed method is equivalently complex to implement and leads to higher extraction accuracy in selected sound scenes.
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Analysis of Head Rotation Trajectories During a Sound Localization Task
Yukio Iwaya, Brian F. G. Katz
[Abstract]
Dynamic changes of the perceived Head-Related Transfer
Function as a function of head movement have been shown
to be an important cue in sound localization. To investigate
the cognitive process of dynamic sound localization,
quantification of the characteristics of head movement is
needed. In this paper, trajectories of head rotation in a sound
localization task were measured and analyzed. Listeners
were asked to orient themselves towards the direction of
active sound source via localization, being one of five
loudspeakers located at 30∘ intervals in the horizontal plane.
A 1 sec pink noise burst stimulus was emitted from different
speakers in random order. The range of expected head
rotations (EHR) for a given stimulus were, therefore, from
30∘ to 120∘. Head orientation was measured with a motion
capture system (yaw, pitch, and roll). Analysis examined
angular velocity, overshoot, and reaction time (RT). Results
show that angular velocity increased as EHR increased. No
relationship between overshoot and EHR was observed. RT
was almost constant (about 260 ms) regardless of EHR. This
may suggest that dynamic sound localization studies could
be difficult for a short stimulus with duration less than 260
ms.
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Real time audio encoding/decoding system using MPEG-H 3D Audio toward advancement of terrestrial broadcasting technology
Takehiro Sugimoto, Shuichi Aoki, Satoshi Oode, Tomomi Hasegawa, Hiroki Kubo, Hiroyuki Okubo
[Abstract]
A real time audio encoding/decoding system using MPEG-H 3D Audio (3DA) was developed. Research and development for advancement of terrestrial broadcasting technology has been carried out in Japan. The authors are engaged in improvement of audio scheme in broadcasting system, specifically on highly efficient audio coding, audio object handling by metadata, and rendering feature adaptive to preference of listeners and playback environment. Adopted coding technology is MPEG-H 3DA standardized in 2015. Coding efficiency of MPEG-H 3DA is twice as high as conventional AAC in 22.2 multichannel (22.2 ch) sound transmission. The developed system supports up to 24-channel beds with 4 objects and can render adaptively to typical playback environments, e.g., 22.2 ch, 7.1.4 ch, 5.1.2 ch, 5.1 surround and stereo. The encoder accepts and interprets audio objects handled with a serial representation of the Audio Definition Model and dialogue enhancement and dialogue replacement are realized using audio object. Transmission experiment with 8K video is also reported.
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Machine-learning-based estimation of reverberation time using room geometry for room effect rendering
Ricardo Falcon Perez, Georg Götz, Ville Pulkki
[Abstract]
The reverberation of sound inside rooms has a significant
impact on sound quality. In virtual reality, the reverberation
of rooms should be reproduced with sufficiently high
accuracy to produce plausible rendering of room acoustics.
One of the most important parameters when rendering
reverberation artificially is the reverberation time, and the
value for it should be estimated from the geometry of the
virtual room, for any given source/receiver position and
orientation. While complete physical models are typically too
computationally costly, an approximation can be made using
simplified mathematical models such as Sabine of Eyring
formulas, at the expense of reduced accuracy. This work
shows a proof of concept for a machine learning based
method that predicts the reverberation time of a room, using
geometric information and absorption coefficients as inputs.
The proposed model is trained and evaluated using a novel
dataset composed of real-world acoustical measurements of
a single room with 832 different configurations of furniture
and absorptive materials, for multiple loudspeaker positions.
The method achieves a prediction accuracy of approximately
90 % for most frequency bands. Furthermore, when
comparing against the Sabine and Eyring methods, the
proposed approach exhibits a much higher accuracy,
especially at low frequencies.
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The Perceptual Effect of Reflective Surfaces on Acoustic Crosstalk Cancelation Using an 8-Channel Linear Loudspeaker Array
Karim Bahri, Jens Ahrens
[Abstract]
We recently presented a user-tracked crosstalk cancelation system based on a linear array of loudspeakers. The system uses Ambiophonics in the low frequency range and a superdirective nearfield beamformer in the mid and higher frequency range. Efficient real-time execution is achieved by parameterization of pre-computed beamformer weights. The perceptual evaluation showed a considerable difference in the perception of binaural content depending on whether the system was operating in a free field or in a room with reflecting surfaces. In this contribution, we investigate the perceptual effect of different reflecting surfaces that are apparent in the reproduction room. Most notably, we separately investigate the effect of the floor reflection, which fuses with the direct sound from the loudspeakers, and the effect of the wall reflections, which arrive at later time instants.
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Organization:
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J. Li |
Q. Meng, Acoustics Lab, School of Physics, South China University of Technology |
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Acoustically transparent sound presentation in hearing devices: algorithms, devices and models
Simon Doclo, Henning Schepker, Florian Denk, Reinhild Roden, Matthias Blau, Birger Kollmeier
[Abstract]
Assistive hearing devices often have a low user acceptance due to a limited sound quality. Recently, the concept of acoustic transparency was proposed to increase the sound quality for hearing-impaired and normal-hearing users. Acoustic transparency of a hearing device refers to the acoustic perception of an open ear canal while the device is inserted in the ear. This can be achieved by applying an equalization filter to the output of the device such that the transmission properties of the open ear are obtained. In this contribution we give an overview of our approach to acoustically transparent sound presentation. We present recent advances in our custom earpiece design with multiple receivers and microphones as well as signal processing algorithms for robust acoustic feedback suppression and sound pressure equalization. We present results from subjective evaluations of a real-time demonstrator in terms of the overall quality compared to the open ear and outline future research directions, e.g., using individualized electro-acoustic models to alleviate the necessity of measuring the acoustic transfer functions within the device and to the eardrum. Funded by the Deutsche Forschungsgemeinschaft (DFG) - Projektnummer 352015383 - SFB 1330 A4 and C1.
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Cochlear-Implanted Children's Perception of Mandarin Tones in Normal Speech and Whispered Speech
Xin Ding, Wentao Gu
[Abstract]
Since cochlear implants lack an efficient coding of fine structures of speech including F0, it is generally difficult for cochlear-implanted (CI) children to perceive linguistic tones, of which F0 is the primary acoustic cue. In whispered speech, however, F0 is missing and no longer serves as an acoustic cue of tones. Using a picture selection task, we compared the identification of Mandarin four tones in isolated syllables between two groups of 4-to-5-year-old children, i.e., CI and normal-hearing (NH) groups, in both normal speech and whispered speech. For normal speech, NH showed significantly higher identification than CI, regardless of tone which showed no significant effect. For whispered speech, both groups identified T3 better than other three tones, possibly due to their use of other cues for T3; Also, no significant difference was found between the two groups on all tones but T3, on which NH showed significantly higher identification than CI, suggesting that NH retrieved the voice quality cue for T3 better than CI. In both groups, identification rate was significantly lower for whispered speech than for normal speech, but the decrease in the rate was greater in NH than in CI, confirming that NH’s tone perception relied more on F0.
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How the temporal amplitude envelope of speech contributes to urgency perception
Masashi Unoki, Miho Kawamura, Maori Kobayashi, Shunsuke Kidani, Masato Akagi
[Abstract]
Speech communicates non-linguistic and para-linguistic
information as well as linguistic information. Our previous
studies on noise-vocoded speech (NVS) showed that
temporal modulation cues provided by the temporal
amplitude envelope (TAE) affect how vocal emotion and
speaker individuality are perceived. However, it is still
unclear if temporal modulation cues affect the perception
of urgency. Here, we experimentally investigated if the TAE
of speech affects the perception of para-linguistic
information, particularly urgency. We compared NVS in
which the TAEs were identical to those of the original
speech, and NVS in which the TAEs had undergone low-pass
or high-pass filtering. Urgent scales were derived from a
paired comparison of the results and used to investigate the
relationship between the temporal modulation components
and urgency perception. Our findings were: (1) the degree
of urgency of the NVS stimuli was perceived as being similar
to that of the original; (2) temporal modulation components
of NVS upwards of 6 Hz were significant cues for urgency
perception; (3) temporal modulation components of NVS
downwards of 8 Hz were significant cues for urgency
perception. The results suggest that temporal modulation
cues in the TAE play an important role in urgency
perception.
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Restoring Lost Speech Components with Generative Adversarial Networks for Speech Communications in Adverse Conditions
Nengheng Zheng, Yupeng Shi, Yuyong Kang, Qinglin Meng
[Abstract]
Speech enhancement has been widely implemented to
restore the speech quality for speech communication
between humans or human and machine. For different
speech communication scenarios and
channel/environmental conditions, the types and degrees
of speech distortion could vary significantly and many
speech enhancement strategies have been developed
accordingly. This study deal with a severe distortion problem,
i.e., part of the spectral and/or temporal components of the
speech are completely lost. The spectral loss is simulated by
a transmission channel with very narrow passing bandwidth
(lower than 2 kHz) which results in severely degraded
speech quality; the temporal loss is simulated by packet loss
up to 20% percent in massive communication which results
in poor speech intelligibility. A generative adversarial
networks (GAN) based speech enhancement scheme is
proposed for restoring the missing spectral and temporal
components with different network structure and
parameters. A set of experiments have been conducted to
evaluate the effectiveness of proposed enhancement
scheme and promising results have achieved.
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Vocal Emotion Recognition in Mandarin-Speaking Cochlear Implanted Children
Haitao Guan, Wentao Gu
[Abstract]
Previous studies have shown that cochlear-implanted (CI) children have big difficulty in vocal emotion recognition in comparison to the normal-hearing peers, mainly due to the lack of efficient coding of fundamental frequency (F0) in the CI devices, but few research has been conducted on preschool CI children, and even fewer on CI children speaking Mandarin, a tone language in which F0 conveys both lexical tone and emotion related intonation. The present study compared vocal emotion recognition between two groups of 4-to-5-year-old Mandarin-speaking children, i.e., CI and normal-hearing (NH) groups. Six semantically neutral sentences were recorded by a female speaker in three basic emotions, i.e., anger, joy, and sadness, among which anger and joy showed greater mean F0, F0 variability and F0 range than sadness. The children were asked to judge the type of emotion from the recorded utterances. For all three emotions, the CI group showed significantly lower recognition accuracy and longer reaction time than the NH group. The confusion mainly occurred between anger and joy in the NH group, while all three emotions were mutually confused in the CI group, confirming that Mandarin-speaking CI children were impaired in perceiving vocal emotions due to the lack of F0 processing ability.
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Intelligent background sound event detection and classification based on WOLA spectral analysis in hearing devices
Feifan Lai, Kaibao Nie
[Abstract]
Audio signals of real-life hearing devices typically contain
background noises. We aim to build a system model which can
automatically separate background noise from noisy speech,
and then classify background sound into predefined event
categories. In this paper, we propose to use weighted overlap-
add algorithm (WOLA) and deep neural networks(DNN) for
sound event detection, from recordings of 11 realistic
background noise environments(cafe, station, ...) combined
with human speech at different SNR levels. In our approach, a
sound trough detection algorithm is used to retrieve important
background sound information. We use WOLA as an advanced
algorithm which extracts spectral features by transforming a tiny
fraction of time domain signal into frequency domain data
represented in 22 channels. Moreover, a feed-forward neural
network with one hidden layer and one output layer is used to
efficiently recognize diverse event feature patterns, and then
produce classification decisions based on confidence values.
The overall detection accuracy has achieved 95%, while the
event ëhallway’ has the lowest detection rate at 85%. This
detection algorithm has the potential for improving noise
reduction in hearing devices by applying distinct gain
compensation in frequency channels for each particular
detected event.
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Maximum Expanded Measurement Uncertainty: Hearing Aids
Zemar Martins Defilippo Soares, Isabella Florêncio Cruz Da Silva, Nelson Mello Do Espírito-Santo
[Abstract]
For the study of measurement uncertainty estimation it is necessary to understand the model used to perform the measurements. In acoustic performance measurements of Hearing Aids the IEC 60118 part zero and part 7 present the method used for the measurements achievements. However, these standards present two possible acoustic environments where acoustic performance measurements of Hearing Aids can be achieved: Anechoic Chamber and Test Box. This work will estimate the maximum measurement uncertainty in acoustic performance measurements of Hearing Aids in Anechoic Chamber and Test Box. For the estimation of the maximum measurement uncertainty will be studied the main sources of uncertainties that lead to the greatest impact in the model used. A strong emphasis will be given to the source of uncertainty regarding the sound field where Hearing Aids are positioned for acoustic performance measurements. This investigation is based on results of experimental measurements. This work shows the differences found in the estimation of maximum measurement uncertainty in the Anechoic Chamber and in the Test Box.
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A pilot study of the relationship between mandarin chinese word and sentence recognition for the elderly
Jianxin Peng, Jiazhong Zeng, Jiamin Liu
[Abstract]
In present study, the simulated binaural room impulse responses
different reverberation time, were convoluted with the test word
and sentence signals, and noise signal and reproduced by
headphone. The subjective word and sentence recognition
scores for elderly under different signal-to-noise ratio and
reverberation time were obtained. The results show that the
shorter the reverberation time, the higher the scores of word
recognition and sentence recognition for the elderly. The higher
the signal-to-noise ratio, the higher the scores of word
recognition and sentence recognition. There is high correlation
between the scores of word recognition and sentence
recognition for the elderly.
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Organization:
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M. Melon, LAUM / Le Mans Université |
G. Behler, RWTH Aachen University |
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Evaluation of Non-linear Distortion in Compression Chamber of a Low Frequency Horn
Tomasz Nowak, Andrzej Dobrucki, Romuald Bolejko
[Abstract]
The subject of the paper is study of non-linear behaviour
of air in horn-loaded low frequency speaker devices.
Research and development of low frequency transducers
lead to improvement their linearity of excursion trough
linearization of suspension compliance and BL-factor as
well as steadily increasing thermal power handling. The
use of those latest transducers promises to achieve
unprecedented efficiency and sound pressure levels,
especially when combined with high compression horn-
loading. The limiting factor in this case is nonlinearity of the
air, which is particularly high in compression chamber. The
volume of this chamber changes during operation of
loudspeaker even by several dozen percent. This paper
gives an insight on the influence of nonlinear compliance
on nonlinear properties of loudspeaker through analyse
and measurement of a prototype horn-loaded low-
frequency loudspeaker device. As the driver an ultra-
linear woofer has been applied. To evaluate the
participation of the horn and compression chamber in
total nonlinearity of the device, the measurement of the
device with and without horn-loading have been done.
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Modelling of a Compression Driver using Lumped Elements
Joerg Panzer
[Abstract]
Modelling linear response functions of a typical electro-dynamic compression driver with the help of lumped elements should be an easy task. However, it turns out that acoustic channels and slits inside such as device can go into strong coupling with the mechanical fundamental resonance. This paper demonstrates an extended electro-mechano-acoustical network and compares its response functions to measurement.
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Acoustic simulation for high intensity sound source with Helmholtz resonator
Zinan Li, Bo Zhang, Qiqi Chen, Liheng Wang, Yutian Bai
[Abstract]
To study acoustic radiation characteristics of high intensity
sound source used for the acoustic test of sound absorbing
materials, a simulation model in the combination of horn-type
loudspeaker and Helmholtz resonator, which are conceived for
the design of a high intensity sound source in acoustic
impedance tube, is developed by using software COMSOL. Fully
coupled modeling related to multiple physical fields such as
electromagnetic, mechanical, and acoustic ones is
considered in this work, letting voltage be the input and sound
pressure the output. Without the help of any simplified lumped
parameter and only through the acoustic model constructed in
COMSOL and accurate material parameters, the signal
frequency response characteristics, output sound pressure, and
electroacoustic transformation efficiency of the high intensity
sound source can be obtained. The results show that the
acoustic radiation characteristic of the high intensity sound
source depends the interactions between horn-type
loudspeaker and Helmholtz resonator, and its equivalent
acoustic impedance characteristics is mainly dominated by
Helmholtz resonator. It is also shown from results that the
resonances of both horn-type loudspeaker and Helmholtz
resonator may not only strengthen sound radiation of sound
source, but enhance its electroacoustic transformation
efficiency observably.
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Measurement of Loudspeaker Parameters: A Pedagogical Approach
Antonin Novak
[Abstract]
Traditional ways of measuring linear parameters of an electrodynamic
loudspeaker, i.e., fitting the measured signals of voltage, current, and
velocity to a given model, are not very intuitive nor pedagogical. The
method proposed in this paper is based on an educational approach
that in the first step estimates the force factor and separates the
electrical and mechanical impedance. In the second step, both
impedances can be studied separately to highlight the importance of
using models incorporating eddy currents on the electrical side and
creep effect on the mechanical side.
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Effective Radiation Area (Sd) for Axisymmetric Loudspeakers Radiating in an Infinite Baffle Using a Near Field Analysis
Luis Angelo Velarde Criado, Jorge N. Moreno
[Abstract]
In this paper, the effective radiation area (Sd) is obtained
(theoretically and experimentally) for different loudspeakers. For
this purpose, is considered a near field approach. This method is
theoretically based in previous works that use the geometrical
properties of different axisymmetric pistons and calculate the
effective radiation area by positioning the pistons in an infinite
baffle. In the case of loudspeakers, the calculations are
redefined with the new geometrical conditions, in order to
increase the precision of the results. Finally, these results are
compared with the listed effective radiation area of the
loudspeakers data sheets, and other techniques, in order to set
the accuracy of the measurement.
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Measurement of the Back Loading of a Loudspeaker Mounted on a Closed Enclosure – A First Approach
Jorge N. Moreno, Richard Rivera, Luis Angelo Velarde Criado
[Abstract]
In this paper, a simple way to measure the parameters related
to the back loading of a loudspeaker mounted on a closed
enclosure is presented. If the electro-mechano-acoustical
parameters of the loudspeaker in free air are known or can be
measure, then by measuring one more time the parameters of
the loudspeaker mounted on a closed enclosure it is possible to
calculate the back loading parameters due to the enclosure,
including the filling material. This method could also be used to
measure the effect of the filling material in the back cavity of
midrange units, tweeters and compression drivers.
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On the relevance of transducer measurements for real-world applications
Martin Schneider
[Abstract]
Transducers are typically measured in laboratory environments like anechoic or reverberation chambers. The developer then can design his device to pre-determined demands. These are often linear frequency response, measured with plane or spherical waves in the free and/or diffuse field, and constant directivity. But demands may also diverge from these basic ideals, depending on the transducer’s intended application.
In practice however, transducers are rarely used in such idealized environments. Transducers may have to be mounted inside a housing, held by clamps and stands, placed near reflective surfaces or bodies, and may even be hand-held. All these factors influence the transducer’s response.
For the developer it is thus essential to measure the transducer in a variety of setups. These may include a set of near and far field measurements, with and without acoustically obstructive items, and may also bring up the necessity to simulate a human person and its body parts as acoustical objects.
Integrating the results of such a set of measurements will hopefully confirm the transducers suitability for the intended application, but it may also lead to modified demands on the transducer design.
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Comparison of numerical methods for miniature loudspeaker modelling
Juha Backman
[Abstract]
The small dimensions of the loudspeakers and earpieces found in portable audio devices pose challenges for the acoustical modelling. The millimetre scale dimensions imply that the sound propagation occurs to a significant extent within the viscous and thermal boundary layers, so viscothermal modelling is needed. The design question is further complicated by the geometric nonlinearity due to transducer diaphragm movement, and sound pressures inside the transducer that approach the levels that usually call for nonlinear acoustics models. The high viscosity forces can also lead to laminar vortex formation, which would then imply that the mo
The paper studies an example of an axially symmetrical miniature transducer using different finite element approaches:
- Small-displacement linear viscothermal
- linear viscothermal with geometric nonlinearity due to the moving boundaries
- nonlinear viscothermal with moving boundaries and nonlinear acoustics
- compressible flow CFC with moving boundaries
The results suggest that taking geometric nonlinearity into account is quite necessary for transducer modelling, while the typical nonlinear acoustics of pressure-dependent air compressibility are less significant. The CFD models indicate that vortex formation is a likely process in miniature transducer, but due to computational complexity the applicability of CFD models in everyday design is still limited
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Electrostatic All-Silicon MEMS Speakers for In-Ear Audio Applications - Acoustic Measurements and Modelling Approach
Lutz Ehrig, Hermann Schenk, Franziska Wall, Anton Melnikov, Michael Stolz, Sergiu Langa, Bert Kaiser, Holger Conrad
[Abstract]
This paper presents the acoustic properties of a novel electroacoustic MEMS transducer. The transducer is based on electrostatic bending actuators with lateral deflection, i.e. in the chip plane. A large number of such bending actuators is arranged in pairs and move in phase opposition. By means of an appropriate sound guidance, areas of opposite phase pressure changes are separated, so that the acoustic short circuit is avoided and audible sound is generated by the system. In this paper, acoustic measurements recorded with a standard coupler are discussed and compared with the results of model calculations. The main focus is on system-specific features that provide important insights for the enhancement of future designs in terms of sound pressure level and distortion.
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Prediction of acoustic emission of a rigid electrodes DEAP loudspeaker
Emiliano Rustighi, William Kaal, Sven Herold
[Abstract]
Dielectric Electro-Active Polymers (DEAPs) are lightweight
materials whose dimensions change significantly when
subjected to electric stimulation. One form of DEAP construction consists of a thin layer of dielectric sandwiched between two perforated rigid electrodes. They can be used as an actuator or a sensor and have the potential to be an effective replacement for many conventional transducers. This paper refers to their use
as loudspeakers. To date, flat DEAP loudspeakers have been prototyped and tested but no numerical prediction of their acoustic performance has been presented. In this paper an elemental model is presented. The electro-dynamic behaviour of the electrodes and dielectric layers is taken into account. The acoustic impedance is calculated assuming baffled conditions. The impedances of the individual layers are stacked together and preliminary results are shown.
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Sensorless Measurement of the Acoustic Impedance of a Loudspeaker
Romain Boulandet
[Abstract]
This paper presents an experimental approach for estimating the load
acoustic impedance of a loudspeaker. The concept of impedance is
commonly used to study the performance of a loudspeaker system.
The acoustic load impedance is the effect of the acoustic field on the
diaphragm that may affect the loudspeaker response, particularly in
the low frequencies due to the coupling with the room modes. This
impedance can hardly be modeled because it depends both on the
physical characteristics of the loudspeaker and the complexity of the
acoustic field in which it radiates. Conventional measurement
techniques require external sensors such as microphones or
accelerometers.
The proposed approach relies mainly on the reciprocal mechanism of
electrodynamic transduction, making it possible to use the
loudspeaker as a sensor simultaneously. This work shows that the
acoustic load impedance can be derived from an input electrical
impedance measurement and the knowledge of the electromechanical
parameters of the loudspeaker. An analytical model is given and data
measured in actual rooms provides proof of concept. Potential
applications are also discussed.
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Modelling of a MEMS transducer using approximate eigenfunctions of a square clamped plate
Karina Abramova, Petr Honzík, Nicolas Joly, Stéphane Durand, Michel Bruneau
[Abstract]
The model of electroacoustic MEMS transducer with a moving
square shaped clamped plate loaded by a thin fluid gap and a
peripheral cavity is presented herein. The behaviour of the
transducer, namely the thermal and viscous boundary layers
effects originating in the fluid gap between the moving
electrode and the fixed one and the strong coupling between
the moving electrode displacement and the acoustic pressure
field in the fluid gap, have to be described correctly by the
model. The modelling approach proposed herein involving the
integral method for describing the acoustic pressure in the fluid
gap requires an analytical expression of eigenfunctions of the
square shaped clamped plate. Such an approximate
expression in form of two-dimensional cosine series proposed
recently suffers from slow convergence and inaccurate
boundary conditions. The solution proposed herein is based on
the series expansion over the system of functions satisfying
exactly the boundary conditions, which leads to faster
convergence, hence lowering computational costs. The
proposed eigenfunctions are described and the difference from
the previous approximations is discussed. Finally, the acoustic
pressure sensitivity of the transducer is presented and compared
to the results of a reference finite element model.
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Temperature effects on the mechanical-acoustic properties of condenser microphones: experimental characterization
Cécile Guianvarc'H, Thomas Lavergne, Lara Risegari, Petr Honzík, Dominique Rodrigues, Roberto M. Gavioso
[Abstract]
Condenser microphones are originally designed for airborne acoustic measurements. For this reason, their electro-mechanical behaviour is well-known and characterized almost in atmospheric conditions. However, more and more applications, such as acoustic thermometry, gas metrology or thermoacoustics, require the use of electrostatic transducers (as acoustic transmitters or receivers) in gas conditions that differ significantly from atmospheric ones.
Previous experiments using the electrostatic actuator technique evidenced the relevant influence of the static pressure and the type of gas on the frequency response of condenser microphones. The same technique is used here to characterize the mechanical properties of a 1/4” microphone’s membrane in vacuum at temperatures between 80 K and 300 K. The resonance properties of the membrane are then obtained and expressed as function of the temperature. Thus, they are used in comprehensive models of condenser microphones whose validity as function of the temperature is experimentally checked. Also, an improved model of condenser microphone coupled to an electrostatic actuator is developed for the sake of this work.
By improving the supporting models and experimental methods, this work is expected to drive the design of new electrostatic transducers and promote an advance of the calibration procedures, as required for specific applications.
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Analytical modelling of a MEMS transducer composed of a rigid micro-beam attached at one end to a flat spring moving against a reduced-size backplate
Petr Honzík, Antonin Novak, Stéphane Durand, Nicolas Joly, Michel Bruneau
[Abstract]
The use of planar micro-beams as moving parts of acoustic and
electroacoustic devices has increased recently because of
their geometrical simplicity, hence lowering fabrication costs.
The precise modelling of such devices is then of interest. The
miniaturized transducer proposed herein is composed of a
planar rigid micro-beam attached at one end to a flat spring
(the other end remaining free) surrounded by thin slits and
loaded by a thin fluid layer (situated in the gap between the
micro-beam and a reduced-size backplate) and a small
cavity, both being placed behind the beam. Such a
configuration reduces the overall size of the device (no need of
an external cavity) and enables to adjust more parameters
comparing to the case of the backplate of the same size as
the one of the micro-beam. The thermoviscous damping effects
originating in the fluid-filled parts of the device (slits, air-gap,
and cavity) are taken into account. As a result of the model,
the displacement of the micro-beam is calculated and
compared with the reference finite element solution, the
acoustic pressure sensitivity of the transducer is finally
presented and discussed.
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Organization:
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W. De Roeck, KU Leuven |
W. Desmet, KU Leuven/Member of DMMS Lab, Flanders Make |
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Generalized Method of Describing Acoustic Duct-Like System as a Multiport
Anna Snakowska, Jerzy Jurkiewicz
[Abstract]
The paper presents general method to analyse acoustic mufflers or other duct-like systems assuming propagation of a multimode wave and applying the multiport method. Apart from the main propagating tube and expansion chambers the analysed device may contain segments of annular duct (concentric or not with the central tube) branch resonators, cavities and cavity diaphragms. Systems of such complicated geometrical structure have been widely analysed under the assumption that the wave propagating through any element is the plane wave. Thus the problem might have been analysed by means of the two-dimensional transfer matrices (or others equivalents) derived from conditions of conservation of the acoustic pressure and the volume velocity across each discontinuity. However, the assumption of the pure plane propagation across every element of the acoustic system imposes strong limitations on the excitation frequency or the radius of the duct segments as the Helmholtz number should not exceed the value 1.93 which represents the non-dimensional cut-on frequency of the first circumferential mode. If the excitation and the system are axially symmetric the limited value of the Helmholtz number raises to 3.83. The analysis is carried out by means of the scattering matrix which is finally transformed into transmit matrix.
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Characterization of the centrifugal fan noise generated in residential HVAC systems using in-duct measurements
Sarah Van Tricht, Pieterjan Broos, Hervé Denayer, Maarten Vanierschot, Wim De Roeck
[Abstract]
The acoustic characterization of a centrifugal fan in a
residential HVAC system is carried out through the
determination of the total source spectrum of the sound
generated by the fan under varying operating conditions.
The most commonly used characterization technique,
described by ISO 5136, requires an anechoic termination
and the possibility to insert a microphone in the duct.
Since this approach largely depends on the efficiency of
the anechoic termination, an active one-port
characterization using flush mounted microphones is
adopted in this research. One of the major advantages of
this approach is the fact that the characterization is
carried out independent of downstream reflections, thus
not requiring the construction of an anechoic termination.
At first, the passive property of the fan, i.e. the reflection
coefficient of the propagating waves at the inlet of the
fan, is determined. Subsequently, the downstream
reflection component of the test rig is characterized.
Finally, both passive models are combined to compute
the active source spectrum describing all noise
generation and dissipation phenomena of the fan. The
acoustic characterization is compared to existing scaling
laws, such as described e.g. in VDI 2081.
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Trained Algorithms for Mode Decomposition in Ducts
Stefan Sack, Mats Åbom
[Abstract]
Acoustic waves in duct systems become non-linear if their pressure amplitudes or particle velocities exceed critical levels. The critical levels depend on the pressure and velocity of the ambient fluid and the geometry of the flow channels. A microperforated plate mounted inside a duct, for example, may have a very large local particle velocity inside the perforations even for relatively low acoustic excitation levels. Classical models for in-duct sound such as the acoustic two-port are only valid within the linear region and, therefore, under certain circumstances describe the scattering through perforations inadequately. Although some attempts have been made to incorporate non-linearity into existing methods, experimental studies still suffer from varying particle velocities depending on the frequency. We developed an extension to the classical two-load method that uses a combination of excitation sources to guarantee a constant particle velocity for all tested frequencies. By means of this method, a non-linear scattering matrix can be extracted. As a test case, the particle velocity-dependent scattering of a perforated panel inside a circular duct is presented.
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Prediction of the nonlinear response of an acoustically forced flow with linearized Navier-Stokes equations
Abel Faure, Claire Bourquard, Nicolas Noiray
[Abstract]
The prediction of self-sustained aeroacoustic oscillations
has attracted
significant research interest for decades. However,
quantitative
predictions of whistling amplitude on practical
configurations are still
challenging. In this study we use particle image
velocimetry to record
flow data at an acoustically excited T-Junction subject to
low-Mach
grazing flow. Data is recorded for several acoustic
excitation
amplitudes and frequencies. This allows to formulate a
model
predicting the non-linear response of the system with
Navier-Stokes
equations linearized around different mean flows, which
were
measured with increasing excitation amplitudes. This
linearized
perturbation approach gives a good prediction of the
spatial
distribution of the velocity fluctuations and predicts their
amplitude to
a reasonable degree. The study concludes with a
discussion on the
benefits and drawbacks of modeling the aeroacoustic
coupling at the
shear layer with the incompressible Navier-Stokes
equations
linearized around the mean flow.
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Self-Excited Vibration of Valve Caused by Hydrodynamic Force
Xiangdong Liang
[Abstract]
Liquid vibration is the source of duct noise and is main reason
of pipe connector to be loosen and leaking out working
medium from the connector also. Besides machine linked to
the pipe line, valve in the pipe line is also one kind of vibration
source to the pipe line because liquid passing through the
valve can cause valve to vibrate and radiate noise.
Thermostat control valve is a kind of self-adjust valve used in oil
pipe line for protecting machines. To control the oil
temperature in selected range the thermostat control valve
can adjust the oil flows to the heating branch pipe and
cooling branch pipe as the oil temperature is changed. But
when it adjusts flow of oil, vibration and noise will be taken
place in the valve. And this kind of liquid vibration may
damage the thermal rod of the thermostat control valve. The
mechanism of how this kind of valve liquid vibration taken
place will be shown in this paper. Some of suggestion about
decreasing or avoiding this kind of vibration on the oil pipe line
will be given in this paper based on the conclusion of
investigation.
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Organization:
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N. Henrich, Université Grenoble Alpes |
M. Döllinger, Division of Phoniatrics and Pediatric Audiology, University Hospital Erlangen |
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Acoustic Source Term Interpolation in Hybrid Aeroacoustic Simulation of Human Phonation
Michael Weitz, Stefan Schoder, Paul Maurerlehner, Sebastian Falk, Michael Döllinger, Manfred Kaltenbacher
[Abstract]
We use a hybrid aeroacoustic approach that relies on an
incompressible flow simulation resolving relevant turbulent
scales. Based on the flow results, we compute the
acoustic source terms on the flow grid and perform a
conservative interpolation to the acoustic grid, on which
we solve the perturbed convective wave equation to
obtain the acoustic field. This conservative transformation
of the acoustic sources from the flow grid to the acoustic
grid is a crucial step to allow coarse acoustic grids without
reducing accuracy. An advanced cut-volume-cell
approach, guaranteeing high accuracy both for regions
where the flow grid is finer and for regions where it is
coarser than the acoustic grid, is employed. In this
framework, the radial basis function method is
incorporated to compute spatial derivatives of the flow
data as necessary in the computation of the acoustic
sources. The simulations demonstrate the applicability of the cut-
volume-cell interpolation method to obtain an accurate
transfer of the acoustic sources from the flow grid to the
acoustic grid. The acoustic results changed very slightly
even for a reduction of elements in the source region of
more than 10 times compared to the initial acoustic grid,
leading to a strong reduction in CPU-time.
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Acoustic characterization of an averaged vocal tract model based on the MRI data of professional tenors
Judith Probst, Alexander Lodermeyer, Sahar Fattoum, Matthias Echternach, Stefan Becker, Stefan Kniesburges
[Abstract]
Human speech is created in the larynx and the vocal tract. For vowels, the basic sound is generated by the oscillating vocal folds and filtered by the vocal tract to produce the typical formants.
In the present study, we investigate the influence of the vocal tract on phonation. Based on Magnet Resonance Imaging (MRI) data, the acoustic behavior of the vocal tract can be characterized using numerical and experimental models. However, previous studies only analyzed single vocal tracts that were directly obtained from individuals. To investigate the basic influence of the vocal tract on phonation, a vocal tract model is necessary that shows preferably little individual but the typical acoustic characteristics of the specific vowel. Hence, we computed an average vocal tract geometry based on six simplified individual vocal tracts that were extracted from MRI images of professional tenors. The resulting vocal tract model was evaluated in an experimental setup including artificial vocal folds. The comparison with the tenors’ vocal recordings shows that the model reproduces the acoustic properties well. Hence, the average vocal tract model presented here is suitable for future investigations of the influence of the vocal tract on phonation.
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Influence of the vocal tract on voice directivity
Rémi Blandin, Manuel Brandner
[Abstract]
Voice directivity induces variations of the sound amplitude
and frequency content with the direction.
Voice directivity is important for the efficient transmission of
speech and singing.
Therefore, it is
taken into account both by the artists and the designers of
concert halls in order to enhance the
quality of artistic performances.
While the shape of the head and the torso has long been
considered to study this phenomenon,
the influence of vocal tract shape has received very little
attention.
However, it has recently been shown that it can influence
voice directivity through the
internal acoustic field and the frequency content.
The contribution of the vocal tract to the voice directivity is
characterized through
physical modelling and direct observation of speakers and
singers.
In particular, the role of higher order propagation modes is
discussed as well as strategies to
control directivity through the frequency content of voice.
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Finite Element Simulation of /asa/ in a Three-Dimensional Vocal Tract using a Simplified Aeroacoustic Source Model
Marc Arnela, Oriol Guasch
[Abstract]
The numerical simulation of fricative sounds in three-dimensional
(3D) vocal tracts typically involves hybrid Computational
Aeroacoustics (CAA) approaches. Unfortunately, those are very
costly and require from super-computer facilities to produce a
mere few milliseconds of sound. The problem becomes
prohibitive if one not only aims at generating a single fricative,
but also sequences containing both, vowels and fricatives. It
then seems wise to try to approximate somehow the flow noise
sources, so that only an acoustic simulation becomes necessary
avoiding the very demanding computational fluid dynamics
step in CAA. This can be done to different levels of precision. In
this work, it is suggested to follow a similar methodology to that
in one-dimensional techniques, but applied to 3D dynamic
vocal tracts. Vowel sounds are generated introducing glottal
pulses at the vocal tract entrance (glottis), while a monopole
source consisting of filtered white noise is activated in a location
immediately downstream of the constriction where a fricative
sound is articulated. Acoustic wave propagation in a 3D
dynamic vocal tract is simulated using a stabilized Finite Element
Method (FEM) for the wave equation in mixed form, set in an
arbitrary Lagrangian-Eulerian (ALE) framework. The sequence
/asa/ is produced as an example.
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Glottal Opening Measurements in VCV and VCCV Sequences
Yves Laprie, Benjamin Elie, Angélique Amelot, Shinji Maeda
[Abstract]
Many studies on speech acoustics and production use articulatory synthesis as a
framework to investigate the relationship between articulatory gestures and acoustic
features. Although many supraglottal articulatory models are available, usually built
from vocal tract imaging acquisitions, glottal gestures are commonly modeled with
simple geometric primitives which do not necessarily reflect reality. This is an importa
issue as glottal opening gestures are very important to control the glottal flow at the
vocal tract input and the supraglottic acoustic flow. This study is a first step towards the
development of a database of realistic glottal gestures which will be used to design the
glottal opening dynamics in articulatory synthesis paradigms. In the first part of this
paper, we present experimental measurements of glottal opening dynamics in VCV and
VCCV sequences uttered by real subjects, thanks to a specifically designed external
photoglottographic device (ePGG). The corpus was designed to highlight the
differences in glottis opening between fricatives and stops. The existence of different
patterns of glottal opening is evidenced according to the class of the consonants. A
numerical study is then used to show the influence of these patterns on the production
of sounds and on the coarticulation.
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Formation of stationary intraglottal vortices during phonation
Liran Oren, Sid Khosla, Charles Farbos De Luzan, Alexandra Maddox, Ephraim Gutmark
[Abstract]
According to the myoelastic-aerodynamic theory of voice production, the vibrations of
the vocal folds occur due to interaction between the fold’s elasticity and intraglottal
aerodynamics. The shape of the glottal opening changes from converging to diverging
during the opening and closing phases of the folds vibration, respectively. The divergent
shape causes the glottal jet to separate from the medial wall and rotational flow
develops in the regions between the wall and the jet. The formation of these intraglottal
flow separation vortices (FSV) was shown both experimental and computationally, but
their role in the phonation process is still an ongoing debate. Few models for the
vibration mechanism have predicted that FSV would form as shedding vortices, thus
convecting downstream with the glottal jet. In the current study, we used time-resolved
phase-locked PIV measurements of intraglottal flow in a canine larynx model. Results
show that FSV develop and remain as recirculation zone near the superior aspect of the
folds. It is important to correctly model FSV because they can produce negative
pressure near the superior aspect of the folds, which we hypothesize can increase
maximum flow declination rate (MFDR). MFDR is important because it is clinically
related to loudness and vocal efficiency.
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Modeling the pre-phonatory vocal fold posture in the larynx model SynthVOICE
Stefan Kniesburges, Reinhard Veltrup, Sahar Fattoum, Anne Schützenberger
[Abstract]
The basic aim of synthetic larynx models is to reproduce the phonation process. Thus, we develop the synthetic larynx model SynthVOICE that incorporates the control of the pre-phonatory posture (ad-/abduction) and the elongation of the vocal folds (VFs). The aim is to reproduce the physiological and pathological phonation process especially dysfunctions of the muscular and nerval control such as muscle tension dysphonia.
The model is manufactured in a multistage pouring process using liquid silicone mixtures with a different stiffness. It is composed of a cylinder mimicking the thyroid cartilage, the VFs and five manipulator devices to control the pre-phonatory VF posture and tension. Both, single- and three-layer VF models can be inserted that are based on the M5 shape. The pre-phonatory settings are controlled by a mechanical setup that exhibits seven linear and two rotational traverse tables. A Matlab script determines the appropriate displacements and angles of the traverse tables to move the VFs in a specific posture.
To validate the SynthVOICE model, we present measurements of the oscillation threshold pressure, the oscillation frequency and the produced acoustic signal for different settings of VFs posture. The Else-Kröner-Fresenius Stiftung is highly acknowledged for its funding (grant-no. 2016_A78).
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Organization:
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S. Moreau, University of Sherbrooke |
S. Becker, Institute of Process Machinery and Systems Engineering, University of Erlangen |
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Analysis of Sound Source Localization in an Axial Fan
Seyed Mohsen Alavi Moghadam, Matthias Meinke, Wolfgang Schröder
[Abstract]
Numerical analyses of the tip vortex system of a ducted axial fan
are conducted by a hybrid fluid-dynamics-acoustics method. First,
large-eddy simulations are performed to investigate the dynamics of
tip clearance flow for various tip-gap sizes and to determine the
acoustic sources. The simulations are performed for a single blade out
of five blades with periodic boundary conditions in the
circumferential direction on a multi-block structured mesh with
140$\times10^6$ grid points. The turbulent flow is simulated at a
Reynolds number of 9.36$\times10^5$ at undisturbed inflow condition
and the results∼\cite{Moghadam2019} are compared with experimental
data∼\cite{Tao2013}. The results show that increasing the tip-gap size
results in various vortices in the tip-gap region, i.e., tip-leakage,
separation, and induced vortices, which enlarge the diameter and the
strength of the main tip vortex. For the largest tip clearance, spiral
vortex breakdown occurs at the design operating condition. Second,
the acoustic field on the near field is determined by solving the
acoustic perturbation equations (APE)∼\cite{Ewert2003} on a mesh for a
single blade consisting of approx. 1060$\times10^{6}$ grid points. The
results show that the larger the tip gap size the higher the broadband
noise level∼\cite{MoghadamACTA2019}. Detailed results will be presented at the conference.
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Prediction Methodology of Broadband Noise from a Cooling Fan
Soichi Sasaki
[Abstract]
The methodology for prediction on broadband noise of a
cooling fans using in HVAC system will be proposed. Because
such the fan cools the heating device by its own wind velocity,
the fan has no duct and also there is little static pressure rise. This
methodology has an originality applying a modified blade
element momentum theory for the prediction of the broadband
noise. The slip factor of the blade element is evaluated based
on the measured aerodynamic characteristics and the Euler
pressure. The slip factor can give the attack angle and relative
velocity of the blade elements in the design phase. The
characteristics of attack angle of the broadband noise
generated from the isolated blade are measured by wind
tunnel test. The broadband noise generated from the cooling
fan is predicted by superimposing the two dimensional blade
elements in the span direction. In the full length paper, the
procedure of the prediction methodology of the broadband
noise, the result of wind tunnel test of the isolated blade and the
comparison of the actual measurement noise spectra
generated from the fan and its predicted value will be
presented.
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Numerical Investigation of Hydrodynamic/Acoustic Splitting Methods in Finite Volumes including Rotating Domains
Joscha Piepiorka, Otto Von Estorff
[Abstract]
In today’s product development, the behavior of fans is increasingly important. Especially the noise emission is a major challenge to be handled by the manufacturers. Alongside expensive experimental investigations, reliable numerical simulations to predict the flow induced noise are gaining more and more importance. The paper will present first numerical results of an axial fan benchmark case [1]. A simplified model of the experimental setup is used for a scale-adaptive CFD simulation. Based on incompressible CFD results, the acoustic quantities are modelled by a finite volume approach, which has been developed in the software framework OpenFOAM. Different hydrodynamic/acoustic splitting methods are used to investigate the flow induced noise including the rotating domain of the fan. Special focus is placed on velocity terms and their influence onto aeroacoustic source and wave propagation effects. For this purpose, different hydrodynamic/acoustic splitting methods are introduced and reformulated. The resulting sound fields are computed and interpreted based on the theoretical assumptions. Finally, the computational effort in the finite volume framework is discussed and the results are compared to the experimental data [1: Zenger, Florian, et al. A benchmark case for aerodynamics and aeroacoustics of a low pressure axial fan. No. 2016-01-1805. SAE Technical Paper, 2016].
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Vortex sound based calculations for the aeroacoustic noise of a centrifugal fan
Hakan Dogan, Martin Ochmann, Chris Eisenmenger, Stefan Frank
[Abstract]
Radial ventilators are widely used in industrial applications such as home appliances, automobile industry and air-conditioning devices. In this work, we investigate the aerodynamic and aeroacoustic performance of a radial ventilator with nine backward curved blades by means of computational fluid dynamics and aeroacoustics. In a few previous publications by the authors, some experimental and numerical results were presented and compared. Regarding the aeroacoustic computations, the far field noise was calculated by solving the Helmholtz equation using the boundary element method (BEM) or the finite element method (FEM). Moreover, FEM calculations using Lighthill’s analogy (Lighthill’s wave equation) were also done. In the current paper, we perform vortex sound theory based aeroacoustic calculations, namely using Powell’s analogy. The results obtained using vortex sound theory will be compared with the theoretical/numerical approaches mentioned above, and with the experimental results. As stated by some previous researchers, although Lighthill’s and Powell’s formulations pose only slight differences mathematically, they exhibit considerably different numerical robustness. The application, numerical stability and accuracy of the two methods with regards to the current problem will be investigated and compared in this paper.
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Experimental Investigation of the Influence of different Leading Edge Modifications on the Sound Emission of Axial Fans downstream of a Heat Exchanger
Felix Czwielong, Florian Krömer, Chaitanya Paruchuri, Stefan Becker
[Abstract]
Axial fans have a major impact on the sound emission of technical systems. One of the common applications of low-pressure axial fans is their usage in air conditioning and cooling circuits. In most cases the fan operates downstream of a heat exchanger in this system. The suction side heat exchanger changes the upstream flow field of the fan and consequently the fan acoustic behavior compared to undisturbed inlet conditions. Investigations on sound emission of airfoils showed that leading-edge modifications like serrations or slits reduce the turbulence-airfoil interaction noise. In this study, the impact of three different leading-edge modifications (straight leading edge, sinusoidal serration and slits) on the sound emissions of a forward skewed low-pressure axial fan was examined. The inflow conditions varied between a free inflow and a disturbed inflow, generated by a heat exchanger located on the suction side of the axial fan. The results show that the modification of the leading edge has a positive influence on the sound radiation of the axial fan for both inlet conditions discussed above.
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Aeroacoustic Simulation and Experimental Validation of Sound Emission of an Axial Fan Applied in a Heat Pump
Andreas Lucius, Marc Schneider, Stefan Schweitzer-De Bortoli, Tom Gerhard, Thomas Geyer
[Abstract]
The reduction of noise emission of heat pumps is a major development target for the appliance manufacturer. An important source of sound is the fan. The fan is operating at disturbed inflow conditions due to installation in the appliance. Additionally small scale turbulence generated by the heat exchanger further increases fan noise emission. Noise emission may be reduced via improved installation or specific fan design.
The aim of the project was to measure flow and turbulence characteristics inside the heat pump via constant temperature anemometry as well as noise emission. The acquired data set was then used for validation of CFD and CAA simulations. The effect of inflow turbulence was analyzed for configurations with and without heat exchanger. Hot wire measurements show increased average turbulence levels of 20 percent downstream the heat exchanger. In a second step URANS simulations were conducted for both configurations. The heat exchanger was modelled as porous medium. Turbulence properties could be matched with experiments with reasonable boundary conditions at the heat exchangers outlet.
A LES-based CAA simulation of the heat pump was done for the configuration without heat exchanger. There is some deviation in flow rate but quite good prediction of sound power.
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Experimental investigation of the noise reduction of a plug fan with leading-edge serrations
Ignacio Zurbano-Fernandez, Alain Guedel, Mirela Robitu
[Abstract]
In previous years, leading-edge serrations have been
successfully used as a noise-reduction technique applied
to airfoils. More recently, this technique has been also
used and validated on axial fans. In this study, sinusoidal
leading-edge serrations have been integrated on a plug
fan (centrifugal fan without volute). The outcome of RANS
simulations of the non-serrated baseline impeller has been
used to define an assumed optimum serration geometry in
terms of noise reduction. This optimum geometry has been
used as a reference to manufacture three impeller
prototypes of various geometries, defined by their
amplitude (2h) and wavelength (λ).
This paper will present the results of the test of the
prototypes in a reverberant room, where noise and air
performance measurements are done simultaneously. For
each impeller, five fan operating points will be presented,
and for each one of the points the sound power level is
measured at both the inlet and the outlet of the fan. This
allows us to assess the effectiveness of serrations at
different operating points, while checking the impact on
the fan efficiency. The results will also be compared with
the non-serrated baseline fan.
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Computational Aeroacoustics of an Axial Fan with Leading Edge Serrations
Stefan Schoder, Florian Krömer, Michael Weitz, Manfred Kaltenbacher, Stefan Becker
[Abstract]
In recent years a trend towards modifications of the leading edge geometry in airfoils for reducing the sound emission and improving the stall behavior has been observed. However, low-pressure axial fan design methodologies for a decreased sound radiation almost rely exclusively on the use of fan blade skew rather than on leading edge modifications so far. Hence, with this study we aimed to investigate the sound reduction capabilities of leading edge serrations applied to low-pressure axial fans using the efficient perturbed convective wave equation.
For this, we used a reference fan with straight leading edges and a fan with serrated leading edges that has been studied in experimental investigations. The comparison of the CFD and CAA simulations with experimental results show good agreement. Furthermore, we analyzed the sound reduction mechanisms of leading edge serrations based on the CAA results. In general a sound reduction was obtained with leading edge serrations without influencing the fan operating point. The study demonstrates the applicability of this aeroacoustic approach for axial fans with modified leading edges.
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Axial Ventilator Tip Gap Noise Prediction from Discontinuous Galerkin based CAA with Stochastic Vortex Sound Sources
Lev Liberson, Markus Lummer, Michael Mößner, Roland Ewert, Jan Werner Delfs
[Abstract]
A prominent noise source of axial ventilators emerges from the complex flow within the tip clearance between fan blades and surrounding encasement. To obtain an aeroacoustic fan design, an associated process must be acceptably efficient while providing a sufficiently amount of physical modeling depth. The mechanisms of noise generation as well as the sound propagation in highly inhomogeneous flow fields have to be captured.
A promising method is developed at the DLR by using the in-house CAA propagation solver DISCO++ coupled with the stochastic acoustic source generator Fast Rand Particle Mesh Method (FRPM). Utilizing the Discontinuous Galerkin method the Acoustic Perturbation Equations (APE) are solved on an unstructured tetrahedral mesh, while the acoustic sources are reconstructed on a Cartesian background grid.
The prediction capability of the method is evaluated by computing the noise generated by a ducted axial ventilator at two different tip clearance variations. While the trailing edge noise is well captured, the tip gap noise increase is not properly predicted in terms of spectral shape - presumably due to anisotropic length scale influence not properly recognized by the current stochastic model. First simulations based on an enhanced anisotropic length scale model will be presented and results discussed.
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Aeroacoustic evaluation of the forward-curved fan inlet flow
Jurij Gostiša, Tadej Novaković, Jurij Prezelj, Marko Hočevar
[Abstract]
Forward-curved fans have been extensively used in various
industrial and residential HVAC applications. Regarding their
favourable characteristic properties, low noise emissions and
small size, forward-curved fans have found use in various
applications with requirement for high flow rates at moderate
pressures and efficiency not being of primary importance.
Design of the fan’s inlet flow channel is heavily influenced by
limited space available, which results in adverse flow
aerodynamics. One of the main reasons for poor efficiency of
the forward-curved fan is the flow separation, which starts at the
inlet, develops into the blade area and then into the volute.
Such vortical flow occupies about a third of the rotor’s width,
resulting in a significant flow rate and efficiency reduction. Local
measurements of flow velocity using hot wire anemometry were
performed and a novelty approach with psychoacoustic
metrics adopted to velocity signal processing was used to
improve the understanding of the aerodynamic phenomena in
the inlet area. The findings exhibit distinct areas and properties
of adverse vortical flow and are used in the further inlet channel
design process.
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Influence of the Mesh Size on the Aerodynamic and Aeroacoustics of a Centrifugal Fan using the Lattice-Boltzmann Method
Rebecca Schaefer, Martin Boehle
[Abstract]
Due to market requirements and ever stricter guidelines, noise emissions of fans are becoming increasingly important in addition to energy efficiency. For this reason, developer aims not only at more efficient running fans but also ever quieter fans. In order to keep the development process cost-effective and short, numerical investigations are used today. The Lattice-Boltzmann method (LBM) is a relatively new method promising to calculate aerodynamics coupled with aeroacoustics.
The present paper deals with the simulation of the aerodynamics and aeroacoustics for a centrifugal fan using the commercial LBM Code PowerFLOW of the Exa Corporation. The focus is on the variation of the mesh size. The aim is to determine the smallest cell size and the geometry defined for refinement in such a way that both the flow and acoustics of the centrifugal fan can be accurately predicted with an acceptable total computing time. At the end a total of four operating points are analysed.
The results of the LBM simulation are compared and validated with measurement results. Flow information in terms of the pressure rise as a function of the volume flow rate is well captured. In addition, the noise is well predicted with respect to blade passing frequency.
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Research on Aerodynamic Noise Calculation and Noise Reduction Design of Multi-blade Centrifugal Fan
Jin An Huang, Yang Xiang, Chaojun Jiang
[Abstract]
Taking the multi-blade centrifugal fan in an air-
conditioning system as the object, the three-dimensional
modeling of the internal fluid of the fan is carried out. The
steady-state and non-steady-state calculations are
carried out in the CFD software, obtaining the internal
flow field information, with calculation result imported
into Noise estimation which was performed in the LMS
Virtual.lab. Besides, the accuracy of the fan
aerodynamic noise calculation was verified against the
experimental results. Through the analysis of the
mechanism of the aerodynamic noise of the fan, the
main noise source of the aerodynamic noise of the fan is
located at the impeller and is closely related to its
internal flow field distribution and its structure. For the fan
products that have been put into production, it is difficult
to modify the main structure including the volute, the
impeller, the blade size, etc., and only local structure can
be redesigned to achieve the purpose of noise
reduction. Therefore, the blade perforation design is
adopted. By setting reasonable perforation parameters,
the eddy current shedding around the blade can be
reduced without changing its performance, so as to
reduce the pressure pulsation of the blade surface and
the aerodynamic noise of the fan.
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Organization:
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P. Croaker, UNSW Australia |
M. Liefvendahl, Swedish Defence Research Agency (FOI) |
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Simulation of flow noise generated by the interaction of inflow turbulence with the leading edge of a foil
Mattias Liefvendahl
[Abstract]
Noise generated by the interaction of inflow turbulence
with a lifting surface may be of interest for a number of
hydrodynamic applications, including a propeller
operating in non-cavitating condition. In the present work, simulation methods are applied for the
prediction of the radiated noise. The experimental set-up
of Santana et al. (2016) will be investigated at a chord-
based Reynolds number, Re = 256 000, and Mach number
of 0.09, with strong homogeneous inflow turbulence
interacting with the foil. The large-eddy simulation (LES) technique will be
employed to predict the flow, with synthetic turbulence
fluctuations applied at the inflow boundary. The LES will be
employed to compute the acoustic source terms of an
acoustic analogy (Liefvendahl and Bensow, 2018). The
spectrum of turbulence upstream of the airfoil leading
edge will also be extracted from the LES and combined
with semi-analytical scattering models for leading edge
noise. The flow field and the radiated noise predicted
with the LES and semi-analytical techniques will be
compared with experimental data from the literature. Liefvendahl & Bensow, 32nd Symp. on Naval
Hydrodynamics, 2018. Santana et al., J. Sound and Vibration, 383:349-363, 2016.
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Noise from a blunt edged flat plate in a reverberant water tunnel
Paul Croaker, James Venning, Mahmoud Karimi, Paul Brandner, Con Doolan, Nicole Kessissoglou
[Abstract]
Large eddy simulations of flow past a blunt edged flat
plate located in a reverberant water tunnel are
presented. The flow is at a Reynolds number based on
chord of 6.8e6 and a Mach number of 5.3e-3.
Hydrodynamic pressures on the surface of the plate are
extracted and processed to investigate the wavenumber-
frequency content of the pressure fluctuations. Unsteady
velocities at the mid-span plane of the plate are also
processed to extract volumetric flow-induced noise
sources in the vicinity of the blunt edge. The surface
pressures and volumetric source terms are then
combined with semi-analytical models of trailing edge
scattering to predict the pressure fluctuations on the wall
of the water tunnel. These analytical models were
originally developed to analyse far-field sound produced
by flow past a structure in air. This work investigates the
suitability of applying these scattering models to near-field
prediction of flow-induced noise in a reverberant water
environment. The results are compared with experimental
measurements taken in the Cavitation Research
Laboratory at the Australian Maritime College, University of
Tasmania.
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Organization:
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T.P. Chong, Brunel University London |
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Aerofoil trailing edge self-noise reduction by Surface Mounted Attenuation Devices
Edvard Schroeder, Tze Pei Chong, M. Kamruzzaman, Jeremy Hurault, Phillip Joseph
[Abstract]
This paper presents the results of a preliminary experimental study into the effect of add-on type finlets on the aeroacoustic performance of a wind turbine aerofoil. The content mainly deals with the turbulent boundary layer -trailing edge broadband noise characteristics subjected to the add-on finlets.
The test program seeks to test various combinations of finlet height and spacing present on the (1) suction side only, (2) pressure side only, and (3) both suction and pressure sides. Each finlet parameter configuration is tested at jet velocity U=30m/s and 45m/s, at geometric angles of attack AOA (geometric) = 10∘, 0∘, -10∘.
Based on the data acquired thus far it is observed that a finlet usually performs best with increasing height and decreasing spacing. Under configurations (1) and (2), finlets are able to produce up to -2dB Sound Power Level broadband noise reductions, where configuration (3) offers broadband reductions of up to -7dB Sound Power Level, with potential to observe more reductions once optimised. The main parameter for improved performance of finlets is the spacing rather than the height. Finlet height does not offer significant impact on the performance if the spacing is not optimal.
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The Effect of flow-permeable material on the Flow Field and the Aerodynamic Noise of Cylinders
Thomas Geyer, Richard Peschel
[Abstract]
The flow around circular cylinders is one of the major aeroacoustic noise source mechanisms. Such a cylinder represents a simple model for technical applications like parts of the landing gear of airplanes, train pantographs, antennas or other protruding parts of vehicles. A reduction of the flow noise is especially desirable since it contains both a broadband component and tonal components due to the regular vortex shedding. One approach to reduce this noise is to cover the cylinder with flow-permeable materials, such as foams or porous rubbers.
In a recent experimental study in the aeroacoustic wind tunnel at the Brandenburg University of Technology, both the aerodynamic noise radiation as well as the flow field around a set of porous covered cylinders were investigated. The acoustic measurements were performed at Reynolds numbers (based on cylinder diameter) between 16000 and 100000 using single microphones, while the detailed flow measurements were done using hot-wire anemometry. The results show the influence of the material properties and of the thickness of the porous layer on the aeroacoustic noise and on the turbulence in the wake of the cylinders.
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On the Effects of Leading Edge Serrations on Aeroacoustic Properties during Stall
Yannick D. Mayer, Bin Zang, Mahdi Azarpeyvand
[Abstract]
The understanding of aerofoil noise and possible mitigation
approaches is of key importance for the aerospace and wind
turbine industry. Despite their relevance and mostly unknown
physical noise generation mechanism, separation and stall noise
have not received much research attention. In order to control
the aerodynamic stall behaviour, serrated leading edges have
previously been employed resulting in a less abrupt stall
behaviour of aerofoils, but their effects on the hydrodynamic
pressure field around aerofoils is not well understood. Hence, the
current experimental investigation aims to further the
understanding of stall and separation noise and its mitigation
using leading edge serrations. Experiments are therefore
conducted in the aeroacoustic facility of the University of Bristol
using a highly instrumented NACA0012 aerofoil with a large
number of static pressure taps and unsteady surface pressure
transducers. A variety of leading edge serrations have been
designed in order to vary the strength of the induced vortices by
the leading edge serrations at high angles of attack.
Additionally, simultaneous hot-wire measurements will allow
correlation measurements to be conducted between unsteady
surface pressure fluctuations and the existing flow field, in order
to improve the understanding of the underlying physics.
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Optimisation of Leading-edge Undulations of a NACA 65(12)-10 Aerofoil for Noise Reduction and Aerodynamic Enhancement
Kwing-So Choi, Jung-Hoon Kim, Giovanni Lacagnina, Phillip Joseph, Seyed Mohammad Hasheminejad, Tze Pei Chong, Muhammad Farrukh Shahab, Mohammad Omidyeganeh, Alfredo Pinelli
[Abstract]
Leading-edge undulations or tubercles of humpback whale
flippers have been known as one of biomimetic technologies
adaptable to flow control of aerofoils, particularly at post stall
conditions. These leading-edge undulations are also known to
reduce noise resulting from an interaction with on-coming
turbulence. We have recently carried out a parametric studiy of
a NACA 65(12)-10 aerofoil with a view to optimise the amplitude
and the wavelength of leading-edge undulations for noise
reduction and aerodynamic enhancement. A 3x3 test matrix
composing of three amplitudes (h = 3%, 6% and 12% chord) and
three wavelengths (Lambda = 10%, 20% and 30% chord) was
used in the investigation, where lift, drag and noise were
measured at the Reynolds number of 100,000. In this test, a
turbulence-generating grid was installed at the inlet of the test
section to increase the turbulence level in the freestream and to
promote transition to turbulence near the leading edge of
aerofoils without a need for a trip device. Within the test matrix
considered, we found that the best improvement in CL,max is
given with the greatest wavelength and amplitude, whereas
the maximum noise reduction is obtained with h/C = 0.09 and
Lambda/C = 0.2.
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Leading- and trailing-edge noise reduction using serrations of new geometry
Benshuai Lyu, Lorna Ayton, Chaitanya Paruchuri
[Abstract]
The leading- and trailing-edge noise due to the interaction
of unsteady flow with the sharp edges of aerofoils are
common in many engineering applications. One of the most
widely used reduction methods is to use serrations, such as
those of a sawtooth and a sinusoidal profile. The optimal
serration geometry, however, remains largely unknown. In
this study, we propose a new type of serration which has an
ogee shape. Both model predictions and experimental tests
show such a new type serration provides improved
performance over conventional sawtooth serrations for
leading- and trailing-edge noise reduction. More
importantly, this new geometry does not introduce noise
increase across the entire frequency range of interest. It is
widely known that the leading-edge noise reduction
appears to agree more favourably with model predictions
than the trailing-edge noise reduction. In this study we also
investigate the possible causes for such a difference by
closely examining the wall pressure wavenumber spectrum
under the turbulent boundary layer.
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Aeroacoustic and Aerodynamic Study of Trailing-edge Serrated Airfoils in Tandem Configuration
Xiao Liu, Syamir Alihan Showkat Ali, Mahdi Azarpeyvand, Yannick D. Mayer
[Abstract]
This paper presents a comprehensive experimental study
on the
application of trailing-edge serrations as a passive control
method for reducing the aerodynamic loading on airfoils
in
several tandem configurations. The purpose of this study is
to
investigate the effectiveness of serrated trailing-edge on
NACA65-710 tandem airfoil to control and regularize the
turbulent flow within the gap area between the two
airfoils.
The
wake flow characteristics for an isolated NACA65-710
airfoil
with
and without the trailing-edge serration treatment have
been
carried out using two-dimensional Particle Image
Velocimetry(PIV) method. The aeroacoustic performance
of the
tandem airfoil configuration is also studied using in-site
pressure
transducers instrumented over the airfoil surface, as well as
far-
field microphones. To reveal the effects of trailing-edge
serration on the tandem configuration, flow experiments
were
performed using a slotted-sawtooth serration, for several
tandem configurations, with different airfoil gap distances.
The
results show that the use of serrations can generally lead to
a
significant reduction in the turbulent kinetic energy within
the
gap, due to the interaction between the flows issued from
the
tip and root of the serration. The far-field noise results have
shown that a significant noise reduction can be achieved
with
the use of the serrations.
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Numerical and experimental investigation on effects of blade tip-rake on vortex structure and aerodynamic noise of axial-flow fans in an outdoor unit of air-conditioners
Seo Yoon Ryu, Cheolung Cheong, Jong-Uk Kim, Byeong Il Park, Se Min Park
[Abstract]
In the case of axial fans, there are more complex mechanism involved in transferring the energy from the fan blades to a flowing fluid and generating aerodynamic noise due to the blade tip vortex. In this paper, the effects of blade tip shape on vortex structure and aerodynamic noise of axial fans which are used as a cooling fan in an outdoor unit of air-conditioners are numerically and experimentally investigated. Two types of axial fans are considered: one is with blade tip-rake and the other is without blade tip-rake. The blade tip-rake in introduced in the axial fan to reduce the tip vortex as the wing fence does in the airplane. First, the flow field around the fans was predicted by solving the three-dimensional unsteady compressible RANS equations using computational fluid dynamics techniques. Then, the validity of the numerical results are confirmed by comparing the predicted flow rates with the measure ones. The characteristics of flow field driven by the fans with and without blade tip rake are analyzed in detail, especially with an emphasis on the coherent vortex structure. Finally, the aerodynamic noise of axial fans are characterized according to the vortex structure related to blade tip shape.
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Near-field Aeroacoustic Characteristics of a Stalled NACA 0012 Aerofoil
Yannick D. Mayer, Bin Zang, Mahdi Azarpeyvand
[Abstract]
Stall and separation noise can be major contributors to the
noise emissions of rotating blades, such as wind turbines,
propellers and fans. However, the underlying physics of the
noise generation mechanism of stall noise has not received the
same research attention as trailing edge noise. Hence, this
paper investigates the effects a separated boundary layer has
on near-field flow structures and turbulent surface pressure
fluctuation statistics, and therefore noise radiation. In order to
investigate the wall bounded flow with flow separation,
experiments are conducted in the aeroacoustic facility of the
University of Bristol using a highly instrumented NACA 0012
aerofoil with 300mm chord. The aerofoil is equipped with 87
static pressure tappings and 91 miniature microphones
measuring surface pressure fluctuations over the aerofoil. The
experiments are conducted for free-stream velocities between
10m/s and 30m/s and a wide range of angles of attack,
including the post-stall regime. In addition, single-wire and cross-
wire measurements are conducted synchronously to the
unsteady pressure measurements, permitting correlation statics
between the collected flow quantities to be determined. This
aims to improve the understanding between eddies present in
the flow field and the resulting wall pressure statistics, and
thereby the noise generation mechanism.
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An Experimental study on Aeroacoustics and Shock Dynamics Associated with Hartmann Whistle
Sonu K Thomas, K Srinivasan
[Abstract]
Hartmann whistle refers to a device that employs a high speed jet impinging on a cavity, which is open at one end and closed at the other. One
advantage of using this device is that, it employs no moving part, which in turn makes it more reliable and safe. This device is capable of
producing high amplitude tones over a wide range of frequencies. The aero-acoustic field produced due to resonance can have high
amplitudes which can be used in Active Flow Control (AFC) applications. Experiments were done to study the interaction between jet and open-
closed end cavity, which in turn is responsible for the high amplitude sound produced. The nature of these sound field produced are not
completely evident from the prior studies. In the full paper, the aero-acoustic measurements, the resonance induced pressure oscillation
measurements in the cavity and the flow visualization using a high speed Schlieren technique will be presented at different nozzle pressure ratios.
The main intend of the present work is to understand the underlying physics of such flows interacting with open-closed end cavity, more clearly.
This combined measurements can shed more light on the high intensity sound fields thus produced.
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Organization:
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M. Taroudakis, FORTH-IACM & University of Crete |
S. Dosso, University of Victoria |
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Extraction of Interface Wave Dispersion Curves from Ocean Ambient Noise
Hefeng Dong, Guoli Wu, Ganpan Ke
[Abstract]
Interface wave dispersion measurements have been widely
used in geoacoustic inversion to estimate seabed shear-wave
speed profile over the past years. This paper extracts interface
wave dispersion curves from the cross-correlations retrieved
from ocean ambient noise. This technique requires an isotropic
distribution of the noise sources in order to ensure reliable travel-
time estimates obtained from the cross- correlations. However,
real ambient noise fields are often influenced by highly
coherent, directional sources generated by ships, earthquakes
and other human activities, resulting in a bias of the travel-time
estimates. In order to reduce the influence of energetic,
directional sources, an eigenvalue-based approach is proposed
and applied to the eigen-spectrum of the sample covariance
matrix. This approach separates the eigen-spectrum into
different components based on the statistical model of the
covariance matrix and attenuates directional noise and self-
noise related components by a sequential hypothesis tests. This
procedure is applied to a dataset of ocean ambient noise
recorded by a cable. The total recording time is 1.02 hours with
a sampling interval of 2ms. The method shows a clear
improvement compared with classical temporal and spectral
normalization approaches and makes it possible to obtain
reliable dispersion measurements over a short recording period.
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Effect of a soft sediment layer on acoustic normal mode propagation and geoacoustic inversion in the New England Mud Patch
Gopu R Potty, James H Miller
[Abstract]
Previous studies have shown that the travel times of acoustic normal modes
corresponding to the Airy Phase regions were found to be extremely sensitive to shear.
Simple inversion schemes were developed to estimate the shear speed in the ocean
bottom by comparing theoretical predictions with experimental data. Modal dispersion
characteristics of broadband data collected during experiments conducted in Middle
Atlantic Bight and New England Mud patch were analyzed and bottom shear speeds
were estimated. The estimated shear speeds were also compared with shear speeds
calculated from core data. In this study the bottom model will be revised to include a
soft sediment layer over the elastic basement. The modal dispersion will be calculated
using this bottom model corresponding to the New England Mud patch environment
using propagation model ORCA. The effect of the addition of soft mud layer on
dispersion of lower order modes will be investigated and presented. The sensitivity of
modal propagation to the soft mud sediment layer will be explored. [Work supported by
the Office of Naval Research].
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Underwater Acoustic Localization of Pulsed Sources with an Array of Three Hydrophones
Emmanuel Skarsoulis, Despoina Pavlidi
[Abstract]
In connection with the development of an acoustic observatory
for sperm whales off south Crete an array of three moored
hydrophones is considered. It is known that range, depth and
bearing of a pulsed source can be estimated using a pair of
hydrophones by exploiting the time differences between direct
and surface-reflected arrivals. However, that approach suffers
from left-right ambiguity, as well as from increased range/depth
and azimuth uncertainty for source locations close to the broad
side and the endfire of the two hydrophones, respectively. The
addition of a third hydrophone, not aligned with the other two,
removes the left-right ambiguity and can also lead to reduced
uncertainties in range, depth and bearing estimation by
increasing the spatial diversity, associated with the different
orientation of each hydrophone pair. Furthermore, this
configuration allows for range approximation by combining
simple bearing estimates based on time differences of direct
arrivals at the different hydrophone pairs, e.g. in cases where
surface-reflected arrivals cannot be resolved. The presentation
will focus on the benefits from the use of the three-hydrophone
configuration compared to localization with a pair of
hydrophones. [Work partially supported by the Stavros Niarchos
Foundation / ARCHERS project].
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Bayesian Geoacoustic inversion for SBCEX17 Reflection, Dispersion, and Ship-noise Data
Stan Dosso, Charles Holland, Julien Bonnel, Dag Tollefsen, Josee Belcourt, Jan Dettmer, David Knobles
[Abstract]
This paper considers a Bayesian inference approach to model
selection, parameter estimation, and uncertainty quantification
in seabed geoacoustic inversion, with application to data from
the 2017 Seabed Characterization Experiment (SBCEX17). The
approach is based on trans-dimensional (trans-D) sampling
which considers the number of seabed layers as an unknown
parameter in the inversion. Trans-D sampling is carried out with
the reversible jump Markov chain Monte Carlo algorithm, which
constructs a Markov chain that samples from the posterior
probability density (PPD) of the geoacoustic model parameters.
Wide but efficient sampling of the trans-D parameter space is
achieved using principal-component reparameterization and
parallel tempering. The data-error model is based on the
assumption of multivariate Gaussian errors with correlations
represented by an autoregressive process-trans-D sampling of
zeroth- and first-order autoregressive processes is applied to
avoid over- or under-parameterizing the error model. The
Bayesian inference approach is applied to three acoustic data
sets frrom the New England Mud Patch during SBCEX17,
including wide-angle (spherical-wave) reflection coefficients
collected using a moored receiver and towed source;
dispersion of water-borne acoustic modes using a single source
and receiver, resolved by warping analysis; and matched-field
inversion of ship-of-opportunity noise recorded at a bottom-
mounted horizontal receiver array.
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Organization:
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M. Taroudakis, FORTH-IACM & University of Crete |
S. Dosso, University of Victoria |
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Acoustic Signal Characterization using Hidden Markov Models with applications in Acoustical Oceanography.
Costas Smaragdakis, Michael Taroudakis
[Abstract]
The work presents a new probabilistic characterization scheme for acoustic signals with applications in Acoustical Oceanography. This scheme assign the stationary wavelet packet coefficients of the signal, to a representative left-to-right Hidden Markov Model (HMM). In particular, the coefficients of the signal are modeled via a set of Gaussian emission distributions and a matrix of transiting probabilities. The training of the HMM is performed using the Expectation-Maximization algorithm. Similarities measurements between signals then are calculated in terms of the Kullback-Leibler divergence (KLD) by comparing their corresponding HMMs. The applications in Acoustical Oceanography consist of both simulated and experimental data, in all test cases, a single acoustic signal due to a known acoustic source, recorded at a single hydrophone is considered for the retrieval of environmental parameters (ocean acoustic tomography or sea-bed classification applications). The inversion process follows an optimization approach on the basis of KLD, through a Genetic Algorithm. Finally, the results are compared with those obtained using the Statistical Signal Characterization Scheme (SCSS) proposed by Taroudakis et al. [JASA, Vol 119, 1396-1405 (2006)] Moreover, the results corresponding to the real experiment are compared to reference values from the literature.
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Target Detection Method for Reverberant Environments in Continuous-wave Active Sonar System based on Group Multichannel Nonnegative Matrix Factorization
Seokjin Lee
[Abstract]
In the underwater environment, the active sonar system emits an acoustic wave and receives echo signals to detect a target. However, the echo signals consist of not only the echo from the target but also the reverberation from scatterers. If the target has a low speed, the detection problem is more difficult because the target echo is interfered by the reverberation severely. In this paper, the target detection method for the reverberant environment is developed based on multichannel nonnegative matrix factorization. To extract the target echo signal mixed with reverberations, received signal is transformed into bearing-time-frequency domain, and analyzed into bearing, frequency, and temporal basis by the multichannel nonnegative matrix factorization. The simulation was performed to evaluate the proposed method, and the result shows that the proposed method has enhanced detection performance than the conventional methods in the simulated reverberant environments.
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Estimation of bottom parameters using measurement and analysis of sound intensity fluctuations in the presence of moving nonlinear internal waves in shallow water.
Boris Katsnelson, Valery Grigorev, Yanyu Jiang, Yun Ren
[Abstract]
Methodology of estimation of bottom parameters is
proposed
using data processing and analysis of sound intensity
fluctuations due to mode coupling in the presence of
moving
nonlinear internal waves (NIW) in shallow water channel. As
an examples experiments Shallow Water 2006 (Atlantic shelf
of USA) and ASIAEX 2001 (South China sea) are considered.
In these experiments NIW generated in area of the shelf
break
were moving approximately along acoustic track with the
length about a few tens of km. Spectra of intensity
fluctuations of the sound intensity of pulses during a few hours
on vertical line array (VLA) due to mode coupling in both
experiments have specific peaks corresponding to the
difference between wave numbers of coupling pairs of
modes. Analysis of fluctuation spectra includes both
temporal
dependence of fluctuations on separate hydrophones and
temporal dependence of intensity summarized over VLA. It is
shown that geoacoustic inversion in given case can be
achieved using comparison of experimental spectral and
temporal dependencies with results of theoretical modeling,
containing bottom characteristics (sound speed, density and
attenuation) as a fitting parameters.
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Method to determine far-field beampattern of long array from subarray beampattern measurements
Donghwan Jung, Jeasoo Kim, Hongju Gu, Sungyong Moon
[Abstract]
Beampattern measurement is essential to verify the performance of
array sonar. However, the classic problem encountered in
beampattern measurement of arrays is that the far-field conditions
for reaching the plane wave are limited due to reasons such as the
measurement space. We aim to measure the beampattern of the
array in a limited space and to evaluate the performance through the
-3dB beam width. Therefore, we propose a method to measure the
beampattern of discrete line array in limited space. The proposed
method is based on the sub-array method. A discrete line array
whose measurement space does not satisfy the far-field condition is
divided into several sub-arrays satisfying the far-field condition, and
the beampattern of the line array is derived through the beampattern
for the sub-array. The proposed method was verified by simulation
and we performed a measurement experiment on a line array of 256
elements with a design frequency of 455 kHz. This method can
measure the beampattern of the line array simply as compared with
the method of converting the beampattern obtained from the
measurement data in the near-field to the far-field.
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Organization:
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J. Abshagen, WTD 71 |
C. Zerbs, Müller-BBM GmbH |
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Towards a Realistic Approach of Sound Propagation in EIAs: How to Represent Moving Sources and Moving Receivers
Uwe Stöber, Lars O. Mortensen, Frank Thomsen
[Abstract]
Recent guidelines for marine mammal protection like those in the US or in Denmark impose dose-based criteria that require to accumulate the noise received by a marine mammal over time. Compared to a static receiver, fleeing animals receive a significantly lower noise dose because they move away from the zone of noise impacts. Animal movement models can therefore be useful to predict more realistic noise impacts on marine mammals. In addition, noise sources are not always static. During seismic exploration for example, air guns are towed behind a vessel and the locations of the airgun sounds are changing with the forward movement of the vessel. Here we present a feasible solution for noise impact assessments that compromises between realistic representations of mammal behavior and sound field on the one hand and computational resources on the other hand to provide the best possible accuracy in a realistic time frame. The approach consists of two steps: First, the sound field is derived from a number of detailed propagation simulations. Then, an Agent Based Model (ABM) simulates animal movement and noise dose accumulation. Our study shows that noise impact assessments benefit from using a more dynamic approach.
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A Computational Investigation into the Influence of the Shear Properties of the Seabed on Sound Propagation in Shallow Water.
Ray Kirby, Wenbo Duan
[Abstract]
The propagation of sound waves in the ocean is influenced
by the acoustic characteristics of the seabed, especially in
shallow water. If the seabed is considered to be an elastic or
viscoelastic structure, then sound propagation will be
influenced by the shear and longitudinal properties of this
structure. Accordingly, this paper examines the sensitivity of
sound propagation to the shear properties of the ocean
floor for shallow waters and when the sound source lies in
the ocean. This is motivated by predictions reported in the
structural mechanics literature, which demonstrate that
wave propagation in viscoelastic structures can be highly
sensitive to the shear properties of the material. This
investigation uses the semi analytic finite element method to
undertake a set of numerical experiments. This numerical
approach provides a computationally efficient way of
solving the fully coupled acoustic problem, with Bergmann’s
equation used for the fluid, and Navier’s elastodynamic
wave equation for the seabed. Both elastic and viscoelastic
structures are examined and predictions are reported for a
range of shear properties, and the relative sensitivity of
sound propagation on these shear properties is then
investigated.
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Three-dimensional acoustic parabolic equation model based on GPU processing
Keunhwa Lee, Woojae Seong, Youngnam Na
[Abstract]
We introduce a three-dimensional acoustic parabolic equation (PE) model based on GPU processing. This model is developed by splitting the full exponential operator in the three-dimensional split-step marching solution in the Cartesian coordinate. Each exponential operator is approximated by the Taylor series and the sum representation of Pade approximation, which gives an opportunity for GPU parallel processing of the 3D PE solver. An advantage of this model is to implement higher-order cross-terms appearing to the operator approximation efficiently. In the talk, we show the computational performance of GPU based PE code with the comparison of a traditional code for several ocean environments.
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The Analysis of Parabolic Equation Model Solutions using Split-Step/ Finite Difference Method
Mustafa Aslan
[Abstract]
Parabolic equation models solved using the split-step
Fourier
(SSF) algorithm are commonly studied for the modeling of
acoustic wave
propagation in the underwater environment. The
smoothing
function, in the solution of realistic problems, is introduced
in SSF
algorithm to treat the density change between the water
column and the sediment. The algorithm with the
smoothing
approach accumulates the phase error with range. As an
alternative method, by Yevick and Thomson, the hybrid
split-
step/finite difference method was developed to numerically
accomplish the density discontinuity issue and significantly
improved the phase error. In the method, the additional
operator containing density dependent terms are
introduced to
implement the hybrid approach. In this work, the results of
SSF
parabolic equation model with the smoothing and the
hybrid
approach are compared and are discussed the effect of
numerical parameters on the model implementation.
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Parallel implementation for Three-Dimensional Acoustic Field Computation in a Penetrable Wedge by Image Source Method
Wenbin Xiao, Yongxian Wang, Wei Liu, Qiang Lan, Xinghua Cheng, Zijie Zhu, Xin Wang, Ben Luo, Dezhi Wang, Jiani Wu, Lilun Zhang
[Abstract]
Present work aims to the open multi-processing (OpenMP)
parallel implementation for the three-dimensional acoustic field
computation in a penetrable wedge. Possible practical
application of the wedge waveguide propagation is extended
to the continental shoulder or shallow water environment. By the
source image method, the acoustical propagation solution is in
conjunction with an extension of reflection analysis of spherical
wave filed from a planar interface. Multiple reflections at the
boundaries are taken as the originating from a source image
depended on the surface and bottom positions. Analytical
expression of acoustical pressure here is coupled with the Bessel
function expansion in complex form. Two range-dependent
cases, such as the along the slope (two dimension) and cross
the slope (three dimension), are tested by the OpenMP
parallelization and performance optimization. The multi-core
platform of ”Knights Landing” or KNL and many-core platform of
”TianHe-II” are applied to the utilization. The performance gain
observed in using OpenMP of the Fortran code on high
performance computing facilities is demonstrated.
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Sound propagation experiments in a Norwegian fjord
Jan Abshagen, Volkmar Nejedl
[Abstract]
Vertical stratification of temperature and salinity along with the increase of hydrostatic pressure with depth result in the formation of sound channels in the ocean as well as in deep fjords. A sound channel may form in the surface layer but also in a deep water layer with its acoustical axis being located below the main thermocline. Due to fresh water inflow, mixing, or tidal currents, sound propagation conditions may vary significantly with time but also display substantial variations between different locations, in particular in a fjord. We have performed experiments on sound propagation with RV Elisabeth Mann Borgese in the Norwegian Sognefjord. The measurements were performed with a freely drifting projector buoy and a vertical receiver array, as well as a towed horizontal receiver array in a deep and in a surface channel, respectively. In order to determine the directional behavior of the sound field, beamforming techniques were applied. This work addresses frequency-dependent propagation losses, as well as the spatial, temporal, and directional variability of sound channel propagation in confined waters.
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An Underwater Vehicle Shape with Reduced Acoustic Backscatter
Tom Avsic
[Abstract]
In some underwater applications, acoustic backscatter might have undesired consequences, such as interferences in underwater communications or echoes in the sonar image. Acoustic backscatter of an underwater object is classically reduced by applying appropriate absorbing materials. Absorbing materials convert the acoustic energy into heat and their thickness is usually of the order of a quarter wavelength. In some cases, the application of absorbing materials might have unacceptable drawbacks on other design criteria. For long-distance acoustic propagation (acoustic channel propagation), it can be shown that undisturbed sound travels nearly horizontally within a limited vertical angle of not more than ± 20∘ from the horizontal plane. Hence, for long-distance applications, the acoustic backscatter can be reduced by proper shaping of the vehicle’s lateral outer hull. In case the outer hull is not opaque enough to shield the vehicle’s inner structures (with potentially high backscatter), reflective material can be applied. The working mechanism of reflecting materials is the impedance mismatch and hence, reflecting materials can be much thinner than absorbing materials. This conference contribution will address the worldwide sound spreading in the ocean and show the guidelines for geometric outer hull shaping.
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Development of Underwater Acoustic Transducers
Yang Liu, Houlin Fang, Liangyong Zhang, Fang Zhang, Deyu Sun
[Abstract]
Underwater acoustic transducer which undertakes the task of signal generation and reception is the core equipment for sonar detection. For understanding the mechanism of underwater acoustic transducers, we have reviewed recent typical transducers based on different working principles and different vibration modes, including flextensional transducers, cylindrical radiating transducers, longitudinal transducers and so on. Our research is of great importance for obtaining underwater acoustic transducer with higher performance.
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Analysis of Displacement Amplification Characteristics of Class IV Flextensional Shell Based on Elliptic Perimeter Approximation Formula
Houlin Fang, Difeng Sun, Fang Zhang, Yang Liu, Tianqing Zhao, Cheng Zhang, Liangyong Zhang, Xubin Liang, Deyu Sun
[Abstract]
As the most widely used underwater acoustic transducer, class
IV flextensional transducer can get larger amplitude flexural
vibration from longitudinal vibration of driven stack by the
displacement amplification effect, thereby radiating more
energy. In order to further quantitatively understand the
displacement amplification characteristics of elliptical tube
shell, this paper proposed to calculate the displacement
magnification using the ellipse perimeter approximation
formula, and the variation law of the displacement
amplification with the compression coefficient and the
deformation rate was obtained, which was analysed and
compared with the results by the method of finite element
analysis. The empirical formula for calculating the finite element
results by the results of the perimeter approximation formula was
got by fitting curves. When the compression coefficient is
0.24∼0.80, the error of the calculation results was not more than
6.11%, and finally the preliminary verification was carried out by
experiments. The research of this paper has certain reference
significance for the estimation of the radiation performance of
low frequency transducers.
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Passive Ship Localization in a Shallow Water Using Pre-trained Deep Learning Networks
Dezhi Wang, Lilun Zhang, Changchun Bao, Shuqing Ma, Yongxian Wang
[Abstract]
Subject to the lack of detailed environmental information, the
classical matched-field processing (MFP) may not be adapted
to the accurate localization of ocean acoustic sources. In this
paper, a framework that applies deep learning techniques
instead of the conventional MFP method is presented for the
localization of ship acoustic sources in a shallow water
environment. The original data in terms of sound pressure is
recorded from a vertical array placed in the costal waters. The
acquired data is converted into normalized sample
covariance matrices (SCMs), which are used as input data fed
into the deep-learning architecture. In particular, a framework
is proposed to predict the range information of a target ship in
a transfer learning manner based on the pre-trained Xception
model that is the state-of-the-art convolutional neural
networks. The proposed method achieves a performance up
to 10 km at range prediction, which is significantly better than
that of conventional MFP method and some other shallow
machine learning methods. Different from the MFP approach,
machine-learning methods are driven directly by the data,
which offers an opportunity to enable them to overcome the
environmental mismatch problem.
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Organization:
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A. Homm, Wehrtechnische Dienststelle für Schiffe und Marinewaffen (WTD71) |
V.F. Humphrey, ISVR, University of Southampton |
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Innovative solutions to reduce the transfer of structure borne noise in couplings
Lothar Kurtze
[Abstract]
Modern couplings for ships and especially for mega yachts
have to fulfill various tasks. Located between gearbox and
propeller shaft, they have to minimize the transfer of structure
borne noise while transmitting high torque levels. In addition,
they have to resist high misalignments due to very softly
mounted frames of ship engine and gearbox. Couplings also
have to deal with sometimes difficult torsional vibration
situations of the drivetrains, they have to be lightweight and in
case of electric motors they often need electric insulation
properties.
To fulfill all these partly contradictory demands, innovative
approaches are essential. In this paper, the development of a
lightweight coupling made of a combination of glass and
carbon fibers, elastomer layers and steel will be presented. After
the design phase, the couplings have been tested at an
acoustic test bed. In the next step they have been tested
together with an Advanced Electric Drive (4 MW-range). And
finally, the new Geislinger Silenco®-coupling has been validated
during sea trials. Thus, the coupling helps significantly to reduce
the noise radiation. This includes the noise reduction in the ship
cabins as well as the noise reduction into the environment.
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Application of the Energy based Finite-Element-Method to determine the sound emission of vibrating ship structures in the High Frequency Domain
Henning Lohmann, Boris Dilba, Olgierd Zaleski, Otto Von Estorff
[Abstract]
For large scale structures the applicability of classical numerical
methods such as Finite-Element-Method (FEM) or Boundary-
Element-Method (BEM) is restricted to the low and mid
frequency range. The fine mesh resolution for higher frequencies
rapidly leads to problem sizes which exceed given
computational resources. Here, the energy based Finite-
Element-Method (EFEM) provides an efficient solution.
The EFEM is based on balancing input, output and dissipated
powers over a specific control element and is applicable to fluid
and structure problems. Exchange of energy between elements
is described by transmission coefficients. The unknowns of the
EFEM are energy densities which aren’t oscillatory in nature.
Thus, the required discretization is independent of the frequency
and results in a smaller system of equations compared to
classical methods.
In this contribution the application of the EFEM to a ship is
outlined. After a brief introduction to theoretical aspects the
focus will be on the coupling of fluid and structure problems
based on the radiation efficiency. In addition to determining the
radiation efficiency the influence of the coupling parameters on
the resulting system of equations is shown and utilized to define
a lower frequency bound for the application of the EFEM to
vibrating ship structures.
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Analysis Methods and Design Measures for the Reduction of Noise and Vibration Induced by Marine Propellers
Julian Kimmerl, Paul Mertes, Vladimir Krasilnikov, Kourosh Koushan, Luca Savio, Mario Felli, Moustafa Abdel-Maksoud, Ulf Göttsche, Nils Reichstein
[Abstract]
The paper presents the ongoing work in the research project ProNoVi. The objective of the project is to improve the numerical and experimental methods for the prediction of noise and vibrations induced by a propeller operating behind ship hull in full-scale conditions. Based on the improved methods practical recommendations for the reduction of noise and vibration levels for different classes of vessels are a further objective. The propeller cavitation noise is identified as the dominating noise source, which coincides with important frequencies of perception of marine fauna and thus may have negative impact on marine life. As a side effect, reducing the noise under water will increase comfort and thus safety for crew and passengers on board. The project aims at delivering a better understanding of fundamental physical mechanisms related to turbulence, induced vorticity and cavitation dynamics, which play a decisive role in generation of tonal and broadband propeller noise. The project partners will give a joint presentation, focusing on state-of-the-art in the field, today’s challenges in both the experimental and numerical approaches, and how the ProNoVi project is addressing those. ProNoVi is funded by the European Union. Further information is available at www.martera.eu/projects/pronovi.
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Numerical modelling of the correction factor for predicting the monopole source level from shallow water noise measurements on vessels
Victor F Humphrey, Yin Cen
[Abstract]
Environmental issues have led to a growing interest in the measurement of the radiated underwater noise from surface vessels. ANSI and ISO have supported this by issuing standards for the measurement of the Radiated Noise Level of vessels in deep water. In addition, a draft standard for the conversion of such measurements into the Monopole Source Level, required as an input to environmental noise models, is at an advanced stage. However, many operators and investigators are constrained to making measurements in shallow water where both sea-surface and seabed effects are important. A numerical study has been performed in order to understand the variability of the correction factor required to convert measurements of received pressure into the equivalent Monopole Source Level in such an environment. These calculations have been performed using an image source model and have included the effects of reflection at both the sea-surface and an attenuating seabed. The influences of the water depth, seabed type, distance of closest approach, and number and location of the hydrophones have been investigated. The results show how the correction factor varies and indicate the extent to which simplified approximations can be used for the correction factor.
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Measurements of Underwater Noise from Pile Driving in Southwest Coast of Korea
Dong-Gyun Han, Daehyeok Lee, Jee Woong Choi
[Abstract]
The interest in underwater noise produced by impact pile driving
has been increasing worldwide. Since the impact pile driving
noise with extremely high sound pressure level can cause
negative effects on marine ecosystem, it is important to
measure the noise and to assess possible risks. Recently, offshore
wind power generation is being actively developed in Korea
and accordingly efforts to assess the environmental impact of
pile driving noise are being made. Measurements of pile driving
noise were carried out twice in the southwest coast of Korea in
2017 and 2018. The variation of the impact pile driving noise with
the number of strikes was analyzed for the noise measured at a
single site. In addition, impact pile driving noise were measured
as a function of source-receiver range, which was compared to
the theoretical transmission loss curves predicted by a damped
cylindrical spreading model and the parabolic equation-based
range dependent acoustic model (RAM).
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Shipping noise propagation at the shallow sea
Eugeniusz Kozaczka, Grażyna Grelowska
[Abstract]
The issue of rapid noise growth in the oceans and seas is
currently a challenge for research teams dealing with
underwater pollution. At present, for the EU water areas, a
directive has been developed, which is intended to cause a
reduction of acoustic background, the main component of
which is underwater noise mainly related to maritime transport.
The paper proposes a method to determine the noise map on
the shallow sea basin based on the classical solution of the
wave equation for specific boundary conditions. As a result, we
get a solution consisting of a superposition of analytical solutions
for the issue of own waves related to the depth in relation to
changes in the distance to source.
For a stationary problem, the expression describing the noise of
the selected vessel in spectral form will be used as a function of
excitation. This formulated issue allows us to estimate the
underwater noise level at any distance and depth from the
source.
The analysis of the superposition of underwater acoustic
disturbances makes it possible to determine the value of the
sound pressure level from the sources of these disturbances.
Details of the full procedure will be presented in the
presentation.
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Noise mitigation for the construction of offshore wind turbines – Overview of Noise Mitigation Systems for complying with noise limits.
Michael A. Bellmann, Robert Kühler, Rainer Matuschek, Patrick Remmers, Siegfried Gündert, Michael Müller, Jan Schuckenbrock
[Abstract]
Underwater noise caused by impact pile-driving during the installation of offshore foundations is potentially harmful to marine life. Therefore, in several nations (e. g. Germany, The Netherlands, Denmark, Taiwan, etc.), noise limiting values were developed to prevent injury of marine life. The experience over the last years shows, that underwater noise produced during pile-driving, depending on many parameters and measurements, and shows values, which exceed the national underwater noise limiting value by up to 20 dB. Therefore, Noise Mitigation Systems (NMS) are required to significantly minimize the underwater noise. Therefore, many different noise mitigation systems were applied during the last years during installation of Offshore foundations and were accompanied by underwater noise measurements in accordance to the ISO 18406. In this paper, a general overview of the existing and tested NMS, including various tested system configurations as well as combinations of different NMS, will be presented. Additionally, the achieved overall resultant noise reductions during pile driving as well as the factors influencing the noise reduction will be discussed. Moreover, it will be explored what implications the development towards larger foundations can have for the state of the art in noise mitigation.
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Uncertain Parameters of the Propagation Path in 3D Pile Driving Noise Modelling
Jonas Von Pein, Johannes Seidel, Elin Klages, Stephan Lippert, Otto Von Estorff
[Abstract]
Nowadays, offshore constructions are mainly mounted to the sea bed by piles. These piles are driven into the sea floor by percussive pile driving. This process leads to high underwater sound pressure levels. Therefore, limiting values have been set in many countries for the protection of the marine fauna. In order to get a profound prognosis of the emitted sound levels, numerical models are essential. These models are mainly rotationally symmetric and based on a finite-element formulation for the close vicinity of the pile and a far-field propagation model for ranges further away than 1 km. In the current contribution the far-field model is a full 3D parabolic equation model. A major problem in pile driving noise radiation modelling is to derive a reliable description of the acoustical properties of the sea bottom. However, due to the inhomogeneous character of the soil these parameters are always subjected to variability. To quantify the effects of this uncertainty in the acoustical soil parameters, a sensitivity analysis is performed for a pile driving scenario assuming a heavily changing bathymetry with and without noise mitigation. Furthermore, the sensitivity analysis is performed with Nx2D and 3D simulations.
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Evaluation of range standards for underwater radiated noise of ship’s in beam aspect.
Hans Hasenpflug, Anton Homm, Layton Gilroy, Stefan Schäl
[Abstract]
The underwater radiated noise levels of a small naval vessel were measured at the Heggernes deep water sound range. The data evaluation and analysis was performed in accordance with the NATO standard STANAG 1136 and the ISO 17208-1 standard. Both standards describe procedures for the determination of underwater radiated noise levels of ships in beam aspect on a deep water sound range.
To assess and compare the results of both standards, the hydrophone configuration at the Heggernes range was utilized at various depth settings. Measurements were made of the ship in two different ship conditions (machinery states) and a variety of sailing speeds. Some of these measurements were repeated a large number of times. The resulting 3rd octave based spectra are used to compare the two procedures.
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Structure-borne Noise Reduction of Gearboxes in Maritime Application
Robin Daniel Seiler
[Abstract]
The reduction of vibration and noise levels on board ships
offers
not only occupational safety for the crew, but also increased
comfort for passengers and less impact on the environment.
In
order to meet predefined noise and vibration requirements,
an
effective noise control concept must integrate a variety of
aspects and includes several coordinated counter measures.
The foundation for this is laid in the engine room, because the
excitement of the ship’s structure in the engine room
decisively
shapes the vibro-acoustic situation on board. Here you can
”gain” the most decibels. Single or double-elastic suspensions
for
ship engines already show high transmission losses, when
rubber
mounts are used. As a result, the structure-borne noise via
gearboxes becomes a crucial sound transmission path.
Against this background, the central role of gearboxes in
powertrains of ships is investigated. The presented
countermeasures include different drive train configurations:
1. Highly flexible couplings with low torsional stiffness
between
engine and gearbox,
2. Propeller shaft couplings in combination with very soft
gearbox mounting systems,
3. Active vibration controls to reduce velocity amplitudes
on the gearbox foundations.
The effects of the outlined solutions on the
reduction of airborne noise levels in receiving rooms on board
are discussed.
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Organization:
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K. Sepahvand, Technical University of Munich (TUM) |
P. Langer, Technical University of Munich (TUM) |
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Reliability of experimentally determined damping values
Christian A. Geweth, Simon Boche, Kheirollah Sepahvand, Steffen Marburg
[Abstract]
In order to increase the reliability and precision of numerical
models for vibro-acoustic problems, it is often still necessary
to extract and validate input parameters by performing
experiments. Among the frequently needed input
parameters for finite element models, damping is one of the
most difficult ones to acquire. One of the reasons for those
difficulties is that the boundary conditions commonly used for
numerical models, e.g. free-free support or ideally clamped
are not precisely feasible in an experimental setup but are
mere approximations. Since the utilized type of support can
affect the obtained experimental damping values, it can be
necessary to pay close attention to the design of the
experimental setup. The focus of this study is set on the influence of different types
of support on experimentally obtained damping values. The
structural dynamical response of the used test specimen with
different types of support on an impulse excitation is
measured by a laser scanning vibrometer. During post-
processing, the damping values are derived from the
measurements and are further analyzed.
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Lumped parameter model and Monte Carlo Simulation to study middle ear uncertainties
Julio A. Cordioli, Lucas Lobato, Stephan Paul
[Abstract]
Over the past 70 years several deterministic mathematical models of the human middle
ear have been developed for assessing its physiological process, study surgical
techniques and to design hearing devices. However, experimental data have shown a
large uncertain concerning the dynamic response of the middle ear over different
subjects. This work aims to present a preliminary study on the middle ear uncertainties
evaluating its dynamics through statistical tools. Our hypothesis for the dynamic
behavior dispersion observed experimentally is based on the mechanical parameters
variability among subjects. A 1D lumped parameter model of the human middle ear is
combined with Monte Carlo Simulations for a probabilistic assessment. It was observed
that the confidence interval of the natural frequencies rises with the mode order, where
the standard deviation reaches more than 10% of the mean for the third mode. In
addition, the third mode has the greater number of outliers in the boxplot analysis. Also,
while the frequency response of the malleus shows the greater confidence interval
above 1 [kHz], the incus frequency response presents the smaller dispersion. At last, our
results suggest the large dispersion seen at low frequency in the experimental data is not
caused by the mechanical parameter variation.
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Organization:
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M. Gaborit, LAUM UMR CNRS 6613, Le Mans Université, France / MWL, KTH, Stockholm, Sweden |
L. Jaouen, Matelys |
P.R. Andersen, The Technical University of Denmark |
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Uncertain acoustic meta-atoms
Felix Kronowetter, Steffen Marburg
[Abstract]
Acoustic metamaterials (AMMs) consist of periodic arrangements of single meta-
atoms (e.g. Brillouin zones). The AMMs can manipulate the acoustic wave
propagation in ways that are not found in nature or conventional materials.
Furthermore, AMMs can have unnatural material properties such as a negative
effective mass or band gaps. One type of meta-atom is based on the principle of
a Helmholtz resonator that is embedded in a fluid matrix. A periodic arrange-
ment of such meta-atoms in the two dimensional space combines the effects of
a resonator and those of phononic crystals. The effectiveness of that kind of
AMM depends on the eigenfrequencies of the resonators and the relative posi-
tion of one meta-atom to one another. Since the production of AMMs is linked
to manufacturing tolerances the perfect periodicity is not fulfilled and can affect
the properties of the AMM.
This work deals with the uncertainties of the meta-atoms concerning the geometry of the embedded resonator. The uncertain geometry parameters are
approximated by spectral expansions. Further, the transfer function and the insertion loss with respect
to the uncertain parameters are analyzed. Finally, the results of the spectral
approach are compared to those of the Monte Carlo method.
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Modeling sound transmission through a periodic acoustic metamaterial grating of finite size
Xiang Yu, Fangsen Cui, Wei Hin Mark Wong
[Abstract]
This study investigates into the sound transmission through a periodic acoustic
metamaterial grating of finite size. The single-layer grating is constructed by
periodically arranging sub-wavelength unit cells in a slab, as a part of a large baffle
between two acoustic domains. The metamaterial unit cell consists of an open duct
decorated with coiled resonators, which intends to suppress sound transmission using its
acoustic stop-band. The space-coiling structure allows the metamaterial to operate at
low frequency with a compact size. Analytical approach to predict the sound
transmission loss of the combined baffle is developed, which is employed to study its
sound attenuation under normal and oblique incidence. Results show that the unit cell
geometry, the periodicity of grating elements, and the angle of incidence significantly
affect the STL. The STL behavior of the acoustic grating is mainly governed by three
physical effects, including the acoustic stop-band, the edge diffraction, and the
destructive radiation interference behind the acoustic grating. Detailed mechanism
studies for these effects are discussed using numerical examples, and experiment is
conducted to validate the proposed numerical approach.
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Development of a metamaterial for acoustic and architectonical improvement of window design.
Gioia Fusaro, Xiang Yu, Fangsen Cui, Jian Kang
[Abstract]
The development of windows which can address the issue of
high noise levels while maintaining natural ventilation has drawn
considerably attentions recently. However, there are limitations
to traditional double glazing together with duct designs. In our
previous studies window systems have been developed based
on the local resonant stopband of acoustic metamaterial
(AMM) to achieve dual functions of noise reduction and natural
ventilation. In this study, further effort is made to develop
window designs based on origami concept, and also with a
different approach for ventilation design. Firstly, such window
designs are created using parametric modelling technique. The
sound reduction performance is then investigated using the
Finite Elements Method (FEM), and the sound reduction index
(SRI) is evaluated. The performance is further optimised using
parametric FEM analysis. The computational platform
developed can also facilitate the optimisation of window
designs.
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Prediction of oblique incidence sound absorption coefficient for microfiber sound absorber
Tatsuhiro Komito, Senji Kitahara, Toshimitsu Tanaka
[Abstract]
Automotive parts are required to reduce weight and size, and such the task is also necessary for sound absorbers as well. So the microfiber sound absorber with high performance whose fiber diameter is several hundreds nanometers or several micrometers is being used widely. On the other hand, it is important to develop techniques for estimating sound absorbing performance when we design the structure of microfiber sound absorber.
However, preceding research is not for estimating oblique incidence sound absorbing coefficient but for estimating normal incidence sound absorbing coefficient. This paper presents numerical calculation and experiment for oblique incidence sound absorbing coefficient of microfiber sound absorber in the free sound field, and refers to consideration on the structure of microfiber sound absorber.
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Off-line Envelope Estimation for Acoustic Screens with Uncertain Properties
Mathieu Gaborit, Olivier Dazel, Peter Göransson
[Abstract]
The development of new solutions to reduce impact of noise receives constant attention both from the research and engineering communities. Even though the proposed systems tend to diversify, multi-layered absorbers based on foams remain the most common. Such structures feature one or several thick slabs of foam and are completed by protective or tuning layers of small thickness inserted between slabs or on the free surface. These layers (acoustics screens or films) bring a number of challenges. Firstly, their thickness makes them hard to characterise and some parameters are uncertain. Secondly, the bonding to the surrounding system alter the effective properties. Given the strong influence the films have on the system’s response, it appears important to account for these uncertainties from the design phase on.
The present contribution introduces a new technique to efficiently account for uncertainty linked to the screens. More precisely, the proposed approach is based on separating the average film influence from the deviation due to uncertainties. This allows to estimate the response envelope using a pre-computed average response and the properties. The results computed for a range of typical and challenging configurations are promising and validate the robustness of the method at any angle of incidence.
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Double Reflections from corrugated surfaces
Jorge Petrosino, Nicolas Casais Dassie, Damian Andres Fernández, Georgina Alejandra Lizaso
[Abstract]
Reflections of acoustic waves from corrugated surfaces are discussed.
A proposal made by Yi-Fan Zhu for a metasurface model considers
that each groove on a corrugated surface is analogous to each
individual source in a line array, where the phase difference between
sources is determined by the depth of each groove. Zhu’s original
proposal suggests using the phase gradient of the arrangement as a
design parameter, and derives conditions for reflections without
dispersion. In this paper, we expose the advantages of considering
time-delay gradient instead of phase gradient. Also, designs that
combine two corrugated surfaces interspersed with different gradients
to produce double reflection patterns are presented. The angles and
delays of both reflections can be selected independently.
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Finite element analysis on the surface characteristics of acoustic resonators with thermal and viscous boundary layers
Naohisa Inoue, Tetsuya Sakuma
[Abstract]
The thermal and viscous boundary layers (TVBL) around a
resonator’s neck play important roles in the absorption,
particularly when there is no porous material near the neck.
This paper presents a simple implementation of a TVBL
boundary condition for the three-dimensional finite element
analysis of Helmholtz equation. The implementation is
verified by comparing normal incidence absorption
coefficients calculated and measured for slit-type resonators.
Good agreement can be observed under the appropriate
calculation setting. Subsequently, a detailed investigation is
conducted on surface impedances of Helmholtz resonators
composed of circular holes and square baffles. Numerical
values are compared with analytical values based on the
classical models such as Maa, Allard-Ingard and so on. It is
demonstrated that the presented method convincingly
captures the surface impedance on an acoustic resonator
unless the TVBL thicknesses are much larger than the
perforation diameter.
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Organization:
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H. Waubke, Acoustics Research Institute, AAS |
W. Kreuzer, Acoustics Research Institute, Austrian Academy of Sciences |
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Numerical Investigation of Indirect Combustion Noise Mechanisms in a Nozzle
Thomas Deconinck, Antoine Vallon, Alexis Appel De Gardane, Yves Detandt, César Legendre
[Abstract]
The entropy and vorticity disturbances generated in the combustion chamber of an aero-engine can be converted into pressure fluctuations while accelerated in the turbine stages : this phenomena is known as Indirect Noise. The mean flow plays a significant role in this process and should be considered in the modeling. As a first step, the present paper focus on modelizing the interaction of entropy and acoustic waves in a nozzle. First, the mean flow inside the nozzle is estimated by means of a steady-state CFD simulation. Second, the wave propagation including the mean flow effects is computed by means of a high-order adaptive Discontinuous Galerkin (DG) scheme in time domain solving the linearized Euler equations (LEE). This scheme is chosen due to its ability to accurately represent the phenomena involved and its high parallel scalability for high frequency problems. The studied cases consist of a convergent-divergent nozzle configuration where entropy waves are injected upstream by means of strong heat pulses. Three nozzle test conditions are investigated: subsonic; sonic and supersonic. Finally, the numerical solutions are compared with available experimental data and analytical models showing good agreement.
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Air Flow Computations for Helmholtz Resonators in a Sound Field
Norbert Gorenflo, Tobias Merkel, Jonas Stein
[Abstract]
We present computations of the air flow near the neck of a Helmholtz resonator which is located in a time-harmonic sound field. Of particular interest are sound frequencies near the resonance frequency of the resonator, where the flow speed is maximal. Besides BEM- and FEM-calculations for the resulting boundary value problem for the Helmholtz equation we have performed also fluid dynamical FEM-computations based on the Navier-Stokes equations and the Bernoulli equation, respectively. In the fluid dynamical case, small amplitudes of the initial values cause only weak nonlinear effects. Thus, if such small initial values are taken from the corresponding solution of the Helmholtz equation, this time-harmonic solution should be similar to the fluid dynamical solutions. This can be used to validate the fluid dynamical solutions. Basically we are interested in excitations of the Helmholtz resonator with greater amplitudes and the resulting different behaviour between the inflow and the outflow at the resonator neck.
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The Time-Explicit Nodal Discontinuous Galerkin Method Applied to Acoustic-Structure Interaction Problems
Kirill Shaposhnikov, Mads J. Herring Jensen, Elin Svensson
[Abstract]
The time-explicit nodal discontinuous Galerkin (dG) method has gained a good reputation as a numerical approach to solving first-order hyperbolic problems for its high accuracy and low memory consumption, great scalability on clusters, and ability to operate on unstructured as well as nonconforming meshes. The method has been proven to be efficient for simulation of transient acoustic phenomena in acoustically large computation domains. Application areas include ultrasound transducers, nondestructive testing, geophysics, etc. In this work, we apply the dG method to study wave propagation in coupled linear elastic and acoustic media. The velocity-strain formulation is used for the structural part of the system, and the velocity-pressure formulation, for the acoustic part. In the three-dimensional space, this results in nine and four coupled hyperbolic equations for the structural and the acoustic part, respectively. The consistent and accurate coupling between the acoustic and structural domains is ensured through the proper upwind flux across the interface. The proposed approach is validated by comparison with results obtained for the second-order time-implicit finite-element formulation of the acoustic-structure system. The effectiveness of the time-explicit dG scheme is demonstrated by its application to large-scale wave propagation problems in coupled acoustic-elastic media.
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Fast Algorithms Applied to the Acoustical Energy Boundary Element Method
Matthias Ram, Otto Von Estorff
[Abstract]
The Boundary Element Method (BEM) is a well-established method for solving acoustic problems. Since the number of degrees of freedom of the discretization depends on both, the geometry and the frequency range of the problem, the memory requirement to store the fully populated system matrix becomes very large for the calculation of high-frequency problems. The area of application is thus limited to low and medium frequencies. The Energy Boundary Element Method (EBEM) represents an approach for tackling high frequency problems. The transition to non-phased energetic state variables removes the correlation between the investigated frequency range and the element size. In the EBEM the number of degrees of freedom is thus determined primarily by the complexity of the geometry.
In order to speed up the computation and to handle structures of very high geometric complexity, a fast multipole algorithm for the EBEM is investigated. It has a great potential to reduce the numerical effort in the BEM. Helmholtz and Laplace problems have been solved very efficiently applying this algorithm. However, the kernels used in the EBEM require an adaptation of the algorithm. In this paper the development of a fast multipole formulation of the EBEM is presented.
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Efficient Evaluation of Flow Induced Sound Sources at Low Frequency by Fast Multipole BEM
Takayuki Masumoto, Yosuke Yasuda, Naohisa Inoue, Tetsuya Sakuma
[Abstract]
Recently, the fast multipole BEM (FMBEM), a highly efficient BEM to which the fast multipole method (FMM) is applied, has been widely used for large DOF acoustic problems. In the flow induced noise calculation based on the Lighthill’s analogy, however, the processing time for source evaluation is remarkably large due to the large number of sound sources, even if FMBEM is used. This markedly impairs the overall computational efficiency. To solve this problem, we have studied on application of the high-frequency FMBEM (HF-FMBEM) for the source evaluation. In this study, the hierarchical cell structure for grouping boundary elements was expanded to a larger structure for grouping sound sources widely distributed outside the boundary. However, since the contributions from quadrupole sources were expanded with Rokhlin’s diagonal form, this method was unstable at low frequencies. In this paper, we propose a stable method at low frequencies using low-frequency FMBEM (LF-FMBEM), which is based on the original multipole expansion theory. We also show its effect on the computational load through a numerical example: a flow induced noise analysis around a side view mirror, where the number of elements and sources are about six thousand and three million, respectively.
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Efficient Evaluation of Sound Radiation of an Electric Motor Using Model Order Reduction
Martin Eser, Caglar Gürbüz, Lennart Moheit, Marold Moosrainer, Steffen Marburg
[Abstract]
Electric motors can suffer from severe noise radiation when the internal electromagnetic forces excite a natural vibration mode of the stator pack. These critical operation points can be identified from a Campbell diagram. The generation of the diagram based on numerical analyses is computationally very costly, in particular when the motor is to be analyzed in the mid-frequency range.
The complex vibration behavior of the motor housing prohibits a straight-forward application of common model reduction processes. Therefore a new multiple-input-multiple-output model reduction process is developed using the moment matching method based on Krylov subspaces. The in-vacou surface velocity modes of the motor housing yield as input load patterns. A structure-preserving second-order Arnoldi algorithm is used for the calculation of the Krylov subspaces. The proposed process enables the direct evaluation of the radiated sound power for all considered frequency points within a frequency band. The reduction process, incorporating an adaptive selection of structural modes and an optimized parallelization of subspace calculations, allows for an efficient evaluation of the radiated sound power of the electric motor.
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Organization:
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H. Waubke, Acoustics Research Institute, AAS |
W. Kreuzer, Acoustics Research Institute, Austrian Academy of Sciences |
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Model Order Reduction with Krylov Subspaces of exterior Acoustic BEM Systems
Dionysios Panagiotopoulos, Elke Deckers, Wim Desmet
[Abstract]
Numerical modelling with the Boundary Element Method is
often hindered due to the excessive computational cost at
higher frequencies. In that context, a suitable model order
reduction scheme is proposed to alleviate the
computational cost needed for a BE fast frequency sweep
acoustic analysis. First, the method utilizes a kernel series
expansion to remove the frequency dependency from the
BEM system matrices. Subsequently, a Galerkin projection is
deployed on the frequency independent matrices to
reduce the size of the system. The basis employed in the
Galerkin projection is assembled by the Krylov subspaces
obtained through an Arnoldi procedure of the BEM system
on a predefined frequency grid. The use of Krylov
subspaces facilitates the definition of an error estimator
that indicates the level of the error expected by the
projection of the system. The efficiency of the model order
reduction scheme is first assessed in terms of algorithmic
efficiency and then validated on a test case.
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Evaluating acoustic properties based on sound energy for interior problems
Caglar Guerbuez, Steffen Marburg
[Abstract]
The analysis of sound radiation of vibrating structures is of great importance in order to develop low-noise emitting structures. For exterior problems, it was shown that the non-negative intensities shall be considered in order to evaluate the acoustic properties of a structure, since using this method accounts for the acoustic short circuit. This acoustic short circuit is a phenomenon which is present when evaluating the sound intensity of a structure. Regarding interior problems, the sound pressure level is commonly analyzed to characterize the acoustic behavior. However, the sound pressure level is strongly dependent on the position of the field point of interest. Thus, the analysis of the sound pressure level is highly sensitive to deviations of the field point position. This paper introduces the sound energy as an appropriate quality criteria for interior problems due to the global characteristic. The sound energies are compared with the sound pressure level for numerical benchmark examples and real world applications. Based on the time harmonic problem formulation, the acoustic interior problem is solved with the boundary element method (BEM).
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Relation between Convergence Tolerance for Iterative Solvers and Calculation Results in Rooms Using BEM
Yosuke Yasuda, Kota Saito, Hidehisa Sekine
[Abstract]
The Boundary Element Method (BEM) has been widely used in
the fields of acoustics, including the room acoustics. In BEM, the
iterative methods are often used to solve the linear system,
especially when dealing with large-scale problems. In general,
an iterative solver requires a convergence tolerance, i.e., a
value for termination of the process, to control the calculation
accuracy. Since the calculation accuracy and time are in a
trade-off relation, it is practically important to set this value
appropriately. It is often set to a smaller value for safety, but if
one limits the target to important quantities in room acoustics,
specifically reverberation decay curves, frequency responses,
sound pressure level distributions, etc., it is possible to set a
necessary and sufficient convergence tolerance for each
quantity. As a result, reduction of unnecessary calculation cost
can be expected. In this study, we investigate the influence of
the convergence tolerance for iterative solvers in BEM on
calculation results in rooms through numerical experiments. In
particular, we focus on the effect of the sound absorption
condition of the surrounding walls and the diffuseness of the
sound field. Finally, a recommendation value of the
convergence tolerance for each quantity in room acoustics is
proposed.
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Integration of FEM shell elements as a “boundary condition” in BEM calculations using different solution methods
Ralf Burgschweiger, Ingo Schäfer, Martin Ochmann
[Abstract]
Within the framework of research projects in the field of ”Computational Acoustics”, a powerful BEM-based code for the determination of the backscattered sound pressure level in the far field has been developed in recent years, using different solution methods (direct and iterative solvers, fast multipole method, ray tracing method etc.).
This code has been extended so that in addition to the ”classic” boundary conditions (pressure, normal velocity, impedance, inertial coupling, etc.), additional FEM shell conditions and thus elastic material properties can be taken into account. For this purpose, an additional FEM equation system is set up for the elements concerned which is integrated directly into the BEM equation system via appropriate transformation matrices.
The solution of this overall system can be done by means of direct equation solvers (based on a complete sparse matrix), by eigenvalue calculations or by the so-called Schur method (using iterative solvers).
The paper deals with the mathematical and physical fundamentals, shows the differences between the solution methods of the entire system and presents the results of test structures. As far as available until then, a comparison is also made with results from ”pure” FEM calculations.
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Numerical investigation of acoustic radiation damping in sandwich structures
Suhaib Koji Baydoun, Steffen Marburg
[Abstract]
Acoustic radiation damping is the primary energy dissipating mechanism for lightweight structures with large radiating surfaces. Attempts to reduce the vibrational response of those lightweight structures by additional mechanical damping can only be successful if the extent of mechanical damping is comparable or larger than the extent of radiation damping. In this regard, engineers are in need of reliable and flexible methods for the quantification of radiation damping as well as for the modelling of its effect in an early stage of the design process. In this paper, we address the equations of time-harmonic elastodynamics and acoustics by means of finite and boundary element methods respectively. The structural modes subject to fluid loading are determined using a nonlinear eigenvalue solver based on contour integration. Thereby obtained complex eigenvalues provide the modal values for radiation damping. The approach is verified based on examples of honeycomb sandwich panels in air and the results are compared to other numerical as well as to experimental results found in the literature.
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Moving sources and the 2.5D Helmholtz Boundary Element Method
Christian Kasess, Holger Waubke
[Abstract]
For perception tests on environmental noise, e.g. evaluating the effect of a noise barrier, it can become necessary to virtualize moving sources. In the framework of the 2.5D boundary element method, which is very useful for long and large structures, the full treatment of moving sources becomes computationally extremely involved. Even in the simplest case, i.e. mono-frequent emission and constant speed, higher speeds result in the necessity of calculating ever higher 2D frequencies beyond the signal’s frequency in order to perform the necessary inverse Fourier transform to acquire the time signal. For wide-band signals a 2D inverse Fourier transform needs to be performed. Adding further a non-uniform motion results in yet another dimension that needs to be integrated over.
The evaluation for sampled signals of sufficiently high sampling rates and durations of a few seconds can thus become challenging. Interpolation of data points in either frequency, space, or time can help to decrease the computational burden in particular when employing different types of demodulation schemes. The aim of the work presented is to investigate for sampled source signals at which points in the calculations interpolation can be applied without introducing too high a computational error.
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Simulation of vibrations from railway tunnels
Holger Waubke, Wolfgang Kreuzer, Tomasz Hrycak, Sebastian Schmutzhard
[Abstract]
Railway traffic in tunnels leads to annoying vibrations and secondary airborne noise in buildings near the tunnel. A model is presented that introduces a horizontal layering of the anisotropic soil coupled with tunnel shell. The horizontal layering allows to include reflections of waves at the interfaces. The tunnel shell and the superstructure are simulated by mean of finite elements 2.5D and the soil is coupled using the boundary element method in 2.5D. The 2.5D model allows using a moving source in a simple manner, which is important for high-speed trains. The difficulty for the model is that the Green’s function of the layered soil in 2.5D is not given explicitly. This problem is solved by using the easily calculated solution in the wavenumber domain. Therefore, the singular and non-singular boundary element integrals are solved in this domain. The additional numerical step needed is the inverse Fourier transformation of the integral values.
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Fundamental solutions in modeling of vibrations radiated from tunnels with 2.5D - BEM
Sebastian Schmutzhard, Tomasz Hrycak, Wolfgang Kreuzer, Holger Waubke
[Abstract]
In this talk, we present a boundary element method used to model vibrations within horizontally layered orthotropic soil containing a horizontal straight tunnel.
We assume that the tunnel extends infinitely along the x direction. After taking the Fourier transform, the three dimensional problem reduces to several two dimensional problems in the wave-number k_x-domain.
Taking the Fourier transform with respect to both horizontal variables x and y, the fundamental solution of the wave propagation has a simple structure in the wave-number (k_x, k_y)-domain. Inside a single layer, the fundamental solution can be represented by a superposition of six waves with complex wave numbers. Their corresponding coefficients are computed by solving a linear system, obtained by a boundary condition at the surface, continuity conditions at the interfaces between the layers, and a radiation condition at infinity. We study stabilizing strategies for the evaluation of the fundamental solution.
We use linear elements, and approximate the solution by linear functions. Exploiting the explicit structure of the fundamental solution in the (k_x, k_y)-domain, we can evaluate one part of the derivation of the BEM-matrix analytically.
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A Domain Decomposition Method with Fast Convergence for the Helmholtz Equation
Denis Duhamel
[Abstract]
Solving the Helmholtz equation by finite element methods is quite important in acoustics. When the frequency or the size of the problem increases, large meshes are necessary and consequently heavy computations are required. One possibility is to use domain decompositions for which the domain is decomposed into subdomains on which the solutions can be computed more easily. This involves an iterative scheme where data are transmitted across the boundary of subdomains. The main problem is to have a low number of iterations so that the problem can be solved in a reasonable amount of time. In this work, we present a domain decomposition method based on two main features. The first one is to use extended domains with absorbing boundary conditions. The second feature is to decompose the whole domain into a two dimensional network of subdomains so that double sweep preconditioners can be used in the two directions. Examples are shown where the number of iterations is usually less than 10. This number of iterations is also shown to depend very slowly on the number of domains and on the frequency resulting in an efficient computational method.
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On the impact of the shape of the artificial boundary in exterior Helmholtz problems
Nick Wulbusch, Reinhild Roden, Alexey Chernov, Matthias Blau, Andrea Moiola
[Abstract]
Acoustical problems can be investigated numerically in the
frequency domain by solving the Helmholtz equation with the
Finite element method (FEM). In exterior problems we
approximate the radiation condition by truncating the
considered domain with an artificial absorbing boundary.
Popular choices include absorbing boundary conditions (ABC)
or absorbing layers (PML), where the implementation of the
latter is usually more involved. Both methods have the
disadvantage (although for PML this is less significant), that they
still reflect small parts of the wave if the domain is discretized for
numerical computation. This reflection error is typically small for
normal (0 degrees) incidence but increases with larger
incidence angles. In this numerical study we consider a model problem in two
dimensions. For various ABCs we study the impact of changing
the shape of the artificial boundary on the reflection error and
the accuracy of the numerical simulations. The outcome of numerical simulations for different ABCs and
methods of shape optimization will be discussed in detail.
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BEM Simulation of tube acoustics using thin elements
Wolfgang Kreuzer, Veronika Weber
[Abstract]
The simulation of sound wave propagation and radiation in
connection with open tubes and pipes has a long tradition in
acoustic engineering. Among the many numerical methods
used in acoustics the boundary element method (BEM) plays
an important role because of its flexibility with respect to
geometry, material parameters and radiation conditions at
infinity. In our talk we present a BEM model for a musical tube
(Boomwhacker, Klangröhre). This plastic tube is 63cm long
with a radius r = 1.75cm and when hit against an object the
tube produces the musical note C. The wall width of the
tube is very small, thus, in the BEM-model a special
formulation with infinitely-thin elements is used. This has the
advantage that no coupling between interior and exterior
domain of the tube and no special boundary conditions at
the tube ends are necessary. Because of the material of the
tube, structural vibrations are neglected.
We introduce the model, present simulation results and
discuss numerical problems like mesh discretization and
numerical stability.
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Flexible multi-level fast multipole BEM with Direct Solver for Industrial Acoustic Problems
Yue Li, Onur Atak, Wim Desmet
[Abstract]
Fast Multipole BEM for acoustics is a very powerful method to
solve ultra large problems, which exploits iterative solvers.
However, for Helmholtz problem it may fail to converge in some
cases. Whereas FM-BEM provides a unique offering for ultra
large problems, many industrial problems are in fact can be
solved with tens of thousands of Dofs and can be considered as
medium to large size. For such problems, direct solvers are still
viable and provides a good option in terms of stability and
convenience of solving many right-hand sides. In this paper, we
present a fast multiple accelerated indirect BEM for solving
medium to large size acoustic problems. The assembly process
of the indirect BEM is accelerated by an efficient direct multi-
level fast multipole method. The hypersingular term of the
indirect BEM formulation in the far-field is directly approximated
with the multipole expansion, while the near field terms are
regularized to accommodate this hybrid approach. A parallel
direct solver is used for solving the linear system. The results show
that the FM accelerated assembly coupled with modern direct
solvers offers a robust approach, where medium to large size
acoustic problems can be efficiently solved on a modern
desktop computer.
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FEM-BEM applications in vibro-acoustics using GypsiLab
Matthieu Aussal, François Alouges, Marc Bacry, Gilles Serre
[Abstract]
The GypsiLab framework [1] is an Open Source Matlab suite of routines that permits the user to develop numerical codes that use the Finite Element Method (FEM) or the Boundary Element Method (BEM). Several cutting edge algorithms are included as, e.g. a complete H-Matrix toolbox that allows for the compression of the dense matrices in BEM. We present the first numerical experiments of the toolbox in fluid-structure interactions, and more precisely in vibro-acoustics. The benchmarks that we chose highlights the strong coupling of the FEM in the structure and BEM for the pressure field in the fluid. The talk will emphasize the ease of use of the toolbox and the handling of the complete algebra for the H-matrix compression, including the computation of hierarchical matrix inverses. A comparison with the literature shows the efficiency of the approach together with the high modularity of the toolbox. [1] F. Alouges and M. Aussal, FEM and BEM simulations with the Gypsilab framework, SMAI Journal of Computational Mathematics, 2018.
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Organization:
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S. Marburg, Technical University of Munich (TUM) |
L. Cheng, The Hong Kong Polytechnic University |
M. Hornikx, Eindhoven University of Technology |
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Numerical simulation of nonlinear viscothermal acoustic wave propagation
Jan Albertus De Jong, Erwin Kuipers
[Abstract]
Behind-The-Ear hearing instruments are typically applied to
assist people with a severe to profound hearing loss.
Accordingly, very high acoustic pressure levels can be
obtained in the ear canal of the wearer. At such levels,
significant distortion is present, leading to a reduced sound
quality. The distortion mainly originates from the balanced
armature receiver driver, operated beyond its linear limit.
However, as the acoustic pressure level in the acoustic
transmission system (basically a set of tubes) can be much
higher than 140 dB SPL, it is of interest to investigate the
amount of acoustic distortion caused by nonlinear wave
propagation. As such, this paper focuses on the prediction of the distortion
due to the combined effect of viscothermal- and nonlinear
wave propagation. To simulate these effects in the tubes, a
modified form of the model of Chester is used. In this model,
the viscothermal effects results in a half order derivative of
the acoustic pressure to time. This fractional derivative term is
approximated using the diffusive representation. At low wave
amplitudes, the correct implementation of the model is
verified using the Low Reduced Frequency model. At high
amplitudes, the results are compared with a full
Computational Fluid Dynamics simulation.
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Adjoint-based computation of shape sensitivity in a Rijke-Tube
Georg A. Mensah, Alessandro Orchini, Jonas P. Moeck
[Abstract]
Geometrical shape is the most important parameter defining acoustic spectra. While adjoint-based sensitivity analysis of shape alterations is a well established technique in other engineering disciplines such as airfoil design, it is less common in acoustic and especially thermo-acoustic contexts. Only recently the thermo-acoustic community has begun to deploy such methods for stability assessment. The present paper aims at continuing this effort by discussing (i) the challenges of domain shape deformations to adjoint-based gradient calclulations, (ii) potential extensions of the theory to higher orders, and (iii) a parameter-free implementation into a state-of the art thermo-acoustic Helmholtz solver. Deformations of a simple thermoacoustic system – a Rijke-Tube – will be studied as an example case to validate the method. Despite this practical focus on thermo-acoustics, the results of the paper are fruitful to other acoustic topics involving stability analyses as the underlying concepts readily apply to any eigenvalue problem linear or non-linear in its eigenvalue. Indeed, the gradient information calculated from first-order theory is the first building block for efficient algorithms to optimize the shape of acoustic devices.
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Exploration of the pseudospectral method for auralizations
Jorge Petrosino, Lucas Fernando Landini, Andres Bonino Reta, Georgina Alejandra Lizaso
[Abstract]
Sound propagation in auralizations can be simulated with geometric
or wave based models. Each one offers particular advantages and
disadvantages. Geometric models lead to the ray tracing method and
the image source method. Both methods achieve good results for
high frequencies and are efficient in large rooms with complex
structures, but are not able to represent in a simple manner specific
wave phenomena, such as diffraction. Typical wave based models
such as the finite element method, the boundary element method, or
the finite difference method are characterized for achieving very
precise results for individual frequencies applied to small and
moderately sized rooms. A typical disadvantage is the need for high
computing power, mainly due to the necessity of running separate
simulations for individual frequencies because of numerical
dispersion. In this paper, auralizations with the k-space
pseudospectral method are explored since it is possible to reduce or
eliminate the problem of numerical dispersion with this technique. For
certain conditions, complete range impulse response can be obtained
from a single simulation execution. The room response, including
absorption, diffraction and diffusion is analyzed for different
scenarios.
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Organization:
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V. Chatziioannou, Department of Music Acoustics, University of Music and Performing Arts Vienna |
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Assessing the role of onsets for musical instrument identification in an auditory modeling framework
Kai Siedenburg, Marc René Schädler, David Hülsmeier
[Abstract]
Sound onsets are long known to provide important cues for
musical instrument identification by human listeners. It has
yet remained unclear whether this circumstance rests on
perceptual or acoustical grounds: Are listeners utilizing
informative features that are only available in the onset, or
are there equally informative acoustic features throughout
sounds that listeners tend to ignore because of their
redundancy? Here we approach this question by using the
automatic speech recognition-based simulation framework
for auditory discrimination experiments (FADE) [Schädler et
al., 2016, JASA] to model data from a recent study on
instrument identification [Siedenburg, under review].
There, listeners were tasked to identify Western orchestral
instruments from 64 ms segments extracted from the onset
or from the middle portion of the sounds. In the present
study, instrument identification is modeled using FADE with
three different feature sets: Log-Mel spectrograms, Mel-
Frequency Cepstral Coefficients, and separable Gabor filter
bank features (SGBFB). Results indicate that all three feature
sets yield a strong decrease of classification performance for
the middle portions of the sounds. This suggests that the
utility of onsets for sound identification may not be based on
specific properties of human auditory perception but on the
particular acoustic richness of sound onsets.
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The Role of Attack Transients in Timbral Brightness Perception
Charalampos Saitis, Kai Siedenburg, Paul M. Schuladen, Christoph Reuter
[Abstract]
Evidence from acoustical modeling and multidimensional scaling of
brightness dissimilarity ratings of natural instrumental sounds [Saitis
& Siedenburg, in prep.] suggest at least two acoustical properties of
tones to play a role in the perception of timbral brightness: a) the
spectral energy distribution, and b) the attack time and/or
asynchrony in the rise of harmonics. To further investigate the role of
these cues in timbral brightness perception with highly controlled
stimuli, 40 musically experienced listeners will perform a MUSHRA-
like task involving synthetic harmonic complexes varying along three
parameters: spectral centroid, attack time, and fluctuation of spectral
centroid over the first 100 ms. The latter models the progressive
expansion of the spectrum toward the higher harmonics. It is
hypothesized that two sounds with same spectral centroid values but
different attack times will be perceived to differ in brightness, with
faster attacks leading to brighter sounds. Similarly, two sounds with
same spectral centroids and attack times but different ”arrival times”
for the higher harmonics will vary in perceived brightness: the sooner
higher harmonics appear, the brighter the sound. Overall, this
experiment will thus help to clarify the relation between two major
factors in timbre perception: onset and spectral energy distribution.
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Non-iterative, conservative schemes for geometrically exact nonlinear string vibration
Michele Ducceschi, Stefan Bilbao
[Abstract]
This work presents novel finite difference schemes for the solution of the geometrically exact nonlinear model of wave propagation in strings. Stability is enforced by ensuring conservation in the discrete setting. The schemes are here derived from a suitable quadratisation of the nonlinear potential, yielding ultimately a non-iterative update equation requiring the solution of a sparse linear system per time step. Comparison with previous iterative schemes is carried out, showing convergence to a unique solution. The new schemes are extremely efficient, yielding speedups of about half an order of magnitude over previously available schemes.
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Numerical simulation of transients in single-reed woodwind instruments
Sebastian Schmutzhard, Montserrat Pàmies-Vilà, Alex Hofmann, Vasileios Chatziioannou
[Abstract]
In this work, we discuss a mathematical model of the sound production in single-reed woodwind instruments focusing on player-instrument interactions. The model consists of an ordinary differential equation, coupled to a system of partial differential equations. The motion of the reed is described by a nonlinear harmonic oscillator, where the nonlinearity stems from the collision of the reed with the mouthpiece lay and the coupling between air flow and pressure. The dynamics of air pressure and air velocity inside the instrument are described by a boundary value problem for the wave equation. We introduce a method to model the interaction between the player’s tongue and the reed, which leads to a further nonlinearity in the reed equation. Moreover, we study strategies to estimate the model parameters relevant for the production of realistic tones, as well as for realistic transients in between tones. Finally, using the proposed model, recordings of real players are resynthesised.
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The effect of fast trombone slide glissandi on the mechanics of artificial lips
Amaya Lopez-Carromero, Jonathan Kemp, Donald Murray Campbell
[Abstract]
The lips of a brass player can be modelled as a mechanical oscillator which couples with the acoustic resonances of the instrument to generate and sustain a musical note. On instruments like the modern orchestral trumpet, operation of a valve changes the length of the resonator without significantly altering the mean pressure of the air column. On a slide trombone, however, a glissando involves a telescopic change in the length of the resonator, resulting in localised changes in the mean pressure in the air column. The consequent low frequency fluctuations in mouthpiece pressure can disturb the equilibrium position of the lips, affecting their dynamic behaviour. This paper describes experimental studies in which an artificial mouth is used to investigate the infrasound mouthpiece pressure gradients generated by fast glissandi on a trombone, and their influence on the mechanics of the oscillating lips.
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A Survey of Recent Studies on Initial Transients in Free Reed Oscillation
James Cottingham
[Abstract]
In recent decades a number of models have been proposed for sustained oscillation of free reeds, including in some cases conditions for the initiation of the oscillation, as well as experimental work testing the validity of the models. Typically these models involve a simplified or generic geometry of the reed tongue. In addition, the motion of the free-reed tongue in the early stages of the initial transient has been studied experimentally with the aim of determining the presence of higher modes and their possible role in the attack transient, as well as the possible relation of the details of reed geometry to details of the initial transient waveform, including the rise time. Comparisons of reed tongues of different design have been made to explore the role of these modes in the initial excitation. There is evidence that the first torsional mode and the second transverse mode may be significant in initiating reed oscillation, so that reed design enhancing the torsional mode may be helpful in alleviating the problem of slow attack. [Work partially supported by United States National Science Foundation Grant PHY-1004860]
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A Sustained Vowel Replacing algorithm based on Iterative Formant Filtering
Chun-Tang Hsu, Yi-Wen Liu
[Abstract]
In electronic music, formant filtering refers to transforming a
source signal to sound like vowels. However, existing plugins,
such as iZotope VocalSynth 2, and The Orb do not focus on
preserving the timbre of the source audio after filtering. In
this paper, an algorithm is proposed for replacing a
sustained vowel of one singer with another vowel while
preserving the timbral characteristics. Here we assume that
a timbre source signal (TSS) and a vowel source signal (VSS)
are available by recording from singer 1 and 2, respectively.
Spectral envelopes below 5k Hz of both signals are
extracted first. Then, the ratio of the two envelopes is used
for fine-tuning the TSS power spectrum and this process will
be repeated iteratively. Upon completion, the output file is
meant to preserve the timbre of singer 1 while its vowel
identity, including the accent, has been changed to that of
singer 2. To evaluate the synthesis quality, sustained vowels
sung by 11 singers were recorded, and the proposed
algorithm was applied for cross-synthesis. 11 subjects are
invited to tell if synthesized timbre resembles singer 1 or
singer 2. Results show that 7 subjects successfully chose
singer 1 with accuracy > 70%.
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Sympathetic vibration in a piano
Jin Jack Tan, Armin Kohlrausch, Maarten Hornikx
[Abstract]
Sympathetic vibration is a common phenomenon in musical
instruments. In the piano, strings that are not struck directly by
the hammer can vibrate sympathetically due to physical
coupling. This effect will, for instance, occur when the una
corda pedal is used in a grand piano. When depressed, the
hammer shifts rightwards so that the leftmost string will be
missed, striking only 1 out of 2 (or 2 out of 3) strings in a note at a
same time. In this study, the una corda effect is investigated by
modeling sympathetic strings in an existing numerical piano
model. The strings are discretized into finite elements and they
are coupled to a soundboard, which is solved modally. An
energy-preserving time-stepping scheme is employed to ensure
numerical stability. It is known within the pianist community that
the una corda pedal changes the timbre and sustain of the
sound. As a preliminary study, the una corda pedal effect is
discussed based on a number of perceptual and acoustic
descriptors which are known to be used in judging the timbre of
piano notes.
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Predictive simulation of mechanical behavior from 3D laser scans of violin plates
Mirco Pezzoli, Riccardo Roberto De Lucia, Fabio Antonacci, Augusto Sarti
[Abstract]
In this paper we present a methodology for the predictive
simulation of the vibrometric behaviour of a violin plate. The
3D
outer shape of the plate is acquired by means of a 3D laser
scanner and then smoothed in order to remove artefacts
and
details that are unnecessary for the acoustics simulation. The
thickness of the plate is incorporated into the model through
a
technique that receives as input the thickness sampled at
some
points of the plate and interpolates it over the entire surface.
We validate this 3D reconstruction technique by comparing
the
vibrometric behaviour of the 3D model with data measured
on the
reference plate, and with simulations on a model with
uniform
thickness.
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Measurement and Modelling of the Japanese koto: Problems and Solutions
Kimi Coaldrake
[Abstract]
There are few studies of the acoustical properties of the
Japanese koto (zither) and little is known about the resonances
arising from its complex organic shape, its length (1.83cm) and
the paulownia wood from which it is made. As part of the
measurement and construction of a high resolution finite
element model of a hand-crafted professional-grade koto using
COMSOL Multiphysics® software, three major problems were
resolved and are discussed. First, lack of information on the
wood for key model inputs required scientific evaluations of
physical properties by microscopy and independent testing. The
traditional grade system relating to cross and straight grains
used in different components was also found to be critical to
model accuracy. Second, issues relating to measuring the
complex geometry were only resolved by obtaining 2400 cross-
sections of the koto in a CT scan. These revealed hitherto
unseen details such as the semi-circular internal top plate
edging and roughhewn interior. Their inclusion in the model
significantly improved its accuracy. Third, the organic shape
and length required special lofting techniques, nearly one
million mesh elements and 34 million degrees of freedom which
in turn necessitated upgrading computer hardware. Initial results
from modelling will also be briefly reported.
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Study of Ivory Alternatives for Koto Bridges: Correlation between Sensory Evaluation and Amplitude Modulation of Fundamental Spectrum
Shuichi Sakamoto, Shunsuke Watanabe, Taku Watanabe, Yuki Sato, Shuma Ito, Ryota Kominami, Yuki Akiba
[Abstract]
The bridges of a koto, a Japanese musical instrument, have traditionally been made of ivory for its superior sound quality and appearance. On the contrary, in the conference of CITES CoP17, 2016, a resolution for the closedown of the domestic market of the ivory was adopted. Therefore, the demand for substitute materials for the ivory is expected to increase in future. Despite this, from the viewpoint of sound quality, the currently available substitute materials used to manufacture koto bridges remain inferior in comparison with ivory. This study investigates the performance of substitute materials by investigating the properties of koto bridges made of metal, resin, and wood. A sensory evaluation of a single tone played by professional and expert koto players was performed for all substitute materials. Additionally, a wavelet analysis of the recorded tone was conducted. The amplitude modulation of the fundamental spectrum level varied in time with respect to each of the chosen materials. The study also includes an investigation of the correlation between sensory evaluation and the recorded amplitude modulation of fundamental spectrum.
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Organization:
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A. Schlesinger, theapro theater projekte daberto + kollegen planungsgesellschaft mbh |
J.M. Kimmich, HTW Berlin |
H. Xie, Chongqing University |
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Face the Music: A 12 year study of the sound of performance and hearing of classical music
Stephen Dance, Douglas John Shearer, Georgia Zepidou
[Abstract]
The Acoustics Group has collaborated with the Royal
Academy of Music since the enforcement of the Control
of Noise at Work Regulations in 2008. Every new student
has attended an awareness course on the sound of
performance and hearing, undertaken health surveillance
and has been questioned as to their aural history. In
addition, the sound of performance, in terms of individual
practice, rehearsal and performance has been measured
for all types of instrument. This dataset has been analysed
to determine the hearing acuity of classical musicians and
their occupational sound exposure. A paper will be
presented on this analysis which suggests that self
produced music has much less impact on hearing loss
than typical noise exposure, as given in ISO 1999:2013.
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The exposure of musicians to sound assessed by two-channel noise dosimetry
Agnieszka Pietrzak, Jan Zera
[Abstract]
The sound exposure doses of musicians may significantly differ depending on the ensemble, repertoire and the intensity of their daily professional activity. In solo performance the sound levels are often asymmetric between the two ears, which mostly results from the directional characteristics of the instrument’s sound radiation, with only a small effect of reverberation, as the player is exposed to the instrument’s sound in the near field. In ensemble playing the sound of all the other instruments is usually the main factor influencing the sound exposure dose. In this study a two-channel noise dosimetry was used to assess the sound exposure doses in the left and in the right ear of music students. The measurements were conducted during rehearsals of chamber music ensembles, symphonic and wind instrument orchestras, big-band, and during individual practicing. The results show an interaural asymmetry in sound level, up to about 6 dB, in musicians playing instruments held asymmetrically to the player’s head and in cases when the musicians were exposed to intense sound of the neighboring instruments. It also was observed that the spread of sound levels was larger during individual practicing (82-103 dBA) than in large ensembles (90-99 dBA).
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Symphony Orchestra Musicians: Reduction of Sound Exposure by Physical Measures
Remy Wenmaekers, Bareld Nicolai, Maarten Hornikx, Armin Kohlrausch
[Abstract]
Since most musicians are reluctant to wear hearing protection devices, physical measures are used in an attempt to control exposure levels in the orchestra. Examples of such measures are increased space between musicians, positioning brass instruments on risers, putting screens between musicians and adding sound absorption. The effectivity of such measures has been shown by the authors in JASA 142, 3154 (2017) and the published paper will be discussed in this presentation. It is difficult to exactly reproduce the music played by an orchestra under different conditions to determine the effect of measures. A solution to this problem was to simulate the effectivity of control measures using a prediction model. The model calculates the equivalent sound levels for a performance of the first 2 minutes of the 4th movement of Mahler’s 1st symphony, representative for loud orchestral music. The model outcome shows good agreement with measurements of the same excerpt. Results show that physical measures are not effective enough to significantly reduce sound levels. It seems that musicians, playing current modern powerful instruments, have no other choice than to protect their ears with ear plugs under all circumstances if they wish to avoid the risk of developing hearing damage.
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Dicriminability of high-resolution audio with regard to the quantization accuracy
Masanobu Miura
[Abstract]
The high-resolution audio has been popular for commercially-sold music delivery services. The high-resolution audio is characterized in both the over 20Hz components and high-quantization accuracy. This presentation focuses on the quantization accuracy in terms of the discriminability of the difference on quantization accuracy. In particular, here employs 24 bits and 16 bits format’s audio as experimental stimuli, where the ones of 16 bits are created by conducting downbit-processing for commercially sold 24 bits audio. Three methods of downbit-processing are employed. 10 university students were asked to listen to the audio and to answer the format, as 24 or 16. The stimuli are simultaneously presented to listeners through headphones. Then the listeners were asked to answer the sound image localization. This method is newly employed here. The method has an advantage that listeners are possible to compare the sound simultaneously and to answer the sound image locations, which is thought as easier to rate than to the conventional listening experiment, just presenting each by each. The experimental result shows significant difference on the discrimination on the difference of audio format. The factor why people are possible to discriminate will be discussed on the conference venue.
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Reducing the sound exposure level in an orchestra pit by a set of tailored measures
Anton Schlesinger, Martin Tschaikner, Martin Ochmann, Jan Michael Kimmich, Stefan Frank, Axel Schlicksupp
[Abstract]
Risk mitigation of a noise-induced hearing loss for musicians of orchestras is generally based on a set of measures. Hearing protection, which is one of the most efficient means, is considered inadequate as the sole solution in orchestras. Technical and organizational measures are attractive because of their appropriateness to dovetail with the conditions of performance practice and architecture.
This contribution presents an attempt to reduce the sound exposure level in the orchestra pit of the Deutsche Oper Berlin by a set of constructional measures. The current conditions and the requirements of the musicians were assessed with acoustical measurements and with a questionnaire, respectively. A hybrid simulation method was subsequently employed to simulate and optimize the sound field in the entire frequency range. Optimization focussed on an improvement of audibility between musicians and the equalization of the areal pattern of room modes. While the former objective is well-know for improving the ensemble play, the latter reduces the exposure as well as the masking of the fundamental frequencies of singers and instruments. This contribution concludes with a set of constructional measures, which are considered viable approaches for reducing the risk of noise-induced hearing loss in the orchestra pit.
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Organization:
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J. Kergomard, LMA CNRS Marseille |
D. Sharp, The Open University |
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Spectral and Mathematical Music Theory Analyses of the Ikoro Drum Using Visualizations and Sonifications of Beat-Class Theory.
Stephen G. Onwubiko, Neilsen Tracianne, Andrea M. Calilhanna
[Abstract]
The Ikoro or Ekwe, is a wooden slit drum, a percussion
instrument of the idiophone family, which by virtue of its
functions and its
predominance in the eastern parts of Nigeria with the Igbo’s
and plays an important role in the Igbo musical culture and
settings.
The Ekwe being a drum made of wood has pitches, and
produces a very deep rich bass sound which can be heard
from a
distance and has a sustained reverberation rendering a
background beat to music or sound of warning. This paper
looks at
frequency domain of the sustained reverberation, measuring
and analyzing the dynamics responses of the structure upon
an
excitation, describes the historical and constructional process
of the Ekwe, and examines the isomorphism between Ekwe
music
and mathematics by applying the narrative of mathematical
music theory developed by Richard Cohn (Yale). This work
explores
how the rhythm, meter and pitch of Ekwe music, can be
represented through using various techniques such as
visualizations and
sonifications, by applying Cohn’s beat-class set theory with
Andrew Milne’s (WSU) instruments of mathematical music
theory, the
cyclic graph XronoBeat, and SkiHill apps. These analyses
provide a deep understanding of the distinct sound of the
Ekwe.
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Ikorodo Music Analyzed Through Visualizations and Sonifications of Beat-Class Theory
Andrea M. Calilhanna, Stephen G. Onwubiko, Tobi Kemewerigha
[Abstract]
This paper explores how Cohn’s (2018) beat-class theory,
represented through visualizations and sonifications using cyclic
graphs, provides a suitable approach for the analysis of Ikorodo
music of the Igbo people of Nigeria. Unlike traditional music
theory, Cohn’s instruments of music theory have the capacity to
represent the listener’s temporal experience of music because
modern meter theory encompasses the recognition and
documentation of the experience of mathematics during
listening. Our presentation explains how embodied
mathematics and music can be represented through
visualizations and sonifications of beat-class theory when
listening to Ikorodo music. Arguably, the use of traditional
Western music theory in classroom textbooks to analyse all
music has resulted in a ëdumbing down’ of music curricula in
general (Calilhanna, 2018, 2019b). However, recent research
indicates the efficacy of teaching the isomorphic relation of
mathematics and music in an interdisciplinary and student-
centered approach (Cohn, 2018; Milne, 2019; Calilhanna, 2017;
Calilhanna and Webb 2018; Calilhanna, 2018; Hilton, Calilhanna
and Milne, 2018; Hamilton, T. et al, 2018). Thus, through the
application of mathematics to analyse Ikorodo music we will
demonstrate how the Igbo’s Ikorodo music can be recognised
for its complexity, beauty, unique characteristics, and cultural
importance.
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Procedure for arranging backing score for ensemble music by evaluating ease of playing on instrumentalists
Nozomiko Yasui, Masanobu Miura
[Abstract]
Since arrangement of sheet music for orchestral ensemble requires a certain musical knowledge and expertise, amateur orchestral musicians often have difficulties when they arrange the sheet. An automatic arrangement method was proposed by employing Eigenmusic which is a set of eigenvector for many music excerpts, in order to evaluate the similarity among phrases. The difficulty in playing the arranged sheet music, however, was not evaluated so whether players feel difficulties on playing was out of discussion. This study proposes procedure for evaluating the degree how easy to play the phrases on sheet music. This procedure generates phrases for woodwind instruments using a MIDI database of backing phrases. Each musical phrases selected from the database are allocated to instrumentalists. Next, the ease of playing each phrases is evaluated based on fingering that is important in performance. Finally, phrases evaluated as the easiest are conjoined in each instrumentalist. We conducted an experiment to investigate the ease of playing for generated sheet music by musician’s rating. Experimental results showed that the procedure for evaluating ease of playing is effective so that proposed procedure can generate easier phrases to play than our previous method.
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Anti-phase synchronization between the oscillation in the pipe and that in the foot of a flue organ pipe
Kin'Ya Takahashi, Sho Iwagami, Shuhei Tateishi, Genki Tsutsumi, Taizo Kobayashi, Toshiya Takami
[Abstract]
In the recent work (Acoust. Sci. & Tech. vol.40, No.1 pp29-39
(2019)), we found with numerical study of a 2D flue organ pipe
mode that the foot works as a Helmholtz resonator. When the
frequency of acoustic oscillation in the pipe is higher than the
resonance frequency of the Helmholtz resonator by almost the
full-width at half-maximum, the most stable oscillation is
observed, and the oscillation in the foot is anti-synchronized with
that in the pipe. However, if the Helmholtz resonance frequency
is nearly equal to the acoustic oscillation frequency, the
oscillations in the pipe and foot become rather unstable and
the oscillation in the foot lags behind that in the pipe by nearly
pi/2. In this paper, we study the phase relations among the
pressure oscillation in the foot, that in the pipe, and the
hydrodynamic oscillation of the jet, which are the key to
understanding the mechanism stabilizing and destabilizing
oscillations of the flue organ pipe.
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Reproducibility of Mode Transition of Edge Tone with DNS and LES
Sho Iwagami, Taizo Kobayashi, Kin'Ya Takahashi, Yuji Hattori
[Abstract]
A jet attacking a edge oscillates spontaneously, and generates
aerodynamic sound celled ”edge tone”. The edge tone is the sound
source of air-jet instruments like a recorder and flute. The jet
oscillation has several modes. While one of the modes is selected, the
frequency increases linearly with the jet velocity. When the jet
velocity exceeds a threshold, the mode transition arises, and it is
hysteretic. The mode transition induces sudden changes of the
frequency and the sound energy. Therefore, the reproducibility of the
mode transition with a numerical calculation is crucial for
understanding the mechanism of edge tone. In this study, the 2D
model of edge tone is investigated with compressible DNS and LES.
The DNS adopted in our study is an exact method without including
artificial viscosity. The mode transition can be accurately reproduced
with DNS. On the other hand, in some cases of calculations with LES,
the mode transition cannot be reproduced even though the
calculation is not broken down. We will discuss the details of this
problem.
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Organization:
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P. Baddoo, University of Cambridge |
L. Ayton, University of Cambridge |
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An aeroacoustic investigation into the effect of self-oscillating trailing edge flaplets
Edward Talboys, Thomas Geyer, Christoph Bruecker
[Abstract]
Aerofoil self-noise is a phenomenon that has been significantly
researched and the mitigation thereof has been extensively
explored. Trailing-edge serrations, porous trailing-edges and
rigid brush like extensions are a few of the previously explored
mitigation strategies. With the rapid progression and interest in
low Reynolds number UAV aircraft in recent years, the topic is
once again a topic of significance. The present study uses
simple passive flexible oscillators (flaplets), which are attached
to the trailing edge of a NACA 0012 aerofoil. A previous study,
by the current authors, on this type of oscillator has shown that a
significant benefit on tonal noise can be achieved, via a
modification to the feedback loop. In the present study, the
aerofoil has been tripped such that the effect of the flaplets can
be characterised for turbulent boundary layer - trailing edge
acoustics. The measurements were carried out in an open jet
wind tunnel at Brandenburg University of Technology, over a
wide range of low (50,000) to medium (350,000) Reynolds
Numbers at varying geometric angles of attack. Sound pressure
level measurements, aerodynamic force measurements and
wake profiles will be presented in the full manuscript.
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The suppressed sound of inhomogeneous flapping airfoils
Michael Weidenfeld, Eran Arad
[Abstract]
We study the aeroacoustic problem of a flapping thin airfoil in
a high-Reynolds and
low-Mach regime, and propose novel means to reduce the
far-field sound originated by
the fluid-structure interaction. The present model consists of a
two-dimensional, elastic
airfoil, periodically actuated at the leading edge. The near-
field is analyzed using thin
airfoil theory, and a discrete-vortex wake-model. The derived
airfoil motion and wake
dynamics are then radiated as noise sources to the far field
using Powell-Howe’s
acoustic analogy. Optimal material properties and thickness
distribution are sought to
reduce the sound of an otherwise rigid airfoil under specific lift
constraints. To this end,
we formulate an optimization problem, yielding a
Pareto front, reflecting the
trade-off between the competing objectives of lower sound
levels and aerodynamic
efficiency. Compared with a rigid heaving airfoil, over 10 [dB]
sound reduction was
obtained for the optimal flexible configuration producing the
same lift amplitude. The
sound reduction mechanism is two-fold: The motion and
wake dipoles are shifted to an
antiphased mode thus reducing the total sound signal, while
the motion dipole is fixed
in magnitude, retaining the lift amplitude. The opposing
sound dipoles follow from the
synchronized disposition of the airfoils’ motion and circulation
signals.
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Acoustic emission of aeroelastic vortex-gust interactions
Huansheng Chen, Justin Jaworski
[Abstract]
The coupled interaction between an unsteady vortical flow
and dynamics of an aerodynamic structure is a canonical
problem for which analytical studies have been typically
restricted to either static or prescribed structural motions.
The present effort extends beyond these restrictions to
include a Joukowski airfoil on elastic supports and its
aeroelastic influence on the incident vortex, where it is
assumed that all vorticity in the flow field can be represented
by a collection of line vortices. An analytical model for the
vortex motion and the unsteady fluid forces on the airfoil is
derived from inviscid potential flow, and the evolution of the
unsteady airfoil wake is governed by the Brown and Michael
equation. Once the evolution of the vortical flow field is
established, the noise radiated to the acoustic far field is
determined by appeal to classical vortex sound theory.
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Aeroacoustic Evaluation of Low-Noise Wind Turbine Blade Tips
Michaela Herr, Claas-Hinrik Rohardt, Benjamin Faßmann, Jorge Pereira-Gomes, Christina Appel
[Abstract]
The planned contribution provides an overview on accomplished wind tunnel testing and simulation activity within the German national wind energy project BELARWEA.
The project aimed at numerical tool development and validation to support the design of both aerodynamically efficient and low-noise wind turbine rotors. The focus was set on the tip region, herein defined as the outer 20 % of the rotor radius, i.e. addressing noise generation from the trailing-edge- and tip vortex regions. Aeroacoustically driven 2D profile design, 3D winglet design and 2D/3D CFD and CAA analysis were supplemented by the transfer of passive noise reduction technologies from aerospace applications to wind turbine blades. Validation experiments were performed in systematic evaluation steps, namely
Ï at 2D blade sections in the Acoustic Wind-Tunnel Braunschweig (AWB) of DLR, and
Ï at 3D blade tip models in the larger acoustic facility DNW-NWB, operated by the German-Dutch Wind Tunnels foundation.
A noise reduction potential of up to 8 dB could be demonstrated in the NWB experiment, confirming the previously documented AWB results. The overall reduction was composed of a 2-4 dB reduction by a new low-noise profile design RoH-W-18%c37 (relative to a NACA 64-618 reference configuration) and an additional ∼6-dB reduction by brush add-ons.
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Analytic solutions for acoustic scattering by blade rows with complex boundaries: porosity, compliance and impedance
Peter Baddoo, Lorna Ayton
[Abstract]
The acoustic scattering by an infinite cascade of flat plates is analysed for a range of boundary conditions including porosity, compliance and impedance. By employing the Wiener-Hopf method, analytic expressions are derived for the pressure field that are uniformly valid throughout the entire domain. This enables exact solutions for the unsteady lift, sound power output and other acoustic quantities. The Wiener-Hopf analysis shows that complex boundaries have a significant effect on the modal structure of the cascade’s inter-blade region, whilst the modes in the far-field regions are unchanged. The results demonstrate the considerable impact of small modifications to the boundary conditions and highlight the importance of the accurate modelling of aeroelastic effects when predicting sound transmission and generation in turbomachinery.
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Organization:
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I. Kuznetsova, Kotelnikov IRE of RAS |
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Digital micro-laboratory application using surface acoustic wave devices
Jun Kondoh, Tomohiko Fukaya, Yota Terakawa, Sota Tsunogaya, Ren Ikeda
[Abstract]
Droplet manipulation is one of the important applications of surface acoustic wave (SAW) devices. When a sensor is fabricated on the droplet manipulation surface, a digital micro-laboratory for droplet manipulation, mixing, and measurement is realized on a piezoelectric substrate. For optimum droplet manipulation, it is necessary to clarify the acoustic streaming caused by the SAW in droplets. We have discussed droplet manipulation mechanism, experimentally. For optimum control of the droplet, it is necessary to measure the droplet position. When the SAW was generated by the short burst wave, reflected and transmitted signals were observed. The origin of the signals is the longitudinal wave propagation in the droplet. Therefore, it is possible to determine the droplet position with the signals. For the droplet position measurements, the longitudinal wave velocity of the droplet is known. If the droplet position is known and the velocity is unknown, it is possible to measure the velocity of the droplet using the SAW device. This is a unique measurement method of a longitudinal wave velocity of droplets.
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A study on the relationship between rock microstructure and wave disperson in carbonates and sandstones
Wei Cheng, Jing Ba, Jose Carcione
[Abstract]
Rock anelasticity, i.e. acoustic wave velocity dispersion and
attenuation, is related to porosity, texture and pore fluids.
The relation is important to provide a basis for further rock
physics studies and field applications of seismic exploration.
In this work, ultrasonic compressional and shear wave
velocities are measured in full-water and full-gas saturated
tight carbonates, sandstones and volcanic rocks. By adding
experimental data from the literature, we evaluate the
compressional wave velocity dispersion as a function of
porosity for all these lithologies (296 samples). It shows that
the dispersion peaks at approximately 15 % porosity for each
lithology, which we consider as a characteristic porosity,
indicating that wave dispersion increases with porosity in the
low porosity range, but decreases in the high porosity range.
The poroelasticity theory is used to model the compressional
wave dispersion. Results reveal that the microcrack radius
and porosity have peaks at moderate porosities (15-19 %) for
the three lithologies. The characteristics may be related to
the diagenesis process and thus to microstructure of
different lithologies. The new findings here reflect the intrinsic
natures of shallow earth rocks, which will contribute to better
understandings on earth rocks for researchers of rock physics
and exploration geophysics areas.
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Acoustic waves in piezoelectric plates in contact with gasoline
Iren Kuznetsova, Boris Zaitsev, Andrey Teplykh, Anastasia Kuznetsova
[Abstract]
We studied the properties of SH0 wave in LiNbO3 plate which was in contact with gasoline. The dependencies of phase velocity vs on parameter hf (h is a plate thickness, f is a wave frequency) were plotted. The influence of various types of gasoline on the wave properties has been studied. It has been found that the velocity of SH0 wave in piezoelectric plate decreases with the increase of permittivity of gasoline. The theoretical results have been experimentally verified by using delay line based on SH0 wave in YX-LiNbO3 plate with a thickness of 200 µm with operating frequency 3.65 MHz. It has been found that the wave frequency decreases with grow gasoline octane number. We have also theoretically and experimentally demonstrated that it is possible to develop gasoline identifier using the piezoelectric lateral electric field excited resonators. In this case the value of real part of electrical impedance on the frequency of parallel resonance uniquely associate with the octane number of gasoline contacting the free side of the resonator. At that the frequency of parallel resonance does not depend on permittivity/octane number of gasoline. Work is done under goverment task of Kotelnikov IRE of RAS.
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Study of shear-horizontal waves in structure “piezoelectric - viscous and conductive liquid”
Vladimir Kolesov, Iren Kuznetsova, Vladimir Anisimkin, Zhenghua Qian, Alena Gorbunova
[Abstract]
Recently it has been found that velocity of shear-horizontal surface acoustic waves initially increase as the conductivity of the layer located on the surface of the piezoelectric medium increases. At a certain value of conductivity the velocity of such waves reaches a maximum and with a further increase in conductivity begins to decrease. The same effect was found for conductive liquid and shear horizontal wave in piezoelectric media. In this work the influence of viscosity of a liquid on the magnitude of this effect is theoretically and experimentally studied. It was found that for surface acoustic leaky wave in 36YX lithium niobate and Bleustein-Gulyaev wave in cadmium sulfate as the viscosity of a liquid increases the value of above effect decreases and at a certain value of viscosity it disappears. As a liquid we used butil acetate characterized by low permittivity as 6. The results obtained can be useful in the development of sensors for biological fluids. Work is supported by the grants #17-51-53101 and #18-57-7802 from Russian Basic Research Foundation.
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Microstreaming patterns induced by shape modes of acoustically trapped bubbles
Sarah Cleve, Gabriel Regnault, Cyril Mauger, Claude Inserra, Philippe Blanc-Benon
[Abstract]
The combination of volumic, translational and nonspherical bubble oscillations induce a relatively slow mean flow, called microstreaming, in the vicinity of the bubble. It is well known that microstreaming and its resulting shear stresses play an important role in medical applications such as sonoporation (permeabilization of a cell membrane). The exact mechanisms are however not fully understood. Only few studies have tried to correlate the relatively slow streaming to the fast temporal bubble dynamics. In the present work, we visualize for the first time microstreaming around an acoustically trapped bubble that is oscillating with axisymmetric shape modes. Different types of streaming patterns can be observed. Correlation to the temporal bubble dynamics allows us to identify several important parameters that define the patterns of microstreaming. As might be expected the mode number plays an important role, nevertheless it is not sufficient to define a streaming pattern. Phase differences between several modal components as well as the bubble size with respect to the resonance frequency have to be taken into account as well.
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Nonlinear shear wave propagation to assess biomechanical properties in soft tissue
Juan Melchor, Guillermo Rus, Antonio Callejas, Inas Faris, Javier Naranjo, Miguel Riveiro
[Abstract]
Ultrasonic-based characterization of soft tissue is an emerging technology with great
potential as a clinical diagnostic tool. There is evidence that abnormalities in the
structural architecture of soft tissues are linked to a broad range of pathologies including
tumors, liver fibrosis, preterm birth...The present work derives different theories of
non-linear formulation of wave propagation using generalized and particular
coordinates on a material characterized by different strain energy functions.
Afterwards, these formulations are adapted to simulate the propagation of shear waves
on nonlinear elastic materials to explain the properties of soft tissue when is measured
under these conditions. The resultant system of equations could be solved under a finite
differences framework and probabilistic inverse problem to explore how improves the
understanding of soft tissues. The response of the system varying the linear/non-linear
parameters of the strain energy is obtained via Finite Differences Time Domain
simulation showing significant variations among the different models. Nonlinear wave
propagation approach implies a new paradigm to model the nonlinear biomechanical
behavior of soft tissue improving the plausibility versus previous elastic or viscoelastic
predictions.
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Organization:
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X. Wang, Beihang University |
W. Yang, The Hong Kong Polytechnic University |
Y.-S. Choy, The Hong Kong Polytechnic University |
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Effect of Acoustic Treatment on Fan Flutter Characteristics
Yu Sun, Xiaoyu Wang, Xiaofeng Sun
[Abstract]
This paper investigates the effect of acoustic treatment on fan
flutter stability. A three-dimensional model is proposed to predict
the flutter characteristics of a fan in a finite-length annular duct
partially lined with acoustic liners. In this model, the
aerodynamic couplings of rotor and liner are analyzed in a strict
sense by employing the transfer element method, with the
unsteady blade loading distribution predicted by the three-
dimensional lifting surface theory and the acoustically treated
wall modelled as the hard wall with the equivalent monopole
distribution. Moreover, combined with a boundary integral
approach, this model implicitly considers the reflections at duct
openings. Within the linear scope, the primary perturbations
induced directly by blade oscillation and the scattering fields
due to aerodynamic couplings are superposed to give the total
blade loadings, such that the flutter stability of the rotor can be
evaluated by its unsteady aerodynamic work based on the
energy method. Under certain circumstances, numerical
experiments indicate noticeable stabilizing or destabilizing
effect of both locally and non-locally reacting liner on rotor
flutter characteristics. And it has been theoretically confirmed
that cascade flutter can be suppressed by the proper
configurations of a kind of non-locally reacting liner with
controllable cavity geometry.
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An Immersed Boundary Method for Fluid-Structure-Acoustics Interaction at Low Reynolds Numbers
Li Wang, Fangbao Tian, Joseph C.S. Lai
[Abstract]
This paper presents an immersed boundary method for fluid–
structure–acoustics interactions involving complex boundaries
at low Reynolds numbers. In this method, the compressible
Navier–Stokes equations are considered, where a fifth-order
accuracy Weighted Essentially Non-Oscillation (WENO) scheme
and a fourth-order central difference scheme are adopted to
discretize the convective and diffusion terms, respectively. The
third order Runge-Kutta method is used for the temporal
discretization. The non-linear flexible structure immersed in the
fluid is numerically solved by using a finite element method. By
using a penalty immersed boundary method, the no-slip
boundary between the fluid and the structure is achieved.
Acoustic waves scattering benchmark problems and sound
generation by a micro flapping vehicle are presented to
validate the present solver. Comparisons with the published
data are presented to demonstrate the good performance of
the method in modelling acoustics. The capability of using this
method to study the sound generation by flapping plates is also
demonstrated.
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Theoretical Model for the Prediction of Sound Radiated from Unbaffled Long Enclosure with Ground Effect
Weiping Yang, Yat-Sze Choy
[Abstract]
An analytical model is presented for the prediction of
sound radiated from an unbaffled semi-infinite long
enclosure with ground effect. This geometrical
arrangement forms an idealized representation of traffic
facilities such as tunnel and railway station where sound
propagates along the enclosure and radiates to the
outside through the openings at two ends. Both the sound
field in and outside the enclosure should be accurately
predicted and thoroughly analyzed so as to determine an
appropriate noise control strategy. In this paper, the
Fourier transform technique and the mode matching
method are applied to reduce the intractable boundary
value problem into a scalar modified Wiener-Hopf
equation. The solution of which contains infinitely many
unknowns satisfying an infinite system of linear algebraic
equations susceptible to numerical treatment. Good
agreement is found between analytical and FEM solutions
in a wide frequency range. The far-field directivity patterns
are emphatically illustrated and the formation
mechanisms of lobes, zeros and the after radiation are
discussed in details. In the end, the advantage and
application prospect of the proposed model are
summarized.
Keywords: Sound radiation, unbaffled long enclosure,
ground effect, Wiener-Hopf technique
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Numerical and Experimental Investigation of the Generation of Wind Noise in Hearing Aids
Jörg Riedel, Florian Krömer, Hartmut Ritter, Kevin Bayer, Jonathan Tamil, Dietmar Lommel, Stefan Becker
[Abstract]
Hearing aid users are often confronted with the problem of wind noise during outdoor activities. The microphones installed in the hearing aids cannot distinguish whether the deflection of the diaphragm is caused by acoustic or hydrodynamic pressure fluctuations. This results in unpleasant background noise, which superimposes the acoustic signal. A dual approach consisting of CFD simulations using Star-CCM+ and measurements in an anechoic wind tunnel was used to investigate geometric features such as the shape of the housing and the rocker-switch with regard to their influence on wind noise generation. Using additive manufactured hearing-aid-prototypes, the input-related time-pressure signal at the microphone membrane was measured in the experimental approach. In the simulation, the corresponding signal was calculated using Delayed Detached Eddy Simulation. A comparison of the obtained pressure spectra from simulation and experiment shows very good agreement in the characteristic low-frequency range. The visualization of the numerical results provides information on which fluid dynamic effects, such as boundary layer separation and vortex formation, lead to the generation of wind noise. The results obtained are useful for estimating the performance of hearing aid models under the influence of wind at an early stage of product development.
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Experimental Investigation of the oscillating flow dynamics at the exit of regenerator meshes with different configurations
Islam Ramadan, Helene Bailliet, Jean-Christophe Valiere
[Abstract]
Regenerators (often consisting in stack mesh-wires) and heat
exchangers are the core of thermoacoustic devices and, as
such, key elements for the efficiency of these systems. In this
study, the oscillating flow velocity field at the exit of different
regenerators has been measured using Particle Image
Velocimetry (PIV) system. Three different mesh wire sizes have
been selected with respect to the size of the viscous penetration
depth. The influence of the randomness of stacking the mesh
wires on the oscillating flow dynamics has also been
investigated. Measurements are performed for different
acoustic levels. In the conference, the structure of oscillating
flow will be discussed, especially in terms of vorticity fields for the
different regenerators at different phases over the acoustic
cycle. The evolution from low to high acoustic amplitudes will be
presented. The generation of flow structures due to impinging of
vortices, formed during ejection phases, with the regenerator
edges will be highlighted.
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Organization:
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M. Guild, US Naval Research Laboratory, Code 7160 |
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An Integration Strategy for Acoustic Metamaterials to Achieve Absorption by Design
Ping Sheng
[Abstract]
While resonance-based acoustic metamaterials can display many novel wave manipulation capabilities, they have the Achilles’ heel of being narrow-frequency in character. It would be most desirable if a sound absorber can be designed to fit the noise spectrum, with a minimum allowed thickness. Such sound absorbing structures can now be realized through a design recipe that incorporates the causality constraint on the acoustic response. We use the causality constraint to delineate what is ultimately possible for sound absorbing structures, and denote those which can attain near-equality for the causality constraint to be ”optimal.” Anchored by the causality relation, an integration strategy can be formulated for realizing structures with target-set absorption spectra and a sample thickness close to the minimum value as dictated by the causality constraint. By using this approach, we have realized a 10.86 cm-thick structure that exhibits a broadband, near-perfect flat absorption spectrum starting at around 400 Hz, while the minimum sample thickness from the causality constraint is 10.36 cm. To illustrate the versatility of the approach, two additional optimal structures with different target absorption spectra are presented. This ”absorption by design” strategy would enable the tailoring of customized solutions to difficult room acoustic and noise remediation problems.
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Metafluid inspired by cereal straws for the perfect absorption of sound
Jean Philippe Groby, Weichun Huang, Logan Schwan, Vicente Romero García
[Abstract]
We present deep subwavelength perfect absorbers for the acoustic energy mimicking cereal straw bale, namely Metafluid inspired cereal straws (MCS). A straw bale is assumed to be constituted of tightly packed straws cut on either side of a node. This anisotropic metafluid efficiently combines visco-thermal diffusion from the stems and strong dispersion arising from the resonance of double quarter-wavelength resonators. We idealise this system by a closely packed square array of double quarter-wavelength resonators made of a open hollow cylinder divided by a rigid wall in the inner part. The viscothermal diffusion is further tailored by accounting for the corrugation arising from the Fused Filament Fabrication technology used to manufacture the MCS. Both the increase of the density of states at low frequency arising from slow sound regime created by the fundamental resonance of the double quarter-wavelength resonators and the attenuation enhancement arising from the highly concentrated arrangement together with the corrugations are advantageously used to design perfect absorber by critical coupling. A perfect absorption is theoretically and experimentally reported at 244 Hz, which corresponds to a wavelength equals to 13 times of the designed MCS thickness.
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Modeling and characterization challenges of multiple dynamics materials (also known as metamaterials)
Luc Jaouen, Fabien Chevillotte, François-Xavier Bécot
[Abstract]
We discuss some challenges related to the developments of
passive acoustical materials exhibiting multiple dynamic
behaviors (also known as acoustical metamaterials).
These challenges can be related to the characterization of
their elastic & acoustic intrinsic parameters or the
development of models. We first present a brief summary of the physical phenomena
used to enhance the absorption or insulation performances
of passive materials with highlights on some modeling
challenges. In a second part, we show examples of challenges related to
the characterizations of these materials. In particular, for
acoustic characterizations, we show how the dynamic mass
density & dynamic bulk modulus or wavenumber &
characteristic impedance are impacted by the
aforementioned physical phenomena. While the
measurement techniques of these dynamic properties do
not differ, parameter estimations are, of course, strongly
linked to the choice of the physical model used to estimate
them.
Some hints on the characterization in specific cases are
finally provided.
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Random Incidence Transmission Loss of Miniature Helmholtz Resonator Embedded Acoustic Meta-material
Jhalu Gorain, Chandramouli Padmanabhan
[Abstract]
An array of light-weight miniature Helmholtz resonators
(HRs), was shown to significantly enhance the low frequency
absorption, when embedded in a PU foam material; this test
was carried out in an impedance tube. This concept was
then used to study the sound absorption in a cylindrical
cavity. A series of tests were carried out by changing
number of HRs, embedded circumferentially in PU foam, at
various radial distances on top of the cavity, with a noise
source at the bottom. About 8-14 dB reduction in noise
levels was observed at the lower frequencies.
In this paper, the idea is now extended to examine the
random incidence transmission loss of such an array of
embedded HRs. Towards, this purpose a twin room
reverberation chamber with a 30 cm x 30 cm panel in
between is used; random incidence tests will be done for a
panel with 20 mm foam alone and the same foam with an
array of miniature resonators. The effectiveness of this
concept to increase transmission loss at lower frequencies
will be demonstrated. Also, by adjusting the resonator
frequencies, by intruding the neck into resonator cavity, s
shown to yield increased tranmission loss over a broader
range of frequencies.
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An Analytical Model for Broadband Sound Transmission Loss of a Finite Single Leaf Wall Using a Two Degree of Freedom Resonant Metamaterial
Javier Hernan Vazquez Torre, Jonas Brunskog, Vicente Cutanda Henriquez
[Abstract]
Acoustic metamaterials (AM) have emerged as an academic
discipline within the last decade. Metamaterials can exhibit high
transmission loss at low frequencies despite having low mass per
unit area. This paper investigates the possibility of using AMs for
increasing the sound insulation of finite single leaf walls (SLW),
focusing on the coincidence effect problem. Formulas are
derived using a variational technique for the forced sound
transmission of finite SLW with a coupled array of two degree of
freedom resonators. An analytical model is presented and the
effects of the band gap in sound transmission and radiation are
analyzed and compared to the single degree of freedom case.
Moreover, numerical simulations verify both the single and two
degree of freedom models. Finally, some conclusions are drawn
regarding the effectiveness of the proposed model, possible
applications, and future work.
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Topological bound states in mechanical graphene
Johan Christensen
[Abstract]
The discovery of topologically non-trivial electronic systems has opened a new age in condensed matter research. From topological insulators to topological superconductors and Weyl semimetals, it is now understood that some of the most remarkable and robust phases in electronic systems (e.g. Quantum Hall or Anomalous Quantum Hall) are the result of topological protection. In this work we demonstrate the acoustic analogue of a topologically bound state, a different class of non-propagating protected state that cannot be destroyed by local perturbations.
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Experimental Observation of Topological Fano Resonances for Audible Sound
Farzad Zangeneh-Nejad, Romain Fleury
[Abstract]
The Fano resonance is a widespread wave scattering phenomenon
observed in many different systems, from cold atom physics, to
electromagnetics, electronic circuits, and acoustics. It is characterized
by a very sharp and asymmetric scattering cross-section spectrum that
switches from a zero to a maximum value for two close by frequencies.
Such extraordinary feature has established Fano resonances as the basic
principle for many applications including efficient sources and emitters
[1], switches [2], interferometers, or supersensitive sensors [3]. Yet, these
resonances are difficult to achieve in practice, as they require tight
geometrical tolerances and large fabrication precision to guarantee
their occurrence in the desired frequency range, and the absence of
parasitic peaks. In this talk, we will discuss a route for leveraging one-
dimensional topological insulators to generate a novel form of sturdy
Fano resonances, and experimentally demonstrate their robustness for
audible sound waves [4]. Extension to other physical systems will be
briefly discussed. 1. K. Nozaki, et al. Optics express 21.10 (2013): 11877-11888.
2. S. Chua, et al. Optics express 19.2 (2011): 1539-1562.
3. C. Wu, et al. Nature materials 11.1 (2012): 69.
4. F. Zangeneh-Nejad, and R. Fleury, Physical review letters. 122 (2019):
014301
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Organization:
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M. Guild, US Naval Research Laboratory, Code 7160 |
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Non-reciprocity in discrete and continuous lattice systems via mechanical modulation
Michael R. Haberman, Benjamin Goldsberry, Samuel Wallen
[Abstract]
Materials with time- and space-dependent effective
properties are of significant interest as a means to
introduce non-reciprocal wave propagation in
engineered material systems. Materials displaying this
unique behavior have significant potential as direction-
dependent devices for mechanical wave sensing and
transmitting capabilities. They also open up the possibility
for the construction of novel materials or structures that
isolate unwanted vibration. Recent analytical models of
spring-mass chains have shown that external application
of a nonlinear mechanical deformation, when applied on
time scales that are slow compared to the characteristic
times of propagating linear elastic waves, may induce
non-reciprocity via changes in the apparent elastic
modulus for perturbations around that deformation
[Wallen et al., Phys. Rev. E., 99, 013001, (2019)]. This work
investigates direct numerical simulation of discrete lattices
and a finite element approach for continuous elastic
systems to simulate elastic wave propagation in
mechanically-modulated metamaterials. Analysis of the
degree of non-reciprocity when varying sub-wavelength
geometry or geometric modulations will be shown. Of
specific interest is the use of FEM to investigate non-
reciprocity in elastic lattices consisting of unit cells with
varied geometric asymmetry and more general forms of
mechanical modulation.
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Sound Insulation using Metaatoms with Willis Coupling
Anton Melnikov, Sebastian Oberst, David Powell, Steffen Marburg
[Abstract]
Willis coupling is a phenomenon, which can increase the effectivity of acoustic metamaterials. We present a concept for sound insulation based on metaatoms with strong Willis coupling. The metaatom geometry is simple and easy to realize in industrial applications. Furthermore, available analytical formulation of its polarizability allows tuning the operating frequency and the Willis coupling magnitude to desired values. One of the outcomes is a metacapsule, which can be beneficial for example in stage machinery applications, where common noise control solutions were not successful. We investigated the transmission properties of an exemplary metacapsule during experiments and using simulation techniques.
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Vibration control by metamaterials consisting of lattices of resonant cavities in a thin plate
José Sánchez-Dehesa, Penglin Gao, Alfonso Climente
[Abstract]
It is shown that crystals made of 1-beam cavity resonators produce low-frequency omnidirectional bandgaps for flexural waves propagating in thin plates. Based on this property, hardly obtained for resonant cavities containing a higher number of beams, we have designed single-phase metamaterial plates for the suppression of low frequency flexural waves in a broad range of frequencies. These metamaterials are obtained by using resonant cavities containing a multiple number M of identical 1-beam resonators uniformly distributed in the cavity. Square lattices of this type of resonators have been studied by using the impedance matrix approach and the multiple scattering method. For the case M =4, we designed composite structures consisting of platonic crystal slabs with complementary bandgaps that easily merge into a broadband wave attenuation region. The normalized width, defined as the percentage of the bandwidth to its central frequency, reaches 95.3%, representing an enhancement of about one order of magnitude compared with the absolute bandwidth obtained for the case of a single 1-beam resonator in the cavity. The gaps can be tuned to lower frequencies by changing the geometrical parameters, such as the length of the beam, the radius and thickness of the smaller circular plate.
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Design of a Resonator-Based Metamaterial for Broadband Control of Transverse Cable Vibration
Lawrence Singleton, Jordan Cheer, Stephen Daley
[Abstract]
Cables are widely used to bear loads, tether objects or
transmit data over a distance. Vibration in cables can cause
excessive wear, acoustic radiation and for sensing arrays,
measurement error. Although a variety of control methods
have been proposed, metamaterials potentially offer a high
level of performance within limited weight constraints. This
paper investigates the design of a metamaterial consisting of
single-degree-of-freedom (SDOF) resonators modelled as
mass-spring-dampers, evenly distributed along a one-
dimensional string model. An initial investigation is
conducted into systematic tuning approaches for distributed
arrays of resonators, with simulations showing significant
reductions in the energy of the cable around the resonance
frequencies. In practice, the characteristics of the cable may
change over time and reduce the effectiveness of the
developed configuration. The robustness of the systematic
design approaches to variations in the cable tension are
investigated and show a reduction in the effectiveness of the
metamaterial. To overcome these practical limitations,
approaches to the robust optimisation of both the tuning and
location of the resonators to control the kinetic energy of the
vibrating cable are investigated and compared to the
systematic methods.
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Airborne and ultrasonic characterisation of acoustic surface waves on structured plates
Tim Starkey, Alastair Hibbins, J Roy Sambles, John Smith
[Abstract]
Acoustic ‘metamaterials’ offer a route to control, guide, or otherwise
manipulate the propagation of acoustic energy for a range of
engineering applications. One class of these materials is the two-
dimensional ‘metasurface’; these are two-dimensional structures,
usually plates or surfaces, that are adorned with periodic patterns of
resonators for the control of radiated sound and bound acoustic
surface waves (ASWs). In this paper, we will show the characterisation of resonant-cavity-
based structured surfaces for the control of sound in air and
underwater. Aluminium plate samples are drilled with hole arrays and
experimentally characterised by spatially-mapping the near acoustic
field of the sample surface following excitation from a point-like
source. Experimental results show that bound acoustic modes are supported
on these surfaces, and can exhibit acoustic beaming and zero-group
velocity. Airborne ASW characterisation of the dispersion relation
show good agreement with FEM simulations when the sample surface
is considered acoustically rigid. Underwater characterisation shows
good agreement only when the samples elastic properties are
accounted for. Having demonstrated these bound acoustic modes experimentally, we
show some new ideas towards coupling them to turbulent flow.
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Design Of Multi-directional Acoustic Cloaks Using Two-dimensional Shape Optimization And The Boundary Element Method
Peter Risby Andersen, Vicente Cutanda Henriquez, Lorenzo Sanchis, José Sánchez-Dehesa
[Abstract]
Acoustic cloaking is a technique that seeks hiding objects in a sound field by reducing or cancelling their scattered sound pressure. Incident waves are restored to as close as possible their original undisturbed form after hitting the cloaked object. One technique for achieving this goal is the design of additional scatterers around the object, which, properly distributed and shaped, can create cloaking at the design frequency. A newly developed numerical technique combining the Boundary Element Method (BEM) with shape optimization is applied in this work for two-dimensional cloaking of a cylinder. The shapes of the scatterers are optimized for the cloaking of the whole setup with waves impinging on the cylinder from several different directions. The results show a measure of the amount of cloaking depending on the direction at the range around the design frequency. The optimization results are compared with experimental work made for one-directional cloaking. The impact of visco-thermal acoustic losses in the cloaking design is also evaluated by means of a BEM implementation with losses.
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Organization:
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A. Morgans, Imperial College London |
D. Yang, Imperial College London |
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Experimental investigation of intrinsic thermoacoustic instabilities in a combustion chamber terminated by a variable aperture
Manmohan Vishwakarma, Sathesh Mariappan, Maria Heckl
[Abstract]
Combustion instabilities arise due to flame-acoustic coupling. In
experimental studies using laboratory test rigs, the exhaust is
typically an open tube end. Acoustic waves are reflected at
such an end with only minor losses and form a standing wave
that interacts with the flame. If the exhaust is fitted with an
aperture, the acoustic losses are enhanced, and intrinsic
thermo-acoustic modes are likely to manifest themselves. This
paper presents an experimental study where the area ratio AR
(area of aperture to cross-sectional area of combustion
chamber) is used as a bifurcation parameter, and the
frequency spectra of the acoustic field are recorded. AR is
varied in the range from 1 (fully open end) to 0.05 (nearly
closed end). Another parameter of interest is the mean flow
velocity. The test rig is a swirl-stabilised premixed combustor with
a settling chamber, burner and combustion chamber. The fuel is
Liquefied petroleum gas. From the properties of the observed
spectra, and from the measurements of the reflection
coefficient at the exhaust, a network model is developed. This
makes it possible to predict which type of mode will occur, and
to calculate its frequency and growth rate.
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The acoustic absorption coefficient of short circular holes sustaining a high Reynolds number bias flow
Renaud Gaudron, Aimee Morgans
[Abstract]
The damping of acoustic waves across short circular holes sustaining a high Reynolds number flow is of interest in many engineering applications, such as aircraft and rocket engines liners. Vortical structures are shed periodically from the perforation rim for these geometries, with the interaction between the convected vortical structures and the acoustic flow field leading to amplified or damped acoustic oscillations within the system. While many theoretical models describing the interaction between the acoustic waves and vortical structures have been proposed since the pioneering work by Howe in 1979, the different models have not previously been interrogated in terms of the overall proportion of acoustic energy absorbed - known as the acoustic absorption coefficient. The objective of this work is to determine and compare the acoustic absorption coefficient across various models of a short circular hole sustaining a high Reynolds number bias flow. The dependence on parameters such as the mean Strouhal number and mean Mach number is discussed and comparisons across the various models performed. Finally, conditions leading to the amplification or damping of acoustic energy are derived for each model, with physical reasons for differences discussed.
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Characterization of the aeroacoustic instability in a T- junction
Claire Bourquard, Abel Faure, Nicolas Noiray
[Abstract]
Aeroacoustic resonance at a T-junction is a well-known
phenomenon that has been studied using a variety of
different approaches. These include vortex-sheet
modelling of the shear layer and Linearised Navier-Stokes
Equations (LNSE) incorporating a turbulence model. The
aeroacoustic instability results from the constructive
interaction of the acoustic waves in the cavity and the
turbulent shear layer. In the present study, measurements
of the acoustic reflection coefficients of the shear layer
and of the cavity are performed at different frequencies
and amplitudes. These measurements are used to derive
a nonlinear model of the aeroacoustic instability and
allow for analysis of linear stability and of nonlinear
dynamics.
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Acoustic scattering in arrays of orifices, slits and tube rows with mean flow: A comparison
Charles Boakes, Aswathy Surendran, Dong Yang, Aimee Morgans
[Abstract]
Thermoacoustic phenomena result from positive feedback
between acoustic fluctuations and fluctuations of heat transfer
rate (release or absorption). The thermoacoustic response of
combustion system heat exchangers in cross flow is assumed to
be dominated by two effects: acoustic scattering at the heat
exchanger tube row and heat transfer from the tube row. While
the latter causes a significant reduction in the net energy of the
flow, the influence of both the effects on the acoustic
fluctuations needs to be addressed carefully to predict the
thermoacoustic response of the heat exchanger. To this end,
we adopt a sequential approach where the acoustic scattering
and the heat transfer are assumed to be independent of each
other. In the present work, we focus on the acoustic scattering
behaviour of heat exchanger tubes (only) by comparing
existing acoustic models for orifices, slits and tube rows and
identifying those operating regimes where the different models
are valid and overlap. This will enable us to formulate
appropriate acoustic models for the various operating regimes
of the heat exchanger tube rows.
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Prediction of Acoustic Response of Tube-rows with Bias-flow Using Linearized Navier-Stokes Equations in Frequency Domain
Wei Na, Susann Boij, Aswathy Surendran, Dong Yang, Aimee Morgans
[Abstract]
Heat exchangers are widely used in the industries where the heat
needs to be transferred from one fluid to another fluid. For example,
there are plate-fin heat exchangers in gas turbine engines, shell and
tube heat exchangers in oil refineries and tube bundle heat
exchangers in domestic boilers, etc. Amongst the different types of
exchangers, tube bundle exchangers are most commonly used heat
exchange equipments. In this paper, the heat exchanger with the
structure of tube row with bias flow and possible acoustic absorption
at tube row is studied for the acoustic reflection and transmission.
The numerical methodology solving the linearized Navier-Stokes
equation in the frequency domain is used. It has the advantage of
taking into account the flow effects, viscous losses as well as thermal
losses in the acoustic propagation. The simplified geometry for the
heat exchanger investigated in this paper is a two-dimensional
rectangular duct with two half cylinders with a bias flow going
through the gap between the cylinders. In the numerical study, the
acoustic response is predicted with and without heat transfer,
respectively, and compared to the experimental data.
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An impedance model for thin microperforated panels
Xianhui Li, Tuo Xing, Liying Zhu, Congshuag Jiang, Wenjiang Wang, Bin Zhang
[Abstract]
The classic model for the sound absorption of
microperforated
panels (MPPs) is well developed. For an ultrathin MPP,
however,
interaction between the air flow on both sides of the panel
is not
negligible, which invalidates the impedance end
corrections in
the classical model. In this paper, a correction length
model is
proposed for the transfer impedance of an ultrathin MPP.
Theoretical and numerical analyses show that the
impedance
jumps take place within the distance less than one half of
the
perforation radius. Provided that the panel thickness is
larger
than the perforation radius, contribution from the end
effect can
be well approximated by adding 1.2 times of the radius to
the
thickness. Sound induced vibration also significantly affects
the
MPP’s impedance. Herein averaged air velocity within a
circular
perforation is solved under the non-slip boundary condition
exerted by the panel vibration velocity. Together with the
flow
continuity at the panel surface in a spatially mean sense,
the
overall acoustic impedance of the MPP is derived. The
experimental results for two different MPP configurations
validate
the model.
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Sound generation by entropy perturbations passing through short circular holes
Dong Yang, Juan Guzmán, Aimee Morgans
[Abstract]
It is well known that unsteady combustion can generate entropy fluctuations (also known as temperature variations or “hot spots”). These entropy perturbations remain silent when advected by a non-accelerated mean flow, but generate additional sound when accelerated. In many laboratory-scale experiments which study this phenomenon, mean-flow acceleration is achieved using a nozzle or a sudden contraction of the flow area such as a hole. Several analytical models that predict entropy noise in nozzles exist for a wide range of parameters. However, these models, which are often based on inviscid flow, and plane acoustic and entropy wave assumptions, are believed to be inadequate for holes. This is because when the entropy wave interact with a short circular hole, oscillating vorticity (shed from the hole inlet edge) is generated and the entropy wave could also become non-planar – these may strongly change the sound generation. In the present work, an analytical model based on the Green’s function method, which can consider the coupling of acoustics, entropy and vorticity, is developed to predict the sound generated by entropy perturbations passing through short circular holes. Numerical simulations which introduce low-amplitude entropy fluctuations upstream of the hole are also performed to validate the model.
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Organization:
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J.B.W. Kok, University of Twente |
M. Heckl, Keele University |
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Background-oriented Schlieren of entropy waves advected in straight duct
Markus Weilenmann, Yuan Xiong, Nicolas Noiray
[Abstract]
Entropy waves generate indirect combustion noise when
accelerated in the turbine stages of aeroengines. This
indirect combustion noise constitutes a significant part of the
overall combustion noise. In addition, the indirect
combustion noise from entropy waves can constructively
interact with flame and induce high amplitude self-sustained
oscillations. A deep understanding of the noise generation at
the vane is therefore very important for the prediction of
noise emissions and thermoacoustic instabilities. This
experimental study contributes to that goal: a highly
turbulent airflow in a rectangular duct at ambient
temperature is subject to the forcing of a siren supplied with
hot air. This pulsed hot-jet-in-cold-crossflow yields coherent
entropy waves. After advection through the channel, these
hot spots are accelerated in a nozzle and produce indirect
noise. The line-of-sight integrated temperature amplitude
and phase are measured using high-speed Background-
oriented Schlieren (BOS), thermocouples and simultaneous
particle image velocimetry (PIV). This non-invasive method
facilitates the observation of low amplitude temperature
fluctuations at high frequencies, with a simple and
inexpensive setup. The noise generated by the passage of
the entropy waves through the vanes is quantified as
function of temperature fluctuations amplitudes and
frequencies.
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Study of the Relation between Entropy Flux Density Production and Thermal Efficiency of a Thermoacoustic Engine
Mariko Senga, Shinya Hasegawa
[Abstract]
When the traveling wave propagates in a bundle of narrow tubes (the regenerator) whose channel radius is smaller than thermal penetration depth and experience a positive temperature gradient, the thermodynamic cycle of working gas in the regenerator undergoes the isothermal reversible thermodynamic cycle similar to that of Stirling cycle. The thermoacoustic engine is classified as a type of heat engine which has high thermal efficiency theoretically. However, in actual, it is difficult to execute the isothermal reversible process because of the irreversibilities such as dissipation and heat transport from hot to cold by oscillating flow in the regenerator. These irreversibilities causes decrease of thermal efficiency. These irreversibilities depends on two parameters: (1) the ratio of channel radius to the thermal penetration depth, and (2) the specific acoustic impedance at the regenerator. By setting these two parameters appropriately, irreversibilities can be reduced. In this study, the entropy flux density production is focused as an indicator of the reversible and irreversible characteristics. The relation between the entropy flux density production and thermal efficiency of the regenerator is investigated by changing two parameters both of (1) and (2).
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Experiments on self-excited thermoacoustic oscillations in an air-filled closed tube
Nobumasa Sugimoto, Keisuke Minamigawa
[Abstract]
This paper is a counterpart of the previous one in which self-excited thermoacoustic oscillations in a looped tube are examined experimentally and discussed (J. Phys. Soc. Japan, 87, 104401 (2018)). Using a closed straight tube filled with atmospheric air and installed a stack subjected to a temperature gradient, onset of thermoacoustic instability and emergence of self-excited oscillations are investigated experimentally. Effects of a pore radius in the stack are examined for four stacks of 50, 100, 200 and 300 cells per square inches. The onset of instability is discussed in comparison with the marginal conditions derived by using Rott’s equation. It is found that the temperature ratios at hot-to-cold ends of the stack are higher in all cases than the ones predicted by the marginal conditions. The pressure profiles in the steady-state oscillations exhibit typical characteristics of nonlinear oscillations, i.e. sharp crest and round trough due to many higher harmonics of oscillations, but no shocks appear. A mean pressure is higher near the closed ends. Among the four stacks, it is observed that the peak-to-peak pressure at the closed end takes the maximum 8% relative to the atmospheric pressure in the case of the stack of 100 cells.
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Amplitude Death in Coupled Thermoacoustic Oscillators with Frequency Detuning
Hiroaki Hyodo, Tetsushi Biwa
[Abstract]
Thermoacoustic oscillation is a harmful vibration in
combustors of a gas turbine engine because it can
damage the combustors. Development of the suppression
method of thermoacoustic oscillation is urgent necessity.
Recently, the coupling-induced amplitude death was
reported experimentally and numerically in
thermoacoustic oscillators of Sondhauss type. Since the
coupling was established only by using hollow tubes that
connect two oscillators, this method can be a simple and
reliable method to stop the oscillations, compared to
active control method involving electronics and acoustic
drivers.
In this study, the tube coupling is further tested for the
coupled thermoacoustic oscillators with frequency
detuning, in order to explore the possibility of stronger
suppression effect. The amplitude death region was
numerically obtained based on hydrodynamic equations,
which agreed with experimental results. We found that the
amplitude death was realized with much smaller tube
diameter than that of coupled thermoacoustic oscillators
with no detuning. Also, we found that the tube length
resulting in the amplitude death was different from that of
coupled identical oscillators.
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A semi-analytical solution for acoustic wave propogation in varying area ducts with mean flow
Saikumar Reddy Yeddula, Aimee Morgans
[Abstract]
Semi-analytical acoustic solution for the propagation of plane sound waves in a varying
area duct for isentropic and non-isentropic cases are presented. The duct also sustains a
mean flow and axial gas property gradients. A second order differential equation(ODE)
in frequency domain is obtained from linearised Euler equations, upon neglecting the
communication between acoustic and entropy disturbances, for each case. This ODE,
with varying coefficients, is solved using adaptive WKB approximation technique, and
the obtained wave-like solution consist of an asymptotic expansion, expressed as
superposition of downstream and upstream propagating wave components. Frequencies
considered here are not very high to assure that acoustic field is predominantly one-
dimensional and diffusive effects are negligible, but large enough for WKB to be valid.
Two industrial gas turbine combustor outlet converging sections with a temperature
gradient and a high mean-flow acceleration are considered as test cases. Analytical
results are compared to two numerically solved solutions: (1) three linearised Euler
equations(3LEE), considering acoustic-entropy coupling; (2) two linearised Euler
equations(2LEE) assuming acoustic and entropy waves as independent, and thus effect
of entropy disturbance is also studied. Semi-analytical solutions agree well with 2LEE
results in low mach-number regions when Documaci’s limiting frequency criteria are
satisfied.
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Organization:
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M. Heckl, Keele University |
X. Wang, Beihang University |
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Extensions of the Born Approximation for Acoustic Radiation Force and Torque to Inhomogeneous Objects and Progressive Spherical Waves
Thomas S. Jerome, Mark F. Hamilton
[Abstract]
A simple volume integral based on the Born approximation was
developed for the acoustic radiation force and torque exerted
by a plane standing wave on objects of arbitrary shape
[Jerome et al., J. Acoust. Soc. Am. 145, 36-44 (2019)]. The
principal restriction is that their material properties are similar to
those of the surrounding fluid, in which case the approximation
is reasonably accurate for objects with dimensions up to about
one wavelength. Results were presented for spheres, cylinders,
and prolate spheroids with homogeneous material properties.
Here, closed-form expressions based on the Born approximation
are presented for spheres and cylinders with compressibility and
density that vary either linearly or discretely within the object.
The expressions reveal the dependence of the radiation force
and torque on the position and orientation of the object with
respect to an incident standing wave. Second, the validity
of the Born approximation for the radiation force on a
homogeneous compressible sphere as a function of its distance
from the center of a diverging or converging spherical wave is
assessed by comparing the closed-form expression obtained in
the Born approximation with the full solution based on spherical
harmonic expansion of the incident and scattered
fields.
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Impact of roughness on shock wave reflection phenomena
Thomas Lechat, Sébastien Ollivier, Didier Dragna, Maria Karzova
[Abstract]
Reflection of pressure shock waves from a flat surface can lead to a 3-shock reflection pattern, called Mach reflection, with reference to Ernst Mach, who provided evidence for this non-linear interaction. This phenomenon has been widely studied in cases of supersonic jets and high-amplitude overpressures generated by moving supersonic bodies or blast waves. In the case of weak shocks (Mach number < 1.47), 3-shock reflection patterns are also observed, and in particular in the case of acoustic shockwaves. If the irregular reflection of shocks from a flat surface is well known, the effects of roughness have been much less studied. In this work we performed numerical simulation of shock propagation over periodic and random surfaces and compared results to experiments. Numerical simulations are based on the temporal integration of axi-symmetric curvilinear Euler equations. Simulations are compared to Schlieren visualizations of the reflection of spark generated N-waves over sandpaper.
The results show that near a rough surface, the pressure level is higher than in the case of a flat surface and the pressure decrease above the surface is also changed. The method will also be applied at larger scales, for example for sonic boom or blast waves reflection over the ground.
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Analytical and computational modeling of viscothermal acoustic damping in perforated microstructures
Vahid Naderyan, Richard Raspet, Craig Hickey, Mohammad Mohammadi
[Abstract]
Predicting the viscothermal acoustic behavior of the
microstructures is crucial in the design of micro-electro-
mechanical systems (MEMS). In MEMS structures the
dimensionless shear wave-number is typically smaller than
one. Therefore, the viscous effects are larger than inertial
effects and are the dominant part of the impedance. In
this work, an analytical solution for the viscothermal
acoustic impedance for perforated microstructures used
in MEMS devices is developed based on the low reduced-
frequency (LRF) method. This solution is based on a full-
plate approach, as opposed to the single-cell approach,
which includes all viscous and thermal losses as well as
compressibility and inertial effects. Additionally, a 3D
viscothermal acoustic model using finite element method
(FEM) is developed to solve the problem numerically for
the case of a MEMS microphone. The results of the
analytical LRF solution are in good agreement with the
FEM results. The results can be applied in designing MEMS
devices to minimize the damping and to optimize the
acoustic performance of the MEMS devices.
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On some properties of magnetoacoustic waves in acoustically active non-adiabatic plasma
Dmitrii Zavershinskii, Nonna Molevich, Igor Zavershinskii
[Abstract]
On some properties of magnetoacoustic waves in
acoustically active non-adiabatic plasma Magnetoacoustic(MA) waves in the plasma with
acoustic activity caused by the non-adiabatic
processes is under investigation. Acoustic activity of the
medium is a consequence of temperature and density
dependence of heating and cooling process which
take place in the medium. It is shown that non-
adiabatic process results in the frequency dependence
of group/phase velocity. Effect of frequency
dependence is most pronounced near the frequency
defined by the inverted heating/cooling time. In the
low-/high-frequency limits effect of frequency
dependence can be neglected. However, in contrast
to high-frequency case where phase/group velocities
equal to their value in medium without non-adiabatic
process,
in the
low-frequency limit both velocities are defined by the
non-adiabatic processes only. Furthermore, frequency
dependence of group velocity in contrast to phase
velocity has as an extrema which can be maxima or
minima depending on type of dispersion (negative or
positive, respectively). This result indicates that some
harmonics in non-adiabatic plasma can propagate
faster or slower than all others. The expression for group
velocity has been obtained under the assumption of
weak dispersion/dissipation. Effects caused by the weak
but finite dispersion/dissipation on group velocity is
analyzed as well.
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Statistical Characteristics of gas metal arc welding (GMAW) sound
Sipei Zhao, Xiaojun Qiu, Ian Burnett, Malcolm Rigby, Anthony Lele
[Abstract]
Gas Metal Arc Welding (GMAW) is an arc welding process to
join two or more metal materials through fusion, where an
electric arc is formed between a consumable electrode and
the base metal. It has been reported that expert GMAW
welders are able to maintain and direct the welding arc based
on the welding sound, and psychoacoustic experiments
showed that the welding performance significantly degraded
without the acoustic feedback to the professional welders.
However, the mechanics of sound production in the GMAW
process has not been understood well. In this paper, a multi-
sensor system is established to measure the welding sound
simultaneously with multiple physical welding parameters to
investigate the generation mechanism of welding sound. An
acoustic model is proposed to predict the welding sound signal
from the physical welding parameters. The current work
enhances the understanding of the welding sound production
mechanism.
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Predicting the noise level during sawing of carbonate rocks from the P-wave velocity
Sair Kahraman, M. Suat Delibalta, Ramazan Comakli
[Abstract]
Millions of employees in the World are exposed to noise at work.
Noise affects humans both physically and psychologically,
whose impacts vary from person to person. Block cutting
machine is one of the most important noise sources in the stone
processing factories. The block cutting machine is one of the
most important noise sources in the stone processing factories.
In this study, the predictability of the noise level during sawing of
carbonate rocks such as limestone, travertine and marbles from
the P-wave velocity in laboratory scale. First, the P-wave
velocities of rock samples were measured in the laboratory.
Then, the core samples were cut by an automatic cutting
machine with diamond saw in the laboratory and noise levels
were measured. A strong linear correlation was found between
the noise levels and the P-wave velocity. After including the
densities of the samples to the analysis, the correlation
coefficient was increased significantly. It is concluded that the
noise level of carbonate rocks can be estimated using the
derived equations from the P-wave velocity.
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Organization:
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L.H. Carney, University of Rochester |
I. Bruce, McMaster University |
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Ray Meddis: A Model Scientist
Christian Sumner
[Abstract]
Ray was a scientist in the classical sense: a creative
intellect who sought the truth. He was also man who had
a profound impact on the career of many hearing
scientists. Ray believed that to understand hearing it was
necessary to replicate its behaviour in a computer model.
His first major contribution in hearing was the ”Meddis
Haircell”, and a series of papers exploring the adaptation
of firing rate in auditor nerve fibres. These were
accompanied by models of stream segregation, of pitch,
the segregation of concurrent vowels, numerous models
of the processing of sound by neurons in the auditory
brainstem, and a non-linear model of the basilar
membrane. In later years, Ray became increasingly
interested in hearing loss, and the role of the olivocochlear
system. Technolology also deserves a special mention:
Ray’s models were implemented in silicon, evaluated
using automatic speech recognition, and even inspired a
smart-phone based hearing aid. Ray lives on in his work,
but also his colleagues: those of us that learnt from him,
worked with him, disagreed with (yes!), were influenced by
his work, and shared his passion for models of the auditory
system.
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Electrocochleography Predictions using a Combined Model of Acoustic Hearing and Electric Current Spread in the Cochlea
Margriet Van Gendt, Kanthaiah Koka, Randy Kalkman, Jeroen Briaire, Leonid Litvak, Johan Frijns
[Abstract]
Electrocochleography (ECochG) through the implant is
advocated as a tool to monitor residual hearing during
cochlear implant surgery. At present it is not known
exactly how hair cell and neural activity contribute to
ECochG responses detected from different locations in
the cochlea. In the current study a model is evaluated
that simulates CM responses to various stimuli at different
intra-cochlear locations. Electric-field spread of electric
dipoles, representing inner and outer hair cells, at 3200
spatially different locations on the basilar membrane was
calculated using a 3D volume conduction model
(Kalkman et al., 2015). Hair cell activation was calculated
using a phenomenological model of the auditory
periphery (Bruce et al., 2018). Both models were
combined to predict the ECochG. The responses over the
array show a narrowing peak that shifts basally with
increasing stimulus frequency. The fact that CM responses
to low-frequency tones can be recorded along the whole
array is a result of cross-turn sensitivity rather than from
excitation of basal hair cells. These simulations show that
the intracochlearly recorded ECochG is a measure of very
local activation, while cross-turn sensitivity broadens
ECochG activation for lower frequencies. Therefore,
ECochG is applicable to measure acoustic tuning curves
of the implanted cochlea.
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Application of a Computationally Efficient Coincidence Detector Model to Simulate Auditory Brainstem Neurons
Go Ashida, Mathias Dietz, Jutta Kretzberg
[Abstract]
Modeling approaches play an important role in the investigation of acoustic information processing along the auditory system. Recent studies found that a relatively simple neuronal model, which produces action potentials depending on the count and timing of modeled synaptic inputs, can simulate a number of response features of auditory brainstem neurons, such as interaural time and intensity difference coding in the lateral superior olive. In this presentation, we review this simple coincidence detector model and discuss its possible applications to several types of auditory neurons. Our applications include principal neurons in the medial superior olive that change their number of action potentials according to interaural time differences and globular bushy cells (GBCs) in the anteroventral cochlear nucleus that send output to the superior olivary complex. By additionally employing an adapting threshold, the model can not only simulate the enhancement of phase-locking relative to auditory nerves but also primary-like-with-notch peristimulus time histograms that are regarded as one of the most important characteristics of GBCs found in vivo. Because of its computational efficiency, the model is also suitable for large-scale simulations that typically include thousands of neurons to study, for example, across-frequency integration and the resulting perception of binaural sounds.
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A model of the prespike of the calyx of Held synapse in the auditory brainstem
Gerard Borst, Martijn C. Sierksma
[Abstract]
Each principal neuron in the medial nucleus of the trapezoid body of the auditory brainstem is typically contacted by a single calyx of Held which covers up to 50% of its somatic surface. The action potentials of this giant terminal can be readily picked up during whole-cell recordings of a principal neuron, and these so-called prespikes have been viewed as a hallmark for the presence of the calyx. We constructed a biophysical model of the calyceal prespike in which the release face was modelled as a current source that was based on the known properties of the calyceal voltage-dependent ion channels. Further, the model consisted of a leak resistance for the synaptic cleft and a passive postsynaptic cell. The model could accurately reproduce size and time course of the prespike. In addition, the model allowed us to systematically test the impact of changes in calyceal properties - contact area, AP waveform, voltage-dependent channel densities - on the amplitude and kinetics of the prespike, and to compare this to experimental manipulations. These results provide support for the idea that the calyceal prespike is a sensitive indicator of capacitive and ionic currents at the release face.
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Detecting interaural incoherence based on variations in the hemispheric balance
Jörg Encke, Lucy Anderson, Werner Hemmert, David McAlpine, Torsten Marquardt
[Abstract]
Human listener demonstrate a remarkable sensitivity in detecting even
slight incoherences within signals presented at the two ears. These
interaural decoherences can also be described as time and frequency
dependent variations in both the interaural phase and level difference
which are important cues for sound localization. Traditionally, the
ability to detect interaural incoherences is explained using delay
line models where the incoherence results in a reduction in the
maximum of the cross-correlation function. Conversely, by analyzing
single-cell recordings from neurons in the inferior colliculus of
anesthetized guinea pigs, this study finds that interaural incoherence
cannot be reliably detected when applying this mechanism. Instead, an
alternative detection mechanism based on a hemispheric balance model
is proposed, where stimulus power fluctuations result in covariant
fluctuations in the two hemispheres so that the balance remains
unchanged, while fluctuations in interaural differences result in
opposing fluctuations. The proposed mechanism thus considers the
disturbances in the hemispheric balance. By re-analyzing the recorded
neuronal responses for variations in the hemispheric balance, it is
shown that this approach considerably improves the performance of
detecting interaural incoherence.
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Predicting speech intelligibility in normal-hearing and hearing-impaired listeners based on a physiologically inspired model of the auditory periphery
Johannes Zaar, Torsten Dau, Laurel H. Carney
[Abstract]
The present study extends and evaluates a speech intelligibility
prediction model for normal-hearing (NH) and hearing-
impaired (HI) listeners recently introduced by Scheidiger et al.
[(2018). Acta Acustica united Ac. 104, 914-917]. The model
combines a non-linear model of the auditory periphery with a
decision process that is based on fluctuation profiles across
characteristic frequency (CF) after modulation analysis in the
range of the fundamental frequency of speech. Specifically,
the short-term across-CF correlation between noisy speech and
noise alone is assumed to be inversely related to speech
intelligibility.
The model has been shown to account for NH listeners’
speech intelligibility in conditions with stationary and fluctuating
interferers. Furthermore, some of the main effects measured in
HI listeners were predicted by adapting the peripheral model
using a simplistic interpretation of the listeners’ hearing
thresholds. Here, the effects of various impairment factors on
the model performance were explored to better account for
the individual HI listener, focusing particularly on the ratio of
inner and outer hair cell loss and on the modulation-frequency
tuning assumed in the model. The results are discussed
regarding the validity of the individual best-matching model
configurations and in relation to a further validation using data
from auditory profiling.
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Hearing-impaired sound perception: What can we learn from a biophysical model of the human auditory periphery?
Alejandro Osses, Frauke Ernst, Sarah Verhulst
[Abstract]
Auditory modelling provides a powerful quantitative framework to study performance in psychoacoustic listening tasks such as signal or modulation detection, and speech perception. Biophysically-inspired auditory models provide the straightforward possibility to account for different aspects of sensorineural hearing loss (e.g., hair cell loss or synaptopathy) and are well-suited to study the origin of individual performance differences on a specific task. Here, we simulated how detection thresholds to low- and high-frequency supra-threshold stimuli changed for various profiles of sensorineural hearing loss: (1) normal and high-frequency sloping audiograms (>1 kHz, 30 dB HL at 8 kHz), and (2) normal or gradual loss of auditory-nerve synapses (removal of low, medium, and up to 50% of high spontaneous-rate fibers). The tasks included detection of pure tones or amplitude-modulated tones (fmod=100 Hz) at carrier frequencies of 0.5, 1, and 4 kHz presented with or without background noise. The simulations were compared with experimental data collected from listeners with normal or high-frequency sloping audiograms. The simulations accounted for about 5 dB of the experimental variability, where the simulated variability was derived from the maximum shift between simulated thresholds in each condition.
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Phenotyping and Computational Modeling of Diverse Forms of Genetic Hearing Loss
Ian C. Bruce, Michael R. Wirtzfeld, Anne Griffin, Amanda K. Morgan, Matthew B. Lucas, Jill Lowther, Terry-Lynn Young, Susan G. Stanton
[Abstract]
Computational models of auditory processing that can incorporate pathology may be helpful in understanding the effects of hearing impairment and in the development of improved devices for those with hearing loss, such as hearing aids and cochlear implants. However, incorporating pathology into computational models to explain a human subject’s experimental data is typically problematic due to a lack of definitive knowledge about the pathology that they have. A number of diverse forms of genetic hearing impairment have been molecularly characterized from the genetically isolated human population of Newfoundland, Canada. Affected members of these families form a useful pool of subjects to target for computational modeling, as within each extended family the specific genetic mutation leads to a similar pattern of pathology and resulting pathophysiology and perceptual deficits, but the types of hearing loss are very different between families due to different gene mutations. We are developing computational models incorporating the suspected cochlear pathology for each subject, based on psychophysical tuning curves and DPOAE growth functions, in addition to routine audiometric measures. The model for each subject will be validated via quantitative predictions of advanced electrophysiological recordings (ABR and ECochG) and word perception in quiet and noise for that subject.
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Clustering in an array of nonlinear and active oscillators as a model of spontaneous otoacoustic emissions
Liv Moretto Sørensen, Peter Leer Bysted, Bastian Epp
[Abstract]
Spontaneous otoacoustic emissions (SOAEs) are the phenomenon of sound emitted by the inner ear in the absence of external stimulation. One potential explanation is that coupled, nonlinear oscillators tend to organize into discrete clusters, oscillating with the same period. While the presence of clusters has been verified in various models, the definition did not consider the phase within a cluster but solely the oscillation period.
This study quantifies the phase relations of oscillators within a cluster relative to each other. The model was built of an array of van-der-Pol oscillators with a linear gradient in the oscillation period and scaling to result in an identical phase portrait. The simulations show different sizes of clusters, depending on the type of coupling (reactive/dissipative), the coupling strength and the absence and presence of stochastic variations in the parameters of the array. The oscillation periods of each element in the array were more stable in the presence of an external tone compared to the absence of an external tone. The phase showed a systematic gradient along the length of the array.
Linking these results to corresponding information in SOAEs will provide information about the coupling, and ultimately about the mechanics in the inner ear.
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Simulation of cochlear response by bone conducted tone
Yasuki Murakami
[Abstract]
To investigate cochlear responses to bone-conducted (BC) tones, a two-dimensional nonlinear cochlear model is proposed in this paper. The proposed model comprises macromechanical and micromechanical behaviors that can be described by the Laplace equation and Neely and Kim’s model, respectively. An outer hair cells model is also included as the source of cochlear nonlinearities. Cochlear responses to air-conducted (AC) tones can generally be simulated through stapes vibration. However, in this study, sound pressure was employed to affect the cochlear duct’s outer wall for simulating cochlear responses to BC tones. Both frequency- and time-domain solutions can solve the proposed model. Through simulation, a forward traveling wave in the cochlea can be observed for both AC and BC tones in the frequency-domain solution. The time-domain solution indicates compressive nonlinearity for both AC and BC tones. However, the degree of compressive nonlinearity is the primary differentiating factor for AC and BC tones. For lower and higher input levels, the degree of compressive nonlinearity for BC tones is respectively higher and lower than that for AC tones. The obtained simulation results thus corroborate that compared to AC tones, BC tones generate a cochlear traveling wave response with marginally different compressive nonlinearity.
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Pulse coding in the ensemble of peripheral fibres and auditory discrimination of the tone burst intensity
Liudmila Rimskaya-Korsakova
[Abstract]
The intensity discrimination of tone bursts with middle
intensity presented either alone or before (after) noise
masker was poorer than the discrimination of the bursts with
low or high intensity. We have offered a general explanation
of the discrimination of the bursts presented in the different
conditions. In simulation experiments, the properties of two
complementary distributions of the spike activity of
ensemble of peripheral fibres were studied. They were the
distribution of the spike occurrence in time and the inter-
spike interval distribution. Model input signals were single
pulses, or combined sounds composed the test pulses that
followed or preceded the masking pulses. The distribution
peaks kept different information about the pulses or the
combined sounds. The peaks were assumed to be grounds
for various subjective features. Changes in the amplitudes of
test or masking pulses caused changes in the weights of the
neural bases for various features and their sensitivities to an
amplitude increment. The results of our simulation and
psychoacoustic experiments revealed a qualitative
similarity. The intensity discrimination depended on the
intensity’s ratio of the test and masking pulses (bursts) but did
not depend on positions and delays of the test pulse (burst)
relative to the masking-pulse (burst).
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Towards a model of electric-acoustic stimulation in cochlear implant subjects with residual hearing
Daniel Alrutz, Waldo Nogueira
[Abstract]
When provided with an electric-acoustic system (EAS), cochlear implant (CI) users with residual low-frequency hearing in the same ear show enhanced speech perception compared to subjects with conventional CIs. However, when the cochlea is simultaneously stimulated acoustically and electrically, masking between both modalities occurs. Masking has been observed from auditory nerve fiber (ANF) spike trains in animals (Miller et al., 2009) and from electrocochleographic (ECochG) responses (Koka and Litvak, 2017) and psychophysical experiments in humans (Krüger et al., 2017; Imsiecke et al., 2018). The origin of these effects remains unclear.
Computational models can be used to better understand the underlying mechanisms of electric-acoustic masking. Existing models either simulate ECochG responses phenomenologically (Fontenot et al., 2017; Frijns et al., OMAI 2018) or calculate ANF spiking for sole acoustic (Meddis et al., 2013; Bruce et al., 2018) or electric (Joshi et al., 2017; Ashida & Nogueira, 2018) stimulation. We present a new model, where we coupled the acoustic model of Bruce et al. (2018) with a model of spike conduction along ANFs. Our model shows appropriate spiking responses to acoustic stimulation of remaining hair cells as well as direct electric stimulation of ANFs.
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Organization:
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B. Epp, DTU, Hearing Systems |
S. Verhulst, Ghent University |
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Selective attention in the brainstem and speech-in-noise comprehension
Marina Saiz Alia, Antonio Elia Forte, Tobias Reichenbach
[Abstract]
Understanding speech in noise is a challenging task.
Moreover, the ability to understand speech in background
noise varies considerably from person to person, even for
people that have normal audiograms and hence no
measurable hearing loss. This variability may result from
cochlear synaptopathy, also referred to as hidden hearing
loss, as well as from damage to the neural pathways in the
brainstem and in the central nervous system that are
responsible for sound processing. Here we describe our
recent work on speech-evoked activity in the auditory
brainstem and its relation to speech-in-noise comprehension.
In particular, we show how the neural tracking of the pitch of
continuous speech in the auditory brainstem can be
measured from scalp electrodes. Moreover, we show that this
neural tracking is modulated by selective attention, and that
there is considerable variation in the strength of the
attentional modulation between subjects. We do not find
evidence that this variability is related to hidden hearing
loss. However, we show that the attentional modulation of
the speech-evoked brainstem activity can partly explain a
subject’s ability to understand speech in noise.
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Intelligibility of Filtered Speech and its Relation to Electrophysiological Markers of Supra-threshold Hearing Deficits
Markus Garrett, Viacheslav Vasilkov, Sarah Verhulst
[Abstract]
The origins and perceptual consequences of supra-threshold hearing deficits are not well understood. Sensory and neural deficits are likely contributors, underlining the need for an objective measure which quantifies the neural contributors and complements audiometric measures of sensory loss. The envelope-following response (EFR) is sensitive to neural fiber loss in animal models and has been proposed as a marker. However, its diagnostic sensitivity in humans and relationship to speech intelligibility are still unclear.
We investigate the relationship between speech intelligibility and EFRs in young normal-hearing (yNH), elderly normal-hearing (oNH) and elderly hearing-impaired (oHI) participants as groups reflecting different degrees of neural (yNH/oNH) or sensory deficits (oNH/oHI). EFR stimuli were 4-kHz-centered amplitude-modulated tones/noise with varying modulation characteristics. To target similar neural generators for EFR- and speech-metrics, we looked at speech reception thresholds in broadband, low-pass (<1.5 kHz) and high-pass (>1.65 kHz) filtered speech (OLSA). Individual differences in EFR strength related to speech intelligibility and this relation grew stronger in the high-pass filtered speech condition, suggesting that it is important to consider the same bandwidths in EFR and speech methods. Our results suggest that, under controlled stimulus conditions, the EFR might be a useful marker of neural deficits underlying degraded speech intelligibility.
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The ORN component as an objective measure of concurrent sound segregation
Alexandra Bendixen
[Abstract]
Electroencephalography (EEG) is a popular technique for
deriving objective measures of auditory function due to its
sensitivity to fine details of auditory processing, its high
temporal precision, and its applicability across a wide range
of participant groups with low testing burden. I will discuss
the suitability of a specific EEG measure as a tool for
objectively studying the perceptual segregation of concurrent
sound sources. Specifically, I will discuss the object-related
negativity (ORN) component of the event-related brain
potential (ERP). The ORN component was introduced by Alain
and colleagues almost twenty years ago to reflect the
perceptual segregation of simultaneous sounds. I will show
recent ORN results from healthy young adults, from elderly
participants with slight hearing impairment, as well as from
sleeping neonates, all of which indicate high sensitivity of the
ORN at group level. I will extend these results towards the
question whether ORN is sensitive enough to measure sound
segregation abilities at single-participant level. Based on
detailed correlation analyses between ORN measurements
and perceptual judgments in young and elderly participants,
I will point out promises but also caveats in using the ORN
for individual diagnostics of auditory function.
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Sensory resolution drives auditory responses in lateral frontal cortex
Antje Ihlefeld, Min Zhang
[Abstract]
An emerging literature reveals that brain mechanisms contribute
to symptoms of hearing loss. To study how hearing loss affects
brain function, several groups including our own have
developed light-based methods via functional near-infrared
spectroscopy (fNIRS). Using fNIRS, we recently confirmed that
lateral frontal cortex (LFC) engages when normal-hearing
listeners direct attention to words in background sound versus
listen passively. We now test how sensory resolution affects LFC
recruitment. Using simulated cochlear implant speech, we ask
listeners to perform a word detection task in a situation with
competing background speech, as a function of the amount of
sensory detail in the cochlear implant simulation. One possibility
is that impoverished sensory cues reduce LFC engagement, as
compared to high-fidelity cues, limiting the potential usefulness
of auditory attention. Alternatively, poorly resolved sensory
detail may increase LFC engagement, causing it to saturate
and limiting overall speech intelligibility. Results reveal increased
LFC recruitment during testing with low sensory detail, supporting
the saturation hypothesis.
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Investigating the Cortical Representation of Speech Melody using Near-infrared Spectroscopy
Kurt Steinmetzger, Martin Andermann, Esther Megbel, Zhengzheng Shen, Mark Praetorius, Andre Rupp
[Abstract]
Objective measures that reliably quantify the listening success with cochlear implants (CIs) are hardly available, since CIs are incompatible with several common neuroscientific methods. However, especially for pre-lingually implanted children, such measures are urgently needed to better monitor the development of their hearing. A promising method that does not interfere with the function of CIs is near-infrared spectroscopy (NIRS). We are assuming that cortical activation patterns which resemble those of normal-hearing control subjects indicate that an implantation has been successful. In a first step, we thus presented normal-hearing subjects with speech-like sounds and vowels that differed regarding their prosodic properties, to evaluate the precision and usability of NIRS. The stimuli either had a more or less clear pitch, or varied regarding their pitch contours. Same as additionally obtained magnetoencephalography (MEG) data, which allow for a clear spatial differentiation of the cortical areas activated by the different stimuli, the NIRS data show different spatial activation patterns, despite the lower spatial resolution of this method. The present results thus suggest that NIRS is suitable for investigating cortical responses to stimuli that vary regarding specific acoustic properties and can therefore be used for the neurophysiological assessment of CI-based hearing.
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Cortical Tonotopic Maps in Tinnitus and Hearing Loss
Elouise Koops
[Abstract]
Hearing loss, which is strongly associated with tinnitus,
induces reorganization of auditory cortical areas. Several
studies have suggested a link between hearing-loss-induced
tonotopic reorganization and tinnitus. Neural plasticity drives
tonotopic reorganization, and is a valuable trait as it allows
our brains to recover from injury and sensory deprivation.
However, it appears to bring tinnitus along as an unwanted
companion. Whereas clinical hearing loss is firmly associated
with neural plasticity, the exact role of cortical reorganization
in tinnitus remains debated. This extensive fMRI study
intended to clarify the relation between tinnitus, hearing loss
and tonotopic reorganization. We included participants with
hearing loss, both with and without tinnitus, and a control
group. In a total of 90 participants, loudness matched pure-
tone stimuli (0.25-8 kHz) were used to construct tonotopic
maps from the bilateral cortical responses to sound. The
bilateral auditory cortex activation was increased in both
hearing loss groups, with and without tinnitus. This effect
was strongest for the hearing loss group without tinnitus.
The tonotopic maps and response amplitudes of participants
with tinnitus point to a more conservative form of
reorganization compared to hearing loss without tinnitus.
This contrasts with the previous notion that tinnitus is related
to excessive reorganization.
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Untangling the Components of Distortion Product Otoacoustic Emissions – Speed, Accuracy and Objective Assessment of Auditory Function
Anthony W. Gummer, Ernst Dalhoff, Dennis Zelle
[Abstract]
Otoacoustic emissions (OAEs) have the potential of providing a differential diagnosis of the function of the cochlear amplifier, as well as the function of the middle ear and medial efferents. OAEs are not used as such in clinical routine because: 1) data gathering is slow, 2) accuracy is insufficient, and 3) models provide only an incomplete mapping of data to function. Here, using the distortion product OAEs (DPOAEs), we circumvent the technical problems by 1) shortening pulse width with increasing elicitor frequency, 2) interlacing the frequency pairs in stimulus blocks, and 3) extracting the nonlinear-distortion component of the DPOAE to remove the interfering coherent-reflection component of the DPOAE signal. Experiments were performed with normal-hearing and hearing-impaired (up to 77 dB sensorineural hearing loss) subjects. Here, we show: 1) using the nonlinear-distortion component, the auditory thresholds can be objectively estimated with high accuracy (7-dB standard deviation), 2) the amplitude of the coherent-reflection component decreases with increasing stimulus frequency and intensity, and 3) the total measurement time for eight stimulus frequencies (1-8 kHz) and six intensities (5-dB steps) is typically about 6 min. With ongoing improvements, we envisage reducing measurement time by about a factor of three, a clinically acceptable value.
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Audiovisual speech processing and listening effort in untreated age-related hearing loss: evidence from functional magnetic resonance imaging
Christiane Thiel, Stephanie Rosemann
[Abstract]
Functional magnetic resonance imaging (fMRI) enables objective measurement of central auditory processing. We used fMRI to investigate neural processing of audiovisual speech in hearing impaired unaided and non-hearing impaired elderly subjects. Speech stimuli were used under four different conditions, visual only, auditory only and two audio-visual conditions, in which subjects could see the speakers face, one with congruent auditory and visual information and a second with incongruent information. While performance in this task did not differ between hearing impaired and non-hearing impaired participants, task-related neural activity differed. Hearing impaired participants showed higher neural activity in medial and lateral prefrontal cortex which correlated with the amount of high-frequency hearing loss. We suggest that the additional recruitment of frontal brain regions may relate to increased effort, which was supported by reports of increased daily listening effort in the hearing impaired group. An additional data analysis on resting state fMRI data in the same group of subjects, showed that this increased listening effort relates to decreased functional connectivity between inferior frontal and auditory cortex. Task and resting state fMRI data therefore provide valuable measures of central auditory processing of speech and increases in listening effort in the hearing impaired.
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Acoustic input impedance of infants with normal and pathological middle ear
Tobias Sankowsky-Rothe, Andreas Becker, Karsten Plotz, Rüdiger Schönfeld, Andreas Radeloff, Steven Van De Par, Matthias Blau
[Abstract]
Newborns and infants often suffer from pathological conditions of the middle ear. In many cases, those conditions comprise liquid in the middle ear, e.g. middle ear effusion, or amniotic fluid during the very first days of life. In order to check the middle ear status in a fast and simple manner, the measurement of the infant’s ear acoustic input impedance could be a useful method.
In an ongoing study, the acoustic input impedance of infants’ ears (aged from 2 weeks up to 5 months) was measured in the frequency range from 100 Hz to 10 kHz at ambient pressure. In addition, the middle ear status was assessed by ENT-specialists, yielding an expert classification of ”normal” and ”pathological” middle ears.
The results, on the one hand, show a great variability at low and very high frequencies, reflecting the susceptibility of the measurements to movements and noise of the subjects. On the other hand, there is a frequency range in which a clear discrimination between normal and pathological middle ears seems to be possible. This discrimination appears to profit from the consideration of both the magnitude and phase information of the measured acoustic impedance, indicating that power-based methods may miss important information.
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Organization:
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T. Neher, University of Southern Denmark |
G. Singh, Phonak AG |
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A Clinical Test Battery for Better hEAring Rehabilitation (BEAR). Towards the prediction of individual auditory deficits and hearing-aid benefit
Raul Sanchez-Lopez, Silje Nielsen, Oscar Cañete, Michal Fereczkowski, Mengfan Wu, Tobias Neher, Torsten Dau, Sébastien Santurette
[Abstract]
One aim of the Better hEAring Rehabilitation (BEAR) project is to define a new clinical profiling tool, a test-battery, for individualized hearing loss characterization. Recently, Sanchez-Lopez et al. (ISAAR 2019) proposed a test battery for hearing deficit characterization. The proposed tests were divided into six categories: audibility, middle-ear analysis, speech perception, binaural-processing abilities, loudness perception, and spectro-temporal resolution. The results of 54 listeners were analyzed using a data-driven approach (Sanchez-Lopez et al., 2018), which provided evidence for the existence of two independent sources of distortion and four different auditory profiles. The classification of the listeners into auditory profiles allows the prediction of the performance of the listeners on different psychoacoustic tasks as well as their expected performance while wearing hearing aids. For the classification, a decision tree with only the most predictive tests is desirable for a correct classification of the listeners. The present study aims to explore the optimal decision tree and to propose a reduced, reliable and time-efficient test battery that can classify listeners into the four auditory profiles in a clinical environment. The clinical test battery will be used in a large-scale study that will help implement a protocol for better hearing rehabilitation.
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Assessing the interaction between different auditory profiles and benefit from six hearing aid processing strategies: Insights from the Better hEAring Rehabilitation (BEAR) project
Mengfan Wu, Raul Sanchez-Lopez, Mouhamad El-Haj-Ali, Silje Nielsen, Michal Fereczkowski, Torsten Dau, Sébastien Santurette, Tobias Neher
[Abstract]
The current study forms part of the Better hEAring
Rehabilitation (BEAR) project, which aims at developing and
evaluating new clinical tools for individual hearing loss
characterization and hearing aid benefit assessment. The
purpose of the current study was to assess the interaction
between four different auditory profiles and two outcome
measures of aided performance obtained for six selected
hearing-aid processing strategies (Sanchez-Lopez et al.,
Euronoise 2018). Sixty older habitual hearing-aid users who
participated in the study were previously classified into four
auditory profiles based on a data-driven approach
(Sanchez-
Lopez et al., Trends in Hearing 2018). All stimuli were
generated with the help of a hearing aid simulator and
presented via headphones. Speech recognition in noise was
assessed at fixed signal-to-noise ratios based on individual
50%-correct speech reception thresholds measured in a
realistic noise environment. Subjective ratings of overall
quality and noise annoyance were measured using a
multiple
stimulus comparison paradigm. It is hypothesized that the
four
auditory profiles will have different needs in terms of
compensation so perceptual outcomes for the six hearing
aid
processing strategies are expected to be different.
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Evaluation of Objective Prediction Approaches for Aided Listening in Complex Acoustic Scenes
Florian Kramer, Marc René Schädler, Anna Warzybok
[Abstract]
Understanding the individual listening performance and the development of objective methods suitable to predict the performance in aided conditions is crucial for improving the rehabilitation strategies with hearing devices. Here, subjective and predicted speech recognition data for realistic acoustic conditions are compared. Speech reception thresholds (SRTs) were measured in three noise conditions (20-talker babble, cafeteria noise and a single competing talker) with 12 hearing impaired listeners. The NAL-NL1 prescription rule with a multiband compressor scheme was applied to compensate for the hearing loss. The measured SRTs were compared to predictions with the short-time version of the binaural speech intelligibility model (BSIM) and the framework for auditory discrimination experiments (FADE).
The results show a significant correlation between predicted and measured SRTs for BSIM overall (p<0.01, R^2=0.62). Especially in the single competing talker condition, BSIM, which predicts SRTs based on weighted signal-to-noise ratios, provides more accurate predictions than FADE which has to recognize the target speech signal.
BSIM also predicts increased SRTs with increasing pure-tone averages (p<0.01, R^2>0.7), indicating less benefit from hearing loss compensation with increasing hearing loss. However, the measured SRTs do not support this trend. The influence of hearing level on SRTs thus appears to be overestimated by BSIM.
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Effects of Personalising Hearing-Aid Parameter Settings Using a Real-Time Machine-Learning Approach
Niels Søgaard Jensen, Laura Winther Balling, Jens Brehm Bagger Nielsen
[Abstract]
In most hearing-aid fittings, gain is prescribed by a fitting
rationale that uses the audiogram as main input. While this
approach may work well in some situations, it falls short in
situations where the user’s listening intention deviates from that
assumed by the rationale. This shortcoming was the motivation
for a new commercially available method to self-adjust hearing-
aid parameters while in a specific situation. The method is based
on use of machine-learning algorithms that utilize user input to
determine a sequence of parameter settings to be assessed in
paired comparisons and to estimate the setting that optimizes
user satisfaction. We present results from a study where 20
participants with hearing loss used the method to adjust
hearing-aid gain in 12 different sound scenarios. In a double-
blind comparison, the participants rated recordings of the
adjusted settings and two prescribed settings with respect to
three different perceptual sound attributes. The results showed a
significant benefit of the method on basic audio quality. A large
spread in the gain adjustments was observed, suggesting a
need for individual adjustments of hearing aids. Data from real-
life use of the same method will be included in a discussion of
clinical and rehabilitative implications.
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User behavior with EVOTION hearing aids
Niels H. Pontoppidan, Lars Bramsløw
[Abstract]
How are hearing aids really used in everyday situations? How can insights into this enhance the rehabilitation process involving the user and the hearing care professional? Data collected with EVOTION hearing aids contribute towards the answers. For 15 years hearing aids have logged data about the sound environment and usage patterns. However, the past logging data provided descriptive statistics about the usage, and not detailed insights with timestamps stating which situations, at what time, and which settings of the hearing aid. The EVOTION hearing aids connects over Bluetooth to a smartphone that stores a snapshot of the sound environment one time every minute to provide enough insight in the sound environments whilst preventing any eavesdropping based on the data. The sound environment data consists of sound pressure level, noise floor, and signal-to-noise ratio and a classification into quiet, noise, speech or speech in noise. Data spanning hearing aid users’ behavior over weeks and months provides an in-depth view in how they cope with different sound environments and throughout the day. Insights from this data enables the hearing care professional to guide the individual user towards optimal outcomes and follow their individual journeys.
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Hearing aid benefit in everyday life
Inga Holube, Petra Von Gablenz, Ulrik Kowalk, Markus Meis, Jörg Bitzer
[Abstract]
Hearing aid benefit is routinely measured with speech tests in quiet or stationary background noise and with questionnaires comprising predefined listening situations. Nevertheless, everyday life of hearing aid users is unique and might not be captured by the traditional tools. An alternative approach employs repeated assessments of individual everyday situations, a method called ecological momentary assessment (EMA). The EMA method was implemented with a smartphone-based system allowing for frequent subjective assessments and objective monitoring of the acoustical environment. In a field study, 20 participants with mild-to-moderate hearing impairment were fitted with hearing aids. The EMA method was applied for several days both before and after hearing aid fitting including an acclimatization phase. The participants categorized their environments and activities and rated loudness, ability and importance of speech understanding, as well as listening effort. By comparing the assessments without and with hearing aids or with old and new hearing aids, the benefit can be estimated. One structural drawback of this approach is the challenge of varying situations between participants but also within participants when comparing different time periods. Having said that, this effect implies the advantage of capturing the hearing-related quality of life on an individual basis.
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Organization:
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P. Majdak, Acoustics Research Institute, Austrian Academy of Sciences |
J. Braasch, Rensselaer Polytechnic Institute |
M.T. Pastore, Arizona State University, USA |
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A likelihood based decoding mechanism for two-channel localization models
Jörg Encke, Werner Hemmert, Mathias Dietz
[Abstract]
Traditionally, the neuronal representation of interaural time
difference (ITD) is often modeled using a delay-line mechanism. In
contrast to delay-line models which inherently decode ITD, the
model proposed here represents interaural phase differences in form of
the relative activity between neuron populations located in the two
brain hemispheres. Therefore, the localization information is only
encoded in specific activities in the two hemispheres. It cannot be
directly determined but has to be decoded first. This study proposes a decoder based on likelihood
calculations. Instead of decoding a single phase difference for a
given hemispheric activity-balance, the likelihood that the observed
activity was due to any possible phase difference is calculated. It is
shown that applying the decoder separately to different frequency
channels, results in a visualization very similar to that of a
cross-correlogram used in delay-line models. It is also shown that the
decoder is able to account for psychoacoustic data, including the
bandwidth-dependent lateralization of 1.5 ms delayed noise that
previously imposed a problem on two-channel models.
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Speech-in-noise performances in virtual cocktail party using different non-individual Head Related Transfer Functions
Lorenzo Picinali, Maria Cuevas-Rodriguez, Daniel Gonzalez-Toledo, Arcadio Reyes-Lecuona
[Abstract]
It is widely accepted that, within the binaural spatialisation domain, the choice of the Head Related Transfer Functions (HRTFs) can have an impact on localisation accuracy and, more in general, realism and sound sources externalisation.
The impact of the HRTF choice on speech-in-noise performances in cocktail party scenarios has though not yet been investigated in depth.
Within a binaurally-rendered virtual environment, Speech Reception Thresholds (SRTs) with frontal target speaker and lateral noise maskers were measured for 21 subjects several times across different days, and using different HRTFs. Results show that for the majority of the tested subject, significant differences could be found between the SRTs measured using different HRTFs. Furthermore, the HRTFs leading to better or worse SRT performances were not the same across the subjects, suggesting that the choice of the HRTF can indeed have an impact on speech-in-noise performances within the tested conditions.
These results suggest that when testing speech-in-noise performances within binaurally-rendered virtual environments, the choice of the HRTFs should be carefully considered. Furthermore, a recommendation should be made for future modeling of the speech-in-noise perception mechanisms to include HRTFs.
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Information-based Source Localization with Distinct Binaural Cues
Patrick Danes
[Abstract]
Audio-motor binaural localization algorithms, which combine directional cues extracted from the sensed signals with the motion of the sensor, are known to overcome shortcomings such as font-back ambiguity and source range non-observability. They can be improved by closing the loop from their output to the control inputs of the sensor, i.e., the sensor motor commands. This paper presents an approach, coined ”information-based feedback control”, which drives in real time a binaural head so as to gather information on the location of a static source. On the one hand, a ”greedy” approach moves the head to its next best position. On the other hand, a multi-step-ahead scheme determines its most effective path over a receding horizon of size N>1, by reasoning on average over yet uncollected audio data. Both methods internally entail the prediction of binaural cues, e.g., ITDs, ILDs or a combination of both. Some results can be given an elegant commonsense interpretation. Implementations on a real-life robotic test bed are presented.
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Human Perception of Dichotic High-Frequency Complex Sounds simulated with a two-channel Count Comparison Model
Jonas Klug, Lisa Schmors, Go Ashida, Mathias Dietz
[Abstract]
The human auditory system uses binaural cues to localize sound sources. Sounds originating from off-midline directions are received with an interaural time difference (ITD) and an interaural level difference (ILD).
Previous models simulating psychoacoustic results extracted ITDs using a cross-correlation approach that resembles so-called delay lines. However, mammals, including humans, are likely to extract ITD information without delay lines. Published physiological data have shown that lateral superior olive (LSO) cells are sensitive to envelope ITDs of sinusoidally amplitude-modulated (SAM) tones. Here, we propose a physiologically plausible model, by combining the well-established auditory periphery model by Zilany et al. (J. Acoust. Soc. Am. 135. 283-286, 2014) with a coincidence counting model of the LSO (Ashida et al. PLOS Comp. Biol. 12, e1004997, 2016). The latter integrates ipsilateral excitatory signals together with negatively weighted inhibitory signals from the contralateral ear using millisecond short coincidence windows.
This version of the model uses very few free parameters to quantitatively account for a wide range of psychoacoustic data. It is demonstrated that the same binaural model neurons account for both ILD-based and ITD-based lateralization as well as their combinations.
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Predicting Externalization of Anechoic Sounds
Robert Baumgartner, Piotr Majdak
[Abstract]
The perceptual association of a sound with a source surrounding the listener is natural
but requires complex signal processing to be achieved in virtual or augmented realities
presented via headphones. Little is known about the set of spatial features required to
elicit an externalized auditory percept, even under anechoic conditions. We investigated
the diversity and relevance of different features by conducting a model-based meta-
analysis of psychoacoustic studies. As potential features we considered monaural and
interaural spectral shapes, spectral and temporal fluctuations of interaural intensity
differences, interaural coherence, and broadband inconsistencies (trading) between
interaural time and intensity differences in a framework of template-based auditory
models. Our model predictions indicate that the monaural spectral shapes and the
strength of time-intensity trading present potent cues to explain most previous results. In
order to better dissociate and validate the relative contributions of the features, we let
listeners judge the effect of monaural spectral gradient distortions and unilateral delays
on the externalization of sounds from various lateral angles. Comparisons between
obtained behavioral results and feature-specific predictions will be discussed.
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Effects of Target Speech Distance on Auditory Spatial Attention in Noisy Environment
Shuichi Sakamoto, Florent Monasterolo, Cesar Salvador, Zhenglie Cui, Yôiti Suzuki
[Abstract]
When people perceive auditory information in a noisy environment, auditory spatial attention plays an important role in extracting the information from the target sound source. For auditory spatial attention, the effects of distance have been unclear while many researchers have pointed out the effects of directional characteristics. In this study, we investigated the relationship between auditory spatial attention and the distance of the target speech sound. In the experiment, background multi-talker noise was presented 1 m from the listeners. Target speech sound was presented at one of four distances: 0.13, 0.25, 0.5, or 1 m. By changing the target’s appearance probability for each distance, the listener’s attention was implicitly focused at a specific distance. Listeners were asked to respond as soon as the target speech sound was perceived. Experimental results clearly revealed that the listeners show faster responses to the target speech sound at the focused distance than to that at other distances. Moreover, when the listeners’ attention was focused at a very near distance, the response became fast even when the target was located at other distances. These results suggest that sounds located at very near distances could enhance the spatial selectivity of auditory selective attention.
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Integrating interaural differences of time and level across frequencies and with each other in a precedence effect model
M. Torben Pastore, Jonas Braasch
[Abstract]
In a series of reports, Pastore and Braasch (2015, and under
review) presented behavioral data for long-duration pairs of
noise stimuli presented over headphones with equal but
opposite interaural time differences (ITDs). One stimulus led
the other by 1-5 ms. Results showed that listeners localized
based primarily on the leading noise, even when the lagging
noise was as much as 6-8 dB greater in amplitude, thereby
demonstrating the precedence effect. Modeling in a recent
report (under review) demonstrated how various forms of
onset dominance in the extraction of interaural time
differences could help to account for these data. However,
interaural level differences (ILDs) created by the physical
interference of the leading and lagging stimuli appear to
have influenced the results for different listeners to varying
degrees. The way in which these ITDs and ILDs may be
integrated across frequencies and with each other will be
explored in new modeling efforts.
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On the evaluation of head-related transfer functions with probabilistic auditory models of human sound localization
Michele Geronazzo, Roberto Barumerli, Federico Avanzini
[Abstract]
Understanding spatial hearing leads to implement efficient and effective auralization rendering algorithms with headphones. Two important aspects contribute to sound localization: (i) acoustic filtering of listener body, and (ii) non-acoustic factors introduced by auditory periphery. Accordingly, head-related transfer functions (HRTFs) describe users acoustics in terms of their spatial filtering. Binaural synthesis through generic HRTFs (commonly a dummy head) is the most simple solution for an auralization framework. In this scenario, a high variability in localization tasks between subjects yields to an unreliable rendering. Listener’s acoustic and perceptual characterization require HRTF modeling and auditory models predictions in order to provide an effective auralization on individual basis.
Systemic comparisons of HRTF approximations and different user profiles can help to predict listener’s performances. We consider several case studies on both vertical and horizontal localization with different HRTFs and probabilistic auditory models. In our analysis, spatial audio rendering with generic HRTFs has a special attention for its commercial relevance compared to unpractical and questionable use of individual HRTFs.
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Better Ear Listening with Recurrent Neural Networks for Speaker Separation
Nils L. Westhausen, Bernd T. Meyer
[Abstract]
The cocktail party problem is still unsolved in real-life conditions, yet recent advances in single channel blind speaker separation based on deep machine learning show promising results that could substantially alleviate this problem. Two prominent approaches in this regard are utterance-level permutation invariant training (uPit) and time-domain audio separation network (TasNet), which however do not consider spatial information relevant in reverberant conditions. This information is essential for using separation algorithms in real-life situations, especially with an application in future hearing devices in mind. The goal of this work is to evaluate single-channel separation procedures in binaural spacial scenes considering four different scenes. Artificial scenarios are created by convolving speech signals produced by different speakers with different measured binaural room impulse responses at different positions. The results are multiple mixtures of two spatial speech signals that should be separated. The separation performance is quantified in terms of the source-to-distortion ratio (SDR) and PESQ. We analyze the degradation that can be expected in the presence of reverberation and also assess the benefit from better ear listening when processing each channel separately, which results in a mean SDR improvement of 2 dB.
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A Binaural Model Predicting The Effect of Hearing Impairment and Noise Level on Speech Intelligibility
Thibault Vicente, Mathieu Lavandier, Jörg M. Buchholz
[Abstract]
A model will be presented that allows predicting binaural
speech intelligibility in noise for normal-hearing and
hearing-impaired listeners. The model inputs are the masker
and target signals at the listener’s ears, as well as their
audiogram that is used to implement an internal noise (IN). The
model computes per time frame and frequency band a
binaural masking level difference (BMLD) when all signal levels
are higher than the IN level, and the signal-to-noise ratio at the
better ear (BE-SNR) using the maximum between the masker
and IN levels. The BMLD and BE-SNR are integrated across time
and frequency and summed to obtain a binaural ratio. The
relative differences of the binaural ratio are compared to
speech reception threshold differences measured in listening
tests.
The model was validated on three experiments involving
normal-hearing and hearing-impaired listeners. Stimuli were
anechoic and played using headphones. The target, in front of
the listener, was presented simultaneously with two speech-
shaped noises or noise-vocoded speech either collocated with
the target or separated at +/- 90∘. The noises were presented at
different sensation levels and the relative target level was
adapted to derive speech reception thresholds. The
experimental data were accurately predicted by the model.
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Binaural Processing in a New Cochlear-Implant Paradigm Inserting Extra Pulses with Short Inter-Pulse Intervals
Martin Lindenbeck, Piotr Majdak, Bernhard Laback
[Abstract]
Cochlear-implant (CI) listeners face degraded selective hearing
because they lack sensitivity to both interaural-time-difference
(ITD) and pitch cues. One major reason is the replacement of
the acoustic temporal fine structure by periodic high-rate pulse
trains to avoid electric-field interactions. A second major reason
is a perceptual limitation of CI listeners in the maximum pulse
rate up to which ITD and pitch cues are salient. Recently, a new
approach relying on inserting extra pulses with short inter-pulse
intervals (SIPIs) into the high-rate carrier pulse trains has been
shown to enhance ITD and pitch sensitivity while likely
maintaining speech understanding. In this contribution, a
concept for a new binaural CI stimulation paradigm which
implements the SIPI approach is presented. In this paradigm, by
applying binaural models of normal hearing, the ITD is extracted
from a binaural audio signal. That ITD, combined with the
envelope-modulation information, defines the positioning of the
SIPI pulses. This concept allows to enhance ITD cue precision
through binaural processing by integrating reliable ITD
information across frequencies or electrodes, respectively. In the
contribution, various models and parameters will be discussed.
Further, strategies how to avoid conflicts in the across-electrode
timing of the SIPI pulses will be outlined.
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Evaluation of an ILD-based hearing device algorithm using Virtual Sound Environments
Ruksana Giurda, Laurent S. R. Simon, Hannes Wüthrich, Norbert Dillier
[Abstract]
Hearing devices are mostly tested using controlled
laboratory environments. These feature simplified auditory
representations of the real conditions under which the
hearing devices will perform once released, such as one
speech signal played from one loudspeaker plus a noise
signal playing from another loudspeaker. It can however
be insufficiently realistic for some of the processing
included in the devices. In order to circumvent this
limitation, Virtual Sound Environments (VSE) can be used to
simulate complex sound fields using headphones or arrays
of loudspeakers, and thus to offer more realistic conditions
than most common hearing device evaluation protocols.
In this paper, a hearing device algorithm based on
Interaural Level Differences (ILD) was evaluated using
Binaural Room Impulse Responses (BRIR, used here as a
reference), and artificial room effect reproduced via
either Higher Order Ambisonics (HOA) or Vector-Based
Amplitude Panning (VBAP). Results show that the algorithm
does not work when using 5th order HOA, and that the
performance of the algorithm is lower when using VBAP
than when using BRIR.
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Multimodal simulation of textiles in virtual and augmented environments
M. Ercan Altinsoy, Alican Alma, Annerose Braune, Sybille Krzywinski, Frank Püschel
[Abstract]
The inspection of remote textiles is of interest for industry, e.g.
the quality control along the supply chain at plants which are
stationed around the world, and for consumer goods sector,
e.g. in E-Commerce as well as for the people with impaired
eyesight and limited mobility. The exploration of textiles using
bare fingers is a multisensory event. In such kind of events, we
obtain information simultaneously from the auditory, tactile, and
visual sensory channels. The aim of this study is to realise the
visual, auditory and haptic representation of textiles in virtual or
augmented environments with multimodal interfaces. The
missing haptic feedback is one of big problems in such kind of
systems. In this study, we developed a haptic interface, which is
able to simulate the roughness of the fabrics and investigated
auditory-haptic interaction to design multimodal stimuli for the
reproduction and to determine the relative contributions of the
auditory and tactile systems on multimodal perception. In this
talk, the importance of the touch induced scraping sounds will
be discussed.
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The 3D Tune-In Toolkit: A C++ library for binaural spatialisation, and hearing loss/hearing aids emulation
Arcadio Reyes-Lecuona, Maria Cuevas-Rodriguez, Daniel Gonzalez-Toledo, Carlos Garre, Ernesto De-La-Rubia-Cuestas, Luis Molina-Tanco, Angel Rodriguez-Rivero, Lorenzo Picinali
[Abstract]
This contribution aims at demonstrating the 3D Tune-In (3DTI)
Toolkit, a C++ standard library for binaural spatialisation, which
includes also hearing loss and hearing aids emulators. Binaural spatialisation is performed through convolution with
user-imported Head Related Transfer Functions (HRTFs) and
Binaural Room Impulse Responses (BRIRs), including additional
functionalities such as near- and far-field sources simulation,
Interaural Time Differences (ITDs) customization, and Ambisonic-
based binaural reverberation. Hearing loss is simulated through
gammatone filters and multiband expanders/compressors,
including advanced non-linear features such as frequency
smearing (using the Moore-Baer model) and temporal distortion.
A generalised hearing aid simulator is also included, with
functionalities such as dynamic equalization, calibration from
user-inputted audiogram, and directional processing. The 3DTI Toolkit is available as an open-source C++ library
through a public GitHub repository. Several other releases of the
3DTI Toolkit are also available, including a Test Application, with
a GUI, OSC remote control capabilities and installers for MacOS,
Windows and Linux; a VST plugin for MacOS and Windows; and
a Javascript wrapper for web-based binaural spatialisation.
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Organization:
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J. Fels, Teaching and Research Area of Medical Acoustics |
M. Marschall, DTU Electrical Engineering |
J.M. Buchholz, Macquarie University |
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LEAP, a new laboratory test for evaluating auditory preference
Karolina Smeds, Martin Dahlquist, Josefina Larsson, Petra Herrlin, Florian Wolters
[Abstract]
When hearing-device signal processing is evaluated in the laboratory, the testing usually incorporates speech testing or sound-quality evaluations with pre-recorded material. Neither real speech communication, nor more passive listening situations are usually included.
A new test, the Live Evaluation of Auditory Preference (LEAP), was developed with the aim to broaden the scope of laboratory testing by focusing on a range of listening intentions. Real conversations between the test participant and one or two test leaders are central to the method, but ecologically valid test scenarios with focused listening (such as watching TV) and scenarios with passive listening are also included.
The test was evaluated with experienced hearing-aid users, who compared two hearing-aid settings using paired comparisons of preference. To validate the method, the results from the LEAP test were compared to results obtained in the field, where a smartphone was used for prompting the assessors to make paired comparisons of preference using Ecological Momentary Assessments (EMA).
When the results from the field and the laboratory were compared, the correspondence was satisfactory. To make the results of a laboratory test indicative of real-life performance, a broadened test scope seems important, and the LEAP could be a step forward in that direction.
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Realistic Audiovisual Listening Environments in the Lab: Analysis of Movement Behavior and Consequences for Hearing Aids
Maartje Hendrikse, Gerard Llorach, Giso Grimm, Volker Hohmann
[Abstract]
With increased complexity of hearing device algorithms a strong interaction between motion behavior of the user and hearing device benefit is likely to be found. To be able to assess this interaction experimentally more realistic evaluation methods are required that mark a transition from conventional lab experiments to the field.
In this presentation we describe audiovisual listening environments that were designed to measure realistic movement behavior in relevant everyday-life situations in the lab. Results of a study that measured movement behavior of young and elderly normal-hearing listeners in these environments are presented. A retest and questionnaires showed that the data and calculated measures were reproducible and reliable, and that the environments were of a sufficient level of realism. Furthermore, results of the movement analysis show that movement behavior depends on the environment and age of the subject, and is predictable in multi-talker conversations and for moving distractors. Based on these results, predictions about the performance of hearing aids in everyday-life situations were made. A second study tested these predictions by estimating the hearing aid benefit related to noise suppression in the presented environments using acoustic simulations with the measured head movement behavior. Results of this second study are also presented.
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Speech Intelligibility in a Realistic Virtual Sound Environment
Naim Mansour, Marton Marschall, Adam Westermann, Tobias May, Torsten Dau
[Abstract]
Employing realistic, yet controlled sound scenarios for the
evaluation of hearing aid (HA) algorithms in a virtual sound
environment (VSE) has the potential to improve HA users’ real-
world listening experience and performance. In the present
study, so-called critical sound scenarios (CSS) were defined as
acoustic scenes that HA users experience as important, difficult
and common, and selected through ecological momentary
assessment (EMA), inspired by results from recent studies. The
scenarios were acquired in real scenes using a spherical
microphone array, and reproduced in an acoustically valid way
inside the VSE. A speech intelligibility task using the Danish
Hearing In Noise Test (HINT) was implemented in one of the
scenes to measure speech reception thresholds (SRT) for normal-
hearing and hearing-impaired subjects. SRTs obtained for the
realistic VSE background noise were found to be higher than
corresponding ones reported in headphone-based studies,
potentially as a result of the increased listening effort required to
separate target speech from the more realistic interfering
speech background.
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Evaluating potential EEG-indicators for auditory attention to speech in realistic environmental noise
Ehsan Eqlimi, Dick Botteldooren, Annelies Bockstael
[Abstract]
The human brain has the remarkable ability to attend to
the target speech among irrelevant information
originating from ecologically valid context, but the neural
mechanism supporting this is still being investigated. Here,
using 64-channel EEG recording from the scalp of 23
subjects engaged in listening tasks, we explored the
potential of different EEG indicators to distinguish
between cortical representation of attentive listening to
13 5-minutes meaningful speech fragments in ecological
background noise (babble, highway and fluctuating
traffic) and inattentive listening to 12 3- minutes of the
same noise.
Firstly, low-frequency power at electrodes selected via an
envelope following response test corresponding to the
speech envelope were investigated.
As a further indicator, the coupling between speech
envelope and EEG signals was examined based on the
cortical entrainment hypothesis.
Furthermore, the cortical lateralization of high-frequency
EEG to examine the effect of linguistic processing in
attentive listening to the speech was analyzed.
Finally, according to the inhibition framework, alpha band
oscillations were studied, which are expected to be
modulated by task-irrelevant signals. Based on multiple
signal classification algorithm and a mixed-model
regression, significant effects of attention demand of the
task, background noise type on the amplitude and
frequency of peak alpha were found.
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Influence of varying room acoustic conditions in a speech-in-noise test with radial moving masker, for young and elderly listeners
Rhoddy Viveros, Lukas Aspöck, Janina Fels
[Abstract]
In real-life listening situations, we often listen to speech in the presence of masking noises inside rooms with different room acoustics conditions. Likewise, most everyday listening situations consist not only of multiple sound sources both stationary and moving, but also of numerous acoustic reflections, reaching our ears at different time and intensity, caused by the physical environment. In a speech perception in noise task, acoustic reflections can be beneficial, increasing the speech signal energy reaching the listener, or detrimental, superimposing on the direct sound altering the waveform. This study aims to show the effect of different room acoustic conditions in a speech-perception task with a radial moving masker. A speech-in-noise test was performed with young and elderly listeners, using binaural room simulation to create a headphone-based virtual acoustic environment. Listeners are asked to identify target digit-triplets, under the presence of a stationary or moving masker. The movement of the masker is a radial movement from a distance of 0.5 m to 1.8 m at a constant azimuth of 70∘. All masker configurations were assessed over three room acoustic conditions: anechoic, acoustically treated room and acoustically untreated room. Results show differences in speech perception between stationary and moving maskers.
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Assessment of acoustical properties and subjective perception in everyday life
Inga Holube, Petra Von Gablenz, Ulrik Kowalk, Jörg Bitzer
[Abstract]
Hearing ability in everyday life can be determined with the method of ecological momentary assessment (EMA). For this purpose, a smartphone-based system was developed including two microphones attached to eyeglasses’ temple arms on each side of the head and a bluetooth transmission. The audio signals recorded by the two microphones are analyzed in segments of 1 min and deleted immediately thereafter. By storing only averaged characteristics of the environment’s acoustical properties, the privacy of the user and bystanders is fully preserved. Further offline analysis allows for own-voice detection and background estimation. The smartphone is additionally used to prompt the user approximately every half hour for subjective assessments of loudness, ability and importance of speech understanding, and listening effort with respect to the situation. 47 elderly participants used the system for several consecutive days in their individual everyday life. They spent most of their time at home. Nevertheless, mobility situations (e.g., in a car) and societal situations (e.g., in a restaurant) are important and require a higher listening effort. This contribution demonstrates the scope of this approach and gives results of acoustical properties and subjective perceptions in everyday life.
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Behavior and Speech Intelligibility in a Changing Multi-talker Environment
Lubos Hladek, Bernhard U. Seeber
[Abstract]
At the auditory cocktail party, we often listen to speech coming from different directions. This project aims to investigate how behavior and speech perception interact when people have different goals when listening in an acoustically complex scene. In the experiment, participants stand in a simulated reverberant room and listen to target sentences of one talker from random directions in the horizontal plane. The targets are presented in an ongoing fashion together with multiple interferers, which are clustered or spatially distributed speakers. Different conditions aim to test behavior following different goals: a) the participants can turn freely to achieve better understanding, b) they are instructed to localize the target sentences, c) in separate sessions speech perception and localization is tested using the same material, but in interrupted fashion with responses provided after each presentation of the target sentence. Stimuli are presented in free-field using the real-time Simulated Open Field Environment. The movement behavior is recorded using a video-based motion-tracking system, and the motion data are analyzed in relation to the spatial distribution of the interferers, and performance in the trial-by-trial task. The results are discussed in the context of developing realistic listening scenes for psychoacoustical and audiological research.
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Effect of test realism on speech-in-noise outcomes with bilateral cochlear implant users
Jörg M. Buchholz, Javier Badajoz Davila, Richard Van Hoesel
[Abstract]
Current speech intelligibility tests conducted in the
laboratory follow oversimplified methods whereby neither
the target speech nor the interfering noises resemble what a
person encounters in their daily lives. This study explores the
effect of the realism of the target speech and noise on word
recall performance of bilateral cochlear implant users. Three
different test paradigms are compared: (1) anechoic BKB-
like sentences presented in babble noise, (2) reverberant
BKB-like sentences presented in three-dimensional recordings
of actual noisy situations, and (3) a variation of the second
test where the sentences were taken from natural effortful
conversations. All the tests were conducted by means of a
spherical loudspeaker array located inside an anechoic
chamber. The comparison was conducted at two different
SNRs: 1.4 and -2.2 dB. Speech intelligibility results from eleven
adult CI users were consistently highest in the first test and
lowest in the third test. These results indicate that participants
could more easily deal with babble noise than with more
realistic noisy situations, and that they could understand
more easily BKB-like sentences than more realistic speech.
The findings of this study highlight the importance of
improving the realism of speech-in-noise tests to provide
outcomes with increased ecological validity.
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Spatial release from masking assessment in virtual reality for bilateral cochlear implants users
Lorenzo Picinali, Marina Salorio-Corbetto, Deborah Vickers
[Abstract]
One of the potential benefits of having bilateral cochlear implants, in addition to enhanced sound localisation abilities, is the improvement of speech-in-noise perception, more specifically the advantages gained by being able to spatially separate the speech and noise sources. This process is known as spatial release from masking (SRM).
In order to improve these skills through hearing training, it is essential to assess binaural hearing performance in a time-efficient manner, being able though to maintain some of the real-world complexities, e.g. multiple sources and reverberation.
Traditional tests of SRM and/or localization are often time consuming, and typically assess these in unrealistic settings, with only one controlled stimulus to attend to. With this in mind, Bizley and colleagues (2015) developed the Spatial Speech in Noise test (SSiN), which allows for the simultaneous evaluation of speech discrimination for various locations, SRM, and relative localisation measures using speech in a background babble.
We have recently developed a Virtual Reality-based binaural training suite named BEARS (https://www.youtube.com/watch?v=E-Rn7ipiJAg). In order to assess its impact in terms of SRM and localisation, but without needing to rely on custom spaces and expensive loudspeakers arrays, we have implemented and evaluated a 3D headphones-based implementation of the SSiN test.
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Organization:
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M. Dietz, Universität Oldenburg |
O. Macherey, Laboratory of Mechanics and Acoustics |
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Improving ITD coding with bilateral cochlear implants through temporal enhancement
Bernhard U. Seeber, Monika-Maria Oster, Aswin Wijetillake
[Abstract]
Interaural time differences (ITDs) help normal hearing listeners perceive, locate and understand sounds in noisy and reverberant conditions, but their benefit is limited with bilateral cochlear implants (BiCI). Monaghan & Seeber (2016) presented a temporal enhancement algorithm that sharpens and deepens modulations at onsets of selected peaks in the signal envelope and showed improved ITD sensitivity with tone-vocoded speech. Based on the short-term direct-to-reverberant ratio (DRR) the algorithm selects peaks that are dominated by the direct sound rather than reflections and applies enhancements at times derived from the timings of envelope peaks. However, extensive analysis revealed that envelope peak timings do not robustly convey the ITD of a source in reverberant rooms. Presented here is an updated version of the algorithm that estimates the short-term ITD from the signal’s temporal fine structure and adjusts the enhancement timings to encode that ITD. Furthermore, the new algorithm enhances envelope peaks based on the short-term interaural coherence. Evaluation with a vocoder revealed that the new algorithm improved ITD sensitivity relative to the old algorithm and unenhanced case, for all DRRs examined, without degrading speech understanding. This suggests that BiCI users in our current study will also demonstrate benefits.
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Electrophysiological and Psychophysical Measures of Amplitude Modulation Discrimination Interference in Cochlear Implant Users
Deborah Vickers, Brian C. J. Moore, Patrick Boyle, Josef Schlittenlacher, Lindsey Van Yper, Jaime Undurraga
[Abstract]
Cochlear implants (CIs) work by dividing the incoming acoustic signal into a limited number of frequency channels, extracting the slowly varying amplitude envelope in each channel, and using this to modulate the level of electrical pulses delivered to the auditory nerve fibres. The amplitude modulation (AM) cues that are transmitted are crucial for speech understanding. The use of AM cues can be affected by interference from adjacent channels caused by current spread, which can be further exacerbated if traumatic electrode placement or poor neural survival occurs. We have tested twelve adult CI users with a measure of across-channel modulation interference (AMCI). The discrimination of different rates of AM in the presence of interfering AM on adjacent channels is measured behaviourally and via the electrically-evoked auditory change complex (eACC), a cortical potential that occurs in response to a change in an ongoing stimulus, AM-rate change in this case. The ability to discriminate AM rate (based on behavioural measures and eACC) will be compared to speech perception scores. If the eACC to AM rate changes proves to be reliable, it can be used with children and also to identify poorly discriminated CI electrodes, to guide re-mapping.
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Spectral Blurring in Cochlear Implants: Association with Channel Interaction and Effects on Speech-in-Noise Perception
Tobias Goehring, Julie G. Arenberg, Robert P. Carlyon
[Abstract]
Cochlear implant (CI) listeners struggle to understand
speech in background noise. Interactions between
electrode channels due to current spread increase the
masking of speech by noise and reduce the effective
number of channels a CI provides. Therefore, strategies to
reduce channel interaction have the potential to improve
speech-in-noise perception by CI listeners. We investigated
the effects of channel interaction on speech-in-noise
perception and its association with spectro-temporal acuity
in a listening study with 12 CI users. By adjusting the spectral
overlap in terms of acoustic bandwidths between electrode
channels (spectral blurring), we introduced channel
interaction effects and measured speech reception
thresholds in noise as a function of the amount of blurring
applied to either all, or 5 out of 15, electrode channels.
Performance for each listener remained roughly constant as
the amount of blurring applied to all channels increased up
to some knee point, above which it deteriorated. This knee
point correlated with performance on a non-speech
spectro-temporal task. Surprisingly, even extreme amounts of
blurring applied to 5 channels did not affect performance
overall. Findings show the resilience of CI listeners against
spectral blurring and illustrate the difficulties faced by
optimisation strategies.
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Coding of Electrical Stimulation Patterns for Binaural Signal Processing in Cochlear Implants
Waldo Nogueira, Tom Gajecki, Reemt Hinrichs, Jörn Ostermann
[Abstract]
A cochlear implant (CI) is an electronic device implanted
in the inner-ear that can restore the
sense of hearing of a profoundly deaf person by electrical
stimulation of the auditory nerve. CI users obtain
nowadays quite good speech understandings in quiet
environments, however they still struggle in more realistic
environments with background noise and reverberation.
CI users need significantly higher Signal-to-Noise Ratios
(SNRs) to achieve the same speech intelligibility as normal-
hearing listeners. For this reason, binaural speech
enhancement techniques have emerged to improve the
SNR in noisy acoustic conditions by making use of the
information captured by the microphones on both the left
and the right ear in a combined signal processing.
However, the transmission of the audio signals from one
ear to the other needs to be done with low latency, low
bitrate and very high quality such that the binaural
algorithms can improve speech performance in noise
without unnecessarily decreasing the battery life of the
devices. We designed a new data reduction method
based on an efficient custom-tailored transmission of the
CI electrical stimulation patterns for meeting the
requirements. The algorithm has been combined with
binaural beamformers and evaluated in speech
intelligibility and quality in CI users.
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Spatial Release From Masking in Bilateral Cochlear Implant Users listening to the Temporal Limits Encoder Strategy
Alan Kan, Qinglin Meng
[Abstract]
Binaural hearing benefits with bilateral cochlear implants
(CI) are usually smaller than with normal hearing (NH) in
the same tasks. This gap in performance has typically
been attributed to a lack of coordinated stimulation
between ears and the high stimulation rates used in
clinical processors. These factors hinder sensitivity to
interaural timing differences (ITDs); an important binaural
cue for NH listeners. The Temporal Limits Encoder (TLE)
strategy was originally designed to encode unilateral low-
frequency temporal fine structure pitch cues into the
signal envelope of CIs. However, TLE also lowers the
stimulation rate on some channels which may potentially
provide useable ITD cues. We measured spatial
release from masking (SRM) in bilateral CI users listening to
TLE vs Advanced Combinational Encoder (ACE) strategy
to determine if TLE provides a benefit. The CCiMobile
research platform was used for testing. Results from eight
listeners showed comparable word recognition
performance in quiet and co-located conditions for TLE
and ACE, even with a short acclimatization period with
TLE. In the spatially-separated condition, performance
across the group was less variable with TLE than ACE, and
more listeners showed SRM. These results indicate that TLE
has the potential for improving binaural hearing benefits
for CI users.
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Influence of asymmetric processing delays on the localization ability of bimodal CI/HA users
Stefan Zirn, Julian Angermeier, Thomas Wesarg
[Abstract]
Under optimal hearing conditions the normal-hearing auditory
system can perceive interaural time differences (ITD) as little as
10 µs. Due to physiological limitations, the largest occurring ITD is
about 700 µs. Differences in processing latencies of digital
hearing aids (HA) and cochlear implants (CI) can be as high as
9 ms in bimodal users. These differences result in an interaural
stimulation timing mismatch that superimposes physiologically
occurring ITDs. This interaural stimulation timing mismatch could
account for the reduced sound localization performance in
bimodal users, which has been reported by other authors. To
investigate this hypothesis, we conducted localization tests in
bimodal CI/HA users with and without a stimulation timing
mismatch. To compensate for this mismatch, we delayed the CI
stimulation according to the individual HA processing latency by
a portable, programmable delay line based on a
microcontroller. First, an initial sound localization test without
compensation was conducted. Then, the delay line was
activated. After a familiarization period to the compensated
stimulation timing mismatch the test was repeated. The results
show a highly significant improvement of the localization ability
in the horizontal plane after compensation.
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Benefits and Challenges of Bimodal Hearing in Children
Melissa Polonenko
[Abstract]
Binaural hearing is important for listening in and navigating everyday acoustic
environments. Unilateral deafness during development drives extensive reorganization
in bilateral auditory pathways, limiting spatial hearing and putting children at risk for
social and educational challenges. In an effort to reduce these listening challenges and
support bilateral development, implantation criteria are expanding to provide children
who have asymmetric hearing loss a cochlear implant for the deaf ear and a hearing aid
to the other ear when needed, known as bimodal hearing. Our work has assessed
hearing development in bimodal users through evoked potentials and behavioural
measures of spatial hearing and speech perception. This talk will discuss how bimodal
hearing can promote symmetric bilateral development but faces limitations for
preventing or reversing asymmetry-driven reorganization when implantation is delayed
or residual hearing is insufficient. Challenges still faced by children who hear bimodally
will also be discussed.
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Organization:
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N. Ma, University of Sheffield |
D. Kolossa, Ruhr University Bochum |
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Auditory Models Comparison for Horizontal Localization of Concurrent Speakers in Adverse Acoustic Scenarios
Roberto Barumerli, Michele Geronazzo, Andrea Almenari, Giorgio Maria Di Nunzio, Federico Avanzini
[Abstract]
This paper aims at comparing and replicating the predictions of two public available computational auditory models for speaker localization in different simulated environments. The direction-of-arrival (DOA) of sound sources in the horizontal plane can be extracted by using binaural spatial cues from room and user acoustics. Since our predictions consider the specificity of both models at the level of peripheral processing, the proposed solution for DOA extraction provides a common multi-conditional training for the Gaussian Mixture Model (GMM) approach. A set of acoustic simulations of adverse conditions (i.e. multi speakers or high reverberant scenarios) support the evaluation phase on robustness of the synthetic auditory process. Our analysis replicates four case studies from the scientific literature in order to investigate the reliability of localization predictions in the frontal horizontal plane. Finally, a newly defined acoustic scenario allows to identify differences between auditory models outcome in the entire horizontal plane. The results show a good agreement with previous literature and our machine learning approach emphasizes peculiarities of each approach for auditory peripheral processing.
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Decoding the neural processing of selective attention to speech
Tobias Reichenbach
[Abstract]
Understanding speech in noisy backgrounds requires selective attention to a particular speaker. Humans excel at this challenging task, while current speech recognition technology still struggles when background noise is loud. The neural mechanisms by which we attend selectively to a particular speech signal remain, however, poorly understood, not least due to the complexity of natural speech. Here we describe recent progress obtained through applying machine-learning to neuroimaging data of humans listening to speech in background noise. In particular, we develop statistical models to relate characteristic features of speech such as pitch, amplitude fluctuations and linguistic surprisal to neural measurements. We find neural correlates of speech processing both at the subcortical level, related to the pitch, as well as at the cortical level, related to amplitude fluctuations and linguistic structures. Our findings may be applied in smart hearing aids that automatically adjust speech processing to assist a user, as well as inform future speech-recognition algorithms.
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Prediction of Human Listeners' Speech Recognition Performance Based on Automatic Speech Recognition
Mahdie Karbasi, Dorothea Kolossa
[Abstract]
There have been many past and recent efforts to improve the
accuracy of instrumental measures in predicting the intelligibility
of speech signals. However, modeling human auditory
perception and, more specifically, human speech recognition,
is a challenging task and it is still difficult to achieve a reliable
prediction that is consistently close to human listening results. In
addition, most of the intelligibility measures introduced so far are
relying on the information acquired using both, the distorted
signal and its correspondent clean version, which constitutes a
serious drawback for such systems.
Automatic speech recognizers have recently been introduced
as a useful tool for predicting the intelligibility directly, without
requiring the clean version of the distorted input signal. In this
work, we will consider some discriminative measures that can be
extracted utilizing the models trained for an ASR and investigate
the accuracy of these measures in predicting the human
listening test outcomes. The ASR-based discriminative measures
are computed for small units of speech like words or phonemes
and can be used to predict the intelligibility of such units of
speech. This study will present a detailed analysis of the
performance of the proposed model-based measures in
comparison to some well-known signal-based intelligibility
measures.
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Modelling of Binaural Speech Intelligibility for Hearing Impaired Listeners Using Intrusive and Non-Intrusive Binaural Speech Intelligibility Models
Christopher Hauth, Marc René Schädler, Anna Warzybok
[Abstract]
Modelling binaural speech intelligibility for hearing impaired listeners and comparing the results to measured speech reception thresholds (SRTs) can help to identify and understand contribution of different factors of hearing impairment (e.g. audibility) to speech perception. This study investigates SRTs obtained for 16 hearing-impaired listeners in a spatially co-located (S0N0) and a spatially separated (S0N90) speech in noise task using the Polish sentence test in noise. The obtained data are predicted using binaural speech intelligibility model (BSIM), which combines an Equalization-Cancellation process as a model of binaural unmasking with the speech intelligibility index. Hearing impairment is modelled by incorporating the individual audiogram as additional internal noise sources into the BSIM. First results indicate that BSIM overestimates the influence of the audiogram on SRTs already in the S0N0 condition. The standard deviation of SRTs across listeners is much smaller than in the predicted data. On the other hand, the predicted binaural release from masking, which is the difference in SRT between the S0N0 and S0N90 scenario, shows good agreement with the individual perceptual data (R2=0.83). The predictions obtained with BSIM are compared with predictions using the non-intrusive ”Framework for Auditory Discrimination Experiments” by incorporating a binaural feature extraction.
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Organization:
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V. Hohmann, University of Oldenburg |
P. Zahorik, University of Louisville |
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Audiovisual Speech Perception: Time for a paradigm shift
Nancy Tye-Murray
[Abstract]
Most adults cannot lipread very well, and when viewing only the
visual speech signal, recognize very little of the content. Even so,
the visual speech signal greatly enhances a degraded auditory
speech signal, as when listening in the presence of background
noise or with hearing loss. For example, in a study performed at
Washington University in St. Louis, older adults recognized less
than 10% of the test words when presented in a vision-only
condition, yet when the visual signal was added to a degraded
auditory signal that was only 540% intelligible, performance
jumped to 80% correct. In this presentation, we will consider
various explanations for this super-additive effect and then
suggest that audiovisual integration is not considered a distinct
ability and the super-additive effect can be completely
accounted for by unimodal performance. Two studies, which
differed in testing conditions and speech material, demonstrate
that listening and lipreading abilities are sufficient to predict an
individual’s audiovisual speech recognition in noisy situations
and that neither age nor an assumed integrative ability further
enhances predictions.
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Audiovisual speech perception in cochlear implant recipients – examinations based on virtual reality
Hartmut Meister
[Abstract]
Cochlear implants (CI) aim at (partially) restoring listening in patients with profound hearing loss by electrical stimulation of the auditory nerve. CI recipients frequently have particularly good skills in using visual speech. This may be due to the need to overcome the relatively limited cues provided by the electrical stimulation thereby supporting the outcome of the CI. Hence, audiovisual speech plays an especially crucial role in CI users. Moreover, assessing audiovisual perception has high ecological validity since daily-life communication typically contains both, auditory and visual speech cues.
However, audiovisual speech is often not systematically considered with CI rehabilitation, partly due to the lack of appropriate speech material. One promising approach is the use of virtual reality (VR) to provide visual speech cues. So called ”talking heads” allow visualization of arbitrary speech material and have the advantage of combining realistic settings with precise control of stimulus presentation.
The present talk gives insight into VR-based audiovisual speech presentation and shows several research areas addressing speech perception in cochlear implant listeners.
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Audiovisual Speech Perception in Children with Autism Spectrum Disorders
Julia Irwin, Trey Avery, Daniel Kleinman, Nicole Landi
[Abstract]
In face-to-face conversation, when a speaker talks, the
outcome of their speech can both be heard (audio) and seen
(visual). We employed a novel visual phonemic restoration
paradigm to assess neural signatures using event related
potentials; ERPs of audiovisual processing in typically
developing children and in children with ASD. Two types of
auditory stimuli were alternately presented with video of a
speaker saying the consonant-vowel syllable /ba/; 1) a
synthesized consonant-vowel syllable /ba/ or 2) a
synthesized syllable derived from /ba/ in which auditory cues
for the consonant were substantially weakened and sounds
more like /a/. The auditory stimuli are easily discriminable,
however, in this novel paradigm, in the context of a visual
/ba/ the auditory /a/ should be perceived as /ba/ (a visual
phonemic restoration), leading to an attenuated ERP
discrimination response. These stimuli were presented in
both passive (simply watch and listen) and active (button
press when /a/ is heard) conditions. Children with ASD
continued to show a large /ba/-/a/ discrimination response in
the presence of video of the speaker producing /ba/ in the
passive, but not active condition, suggesting reduced
influence of visual speech may be modulated by attention.
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Audio-visual stimuli for the evaluation of speech-enhancing algorithms
Giso Grimm, Gerard Llorach, Maartje Hendrikse, Volker Hohmann
[Abstract]
The benefit from speech-enhancing algorithms in hearing devices may depend not only on the acoustic environment, but also on the audio-visual perception of speech, e.g., when lip reading, and on other visual cues. In particular, the functioning of speech-enhancing algorithms depends on the motion behaviour of the user, which in turn depends on visual cues. In this presentation we present various audio-visual stimuli used for evaluation of speech-enhancing algorithms in hearing devices. The stimuli include video recordings of the Oldenburg sentence tests (OLSA), real-time animation of lip movement for animated characters, and complex audio-visual environments. We will discuss the effects of the material on speech perception and gaze and head motion behaviour, and outline applications of these stimuli. Results show that video recordings of the OLSA material are benefitial in speech reception thresholds. This benefit can not be found with the animated lip movement using a simple vocal tract model. However, it was found that the animations are sufficient to achieve natural motion behavior. This presentation is related to contributions at this conference of Llorach et al. on details of audio-visual speech test material, and Hendrikse et al. on natural motion behavior and the influence on hearing aid algorithm performance.
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Audio-visual scene analysis in reverberant multi-talker environments
Axel Ahrens, Kasper Duemose Lund, Torsten Dau
[Abstract]
Normal-hearing subjects are accurate in localizing sound
sources even in reverberant multi-talker environments (e.g.,
Kopčo, 2010; Weller, 2016). Weller et al. (2016) showed that
subjects can accurately analyse reverberant multi-talker scenes
with up to four simultaneous talkers. While multi-talker scene
analysis has mainly been investigated with only auditory
information, the addition of visual information might influence
the subjects’ perception. To investigate the visual influence,
audio-visual scenes with a varying number of talkers and
degrees of reverberation were considered in the present study.
The acoustic information was provided using a spherical
loudspeaker array and the visual information was provided
using head-tracked virtual reality glasses. The visual information
represented various possible talker locations and the subjects
were asked to identify the number of talkers and their specific
locations. For the identification of talkers, subjects had to label
visual locations with headlines from the talker’s speech topic. It
was hypothesized that the addition of visual information
improves subjects’ ability to analyse complex auditory scenes,
while the amount of reverberation impairs the overall
performance.
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Loudness and distance estimates for noise bursts coming from several distances with and without visual cues to their source
Gauthier Berthomieu, Vincent Koehl, Mathieu Paquier
[Abstract]
The at-ear sound pressure decreases with the distance to the source. Since pressure plays an essential part in loudness judgments, loudness could be expected to decrease similarly with the distance to the source. Yet, several studies reported conflicting results when assessing loudness as a function of sound source distance. The present study gathers loudness judgments for noise bursts displayed from several distances from listeners in a virtual environment. A visual representation of the distant sound sources could or could not be available to the listeners in order to modify the perceived distance of these sounds since visual distance cues affect auditory distance perception. Sounds were displayed in free or diffuse fields in order to also provide auditory distance cues. The results of this study show that visual cues had no significant effect on loudness estimates, even though they significantly affected apparent source distance judgments. The discrepancy between these results and the generally accepted loudness constancy phenomenon could come from the instructions that were given to the listeners, which focused on “at-ear” loudness judgments whereas loudness constancy is predominantly obtained when asking the listeners to make loudness judgments based on the sound source power.
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Auditory perception of distance to rattlesnakes in an audio-visual virtual environment
Michael Schutte, Michael Forsthofer, Boris Chagnaud, Lutz Wiegrebe
[Abstract]
Rattlesnakes shake their rattles to drive away animals which they perceive as threatening. Current research has shown that they change the rate at which they rattle according to the distance of an approaching object: the closer the object, the more rattling sounds per second they produce. In an audio-visual VR experiment, we test the hypothesis that this behaviour makes human listeners underestimate their distance to a snake. Humans are passively moved towards an invisible snake in a visual virtual environment (a long-grassed meadow), starting at a distance between 8 m and 1.4 m. Rattling sounds are synthesised and rendered via a vertical loudspeaker array that faithfully transmits loudness and elevation cues. Humans are instructed to stop their movement when they estimate the snake to be exactly 1 m away. We observe that humans stop at significantly larger distances when we simulate a rattling behaviour true to biology, i.e., including the (nonlinear) distance-dependent change in rattling frequency, compared to a virtual snake that rattled at a constant rate. Our data show that the distance-dependent rattling mediates systematically erroneous but useful distance information and is suitable to make human listeners think that they are closer to the snake than they really are.
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Organization:
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J. Bizley, University College London |
R. Maddox, University of Rochester |
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Reaction times in multisensory localization tasks
Yi Zhou, Colton Clayton
[Abstract]
The encoding of space between the visual and auditory systems does not always align. For foveal species like humans and monkeys, the visual field is restricted to the frontal space, whereas the auditory field is panoramic, covering the entire surrounding space. The rear sensitivity brought by spatial hearing is critical for avoiding unseen dangers. Our recent study shows that, in humans, vision’s influence on auditory perception can extend to unseen rear space (Montagne and Zhou, 2018). The present study further investigates the effects of visual stimulation on reaction times for localizing an auditory event presented from the front or rear. Experiments are designed to simultaneously survey two types of reaction times: (1) choice reaction time for indicating front or back direction of a sound by pushing a button; (2) saccade reaction time for indicating the lateral direction of a sound by making a gaze shift. Our results show domain-specific effects of visual capture on both types of reaction times and support our previous findings that audio-visual interactions are not limited by the spatial rule of proximity.
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Relatively Speaking: Spatial Discrimination Tasks As Tools for Studying Multisensory Integration
Ross Maddox
[Abstract]
Both the visual and auditory systems provide spatial
information. Because many objects can be both seen and
heard, the brain combines information from each sensory
modality to form a single estimate of an object’s location.
When there is conflict between the cues from a single
object, those cues are weighed by their reliability, which
usually results in a visually-driven bias of auditory
localization, namely the ”ventriloquist effect.” Numerous
studies of crossmodal integration have described the
phenomenon behaviorally, modeled it, and explored its
neural underpinnings. The overwhelming majority of these
studies have used absolute localization tasks, for which
localization bias is the principal outcome, rather than
discrimination tasks (i.e., relative localization), which
measure spatial acuity. Studies from our lab suggest this
represents a missed opportunity. We have used spatial
discrimination to show effects of eye gaze on auditory
spatial coding unrevealed by absolute localization tasks.
We have also used discrimination in tasks where auditory
and visual stimuli are collocated, so no localization bias is
expected. Finally, our preliminary results suggest that
spatial discrimination tasks reveal abnormal auditory
spatial processing in people with hemianopia (one-sided
blindness due to a visual cortical lesion), where studies of
auditory localization have suggested normal processing.
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Visual Recalibration of Auditory Spatial Perception Decays at Different Time Scales
Patrick Bruns, Brigitte Röder
[Abstract]
In human observers, exposure to audiovisual stimuli with a consistent
spatial disparity results in an adjustment of auditory spatial
representations that corrects for the mismatch between auditory and
visual input (known as the ventriloquism aftereffect). Our recent
findings suggest that such adjustments occur in parallel at multiple
time scales. However, it is not clear if recalibration of auditory space,
once emerged, remains stable until new audiovisual evidence is
encountered, or would decay over time. Here we reanalyzed data
from a previous study to answer this question. In this study, human
participants localized sounds in complete darkness, immediately
before and after they were exposed to 200 audiovisual stimuli in
which the visual stimulus was consistently displaced to the right of
the sound source. A trial-by-trial analysis of the localization
responses in the auditory localization posttest suggested that the size
of the ventriloquism aftereffect was strongest at the beginning and
decreased over the course of the 96 unimodal localization trials.
Thus, cross-modal recalibration decayed although participants did not
experience any new audiovisual evidence. Despite this rapid decay, a
stable carryover effect on sound localization was observed on the
next day, suggesting distinct underlying mechanisms for immediate
and sustained cross-modal recalibration effects.
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Sensory integration in parietal but not auditory cortex mediates multisensory integration and recalibration
Hame Park, Christoph Kayser
[Abstract]
During the ventriloquist effect (VE) a sound is mislocalized
towards a visual cue. This VE induces a change in the
perceived location of a subsequent unisensory sound - the
ventriloquist after-effect, a form of multisensory recalibration
(VAE). We investigated the neural mechanisms of these
phenomena.
We measured magnetoencephalography (MEG) while
participants performed an auditory localization task.
Psychometric data confirmed both VE and VAE.
We used cross-validated classification to localize MEG source
activity encoding current and previous sensory information,
and regression modelling to determine when and where
neural activity reflecting auditory encoding is influenced by
the previous sensory information, and where this is predictive
of the VE and VAE response biases.
We found neural signatures of audio-visual integration (the
VE
bias) in superior temporal and parietal cortex. Of these,
activity in parietal regions mediated the behavioural
recalibration of unisensory auditory perception (the VAE
bias).
This suggests that parietal, but not auditory cortex, integrates
audio-visual information and combines this with persistent
representations about prior stimuli to guide adaptive
multisensory behaviour.
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Temporal attention and comodulation in multisensory causal inference
Virginie Van Wassenhove, Daria La Rocca, Denis Alexander Engemann, Philippe Ciuciu
[Abstract]
Perception relies on inferences about the causal structure of the
world provided by multiple sensory inputs. In ecological settings,
multisensory events that cohere in time and space benefit such
inferential processes: hearing and seeing a speaker enhances
speech comprehension, and the acoustic changes of flapping
wings naturally pace the visual scene crowded by a flock of
birds. I will illustrate by means of psychophysical and
magnetoencephalographic (MEG) studies how the human
brain may synthesize and represent multisensory temporal
comodulation in some abstract form, which may benefit (as
top-down predictions) the analysis of incoming sensory signals.
For instance, one recent finding (under review) shows the
emergence of large-scale synchronizations in high gamma (60-
120 Hz) and beta (15-30 Hz) bands following experiencing
temporally comodulated multisensory signals. The engagement
of prefrontal, parietal, and visual cortices suggest the possibility
that a short experience of temporally comodulated stimuli
signaling the same information may functionally re-route
unisensory processing.
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The Role of Audiovisual Temporal Coherence in Auditory Scene Analysis
Jennifer Bizley, Huriye Atilgan, Ana Isabel Sanchez Jimenez
[Abstract]
Perception is seamlessly multisensory, and interpreting an object
through multiple modalities offers advantages over unisensory
processing. We have recently demonstrated that audiovisual
temporal coherence can modulate the performance of human
listeners in an auditory selective attention task (Maddox et al.,
2015) and the responses of single neurons in ferret auditory
cortex to an acoustic scene (Atilgan et al., 2018). Stimuli were
independently amplitude modulated acoustic streams and a
radius (for human psychophysics) or intensity (for recordings in
ferrets) modulated visual stimulus. We demonstrated preferential
processing for sounds that were temporally coherent with the
visual stimulus. In these studies the coherent changes in visual
size/luminance and auditory amplitude bound information
across modalities. Critically, these benefits extended to non-
binding features of the sound - auditory pitch and timbre-
consistent with the formation of an audiovisual object (Bizley et
al., 2016). Training listeners to better discriminate audiovisual
temporal coherence increased their performance in the
auditory selective attention task and changed the way in which
they were able to benefit from a visual stimulus such that
temporal coherence between both target and masker sounds
was advantageous compared to an independently modulated
visual stream (Atilgan and Bizley, submitted).
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Multisensory Influences on Human Auditory Communication
Katharina Von Kriegstein
[Abstract]
Understanding what is said and recognising the identity of the talker are two important tasks that the brain is faced with in human communication. For a long time neuroscientific models for speech and voice processing have focused mostly on auditory language and voice-sensitive cerebral cortex regions to explain speech and voice recognition. However, our research has shown that the brain uses even more complex processing strategies for recognising auditory communication signals, such as the recruitment of dedicated visual face areas for auditory processing. In my talk I will give a brief introduction to this work and show how the multisensory influences on auditory processing can be harnessed to improve auditory learning.
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Ageing increases the impact of audiovisual synchrony on speech comprehension in adverse listening situations
Giulio Degano, Samuel Jones, Uta Noppeney
[Abstract]
Age-related hearing loss profoundly impairs speech
comprehension in noisy environments. Combining
psychophysics and fMRI we investigated the role of audiovisual
asynchrony (-250, 0, +250 ms) on intelligibility of audiovisual (AV)
speech in younger and older adults. AV speech was presented
with and without babble noise. Older and younger adults did
not differ in their ability to discriminate between synchronous
and asynchronous speech irrespective of the presences of
noise. Nevertheless, older participants’ speech comprehension
was significantly more impaired for asynchronous relative to
synchronous speech, when the speech was presented with
babble noise. At the neural level, we observed that this
decrement in speech comprehension for asynchronous AV
speech was mediated by activations in a widespread neural
system including the superior temporal sulci. These regions
showed an activation increase for synchronous relative to
asynchronous speech in noise in older adults, but a decrease in
younger adults (i.e. significant cross-over group x (a)synchrony
interaction). Collectively, our results demonstrate that
audiovisual synchrony can impact the neural processes
underlying speech comprehension differently in older and
younger adults, even though both groups are equally good at
discriminating between synchronous and asynchronous speech.
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Organization:
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B. Chandrasekaran, University of Pittsburgh |
L. Holt, Carnegie Mellon University |
A. Tierney, Birkbeck College |
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Promoting Efficient Sound Category Learning Using an Incidental Task
Sung-Joo Lim
[Abstract]
Speech signals are acoustically variable. Listeners must perceptually
map these acoustically variable speech signals onto linguistically-
relevant categories learned from one’s native language experience.
But how do listeners naturally acquire the speech categories from
their experience without any explicit training or directed feedback? I
will present a series of behavioral and neuroimaging studies that
demonstrate whether and how adult listeners successfully learn
sound categories without explicit training. I employed a videogame
task (Wade and Holt, 2005) in which sound category exemplars
incidentally guided successful game actions without feedback about
sound categorization. First, we found the incidental videogame
training effectively promoted non-native speech categories, such that
2.5 hours of videogame play resulted in learning gain on par with
weeks of standard training. Second, we used fMRI to examine the
neural mechanisms underlying the incidental sound category learning.
During game play, the posterior striatum was recruited and the
striatal connectivity to the left superior temporal sulcus predicted the
generalization of learned sound categories. In sum, these data
demonstrate that the striatum and its interaction with the auditory
cortex promote incidental learning of sound categories emerging from
naturalistic learning environments, and provide perspective on
optimal training protocols to promote efficient sound category
learning.
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Speech sound training alters auditory processing in rats
Crystal T Engineer
[Abstract]
Speech sounds evoke unique neural activity patterns in the
primary auditory cortex (A1) of rats. Like humans, rats can easily
learn to discriminate between speech sounds. Behavioral
discrimination accuracy can be predicted by the neural
similarity of the A1 response pattern to pairs of speech sounds.
For example, pairs of sounds that evoke very similar A1 activity
patterns are difficult for rats to discriminate, while pairs of sounds
that evoke very distinct A1 activity patterns are easy for rats to
discriminate. Rat models of autism exhibit many of the classic
neural and behavioral deficits observed in individuals with
autism. Extensive speech discrimination training alters auditory
cortex responses in both experimentally naïve rats and in rat
models of autism. However, in some autism models, training
alone is insufficient to reverse neural processing deficits. Vagus
nerve stimulation (VNS) triggers rapid, phasic release of plasticity
promoting neuromodulators, which enhances plasticity in the
auditory network when delivered with sound presentation. For
example, pairing the sounds ërad’ and ëlad’ with VNS increases
the A1 response strength to the paired sounds. Ongoing work
involves pairing VNS with auditory training in rat models of
autism to improve both sound discrimination ability and the
neural processing of sounds.
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Effects of Short-term Training on Attentional Modulation of Neural Phase
Adam Tierney, Fred Dick, Aeron Laffere
[Abstract]
How does the brain follow a sound that is mixed with
others in a noisy environment? One possible strategy is to
allocate attention to task-relevant time intervals while
suppressing irrelevant intervals - a strategy that could be
implemented by aligning neural modulations with critical
moments in time. In this talk I will report results from several
experiments demonstrating that selective attention to
temporally interleaved non-verbal tone streams is linked to
phase shifts in EEG activity at the rate of within-stream
tone presentation. This paradigm provides an implicit
measure of non-verbal selective attention which strongly
correlates with performance in both adult and school-
aged populations. Moreover, we find that attentional
modulation of neural phase is stronger in trained musicians
and can be enhanced by as little as two hours of online
selective attention training. These results suggest that
phase timing is a robust and reliable marker of individual
differences in auditory attention and demonstrate the
possibility of rapid short-term plasticity in the mechanisms
of selective auditory attention. Finally, I will present pilot
data relevant to the question of whether short-term
training programs can be a successful remediation
strategy in populations who struggle to control attention.
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Organization:
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M. Chait, University College London |
Y. Boubenec, École Normale Supérieure |
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Neural Mechanisms of Temporal Processing in the Central Auditory System: A Theory of Gap Detection
Jennifer F. Linden
[Abstract]
Humans are remarkably sensitive to brief interruptions of ongoing sound. Gap-detection
thresholds are typically less than 6 ms in normal young adults, and often higher in older
adults, patients with developmental disorders, or subjects with auditory processing
difficulties. Gap-detection thresholds are therefore routinely measured in audiological
clinics to assess auditory temporal acuity. However, despite the simplicity of the gap-in-
noise detection task and its importance as a clinical tool, the neural mechanisms of gap
detection are still poorly understood. Here I describe recent insights into the neural mechanisms of gap detection gained from
studies of an unusual mouse model of gap-detection deficits. Neurophysiological data
and computational modelling reveal that central auditory responses to sound offsets
(disappearances) play a key role in defining the limits of gap-in-noise acuity.
Additionally, adaptive gain control in higher auditory brain areas increases gap-in-noise
sensitivity. These results indicate that gap-in-noise detection relies not only on
peripheral and brainstem mechanisms that produce precisely timed neural responses to
sound offsets and onsets, but also on higher central auditory mechanisms of adaptation
and intensity gain control. Thus, elevated gap-detection thresholds in patients with
auditory perceptual difficulties could arise from abnormalities in many different
auditory brain areas.
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Functional segregation of Ferret Auditory Cortex probed with natural and model-matched sounds
Agnès Landemard, Célian Bimbard, Shihab Shamma, Sam Norman-Haignere, Yves Boubenec
[Abstract]
Natural sounds exhibit a rich variety of statistics which are gradually processed, extracted, and explicitly encoded along the auditory pathway. At the top of this hierarchy, it is established that auditory cortex analyses sounds to provide a multi-dimensional representation of their time-frequency dynamics through a variety of modulation filters. Here we wanted to directly probe the similarity of sound representations between the activation of these spectro-temporal acoustics features and sound-evoked cortical responses. We took advantage of functional UltraSound imaging, a recent neuroimaging technique, to access hemodynamic responses throughout the auditory cortex of awake ferrets with a high spatial resolution (∼200µm). Using multi-dimensionality scaling, we observed a consistent match between the organization of natural sounds in the spectro-temporal space and in the neural space. Dissimilarity matrices showed similar clustering of sounds in both spaces. Interestingly, we show for the first time that the spatial pattern of hemodynamic responses, and thereby their representational similarity with the spectro-temporal features, is stable throughout the presentation of sounds (10 seconds).
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Theoretical Underpinnings of Statistical Processing of Complex Sounds
Benjamin Skerritt-Davis, Mounya Elhilali
[Abstract]
To make sense of our auditory surroundings, the brain tracks sound
sources as they evolve in time, collecting information over the recent
context to construct a model of the external world. Two processes are
at work here: 1) inferring the temporal window of relevant context,
and 2) building efficient representations of contextual information.
Though both of these processes are pervasive in everyday listening,
the computational mechanisms behind them are poorly understood.
We explore a perceptual model rooted in Bayesian theories of
perception that incorporates both of these processes, inferring the
relevant context window from the acoustic input and sequentially
building statistical representations of sounds. Using experimental
results from the laboratory, the model is used as a springboard to
interpret processes of statistical tracking of the acoustic world and its
role in facilitating perception of complex sounds.
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Temporal Integration Windows for Auditory Statistical Estimation
Richard McWalter, Josh H. McDermott
[Abstract]
Sound textures result from the superposition of many similar
acoustic events, as arise from falling rain or galloping horses.
Such sounds are often approximately stationary, and are
thought to be represented in the auditory system with time-
averaged statistics. Recent work has suggested that texture
statistics can be averaged over windows as long as several
seconds [McWalter and McDermott, 2018]. However, it remains
unclear whether all statistics are averaged over the same
extent, or whether averaging windows might differ across
statistics, potentially adapted to the variability of the statistic.
We measured integration windows for individual classes of
statistics from an auditory texture model [McDermott and
Simoncelli, 2011]. In a psychophysical experiment, listeners
judged which of two sound textures was most similar to a
reference texture. We measured performance for different
stimulus durations, using textures synthesized to vary in individual
classes of statistic. We found texture discrimination improved
with stimulus duration but then leveled off, presumably signaling
the averaging window used to estimate the statistics. However,
the performance asymptote occurred at different durations for
different statistics (∼150ms to ∼2.5s). The results reveal that
the extent of time-averaging varies across texture statistics, as
might be required for robust estimation of the statistics.
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Foreground-background decomposition in complex auditory scenes
Sabine Thomassen, Alexandra Bendixen
[Abstract]
Tremendous insight into auditory scene analysis has been
gained by using the ,ABA_ë auditory streaming paradigm as a
simplified model scenario. Since natural acoustic scenes
usually contain more than two potential sound sources,
recent efforts strive towards studying scene analysis with
more complex source configurations. We will present our
current work along these lines, focusing on the question of
how perceptual foreground and background are decomposed.
In subjective-report procedures, we show that listeners
spontaneously report hearing more than one sound source in
the perceptual foreground at once. To examine the validity of
listeners’ subjective reports, we are developing audio-visual
stimulus protocols that allow us to collect an objective
indicator of auditory foreground formation via
eye-tracking. The eye-tracking results are in good agreement
with the indicated auditory percepts, suggesting that
foreground perception is validly reported. To study how the
perceptual background is structured in complex auditory
scenes, we combine behavioral measures with
electroencephalography (EEG). Our results suggest that
integrated and segregated representations of the perceptual
background are held in parallel. We will discuss our findings
in the light of current theories of auditory scene analysis,
with special emphasis on the mutual exclusivity assumption.
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Emergence of deviance detection along the auditory neuroaxis and beyond: A neuronal correlate for predictive coding?
Manuel S. Malmierca
[Abstract]
Stimulus-specific adaptation (SSA) is the reduction in the responses to a common sound relative to the same sound when rare. SSA was originally described in the primary auditory cortex as the neuronal correlate of the mismatch negativity (MMN). The neural sources of MMN are located mainly within non-primary auditory cortex in humans and animal models. Moreover, SSA is also present as early as in the auditory midbrain and thalamus (IC and MGB).
In this talk, I will show our recent findings on recordings from single neurons in the IC, MGB and auditory cortex (AC) of anaesthetized rats and awake mouse to an oddball paradigm similar to that used for MMN studies. Our data demonstrate that most neurons in the non-lemnical divisions of the auditory brain show strong SSA and that there is a hierarchical emergence of prediction error signals along the central auditory system.
Taken together our results unify three coexisting views of perceptual deviance detection at different levels of description: neuronal physiology, cognitive neuroscience and the theoretical predictive coding framework. Financial support was provided by the Spanish MINECO (Grant # SAF2016-75803-P), and the European Union’s Horizon 2020 Research and innovation programme under the Marie Skłodowska-Curie grant agreement No 72209.
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Short-term Statistics and Lexical Experience Drive Predictions and Prediction Errors Along the Auditory Pathway
Carles Escera, Jordi Costa-Faidella
[Abstract]
Perception can be thought of as an act of inference, and the
brain as a predictive machine continuously generating internal
models of the causal dynamics of the world in an attempt to
interpret its observations. According to the predictive coding
theory, feedforward prediction errors and feedback predictions
flow across hierarchically organized brain areas in distributed
neuronal assemblies that support neuronal representations.
Despite the prevalence of this theory in the interpretation of
brain function and the wealth of studies showing indices of
prediction errors in neural responses, very few studies have
actually found evidence for predictive brain activation, nor it is
clear how far high-order predictions can travel upstream the
hierarchy. Here we present two studies aiming to shed some
light on these questions. In the first study, we show how the
hemodynamic response of primary and secondary auditory
areas is modulated by expectancy, increasing towards the
predicted appearance of a sound according to local
stimulation probabilities. In the second study, we show that
lexically driven predictions about word endings generate early
prediction error signals at different stages of the auditory system
hierarchy, and that such predictions are even capable to
modulate the early stages of acoustic encoding.
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The influence of nested patterns on sensitivity to pattern violations.
Juanita Todd
[Abstract]
In studies of auditory perceptual inference a violation of a locally
established pattern will elicit a component of the evoked potential
known as mismatch negativity (MMN). The amplitude of MMN is
considered to be ”precision-weighted” where MMN is largest when the
predictable pattern is very stable. Over a series of studies, we have
demonstrated that the amplitude of MMN to a local pattern violation
is subject to modulation by longer-term assumptions about the sound
sequence structure. In this presentation two independent datasets
using ”multi-timescale sequences” will be used to expose how MMN
amplitude to a local pattern violation is significantly reduced after a
longer-term assumption about the sequence has been violated. This
higher-order modulation is so strong that it yields a stronger
influence over MMN amplitude than the period of stability in the local
pattern. The data speak to the ability of the auditory system to learn
predictable structures on multiple timescales simultaneously and to
how layers of prediction interact to impact auditory evoked potentials.
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How the brain tracks the unfolding statistics of rapid sound sequences - evidence from brain imaging and pupillometry
Maria Chait
[Abstract]
I will present ongoing work in my lab using brain imaging (EEG, MEG and fMRI), behavioural and eye-tracking (pupillometry) experimentation to reveal how human listeners discover patterns and statistical regularities in rapid sound sequences. Sensitivity to patterns is fundamental to sensory processing, in particular in the auditory system, and a major component of the influential ëpredictive coding’ theory of brain function. Supported by growing experimental evidence, the ëpredictive coding’ framework suggests that perception is driven by a mechanism of inference, based on an internal model of the signal source. However, a key element of this theory - the process through which the brain acquires this model, and its neural underpinnings - remains poorly understood. Our experiments focus on this missing link. The research approach, based on measuring behavioural and brain responses to rapid tone-pip sequences governed by specifically controlled rules along a variety of feature dimensions enables us to address questions related to (1) how the brain discovers patterns in sound sequences, (2) which neural mechanisms are involved, (3) to what degree the process is automatic or susceptible to attentional state and behavioural goals of the listener.
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Assessing the level of consciousness for individual patients using complex, statistical stimuli
Urszula Gorska, Bernhard Englitz
[Abstract]
Patients with prolonged disorders of consciousness (PDOC) are
often unable to communicate their state of consciousness.
Determining the latter is essential for the patient’s care and
prospects of recovery. Auditory stimulation in combination with
neural recordings has recently been highlighted as a promising
technique towards objective estimation of conscious
awareness. Here, we investigate the potential of statistically
defined acoustic stimuli to elicit EEG responses suitable for
distinguishing between 5 subject groups, ranging from
unconscious to responding.
We presented naturalistic textures to listeners, which
changed their statistics unexpectedly. Active listeners indicated
the change by button press, while all other groups (passive,
asleep, minimally conscious (MCS), unresponsive wakefulness
(UWS)) listened passively. We quantified the evoked potential at
onset and change in statistics and their descriptional complexity
(Lempel-Ziv complexity (LZC)).
On the group level, evoked potentials can separate
patients from healthy controls, but fail to differentiate between
the patient groups. LZC reliably differentiates between
individual patients, when compared with the classical
behavioral scale.
In summary, EEG signal complexity during changes of
complex acoustic stimuli provides an objective criterion for
distinguishing states of consciousness in clinical patients. These
results suggests a path toward a cost-effective tool to choose
appropriate treatments for non-responsive PDOC patients.
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Organization:
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A. Noyce, Boston University |
I.S. Johnsrude, The University of Western Ontario |
A.J. Billig, UCL Ear Institute |
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Mapping auditory specialization within human frontal cortex
Abigail Noyce, Sean Tobyne, Ray Lefco, James Brissenden, David Somers, Barbara Shinn-Cunningham
[Abstract]
Large swaths of lateral frontal cortex (LFC) are often
characterized as domain-general or multiple-demand, due to
their apparent recruitment in a wide range of cognitive tasks
(e.g. Duncan 2010, Fedorenko 2013). However, using fMRI to
contrast brain activity during auditory attention or working
memory with that brain activity during matched visual tasks has
robustly identified a number of bilateral auditory-biased regions
in lateral frontal cortex (Michalka 2015, Noyce 2017). The
supplementary motor area (SMA), transverse gyrus intersecting
precentral sulcus (tgPCS), caudal inferior frontal sulcus/gyrus
(cIFS/G), anterior central operculum (aCO), and frontal
operculum (FO) are bilaterally recruited for demanding auditory
- but not for visual - tasks. Resting-state functional connectivity
analysis confirms that these regions participate in a network with
other auditory structures in superior temporal lobe. A seed-to-
whole-brain analysis of functional connectivity (Tobyne 2017)
revealed additional regions of frontal cortex that are
significantly more strongly connected to the auditory network
than to the corresponding visual network. Both the auditory
frontal network and these extended network regions show
substantially lower levels of domain-general recruitment than do
corresponding visual network regions, suggesting that human
auditory cognition may rely on specialized cortical machinery.
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Spectral Properties of Primary and Non-primary Auditory Cortical Activity
Alexander J Billig, Björn Herrmann, Ariane E Rhône, Phillip E Gander, Kirill V Nourski, Beau F Snoad, Christopher K Kovach, Hiroto Kawasaki, Matthew A Howard, Ingrid S Johnsrude
[Abstract]
Direct recordings from neurosurgical patients undergoing
invasive monitoring for epilepsy provide an opportunity to
resolve distinct auditory cortical fields on the supratemporal
plane. Short-latency (< 20 ms) responses and prominent phase
locking to 100-Hz click trains are consistently found in
posteromedial Heschl’s gyrus (HG), indicating a primary field
(Brugge et al., 2009, J. Neurophysiol.). More anterolateral
recording sites, presumably belonging to a non-primary field,
exhibit longer latency responses with considerably less phase-
locking. To more fully characterise electrophysiological
properties of primary and non-primary auditory cortex we
analysed activity in the frequency domain, prior to and during
the presentation of clear and degraded sentences. In contrast
with posteromedial HG, where power in the high gamma range
(70-150 Hz) dominated, anterolateral HG was characterised by
high spontaneous alpha (7-10 Hz) activity that was strongly
suppressed from 300 ms after the onset of a stimulus until its
offset. This suppression was more pronounced in response to
clear than to degraded speech. We consider possible
explanations for differences in the spectral profile of primary
and non-primary auditory cortical activity.
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Temporal Context Invariance Reveals Neural Processing Timescales in Human Auditory Cortex
Sam Norman-Haignere, Laura Long, Orrin Devinsky, Werner Doyle, Guy McKhann, Catherine Schevon, Adeen Flinker, Nima Mesgarani
[Abstract]
Natural sounds like speech and music are structured at many
timescales. But it remains unclear how these diverse timescales
are cortically represented. Do processing timescales increase
along the putative cortical hierarchy? Are there distinct
processing timescales for speech or music? Is there hemispheric
or anatomical specialization for processing particular
timescales? Answering these questions has been challenging
because there is no general method for estimating integration
periods: the time window within which stimulus features alter the
neural response. Here, we introduce a simple experimental
paradigm (the ”temporal context invariance” paradigm) for
inferring the integration period of any time-varying response. We
present sequences of natural sound segments in which the
same segment occurs in two different contexts. We then
measure how long the segments need to be for the response to
become invariant to the context. By applying this paradigm to
human electrocorticography data from epilepsy patients, we
map neural processing throughout human auditory cortex. This
map reveals a clear gradient in which integration periods grow
substantially as one moves away from primary auditory cortex,
providing support for hierarchical models. We also show that
speech selectivity first emerges at timescales of approximately
300-500 ms, suggesting selectivity for syllabic or word structure.
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Spatial processing in the auditory cortex for stream segregation and localization
Martha M Shiell, Elia Formisano
[Abstract]
The further apart two sounds are along the horizontal azimuth, the more likely they are to be heard as two separate sources. Previous research with neuropsychology patients indicates that this function of spatial stream segregation (SSS) may be dissociable from that of locating sound sources. Here, we test if these functions can be dissociated in the primary and posterior auditory cortex. We measured brain activity in ten normal-hearing adults with 7T functional magnetic resonance imaging. Participants listened to two concurrent auditory streams spatially-separated along the horizontal azimuth. The streams were either (1) spanning the frontal left quadrant, (2) clustered near the midline, or (3) clustered in the far-left periphery. Due to differences in spatial separation, SSS was easier in condition 1 than in conditions 2 and 3. A preliminary multivariate pattern analysis indicates that conditions that differed in SSS (e.g. conditions 1 vs. 2, and 1 vs. 3) have corresponding brain activity patterns that are better decoded than conditions that differ only in location (e.g. conditions 2 vs. 3), suggesting that spatial processing in the auditory cortex is more related SSS than to localization. In subsequent analysis we will compare the distribution of activity patterns in each of these functions.
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Assessing Perinatal Maturation of Human Primary and Nonprimary Auditory Cortex
Brian B. Monson
[Abstract]
In general, primary sensory cortex matures in advance of nonprimary sensory cortex, but the timing differences between primary (pAC) and nonprimary auditory cortex (nAC) maturation in humans is unclear. Here we used diffusion MRI to characterize the maturational timeline of auditory cortex microstructure for preterm infants.
We analyzed longitudinal data from infants born preterm who each underwent diffusion MRI up to four times between 26 and 40 weeks postmenstrual age. We examined the variation in diffusion MRI parameters along the length of Heschl’s gyrus. We tested for associations between diffusion metrics in infancy and language developmental outcomes at age 2 years.
We were able to distinguish between pAC and nAC as early as 28 weeks postmenstrual age, a time at which the sulcal boundaries of Heschl’s gyrus are just beginning to appear. Our analysis revealed differing rates of maturation along the axis of Heschl’s gyrus as cortex transitions from pAC to nAC. While pAC was further advanced along the developmental timeline at each timepoint, nAC showed much larger changes from 26 to 40 weeks postmenstrual age than did pAC. Disturbed maturation of nAC (but not pAC) was associated with poorer language performance at age two years.
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What should inter-individual and inter-primate differences tell us about auditory cortical organization?
Fred Dick, Francesco Caprini, Marty Sereno
[Abstract]
Our current conceptualization of human auditory cortical
organization is largely based on a synthesis of results from a
relatively small number of exceptionally detailed
electrophysiological and histology studies in macaque, owl,
and squirrel monkey from the 1970s-early 2000s. Such a
synthesis requires mentally quilting together small, partially
overlapping patches of mapped cortex from multiple
individuals, with the necessary assumption that individuals within
a species - and across related species - share a common
organizational scheme. This approach has been quite
successful in predicting patterns of human auditory
regionalization as shown in fMRI and electrocorticography
studies. However, the power of MRI to image all of cortex in
many individuals at multiple timepoints has also shown that the
topography of tonotopic and inferred ’myelin maps’ can vary
considerably across people, and even between the two
hemispheres of a single person. Here, we compare individual
auditory maps from a large (>50) combined tonotopy and
multiparameter map study to each other, and to previously
published human and non-human primate auditory maps and
theoretical schemes. We show there is significantly more ’true’
variability in auditory organization than is generally
appreciated, and suggest ways that such variability can inform
models of cortical development and regionalization.
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Organization:
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M. Cooke, University of the Basque Country |
V. Hohmann, University of Oldenburg |
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Speech enrichment: Listening effort and intelligibility
Anna Exenberger, Paul Iverson
[Abstract]
In this study, we investigated differences between a hearing impaired (HI) and a normal hearing (NH) group on subjective, behavioural and neurophysiological measures of speech processing and listening effort. We fitted both groups with a hearing aid (HI for hearing loss compensation and NH with minimal amplification for comparison), and asked them to perform a speech-in-noise task. The target speech was presented in multi-talker babble noise in two SNRs (+5dB and 0dB). The hearing aid had three different settings: hearing loss compensation only, noise cancelling and beam forming. EEG (64 channels) was recorded during the experiment in order to analyse cortical entrainment as well was intersubject correlation. Results revealed a marked difference between cortical entrainment and other measures; whereas the subjective listening effort measure, behavioural performance (intelligibility) and intersubject correlation (as a measure of auditory attention) showed relatively small differences between the NH and the HI group, we found substantially larger cortical entrainment for the HI group, specifically in higher SNRs (+5dB). We suggest that these results might indicate effects of listening effort measurable specifically in the auditory processing stage.
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Fundamental research on the verbal transformation effect in Japanese
Seiya Funatsu, Masako Fujimoto
[Abstract]
Verbal transformation effect is the phenomenon when a person hears a repeated single word without pause, illusory changes of the physically unchanging word are induced. In Japanese, there are few studies of verbal transformation effect. Therefore, in this study, we examined whether the verbal transformation effect occurs related to the change of inter stimulus interval (ISI), and a word that has an emphasized syllable and non-sense words induce verbal transformation effect in Japanese. ISI was 0, 0.02, 0.05, 0.1, and 0.15 sec. The subjects were 16 native Japanese. Stimuli consisted of words with 3 syllables. Regarding ISI, the perceptual transition time changed with ISI change. The mean number of stimulus repetition until perceptual transition occurred was 24.8 times in ISI 0 sec, but 57.3 times in ISI 0.15 sec. ANOVA showed a significant effect in the number of stimulus repetitions (p=0.0440). Moreover, post hoc test revealed that the number of stimulus repetitions in ISI 0.15 sec was significantly larger than that in ISI 0 sec (p=0.0051). Although in the case of non-sense words, the verbal transformation effect occurred, perceptual transition time varied according to each word. The mechanisms of the verbal transformation effect are discussed.
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Talker intelligibility and listening effort: The role of speaking rate
Maximillian Paulus, Valerie Hazan, Anita Wagner, Patti Adank
[Abstract]
Listening to speech in noise induces increased cognitive effort and
fatigue (Pichora-Fuller et al., 2016). Effort is mediated by
characteristics of the listener such as nativeness or hearing status.
Moreover, the issues with noise can be exacerbated by talker
differences (Hazan & Markham, 2004). Nevertheless, past research
has so far not accounted for cognitive effort. Based on an
anechoically recorded corpus of sixteen Southern British English
speakers, we conducted listening experiments in combination with
pupillometry. We presented temporally or spectrally distorted speech
(using time-compression and noise-vocoding, respectively).
Furthermore, undistorted speech was presented in quiet and with
speech-shaped masking noise. Intelligibility scores were obtained
based on keywords recognised correctly. Listening effort was
measured by tracking pupil size over time. Our results are in line with
previous studies measuring talker intelligibility in noise. Pupil dilation
was increased for both, distorted and masked speech. Pupil dilation
was also increased for less intelligible talkers, indicating increased
listening effort. Further studies are necessary to disentangle talker
intelligibility and listening effort. The current results are interpreted in
the context of models of listening effort such as the Framework for
Understanding Effortful Listening (FUEL; Pichora-Fuller et al., 2016).
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Differences between Native and Non-Native Lombard Speech in terms of pitch range
Katherine Pearl Marcoux, Mirjam Ernestus
[Abstract]
Lombard speech, speech produced in noise is acoustically
different from speech produced in quiet (e.g., Summers et al.,
1988) and is more intelligible in noise compared to speech
produced in quiet (e.g., Dreher & O’Neill, 1957). Extensive
research on native produced Lombard speech does not
consider difficulties non-natives may encounter. Non-native
speakers experience a higher cognitive load (e.g., Kormos,
2006) and their native and non-native languages may have
different acoustic patterns (e.g., Van Bezooijen, 1995). We
therefore investigated several acoustic measurements in native
and non-native Lombard speech, which may affect intelligibility. Thirty Dutch and nine American-English speakers read 144
English question-answer pairs in quiet and in noise. These pairs
had early- or late-focus, which we expected to behave
differently due to post-focus compression (e.g., Xu 2011). We
conducted lmers on pitch range, finding no difference in the
two groups for early-focus, only the expected effect of noise,
resulting in a larger pitch range. In late-focus, the native
American-English had a larger increase in pitch range in noise
than the non-native Dutch-English. In addition to pitch range,
we will discuss other acoustic measurements including median
pitch, duration, and intensity.
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Benefits of the WaveNet-Based Speech Intelligibility Enhancement for Normal and Hearing Impaired Listeners
Muhammed Shifas Padinjaru Veettil, Carol Chermaz, Theognosia Chimona, Vassilis Tsiaras, Yannis Stylianou
[Abstract]
Speech perception becomes challenging in adverse listening conditions, hence increasing mental effort. Modification of recorded clean speech with the goal to increase its intelligibility is one approach for improving listening experience. Recently, we suggested a data-driven WaveNet-like speech intelligibility enhancement method which is based on the Spectral Shaping and Dynamic Range Compressions (SSDRC) approach. Both approaches achieve intelligibility gains by relocating energy in the time-frequency plane. In this study we have assessed the performance of the WaveNet intelligibility enhancer for both normal and hearing impaired participants using formal listening tests, in terms of intelligibility and sound quality. Intelligibility was measured as a percentage of correct words recalled (CWR) and quality was assessed via a comparative mean opinion score (CMOS). Furthermore, we compared performance among native and non-native listeners using English and Greek stimuli. We observed that modified speech was significantly more intelligible than plain across all listener groups. Hearing impaired candidates rated modified speech 30% higher on average (on a CMOS scale) than normal hearing subjects. The positive impact for the hearing impaired listeners might be attributed to the reallocation of energy to perceptually relevant frequency bands in which the hearing impaired are more sensitive.
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Individual Articulatory Control in Speech Enrichment
Chen Shen, Martin Cooke, Esther Janse
[Abstract]
Individual talkers may use various strategies to enrich their
speech while speaking in noise (i.e., Lombard speech) to
improve their intelligibility. The resulting acoustic-phonetic
changes in Lombard speech vary amongst different speakers,
but it is unclear what causes these talker differences, and
what
impact these differences have on intelligibility. This study
investigates the potential role of articulatory control in talkers’
Lombard speech enrichment success.
Seventy-eight speakers read out sentences in both their
habitual
style and in a condition where they were instructed to speak
clearly while hearing loud speech-shaped noise. A
diadochokinetic (DDK) speech task that requires speakers to
repetitively produce word or non-word sequences as
accurately
and as rapidly as possible, was used to quantify their
articulatory
control.
Individuals’ predicted intelligibility in both speaking styles
(presented at -5 dB SNR) was measured using an acoustic
glimpse-based metric: the High-Energy Glimpse Proportion
(HEGP). Speakers’ HEGP scores show a clear effect of
speaking
condition (better HEGP scores in the Lombard than habitual
condition), but no simple effect of articulatory control on
HEGP,
nor an interaction between speaking condition and
articulatory
control. This indicates that individuals’ speech enrichment
success as measured by the HEGP metric was not predicted
by
DDK performance.
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Near End Listening Enhancement in Realistic Environments
Carol Chermaz, Cassia Valentini-Botinhao, Henning Schepker, Simon King
[Abstract]
Speech playback is harder to understand in noise. Near End Listening
Enhancement algorithms try to overcome the problem by enhancing
the speech signal before it is played by a device. Different strategies
have been tried, achieving variable degrees of success in specific
noise conditions. Such technologies, however, are often tested in
artificial settings - with controlled noise sources and no reverberation.
The purpose of this study is to compare a set of state-of-the-art
algorithms based on different approaches (adaptive vs non-adaptive,
with or without a compensation for reverberation) in simulated real-
life scenarios. Binaural noise recordings and impulse responses of real
environments have been used to create two representative scenarios
in which speech playback may occur, namely a domestic and a public
space. A preliminary study with N=24 subjects revealed the need for
higher SNRs in the realistic settings (in comparison to controlled
noise conditions) in order to obtain the same levels of intelligibility for
plain speech. The goal of the main study is to assess the impact of
noise adaptivity and reverberation awareness in realistic scenarios, in
order to better understand how to make speech playback more
robust to noise in real-life situations.
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Listeners' Speech Rate Preferences in Stationary and Modulated Maskers
Olympia Simantiraki, Martin Cooke
[Abstract]
Durational changes to speech can improve intelligibility in noise
[1, 2], but what speech rate would a listener choose in such
conditions if allowed to do so? We introduce a technique which
enables listeners to adjust arbitrary speech properties in real
time while performing a speech-in-noise task, and apply it to the
question of preferred speech rate. Native listeners adjusted
speech rate while identifying word sequences in quiet,
stationary noise (3 SNRs) and modulated noise (5 modulation
rates). Listeners preferred faster rates in quiet, increasingly slower
rates in stationary noise as SNR decreased, and increasingly
slower rates in modulated noise as envelope modulation rate
increased. These findings complement those obtained using
traditional intelligibility measures. We speculate that
for tasks where intelligibility is near to ceiling, any consistent
listener preferences might be used as an indicator of optimal
listening effort and applied in the design of speech enrichment
algorithms. The listening preference technique can be
adapted readily to many scenarios such as different masking
conditions (e.g. competing talkers) and applied to other
speech features such as F0 and spectral tilt. [1] Adams et al (2012) Am. J. Audiol. 21:22-32. [2] Cooke and Aubanel (2017) JASA 141: 4126-4135.
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A Multifaceted Enrichment of Oesophageal Speech
Sneha Raman, Inma Hernaez, Eva Navas, Luis Serrano
[Abstract]
Post a laryngectomy, oesophageal speakers struggle with communication owing to limitations in producing speech. Oesophageal speech (ES), therefore, is of poor intelligibility and demands increased listening effort. With the aim of improving intelligibility in an Automatic Speech Recognition (ASR) context, we performed three enrichment tasks on oesophageal speech: one heavy-weight strategy (voice conversion) and two light-weight strategies (cleaning up of undesired silences and synthesis with Wavenet vocoder that generates high quality natural sounding speech). While voice conversion (targeted at a healthy speaker) is the conventional approach to improve oesophageal speech, we were motivated to try the aforementioned light-weight strategies, as they are simple and straightforward, and any improvement with these light-weight methods will add to the benefits gained from other methods. Voice conversion was the most useful enrichment technique in improving ASR scores. Synthesizing with Wavenet improved the voice quality of some parts of ES but it also added some unwanted glitches. As a result, it did not improve the ASR scores for ES. Removing undesired silences did not have any improvement in ASR for a well-trained ES speaker as there were few instances of undesired silences. However, for a poorly intelligible speaker it showed marginal improvement.
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Weighted Generative Adversarial Network for many-to-many Voice Conversion
Dipjyoti Paul, Yannis Pantazis, Yannis Stylianou
[Abstract]
The goal of voice conversion (VC) is to convert speech from a source speaker to that of a target, without changing phonetic contents. VC usually relies on parallel data for training, which limits its practical applications. Existing approaches are also limited in handling multiple speakers, since different models should be built independently for every speaker pair. To tackle that, variants of Generative Adversarial Network (GAN) were introduced that allows many-to-many mapping instead of learning all the pairwise transformations. Moreover, GAN-VC can handle non-parallel data, i.e., speakers do not need to utter the same sentences. In this paper, we suggest an algorithmic variation of GAN training where suitable weights are multiplied to the gradient of the Generator. Weights are calculated by putting more power to fake samples that fool the Discriminator which results in a stronger Generator. We refer to this variation as weighted-GAN (weGAN). In weGAN, the convergence of the training performance is accelerated. The suggested weGAN-VC achieves over 10% relative improvement against conventional GAN-VC concerning speech objective quality for the same number of epochs while informal subjective quality shows improvement for both speech quality and speaker similarity. Formal listening tests are undergoing and will be reported during the conference.
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Towards a Neural-Based Single Channel Speech Enhancement Model for Hearing-Aids
Muhammed Shifas Padinjaru Veettil, Claudio Santelli, Yannis Stylianou
[Abstract]
Advancements in machine learning techniques have promoted the use of deep neural networks (DNNs) for supervised speech enhancement. However, the DNN’s benefits of non-explicit noise statistics and nonlinear modelling capacity come at the expense of increased computational complexity for training and inference which is an issue for real-time restricted applications, like hearing aids. Contrary to the conventional approach which separately models the feature extraction and temporal dependency through a sequence of convolutional layers followed by a fully-connected recurrent layer, this work promotes the use of convolutional recurrent network layers for single-channel speech enhancement. Thereby, temporal correlations among inherently extracted spectral feature vectors are exploited, while further reducing the parameter set to be estimated relative to the conventional method. The proposed method is compared to a recent low algorithmic delay architecture. The models were trained in a speaker independent fashion on the NSDTSEA data set composed of different environmental noises. While objective speech quality and intelligibility measures of the two architectures are similar, the number of network parameters in the suggested enhancement method could be reduced by 50-60%. This reduction is highly beneficial for storage and computation constraint applications.
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Word error and confusion patterns in an audiovisual German matrix sentence test (OLSA)
Gerard Llorach, Volker Hohmann
[Abstract]
One of the established tests for speech intelligibility in quiet and
in noise is the Oldenburg sentence test OLSA. This test evaluates
the speech reception threshold (SRT), i.e., the speech level or
signal-to-noise ratio that leads to a specified word intelligibility,
using phonetically balanced 5-word sentences, where each
word is equally intelligible. The standard matrix sentence test is
audio-only, although it is known that adding visual information
to a speech intelligibility task can change the outcome, e.g.,
increasing the speech intelligibility, or inducing the McGurk
effect. Adding visual cues, i.e., being able to see the speaker,
could alter the homogeneity of the words with respect to their
intelligibility and the expected outcome of the matrix sentence
test. In this work we compare the word error and confusion
patterns of the audio-only trials versus the audio-visual ones.
Possible reasons of these effects are discussed.
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EEG alpha power as a measure of listening effort reduction in adverse conditions
Amy Jane Hall, Axel Winneke, Jan Rennies-Hochmuth
[Abstract]
Noise levels in everyday environments are typically low enough for speech comprehension, but over extended periods of time, listeners may experience fatigue. We measured the neurocognitive compensation required to separate speech from noise - listening effort - using electroencephalography, for unprocessed and AdaptDRC-enhanced speech.
AdaptDRC is a noise-dependent near-end-listening-enhancement algorithm which significantly improves the intelligibility of speech in noise and reduces subjectively rated listening effort (Rennies et al., 2018).
Participants (N=27) completed a listening effort task using a categorical listening effort scale, in which sentences were presented at five SNRs and in two types of noise. We also measured speech intelligibility, hearing levels, and cognitive abilities.
Subjective listening effort was significantly lower, and speech intelligibility was significantly higher for AdaptDRC speech than unprocessed speech (p<.001). Analyses of the neurophysiological data show that there is a non-linear relationship between SNR and spectral alpha power with a peak between 0-5dB, and that alpha power is sensitive to noise type and speech processing. Better working memory and selective attention scores predict lower subjective listening effort ratings.
These findings provide insight into the neurophysiological correlates of listening effort and aid the development of an objective measure of cognitive load for speech in noise.
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Differences in Processing Speech-on-Speech Between Musicians and Non-musicians: The Role of Prosodic Cues
Elif Canseza Kaplan, Deniz Baskent, Anita Wagner
[Abstract]
Musical training was suggested to grant normal hearing listeners an
advantage in speech-on-speech (Başkent & Gaudrain, 2016;
Swaminathan et al., 2015). Furthermore, musical training appears to
affect the timing of lexical resolution when processing quiet vs
masked speech (Kaplan et al, in prep): musicians exhibited less
lexical competition when a two-talker masker was added to target
speech, while there was no difference between groups in quiet. In the
current study, we investigate the underlying reasons for the
difference observed in processing speech-on-speech in both groups.
We employ the visual world paradigm, where participants are asked
to identify the image of the target word among images of a
phonological competitor (i.e. ham-hamster) and two unrelated
distractors. We manipulated the prosodic boundaries, which are
known to play a role in lexical resolution in quiet speech (Salverda,
2003): the target words either have regular or lengthened durational
cues. If musical training enables higher sensitivity to the acoustic
correlates of prosodic boundaries in processing masked speech, we
expect to observe more lexical competition and delayed lexical
resolution in musicians in the longer duration condition. We will
compare gaze-tracking and pupil data of both groups across
conditions.
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Evaluating Cognitive Load of Text-To-Speech (TTS) synthesis
Avashna Govender, Simon King, Cassia Valentini-Botinhao
[Abstract]
Current evaluation methods for text-to-speech (TTS)
synthesis rely solely on subjective rating scores. These tests
typically account mostly for how natural or intelligible the
voice is. With state-of-the-art systems, these measures are
approaching ceiling and therefore alternative measures such
as the cognitive load may become more meaningful. To our
knowledge, there is little or no recent work evaluating the
cognitive load of state-of- the-art text-to-speech systems.
We use pupillometry as a measure of cognitive load. The
pupil has been found to dilate upon increased cognitive effort
when carrying out a listening task. Currently we are
evaluating speech generated by a Deep Neural Network TTS
synthesiser. In our method, we generate stimuli that step
incrementally from natural speech to synthesized speech by
changing only a single feature at a time. Stimuli are
presented in both quiet and noisy conditions. Our hypothesis
is that poor F0 and spectral predictions in the synthesiser
demand more cognitive effort in the listener.
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Directional selectivity of auditory spatial attention in multi-talker environment.
Ryo Teraoka, Shuichi Sakamoto, Zhenglie Cui, Yôiti Suzuki, Satoshi Shioiri
[Abstract]
When people direct the auditory selective attention in a specific direction, the information from that
direction can be perceived easily even in a noisy environment. Understanding how the effect is
distributed spatially is important. The extent of spatial distribution may depend on the direction of
the attention focus, that is, it may differ between frontal and oblique directions. In the present
study, we investigated the dependence of the spatial tuning of auditory selective attention on the
focus azimuthal direction. To examine the spatial tuning, speech intelligibility in a complex multi-
talker environment was measured. In the experiment, a target sound and multiple distracting
speech sounds were presented from loudspeakers surrounding observers. By manipulating the
probability of the target presentation directions, the listener’s auditory spatial attention was
directed to a specific loudspeaker. The attracted direction was selected as -30, 0, or +30 degrees.
Results showed that the greatest improvement in intelligibility was observed at the attracted
direction, for all attracted directions. The spatial tuning of attention was narrower for the attracted
directions of both -30 and +30 degrees than that of 0 degrees. This means that the attention
spotlight is modulated by the direction of the attention.
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Ocular Correlates of Auditory Emotion Recognition in Hearing-Impaired Listeners
Julie Kirwan, Anita Wagner, Deniz Baskent
[Abstract]
Hearing-impaired (HI) individuals are shown to perform worse in
auditory emotion recognition tasks compared to normal hearing (NH)
individuals, however, it is still unclear if this is a top-down, or
bottom-up issue (Picou et al., 2018) i.e. whether the problem is
based on how hierarchically low auditory regions are unable to
process emotional stimuli accurately or how top-down processes are
incapable of regulating and classifying emotional stimuli. An index of
emotion recognition can be observed in pupil dilations, which have
been shown to dilate more for emotional stimuli in comparison to
neutral stimuli for visual emotion recognition, and recently also for
emotional compared to neutral speech. In this study, we investigated
pupil dilation and gaze fixations as a potential marker for emotion
recognition in the HI population. We recorded pupil dilations and
behavioural responses of NH and HI participants as they listened to
emotional speech and vocalisations. By investigating the pupil’s
response in this way, we present here insights into the ocular
correlates of vocal emotion recognition in the HI population.
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Organization:
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H. Fastl, AG Technische Akustik, MMK, TU München |
S. Kuwano, Osaka University |
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Binaural Modeling for Complex Environments
Jonas Braasch, Jens Blauert
[Abstract]
Functional binaural models have been used since the mid-20th
century to simulate laboratory experiments. The goal of this
chapter is to extend the capabilities of a cross-correlation
model so it can demonstrate human listening in complex
scenarios found in nature and human-built environments. A ray-
tracing model is introduced that simulates a number of
environments for this study. This paper discusses how the
auditory system is used to read and understand the
environment and how tasks that require binaural hearing may
have evolved over the course of human history. As use cases,
sound localization in a forest is examined, as well as the binaural
analysis of spatially diffuse and rectangular rooms. The model is
also used to simulate binaural hearing during a walk through a
simulated office-suite environment.
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Limits of Mixing Background Sounds to Foreground Sound Samples in Psychoacoustic Laboratory Experiments on Noise Annoyance
Eduardo Pelizzari, Armin Taghipour
[Abstract]
A variety of noise annoyance investigations are conducted as laboratory experiments. To simulate a more natural and realistic scene, it is preferable to add background ambient sound samples to the noise samples. However, possible effects of background sounds on annoyance ratings to foreground sounds should be either minimized or quantified. A laboratory experiment was conducted, in which background ambient sound samples were mixed to foreground helicopter noise samples. In a partially balanced incomplete block design, three factors were varied: noise’s sound exposure level (6 categories), background sound type (4 categories), and background sound pressure level (3 categories). The design was complemented by six treatments in which only helicopters and no background sounds were present. Each subject received 40 of the 78 treatments. Annoyance to each individual 20-s stimulus (i.e. treatment) was rated on the ICBEN 11-point numerical scale. It was investigated whether background sound type and/or level affected annoyance. The major predictor of annoyance was helicopter’s sound exposure level. Stimuli containing eventful background sounds were associated with higher annoyance ratings than those with less eventful or no background sounds. Background sound level did not affect annoyance significantly, as long as L_mix - L_flight < 1 dB.
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Annoyance modeling of construction noise using acoustical features, noise sensitivity and health condition
Jae Kwan Lee, Seo Il Chang, Soo Il Lee, Jae Woong Jang
[Abstract]
The construction site noise is the biggest cause of the noise
dispute in Korea. A listening experiment was conducted to
produce annoyance prediction models for four kinds of
construction noise that cause the most complaints. Those were
generated by the construction machineries such as pile driver,
excavator, pumping vehicle, and concrete mold removal
working. These four kinds of noise recorded at the construction
site were adjusted to the stimuli with 35 ∼ 80 dB(A), and they
were played to the subjects and the annoyance scores were
surveyed. The objective indicators of the stimuli and the survey
results were used to develop the annoyance models caused by
the construction noise. The annoyance model was developed
by multiple regression analysis and its acoustic features include
traditional indicators such as Leq, Lmax as well as indicators of
spectral feature and temporal feature proposed in recent
studies. The mean, maximum, and percentiles values of sound
quality indices such as loudness, sharpness, roughness and
fluctuation strength were used for the modeling. Noise sensitivity
and health condition which were obtained from subjective
questionnaire also incorporated into the annoyance prediction
models.
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Loudness of ramped and damped sounds that are temporally shifted across ears
Josef Schlittenlacher, Robbie Zhao, Brian C. J. Moore
[Abstract]
Previously we showed that amplitude-modulated sounds
were
louder when their modulation was out of phase across the
two
ears than when it was in phase. The level difference
required for
equal loudness (LDEL) between diotic presentation and an
interaural modulation phase difference of 180∘ was about
2 dB.
This effect could be explained by a loudness model in
which
binaural summation lags behind binaural inhibition. The
present
study investigated the binaural loudness of ramped and
damped sounds in a similar manner. Stimuli consisted of
trains of
1000-Hz tone pulses with linear rise and fall times with ratios
of
1:10 (damped sounds) or 10:1 (ramped sounds). Stimuli
contained 28 55-ms pulses, 14 110-ms pulses or 7 220-ms
pulses,
resulting in a stimulus duration of 1540 ms plus half the
pulse
duration for the interaurally shifted stimuli. The LDEL
between
diotic and interaurally shifted stimuli was close to 0 dB in all
of
these conditions. For a single 220-ms pulse, the LDEL was
1.4 dB
for damped sounds, and 3.0 dB for ramped sounds, the
diotic
sounds being louder. The difference between a single
pulse and
a pulse train suggests differences between short-term and
long-
term loudness judgments.
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Annoyance penalty of amplitude-modulated sound
Valtteri Hongisto, Petra Karoliina Virjonen
[Abstract]
Amplitude-modulation (AM) appears in various kinds of noise sources. The nature of periodic AM sound can be described by e.g. AM frequency, AM depth, and equivalent spectrum of sound pressure level (SPL). There is little research concerning the effect of these factors on annoyance. Our purpose was to determine the penalty of periodically AM sounds for a range of AM frequency, AM depth, and SPL spectrum. Forty participants rated the annoyance of 70 AM sounds and 22 reference sounds. The three independent variables of AM sounds were AM frequency (7 levels, from 0.25 to 16 Hz), AM depth (5 levels, from 1 to 16 dB), and SPL spectrum (2 alternatives). All AM sounds were played at 35 dB LAeq. Reference sounds were played at levels from 29 dB to 49 dB LAeq. They enabled the determination of penalty of AM sounds. All sounds were created from pseudorandom noise. The results showed a significant effect of all three independent variables on annoyance. The effect of both AM frequency and AM depth was very large. The penalty values varied from -1 to +12 dB. The results can be used in the assessment of the penalty of AM sounds.
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Annoyance of impulsive sounds – a psychoacoustic experiment involving synthetic sounds
Ville Rajala, Valtteri Hongisto
[Abstract]
BACKGROUND. NT ACOU 112 method presents a procedure to assess the prominence (an adjustment to LAeq) of impulsive noise using onset rate (OR) and level difference (LD). Our purpose was evaluate the method with respect to the threshold where the impulsiveness causes additional annoyance. METHODS. Thirty-two persons participated in the psychoacoustic experiment. The participants rated the annoyance of sounds using numerical rating from 0 (not at all) to 10 (extremely). The experiment involved eight reference sounds and 33 impulsive sounds. The level of all impulsive sounds was 55 dB LAeq. The impulsive sounds were created synthetically by changing the temporal pattern of wide-band noise. The OR ranged from 5 to 800 dB/s and the LD ranged from 5 to 40 dB. The reference sounds were created from steady-state broadband noise within 49-70 dB LAeq in 3 dB steps having the same spectrum as the impulsive sounds. They were used to assess the annoyance of impulsive sounds. RESULTS AND DISCUSSION. Statistically significant increment of annoyance was usually observed when OR≥10 dB and LD>5 dB. The findings can be exploited e.g. in the development of ISO PAS 1996-3 standard.
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Practical Experience with Psychoacoustics in Automotive Engineering
Uwe Letens, Arne Oetjen, David Goecke, David Maiberger
[Abstract]
Throughout the last decades the noise emission of passenger cars has been reduced considerably. Therefore today the development process of a new car regarding interior and exterior noise is more focussed on the complex interaction of several sound components instead of the sound pressure level. The application of psychoacoustics in the noise evaluation is of increasing importance. The sensory measures loudness, sharpness, tonality, roughness and impulsiveness are the most common ones.
After some basic considerations this paper will present a few selected samples of interior and exterior noise studies on passenger cars. By using the appropriate measures psychoacoustics can help to identify annoying noise components (for the purpose of ”sound cleaning”) and assists to set targets for the final vehicle sound (”sound design”).
Once the basic psychoacoustic measures are elaborated it’s possible to develop a ”noise metric” which can forecast a holistic noise evaluation. Similar to a real car driver it selects an appropriate set of driving conditions and a corresponding set of acoustic and psychoacoustic measures. In a multi-step evaluation process all results are normalized and weighted as in a multiple regression analysis and are summed up into a ”noise index”.
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Subjective and electrodermal responses to annoying vehicle sounds: Role of task load and noise sensitivity
Wolfgang Ellermeier, Florian Kattner, Ewald Klippenstein, Michael Kreis, Catherine Marquis-Favre
[Abstract]
In order to study noise annoyance in the laboratory, a total of N=59 participants were exposed to short 5-s segments of pass-by recordings of motor vehicles presented at levels ranging from 50 to 70 dB(A). In Experiment 1, they just rated the annoyance due to sounds on a visual-analogue scale, in Experiment 2, they had to simultaneously perform a visual multiple-object tracking task. Annoyance was judged to be lower when participants were engaged in the cognitively demanding task (in Experiment 2). Presenting sounds spanning two different decibel ranges in separate blocks (in Experiment 1) did not affect the function relating level to perceived annoyance very much, suggesting the judgments to be robust with respect to stimulus range effects. Across both experiments, the magnitude of simultaneously recorded phasic skin conductance responses increased significantly with the sound pressure level of the annoying vehicle sounds. Finally, annoyance ratings tended to be elevated in participants describing themselves as noise sensitive, as assessed using a 52-item questionnaire completed after they had judged the sounds. Observing such systematic relations between noise sensitivity, annoyance, task load during exposure, and psychophysiological responses in the laboratory might contribute to enhancing our understanding of noise annoyance.
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Subjective and Objective Assessments of Noise Barriers in Terms of the Loudness Level
Marcin Piotr Nowak, Piotr Kokowski
[Abstract]
The common measure of noise reduction caused by acoustic barrier is its insertion loss, IL. It is expressed in A-weighted decibels, ILA, as a difference between A-weighted time-averaged sound pressure levels, LA, calculated or measured without and with the presence of acoustic barrier. By definition, insertion loss ILA does not depend on the absolute value of the A-weighted sound pressure level. Thus, for example in case of road traffic noise, the ILA is insensitive to the change of factors influencing sound level, like vehicle speed, fleet composition, distance to the road and so on. On the other hand, it is widely known that for some environmental noises LA poorly correlates with subjective noise assessment. A better correlation can be observed for more sophisticated noise indexes, like Zwicker loudness, expressed in sones. The insertion loss in sones, ILN, is to be determined as the ratio between the loudness without (N) and with (Nb) the barrier. The results of both, numerical calculations as well as psychoacoustic experiments, will be presented to show the relation between ILA and ILN and their correlation with the subjective assessments of barrier insertion loss as a function of different traffic noise levels.
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Improving defect detection in wind turbine blades with psychoacoustic means and prediction models
Bernhard U. Seeber, Gaetano Andreisek
[Abstract]
Wind power plants, and particularly their blades, must withstand significant environmental stresses. Regular inspections of the blades’ structural integrity are essential to ensure a projected lifetime of fifteen to twenty years. Besides a visual inspection, experienced engineers tap on the blades surface with a suitable tool to detect invisible defects inside the blade’s cross-section. By listening to the emitted tap sounds, the human inspector can assess the material integrity. This work aims at identifying acoustic features that are key to building an acoustic analysis algorithm of tap test recordings. The process involved two approaches: 1) Listening tests with ten engineers familiar with the inspection of blades, in which audible differences between tap test recordings from intact and defective material were rated using a set of defined adjectives (semantic differential rating). 2) Correlation of acoustic features, which were extracted with our Matlab toolbox, with a detailed assessment of defects in the vicinity of the respective tap points. As a result, an informed (based on the inspectors’ knowledge), multi-layered regression algorithm could be built, which takes single-valued acoustic features as inputs and successfully rates the material integrity and damage size around a tap point.
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Activity Disturbances by a Step Change in Aircraft Noise Exposure around Hanoi Noi Bai International Airport
Takashi Yano, Makoto Morinaga, Shigenori Yokoshima, Thulan Nguyen, Thao Linh Nguyen
[Abstract]
A new terminal building was operated at Hanoi Noi Bai International Airport (HNBIA) in December 2014 and the number of flights increased 1.2-1.3 times. In order to investigate community response to the step change in aircraft noise exposure, socio-acoustic surveys were conducted at 13 sites around HNBIA once before and twice after the operation. The sample sizes in the 1st, 2nd, and 3rd surveys were 891, 1121, and 1287, respectively. Since the usage of the two runways changed before and after the operation, the noise exposures were not monotonously increased from 1st to 2nd or 3rd surveys. Thus the difference in Lday and Lnight between before and after the operation (ΔLday and ΔLnight) ranged from -9 to 5 dB and from -2 to 8 dB, respectively. In this paper activity disturbances such as listening, resting, and sleep disturbances by the step change are investigated. As ΔLday increases, the exposure-response relationships for listening and rest disturbances become higher than that for the 1st survey even if ΔLday was minus. As ΔLnight increases, the exposure-response relationships for sleep disturbance becomes higher than that for the 1st survey when ΔLnight was plus but lower when ΔLnight was minus.
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Improvement of Copy Machine Noise
Takeo Hashimoto, Shigeko Hatano
[Abstract]
We use copy machine in our daily life for the hard copy of our manuscript which is useally coneted to PC. The copy machine is useally located close the user, it is important to improve the noise emitted from the machine for good sound atmophere. When we think of the noise from the copy mmachien there are several causes that create unwanted noise from the machine. Here, we would like to show the result of the improvement of noise emitted from the copy machine by treating the time envelope of the noise for better impression.
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Applications of Psychoacoustics in Dental Drill Noise Evaluation
Tomomi Yamada, Sonoko Kuwano, Mikako Hayashi
[Abstract]
The sound produced by dental drills such as a dental air turbine handpiece or a dental ultrasonic scaler can markedly influence the sound environment in a dental clinic. Indeed, many patients report that the sound of a dental drill elicits unpleasant feeling. Comfortable sound environment for patients is pursued in a field of medical care as well as in daily life. The sounds emitted by dental drills at dental clinics has high frequency components up to 20,000 Hz with high sound pressure level. In this study, the effect of sound level or frequency components in high-frequency region on the subjective impression of dental noise was systematically investigated in order to find clues to create comfortable environment in dental clinics. We will report the findings.
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Annoyance of Noise in the Infrasound Range; Study Design and Acoustic Presentation
Detlef Krahé, Sarah Leona Benz, Christian Eulitz, Stephan Grossarth, Ulrich Möhler, Uwe Müller, Dirk Schreckenberg
[Abstract]
An increasing number of infrasound sources and the success in decreasing the acoustical emission and immission caused by noise sources at higher frequencies let the noise below 20 Hz come more in the focus. Many investigations concerning infrasound are done in the past, e.g. regarding the threshold of perception. However none of the results gives a sufficient answer, which can essentially explain the number of infrasound complaints and can give a reliable recommendation for a limitation of the exposure to infrasound. Therefore, in a special laboratory study on behalf of the Federal Environmental Agency an interdisciplinary team of experts tried to answer this question. The subjects should hear several noise scenarios in a living room atmosphere in a very silent house for a scenario period of 30 minutes. They should hear the sound in an accustomed way, that means by loudspeaker, where the driver has to have low distortions in order to be sure, the reaction is exclusively caused by infrasound. The actions and organization to create these conditions for the investigation are described in this paper. In the papers of Müller et. al. and Schreckenberg et al, the physiological and psychological effects of infrasound in this study are described.
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Organization:
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S. Van De Par, University of Oldenburg |
L.H. Carney, University of Rochester |
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Improved binaural speech intelligibility by adding reverberation to the target speaker
Julian Grosse, Steven Van De Par
[Abstract]
In anechoic, multi-source listening situations, the auditory system is remarkably capable of understanding speech. An improvement in speech intelligibility is observable when interfering and target speakers are spatially separated compared to being co-located.
The present study investigates the influence of reverberation on speech intelligibility of a reverberated target placed in front of the listener in the presence of symmetrically placed anechoic interferers. A simulated binaural room impulse response (BRIR) is systematically truncated at different durations to investigate the effect of reflections. Measurement conditions consisted of co-located and separated speakers that were presented, using BRIRs for the target speaker and HRTFs for the interferers.
It was found that reflections can have an opposite effect on speech intelligibility. Whereas for spatially separated speakers, added reverberant components impair speech intelligibility, for co-located conditions speech intelligibility improves. This suggests that the auditory system may utilize the spatial diffuseness of the target source in co-located conditions to perceptually segregate it from the interfering anechoic sources. In spatially separated conditions, the difference in perceived location of the sources may be the dominant cue for segregation such that diffuseness caused by adding reverberation does not further improve intelligibility.
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Modeling binaural detection of a Gaussian noise target in the presence of a lead/lag masker
Jonas Braasch, M. Torben Pastore, Yi Zhou
[Abstract]
Recently, we presented the results of a binaural masked
detection experiment in which a noise target was temporally
embedded within a lead/lag noise masker pair [J. Acoust. Soc.
Am. 141, 3639]. The results show that the inter-stimulus interval
(ISI) between the masker and its reflection changed the
detection threshold significantly. For low ISIs of 2-ms, the
average masked detection was -11 dB, but for a greater ISI of
20 ms, the threshold was much higher (-2 dB). In the experiment,
masked detection thresholds did not depend on whether the
masker lead was on the same side as the target (with the lag on
the contralateral side) or the other way round. Different model
approaches are presented to simulate the psychophysical
results, including a precedence effect (PE) model that was
extended using a cepstrum-based method to determine the
localization cues for the direct sound of the masker. The PE
model is used in conjunction with an EC model and a
binaural/monaural detector using optimal frequency-weighting
functions that were calculated during the above-threshold
conditions during the adaptive experiment. The modeling results
show that the PE model is not needed to explain the
psychoacoustical results. [Work supported by NSF BCS-1539276
and NSF BCS-1539376.]
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Reconsidering binaural phenomena in terms of interaural neural fluctuation differences
Laurel H. Carney
[Abstract]
Binaural phenomena have been interpreted in terms of a variety of acoustical parameters, including interaural differences of time, level, and envelope. Models based on these parameters often include physiological transformations, such as peripheral filters, possibly even nonlinear filters. However, transduction of the mechanical response in the cochlea into an electrical response is generally not included in binaural models, except for the bandwidth limitations imposed by the low-pass filtering of the OHC membrane filter. The influence of inner-hair-cell (IHC) saturation, and its interaction with basilar-membrane compression, have a qualitative effect on the signals that drive the inputs to the binaural nervous system. Saturation of the IHC voltage strongly shapes the responses to stimulus parameters that are conveyed to the central nervous system. IHC saturation influences the time course of auditory-nerve (AN) responses, and in particular affects the amplitude of low-frequency fluctuations in AN rate functions in response to complex sounds. Here, we re-examine some classical binaural phenomena in terms of interaural differences of neural fluctuation amplitudes of AN rate functions. We also examine response profiles across populations of model midbrain neurons that are sensitive to both neural fluctuation frequency and interaural differences. (Supported by NIDCD-R01-010813)
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Binaural masking level difference as a function of noise bandwidth and noise delay
Mathias Dietz, Kristin Bracklo, Stephan D. Ewert
[Abstract]
The human binaural system can exploit differences between the
interaural phase of a noise and a target tone to improve
detection thresholds. The maximum masking release is obtained
for detecting an antiphasic tone (Sπ) in diotic noise (N0). It has
been shown in several studies that this beneft gradually declines
as an interaural time difference (ITD) is applied to the N0Sπ
complex. This decline has been attributed to the reduced
interaural coherence of the noise. Here, we report detection
thresholds for a 500 Hz tone in masking noise with up to 8 ms ITD
and bandwidths from 25 to 1000 Hz. When reducing the noise
bandwidth from 100 to 50 and 25 Hz the masking release at 8 ms
ITD increases, in part because the narrower bandwidths result in
a higher coherence length. For bandwidths of 100 to 1000 Hz,
however, no significant difference was observed, indicating that
an auditory filter with a bandwidth <100 Hz is operational and
produces identical coherence for this group. Thus, our
coherence-based model requires an effective auditory filter
bandwidth <100 Hz, in line with established monaural models
but in contrast to delay line-based models for N0Sπ detection.
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Spatial unmasking of circular moving sound sources in the free field
Norbert Kolotzek, Bernhard U. Seeber
[Abstract]
This study investigates the contribution of interaural phase and level differences to the unmasking of a moving tonal source in noise under free-field presentation.
Thresholds of static and circular moving 500 Hz sine tones presented in octave-band noise from the front were measured with seven normal hearing listeners with an adaptive forced choice method. The stimuli were generated with 17th order Ambisonics and played via 36 horizontally arranged loudspeakers in an anechoic chamber. Six velocities were tested from 10∘/s to 200∘/s.
Spatial unmasking increased with increasing velocity up to 11 dB for 200∘/s but unmasking remained lower than for static sound sources at the most lateral position reached in the movement trajectory. This is consistent with observations in the literature on binaural sluggishness. Even for a small velocity of 30∘/s and therefore relatively slow changes in interaural phase, spatial unmasking decreases by 5.3 dB. This velocity is comparable to that of a walking person in 2 meters distance. It seems that in the free field, the auditory system cannot fully exploit interaural phase differences even when they are only changing slowly. This finding is important for extending binaural models to more realistic, dynamic listening situations in the free field.
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Understanding auditory motion perception: the role of temporal fine structure and envelope cues
Michaela Warnecke, Ruth Litovsky
[Abstract]
Auditory objects typically involve motion, either because the
sound source is in motion, or the listener is. Yet, most
psychoacoustic research has focused on investigating sound
localization for stationary sources. In normal hearing (NH)
listeners, research utilizing ”chimera” signals showed that
envelope cues are sufficient for robust speech perception,
while temporal fine-structure (TFS) cues govern binaural
sound lateralization. By contrast, listeners who are deaf and
use bilateral cochlear implants (BiCI) do not have access to
TFS cues through clinical processors. Recent work has
demonstrated poor auditory motion perception in BiCI
listeners compared to NH listeners. In this study, we
investigate the impact of TFS on auditory motion perception
and speech understanding by simulating both stationary and
moving sounds consisting of chimera-style speech and noise
stimuli in virtual auditory space. Perceptual measures
include: identifying whether a sound is stationary or moving,
discriminating the direction of motion and explicitly reporting
the absolute range of motion. The results of this study
provide a baseline for measurements to be conducted with
BiCI listeners. Knowledge of the cues that are lost in BiCI
users have the potential to provide insight for improved
engineering approaches for BiCIs.
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Hearing protections: effects on HRTFs and localization accuracy
Lorenz Kroener, Alexandre Garcia, Véronique Zimpfer, Christophe Langrenne
[Abstract]
With a healthy natural hearing sound source localization is a simple task, which is performed by the brain with high accuracy. The individual Head Related Transfer Function (HRTF) filters the incoming sound depending on the angle of incidence. The HRTF is defined by the individual geometry of the listener’s anatomy (torso, head, and pinna) and the listener’s brain is well trained to his HRTF. Protecting the auditory system from dangerous sound pressure levels, by wearing hearing protections or wearing Tactical Communication And Protection Systems (TCAPS) modifies the geometry of the pinna. The listener cannot adapt instantaneously to the modified HRTF, leading to a less accurate sound source localization. Listening test showed that front-back confusions is the major error type. Three different models of TCAPS were analyzed to gain information about their influence on sound source localization capabilities and on the modifications in the listener’s HRTF. HRTF measurements on different dummy heads were done during four scenarios: dummy head without TCAPS, dummy head wearing two different in-ear TCAPS (earplugs) and one on-ear TCAPS (earmuffs). Under the same scenarios sound source localization tests were conducted with subjects. In order to improve the sound source localization, prototypes of new generation TCAPS are presented.
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Organization:
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A. Fiebig, TU Berlin |
T. Hashimoto, Seikii University |
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Cognitive load influences the evaluation of complex acoustical scenarios
Jochen Steffens, Franz Müller, Melanie Schulz, Samuel Gibson
[Abstract]
Sound evaluations depend on sound characteristics as well as on situational and person-related factors. Aim of this study was to investigate the effects of temporal dynamics of a sound, a person’s cognitive load, personality traits (noise sensitivity, extraversion) and their interactions on the overall retrospective evaluation of complex acoustical scenarios. In the course of a laboratory experiment, 62 participants were presented eight two-minute sound examples, whereby the position and valence of the peaks were varied. Task of an experimental group was to perform the so-called Stroop test of color-to-word interference during the sound presentation, and to retrospectively rate the sounds in terms of loudness and pleasantness. A control group only rated the sounds during and after listening to them. Results revealed that, on average, participants in the ”inattentive” experimental group rated the sounds as 6.6% less loud and 6.7% more pleasant compared to the participants of the ”attentive” control group. Peak position and valence also showed main effects but did not interact with the effect of the treatment. Moreover, personality traits did not have a significant main or moderating effect on sound evaluations. Findings highlight the important role of cognitive load and attention on the evaluation of complex acoustical scenarios.
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The influence of extreme response style on sound evaluations
Christoph Jakobs, Dustin Selbach, Sebastian Böldt, Jochen Steffens
[Abstract]
This paper is the first to examine the influence of
sociodemographic characteristics of listeners on the tendency
to extreme response style (ERS) in psychoacoustical
experiments. ERS can be described as the proneness to
exploit the edges of rating scales in quantitative surveys
leading to systematic deviations of the respondents’ reported
values from the true values.
In line with the psychological literature, we assumed that
ERS is more pronounced in women (compared to men) and
persons with low educational level (compared to higher
educated persons). Furthermore, we expected that ERS
increases with ascending age of people and increasing survey
duration. To test our hypotheses, we re-analyzised the data
obtained by Weinzierl et al. (2018). In their study, 190
subjects rated their acoustical impression of 35 binaurally
simulated rooms by means of the Room Acoustical Quality
Inventory (RAQI). Results of linear mixed-effect models
confirmed our assumption that women show a significantly
higher tendency of ERS than men and that increasing survey
duration leads to a higher ERS. In contrast, no significant
influences of education and age on ERS were observed. The
results highlight the role of ERS in sound evaluations and the
need to consider this moderating factor when conducting
psychoacoustical experiments.
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Feature analysis of spectral cues and loudness for perception of sound direction by people with unilateral hearing loss
Fumikazu Saze, Kan Okubo, Kazuhiro Iida
[Abstract]
People with unilateral hearing loss (UHL) commonly have
difficulty judging sound direction because of a lack of interaural
time difference (ITD) and interaural level difference (ILD),
although some earlier reports have described that some are
able to judge direction to some degree. In this study, assuming
UHL as loss of hearing on only one side, we aim to overcome
that difficulty through analysis of sound direction perception. We
conducted experiments of direction recognition for persons with
UHL using Gaussian white noise. A stimulus was delivered at 30-
70 dB SPL, with the sound level selected randomly. Experiment
results suggest that persons with UHL can perceive sounds from
a specific direction (hearing side and front side), although they
have some difficulty distinguishing deafness side stimuli from rear
side stimuli. Additionally, we discuss spectral cues and loudness
in the respective directions. This examination revealed the
possibility that the relation of spectral cues and loudness
influences the perception of sound direction for persons with
UHL.
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Organization:
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J. Becker, HEAD acoustics GmbH |
F. Völk, WindAcoustics UG |
P. Davies, Purdue University |
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Temporal weights in loudness judgments: A review
Daniel Oberfeld-Twistel, Alexander Fischenich, Jan Hots, Jesko Verhey
[Abstract]
During the past 10 years, a significant number of studies
have shown that, in contrast to a sound level meter and
widely used measures like the LAeq, human listeners do not
weight all temporal portions of a sound equally when they
evaluate its overall loudness. Instead, the psychoacoustic
data very consistently show a primacy effect: The beginning
of a sound is more important for the loudness judgment than
later temporal portions. This talk provides an overview of the
state of the art concerning temporal weights in loudness
(data and modeling approaches). We will discuss a) how
the temporal dynamics of the primacy effect can be
modeled and how it depends on the sound duration and
the duration of the temporal segments, b) how the temporal
weights are affected by the presence of a background
noise and by variations in sound level, c) how the primacy
effect recovers during silent gaps in the sound, and d)
whether the temporal weights are applied globally or on a
frequency-channel specific basis. The data provide a
challenge for current dynamic loudness models and
improved modeling strategies will be suggested.
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The loudness detection of the first and last pulse of a periodic sequence: role of temporal integration and masking.
Liudmila Rimskaya-Korsakova, Dmitriy Nechaev
[Abstract]
The study is attempt of searching the speech intelligibility
reduction causes. A pulse train containing 12 pulses with inter-
pulse intervals of 20 to 110 ms was considered a model of the
test and masking phonemes. The first or last pulse of the train
varied in level; they were considered tests to measure the
threshold difference between the test pulse alone and the
same pulse in the train (dIso) and the minimum inter-pulse
interval (Tmin) for the test pulse detection. The lowest Tmin and
dIso was found in normal-hearing listeners. Higher dIso were
obtained in the listeners with a prolonged temporal integration
of periodic sounds independently of hearing sensitivity.
Increased dIso values were observed at Tmin despite of the test
pulse position in the pulse train. The temporal integration and
masking determined the values of dIso and Tmin. Thus, within-
train pulses affected the detectability of the first and last pulses.
It is suggested that this effect is one of causes of changes in
phonemes loudness and the speech intelligibility reduction.
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Pitch Strength and Asymmetry in Global Loudness between Rising- and Falling-Intensity Sounds
Sabine Meunier, Jacques Chatron, Patrick Susini, Emmanuel Ponsot
[Abstract]
A sound is judged as overall louder if it increases rather than if it decreases in level over time; this is the so-called ”asymmetry in global loudness”. Several studies have shown that this effect is larger for pure tones than for broadband noises. Results of a previous study suggested that this difference is probably not related to the spectral width of the sounds per se, but by whether the sounds elicit a sensation of pitch or not. Based on the well-established finding that loudness and pitch interact with each other - e.g. a pure-tone that only increases/decreases in level is perceived as increasing/decreasing in pitch - we wondered whether this could explain why stronger asymmetries are found for tones vs. broadband noises. We hypothesized that the combination of loudness and pitch variations may reinforce the asymmetry in global loudness for sounds that elicit pitch. To test this hypothesis, we measured in this study the exact relationship between the size of the asymmetry in global loudness and sound’s pitch using sounds whose pitch strength relative to the noise percept can be varied by modifying parameters (iterated ripple noises, bandpass noises), thus asymmetry in global loudness and pitch strength can be compared.
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Progress in Tonality Calculation
Julian Becker, Roland Sottek, Thiago Lobato
[Abstract]
The perception of tonal sounds is one of the most important psychoacoustic sensations for product sound quality and environmental noise. In the past, a lot of work has been carried out to automatically quantify tonal sound events. This work resulted in several methods for quantifying tonalities, such as Tone-to-Noise Ratio, Prominence Ratio or the Psychoacoustic Tonality, which was standardized in the 15th Edition of the ECMA-74 standard. The Psychoacoustic Tonality is based on a hearing model which emulates the processing of human hearing. Thus, it is able to predict human perception better than other methods which are only partly based on psychoacoustics.
Since the publication of the Psychoacoustic Tonality, the algorithm is constantly being improved. In this paper, the latest progress in the calculation of the Psychoacoustic Tonality is presented. The new developments include improvements for low frequencies, identification of tonal components and a higher frequency resolution of the detected tonalities.
These improvements will likely be included in future editions of the ECMA-74 standard.
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Spectral prominence influencing the perceived strength of psychoacoustic measures
Arne Oetjen, Steven Van De Par
[Abstract]
The perceived strength of many psychoacoustic measures such as roughness, impulsiveness or beating strength all depend to some degree on the overall sound pressure level. In previous studies this effect was mainly observed using synthetic sounds. In environmental sounds these psychoacoustic phenomena often are only present within a certain limited frequency band. For such a frequency band, only a very small change in sound pressure level in relation to the neighboring bands can result in a very large change in the perceived strength of the psychoacoustic measure and thus the dependence of the psychoacoustic measure on the sound pressure level is much stronger than in isolated broad- or narrowband conditions. This is shown in listening tests where for example modulations were only applied to a limited band within a broadband noise. The level of only this band was then altered in small steps. The resulting changes in the spectral prominence of the modulated band caused a large change in the perceived strength for some psychoacoustic measures. This effect is relevant for environmental sounds such as vehicle interior noise which can exhibit different degrees of spectral prominence. Different interpretations of these data will be discussed.
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Expression of the Feelings to Noise using Cepstral Parameters
Masanori Akita, Wataru Saito, Satoshi Ito, Yoichi Midorikawa
[Abstract]
The influence that noise environment gives in human feelings
becomes one important topics of the environmental problem.
This paper shows the unpleasant feelings of human when
hearing different type of noises. Previous our research showed
that the unpleasant feelings are changed even if the sound
pressure level hearing noise is same. In this paper, we define the
feature parameters using spectral envelopes of noisy signal
using cepstral coefficients. Relation between the feeling test of
examinees and the feature parameters are compared. The
results show that the spectral level on high frequency domain
affect the feelings. Next, examination between the feelings and
the Lombard effect are tested. The amount of the Lombard
effect are defined using the center of the gravity of the
pentagon by the first and the second formant frequency of
Japanese 5 vowels or the average of the moving distance of
the selected vowels. Examinations show some relationship
between amount of the effect and feelings.
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Modelling Monaural and Binaural Audio Quality
Stephan D. Ewert, Jan-Hendrik Fleßner, Thomas Biberger
[Abstract]
In today’s entertainment, communication, and hearing rehabilitative applications numerous signal degradations, ranging from linear to nonlinear (monaural) to spatial (binaural) distortions, can occur affecting the perceived audio quality. For many applications tools for instrumental quality prediction are desirable. To cope with the large variety of different signal distortions, recently the monaural generalized power-spectrum model for quality [GPSMq, Biberger et al., JAES, 66, 578 (2018)] and the binaural auditory-model-based quality prediction [BAM-Q, Flessner et al., JAES, 65, 929 (2017)] have been suggested. Here the underlying auditory perception principles utilized in the models are outlined and the two models are combined investigating how monaural and binaural quality aspects contribute to overall audio quality. In a listening experiment, the effect of different monaural and binaural distortions on music, speech, and noise, applied either in isolation or in combination, was examined. The resulting database was used for assessing different methods for combining GPSMq and BAM-Q to joint overall audio predictions for monaural and binaural signal distortions. The results indicate that overall audio quality depends on the lower quality aspect, either monaural or binaural. Neither monaural or binaural quality aspects dominated overall quality per se.
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Organization:
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P. Davies, Purdue University |
R. Sottek, HEAD acoustics GmbH |
K. Yamauchi, Kyushu University, Faculty of Design |
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Simulated Transfer Path Accuracy vs. Sound Perception
Giorgio Pulvirenti, Nicolas Totaro, Etienne Parizet
[Abstract]
The use of numerical methods is becoming more and more common in industry design processes. However, to finely represent the physics of the problem, numerical models are always more complex and time consuming. In the meantime, sound perception is now a very important topic for car manufacturers, who try to predict sound quality during the development phase of the vehicle. It is therefore useful to get some ideas about how accurate a simulation should be in order to be used in sound quality applications. The goal of this paper is to understand the minimum accuracy of simulated transfer paths (TP) required to give meaningful psychoacoustic results. A physical model is build and the associated Finite Element (FE) model is set up, validated and taken as reference. Then, some deteriorations of the FE model are introduced and a sensitivity analysis checks how those changes affect psychoacoustic characteristics (sharpness, loudness) of the TPs. Laslty, hearing tests are performed in order to understand how people percieve the differencies in the various simulated sounds.
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Modulation masking patterns for complex-shaped envelopes
Arne Oetjen, Julian Schiller, Steven Van De Par
[Abstract]
The concept of modulation filters is widely applied in in models of auditory perception. This concept was derived from modulation masking experiments using a narrowband masker and a sinusoidal target modulation. Using other periodic waveforms as a target modulation implies that ”overtones” are added to the fundamental modulation frequency. This target modulation containing overtones will result in changes across the output of multiple modulation filters. In order to investigate this phenomenon listening tests measuring modulation masking for different target modulation waveforms, varying in both the modulation magnitude and phase distribution of the overtones, were carried out. These data can not be reproduced by a model which is simply adding up the output from multiple modulation filters or just considering the modulation filter with the highest output. Instead, a new model approach using additional onset information for processing the output from multiple modulation filters is developed. This approach can reproduce the experimental data qualitatively and quantitively.
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Audibility of Spectral Dips and Peaks in Broadband Noise
Léopold Kritly, Vojtech Chmelík, Yannick Sluyts, Christ Glorieux, Monika Rychtarikova
[Abstract]
The paper reports on the ability of people to discriminate between sounds with slightly different frequency spectra. This is relevant in architectural acoustics, for people with a visual handicap who are navigating space by means of echolocation and acoustic recognition of spaces. This is done in room and building acoustics contexts respectively by interpreting just noticeable differences in colour of sound reflection from interior surfaces; and by judging sound transmission spectra, such as the relevance of subtle sound insulation deteriorations of improvements of walls.
Stimuli were created by damping or amplifying a frequency band centred around 125Hz or 2kHz of broadband pink noise with varying bandwidth and reduction/enhancement factor. Perception tests involving 12 human subjects confirm expectations. On average spectral dips are more difficult to detect than peaks. No statistical significant difference was found in the performance of subjects at different presentation levels (Lp,A=35 or 50 dB).
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Effects of Length of Carrier Phrase on Release from Masking in Multi-talker voice guidance
Hayato Sato, Masayuki Morimoto, Kazuhiro Iida, Hiroshi Sato
[Abstract]
Multilingualization of voice guidance is necessary for
international passenger facilities and world famous tourist
sites. Where the voice guidance is provided by public address
systems, the same content is repeatedly broadcasted in
several languages in usual. The authors are advancing a
challenging project to shorten the required time by
broadcasting all of the guidances at once. In this paper, as a
fundamental study, a listening test on word intelligibility of
multi-talker broadcasting was performed to clarify effects of
length of carrier phrase. The target stimulus was a
simultaneous reproduction of two different Japanese words
spoken by two different speakers. The task of the listener
was to answer both of the two words. As a carrier phrase,
other Japanese words spoken by the two speakers was
sequentially reproduced. The carrier phrase and the target
stimulus were presented from a loudspeaker installed in front
of the listener. The parameters of the test were length of the
carrier phrase and combination of gender of the speakers. As
a result, effects of the carrier phrase were found only when
the gender of the speakers were the same, and word
intelligibility scores increased as the carrier phrase became
longer.
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Influence of Full-Body Vibration Adapted to Foreground Components on High-Level Perception of Reality
Shota Abe, Zhenglie Cui, Shuichi Sakamoto, Yôiti Suzuki, Jiro Gyoba
[Abstract]
To develop advanced multimedia communications systems,
understanding how humans perceive reality from the media
presented by the systems is important. Various indexes can be
used to evaluate the sense of reality. Whereas the sense of
presence is an evaluation index that includes the background
information, the sense of verisimilitude focuses on the
foreground information. By adding appropriate sensory
information related to the foreground or background
component, these perceptual realities could be enhanced.
Based on this concept, we investigated the effect of vibration
information added to the audio-visual content. In this study,
vibration information was generated from the sound that
included rich information about the contents. We generated
nine types of vibration by adjusting the cutoff frequency and
the carrier frequency of the sound. The results showed that
higher verisimilitude was observed when vibration closely
connected to foreground components was added in a scene.
Moreover, under that condition, the sense of presence was
hardly affected even when the vibration was added to the
content. These results suggest that generating more realism is
possible by enhancing vibrations if the processing is performed
to match the foreground components in the scene.
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Organization:
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B.F.G. Katz, Sorbonne Université, CNRS, Institut d'Alembert |
M. Otani, Kyoto University |
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Environmental Audio Scene Description and Rendering Model for Virtual or Augmented Reality
Jean-Marc Jot, Rémi Audfray, Sam Dicker
[Abstract]
New virtual and augmented reality audio experiences require rendering multiple digital objects and supporting navigation though complex scenes for a listener using earphones or a head-mounted audiovisual display device. Practical audio scene rendering and creation tools developed for these applications are evolved from game audio systems and address the effects of acoustic reverberation, reflectors and obstacles in the virtual and/or physical environments, using combinations of pre-computed and real-time propagation models. It is desirable to separate the characterizations of the environment, the listener, and the
individual sound sources, in a manner that enables minimizing the per-source propagation and rendering computations to be executed in real time according to source or listener positions, orientations and movements. In this paper, we review the OpenAL EFX scene description and rendering model and demonstrate how it can be extended to meet these objectives by exploiting statistical properties of diffuse
reverberation decays in enclosures. In augmented reality applications, for instance, this enables the realization of a computationally efficient real-time binaural rendering engine that can blend multiple virtual audio objects within the acoustics of a user’s local environment, detected at rendering time.
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Attempt to Improve the Total Performance of Sound Field Reproduction System: Integration of Wave-Based Methods and Simple Reproduction Method
Hiroshi Kashiwazaki, Akira Omoto
[Abstract]
The sound field reproduction system can be applied to various
applications. The primary objective of the system is to physically
reproduce an arbitrary sound field correctly. However, physical
accuracy is not the only requirement in a situation where the system
is actually used. Through several applications, we assumed the
following four factors as total performance; A) physical accuracy, B)
robustness against disturbances, C) flexibility for additional direction,
and D) capability of integration with visual media. We used 24-
channel narrow directional microphone array and 24-channel
loudspeaker as a platform to be considered and worked to improve
overall performance. As one of the reproduction methods, directional
information can be easily reproduced by amplitude panning which
relies on the directivity of the microphone themselves, but the
microphone leakage effect occurs in the low frequency. On the other
hand, wave-based methods such as boundary surface control or
higher-order ambisonics are effective for physical accuracy, but
control of high frequency is difficult. Therefore, if wave-based
methods and simple reproduction method are combined, it is possible
to reproduce a wide frequency range well. We discuss the total
performance of this reproduction method through ITD/ILD
measurement and visualization of a wide range of wavefront.
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Design and simulation of a benchmark room for room acoustic auralizations
Fotis Georgiou, Baltazar Briere De La Hosseraye, Maarten Hornikx, Philip W. Robinson
[Abstract]
In order to achieve accurate acoustic simulations of a room to
obtain an authentic auralization, the following aspects need to
be quantified: the geometrical details of the room, all material
properties and the characteristics of the source and the
receiver. This paper presents the design of a benchmark room
for this purpose, including all this information. The room is an
empty building acoustics transmission chamber with thick
concrete walls. This room was then acoustically treated to
achieve an acoustic environment close to a day-to-day office
room. The surface impedances of the materials additionally
installed in the room were measured with both in the
impedance tube as well as with a pressure-velocity sensor.
Furthermore, the directivity of the measurement source and the
binaural receiver were measured in order to be included in the
simulations. Impulse responses of this benchmark room have also
been obtained from simulations with the in-house discontinuous-
Galerkin method (DG) with time-domain frequency-dependent
boundary conditions, as well as source and receiver directivity.
Time and frequency domain results from the both the
measurements and simulations are presented, showing a close
agreement.
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Individualized dynamic binaural Auralization of Classroom Acoustics using a Virtual Artificial Head
Mina Fallahi, Martin Hansen, Simon Doclo, Steven Van De Par, Dirk Püschel, Matthias Blau
[Abstract]
In order to account for individual acoustic properties as well as room-related information in measurement-based room acoustic auralizations, individual Binaural Room Impulse Responses (BRIRs) are needed. However, in a dynamic scenario with head movements, measuring individual BRIRs can be very time consuming. In this study, a planar microphone array with 24 microphones, referred to as Virtual Artificial Head (VAH), was used to measure Room Impulse Responses (RIRs) for a single receiver position with respect to different source positions in a lecture room. In order to synthesize individual BRIRs for different head orientations, VAH filter coefficients for different head orientations were calculated and applied to the measured RIRs. To this end, individual Head Related Transfer Functions (HRTFs), measured in an anechoic condition, were required. The VAH filter coefficients were calculated by solving a constrained optimization problem, i.e. by minimizing a narrow-band least-squares cost function subject to constrains on spectral magnitude error and the mean White Noise Gain. The quality of the synthesized individual BRIRs was evaluated based on subjective listening tests in a dynamic binaural scenario.
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Anechoic audio and 3D-video content database of small ensemble performances for virtual concerts
David Thery, Brian F. G. Katz
[Abstract]
This paper presents a public database of anechoic audio and
3D-video recordings of several small music ensemble
performances. Musical extracts range from baroque to jazz
music. This work aims at extending the already available
database of anechoic stimuli, providing the community with
flexible audio-visual content for virtual acoustic simulations. For
each piece of music, musicians were first close-mic recorded
together to provide an audio performance reference. This
recording was followed by individual instrument retake
recordings, while listening to the reference recording, to
achieve the best audio quality. In parallel, 3D-video content
was recorded of each musician, employing three Kinect 2 RGB-
Depth sensors, allowing for the generation and easy
manipulation of 3D point-clouds. The position/orientation of
moving instruments extracted from these videos allows for the
incorporate dynamic directivity in the auralizations. This article
details the choice of the musical pieces, the recording
procedure, and technical details on the system architecture
including post-processing treatments to render the stimuli in
immersive audio-visual environments.
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Effects of the order of Ambisonics on localization for different reverberant conditions in a novel 3D acoustic virtual reality system
Hermes Sampedro Llopis, Finnur Pind, Cheol-Ho Jeong
[Abstract]
This study presents an acoustic virtual reality technique based on
geometrical acoustics simulation and high-order Ambisonics to
improve the performance of sound source localization and the realism
in enclosed 3D virtual spaces. This tool is based on pre-calculated B-
Format impulse responses at a grid of receiver positions that is
decoded into a virtual array of loudspeakers fixed at the listener’s
head. The virtual loudspeaker signals are synthesized for a binaural
representation via head-related transfer functions. This tool is
assessed in terms of sound localization. The 1st and 2nd order
Ambisonics are compared by varying different visual and head
movement conditions to see if they affect the localization
performance. Moreover, this study tests the localization in full 3D
challenging reverberation time conditions (0,6s up to 2s). The
perceived realism, sound source width, and sound continuity are
tested under different grid resolutions. Results show that significant
outperformance of the 2nd order technique over the 1st order is
found in all head movement and visual conditions. It has also been
discovered that when using a coarse grid, the perceptual sound
source size is increased and spread out. Moreover, realism and sound
continuity are not affected significantly by the resolution of the grid.
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Binaural reproduction capability for multiple off-axis listeners based on the 3-channel optimal source distribution principle
Motoki Yairi, Takashi Takeuchi, Keith Holland, Dylan Morgan, Laurence Haines
[Abstract]
This paper presents a numerical and experimental studies for synthesizing virtual auditory space to multiple listeners using binaural synthesis over loudspeakers. The binaural principle is based on the optimal source distribution (OPSODIS), which utilizes the idea of a pair of conceptual monopole transducers (left and right) whose azimuthal location varies continuously as a function of frequency. The authors have revealed so far that this makes it possible to provide the same binaural signals not only to the on-axis target listener but also to multiple off-axis listeners. The OPSODIS can be augmented to 3 channels by adding a center channel to the left/right configuration described above (3-channel OPSODIS). The 3-channel OPSODIS is potentially superior to the conventional one as for the on-axis target listener’s sweet spot. In this paper, the authors discuss whether the sweet spot of off-axis listeners can also be enhanced by the 3-channel OPSODIS. Numerical simulations under free-field conditions and experiments in an anechoic chamber are carried out. It is proved that the 3-channel OPSODIS can also provide the additional controlled regions for the off-axis listeners, which are more robustness than those given by the conventional one.
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Organization:
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S. Bilbao, Acoustics and Audio Group, University of Edinburgh |
B. Hamilton, Acoustics & Audio Group, University of Edinburgh |
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A locally implicit time-domain FEM for room acoustics simulation including permeable membrane absorbers
Takumi Yoshida, Takeshi Okuzono, Kimihiro Sakagami
[Abstract]
First-order ordinary differential equations (ODEs) based
time-domain FEM (TD-FEM) is an attractive time-domain
solver for room acoustics simulation. For an idealized case
the TD-FEM has fourth-order accuracy in both space and
time with explicit algorithm. This paper presents a sound
absorber modeling in the first-order ODEs based TD-FEM,
addressing permeable membrane (PM) absorbers, which
have been used to create comfortable acoustic
environments in buildings such as conference rooms,
stadiums and swimming pools. However, simple
implementation of numerical PM absorber model to the
TD-FEM engenders fully implicit algorithm. To overcome
the difficulty, an iterative solver for sparse linear systems is
locally applied to the TD-FEM. As a consequence, locally
implicit first-order ODEs based TD-FEM for sound field
analyses including PM absorbers is presented. Numerical
experiments including acoustics simulation in a real sized
room showed that the presented locally implicit TD-FEM
performs better than the fully implicit implementation
without the reduction of accuracy.
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Modelling Boundary Conditions in High-Order, Nodal, Time-Domain Finite Element Methods
Finnur Pind, Cheol-Ho Jeong, Jan S. Hesthaven, Allan P. Engsig-Karup, Jakob Strømann-Andersen
[Abstract]
Accurate modeling of boundary conditions is an important aspect
in room acoustic simulations. It has been shown that the
acoustics of rooms is not only dependent on the frequency
characteristic of the complex boundary impedance, but also on the
angle dependent properties of the impedance (“extended
reaction”). This paper presents a computationally efficient
method for modeling local-reaction (LR) and extended-reaction
(ER) boundary conditions in high-order, nodal, time-domain finite
element methods, such as the spectral element method (SEM) or the
discontinuous Galerkin finite element method (DGFEM). The
frequency and angle dependent boundary impedance is mapped to a
multipole model and formulated in differential form. The solution
of the boundary differential equations comes with minimal
computational cost. In the ER model, wave splitting is applied at
the boundary to separate the incident and reflected parts of the
sound field. The directional properties of the incident sound
field are determined from the incident particle velocity and the
boundary conditions are adjusted continuously according to the
wave angle of incidence. The accuracy of the boundary condition
model is assessed by comparing simulations against measurements,
where a significantly improved match between simulations and
measurements is found when the ER model is used.
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Broadband Time-domain Impedance Boundary Modeling with the Discontinuous Galerkin Method for Room Acoustics Simulations
Huiqing Wang, Maarten Hornikx
[Abstract]
The time-domain nodal discontinuous Galerkin (TD-DG)
method is emerging as a potential wave-based method
for three-dimensional (3D) room acoustics modeling,
where high-order accuracy, geometrical flexibility and
accurate modeling of boundary conditions are of critical
importance. In this work, an accurate and efficient
formulation of broadband time-domain impedance
boundary conditions of locally-reacting surfaces is
proposed in the framework of the TD-DG method. The
formulation of the time-domain boundary condition is
based on the plane-wave reflection coefficient at normal
incidence and its approximation in the frequency domain
using a sum of rational fractions, which can be obtained
from analytical models or measured impedance values.
To verify the performance of the formulation, a single
reflection scenario is considered and the reflection
coefficient obtained from the numerical tests is compared
with the analytical one based on a locally reacting
surface impedance. Also, the effects of the coefficients of
rational functions on the accuracy are investigated. The
broadband time-domain impedance boundary
formulation accurately predicts both the amplitude and
the phase of the reflection behaviour.
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Organization:
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P. D'Antonio, RPG Acoustical Systems, LLC |
T. Cox, University of Salford |
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Design of a New Testing Chamber to Determine the Absorption, Diffusion and Scattering Coefficients
Peter D'Antonio, Mélanie Nolan, Efren Fernandez-Grande, Cheol-Ho Jeong
[Abstract]
This presentation will describe a design for a new 286 m3
testing chamber that will allow the measurement of the
random incidence absorption coefficient, according to ISO
354, the scattering coefficient, according to ISO 17497-1,
the diffusion coefficient, according to ISO 17497-2 and the
normal incidence low frequency absorption coefficient,
according to ISO 10534-2. Optional dimensional ratios were
determined by minimizing the standard deviation of the
modal response. Both dome-shaped boundary diffusors
and hanging panels are used to provide satisfactory
diffusivity, which is verified using a 200 mm reference
absorber according to ISO 354. The isotropy of the incident
sound is measured with a new Sound Field Analysis
Recorder (SOFAR) and quantified by an analysis of the
wavenumber spectrum. A sufficiently large reflection free
volume is provided in the chamber for boundary method
diffusion coefficient scale model goniometer
measurements. A new FDTD simulation program will be
described to determine the diffusion and scattering
coefficients of any 3D surface in the near and far field. A
steel and concrete lined low frequency impedance tube
measuring 600 mm x 600 mm x 5.8 m long and 7 tons will be
used to measure absorption from 25-250 Hz.
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3D Printed Quadratic Residue Metadiffuser – Design and Measurements of an Optimized Deep-Subwavelength Sound Diffuser
Eric Ballestero, Vicente Romero García, Noé Jiménez, Jean Philippe Groby, Stephen Dance, Haydar Aygun
[Abstract]
Heavily based on slow sound propagation produced by
the strong dispersion in waveguides loaded by side
branch resonators, metadiffusers take the form of slotted
panels with thin slits, each one loaded by a set of
Helmholtz resonators. This type of structure was previously
used as deep-subwavelength perfect absorber. Instead
of focusing on absorptive and transmitting properties,
metadiffusers emphasize on perfect or pseudo-perfect
reflection of sound. Here a particular interest is given to
the scattering patterns obtained by the customization of
metamaterial’s geometry. The deep-subwavelength
nature of the structure leads to dimensions 20 to 46 times
smaller than the design wavelength, i.e. about 1/20th to
1/10th of the thickness of traditional designs. This
presentation aims at introducing the physical aspects of
sound propagation in and out of the metadiffuser,
emphasizing on strong dispersion relations, on the
consequent deep-subwavelength features arising from
the slow sound propagation, and on the custom
scattered fields. This theoretical overview will then be
paired with numerical and experimental data of a 3D-
printed quadratic residue metadiffuser. Finally, potential
applications of such structures will be discussed in critical
industry environments.
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Beyond phase grating diffusers using locally-resonant metamaterials
Noé Jiménez, Trevor Cox, Jean Philippe Groby, Vicente Romero García
[Abstract]
We present sound diffusers based on locally resonant
metamaterials that are one order magnitude thinner than
the classical designs based on phase gratings. Using a set
of waveguides loaded by an array of optimized Helmholtz
resonators, we induce strong dispersive propagation inside
the material. Doing this, waves in the low-frequency
regime are almost frozen and, because of this, a deep-
subwavelength resonance is produced. In addition, this
design allows an accurate tuning of the thermoviscous
loses. In this way, the reflection coefficient can be spatially
tuned in both phase and magnitude, and we are able to
mimic the classical QRD, PRD or Binary/Ternary sound
diffusers. Beyond these designs, we also propose an
optimized panel showing efficient and broadband sound
diffusion ranging from 250 Hz to 2 kHz using only 3 cm
thickness, i.e., 46 times thinner than the wavelength at the
lowest working frequency.
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Perfect acoustic absorption in deep sub-wavelength structures for the ventilation problems with degenerate resonators
Vicente Romero García, Noé Jiménez, Olivier Richoux, Georgios Theocharis, Jean Philippe Groby, Vincent Pagneux
[Abstract]
In this work we report a mechanisms providing symmetric perfect acoustic absorption by sub-wavelength structures in the ventilation problem. While in the problem of reflection the mechanism consists in critically coupling a single resonance independently of its nature, the problem of transmission becomes more complicated and a degenerate resonator with symmetric and antisymmetric resonances should be designed with both resonances critically coupled. The system analyzed in this work is made by a panel with a periodic distribution of slits, the upper wall of which are loaded by Helmholtz Resonators. The propagation in the slit is highly dispersive due to the presence of the resonators, producing a slow sound propagation before the resonance, and down shifting it to low frequencies. By controlling the geometry of the resonator, the visco-thermal losses can be tuned in order to compensate the leakage of the system and fulfill the critical coupling condition so activating the perfect absorption of sound. In the case of transmission, a double slit is needed, one acting as symmetric and the other one as antisymmetric resonators.
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Pattern and Orientation of Diffusers in Rooms with an Absorbent Ceiling
Emma Arvidsson, Erling Nilsson, Delphine Bard-Hagberg, Ola Karlsson
[Abstract]
Lately have the demands on good room acoustics in public
ordinary rooms increased. Traditionally has only reverberation
time been considered in this type of rooms but complementary
parameters as sound strength and speech clarity are more
commonly used today. This trend naturally implies higher
demands on the acoustic treatment. Typical for public ordinary rooms is to use an absorbent ceiling.
One way to further improve the acoustic properties is to
complement the ceiling treatment with diffusers. To investigate
the effect of combining these two types of acoustic treatments
have measurements been carried out combing different setups
of diffusers in a reverberant room with an absorbent ceiling. The
impact on reverberation time, speech clarity and sound
strength have been studied. In this paper we present how diffusers can be used in order to
improve the acoustic properties in a public ordinary room with
an absorbent ceiling and how the pattern and orientation of
diffusers can be used to adjust different acoustic parameters.
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Advances in diffusive surface design using 3D architectural parametric modeling programs
Louena Shtrepi
[Abstract]
Diffusive surface need to be optimally designed for both acoustic performance and aesthetic values. These aspects are at the heart of the design workflow and should respond to the requirements of both architects and acousticians. Advances in parametric modelling through digital tools have led to the integration of performance investigations and architectural design process. Moreover, parametric modeling software are also useful tools for manufacturing. Indeed, they can reduce the production time as they can easily be integrated inside the manufacturing process.
Although, it is more than a decade that the standard ISO 17497:2004 has proposed the scattering and diffusion coefficients measurements, further work is needed in order to increase designers’ awareness on diffusive surface design through simple design rules and approaches.
This research goes through three steps. First, it investigates the diffusive surface properties databases in order to extrapolate basic geometric guidelines for diffusive surface optimization. Second, it analyses a series of case-studies that apply the parametric modeling to the design of diffusive surfaces. Finally, it suggests a design process for diffusive surfaces by integrating parametric models and acoustic simulation aiming to provide architects and designers with rapid visual and acoustic feedback at a preliminary stage of their design.
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Application of multi-objective optimization techniques to the design of sound diffusers
Javier Redondo, Juan M. Herrero, Luís Godinho, Ricardo Patraquim, Trevor Cox
[Abstract]
Multi-objective optimization (MO) techniques have proven to
be a very interesting methodology for solving design problems in
different areas of engineering. Diffuser design problems are
another example. When designing a diffuser, different features
are to be optimized simultaneously, which are usually in
opposition. It is precisely the latter that encourages the use of
MO techniques. The designer chooses the particular features to
be optimized independently, carries out the optimization,
studies the resulting Pareto fronts, and finally chooses the
optimal diffuser using his preferences a posteriori. This results in
more informed and therefore more conscientious decision
making. This work shows how the MO techniques can be
applied in the design of diffusers through a series of examples
where different characteristics are taken into account: the
maximization of the diffusion coefficient at a particular
frequency range or the minimization of the depth of the diffuser.
In the last case, we will explore the possibility of changing not
only the depth of the wells but as well their widths. The study of
the Pareto fronts presented shows how the optimal diffuser to
choose can be susceptible to be different depending on the
characteristics most wanted by the designer.
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Room acoustic texture: a methodology for its quantification
Alejandro Bidondo, Leonardo Pepino
[Abstract]
Room acoustic texture is defined by Beranek as the subjective impression
listeners derive from the temporal and amplitude patterns of early
reflections, at the receiver’s locations inside the room. Traditionally, room
acoustic texture was qualified by visual inspection of the room impulse
responses (RIR) or counting reflection’s peaks. Taking into account that
the reflections in the later part of the RIR follow a Gaussian probability
distribution and are totally mixed, we define the early reflections as every
amplitude outlier present in a RIR, and mixing time as the instant when
their cumulative energy reaches 99% of their total energy. For the
amplitude outliers estimation, a moving median filter was applied to the
energy time curve and subtracting the result from the full room impulse
response. With the outliers impulse response identified, the particular
echo density function is defined as its cumulative energy in time. From
this processing, a group of descriptors were defined that jointly describe
the room acoustic texture, at one point in the sound field. Among these
descriptors are the mixing time, expected texture and distance between
models. Applications of these descriptors and their spatial standard
deviation in rooms seem to be very broad, describing their temporal
fine-structure.
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Anisotropic sound fields in reverberation-room measurements of sound absorption coefficients: Wavenumber spectrum theory
Mélanie Nolan
[Abstract]
Measured values of absorption coefficients obtained from
standardized reverberation-chamber measurements often
differ across laboratories, mainly due to non-isotropic sound
incidence on the absorbing specimen, and differences in the
chambers’ shapes and dimensions. The present study
describes an experimental method for characterizing sound
field isotropy in reverberation rooms. The methodology relies
on a plane wave decomposition (i.e., estimation of the
wavenumber spectrum) to describe the directional properties
of the sound field in the vicinity of the absorbing specimen.
By separating the incident from the reflected components in
the wavenumber spectrum, it is possible to characterize the
distribution of sound incidence on the sample, and to infer
angle-dependent absorption properties. Measurements are
conducted in a reverberation room with absorbing material
on the floor using a programmable robotic arm to scan the
sound field. The results confirm that the distribution of sound
incidence on the sample is not isotropic in the steady state.
The measured angle-dependent absorption coefficients show
good agreement with theoretical predictions.
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Sound Absorption of Thin Resonators including a Winding Neck Extension in Surface Panel
Shinsuke Nakanishi
[Abstract]
Acoustic resonant absorber like a perforated panel or a Helmholtz resonator can be tuned at a low frequency by extending its neck or enlarging cavity volume. However, a total size of the resonators is often quite large when the neck or the cavity is simply extended for tuning at a low frequency. Previous researchers have studied Helmholtz resonator shortened in its size by subsided neck into back air cavity, and confirm that this resonator is tuned at low frequency without a deep cavity. The author has studied effects of a winding built-in neck extension to sound absorption of perforated panels and Helmholtz resonators, which show same effects as the subsided neck into back cavity. This study measures and discusses sound absorption of thin resonator tuned at a low frequency. Test pieces have various neck extensions built in a surface panel and 15 mm total thickness of resonators to propose as sound absorption tiles attaching on walls or partitions. Discussions in this paper focus effects of opening size of the resonator, path length, patterns or number of turns of the winding neck extension and cavity volume to the sound absorption of the thin resonator.
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Organization:
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F. Martellotta, DICAR-Politecnico di Bari |
S.G. Zühre, Bilkent University |
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Unveiling the acoustics of the Cathedral of Santiago de Compostela using 3D impulse responses
Francesco Martellotta, Angel Alvarez-Corbacho, Lidia Alvarez-Morales, Francesca Balestra, Federica Ciani, Miguel Galindo Del Pozo, Juan José Gomez Alfageme, Pedro Fernando Nogueira Lopez
[Abstract]
The Cathedral of Santiago de Compostela in Spain is the
arrival
point of hundreds of thousands of pilgrims that every year
get to
the city along the many paths of the ”Camino de
Santiago”.
Along with the oldest part of the city, the cathedral is listed
as
UNESCO World Heritage, and several restoration works
have
been done and still are under way. Its acoustics is the result
of
centuries of evolution since the original Romanesque
building
was started, with contributions from nearly every
architectural
style. The interior arrangement of the church is significantly
different from the majority of Spanish cathedrals, where
the
choir typically breaks the main nave creating many sub-
spaces.
Here the plan follows the Latin cross shape allowing the
pilgrims to look at the altar from nearly everywhere,
allowing
source-receiver distances up to 70m. An acoustic survey
that
involved different laboratories allowed the collection of
monaural, binaural, and B-format impulse responses in
different
source-receiver combinations. In the present paper, the
results
pertaining to B-format measurements are presented,
discussing
the spatial distribution of the acoustic parameters as a
function
of the directional pattern of the reflection and of the
energy
distribution among the different spaces of the church.
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Mapping the acoustics of Ripon cathedral
Lidia Alvarez-Morales, Mariana Lopez, Angel Alvarez-Corbacho, Pedro Bustamante
[Abstract]
Cathedrals are amongst the most significant buildings of
European cultural heritage and are of great significance for the
study of intangible heritage. This paper presents a detailed
acoustic study of Ripon Cathedral, an English medieval
cathedral whose history goes back to the 7th Century.
Experimental and simulation techniques have been applied in
order to develop a better understanding and preservation of
the acoustic environment of Ripon cathedral. This research
addresses the acoustical conflict inherent to this kind of spaces:
its extreme reverberation builds a unique environment for vocal
and instrumental music, specially organ music, but at the same
time those same characteristics are a challenge for the
intelligibility of the spoken word, which is crucial to the delivery
of sermons. Acoustic maps have been created based on the
diverse uses of the space, the history of the building, and its
architectural features. Such maps have been used to analyse
the acoustic characteristics throughout the cathedral’s interior
space. This work is part of the EC-funded Marie-Sklodowska-
Curie Fellowship ëCathedral Acoustics’.
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Acoustics of Portuguese Romanesque churches
Antonio Carvalho, Francisco Pereira
[Abstract]
This paper summarizes the results of a series of recent measurements done in some very typical Romanesque churches in the North of Portugal to acoustically characterize this type of building. In situ measurements were done regarding Reverberation Time and RASTI (Rapid Speech Transmission Index) in empty churches. The results are presented and compared with some architectural characteristics of the churches (area, volume, ceiling height, length, width, etc.). Statistical correlations and prediction equations are presented among architectural and acoustic parameters.
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The Difficult Compromise between Conservation and Modern Functions of Heritage Buildings; the Case of the Restoration of the “Abbaye des Anges” and its 110 Acoustic Pots
Jean-Christophe Valiere, Karin Le Tyrant, Michel Le Gentil, Bénédicte Bertholon
[Abstract]
The ”Abbaye des Anges” is a Franciscan Abbey founded in 1507 located in the far west of Britany. After the French revolution, it was sold by the French State and became a hotel. This Abbey Church was abandoned and the carpentry wood was given to the American army during the First World War in order to build hangars for planes. This Abbey has a strong symbolic meaning for the local population and the new owners decided in 2000 to restore it, transforming its church into a cultural center. Beyond this, it contains in its walls the largest ensemble of acoustic pots attested in France; and probably one of the most ones in Europe. Therefore, the National Heritage is particularly concerned by this restauration and has required the owners to respect strict specifications, especially concerning the respect of the pots device. The pots were restored or replaced according to their conservation and tested with using methods similar to the historical conditions. Eventually, solutions (walls and roofing treatments and sound system) were found in order to reduce the reverberation time and echoes taking into account National Heritage specifications, cost control, and configurations that should be as invisible and reversible as possible.
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Acoustics Timeline of Hagia Sophia and Süleymaniye Mosque in Istanbul
Zuhre Su Gul
[Abstract]
Hagia Sophia built in 6th century and Süleymaniye Mosque in
16th century are the two major monuments of Istanbul World
Heritage Site. Different than Süleymaniye Mosque, Hagia Sophia
originally built and functioned as a church and later as a
mosque while currently serving as a museum. Their historical
significance as of being cultural heritages and religious uses in
relation to acoustics have motivated many researchers to test
and discuss characteristics of their interior sound fields. A
comprehensive study is necessitated to assess the acoustical
conditions of both structures in relation to their architectural
features and interior finish materials and to examine the
acoustical changes occurred due to the main repairs
undergone their life time. One aim of this study is to review the
acoustical field test results held in different periods by
comparison of common room acoustics parameters as of
reverberation time and clarity. Another point is that, with their
huge scale and multi-domed upper structures both monuments
exhibit multi-slope sound energy decay formation. For that
reason, relevant acoustical predictors including decay rates
and decay times are discussed in light of different functional,
spiritual and acoustical needs expected from such monumental
sacred spaces.
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Acoustical and Architectural History of the Thomaskirche in Leipzig
Braxton Boren, Jack Anthony
[Abstract]
The Thomaskirche in Leipzig has a rich musical and cultural
history, marked not only by its association with the Lutheran
Reformation in the 16th century but especially by the tenure
of J.S. Bach as cantor from 1723-1750. The church’s interior
has undergone several renovations of differing magnitudes
since the earliest building constructed in the 12th century. In
the 20th century, acoustician Hope Bagenal and Bach
biographer C.S. Terry argued that the architectural
modifications made by the Lutheran Reformers in the 16th
century directly affected Bach’s compositional palette by
reducing the reverberation time and allowing more
polyphonic and temporal variation in the liturgical works
composed for the space. Here the history of the building and
its renovations are presented, as well as acoustical
measurements in the current church, which has been
renovated since Bagenal’s time to more accurately match the
interior as it existed in Bach’s time. Some comments are
provided Bagenal’s theory as well as the effect of the
material and geometric changes made by the Reformers on
the performance of Bach’s music during his lifetime.
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UNESCO Wooden Churches of Southern Lesser Poland
Pawel Malecki
[Abstract]
Wooden Churches of Southern Lesser Poland are one of the
fourteen objects in Poland inscribed under the UNESCO’s World
Heritage List. Their architecture remains as a topic for many
studies. On the other hand, its acoustics has not been given
enough attention so far. Spatial Impulse Response
measurements were made and acoustic parameters were
calculated in all 5 wooden churches that are entered to the
UNESCO list. Then, the churches came up for comparative
analysis with reference to other objects of that kind.
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A Virtual Acoustic Restitution of St John's Baptistery in Pisa
Anna Rovigatti, Elena Rossi, Giulia Fratoni, Dario D'Orazio, Massimo Garai
[Abstract]
Virtual acoustics is a useful tool for studying ancient environments now part of our cultural heritage. The acoustic restitution of an existing building, or even of a no more existing building, allows understanding its acoustic and architectural evolution. St. John Baptistery is located in Piazza dei Miracoli, which is a homogenous architectural ensemble including the Leaning Tower and the Cathedral. Due to the succession of different masters, the Baptistery was built in stylistic disagreement with the earliest design, resulting in the actual architecture that provides interesting acoustic peculiarities. This work presents the virtual acoustic restitution of a potential first design, based on previous studies. The accurate calibration of the existing Baptistery model enables us to achieve a believable acoustics restitution through auralisation techniques. More in details the acoustic coupling between the volumes of the building - respectively the dome, the matroneum and the ambulatory - was studied with acoustic measurements and numerical simulations. Concerning the influence on absorption at low frequencies, the construction techniques used in the vault were discussed. Furthermore, the layout of a Gregorian soloists in the choir part of the Baptistery was simulated according to musicological researchers, both in unoccupied and occupied conditions.
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Organization:
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F. Martellotta, DICAR-Politecnico di Bari |
Z. Sü Gül, Bilkent University |
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The Teatro Colón in Buenos Aires as a Double-Function Hall
Gustavo Basso
[Abstract]
The Teatro Colón in Buenos Aires is well known for its acoustic
quality. It is considered not only one of the best opera houses in
the world, but also an exceptional space for symphonic music,
as highlighted by Leo Beranek and colleagues in two articles
published in 2000 and 2003. The architectural and acoustical
analysis of the hall shows some of the possible causes of this
dual behaviour, derived from a specific combination of
architectural features such as the shape of the horseshoe, the
height and depth of the boxes at the upper levels and the
design of the proscenium and pit. These components contribute
to create a very enveloping sound field when used for
symphonic music, similar to those of a shoe-box auditorium, and
allow the maintenance of the acoustical balance between the
singers and the orchestra when used as an opera theatre. As
can be seen from audience opinion polls on the sound
perceived, acoustic measurements and the outcomes of a
digital model, this behaviour is more noticeable on the main
floor. In this work, the acoustic quality of the Colón Theatre as a
double-function hall is analysed.
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Physical measurements vs. auditory assessment of a concert hall by different groups of users: a case study
Tadeusz Fidecki, Jan Zera, Andrzej Miśkiewicz, Barbara Okoń-Makowska, Tomira Rogala, Teresa Rościszewska, Ewa Więckowska-Kosmala, Maciej Jasiñski, Maciej Łukaszewicz
[Abstract]
The acoustical quality of a medium size (6200 m3 volume, 880 seats) shoebox shaped symphonic concert hall, renowned for its acoustics and protected by the cultural heritage law, was evaluated by the measurement of acoustic parameters (ISO 3382-1), and by auditory assessment of sound with the use of survey questionnaires which required from the respondents to judge various attributes of sound and also comment on their experience with the hall. The judgments were obtained from the conductor, the soloists, orchestra members, choir singers, recording engineers, and from the audience. The hall’s acoustics received generally high ratings from the audience. Orchestral musicians highly estimated the audibility of one’s own instrument and other instruments in the orchestra. Choir singers judged the audibility of other voices in the choir as satisfactory. The conductor assessed the overall sound quality as good, but also pointed at insufficient fusion of sound across the orchestral instrumental groups. Sound engineers judged the hall’s acoustics as very good for recording of chamber music and small orchestras, but not for very large ensembles. An attempt is made in this paper to relate the results of auditory judgments of the hall’s acoustics to its measured acoustic parameters.
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Acoustical Conservative Rehabilitation of St. Roque Church, Tollecantto - Goa
Menino Allan S. M. Peter Tavares
[Abstract]
St. Roque Church, Tollecantto, Goa originally a Chapel was built in 1883. The conservation of the church edifice is in progress. Preservation of existing worship ambience is essential in any conservative rehabilitation of a heritage church. It is crucial to have correct reverberation time in the church so as to allow a simultaneous optimization of the worship space for speech and music. In this design, Reverberation Time (RT60) of a sound signal inside the conserved church is established as a conservation reference parameter. Ideal Reverberation Time (RT60) for St Roque church was calculated and found to be 1.33 s. Impulse Response Tests were conducted in the church for three different source positions (original High Altar; Existing Altar; Proposed altar) with recordings done at ten locations. Almost all the recording locations have recorded a RT60 value of 1.4s. Therefore, the value of measured RT60 = 1.4s is established as a reference yardstick to monitor the rehabilitative expansion of the church to accomodate another 150 faithful in the worship space.
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The Acoustical Study of Heritage Buildings: A Holistic Approach
Antonio Pedrero, María Ángeles Navacerrada, Daniel De La Prida, César Díaz
[Abstract]
In recent years there has been a growing interest in the
acoustics of historical sites, resulting in numerous scientific
publications in which the acoustics of different types of
heritage buildings are analyzed.
The approach and methodology to be used to carry out such
studies depends on the objective to be achieved with them.
In some cases, such as when the aim is to assess the
suitability of the building for a specific application (musical
concerts, lectures, etc.), a purely acoustic approach may be
appropriate. However, when it comes to explaining the
acoustic behavior of certain venues in a historical context,
not only acoustics, but also other disciplines such as history,
archeology, sociology or musicology, among others, may be
necessary to carry out a rigorous study.
In this communication the main challenges of historical study
of the acoustics of the heritage buildings are analyzed, and
some methodological aspects to address such studies are
proposed.
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Heritage and Avant-garde - On the Acoustic Design of a Series of 19th Century and Modern Australian Multipurpose Halls
Claudiu Pop
[Abstract]
The acoustic design of multipurpose halls has always been a
challenge in terms of optimising the acoustic characteristics of the
halls to accommodate various performance types. This study
investigates and presents the findings of a series of halls ranging
from old to new and rated anecdotally from poor to excellent. The
ambisonic-technique impulse responses measurement is followed by
computer acoustic 3D modelling and systematic investigation of the
acoustic qualities. A discussion of the suitability of small unamplified
spaces for both speech and music is provided, as well as results of
the computer modelled alterations of the halls’ properties to provide
suggested improvements.
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Analysis of the Acoustic characteristics of a Museum of Modernist Architecture - Art Museum São Paulo Assis Chateaubriand
Marselle Nunes Barbo, Eric Brandão
[Abstract]
The architectural design involves the knowledge of
variables of
fundamental importance to a quality result. Acoustically
characterizing museums delivers design directives of these
buildings according to the precepts of acoustic comfort of
the
users. The museums have an important educational role,
therefore valuing these buildings goes beyond the
principles of
museology, such as the rules for conservation of works. This
work
deals with the acoustic characteristics analysis of the Assis
Chateaubriand São Paulo Art Museum, located in São
Paulo
city, Brazil, being the most important collection of
European art
in the Southern Hemisphere. The architectural design of
modernist style is authored by architect Lina Bo Bardi
becoming
a landmark of 20th century architecture. The exhibition
mode of
the works, which instead of the traditional linear exhibition
through the installation of the works on the walls uses
crystal
easels in the middle of the gallery is also a feature that tells
how
avant-garde the design of this museum. The acoustic
studies
were performed through simulation of the three-
dimensional
model in computational software with comparisons
between
source position and microphones and also the way of
exposition
of the collection. This work is part of a PhD research that is
underway.
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Acoustic suitability of heritage-listed buildings using BRASS software: a case study of Armando Gonzaga Theater, Rio de Janeiro, Brazil
Guilherme Coutinho Fagerlande, Maria Lygia Niemeyer, Julio Cesar Boscher Torres
[Abstract]
The Armando Gonzaga Theater was designed by one of the greatest modern Brazilian architects, Affonso Eduardo Reidy. Protected by INEPAC (State Institute of Cultural Heritage), this theater is located in the neighborhood of Marechal Hermes, a district of lower purchasing power in the west zone of Rio de Janeiro, far from the city center, and unique cultural facility on the spot. Being a heritage-listed building used for different purposes, from plays to acoustic and amplified concerts, it is necessary to adapt it in order to provide better acoustic performance using the new technologies. However, an intervention project in this building must respect the requirements imposed by INEPAC. The aim of this paper is to study alternatives for acoustic adequacy of this theater that are compatible with the heritage limitations required. The methods used are the simulations of the current situation compared to the measurements in the place, and the simulations of situations with new interventions, using BRASS software (developed in the Urban Engineering Program of the Federal University of Rio de Janeiro). The results show that it is possible to make these proposed adjustments, improving the acoustic performance of the building, but respecting its original characteristics.
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Determination of the characteristics of contemporary Turkish mosque and its acoustical properties
Elma Alic, Aslı Özçevik Bilen
[Abstract]
In the last 10 years in Turkey the number of a place of worship of the followers of the religion of Islam (mosque) has increased. Mosque, as a multi-functional place designed to be used technically for speech and music performances, is an active research area for experts of acoustics.
In this study, a section of a comprehensive study focused on speech intelligibility in mosques is presented. The aim is to determine Turkish mosque types which are commonly built since the year of 2000 and to evaluate of its acoustical properties. Mosques built since the year of 2000 in a city which has the average population of the Turkish urban areas, are selected to be analyzed for determining the specification of the sample mosque models’ physical characteristics. Method used in this study is; collection of the determined mosques’ properties; comparative analyze of the collected data; synthesizing the design of the sample mosque and obtaining the room acoustic parameters by using simulation techniques. Finally, acoustical parameters of sample mosque are compared with the acoustical parameters of mosques suggested in the literature. Also, by this research the base model of the contemporary Turkish mosque is provided using for future studies on mosque acoustics.
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Organization:
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M. Garai, University of Bologna |
C. Scrosati, Construction Technologies Institute of the National Research Council of Italy |
M. Nolan, DTU Electrical Engineering, Acoustic Technology |
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On the Revision of ISO 354, Measurement of the Sound Absorption in the Reverberation Room
Martijn Vercammen
[Abstract]
The measurement method for diffuse field sound absorption (ISO 354) is troubled with low reproducibility, far worse than can be accepted in respect to design of spaces, control of quality and legal security. This is caused by deviations from the basic assumption for the relation between sound absorption and reverberation time: the presence of a homogeneous and isotropic sound field. Research has been done to increase and quantify the diffusivity of the sound field in the room.
It was found that indeed it is possible to improve the diffusivity but the only reliable indicator of the diffusivity is the sound absorption of the sample itself. Only improving the room will not take away the differences between laboratories. An ISO working group has made a working draft for the revision of ISO 354, that contains a qualification of the room with a minimum sound absorption of a reference absorber and correction of the outcome of every sample, based on the measurement result of the reference absorber.
This working draft was circulated and commented. A basic outline of the method will be presented as well as a summary of the comments given.
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Design Principles of the Italian Round Robin Test on Reverberation Rooms
Chiara Scrosati, Diego Annesi, Luca Barbaresi, Roberto Baruffa, Filippo D'Angelo, Giuseppe De Napoli, Michele Depalma, Antonino Di Bella, Sabato Di Filippo, Dario D'Orazio, Massimo Garai, Nicola Granzotto, Valter Lori, Francesco Martellotta, Antonio Moschetto, Francesco Pompoli, Andrea Prato, Pietro Nataletti, Fabio Scamoni, Alessandro Schiavi, Fabio Serpilli
[Abstract]
This paper describes the design principles at the basis of
the Round Robin Test carried out among the laboratories
of the Sound Absorption working group of the Italian
Standardization Body (UNI). In recent years, the attention
to the determination of the measurement uncertainties in
building and room acoustics fields raised up. Moreover,
sound absorption has always been difficult to be
determined, in particular for very highly absorptive
materials, e.g. due to the edge effect, and the results
coming from different laboratories cannot always be
compared, due to differences in the actual sound field
found in nominally equivalent facilities. During this RRT,
carefully selected samples have been measured by the
different laboratories participating to the RRT in their
reverberation rooms. The procedure and the main results
will be shown.
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Some comments on using a reference absorber for absorption measurements in reverberation rooms
Volker Wittstock, Heinrich Bietz, Sylvia Stange-Kölling
[Abstract]
The uncertainty of absorption measurements in reverberation rooms is considered to be unacceptably large for certain applications. A long-discussed possibility to reduce the uncertainty is to measure the absorption relative to the known or defined absorption of a reference absorber. To check this approach, measured absorption coefficients from a large data base with round robin results are explored. This check is possible because several different specimen have been measured in the same set of reverberation rooms in some of the round robins. Furthermore, a special reference absorber has been proposed recently. This material was used for a comparison measurement at PTB. About 10 different laboratories came to PTB’s reverberation room and measured the sound absorption of the proposed reference absorber using their own measurement equipment and data analysis. Additional investigations involved measurements of the reference absorber and some other specimens in two different reverberation rooms at PTB. Measurement and analysis results are presented and discussed in view of the current proposals for the revision of ISO 354.
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The effect of absorber placement on absorption coefficients obtained from reverberation chamber measurements
Jamilla Balint, Florian Muralter
[Abstract]
The standard procedure of sound absorption measurements in reverberation chambers
relies on the well-established reverberation theory, which assumes isotropic sound
incidence on the absorbing specimen. Yet, due to the non-uniform placement of
absorbing material in the room, this condition cannot be fulfilled in practice. Some
angles of incidence are emphasized over others, especially at low frequencies. The
current practice to obtain a more isotropic distribution of sound incidence is to install
sound scattering elements in the form of panel or boundary diffusers. However, recent
studies have shown that in spite of the presence of scattering elements, the sound field
incident on the sample is not isotropic. This study investigates the effect of different
absorber placements on the decay rates and absorption coefficient values.
Measurements are conducted in a reverberation chamber with absorbers mounted in the
corners and edges of the room. Such setup is expected to allow for a more uniform
distribution of sound incidence, leading to increased decay rates. In this work we will
discuss if such mounting setup should be added to ISO 354.
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Measurement of Diffuse-Field Sound Absorption Coefficient of Materials Using the Two Microphones Method
Paulo Medeiros Massarani, Ricardo Villela, Daniel Pazos
[Abstract]
Projects in room acoustics usually demand values of sound
absorption coefficients of materials measured in diffuse field.
The most widely used method for diffuse field measurement is
the reverberant chamber method described by ISO 354. Diffuse
field coefficients can also be estimated by integration of
absorption values obtained from measurements for discrete
sound incidence angles in free field. Furthermore, some
alternative methods use sound intensity sensors for the
measurement of absorption coefficients in reverberation rooms.
This work presents a case of absorption coefficient
measurement in a reverberation room using the two
microphones method. The concept of a single normal incident
plane wave, as occurs in the Kundt’s tube, was extend for the
incidence of multiple plane waves from the various propagation
paths in the room. Absorption coefficients of glass wool panels
were obtained experimentally for samples laid down at the
room floor, by using an omnidirectional loudspeaker at three
points of the room and two microphone positions close to the
sample surface. Transfer functions from the source to the
microphone were measured using swept sine excitation, and
the coefficient calculation was performed afterwards.
Assessment of measurement results shows how sample size and
microphone positioning can affect results and measurement
accuracy.
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Design of a New Testing Chamber to Measure the Absorption Coefficient Down to 25 Hz
Peter D'Antonio, Mélanie Nolan, Efren Fernandez-Grande, Cheol-Ho Jeong
[Abstract]
This presentation will describe a new 286 m3 testing
chamber that will allow the measurement of the random
incidence absorption coefficient, according to ISO 354 and
the normal incidence low frequency absorption coefficient,
according to ISO 10534-2. The rev room’s cuboid
dimensions were determined from an optimization
procedure that minimizes the standard deviation of the
modal response. Both dome-shaped boundary diffusors
and hanging panels are used to provide satisfactory
diffusivity, which is verified using a 200 mm reference
absorber according to ISO 354. The isotropy of the incident
sound is measured with a new Sound Field Analysis
Recorder (SOFAR) and quantified with an analysis of the
wavenumber spectrum. 12 impulse responses are collected
from 2 sources and 6 distributed microphones. The MLS
impulse response data is processed with a proprietary
Matlab program, which automatically removes the rms
noise floor, creates an energy decay from the optimal
linearity limit, achieving maximum dynamic range, and
determines an averaged plot of T60 versus frequency, from
all of the microphone-speaker pairs. A steel and concrete
lined low frequency impedance tube measuring 600 mm x
600 mm x 5.8 m long and 7 tons is used to measure
absorption from 25-250 Hz.
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Experimental characterization of the decaying sound field in a reverberation room
Mélanie Nolan, Marco Berzborn, Efren Fernandez-Grande
[Abstract]
Reverberation-chamber measurements of sound absorption
coefficients are based on the assumption that the sound is
diffuse (i.e., homogeneous and isotropic). Such sound field
should be established both before and during the sound
decay. Yet, the test conditions for absorption measurements
render the establishment of a completely diffuse sound field
difficult, if not impossible. It is therefore of interest to discuss
the concept of acoustic diffusion in a reverberation chamber.
A recent investigation examined sound field isotropy in
reverberation rooms in the steady state, based on an
analysis of the wavenumber spectrum in the spherical
harmonics domain. The purpose of the present study is to
analyze the isotropy of the sound field in a reverberation
chamber across time. The evolution of the sound decay is
examined by estimating the wavenumber spectrum in
successive time instances, leading to a time-dependent
analysis of isotropy. Experimental results based on
measurements in a reverberation room in two damping
conditions (empty and with absorbing material on the floor)
are presented.
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On the Directional Properties of Energy Decay Curves
Marco Berzborn, Mélanie Nolan, Efren Fernandez-Grande, Michael Vorländer
[Abstract]
A pre-requisite for the absorption coefficient estimation in accordance with ISO 354 is that the sound field in the used reverberation room is diffuse. A diffuse sound field is generally defined by the isotropy condition - requiring the sound field to be composed of infinitely many sound waves from uncorrelated sources with directions of arrival uniformly distributed over a sphere.
Microphone arrays allow for the decomposition of the sound field into a continuum of plane waves composing it and therefore directly allow for the analysis of the isotropy condition. By extending this concept to the decay process of the sound field we propose and investigate the directional energy decay curve as an analysis framework of the directional properties of the decaying sound field in a reverberation room.
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A case study on the new reverberation room built in University of Technology Sydney
Xiaojun Qiu, Qiaoxi Zhu, Shuping Wang, Jiaxin Zhong
[Abstract]
A new reverberation room has just been built at Centre for
Audio, Acoustics and Vibration in University of Technology
Sydney. The paper reports some key parameters of the room,
which include the background noise and its spectrum, the
number of modes in low frequency bands, the cut-off
frequency, the standard deviation of the spatial variations of
the reverberant field, the decay curves of the sound pressure
level versus time, the reverberation times, and the absorption
coefficients. These parameters are compared with the
published data of other reverberation rooms and some
important issues for making accurate measurements are
discussed.
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Practical aspects related to the measurement of the diffuse field absorption coefficient in scaled reverberation rooms
Louena Shtrepi, Francesca Latorella, Andrea Prato, Alessandro Schiavi, Marco Masoero, Arianna Astolfi
[Abstract]
The scaled reverberation room has proven to be an excellent tool for acoustic consultants and professionals alike to carry out comparison tests between acoustic materials and 3D systems or structures. The suitability of a scaled reverberation room is currently under investigation for the evaluation of the frequency-dependent sound absorption. At present, no standard is available on the methods that could be used to derive acoustic absorption coefficients from scaled measurements. In this work practical aspects of ISO 354 have been investigated within a 1:5 scaled reverberation room: a balance has been sought between reducing sample size, to reduce the manufacturing costs of materials, and finding the appropriate sample area, thickness, orientation and edges treatment, to obtain reliable values at mid and high frequencies. Four different materials have been tested. The paper discusses some of the findings of the measurements conducted on the same materials in a full-scale reverberation room according to ISO 354, in a 1:5 scaled reverberation room and in an impedance tube according to ISO 10534-2. The absorption coefficients data collected have been effective in proving that small reverberation room tests can provide compatible results compared to standard ones in 400-5000 Hz frequency range.
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Prediction of the Sound Absorption of Micro-Perforated Panels whose Holes are Extended in Length by Tubes
John Laurence Davy, Mohammad Fard, Qian Zhang, Yadong Lyu, Jun Yang
[Abstract]
Maa’s theory, which models the specific acoustic impedance
of a narrow air-filled hole in front of an air cavity using a lumped
parameter model, works well for short holes at low frequencies.
However, as the length of the hole or the frequency increases,
the acoustic velocity of the air in the hole varies along the
length of the hole as well as radially. It is necessary to use the
transfer matrix method (TMM) rather than Maa’s lumped
parameter model to calculate the impedance of the hole and
the cavity behind it. The peak sound absorption of a
microperforated panel (MPP) in front of an air cavity can be
moved to lower frequencies by extending the length the holes
in the MPP with rigid or flexible tube bundles in the air cavity
behind the MPP without increasing the total depth of the sound
absorbing system. Previously only Maa’s lumped parameter
model has been used to predict the sound absorption
coefficients of these tube bundle sound absorbers. These
predictions are only correct up to a certain frequency which
depends on the length of the tube extended holes. This paper
attempts to accurately predict the sound absorption to higher
frequencies using the TMM.
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Some notes about low frequency measurements in a reverberation room
Dario D'Orazio, Luca Barbaresi, Massimo Garai
[Abstract]
The modal field influences reverberation time measurements under the Schroeder frequency, as it is well known, but the reverberation room is an extreme environment that deserve some further considerations. Indeed the Schroeder frequency depends on the mode distribution, which in turn depends on the peculiar mode shapes, due to the room shape. For a typical reverberation room the boundary frequency separating the modal regime from the statistical one may extend up to three times the Schroeder frequency. Moreover, the reverberation time in the modal regime depends on the proper excitation of the sound field; this is the reason why some authors use the ’modal reverberation time’ for each mode instead of the average on a frequency band. Notably, the recent ISO 354 draft recommends exciting the sound field by placing the sound sources in the corners. The present work shows some practical experiences concerning the use of various kind of sound sources, changing the loudspeakers size, the distance from the corners, the excitation techniques. The results show how a proper sound source, whose electroacoustic characteristics are optimised for the ISO 354 frequency range, may improve the sound absorption measurements and may influence some metrics related to the sound field diffusion.
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Building FEM Low Frequency Room Models through Modal Decay Time Measurements
Roberto Magalotti, Valentina Cardinali
[Abstract]
The decay time of modal resonances in rooms can be evaluated from measurement of one or more impulse responses. The authors have analyzed the results of measurements in different environments, including a reverberation room. Comparison with Finite Element Method simulations helped in identifying the eigenfunction related to each eigenfrequency; furthermore, measurements with different source and receiver position have evidenced that computed decay times were, to a large extent, stable and repeatable. On this foundation, new FEM low frequency models for the measured environments were built by optimizing the acoustic impedance condition at the boundaries. The paper compares the outcome of the simulation models to the measured behavior and discusses the relationship between impedance boundary conditions and absorption coefficients.
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On the Use of Geometrical Acoustic Models of a Reverberant Chamber to Improve the Reliability of Sound Absorption Measurements
Francesco Martellotta, Ubaldo Ayr, Chiara Rubino
[Abstract]
Measuring sound absorption in reverberant chambers, even
when complying with ISO 354 standard, may often return
results
affected by significant uncertainty, while repeatability is also
a
well known issue when different laboratories are involved.
When
considering results from a single laboratory, obtained from a
specific chamber, several elements contribute to the quality
of
the measurement, some of them are more evident (like the
diffusers), while others are more difficult to account (like the
sample position, mounting, and sealings). With reference to
the
diffusers, one effect typically related to insufficient diffusion is
the
fact that calculated absorption coefficients are higher than
unity. The present paper investigates how these effects may
be
better understood by means of geometrical acoustic
modelling,
analyzing the variations in reverberation time (and hence in
absorption coefficients), resulting from different positioning of
diffusers, as well as locations of sample, sources and receivers
in
real rooms and comparing them with simulated values.
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Organization:
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A. Astolfi, Politecnico di Torino - Department of Energy |
N. Prodi, Department of Engineering, University of Ferrara |
D.S. Woolworth, Roland, Woolworth & Associates |
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Qualitative Evaluation of a Classroom Redesign
Ana M Jaramillo, Bruce C Olson, Peggy Nelson, Sarah Bochat, Michael Doing
[Abstract]
Shevlin Hall is home to the University of Minnesota’s
Speech-Language-Hearing Sciences department. Besides
the typical needs of any educational space, some of the
rooms have additional requirements to meet the needs of
a hearing-impaired population. In a previous study, we
presented the results of a renovation of classroom 110 that
addressed the room acoustics and electro-acoustics
redesign, to achieve adequate speech intelligibility. This
study presents the results of a qualitative study of the
impressions of the building users before and after the
renovation of classroom 110.
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Signal-to-noise ratio in university lecture halls with low intelligibility
Dario D'Orazio, Domenico De Salvio, Laura Anderlucci, Massimo Garai
[Abstract]
Speech intelligibility plays a key role in determining the
quality of verbal communication. It depends on the
acoustic characteristics of the room and the signal-to-
noise ratio (SNR). Background noise strongly shapes the
intelligibility, so being aware of its components is
fundamental. In fact, HVAC noise, anthropic noise and
external activities noise contribute to background noise at
different frequencies. In this work it is shown how it is
possible to distinguish the various noise sources which
contribute to background noise applying statistical
techniques to sound level meter measurements taken
during lectures. The same techniques are used to
characterize the speech level, which is the signal in the
SNR. Organizing the data collected with a sound level
meter, an asymmetrical distribution is built and studied.
Various techniques permit the association of each sound
source to the corresponding sound level: percentile levels,
Gaussian mixture model based on peak detection,
Gaussian mixture model and k-means clustering. In the
present study these techniques are applied to several
university lessons in order to identify different sound signals,
in particular the received speech level and the student
activity. Results are compared and discussed.
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Adaptation to Room Acoustics and Its Effect on Speech Understanding
Pavel Zahorik
[Abstract]
Effective sound transmission between source and receiver is
essential for good communication and sound quality in learning
spaces. Of course, sound transmission can be significantly
affected by the acoustics of the space. Under many
circumstances, however, these acoustical effects have
relatively minor perceptual consequences. This may be
explained, in part, by listener adaptation to the acoustics of the
listening environment. Here, evidence that room adaptation
improves speech understanding is summarized. The adaptation,
is rapid (around 1 second), strongest for rooms with
reverberation times between 0.4 and 1 second, and observable
for a variety of speech materials. It also appears to depend
critically on the amplitude modulation characteristic of the
signal reaching the ear. A better understanding of room
adaptation effects can inform and contribute to methods for
improving speech understanding and sound quality in rooms for
both normally-hearing and hearing-impaired listeners.
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Acoustic treatment of school spaces and its impact on students and teachers. Users’ self-assessment.
Mikolaj Jarosz, Irena Polewczyk
[Abstract]
The paper presents the latest research concerning the correlation between changes in the school building room acoustics and noticeable changes in the communication, behaviour and wellbeing of students and teachers.
The primary school covered by the research is the second largest school of this type in Poland. There was a comprehensive acoustic intervention in 2018 as a part of a pilot study for Warsaw municipality preparing for systematic improvement of acoustics in capital schools. This case gave an excellent opportunity to assess the impact of changes in room acoustics on pupils’ school performance, teachers’ effort and wellbeing of both groups.
The research used the Acoustic Change Feelings Scale (SOZA) for teachers, parents and students, analysis of documentation and interviews. The questions concerned wide range of scholar activity aspects: concentration, speed of work, level of task fulfillment and school achievements, short term memory capacity, fatigue, pupils behaviour during lessons and breaks, level of excitement and aggression or teachers’ voice effort.
Approximately 1,000 students, and 150 teachers were included in the study. Acoustic measurements were made (before and after acoustic treatment), including measurement of the equivalent sound level LAeq, reverberation time T, speech transmission index STI and spatial decay of sound.
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Association between measurement and modeling results of room acoustics in open-plan learning spaces
Joose Takala, Jesse Lietzén, Saveli Valjakka, Henry Niemi, Mikko Kylliäinen
[Abstract]
The measurement methods presented in standard ISO 3382-3
have been suggested for evaluation of acoustics in open-plan
learning spaces. This means that there should also be a suitable
design method to ensure the acoustics beforehand. The
purpose of this study is to find out whether the results of the
measurements carried out in recently built open-plan learning
spaces correspond to the results of room acoustic modeling
performed in design phase of the spaces. Measurements were
done in five new open-plan schools. The results show that room
acoustic modeling can be used as a design method for open-
plan learning spaces.
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Well-being and noise annoyance outcomes from first graders and relationships with classroom acoustics
Arianna Astolfi, Giuseppina Emma Puglisi, Andrea Prato, Silvia Murgia, Greta Minelli, Tiziana Sacco
[Abstract]
The question on the extent to which environmental noise exposure and excessive
reverberation affect well-being of children at school is still largely unexplored. Younger
children of kindergarten and primary school can suffer more than older pupils of well-
being and related mental health issues. Well-being measures are focused on self-esteem,
emotional health and resilience, quality of the relationships with friends and family,
satisfaction of school, life satisfaction. Noise exposure to traffic noise was demonstrated
to be associated with mental health and behavioral problems in children, but nothing has
been found until now that relates noise and excessive reverberation at school with well-
being and mental health issues, despite the bad acoustic conditions that characterize
classrooms in most of Europe. In this study well-being and noise annoyance from
different noise sources were investigated with a subjective survey based on validated
questionnaires. Twenty-five first-grade classes of 20 primary schools and 5 kindergarten
classes of Turin (Italy), participated in the study. Over 366 and 75 pupils answered the
questionnaires, respectively. Classrooms were characterized by different reverberation
times and noise exposure. The happiness scale allowed to explore differences between
happy and unhappy children. Relationships with good and bad classroom acoustics have
also been investigated.
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The sound environment of German preschools and preschool teachers’ thoughts about sound generated by children
Saki Noguchi, Hisako Yoshizawa, Masayuki Sato, Tadashi Konishi
[Abstract]
Noticeably, preschool children develop language and communication, so improvement of the preschool’s sound environment is important. Recent Japanese preschool facilities have experienced problems with noise in the sound environment. One cause is a lack of sound absorption. Moreover, the loudness of the generated sound is a second cause as preschool teachers generally think that children’s voices should not be muted. Therefore, approaches from both the acoustic environment and people are needed to improve the sound environment. Meanwhile, in Germany it was specified in DIN 2004 that the proper conditions of sound absorption in nursery classrooms are superior to those of school classrooms; consequently, nursery rooms began to absorb sound (Kawai, et al., 2018). We researched the sound environment and preschool teacher’s thoughts about it at German preschools to identify how a comfortable sound environment is created.
It was found that sound absorption of nursery classrooms has become established and preschool teachers’ awareness of the acoustic environments is high, and the generated sound tends to be low because teachers think children should learn to use a suitable voice level. Consequently, it was suggested that proper acoustic performance improves human awareness and leads to the creation of a better educational environment.
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Children's perception of degraded speech at normal vs. fast speech rate
Isabel S. Schiller, Dominique Morsomme, Malte Kob, Angélique Remacle
[Abstract]
This study investigated the effect of degraded listening conditions and speech rate on children’s answer accuracy and response time in a speech perception task. Fifty-three normally-developing children (aged 5-6 years) listened to 72 pseudo-word pairs presented at two different speech rates (normal and fast) and four different listening conditions (healthy voice in silence [control], hoarse voice in silence, healthy voice in speech-shaped noise (SSN) at 0 dBA SNR, and hoarse voice in SSN at 0 dBA SNR). The participants had to decide whether the words were the same (e.g. /filam/ - /filam/) or different (e.g. /mafin/ - /nafin/). For either speech rate, degraded listening conditions were found to significantly decrease answer accuracy and increase response times. Regarding response time, speech rate interacted with listening condition: At fast (as opposed to normal) speech rate, children responded slower to healthy voice in SSN but faster to the three other listening conditions. Our findings suggest that speech signal degradations may disrupt children’s speech perception even at normal speed. Speech rate might influence the extent of listening effort associated with different listening conditions. A good quality and adequate transmission of the speech signal may help children to listen effectively.
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Speech Intelligibility in Classrooms: What if the Teacher is Dysphonic?
Pasquale Bottalico, Keiko Ishikawa
[Abstract]
Several studies have analyzed the effect of noise and poor classroom acoustics on the learning process. The acoustic complexity of a classroom is often compensated for by increased vocal effort of teachers. This explains the high prevalence of voice problems among teachers (about 60%). The objective of this study is to determine the effect of a dysphonic voice on speech intelligibility in a real classroom. The test was administered in the presence of artificial noise to 83 college students. The speech from 7 takers with normal voice quality and 7 with dysphonic voice was used during the test. Two different vocal efforts were used, a raised and a loud level. Results indicate that in a range of SNR between 0 dB and -10 dB dysphonic voices are not improving in intelligibility while voices with normal voice quality on the same SNR range showed an intelligibility improvement of about 25%. In conclusion, teachers’ voice problems lead to a decrease in speech intelligibility. Therefore the acoustical parameters necessary to obtain optimal intelligibility should be more restrictive when the teachers suffer from voice problems.
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Studying for an exam in an open-plan study environment: Does background noise impair performance?
Ella Braat-Eggen, Jikke Reinten, Maarten Hornikx, Armin Kohlrausch
[Abstract]
Students in higher education work for a large part of their
time in rather noisy open-plan study environments (OPSEs).
A recent field research has shown that almost 40 percent
of the students were ’much’ to ’very much’ disturbed by
background noise while working in an OPSE. Moreover,
the students indicated that they were very disturbed by
the background noise when they were studying for an
exam. Although the influence of background noise on
different cognitive tasks has been shown in many studies,
little is known about this particular student task. Therefore,
the aim of this work is to analyze the influence of
background noise on the ’studying for an exam’ task in an
OPSE. For this experiment, a set of tasks was developed to
simulate ’studying for an exam’. As output measure, the
performance of students was measured when carrying
out the simulated studying task in different realistic
background sound scenarios. These sound scenarios vary
in background speech and room acoustics. They were
developed, suitable to play back through headphones,
by convolving recorded student voices with binaural
impulse responses obtained from computational acoustic
modelling of an existing OPSE. In the current paper the
research design and first results are presented.
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Exploring the effect of ventilation type on the acoustics of primary and secondary school classrooms
Kieren Smith, Lily M. Wang
[Abstract]
To better understand the effect of indoor environmental
conditions on primary and secondary student performance,
measurements over six school days were logged in each of 218
classrooms at schools in the midwestern region of the United
States. Results from statistical analyses demonstrate the effects
of classroom conditions on student achievement scores.
Specifically, structural equation modeling has been used to
determine the effects of classroom speech levels, non-speech
levels, and room reverberation conditions on math and reading
achievement, while controlling for student demographics.
Information regarding heating, ventilation and air conditioning
(HVAC) system types was also gathered in each classroom. This
paper reports on the correlation between HVAC system type
and the measured speech or non-speech levels in classrooms
during the school day. Additionally the correlations between
measured acoustic and indoor air quality parameters over time
are presented, with specific regard to how HVAC system type
might affect the correlations.
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Speech comprehension and intelligibility in noise in 11 to 13 years old children: what is the relationship?
Nicola Prodi, Chiara Visentin, Erika Borella, Irene Mammarella, Alberto Di Domenico
[Abstract]
In classrooms, the correct reception of what the teacher says
is a necessary condition for learning, but it is not sufficient to
ensure that the content of the message is understood.
Notwithstanding this, usually a direct positive correlation
between the performance in speech reception and speech
comprehension (SC) is assumed. In fact this idea is the
ground for the use of speech intelligibility (SI) in the design of
learning spaces. In this study the relationship between SC
and SI is investigated in three listening conditions, with
reference to 330 school-age children, aged 10 to 14 years
old, with normal hearing.
Participants completed two tasks designed and validated to
measure SI and SC, which were presented in three
conditions: quiet, students’ activity noise and road traffic
noise. The tasks were presented to groups of children within
their classrooms, via tablet and loudspeakers, within a one-
hour session. The order of the tasks was counterbalanced
between the participants.
For each task, results in quiet were compared to the results in
noise, assessing the specific effect of each masker;
furthermore the size of the effects was compared across the
two tasks to gain insight on their relationship.
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Optimising the Acoustic Design for Multi-purpose Rooms used for a Variety of Speech Communication Activities
Colin Campbell, Helene Sallenhag, Erling Nilsson, Emma Arvidsson
[Abstract]
The sound environment for a room designed for speech
communication can be influenced by many factors. In a newly
completed multi-purpose lecture room, we wanted to see the
effects of the different design elements during the process
including; utilising predictive modelling, the surface materials in
terms of absorption performance, diffusion, amount and
placement as well as other design details influencing the overall
sound environment. We have followed this project though the
build and fit out process making acoustic measurements at
different stages. Typically, a room of this type might have
acoustic criteria for sound insulation and reverberation time.
However, this can be somewhat limited when we compare the
theoretical acoustic predictive modelling versus how the
occupants might actually perceive the room in real life
situations. We have measured additional room acoustic
parameters such as speech clarity and the difference in sound
levels. We will identify the acoustic differences as the room is
built up with ceiling, wall and floor finishes, in addition to the
design elements and furniture. Finally, we will discuss design
configurations which can make a room fit for multi-purpose
speech based activities.
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Background Noise and Phonological Processing in Second Language Learners
Emil Holmer, Thaisy Costa, Mary Rudner
[Abstract]
According to the Framework for Understanding Effortful
Listening (FUEL), adverse listening conditions deplete the
cognitive resources available for other tasks. For example,
in a noisy classroom, students may not be able to devote
their full cognitive capacity to learning. This may be a
particular problem for second language learners. In an
experiment, 20 international exchange students,
performed a Swedish visual rhyme task in quiet and in
noise. The rhyme task required the activation of
phonological representations of Swedish but on average
the exchange students had been learning Swedish as a
second language in Sweden for only 3 months. T-testing
showed that the exchange students had poorer Swedish
vocabulary and lexical access skills than a Swedish
comparison group but similar cognitive skills.
Unexpectedly, repeated measures ANOVA showed no
statistically significant effect of noise on visual Swedish
rhyme judgment and no significant interaction between
noise condition and group. This suggests that phonological
processing based on pre-existing representations is
protected from the deleterious effects of adverse listening
conditions, even when those representations are
incompletely established in second language learners.
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Towards Accessible Acoustic Criteria for Inclusion in Mainstream Classrooms
Emma Greenland, Bridget Shield
[Abstract]
New and refurbished school buildings in England and Wales need to comply with criteria for
acoustics set out in Building Bulletin 93 (BB93). However,
the acoustic design strategy to comply with Equality law and to ensure that
ëpeople with disabilities, including those with hearing impairment, must not
be placed at a disadvantage’, is unclear and is complicated by other design
factors and current funding constraints. This paper sets out to establish
acoustic criteria for inclusion of children with special, hearing and
communication needs (SHCN) in mainstream classrooms, using a ëreasonable
adjustments’ approach (which underpins Equality law). Listening disadvantage
is quantified by each type of need according to a review of the speech
perception studies evidence base. Holistic design and management of the
classroom is considered, by quantifying the prevalence of children with
SHCN; examining classroom management strategies and provision of assistive
listening systems; accounting for teacher’s voice protection; and
considering other design aspirations (such as sustainability) alongside
acoustic design objectives.
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Room acoustic conditions of schools, a church, and a residential premise for hearing impaired people
Cheol-Ho Jeong, Wan-Ho Cho, Ji-Ho Chang, Seong-Hyun Lee, Chang Wook Kang, Jin Gyun Park
[Abstract]
It has been known that hearing impaired people need more strict room acoustic requirements than normal hearing people for verbal communications. Therefore, The American speech-language-hearing association (ASHA) recommends a maximum reverberation time of 0.4 s in classrooms, signal to noise ratios (SNR) higher than 15 dB, and ambient noise levels lower than 35 dBA. We measured noise levels and room impulse responses from 18 rooms in two schools for hearing impaired children, a dormitory with 3 rooms, and a church mainly for the hearing impaired in the Republic of Korea and extracted room acoustic parameters. A questionnaire was distributed to obtain subjective reverberance and quality of verbal communication and well-being. The present study will discuss the correlation between the objective acoustic parameters and users’ satisfaction in terms of acoustic conditions.
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The need for comfortable and inclusive acoustical learning spaces for children with autism spectrum disorder
Hidetoshi Takahashi, Kanako Ueno
[Abstract]
Sensory abnormalities have often been reported as symptoms of ASD, and auditory over-responsiveness (AOR) is the most common sensory-perceptual abnormality in individuals with Autism Spectrum Disorder (ASD). This abnormality has been reported to interrupt behavioral adaptation, and sometimes even requires therapeutic intervention.
Recently, we reported that a greater acoustic startle reflex magnitude in response to weak stimuli of 65-85 dB was found in children with ASD compared to those with typical development, and that this index was related to various clinical features, including sensory problems, autistic traits, emotional/behavioral difficulties, and adaptive/maladaptive behaviors in ASD children. This intensity level of acoustic stimuli is frequently experienced in everyday situations including the classroom environment. We addressed sound environmental issues in several different types of school, including special-needs schools and nursery schools on a remote island in Japan. Creating an auditory-friendly school environment that provides a comfortable and inclusive learning space is an especially important issue to address for the healthy development of ASD children with AOR.
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Organization:
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V. Hongisto, Turku University of Applied Sciences |
A. Liebl, HSD Hochschule Döpfer |
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Physiological, psychological and performance effects of office noise
Jenni Radun, Valtteri Hongisto, Henna Maula, Ville Rajala, Darin Al-Ramahi, Mika Scheinin
[Abstract]
BACKGROUND. Background speech is a common disturbance in offices, especially in open-plan offices. Background speech influences performance and subjective estimations. However, not many studies have examined simultaneously physiological, performance and psychological consequences background speech has on humans. AIM. To examine the influence of background speech on subjective experience, performance, and physiological measures. METHODS. We examined the influence of background speech compared to silence in a between-group laboratory experiment. We tested 40 subjects: 21 subjects in the speech group and 19 subjects in the silence group. Background speech was a radio dialogue played at 65 dBA level, which people were supposed to ignore while making cognitive tasks and answering questionnaires. Background speech condition lasted for 50 minutes. Participants’ performance, subjective experience and various physiological stress reactions (e.g. stress hormone level, heart rate variability) were measured. RESULTS. Compared to the silence group, the speech group had lower performance and higher physiological stress level. Speech was subjectively estimated more annoying and loading, but less tiring than silence. PRACTICAL IMPLICATIONS. This study shows that background speech influences experience, performance, and physiological stress level. Therefore, its influence should be minimized in offices, where work requiring concentration is needed.
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A comparison of Cognitive Performance and Listening Effort test procedures
Alexander Markus Dickschen, Andreas Liebl, Stefan Bleeck
[Abstract]
Workers in open-plan offices are objected to irrelevant speech, which leads to decreased cognitive performance. With acoustical treatments, such as sound masking, the cognitive impact can be reduced, but there is no objective procedure to assess this improvement in real office environments. For in-situ evaluation, a robust test procedure is required. The serial recall test used in laboratories requires masking levels to be higher than the speech concealed, which seldom is the case in real offices. Therefore, a dual-task paradigm was investigated, which measures latencies for auditory stimuli presented in noise and silence. The method, originating from audiology for listening effort measurement, showed significant differences even for little masked, clearly intelligible speech. The underlying cognitive processes leading to those differences in response times need to be understood. Experiments with 21 normal hearing participants were conducted to investigate the memory component and the auditory identification process in the listening effort experiment. A comparison to the serial recall test was carried out. The findings suggest an arousal effect at work, with faster replies for higher masking in serial recall and listening effort test. The two tests seem to analyse different cognitive aspects.
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How Office Workers Cope with Distraction by Sounds in the Open Plan Office
Sven Steps, Rianne Appel-Meulenbroek, Theo Arentze, Remy Wenmaekers
[Abstract]
The open plan office is a complex sound environment
existing of speech, office equipment sounds, sounds from
building services and other background sounds. Each
type of sound can have a different influence on the
performance or well-being of the office workers. Much
research reports which sound sources are perceived to be
most disruptive or annoying. To some extent, acoustic
measures such as sound absorption and screens can be
used to reduce the intelligibility of speech or the sound
level of noise sources. Nevertheless, office workers need to
deal with the acoustic environment they work in. In this
research it is investigated which coping strategies are
used by office workers for different sounds and how these
strategies are perceived to be effective for improving
productivity. Questionnaires were distributed among
workers in three different companies that have problems
with sound. Results show that ”discussing the noise problem
with colleagues” is perceived as the most effective
strategy when intelligible speech is the problem. Less
effective, but more often applied, are so-called
avoidance strategies such as ”put work off” and ”interrupt
work”, which is striking. Ear plugs are only perceived to be
effective when speech is unintelligible or other sounds are
disturbing.
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Liveliness as a design parameter for open plan offices
Tom Bouwhuis, Theodoor Höngens
[Abstract]
Human activity is crucial with regard to the sound level in open
plan offices. However, design and evaluation standards such as
ISO 3382-3, as well as most measurements in office
environments, do not take into account how much noise is
being produced or acceptable to receive in actual work
situations. The Liveliness index - classifying sound environments as
’quiet’, ’tranquil’, ’lively’ or ’turbulent’ - aims to fill this gap. Different types of work come with different requirements for, or
sensitivity to sound production. Using a high density sensor grid,
the Liveliness method allows for accurate sound activity
measurements specific to each type of work present in a
certain organisation. Combined with room acoustical
parameters, these can be used as a design tool, indicating the
desired privacy or nuisance distances specific to the types of
work that characterise an organisation. Such dedicated design
helps avoiding discomfort and productivity loss. Since the
method uses descriptions rather than dB’s to qualify the sound
environment, communication with people unfamiliar with
acoustics is also simplified. This paper suggests how to use
Liveliness as a design input, and summarises our recent
experience applying it in different office environments. keywords: human activity, design tool, zoning, noise prevention
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Harmonizing Different Metrics for Speech Privacy
Markus Müller-Trapet
[Abstract]
It is regularly debated which metric to use for the assessment of speech privacy in open-office as well as closed-office environments. Metrics commonly used for this purpose include the Privacy Index (PI), which is derived from the Articulation Index (AI) determined according to ASTM E1130 as well as the Speech Privacy Class (SPC) determined according to ASTM E2638. Regardless of the method of determining these values, it has recently been shown that both metrics are mathematically related. In light of this information, it seems that the discussion might be better directed at different questions, for example what the proper values are for different situations.
This paper will highlight the similarity between Speech Privacy metrics as well as the commonly recommended values for open-plan and closed offices and how they relate to each other. The analysis and discussion is based on data from extensive field measurements as well as measurements under laboratory conditions. The data of the field measurements is also compared to subjective responses to assess the perceived privacy.
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Objective and subjective assessment of acoustics in open-plan offices
Karin Loh, Eric Kurz, Janina Fels
[Abstract]
The acoustic condition in open-plan offices can significantly
affect the work performance and job satisfaction. The main
problem in this case is mostly the noise caused by co-workers
and intensified by suboptimal room acoustics. To fully assess this
complex acoustic scene and to relate it to subjective
perception, it is important to use aurally-accurate measurement
methods as well as to consider psychoacoustic parameters,
which contain information on the temporal and spectral
structure of the noise and sound environment. Furthermore,
measurements are conducted in the daily working condition in
the presence of the workers (in situ) which differs significantly to
the noise generated by the building measured in the absence
of the working people. This works provides an overview on the
acoustic settings in German open-plan offices using objective
measures, such as room acoustic parameters and
psychoacoustic parameters, considering aurally-accurate
assessment measured in situ and in silence. Additionally, these
results are linked to the subjective perception assessed using
questionnaires in parallel.
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ISO 3382-3: Necessary But Not Sufficient. A New Approach To Acoustic Design for Activity-Based-Working Offices
Jack Harvie-Clark, Felix Larrieu, Cecilie Opsanger
[Abstract]
Since the room acoustic parameters in ISO 3382-3 were
published, there have been various international attempts to
classify offices achieving different values. However, recent
summaries of acoustic satisfaction demonstrate little
correlation with the room acoustic parameters that may
drive the design. A new case study illustrates significant
differences in acoustic satisfaction for people on two
different floor plates within the same building; these have
very different workplace design, but have almost identical
parameters according to ISO 3382-3.
A new approach is proposed to assess acoustic conditions
within activity-based working (ABW) offices. The method to
characterise the acoustic environment in terms of Liveliness
is used to determine appropriate noise levels for different
activities. A matrix of signal-to-noise ratios are postulated to
account for source level vibrancy and receiver sensitivity.
This enables a design framework for the in-situ attenuation of
speech between workstations, Dn, A, s. From the in-situ
attenuation requirement, the layout design can be
developed accordingly. This method identifies conflicts
between different types of use or activity where there is
insufficient attenuation between them. This informs the
workspace designer where enhanced in-situ attenuation
can improve the acoustic conditions for occupants.
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Comparison of models to predict the effect of background speech on work performance in open-plan offices
Tobias Renz
[Abstract]
There are different models to estimate the relationships between measurable acoustical
parameters and office workers’ perception, well-being and performance. These models
are based on Speech Transmission Index (STI), Fluctuation Strength (FS) or percentile
level statistics. Whilst STI requires a loudspeaker and is measured in unoccupied
conditions, other metrics, such as FS or percentile level statistics, can be determined in
situ during usual working hours. Nonetheless, the established models have some
shortcomings. STI can estimate the effect of one simulated speech source on short-term
memory performance but it cannot assess the effect of office noise exposure in occupied
offices. FS correlates with the impact of temporal-spectral variability of a background
sound on short-term memory performance but it depends on the sound pressure level
which does not have an effect on the short-term memory performance. Percentile level
statistics correlate with the speech-to-noise ratio: Higher differences between the 10th
and 90th percentile levels measured with fast time weighting lead to lower number
recall performances. As part of this study 100 sound conditions under which subjects
have to complete a number recall task are evaluated with respect to their relationships
with these three acoustical parameters. A cross-validation reveals comparable prediction
qualities of the models.
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Evaluation of Measurement Uncertainties of the D2S in Open-Plan Offices
Lucas Lenne, Patrick Chevret, Etienne Parizet
[Abstract]
Measuring the acoustic performance of open-plan offices is an
increasingly important issue for companies and employees.
Indeed, it has been highlighted in several studies that noise in
such workplaces reduces employees’ performance and makes
them be unsatisfied with their work environment. This is
especially important for conversational noise. The ability of a
room and its layout to reduce conversational noise can be
assessed using acoustic indicators and in particular the D2S
(spatial decay of the A-weighted level of a speech signal when
doubling the distance from the source on a line passing over
workstations). This indicator is now widely used in the field
because it is recommended by several national standards that
give values to be achieved. However, to date, there are no
studies that discuss the accuracy of a measurement in an
office. The purpose of this presentation is to provide an analysis
of the measurement uncertainties for this indicator. To that end,
the measurement principle of D2S will be presented as well as
an analysis of uncertainties based on their theoretical
developments. This analysis will be conducted using real cases
(measurements) and simulated cases using a room acoustics
software.
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Effect of office screens on the spatial decay of sound pressure level in open-plan offices
Tobias Renz
[Abstract]
Acoustical privacy is one of the most crucial, yet least satisfying aspects in open-plan
offices. Background speech often causes high distraction and dissatisfaction. At
medium
to large distances speech intelligibility should be minimised to reduce the disturbing
impact whereas at short distances intelligibility may remain high to support
communication within teams. A holistic acoustical design takes the requirements of the
occupants into consideration. Besides the zoning of teams and the application of an
appropriate layout, the combined use of sound absorbers, screens and masking can
diminish the impact of disturbing speech at medium to large distances. The height of the
used screens can have a high impact on the spatial decay of sound pressure level and on
the resulting speech privacy at medium distances. As part of this study sound pressure
level measurements of 15 offices are clustered into three groups, namely ’no screens’,
’half-high screens’, and ’room-high screens’. The resulting level differences at medium
distances of 5-10 m are significantly different: Whilst ’no screens’ and ’half-high
screens’ result in level differences of 10-25 dB, ’room-high screens’ yield level
differences of 20-40 dB. This implies that only ’room-high screens’ can reduce the
disturbance by background speech at medium distances sufficiently.
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How ISO 3382-3 Acoustic Parameter Values are affected by Furniture, Barriers and Sound Absorption in a Typical Open Plan Office
Remy Wenmaekers, Nicole Van Hout
[Abstract]
ISO 3382-3 offers a set of acoustic parameters to describe
acoustic conditions in open plan offices such as speech
transmission index, spatial decay rate of speech and
background noise level. In this research, these parameter
values were measured in a typical open plan office while
varying furniture, barriers and sound absorption. In many
open plan offices a high acoustic quality may be
desirable, requiring significant amounts of barriers and
(additional) sound absorption. Acoustic designs of open
plan offices can be tested using modelling or
measurements. In practice one may get very similar
design questions such as: ëDo we need screens and how
high do they need to be?’. Measured results from offices
are available in international literature, but direct
comparison is difficult because of different building
dimensions, floor plans and furniture elements. This
research presents results from measurements in a single
open plan office that is representative for many Dutch
offices. Furniture, barriers and sound absorption were
varied to study the feasibility of achieving certain
parameter values. The results from this study give insight in
the acoustic design that is necessary to achieve a certain
acoustic quality in the open plan office.
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Variability in the ISO 3382-3 metrics based on repeated acoustic measurements in open-plan offices
Manuj Yadav, Densil Cabrera, James Love, Jungsoo Kim, Jonothan Holmes, Hugo Caldwell, Richard De Dear
[Abstract]
This paper investigates variability in the key ISO 3382-3:2012 metrics based on two types of repeated measurements in open-plan offices - Type1, where the same path over workstations was measured from opposite ends, and Type2, where two different measurement paths were measured. Per metric, analyses performed used (i) the range of observed values; and (ii) the observed values. Results from category (i) analysis: mean(ΔType1), and 95% confidence intervals were 1.2 m(0.9,1.5) for rD; 0.8 dB(0.6,1.0) for D2,S; 1.2 dB(0.8,1.5) for Lp,A,S,4 m; and 1.2 dB(0.7,1.7) for Lp,A,B. mean(ΔType2) were between twice and thrice the respective values of mean(ΔType1). Results from category (ii) analysis: the reliability, based on intra-measurement correlation coefficient, was fairly high for all metrics, except for Lp,A,S,4 m for Type2 repeats. The repeatability limit/coefficient (r) was 2.5 m for rD; 1.7 dB for D2,S; 2.9 dB for Lp,A,S,4 m; and 3.0 dB for Lp,B, for Type1 repeats. The r values for Type2 repeats were substantially higher except for D2,S. Overall, most of the Type1 results seem reasonable considering repeats were conducted in complicated room acoustic environments, while Type2 repeats needs larger sample sizes in future studies. Some recommendations are outlined for ISO 3382-3 vis-à-vis Type1 and Type2 repeats.
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Numerical and experimental evaluation of a working environment on the basis of a speech intelligibility mapping
Marcus Maeder, Steffen Marburg
[Abstract]
In the area of workplace design, both the technical requirements of the workplace and inter personal concerns of the employees are of decisive importance in order to establish a suitable working environment. Often conflicts of interest arise due to the antagonism between disturbing noise and desired speech intelligibility. An employee who has to concentrate a lot at his/her workplace can be distracted both by disturbing background noises (air conditioning, fans, aggregates etc.) caused by low sound absorption as well as disturbing high speech intelligibility of background conversations. When assessing such situations, the speech transmission index (STI) is an important parameter. Influenced by the reverberation time and the signal-to-noise ratio, calculating the STI is a powerful tool for assessing demanding work environments.
The present work deals with the determination of an STI mapping on the basis of numerical simulations using ray tracing methods as well as calculations of the sound energy density distribution in room acoustics. The experimentally validated models can be used to optimize the workplace configuration to meet technical requirements as well as social concerns of employees.
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A statistical analysis of noise sources in open plan offices
Dario D'Orazio, Elena Rossi, Domenico De Salvio, Massimo Garai
[Abstract]
Acoustical comfort inside open-plan offices is necessary for
optimal work performance. In fact, it is well known that
productivity is closely related to the acoustic conditions of the
working environment. Noise inside offices can be due to various
sources: HVAC devices, office equipment (printer, phones, etc.)
and anthropic activities (human noise). The combined effect of
these noise sources may play a key role in privacy metrics as the
spatial decay of speech intelligibility. Starting from sound level
measurements, noise sources are identified using statistical
methods. The probability distribution of each source can be
highlighted, conferring a different sound pressure level to each
one. In order to investigate the dynamic behaviour of privacy
criteria, these analyses are carried out for each octave band
frequency. This work presents the results of different statistical
algorithms, applied to data coming from several measurement
campaigns in different offices.
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Influence of Active-Noise-Cancelling Headphones on Cognitive Performance and Employee Satisfaction in Open Space Offices
Benjamin Johannes Mueller, Andreas Liebl, Noemi Martin
[Abstract]
A high speech intelligibility in open space offices leads to
problems. As a result of the irrelevant sound effect it influences
cognitive performance negatively. Roomacoustic measures
alone do not lead to improvements if an open space office
design is maintained.
In a study it was examined if active noise cancelling headsets
influence cognitive performance and the subjective feeling in
an open space office. This was done with a cognitive task
(serial-recall) and a survey after every term. No significant
difference between the condition with active noise cancelling
headsets switched on and switched off as well as without
headsets could be determined. However, active noise
cancelling had an influence on subjective rating. The
background noise with active noise cancelling was rated as
significantly less annoying, ability to concentrate significantly
higher and the speaker distance was rated significantly larger in
comparison with active noise cancelling switched off.
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If speech is the source of noise, how can the source be eliminated or reduced?
Andreas Liebl, Noemi Martin, Benjamin Johannes Mueller
[Abstract]
The exposure to noise can be reduced by eliminating the source of noise (if possible), substituting the source with a quieter one, applying engineering modifications, using administrative controls, and by using protective equipment. It is generally agreed that the solutions controlling the source are more cost effective than those controlling noise along the propagation path. Administrative controls and the use of personal protective equipment are measures that control the noise at the receiver and are supposed to be applied only if measures at the source are not possible. However, in this line of argumentation human beings are usually only treated as receivers of noise but not as noise sources. Results are reported in which an inverse Lombard
effect or rather the Fletcher effect is used to reduce the speech
effort and thus the speech level of persons. A feasibility study shows that the speech level can be lowered by up to 3 dB without the necessary manipulation being perceived to be distracting.
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Labeling and exposition time affect subjective assessments of partial maskers for office noise
Sabine Schlittmeier
[Abstract]
Complaints in group and open-plan offices often concern office noise, in particular colleagues speaking on the phone or with other colleagues. One way of combating office noise is to introduce additional sound, a so-called partial masker, that reduces intelligibility of background speech and masks further disturbing sounds. At present, sound masking systems usually emit continuous noise, although this is said to have low acceptance rates among those affected. Two studies are reported which show that subjective assessments of partial maskers can vary with the labelling of the masking sound as well as with exposition time to it. In Experiment 1 (n = 25), the same continuous sound was evaluated differently depending on whether it was introduced as a ”natural noise of a waterfall” or as an ”artificial noise”. Experiment 2 (n = 72) verified that the subjective evaluations of partial maskers vary depending on whether judgements are given after short listening (30 s) or after one hour of cognitive work during it. The empirical results are discussed from psychoacoustic, cognitive psychological and applied perspectives.
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Water-based sound masking - An experimental study in an open-plan office
Valtteri Hongisto, Annu Haapakangas
[Abstract]
In Finland, the new governmental regulations require the Speech Transmission Index STI to be below 0.50 in open-plan offices. It requires both large amount of absorption materials and artificial sound masking due to low ventilation noise levels. Commercial sound masking systems are based on pseudorandom noise. Based on laboratory experiments, water sound has been experienced as more pleasant than pseudorandom noise when performing cognitive tasks. Our purpose was to determine how different water-based masking sounds are perceived at a real workplace. The study was conducted in an open-plan office, where sound masking based on pseudorandom noise had been used for 2 years at 44 dB LAeq. Our experiment included six conditions: original pseudorandom noise (P1a), four different water-based sounds (N1é{N4), and original pseudorandom noise (P1b). Each condition lasted for 3 weeks. The employees were asked to respond to a questionnaire during the third week at each condition. Each sound was presented at 44 dB LAeq. Eighteen employees out of 77 responded to all six questionnaires and their responses were analyzed. Water-based sounds appeared to be more disadvantageous than pseudorandom noise for almost all questionnaire measures. Due to some methodological weaknesses, the suitability of water-based masking sounds requires more research.
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Organization:
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A. Carullo, Politecnico di Torino - Electronics and Telecommunications Department |
A. Castellana, Politecnico di Torino - Electronics and Telecommunications Department |
M. Kob, Erich Thienhaus Institute, Detmold University of Music |
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Accounting for Variability over the Voice Range
Sten Ternström, Peter Pabon
[Abstract]
Researchers from the natural sciences interested in the performing arts often seek quantitative findings with explanatory power and practical relevance to performers and educators. However, the complexity of singing voice production continues to challenge us. On their own, entities that are readily measurable in the domain of physics are rarely of direct relevance to excellence in the domain of performance; because information on one level of representation (e.g., acoustic) is artistically meaningful mostly when interpreted in a context at a higher level of representation (e.g., emotional or semantic). Also, practically any acoustic or physiologic metric derived from the sound of a voice, or from other signals or images, will exhibit considerable variation both across individuals and across the voice range, from soft to loud or from low to high pitch. Here, we review some recent research based on the sampling paradigm of the voice field, also known as the voice range profile. Despite large inter-subject variation, the localizing by fo and SPL in the voice field will make the recorded values very reproducible within subjects. We demonstrate some technical possibilities, and argue the importance of making physical measurements that provide a more encompassing and individual-centric view of singing voice production.
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A New Paradigm of Effective Communication based on Voice Shapes
Alessio Carullo, Adriano Anibaldi, Arianna Astolfi, Alessio Atzori, Viviana Cennamo, Giovanni Zito
[Abstract]
A new relational paradigm is proposed that is based on voice shapes, which represent the speech style used to establish an effective communication. The following voice shapes are defined: ”rounded”, which means that a colloquial and empath communication is established; ”triangular”, which means transmitting energy, joy and interest; ”squared”, which highlights competence and solidity. These functional shapes can be strengthened by the additive component ”heart”, which emphasizes relational quality and the transmitted emotions. Two dysfunctional shapes are also defined: ”flat”, which is a monotone style that does not capture the listener attention; ”spiky” or ”X”, which is an aggressive style that transmits anger or blame towards the listener and does not match relational standards. An attempt has been made to match the voice shapes to acoustic parameters of the vocal signal, starting from 200 different-shape recordings. Such recordings have been processed in order to extract several voice features, such as the histograms of occurrences of: voiced and unvoiced periods; fundamental frequency; intensity; Cepstral Peak Prominence Smoothed. Preliminary results have shown a good discrimination power among the voice shapes using four parameters: 5th percentile and kurtosis of CPPS distribution, mode of unvoiced-period distribution and standard deviation of intensity distribution.
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Pitch Accuracy in Relationship to External and Internal Auditory Feedback
Pasquale Bottalico, Eric J Hunter
[Abstract]
Auditory feedback has a strong effect on the performance of musicians, and an assessment of such feedback is important in the search for the ideal sound. Singers rely not only on external auditory feedback but also on proprioceptive feedback associated with internal voice sensitivities. To better understand the effect of auditory feedback on pitch accuracy, eight professional singers and eight amateur singers were recorded singing two-octave arpeggi with three different levels of external and internal auditory feedback, with two different articulations (staccato and legato), and with different tempi (fast and slow). Professional singers’ pitch inaccuracy was not influenced by articulation. Pitch inaccuracy was reduced in the slow tempo relative to the fast tempo and in the ascendant semi phrase relative to the descendent phrase. The lowest pitch inaccuracy occurred in the High Tessitura. Pitch inaccuracy was influenced by auditory feedback. When both external and internal feedback was dampened, the greatest pitch inaccuracy was observed, while the other feedback conditions did not differ from one another.
In conclusion, it can be stated that the most important role in pitch regulation in singing is played by internal auditory feedback.
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Objective Assessment of the Effects of Semi-Occluded Vocal-Tract Techniques on Vocal Performance
Alessio Carullo, Arianna Astolfi, Alessio Atzori, Vittoria Carlino, Antonella Castellana, Claudio Fabro, Marco Fantini
[Abstract]
Semi-Occluded Vocal Tract Exercises (SOVTE) are a set of techniques that are based on a common principle, which is the increase of the vocal impedance through the semi-occlusion of the mouth. These exercises are mainly used by speakers and singers, but their use is increasing also in voice therapy. A new device has been developed (Vocal Feel) that combines the benefits of SOVTE with the vocal fold hydration. It is made up of a facemask, a humidifier-filter with a unidirectional valve and a bubble equipment. Preliminary tests, which use both static and variable impedance techniques, have shown promising results in the increasing of voice resonance and proprioception of singers. An experimental campaign has started to objectively assess the effects of such an equipment on the vocal performance that is based on the estimation of parameters of the recorded vocal signal of singers, who are equipped with a contact microphone and a microphone in air. The former allows the signal related to the vocal-folds vibration to be sensed, while the latter is used to acquire the signal at the output of the mouth. Example of estimated parameters are sound pressure level, fundamental frequency, Cepstral Peak Prominence Smoothed and singing power ratio.
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Relationship between brain activity and voice acoustic parameters during singing
Lady Catherine Cantor-Cutiva, María Del Pilar Rodríguez, Miguel Suárez-Russi
[Abstract]
BACKGROUND: The EMOTIV Epoc is a non-invasive neuro-
electric detection system that captures and amplifies brain
waves and facial movements. This system is also an electro-
encephalographic device with 14 electrodes and 2 reference
electrodes distributed around the head that captures and
transmit information about brain activity. Acoustic voice analysis
allows to characterize voice production. OBJECTIVE: To identify
the relationship between brain activity and acoustic parameters
of voice during singing. METHODS: This is a cross-sectional study.
Participants will be trained (or in training) singers who were
requested to sing the National Anthem and their favorite song.
Recordings will be performed in a sound booth with low
background and reverberation conditions. During the
recordings, participants will wear the EMOTIV Epoc. Voice
productions will be recorded with a microphone AKG P120
located at 30cm from the mouth of the singers. The analysis will
include the analysis of tendencies between the brain activity
information and changes on fundamental frequency, vocal
sound pressure, and CPPs. RESULTS: We expect to identify the
tendency between the brain activity and the voice acoustic
parameters during singing. We hypothesize that this tendency
may varies depending the musical gender.
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Relationship among brain activity, emotions and singing voice
Lady Catherine Cantor-Cutiva, María Del Pilar Rodríguez, Ferney Beltrán, Andrés Ojeda-Sánchez, Miguel Suárez-Russi
[Abstract]
OBJECTIVE: To determine the relationship among brain activity,
emotions and acoustic parameters of voice during singing
among singers under training. METHODS: This is a cross-sectional
study. Participants were singers under training who were
requested to sing the National Anthem and their favorite song.
Before the singing samples, participants were requested to fill in
a survey about their emotions sat the moment, and their voice
functioning and history of voice disorders. After the recording of
the singing samples, participants were requested to fill in, again,
the survey about the emotions. Recordings were performed in a
sound booth with low background and reverberation
conditions. During the recordings, participants wore the EMOTIV
Epoc. Voice productions were recorded with a microphone
AKG P120 located at 10cm from the mouth of the singers.
RESULTS: Preliminary analysis suggested a reduction in the
perception of happiness after the experiment among males
and females, which may be related with the condition of being
under examination. There was also a tendency of having a
lower degree of voice breaks (%) during the chosen song than
during the National Anthem. Among men, Harmonics-to-Noise
Ratio was bigger during the chosen song, whereas, women,
had smaller Harmonics-to-Noise Ratio.
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Organization:
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V. Lyberg Åhlander, Lund University/ Logopedics |
P. Bottalico, University of Illinois |
L.C. Cantor-Cutiva, Universidad Nacional de Colombia |
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Voice and Noise Dosimetry in College Voice Teachers
Pasquale Bottalico, Yvonne Redman, Lady Catherine Cantor-Cutiva, Chiara Vercelli
[Abstract]
Singing teachers are one the largest occupational groups developing voice-related problems. They use their voices at high intensities for long periods during their workday. This project aimed to describe hearing and voice behaviours of voice teachers. Eight voice teachers were involved in the study. In regards to voice, there was (1) an evaluation before and after voice classes; (2) voice dosimetry during classes. During the evaluation before and after classes subjects were asked to produce sustained vowels and a speech sample in three different voice levels. Several voice parameters were measured from the recordings. During the voice dosimetry, subjects wore the VoiceCare dosimeter while teaching.
In regards to hearing and noise exposure, there was (1) a hearing assessment before and after voice classes; (2) noise dosimetry during classes. Even if high vocal loads were measured during the working time, the analysis of the parameters pre/post showed an improvement of the voice quality of the teachers after class. As it concerns hearing, the noise exposure was on average 85 dB(A). Hearing assessments showed majority of subjects with hearing loss.
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A pilot study in primary school on the effect of a noise monitoring system with lighting feedback on teachers’ voice parameters, noise levels and subjective assessments
Sonja Di Blasio, Giuseppina Emma Puglisi, Chiara Gervasi, Antonella Castellana, Silvia Murgia, Greta Minelli, Giuseppe Vannelli, Simone Corbellini, Alessio Carullo, Arianna Astolfi
[Abstract]
Teachers’ vocal behaviour is affected by noise generated
by pupils that talk and move in classroom. Under noise
conditions, teachers are likely to increase their voice level,
thus causing possible negative consequences on their
vocal health. In this pilot study a noise monitoring system
with a lighting feedback, namely SEM (Speech and Sound
SEMaphore), has been used as an educational tool. It
encourages pupils to reduce their voice volumes through
a coloured lighting feedback. Seven teachers from four
classes of a primary school in Turin (Italy) have been
monitored over two or four working days with SEM
switched on and off. The results have shown that the
background noise level averagely decreased of about 2
dB after the SEM was switched on. With SEM switched off a
Lombard effect has occurred, i.e. an increase in the
speech level with background noise level at a rate of
about 0.4 dB/dB. Conversely, such effect has not been
found with SEM switched on, that is when pupils effectively
lowered their voice volume in answer to the lighting
feedback. Self-reported responses of teachers indicated
that SEM effectively reduces vocal effort and improves
speech intelligibility in the classroom.
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Effects of Background Noise on Acoustic-Perceptual Correlates of Voice
Supraja Anand
[Abstract]
Introduction: Professional voice users or vocal athletes are at
a higher risk for developing voice disorders due to heavy
vocal demands as well as environmental loading factors. This
presentation will review different types of occupational voice
use and associated vocal ergonomics. Specifically, will focus
on voice users working inside transportation vehicles; a less-
studied population within occupational voice research,
despite reports of high noise levels (over 65 dB) therein. Goal: To investigate the effects of background noise on
acoustic-perceptual correlates of voice among train
conductors/engineers. Methods: Twenty participants were instructed to speak in the
presence of three levels of background noise (mimicking
those in trains) with and without hearing protective devices in
a sound booth. Acoustic measures including fundamental
frequency, intensity, pitch strength, and acoustic voice
quality index were extracted from the speech signal. In
addition, participants provided perceptual ratings on
standardized (e.g., Borg vocal effort scale and vocal fatigue
index) as well as visual-analog scales. Results & Discussion: Results will be discussed in the context of
how well the acoustic correlates predicted the perceived
ratings of vocal effort/fatigue under different experimental
background noise conditions. Furthermore, the benefits of
applying such knowledge to the clinical voice evaluation will
be highlighted.
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Performance Evaluation of Autocorrelation Technique for Automatic Speaker Identification in Various Environments
Noha Korany
[Abstract]
The paper aims to improve the performance of automatic
speaker identification in various environments. The
autocorrelation technique and the discrete cosine transform
are applied for feature extraction of speech signals. Then, those
features are employed by the Gaussian mixture model classifier
to determine the speaker identity. The performance of speaker
identification is investigated for various settings of the applied
technique. Moreover, the effect of additive and convolutional
noise on the performance of the identification process is
investigated.
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Semantic Coherence and Speech Production in Adverse Listening Conditions
Carine Signoret, Emil Holmer, Mary Rudner
[Abstract]
According to the Framework for Understanding Effortful
Listening (FUEL), listening difficulties deplete cognitive
resources and influence how they are deployed. This may
affect the ability to hear an interlocutor correctly but also
the ability to make an appropriate response. We
investigated this as follows: Twenty young adults listened to
prerecorded spoken sentences that were either
semantically coherent, e.g. ”Her daughter was too young
for the disco” or not, e.g. ”Her hockey was too tight for the
cotton”. Then, they either repeated the sentences or
recast them as questions, e.g. ”Was her daughter too
young for the disco?”. The two tasks were presented in
blocks with equal numbers of coherent sentences. Sound
quality was either clear or degraded (6-band noise-
vocoding). Repeated measures ANOVA showed
statistically significant main effects of all factors
(coherence, task and sound quality), as well as significant
2-way interactions between task and coherence, and
sound quality and coherence. This suggests that semantic
coherence was more important in the recast than repeat
task and when signal quality was poor. These results
suggest that adverse listening conditions have a similar
effect on speech reception and production and that
semantic coherence may be more important for
speech production than reception.
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The potential of restorative spaces on noise-related wellbeing in schools
Ingrid Verduyckt, Rachel Bouserhal, Adriana Lacerda, Cécilia Borges, Annelies Bockstael
[Abstract]
The sound environment in schools causes stress, mental and physical fatigue in children and teachers, and disrupt student learning. Traditional noise abatement strategies mainly focus on reducing noise levels by room acoustics techniques. These approaches have some shortcomings 1) they are costly and hard to implement in existing infrastructure; 2) they fail to recognize the complexity of noise as a factor with both negative and positive aspects; 3) they have a top-down approach offering little opportunity for engagement and learning. We believe new complementary ways of improving noise related wellbeing in teachers and students are warranted. Objectives: Improving our understanding of restorative spaces’ impact on noise-related well-being in schools through the study of three such spaces in three different schools. Methods: Objective descriptions of the soundscapes and visual scapes of the three spaces are obtained through recordings of acoustical parameters following ISO 3382 and through visual urbanistic quantification methods. Subjective descriptions are obtained through focus group discussions with students and teachers around recordings and photos obtained in the three spaces. Habits of use and appreciation of the spaces and their impact on well-being are collected through self-assessment questionnaires by teachers and students using them.
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Improvement of the sound environment for supporting communication at nursery facilities: Approaches for changing room acoustics and users’ minds
Saki Noguchi, Kanako Ueno
[Abstract]
Recently, nursery facilities in Japan have been found to experience problems pertaining to the sound environment, such as insufficient sound absorption and loudness of the generated sound. Therefore, we examined approaches to create a sound environment to support communication in the classroom.
To address room acoustics, we installed sound absorbing materials in the classroom. As a result, with an average sound absorption coefficient of 0.25 or more, in addition to improving the impression of sound, the teacher reported ease in listening to children’s voices when many sounds were generated, such as during free play. To address users’ minds, we conducted a workshop for 5-6-year-old children and their nursery teachers to make them conscious to listen to the daily sounds, identify pleasant sounds, and feel the effect of corners using sound absorbing materials. As a result, the children and nursery teachers understood that an unnecessarily loud voice leads to noisiness, and that they can relax and enjoy sounds in the space surrounded by sound absorbing material.
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Acoustic Quality of University Classrooms: a Subjective Evaluation of the Acoustic Comfort and Conditions at the University of Sharjah Classrooms
Hussein Elmehdi
[Abstract]
In a previous study, the noise levels in classrooms and lecture
halls at the University of Sharjah (UoS), United Arab Emirates
(UAE) were identified as high and in many cases above the
standard limits. To assess the effect of such high noise levels on
students, we have conducted a campus-wide survey to
investigate the effects of such noise on students’ learning
experience. The survey aimed at gauging students’ perception
of noise levels at various classrooms and impact of noise levels
on students’ academic performance.
The results showed that students were able to reliably identify
the acoustic conditions that interfere with their learning. Strong
correlations (R2 = 0.87) between the background noise and
persistent noise and its influence on voice sensitivity. Correlations
between noise levels and degree of annoyance showed strong
relationship between noise levels and percentage of highly
annoyed respondents. More than 60% of the students felt that
noise interfered with their learning and had an impact on their
achievements with no gender bias, but at lower percentage for
advanced classes. In addition, the results indicate that students
were not comfortable with the noise in the classrooms including
the background noise arising from various internal as well
external sources.
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Organization:
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C.-H. Jeong, Technical University of Denmark (DTU) |
M.-B. Beldam, Saint-Gobain Ecophon |
P. Barach, Wayne State University School of Medicine |
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Next Steps in Hospital Noise Research
Ilene Busch-Vishniac
[Abstract]
Recent changes in government policies have prompted greater attention to noise in
hospitals, which has long been one of the top complaints of patients, staff, and visitors.
This has enabled significant progress in understanding how the hospital soundscape
affects staff and patients, but many questions remain. In this talk we examine the next
set of research issues to address, putting them into the context of what is currently
known. Topics to investigate include measures good at predicting human reactions to
the hospital soundscape, a greater understanding of speech intelligibility in hospitals, an
investigation of the relative importance of various sound sources in shaping the reaction
to the hospital soundscape, and a much more complete understanding of the impact of
the soundscape on medical outcomes of patients and on staff wellbeing. Of particular
note is the use of soundscape analysis techniques similar to those used in examining the
urban landscape. Recent work in this approach at hospitals has provided very
interesting results that suggest avenues for soundscape improvement and that can be
built upon to provide a much more nuanced understanding of the hospital soundscape.
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An investigation of room functions and acoustic demands in selected departments in three Danish hospitals
Thea Mathilde Larsen, Cheol-Ho Jeong, Mai-Britt Beldam, Jonas Brunskog, Christoffer Weitze
[Abstract]
Hospitals contain many different room types, each having
its own contribution to the main function of treating
patients. Different room types with different functions
demand different acoustics. Currently only a few
recommendations in terms of room acoustics are
mentioned in the Danish Building Code, BR18.
Recommendations, only to a limited degree, fulfill the
demand for a sufficient acoustic environment for staff and
patients. In this study different room types were
investigated from selected departments in Bispebjerg,
Frederiksberg and Hvidovre hospitals. Four room types
were investigated; examination rooms, control rooms,
nurse stations and lunch rooms. From interviews with staff
and questionnaires, information about functions and
acoustic demands were collected for all four room types.
Furthermore, acoustic parameters, T20, EDT, C50 and STI
were measured in the rooms. Recommendations in BR18
were found to be insufficient to meet the acoustic
demands in several of the rooms. Future
recommendations should be formulated to ensure a
satisfactory acoustic environment, which meet the
function demands in the rooms. A systematic investigation
of room functions and use of facilities in hospitals in
regards to noise is needed to be able to make such
recommendations.
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Cognitive Effects of Noise on Hospital Emergency Department Staff
Khaleela Zaman, Peter Dodds, Ning Xiang, Paul Barach
[Abstract]
Hospital emergency departments today are undoubtedly noisy,
and this noise can cause human error, adverse medication
events, and limit the ability of clinicians to provide high-quality
and safe patient care. The average sound pressure levels,
analyzed in previous studies for hospitals around the world,
substantially exceed the World Health Organization
guidelines. These sounds-emitting from sources ranging from
monitor alarms, ring binders, overhead paging, trash bins, and
echogenic surfaces, to patients crying out-can be abrupt, yet
not sustained. The problem of hospital noise requires deep
learning and insight beyond those provided by mere sound
pressure level measurements and even loudness evaluations. In
order to evaluate how noise impacts hospital staff distraction
and performance, and its subsequent propensity for human
error, we conducted study at a busy, urban hospital emergency
department. The effects on physician cognitive load and
working memory due to various sonic occurrences within the
hospital emergency department were assessed using cognitive
executive function evaluations with binaurally-augmented
acoustic environments as the backdrop. The paper will discuss
the methods deployed for cognitive testing using binaural
augmentation, share initial results, and offer meaningful
interpretations as well as potential recommendations which
address these results.
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Experimental Design to Measure the Effect of Room Acoustics on Prospective Memory of Hospital Nurses
Jikke Reinten, Ella Braat-Eggen, Maarten Hornikx, Helianthe Kort, Armin Kohlrausch
[Abstract]
A prominent cognitive aspect of a hospital nurse’s job is
to form,
imprint and retrieve intentions for prospective memory
(PM)
tasks. During the verbal change of shift report and visual
inspection of patient files, generally performed in the
nurses’
station at the start of a shift, important PM intentions are
formed
and imprinted. The correct retrieval of these intentions
during
the course of a shift is important for both patient safety
and
satisfaction. The nurses’ station can be a hectic
environment
with multiple conversations going on at the same time.
While
research has shown that the acoustic environment,
speech in
particular, has a significant influence on several
cognitive tasks,
its effect on PM is not known. The acoustic environment
of a
nurses’ station potentially influences the forming and
imprinting
of PM intentions. The current paper presents the design
of an
ecologically valid experiment with the goal of
measuring the
effect of realistic background speech and room
acoustic
conditions on the forming and imprinting of prospective
memory intentions. While the experimental design is
closely
linked to a hospital situation, understanding the effect
of
acoustic conditions on PM might also be valuable in
other
domains.
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A targeted noise reduction observational study for reducing noise in a neonatal intensive unit
Paul Barach, S Chawla, M Dwaihy, D Kamat, S Shankaran, B Panaitescu, B Wang, G Natarajan
[Abstract]
BACKGROUND: Sleep is critical to patient recovery in the hospital, hospitalization is not restful, and inpatient sleep deprivation has been linked to poor outcomes. Excessive noise in neonatal intensive care units (NICUs) can
interfere with infants’ growth, development and healing. Sound levels in many
NICUs exceed the recommended levels by the World Health Organization.
METHODS: We implemented a unit-based nursing empowerment approach
focused on noise reduction strategy in an urban, tertiary academic medical center
NICU that included baseline noise measurements. We conducted a survey
involving staff and visitors regarding their opinions and perceptions of noise
levels in the NICU. Ongoing feedback to staff after each measurement cycle was
provided to improve awareness, engagement and adherence with noise reduction
strategies. After widespread discussion with active clinician involvement,
consensus building and iterative testing, changes were implemented including:
lowering of equipment alarm sounds, designated ëquiet times’ and implementing
a customized education program for staff.
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Staff experience of sound environment in operating rooms built with non-absorbing modules
Maria Quinn
[Abstract]
Modular solutions for operation rooms made of glass and metal adversely affect the sound environment as the absorption of sound is non-existent. The requirement for Reverberation Time (T20) in operating rooms is according to Swedish Standard 25268 < 0.6 seconds. Modules without additional acoustic supplement are calculated to around 2.0 seconds.
In our research the modular ceiling was replaced with an Absorption Class A ceiling according to ISO 11654. Both normal sized rooms (65m2) and the Hybrid room (105 m2) were treated with absorbents. To gain better understanding of the sound environment we also measured Speech Clarity (C50) in addition to reverberation time (T20). All acoustic requirements were fulfilled and this paper aim to evaluate how the staff perceives the sound environment in these rooms. Information will be obtained by questionnaires sent to 78 staff at Sundsvall hospital. The hospital performs 11000 procedures a year in 15 operating rooms. Procedures contains of common surgical, gynecological and orthopedic operations, as well as hernias, cancer, ear-nose-throat and fractures. Operating times varies from short to long and staff alternates within all procedures, the unit is open during evening and night for emergencies. Questionnaire consists of both multiple-choice and open questions.
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Time-based soundscape evaluation of third-class hospital ward
Anugrah Sabdono Sudarsono, Sugeng Joko Sarwono, Aisyah Shabrina, Laudita Natasha Tamrin
[Abstract]
In Indonesia, hospital ward is classified according to the number of beds inside the ward. Third class ward is the lowest class which consist of six or eight beds inside the ward. In this study, the soundscape of third-class hospital ward is analysed on different time of the day. The analysis is conducted based on noise level measurement, sound source identification, annoying sound source identification, and semantic scales.
The noise level in the ward fluctuates between 48.3 - 60.1 dBA with the highest noise level measured during the afternoon. The dominant sound sources in the third-class ward, which are similar in the morning and the afternoon, are the sound of fan and visitor of the patients. The three most annoying sound sources in the morning are the sound of medical instrumentation, curtain, and tableware. This result is slightly different from the afternoon survey, which shows that the three most annoying sound sources are the sound of the mobile phone, curtain, and tableware. The semantic scores show a different score in the morning and the afternoon. This study shows that the soundscape in the third-class ward is changing from time to time and the soundscape intervention must consider this perception difference.
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Characterizing Community Noise in Hospital
Chiung Yao Chen
[Abstract]
The present study was conducted in the lobbies of 16 Taiwanese
urban hospitals to establish what contributes to the degree of
noisiness experienced by patients and those accompanying
them. Noise level measurements were then conducted by 15
min equivalent sound pressure levels (LAeq, 15m, dB) during
daytime hours. The average LAeq itself was found to be poorly
related to perceived noisiness. Levels variation were better
correlated, more continual noise may actually be perceived as
noisier. According to the correlation (r = 0.91, R2 = 0.83)
between objective and subjective measures, 1/ (L5 - L95) shown
to have the largest effects on perceived noisiness. These results
resemble previous studies that had assumed that a larger
fluctuation of noise level corresponds to less annoyance
experienced for mixed traffic noise. As an advanced approach,
for hospital noise that consisted of 12 audible noise events,
subjective noisiness were evaluated by autocorrelation of
measured noise with loudness and levels variation. And those
with shorter reverberation times showed lower noisiness levels
which deduced better speech intelligibility in such spaces.
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Effect of Head-Movement on Sound-Field Auditory Steady- State Response Measurements
Sreeram Kaithali Narayanan, Søren Laugesen, Valentina Zapata-Rodriguez, Jonas Brunskog, Cheol-Ho Jeong
[Abstract]
Sound-field auditory steady-state response (ASSR) is an
objective alternative for hearing aid fitting validation in infants.
For including the hearing aid in the signal path, the stimulus is
presented via a loudspeaker. In this way, ASSR is affected by the
head orientation of the test subject and the room acoustics.
Eleven normal hearing participants were tested for three static
head-orientations, defined according to the position of the
stimulated ear relative to the loudspeaker in two different
rooms. A speech modified NB CE-Chirps stimulus was used in
order to eventually force the hearing aid under test to provide
the correct gain. The rooms chosen for measurement were an
IEC listening room (T30∼ 0.5 s) and an anechoic chamber
(reference condition). A dynamic head-orientation condition
comparable to a real head-movement was simulated by
randomly combining the responses from the three static head-
orientation measurements. The results show a limited influence
of head movement on ASSR level. However, in the 4000 Hz
band, a significant reduction in ASSR level was observed when
the test ear turned away from the loudspeaker. The overall
mean ASSR level in the IEC room was reduced by 2.5 dB with
reference to the anechoic condition.
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Organization:
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M. Vigeant, Penn State University |
S. Weinzierl, Technische Universität Berlin |
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The CHORDatabase: a twenty-one concert hall spherical microphone and loudspeaker array measurement database
Matthew Neal, Michelle Vigeant
[Abstract]
Repeatability and realism are both highly important
goals in concert hall perceptual studies, but it is hard to
satisfy both simultaneously. Auralization from room
impulse response (RIR) measurements allows for
repeatable listening conditions between halls, but often
measurement loudspeakers do not realistically
represent an orchestra. To support both goals, the
Concert Hall Orchestral Research Database
(CHORDatabase) has been generated. This database
consists of 32-channel spherical microphone array
measurements in fifteen North American and six
European concert halls. First, measurements were made
at 7 to 15 seats in each hall using a three-part
omnidirectional loudspeaker, maintaining
omnidirectionality up to 5 kHz. These measurements
enable standard room acoustic parameter calculations
and higher-order spherical array beamforming
techniques. The overall variety of the database will be
demonstrated using both objective analyses. Also, a
compact 20-element spherical loudspeaker array was
used to provide realistic source directivities for RIR
measurements. This array can accurately represent the
frequency-dependent radiation patterns of different
instruments up to 3-4 kHz. Measurements were made at
20 orchestral source positions and one seat in each hall,
15 meters from stage. Full-orchestra auralization
techniques will be presented, and the future potential
of this database will be discussed. [Work supported by
NSF Award 1302741.]
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Does the method matter? A review of the main testing methods for the subjective evaluation of room acoustics through listening tests.
Daniel De La Prida, Antonio Pedrero, Luis A. Azpicueta-Ruiz, María Ángeles Navacerrada, César Díaz
[Abstract]
Listening tests are experimental procedures in which a group of
participants are asked to express their opinion about certain
questions regarding some stimuli that they have been urged to
listen to. The listening tests have proved their effectiveness as
methods for the subjective assessment of sound perception in
different fields of acoustics, and particularly in room acoustics
and sound insulation. The way in which the listening test is
conducted, how the participant’s response is expressed and
how the stimuli are presented to the participants define what it
is known as the testing/query method. There is not yet a
common methodological framework that defines which testing
method is the best for each purpose and under what conditions
the experiments are to be carried out. The purpose of this
communication is to review the most commonly used testing
methods in the subjective assessment of sound and to present
the advantages and disadvantages of each of them when they
are used for the evaluation of the subjective perception of a
sample of population.
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Loudness in different rooms versus headphone reproduction: Is there a mismatch even after careful equalization?
Michael Kohnen, Florian Denk, Josep Llorca-Bofí, Michael Vorländer, Birger Kollmeier
[Abstract]
Even though our expectation is that sound reproduced by loudspeakers in a room should exert the same perceived loudness if reproduced by individually equalized headphones, findings from the literature (e.g., Munson 1952, Fastl et al. 1985) indicate a mismatch: Equal sound pressure levels at the eardrum do not guarantee an equal perceived loudness. The aim of this study is to characterize this mismatch as a function of room acoustics and headphone type. Subjects balanced the perceived loudness between presentations of an identical stimulus from a loudspeaker and a headphone for three different rooms (anechoic, office room, reverberation chamber) and two different headphones (Sennheiser HD650 and Beyerdynamic DT770). Additionally, a comprehensive acoustic characterization of the headphones with different measurement devices was conducted.
The results confirm a loudness mismatch with a weak effect of the room type, indicating that room acoustics (and perhaps the individual expectation and the subjective spatial impression of the room) biases our loudness judgement. This is important for sound reproduction systems in rooms and the limited fidelity of room simulations even with carefully individualized headphones. The consequences of our findings and necessary steps to pin down the origin of the observed mismatch will be discussed.
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Sound image localization by bone-conducted sound
Takai Kazuki, Asakura Takumi
[Abstract]
This study aims at efficient voice-guidance technique by
using bone-conducted sound. By adopting binaural
reproduction by using the head-related transfer function,
the playback sounds for guidance are reproduced to the
users in order to indicate the traveling direction. However,
the binaural reproduction technique by the bone-
conducted sound has not been established yet
compared with the air-conducted sound. Therefore, in this
study, a sound image localization experiment was carried
out using a bone-conducted sound reproduction method
in which the intended direction of the reproduced sound
images are quickly panned to left and right in order to
enhance the perception of the sound direction. In this
study, the results of the sound localization experiment with
the panning angle of 10, 20, 30 degrees were
parametrically discussed.
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Accurate reproduction of binaural recordings through individual headphone equalization and time domain crosstalk cancellation
David Hadley Griesinger
[Abstract]
Acoustic quality of spaces of all sizes depends on methods that
can instantly and precisely compare the sound of different
seats and spaces. Complex systems using many loudspeakers
have been developed that hopefully achieve this goal. Binaural
technology offers a simpler solution. The sound pressure at a
listener’s eardrums can be measured with probe microphones,
and reproduced with headphones or speakers calibrated at
the eardrums. When carefully done the scene is precisely
reproduced. But due to the variability of ear canal resonances
such recordings and playbacks are highly individual. In this
paper we present methods that are non-individual. Our
recordings are equalized to be essentially frequency linear to
sound sources in front, giving the recording head the frontal
frequency response of studio microphones. The recordings are
played back either through headphones equalized at the
eardrum to match the response of a frontal source, or played
back with a simple, non-individual crosstalk cancelling system.
Widows apps for both methods are available from the author
upon request. We equalize headphones at an individual’s
eardrums using equal loudness measurements, and generate
crosstalk cancellation with a non-individual algorithm in the time
domain.
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Identification of the Room Characteristics Using a Spherical Microphone Array
Frederico Heloui De Araujo, Julio Cesar Boscher Torres, Fernando Augusto De Noronha Castro Pinto
[Abstract]
Microphone arrays have been used in the last decades to
identify sound sources in several environments. One of the most
used techniques is the beamforming, that can be understood as
a spatial filter for selecting sound waves coming from the
direction of interest. Nowadays, due the acquisition hardware
and the processing software capacity, it is already possible
apply the beamforming using high numbers of microphone in
the array. This condition allows, for example, use a spherical
microphone array (SMA) for monitoring all directions in space
with high resolution. This research proposes to use a SMA aiming
not only identify sound sources, but also the characteristics of
the environment, such as dimensions, baffles positions, specular
reflection coefficients and others. This will be possible with a
rotating microphone array that provides 98 measure points in a
virtual spherical surface. The processing that will be adopted
consist in applying the spherical harmonic decomposition in the
sound field around the spherical surface created by a known
sound source and using the Wavelet Transform to identify the
directions and time of arrival of the direct sound and the
reflections. It will be presented simulations and measurements in
a real environment.
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The role of median plane reflections in the perception of vertical auditory movement
Florian Wendt, Matthias Frank, Robert Höldrich
[Abstract]
Perception of auditory motion includes two principal localization subsystems. A source moving along the horizontal plane induces coherent variation in inter-aural level and time differences evaluated by the auditory system. Perception of vertical source movement on the other hand is caused by spectral variations induced mainly by the filtering of the pinna.
In real-life situations the direct sound of an emitting source is typically followed by a strong floor reflection. The physical elevation of the source determines delay time and level of this reflection providing additional information about the elevation of the source. Our contributions studies how delay alterations of a median plane reflection influence the vertical auditory movement. Depending on the excitation signal we show two partly contrasting effects dominating our perception: For broadband noise signals a delay modification between direct sound and reflection yields comb filtering and the auditory movement is explained by the pitch-height effect. In contrast, results obtained for speech signals indicate that temporal cues induced by the interference pattern of direct sound and reflection influences our perception. Respective movement directions are deduced from the movement of a physical sound source, suggesting that the auditory system learns to associate these interference patterns to sound source movements.
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Experiencing Room Acoustics Through a Library of Multichannel High-Resolution Room Impulse Responses
Wieslaw Woszczyk, David Benson
[Abstract]
A large room impulse response library being developed at
McGill University consists of more than 10,000 individual IRs
captured using multichannel microphone techniques in a large
variety of acoustic spaces. The challenge is to maximize the
learning value of this resource by maximizing the way the library
can be experienced. To accomplish this a search interface is
developed for the library that uses audio signal features, and
statistical and analytical treatment, to add rich metadata
describing perceptual qualities, acoustical parameters, and
physical characteristics. The user is able to explore 3D graphic
models of venues and audition different locations of sources
and receivers, using a variety of anechoic sources. Moreover,
an integrated convolution reverberation plugin enables the user
to employ searched spaces for live music productions and post-
production. Strict criteria are followed to maximize the sonic
quality of rendering and of the exploration experience.
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Organization:
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U. Stephenson, HafenCity Universität |
M. Ochmann, Beuth Hochschule für Technik Berlin |
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Sensitivity analysis for hybrid room acoustic simulation regarding spatial data of receiver
Philipp Schäfer, Michael Vorländer
[Abstract]
To overcome the limitations of methods using geometrical acoustics, Ray Tracing and Image Source algorithms can be combined with well-known numerical approaches - such as Finite Element Method (FEM) or Boundary Element Method (BEM) - which consider wave-based effects and therefore produce more sophisticated results in the lower frequency range. For this purpose, a room is modeled for both methods and then simulated separately. In a second step, the overall result can be received by uniting the distinct results in the frequency domain. A challenging aspect of room acoustic simulations is the acquisition of input data such as position and orientation of (binaural) receivers. The question arises how accurate this data has to be measured for the purpose of room acoustic simulations. Thus in the present work, a sensitivity analysis is conducted to determine the influence of deviations in spatial data of receivers on the simulation result.
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Preliminary results of scattering surface modeling and perceptual aspects in wave-based acoustic simulations
Louena Shtrepi, Brian Hamilton, Arianna Astolfi, Marco Masoero
[Abstract]
The level of detail of a simulated model as well as the assignment of the materials acoustic properties have been largely debated and optimal guidelines have been determined based on the approximations of the scattering algorithm of the simulation tools. These aspects are of great importance when investigating the differences between geometrical-acoustic (GA) based and wave-based methods. To this aim the present study refers to objective and subjective results comparisons of wave-based simulations (FDTD- finite difference time domain method) in a shoebox concert hall, which has been previously studied through GA-based methods. Three models, that consider a 1) reflective, 2) low scattering, and 3) high scattering conditions of one of the long lateral walls, have been simulated in order to determine the conventional acoustic parameters such as early decay time (EDT), reverberation time (T30), clarity (C80), definition (D50), and interaural cross correlation (IACC). The differences of the GA-based and wave based simulation results have been outlined in terms of simulation accuracy. Furthermore, a preliminary subjective investigation has been performed in order to determine the sensitivity of listeners to the surface diffusivity variations in different listening positions.
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Numerical simulations of Italian Opera Houses using geometrical and wave-based acoustics methods
Dario D'Orazio, Giulia Fratoni, Anna Rovigatti, Brian Hamilton
[Abstract]
The acoustics of Opera houses represents an interesting topic in
scientific literature due to their architectural and morphological
characteristic traits (shape, materials, coupled volumes). A way
to exploit the outstanding value of Italian theatres acoustics is to
perform different typologies of music while adding or removing
acoustically significant elements on the stage. Numerical
simulations may be used as useful tools for designing distinct
layouts and for optimising such performances. The present
paper investigates strengths and weaknesses of two different
room acoustics modelling approaches: geometrical acoustics
(GA) and wave-based methods. The commercial GA software
ODEON and a hybrid ray-tracing and finite-difference time-
domain (FDTD) approach are applied in this work.
Measurements carried out in a sample of Italian Opera houses
provide reference data for 3D virtual models calibration.
Preliminary comparisons between room acoustical simulation
methods are shown and discussed.
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Combining Image and Equivalent Sources for Room Acoustic Simulations
Boris Mondet, U. Peter Svensson, Jonas Brunskog, Cheol-Ho Jeong, Claus Lynge Christensen, Jens Holger Rindel
[Abstract]
Room acoustic simulation tools based on geometrical acoustics
fail to predict accurately the behavior of rooms at low
frequencies. This inaccuracy can lead to poor acoustic
conditions especially in rooms of moderate volume, such as
classrooms and meeting rooms. The present paper tackles this
problem by introducing a simulation technique for room
acoustics based on the Image Source Method and the
Equivalent Source Method. These methods are implemented in
the time domain to achieve efficient calculations with reliable
results for low to medium frequencies. Given a source emitting a
pulse signal in a room, image sources up to different orders are
defined to determine early reflections. With the direct sound
field from the original source and the contributions of the image
sources, the remaining part of the sound field is solved by the
Equivalent Source Method to account for late reflections and
diffraction effects. The results of the test simulations carried out
show that this combined Image-Equivalent Source Method is a
suitable option for room acoustics.
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Study of DIF Boundary Model with Rectilinear FDTD Scheme in Voice Booths
Vito Romanelli Tricanico, Marcel Borin, Carolina Monteiro, Rânnely Araújo
[Abstract]
Computational simulations are helpful when studying acoustical properties of rooms. This represents a challenge when very small rooms, such as office video/phone booths, are taken into account. The volume of these rooms generally does not exceed 15 m3 and the first modes occur at higher frequencies if compared to larger rooms. Computational room acoustics considers two main techniques of modelling sound propagation: geometrical and wave-based. As geometrical modelling is not the most suitable approach for low frequencies since the phase performs an important part in the physics at this frequency range, in the present study, a rectilinear FDTD scheme routine was implemented to obtain the impulse responses for low frequency bands of a voice booth, using proper boundary conditions derived from digital impedance filters (DIF). In order to obtain wideband impulse responses, high frequency impulse responses were acquired using a commercial software based on geometrical methods. Both responses were later filtered with low-pass and high-pass filters based on Schroeder frequency, and then assembled. The results were compared with data obtained from field measurements and, finally, listening tests were carried out to compare stimuli obtained from auralized and recorded impulse response of the room.
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Validation of ray-tracing semiclassical (RTS) low frequency acoustic modelling for rooms with curved boundaries
Rok Prislan, Daniel Svenšek
[Abstract]
The acoustic ray-tracing semiclassical (RTS) method is a frequency domain phased geometrical method which directly reproduces the acoustic Green’s function. It has been previously demonstrated that the method is capable of modeling the lowest modes for the case of local- and extended-reaction boundary conditions in a rectangular room. In the presented study we validate the RTS method for the case of curved boundaries, where the density of paths does not obey the free field dependence appearing with flat or piecewise flat boundaries. The RTS numeric implementation was upgraded to propagate two additional neighbouring rays. Those are emitted from the same source as the original ray but in slightly different directions. The divergence of the rays is used to model the (de)focusing effects that are present due to the reflections on the curved boundaries. The behavior of the RTS method for a representative test case of a room confined with curved weakly absorbing boundaries is presented. Results are compared to those obtained by the FEM and analyzed in relation to the absorbing properties of the boundary. Several interesting peculiarities and limitations of the method are presented. The obtained results are important for validating the RTS method in more general room geometries.
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Lower Bound on Frequency Validity of Energy-Stress Tensor Based Diffuse Sound Field Model
Aidan Meacham, Roland Badeau, Jean-Dominique Polack
[Abstract]
A lower bound on the frequency validity limit is established for an energetic wave equation derived from the energy-stress tensor, examined in the one-dimensional case [Dujourdy et al, Acta Acustica united with Acustica 103:480-491, 2017]. The method efficiently models diffuse sound fields that dominate reverberation at higher frequencies and larger distances. Initially noted in the course of an exhaustive search of the solution space of all valid model parameters, the low-frequency cutoff has implications for the utility of the method in a hybridization context. In practice, the bound is encountered when determining the absorption and diffusion coefficients by iteratively approaching the temporal and spatial decay of measured data. As the test frequency decreases, the ranges of coefficient combinations that result in less than 10% variation from each decay measure become increasingly thin until the region where both measures are satisfactory (the intersection of the two domains) disappears. Further evidence for the bound is provided through comparison with measurements of a long hallway, and stability concerns in the cases where both coefficients are very small are addressed.
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Single versus multi-domain analysis in diffusion equation modeling for sound field analysis of distinct room shapes
Zuhre Su Gul, Erinc Odabas, Mehmet Caliskan
[Abstract]
Diffusion equation modelling has gained popularity
recently in the application of room acoustics analysis mostly due
to its computational efficiency. The method has been used in
proportionate rooms confidently, while some research has
highlighted its applicability in disproportionate rooms
as well. In this study specifically, coupled
rooms and long enclosures that cannot be very well defined by
statistical theory, or under-estimated by ray tracing are studied
by DEM analysis. The diffusion coefficient, dependent upon the
volume and surface absorption area of individual domains, is
one major indicator in this analysis. Defining domains in coupled
volumes or in coupled long rooms is an important issue. The
limits where the domains can be
considered as a single space or as a multi-domain system are
quite vague and necessitate a systematic investigation. Some
real structures including multi-volume monuments and two
tracked subway stations are examined in that respect, by
applying DEM in a finite element medium. The results of this
investigation are assistive in regards to the coupling factor limits
when different volumetric and aperture size relations are
established. The pre-knowledge of applying either a single
domain or a multi-domain solution in a specific combination of
volumes will augment the speed of analysis.
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An Energy Model for Calculating Room Acoustic Parameters in Rooms with Absorbent Ceilings
Erling Nilsson
[Abstract]
An energy model has been developed for calculation of room acoustic parameters in rooms with absorbent ceiling. The model is based on a SEA approach to handle the non-diffuse conditions that often appears in this type of rooms. Since, in practice, an absorbent ceiling is the most common acoustical treatment in ordinary public rooms the non-diffuse conditions are more a rule than an exception. The purpose of the model is to get an estimation of room parameters related to reverberation time, speech clarity and sound strength, taking into account the non-uniform distribution of the absorbent material. The sound field is subdivided into a grazing and non-grazing segment referring to the incidence angle towards the ceiling absorber. A decay constant for each segment is defined as well as a representative number of modes. The scattering effect of furniture and other interior fittings is interpreted as the energy transfer from the grazing to the non-grazing group. Special circumstances at grazing incidence and low frequencies is considered.
The theoretical considerations behind the model will be presented as well as comparison between measurements and calculations
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Pertinence of a Simplified Plane Wave Model for Reverberation Energy Decays in Rooms with a Pair of Parallel Surfaces.
Jean-Jacques Embrechts
[Abstract]
An analytical model has been developed recently to take into account specular and diffuse reflections in rooms with a pair of parallel surfaces. Analytical models can help to better understand the influence of scattering coefficients’ values in those rooms in which unusually long reverberation times can be created by locked-in rays. Also, analytical models can be used complementary to ray-tracing methods to quickly estimate the late part of reverberation decays. The developed model is based on the formalism of sound particles emitted in the room by a point isotropic source. Some approximations lead to an intuitive analytical model including the influence of absorption and scattering coefficients on reverberation. The purpose of this paper is to develop and analyse the pertinence of an additional simplification to the model, based on the assumption of plane waves propagating between the two parallel walls. In this case, the model predicts a reverberation in the room composed of two exponential decays. The pertinence will be analysed by comparing this simplified plane wave model with reverberation decays computed by a sound ray program.
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Dispersion-Reduced Time Domain FEM for Room Acoustics Simulation
Takeshi Okuzono, Kimihiro Sakagami, Otsuru Toru
[Abstract]
The present paper gives an overview of the applicability of
an implicit time-domain FEM (TD-FEM) for room
acoustics simulation, which has fourth-order accuracy in
both space and time. First, the algorithm of TD-FEM
is presented with explanations of dispersion reduced finite
elements for spatial discretization and a time
integration method for time discretization. The theoretical
dispersion error property is also shown. Use of the
dispersion-reduced scheme has advantageous in terms of
accuracy, stability and convergence of iterative
solver, which are directly related to computational cost.
Then, the applicability of TD-FEM is demonstrated
through two numerical examples on room acoustics
simulation at kilohertz frequency range, which are
acoustics simulation of a simple-shaped concert hall and
simulation of reverberation absorption coefficient
measurements. In the numerical examples of concert hall
analysis, a comparison with a conventional TD-
FEM shows the efficiency of the dispersion reduced TD-FEM.
Modeling accuracy of non-locally reacting
permeable membrane sound absorbers for the TD-FEM is
presented through the simulation of reverberation
room method.
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Comparison of sound pressure levels and reverberation times computed by the boundary element method and the radiosity method
Christian Steuck, Uwe Stephenson
[Abstract]
Boundary Element Method (BEM) and Radiosity Method (RadM) are established methods for numerical sound field computation. The BEM yields sound pressure amplitudes mainly for low frequencies, whereas for high frequencies geometrical methods are more efficient: ray and beam tracing and, well known in illumination simulation, the RadM. The latter, in contrast to the BEM, is based on an energy balance described by Kuttruff’s integral equation. A comparison of BEM and RadM is interesting as both are based on a very similar data structure: a subdivision of the room’s surface into a large number of patches and a matrix of factors describing the geometrical relationships between them. While the BEM requires surface impedances, the RadM uses absorption degrees, assuming perfectly diffusely reflecting walls and yields, in its stationary version, the illumination distribution and from that the local sound energy densities and levels. Its iterative solution in time domain yields energy decay curves and thus reverberation times. With the BEM, these are computed by inverse Fourier Transform. A set of rooms with different local distributions of the absorption coefficient is examined, the results of both methods are compared with regard to sound pressure level and reverberation time.
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Numerical Simulation Round Robin of a Coupled Volume Case as Compared to Scale Model Measurements
Antoine Weber, Brian F. G. Katz
[Abstract]
The advantages and limitations of the principle numerical methods
used in room acoustics have been primarily evaluated in single-volume
room conditions, placing the emphasis on early reflections and the
early part of the room acoustic impulse response. Few studies have
examined the capabilities of numerical simulations to model correctly
the case of coupled volumes, where the late part of the impulse
response needs to be accurately represented. This work presents the
preliminary results of a recent round robin type study comparing
coupled volume theory and numerical simulation results with
measurements carried out in a physical scale model. Numerical
methods include geometrical acoustic solutions with several ray-tracing
approaches and wave-based methods, including FDTD
implementations. A scale model was used as a reference (corrected to
1:1 scale) as this physical method allows for observation of sound
fields in coupled spaces, reproducing the acoustic energy transport
through coupling surfaces between volumes. All room model
coordinates were provided to participants. To avoid issues regarding
variations in implementation of material definitions and scattering
behaviors across different methods, the acoustics of the individual
volumes was prescribed, through the reverberation time, in the
uncoupled condition. The volumes were then coupled and the results
analyzed.
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Open-source platforms for fast room acoustic simulations in complex structures
Matthieu Aussal, Robin Gueguen
[Abstract]
This article presents new numerical simulation tools, respectively developed in Matlab and Blender softwares. Available in open-source under the GPL 3.0 license, it uses a ray-tracing/image-sources hybrid method to calculate the room acoustics for large meshes. Performances are optimized to solve problems of significant size (typically more than 100,000 surface elements and about a million of rays). For this purpose, a divide and conquer approach with a recursive binary tree structure has been implemented to reduce the quadratic complexity of the ray/element interactions to near-linear. Thus, execution times are less sensitive to the mesh density, which allows simulations of complex geometry. After ray propagation, a hybrid method leads to image-sources, which can be visually analyzed to localize sound map. Finally, impulse responses are constructed from the image-sources and FIR filters are proposed natively over 8 octave bands, taking into account material absorption properties and propagation medium. This algorithm is validated by comparisons with theoretical test cases. Furthermore, an example on a quite complex case, namely the ancient theater of Orange is presented.
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Estimation of the background noise levels in large atria with known room acoustic properties and function
Richard Šimek, Vojtech Chmelík, Jarmila Húsenicová, Monika Rychtarikova
[Abstract]
One of the major identified acoustic problems in atria is
continuous unpleasant background noise typically caused
by long reverberation of sound. Levels of background noise
caused by internal sound sources depend on activity
performed in the place on one hand and on room acoustic
conditions on the other hand. Present noise can be relatively
easily measured and evaluated by means of statistical noise
analysis that can be expressed e.g. by histogram of noise
levels. However, the prediction of statistical noise levels, in
rooms is very difficult, as it doesn’t depend only on the overall
sound power spectrum of sound source, but also on the time
varying character of sound. Function and thus activity
performed in an atrium plays therefore an important role in
terms of acoustic comfort analysis and without the
information on time variation in source signal, parameters
such as L90 or L5 cannot be successfully predicted. In this
paper several typical signal sounds are synthesized and
auralized in Odeon software in parametric study of an atrium,
followed by statistical analysis of predicted auralized wave
files. Correlation between the activity and room acoustic
parameters is shown and estimation of background noise
level L95 and L90 is shown.
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Boundary element methods for acoustic simulations in the time domain
Heiko Gimperlein
[Abstract]
The talk discusses recent progress for the time-dependent simulation of sound emission and sound scattering using boundary element methods. We particularly consider methods adapted to complex and singular geometries or short impulses. Stable numerical formulations of the relevant equations are discussed, and they are exploited to compute the sound reflection depending on the absorption and impedance. We discuss the performance of our methods in applications such as tire noise.
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Acoustics inside a gypsum sphere with 7 m of diameter
Ernesto Accolti, Fernando Di Sciascio
[Abstract]
Concave surfaces are known to focalize or diffuse sound depending on the curvature and the source position regarding the surface. Spherical rooms are frequently found as theoretical cases in the literature. In this work, results from models and measurements of the acoustics inside a 7-m-diameter gypsum sphere are compared.
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Organization:
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M. Rychtarikova, KU Leuven, Faculty of Architecture |
L. Kritly, KU Leuven, Faculty of Architecture |
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Calculating the Speech Transmission Index in fluctuating noise: a data-driven approach in the short-term implementation
Nicola Prodi, Chiara Visentin
[Abstract]
Everyday communication takes place in the concurrent presence of reverberation and background noise; the latter may have a fluctuating character and a speech-like spectrum, being for instance the result of multiple speakers talking together in the background (i.e., babble noise). The objective characterization of these listening conditions can be achieved by using a time-frame implementation of the Speech Transmission Index (STI) in the indirect scheme, named eSTI. One prerequisite of using the method is that the optimal time frame has to be determined.
In this study, an experimental approach was used to determine the optimal time frame, defined as the one that provides coincident psychometric curves under stationary and fluctuating background noises. Matrixed-word listening tests were presented to 79 young adults with normal hearing. The speech reception task was presented under 28 listening conditions, created by varying signal-to-noise ratio, reverberation and noise type. By comparing the psychometric curves for the two noises, an interval of suitable frame durations was identified, ranging between 200 and 345 ms. Using a time frame within this interval thus ensures that the same eSTI value corresponds to the same predicted intelligibility, irrespective of the noise type.
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Multidimensional Visual Cluster Analysis of Room Acoustical Parameter Values as Means to gain Scientific Insights and Design / Consulting Tool
Klaus-Hendrik Lorenz-Kierakiewitz, Benjamin Pfändner, Christoph Reuter, Stefan Ostrowski
[Abstract]
At the conference DAGA 2017, a method of multidimensional visual analysis of basic room acoustical parameter values RTMid, strenght GMid and volume V was presented. Using this method, it is possible to demonstrate, that different room types as e.g. concert, opera or chamber music halls shape separate clusters in partial volumes of the space spanned by the acoustical parameters. For concert and opera halls it has been shown that halls rated regarding the acoustical quality as ”good” are localized in narrow partial volumes within the centers of the room type clusters. At the DAGA2018, this approach was extended by further room types and by comparisons of the spatial positions of several halls before and after renovations.
In this contribution, the benefit of this analysis method will be shown as a valuable tool to obtain new scientific insights as well as a helpful planning tool: For the room type ”concert hall” the historical development is analyzed from the first examples to the 21st century yielding characteristical trajectories across the parameter spaces clarifying the evolutionary trends as well as geographical differences. The immediate advantage of the method described as consulting and design tool will be demonstrated by application to orchestral rehearsal rooms.
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Acoustics of Multipurpose Halls in Croatia
Marko Horvat, Kristian Jambrošić
[Abstract]
In the modern-day era, halls are usually built to host a variety
of events that can range from movie projections, congresses
and meetings, theatre plays, sporting events, music
performances in various genres, etc., all of it with or without
sound reinforcement. Each type of event requires specific
acoustical conditions in the hall, which makes the acoustics
of a multipurpose hall a challenge to be dealt with mostly
by making compromises. In smaller communities, there is
often only one hall large enough to accommodate all the
visitors interested in attending an event, usually a cultural
centre or a sports hall of a school. However, the trend of
building multipurpose halls is present in cities as well, in order
to use such spaces as much and as often as possible, to
maintain their self-sustainability. To address the acoustics of
multipurpose halls in Croatia and make an overview, this
paper gives a comparison of basic acoustical parameters of
20 different halls. The parameters were either measured as a
way of acoustical analysis, or simulated during the acoustic
design process. The acoustics of these halls is analysed with
regard to their size, intended use, and the values of relevant
acoustical parameters.
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Acoustics features of sports facilities on the example of FIFA 2018 football stadiums in Russia
Anton Peretokin, Anatoly Livshits, Alexey Orlov, Natalia Shirgina
[Abstract]
The FIFA World Cup 2018 an international football
tournament took place in Russia at 12 stadiums in 11
cities: Kazan, Kaliningrad, Moscow, Nizhny Novgorod,
Rostov-on-Don, Saint Petersburg, Samara, Saransk, Sochi,
Volgograd and Yekaterinburg. Some stadiums were
greenfield projects, others were adapted to FIFA
requirements. Acoustic parameters as reverberation time
and speech transmission are standardized for FIFA
stadiums. After the tournament is finished, the stadiums
are to be used both for football games and commercial
purposes: concerts and shows with the use of PA Systems.
If stadiums are used as concert venues, their acoustics
should be high.
An acoustic survey of the stadiums was carried out.
Acoustic parameters were measured in unoccupied
stadiums. Recalculation of occupied stadiums was made in
computer simulation models. The measured values of
acoustic parameters were analyzed and compared with
FIFA acoustic requirements. Positive and negative acoustic
characteristics of the stadiums were determined. Acoustic
characteristics were compared with architectural features.
Recommendations were made on how to improve stadium
acoustics when design and build new stadiums.
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Perceptual comparison of measured, modeled and simulated OBRIRs for virtual human echolocation
Annika Neidhardt, Christian Schneiderwind
[Abstract]
Oral binaural room impulse responses (OBRIRs) describe the
transfer path from a person’s mouth through the room to the
ears of the same person. OBRIRs can be used to realize human
echolocation scenarios in virtual acoustic environments for
headphone based reproduction. In this paper different
approaches to create OBRIRs for the case of a single reflecting
surface close to a person’s head in an anechoic environment
are studied regarding their perceptual deviations. This includes
OBRIRs measured with a head, torso and mouth simulator,
OBRIRs simulated with a raytracing-based toolbox and OBRIRs
generated with a spherical head model with a circular piston. A Repertory Grid experiment was conducted with 20 mostly
naïve listeners to find suitable constructs for evaluating the
audible differences. In this investigation the participants had to
work with given speech instead of talking themselves to focus
on the effect of the different approaches. A detailed analysis of
the results is provided.
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Echoidentification: Using reflected sound to identify objects and their characteristics
Dawn Anderson, Sarahelizabeth Baguhn
[Abstract]
People who are blind and visually impaired use a wide variety of sensory information to understand the world around them. Hearing is a particularly useful sense because of the range. Many persons who are blind or visually impaired use some form of echolocation to monitor the space around them. The sound of a cane tip reflected off a wall is a common way to keep a straight path without trailing.
Orientation and Mobility Specialist teach clients to make a crisp clicking sound, and attend to the reflections of that sound. Students with varying levels of hearing have demonstrated the ability to discern a difference in sound to identify the presence of an object. Clients can be trained to go beyond just locating the objects reflecting sound. The relative size and distance to the object, its overall shape, as well as its hardness can be discerned. This process of identifying the properties of an object may be referred to as echoidentification. This skill is very useful in identifying where doorways are while walking along a city block. The leafiness of a bush sounds different than the wood of bench.
A demonstration of echoidentification will be part of this presentation.
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Organization:
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B. Schulte-Fortkamp, Technische Universität Berlin |
A. Fiebig, TU Berlin |
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Soundscape - from Noise reduction to Perception-based Decisions for Livable Environments
Brigitte Schulte-Fortkamp
[Abstract]
Soundscape is about serving the economy, community, and the environment to enhance the livability in urban areas particularly with regard to the acoustic environment. Soundscape analysis and application is a proven method of providing an improved acoustic environment for urban dwellers, now also addressing a significant portion of the smart growth agenda. As such, it requires a truly interdisciplinary process. The new ISO/DTS 12913-2 Acoustics - Soundscape - Part 2: Data collection and reporting requirements provides evaluation processes and information about participation of stakeholders, based on years of discussion focused on integrating the people concerned into urban planning processes. This paper will discuss the participation processes that follow the new TS and that contributed to its development and standardization. In addition, discussion will focus on the need for holistic, interdisciplinary approaches and platforms within the planning of urban areas, allowing the direct contributions of scientists and practitioners along with the people directly concerned.
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Soundscape standardization dares the impossible – Case studies valuing current soundscape standards
André Fiebig
[Abstract]
Recently, the ISO/TS 12913-2, the second part of the ISO soundscape series, was published supporting detailed information on data collection and reporting for studies using the soundscape approach. This part includes guidelines for soundwalks, interviews, acoustical measurements and defines minimum reporting requirements. The conceptual framework and soundscape related definitions are provided by the ISO 12913-1. Finally, part 3 of ISO 12913 is underway addressing aspects and needs of the data analysis.
This ISO soundscape series provides a foundation for communication across disciplines and offers guidance on how to perform soundscape investigations, but at the same time the question could be raised whether a concept asking for an holistic, context-sensitive view can be subject to standardization at all. The ISO working group had to address this issue in detail. The paper will reflect on the apparent contrast between identifying a common methodological basis of soundscape studies for standardization and the general holistic requirement of the concept soundscape. Based on case studies, views on the apparent conflict are discussed and the value of soundscape standards harmonizing the data collection process is assessed by considering the outcome of soundscape case studies.
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A pilot study on the analysis of soundwalk data
Koji Nagahata, Tomoki Manabe, Shuto Okada
[Abstract]
In soundscape studies, it is quite important to reveal the relationship between qualitative data and acoustic data. The annex of ISO/TS 12913-2 proposed to ask the soundwalk participants to list sound sources. Also, binaural recording is required to get acoustical data. In this study, a simple method to express the relationship between perceived sound sources and acoustic data was explored. In addition, a method to analyze speech data obtained at the discussion after the soundwalk was
examined.
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The Implementation of Acoustic Environment Simulator to Improve The Soundscape of Iconic Space
Ranti Dwi Tassia, Anugrah Sabdono Sudarsono, Sugeng Joko Sarwono
[Abstract]
In general, soundscape study conducted by letting the participants to have the experience of the actual soundscape. This method usually conducted using in situ soundwalk or soundscape reproduction. Soundscape composition has become one of the methods to understand the soundscape of urban area based on human memory. In this study, the concept of soundscape composition with acoustic environment simulator has been implemented to understand and improve the soundscape of an iconic space. Two experiments were conducted in this study; first, the soundscape composition of Bandung’s City Square (the existing condition); second, the improvement of the soundscape of Bandung City Square. This study has identified the soundmarks of Bandung’s City square. The soundmarks are the sound of children, the sound of adhan, conversation, the sound of people sweeping the floor, and the sound of footsteps. The sound of traffic is used in both conditions to represent the urban area as the participant tent to keep this sound in the improved composition. This study also has determined the sound addition which can be used to improve the sound environment quality (the addition of birds chirping and the water). In conclusion, soundscape composition successfully identify and improve the soundscape of the iconic area.
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Classifying urban public spaces according to their soundscape
Kang Sun, Karlo Filipan, Francesco Aletta, Timothy Van Renterghem, Toon De Pessemier, Wout Joseph, Dick Botteldooren, Bert De Coensel
[Abstract]
Cities are composed of many types of outdoor spaces, each with their distinct soundscape. Some of these soundscapes can be extraordinary, others are often less memorable. However, most locations in a city are not visited with the purpose of experiencing the soundscape. Consequently, the soundscape will not necessarily attract attention. Existing methods based on the circumplex model of affect classify soundscapes according to the pleasure and arousal they evoke, but do not fully take into account the goals and expectations of the listener. Therefore, in earlier work, a top-level hierarchical classification method was developed, which distinguishes between spaces based on the degree to which the soundscape creates awareness of the acoustical environment, matches expectations and arouses the listener. This paper presents the results of an immersive laboratory experiment, designed to validate this classification method. The experiment involved 40 participants and 50 audiovisual recordings drawn from the Urban Soundscapes of the World database. It is shown that the proposed classification method results in clearly distinct classes, and that membership to these classes can be explained well by physical parameters, extracted from the acoustical environment as well as the visual scene.
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A soundscape study in New York. Reflections on the application of standardized methods to study everyday quiet areas
Antonella Radicchi
[Abstract]
In 2014 and, more recently, in 2018 the ISO norms on
soundscape were released with the aim of providing a
conceptual framework and standardized data collection
methods for the international community of scientists and
professionals involved in the study of soundscape. This paper
presents a soundscape study conducted in New York where,
following the ISO norms on soundscape, methods are applied
to study everyday quiet areas. Firstly, the paper introduces
the research questions of the soundscape study. Secondly, it
outlines the procedure and methods of the study, consisting
of on-field data collection through soundwalks and the use of
the Hush City app. Thirdly, it discusses initial results and
limitations of the study. In conclusion it reflects on the
application of standardized methods to study everyday quiet
areas in densely populated, large cities, such as New York.
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Soundscape reproduction using headphones for web-based listening test
Anugrah Sabdono Sudarsono, Sugeng Joko Sarwono
[Abstract]
One of the methods to analyse a soundscape is by reproducing the soundscape. The acoustic environment reproduction usually conducted using multi-speaker systems in a laboratory. The experiment usually conducted individually resulting limited respondent of the experiment. A web-based experiment is one of the alternatives to do the soundscape study. By using the internet, the experiment can be done easier with more respondents.
The experiment using the internet usually conducted using an online questionnaire which can play a stereo sound or a video. In this study we are testing the validity of XY, Mid-Side, and Head Tracking decoders to reproduce acoustic environments using headphones. The XY and Mid-Side decoders were used to decodes B-Format signals to stereo signal which can be played directly by the online questionnaire. The head tracking system is used to represent the experiment done using Youtube 360 of Facebook 360.
Analysis based on semantic score indicates that the listening test conducted using stereo decoders can gives the similar score with the listening test conducted using ambisonic reproduction system. Further analysis using semantic differential analysis shows that the experiment conducted using Mid-Side decoders can deliver the same soundscape dimensions with the in-situ experiment.
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A review of regression analysis methods: Establishing the quantitative relationships between subjective soundscape assessment and multiple factors
Ming Yang
[Abstract]
Soundscape research has long been exploring factors/indicators that would impact
human perception, assessment, behaviour, health, well-being, etc. Since soundscape is
intrinsically a complex system, a wide variety of factors have been suggested by
intensive studies and also ISO/TS 12913-2 standard, ranging from acoustics,
psychoacoustics, sound source composition, to demographics, and other
personal/social/cultural factors. To examine subsequently the quantitative relationships
between such indicators and soundscape impacts/descriptors, data statistical methods
are essential, which include a wide range that correspond to different specific research
tasks, e.g. correlation analysis, analysis of variance, factor analysis and cluster analysis.
While soundscape is affected by not only any single factor but multiple factors
simultaneously and interactively, this paper focuses on the statistical methods that can
investigate multiple variables to study the relationships and predict soundscape
descriptors from multiple possible indicators, e.g. regression analysis, fuzzy logic model
and others that have been commonly used in soundscape studies. It provides a brief
review of the methods and their variations/derivations for analysing the different types
of variables/factors, i.e. ratio/interval, ordinal and nominal, exemplified by a number of
effective prediction models from previous studies on subjective soundscape assessments
(e.g. soundscape quality, annoyance, and pleasantness) - a crucial part in soundscape
research.
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Organization:
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J. Hong, Nanyang Technological University |
T. Oberman, University College London |
H. Xie, Chongqing University |
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Towards soundscape indices
Jian Kang, Francesco Aletta, Tin Oberman, M Erfanian, M Kachlicka, M Lionello, A Mitchell
[Abstract]
The growing field of soundscape studies considers sound environments as perceived, in context, with an interdisciplinary approach. This paper outlines an ongoing European Research Council (ERC) Advanced Grant project, which aims to establish ”soundscape indices” (SSID). By taking psychological, (psycho)acoustical, neural and physiological, and contextual factors into account, SSID will adequately reflect levels of human comfort to integrate (and eventually replace) decibel-based metrics commonly used in existing (international) regulations, shifting the focus from noise control to a more holistic approach. Steps to achieve this include: to characterise soundscapes, by capturing acoustic environments and establishing a comprehensive database; to identify key factors and their influence on soundscape quality based on the database, by conducting laboratory psychological evaluation, acoustical/psychoacoustic factors analysis, and also, to research the neural and physiological underpinnings of soundscape perception, by applying techniques such as functional magnetic resonance imaging (fMRI) and Skin Conductance Response; to develop, test and validate the soundscape indices, through analysing the influences of various factors; to demonstrate the applicability of the soundscape indices in practice, by establishing frameworks for soundscape prediction, design, and standardisation. Ultimately, the findings of SSID will allow for easy assessment of public spaces and the increase of the noise management impact.
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Prediction models of desirable levels of birdsong and water sound in a noisy environment: A laboratory experiment based on virtual reality
Zhen-Ting Ong, Jooyoung Hong, Bhan Lam, Kenneth Ooi, Woon-Seng Gan, Samuel Jeong, Irene Lee, Tse Tiong Tan
[Abstract]
Augmenting pleasant natural sounds in the noisy urban
environment is a key strategy in soundscape design. There have
been numerous studies on the positive effects of natural sounds
on soundscape quality. However, little attention has been
directed to predictive models that suggest appropriate levels of
natural sounds at specific ambient noise levels. These models
provide a blueprint for practical soundscape design. This study,
aims to develop prediction models of desirable natural sound
levels to enhance the soundscape quality through laboratory
experiments based on virtual reality. The laboratory test consists
of two sessions (I and II). In session I, participants were instructed
to evaluate traffic sound scenes, ranging from 55dB to 76dB, in
terms of perceived loudness of noise (PLN) and overall
soundscape quality (OSQ). In session II, participants were
instructed to adjust the audio levels of bird and water sounds
augmented to each traffic scene to the most desirable levels.
At the end of each adjustment, participants were instructed to
assess PLN and OSQ for the combined sounds composed of
natural sounds mixed with target traffic sounds. Based on the
results, soundscape predictive models were developed using
acoustic indicators to predict desirable natural sound levels
corresponding to traffic noise levels.
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Evaluation of preferred levels of natural sounds in-situ environment through an augmented reality device
Bhan Lam, Jooyoung Hong, Zhen-Ting Ong, Kenneth Ooi, Woon-Seng Gan, Samuel Jeong, Irene Lee, Tse Tiong Tan
[Abstract]
Many studies have investigated the effect of natural sounds on
improving soundscape quality under controlled laboratory
conditions. The laboratory settings are, however, limited in
reflecting real-life settings, resulting in low ecological
validity. In-
situ experiments can provide a realistic representation of the
real-life settings, which can guarantee high ecological validity.
Recently, augmented reality (AR) technology provides an
avenue for virtual augmentation of visual and audio elements in
the real environment. Therefore, this study investigates the
preferred levels of two types of natural sounds (bird and stream)
in a noisy in-situ environment using an AR head-mounted device
(HMD). Three locations in an outdoor residential area, with
background traffic noise levels varying from 60 to 70 dBA, were
selected for the in-situ experiment. Participants were instructed
to adjust the volume of the natural sounds playing via the
loudspeakers of the AR HMD to their most preferred levels at
each location. At the end of each volume adjustment,
participants were instructed to evaluate the perceived loudness
of noise and soundscape quality. Based on the results of the
experiment, the effects of types of natural sounds and
background noise levels on determining the preferred levels of
natural sounds were examined.
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Passenger's Train Soundscape: Identification of Activities
Nurul Hidayah, Gumilang Paramarta Saniskara, Nurfitriana Muharami, Keysha Wellviestu Zakri, Anugrah Sabdono Sudarsono, Sugeng Joko Sarwono
[Abstract]
Passenger’s train is an environment where people are exposed
by mechanical sound and the sound of human activity. The
passengers need to be in the same acoustic environment for a
period of time, so a comfortable environment is a necessity.
Soundscape intervention can be applied in this case with the
consideration of passenger’s activity. This study tries to identify
the activities in a passenger’s train with 9 hours (07.20-16.20)
journey.
There are several activities identified in the train: sleeping,
eating, going to the toilet, and using mobile phone. In general,
when the passengers are not sleeping, they are using mobile
phone. The passenger tends to sleep during three periods of
time. The eating activity happens once after the first sleeping
period. The passenger’s going to the toilet twice during the
journey. First, after two hours of journey and after 5 hours of
journey. The study has identified the passenger’s activity during
nine hours train journey. The result of this study can be beneficial
to design acoustic environment scenario which considers
different activity.
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The relationship between activities and human perception in urban area
Andini Hapsari, Helga Salim, Anugrah Sabdono Sudarsono, Sugeng Joko Sarwono
[Abstract]
Human activity in urban area can be affected by various
factors such as the sound sources, noise condition, and
perception in that area. This study tries to analyse the
relationship between human activity and the perception
expectation of urban area. The experiment was conducted
using an acoustic environment simulator. In this study, the
participants were requested to compose several soundscape
compositions of urban area which is suitable to do certain
activity. The composition must represent the environment for
doing individual activities (e.g. reading and relaxation) and
social activities (e.g. talking and playing with children). Then, the
participants were asked to rate their compositions according to
the perception of comfort, dynamic, and communication.
The result shows that the perception’s rating from two different
activities is significantly different. The participants need to have
a comfortable soundscape for doing both individual and social
activities. However, they need more comfortable soundscape
for the individual activity. Regarding the perception of dynamic
and communication, the participants prefer more varied and
communal soundscape for doing social activities while
individual activity needs to have a simple and private sonic
soundscape.
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Characteristics of urban soundscapes worthy of preservation
Yihong Jia, Jian Kang, Hui Ma
[Abstract]
Sound environment is an essential component of overall
environment and reflects the characteristics of a city.
Unfortunately, some precious soundscapes are gradually
disappearing with the rapid development of cities. Therefore,
preserving urban soundscapes is as important as noise
control in
order to create a good urban environment. In this study, a
questionnaire survey was first carried out in Tianjin to
investigate the urban soundscapes that should be preserved.
Totally 2504 respondents, including residents and tourists
participated in the survey, and 158 urban soundscapes were
chosen
by them as being worthy of preservation. Twelve of these
soundscapes were selected as representatives for a group
soundscape walk to explore the characteristics of
soundscapes
worthy of preservation. The result shows that relaxation,
vibrancy, representativeness, strength, and richness were
prominent characteristics of the soundscapes deemed
worthy of
preservation. Meanwhile, the preservation level were
positively
correlated with their relaxation and representativeness
characteristics.
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VR Environment-based Evaluation of Impact Factors on the Urban Soundscape Recognition
Rosa Seo, Hyun In Jo, Jin Yong Jeon
[Abstract]
To examine the impact of soundscape spatial change factors in a pleasant urban environment, we evaluated a soundscape in a virtual reality (VR) environment and investigated the audiovisual impact factors on the soundscape. First, nine evaluation points were selected around Area A, and three-minute sound environments were selected at each point using a 360-degree camera (Insta360) and a SoundField (SPS200) microphone to investigate changes in the sound pressure level and frequency characteristics. Subjective responses, such as overall satisfaction, soundscape evaluation factors, and landscape evaluation factors, were evaluated in three environments (audio, visual, and audio with visual) based on the sound environment measurement data. To realize the immersive environment, the subjects were allowed to move freely in VR by reflecting the direction of the sound source and head rotations. Assessments of the level of immersion and sense of reality were used to verify the experimental environment. The experiment results showed that, in terms of the soundscape evaluation factors, overall satisfaction decreased as traffic noise increased and human sounds decreased. As for the landscape evaluation factors, the vegetation factor was the most dominant; as the green area increased, the overall satisfaction increased despite the noise factor.
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Characterization of the urban sound environment from commercial pedestrian streets: a first approach in São Paulo- Brazil.
Bruno Xavier Rego, Haryadne Carniel, Marcos Holtz, Graziela Caruso
[Abstract]
Environmental noise management is in progress in Brazil, and
São Paulo was the first city to establish the mandatory
implementation of its noise map in 2016.The law regulation is
under development by the city council, and to respond to the
technical challenges that will be faced, different studies are
being carried out as there are no environmental noise
calculation models or standards for Brazilian cities
characteristics. One of the main issues to be solved will be how
to incorporate the mesh of commercial pedestrian streets on
acoustic simulations. In this study a pilot study was carried out at
”Calçadão de Osasco”, a pedestrian street in the metropolitan
area of São Paulo, with a high crowd density, packed with
shops and street traders, very representative of the local
soundscape. Different techniques of field measurements and
soundscape evaluation are applied to propose a first
characterization. It is evaluated how this proposal can be
incorporated to prediction softwares and aims to inspire the
study of the urban sound environment of commercial streets in
Sao Paulo.
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Acoustic Vehicle Alerting Systems (AVAS) of electric cars and its possible influence on urban soundscape
Felix Laib, J. Alexander Schmidt
[Abstract]
From July 1, 2019, all new types of hybrid electric and pure electric vehicles in the EU must be equipped with an Acoustic Vehicle Alerting System (AVAS), in particular to warn pedestrians at low driving speeds. In a case study, comprehensive measurements of the pass-by levels on 7 current electric vehicles were carried out at constant pass-by speeds of 10 km/h to 50 km/h. This involved recording the sound pressure level including and without AVAS. The results show that in some cases the pass-by level is considerably increased by the AVAS. Due to the frequently low speeds in urban traffic, the AVAS will play a relevant role for the future acoustics and the soundscape of cities. On the one hand, electric vehicles generally induce less noise in urban traffic than vehicles with combustion engines - a chance to reduce health threatening urban noise. However, since the AVAS of electric vehicles is intended to attract attention, it can on the other hand also generate additional noise annoyance. Considering an increasing share of electric vehicles in urban traffic in the future, the effects on urban quality and the quality of life are investigated. Keywords-AVAS, electric vehicles, urban quality of life, urban soundscape
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Research on Soundscape Identification – A Case Study in Shenzhen, China
Yu Lei, Jian Kang
[Abstract]
Subjective evaluation of soundscape is much related with the sound sources it contains. Due to ambiguous understanding of annoy perception of a noise, a large number of soundscapes were collected and studied to depict how various sounds influencing subjective evaluations. Forty-four sound samples were extracted from thousands samples by using Co-kriging method. The soundscape samples were obtained via 3-yrs field studies covering spatial and temporal variations of Shenzhen, a south-east costal metropolitan in China. Through experimental research on subjective evaluation of soundscapes to two hundred participants, five soundscape categories were elicited based on cluster analysis to subjective evaluations. It is found that soundscapes having more natural sounds were perceived less annoy whereas the ones having more mechanical sounds perceived more annoy; however, to some soundscapes containing more urban living sounds or hybrid sounds, subjective evaluations were not definite. A reason might be related with an individual’s social and cultural background linking with former experiences from once happened incidents linking with social activities/functions.
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Organization:
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J. Kang, University College London |
F. Aletta, University College London |
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Evaluating Sound Environment in Shenzhen by Using Artificial Neural Networks
Chen Xing, Zhixiang Tao, Yu Lei, Jian Kang
[Abstract]
Physically controlling urban noise is not efficient in which sound
meanings has to be included. As evolving from fish-village areas with
a hilly coastal topography, Shenzhen possesses a diversity sound
environment. Therefore, various ecological sound sources exists. In
order to get right knowledge of a sound environment, not only sound
levels but studies in Shenzhen, ten-thousands sound environment
samples were recorded. Statistical analyses of heterogeneity of sound
environments in Shenzhen were firstly examined. On a basis of
previous sound level Artificial Neural Network (ANN) models, sound
environmental quality in terms of levels were studied. Furthermore,
subjective evaluations of sound meanings to different sound
environments have been made. In order to provide a feasible tool in
measuring a sound environment referring sound meaning differences,
a number of ANN models attributing to three kinds of input variables
in predicting sound meanings were developed. The output was
derived from subjective evaluations of different sounds. Finally,
combining the results of sound levels and sound meaning, predicting
models to a sound environment in Shenzhen were given to provide a
feasible tool in measuring and monitoring Shenzhen sound
environments.
Keywords: Subjective evaluation, sound environment, Shenzhen,
Artificial Neural Network (ANN)
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Towards a Soundscape Surround Index
Tin Oberman, Kristian Jambrošić, Francesco Aletta, Jian Kang
[Abstract]
Sound sources are being spread in public spaces during
urban design and planning processes, yet little is known
about the effect their spatial pattern can have on
soundscape. The aim is to investigate if there are
associations with the perceived soundscape dimensions. The
dynamic relation between static sound sources and listener is
investigated in a virtual reality simulation made using
spherical photographs and the Odeon application, based on a
small enclosed square in Sheffield featuring following sound
sources: water sounds from the fountain, bird sounds from
the trees, traffic noise and sounds of people walking and
talking. The participants assessed different acoustic scenarios
wearing a pair of headphones and a virtual reality headset
with head movement tracking, while experiencing reproduced
environments through fixed five evaluation points. An index
for measuring spatial variability of sound sources in an urban
acoustic environment is proposed to describe physical
features of the investigated phenomena and help to predict
its influence on the perceptual outcomes.
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Acoustic comfort evaluation in residential buildings: modeling associations of acoustic data to subjective responses.
Nikolaos-Georgios Vardaxis, Delphine Bard-Hagberg
[Abstract]
Acoustic comfort is a complicated concept in dwellings. This article
presents a study aiming to investigate and evaluate acoustic comfort
in residential buildings. Various dimensions and factors of acoustic
comfort were studied in a sample of Swedish buildings. Acoustic data
was associated to self-reported responses acquired by a survey: a
questionnaire was setup researching the characterization of living
sound environment by residents, the response to noise annoyance
from multiple sources within a building and the emotional reactions of
tenants to acoustic climate at home. The total study sample includes
375 subjects from 34 various structure types (101 building units).
Statistical models were developed for the evaluation of subjective
noise annoyance for the cases of airborne and impact sound related
annoyance, for the identification of dimensions underlying the
concept of acoustic comfort as well as combined models for the
association of several technical parameters or subjective attributes to
acoustic comfort dimensions. KEYWORDS: acoustic comfort, field measurements, subjective
responses.
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A widened array of metrics (WAM) approach to characterizing urban soundscapes – the example in SALVE
Bryce Timothy Lawrence, Robynne Sutcliffe, Salman Ahmed, Susanne Moebus, Dietwald Gruehn
[Abstract]
The emergence of soundscape ecology provides new tools to understand the landscape. Soundscape ecology studies include over 36 soundscape indices (metrics), based on the factors sound pressure (dB), frequency (Hz) and time. Many metrics have been developed for animal observation, whereas fewer specific metrics for urban soundscapes exist. Decibels have been the traditional ënoise’ measurement worldwide. The authors argue that utilization of a single factor (dB) has limited value to characterize urban soundscape complexity. Rather, soundscape analysis should address urban soundscapes as holistic compositions of dB and Hz, over spatial-temporal extents, both within and amongst observations. Therefore the MERCUR supported SALVE project - acouStic quAlity and heaLth in urban enVironmEnts - proposes a widened array of metrics (WAM) to enable the broadest possible understanding of ësoundscape.’ The WAM approach will provide a multi-dimensional understanding of sound as signatures, clusters and spatial disparities of sound phenomena, which characterize the temporal/sonic heterogeneity of urban environments. WAM will be used to analyze the interrelationship of urban soundscapes, the built environment and the health and well-being of the urban population. This paper justifies the metrics included in the WAM approach, the related software, and the interpretation of sound signatures against spatial and health data.
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Ambient Soundscape Assessment of Wind Parks
Tianhong Yu, Holger Behm, Ralf Bill, Jian Kang
[Abstract]
The potential conflicts between wind parks and the public are
enormously enhanced after a rapid increase of wind parks.
Assessment and management of environmental noise have
played a vital role in the reaction of people concerning wind
parks. This study uses an aural-visual preference survey to
systematically investigate effects of background soundscapes
and non-aural factors on annoyance toward wind parks.
Visual and acoustic data were captured from seven German
rural wind park sites. Laboratory experiments were then
carried out with 40 participants to investigate the perception
of the wind parks in rural areas. The tests consisted of three
parts: 1) visual-only condition, 2) aural-only condition, and
3) combined aural-visual condition. Participants were
immersed through google cardboard in laboratories using
virtual reality technology to evaluate properties of sound and
visual environment. Results suggest that the annoyance of
wind parks strongly correlated with sound level ambient wind
parks. However, the visual information did not demonstrate
substantial effects on the people’s opinions of wind parks.
Further results show that the soundscape ambient wind
parks associated with factors including
”calmness/relaxation”, ”naturality/pleasantness” and
”diversity.” Additionally, ”calmness/relaxation” and
”naturality/pleasantness” were found to have a strong impact
on the evaluation of wind park landscapes.
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Soundscape cost index: a case study in Aachen
Margret Sibylle Engel, Carmella Pfaffenbach, Janina Fels
[Abstract]
Soundscape studies are showing the importance and the
benefits regarding the sonic the environment in our daily lives.
Through the soundscape paradigm shift, not just negative
aspects, common on environmental noise studies, are
highlighted, but also how the sonic environment can be
recovering to the welfare of communities. Thinking of the
valuation of environments, past studies focused mainly on how
much the environmental noise does cost for the society. This
kind of valuation followed several methods like hedonic
pricing, contingent valuation, and benefits transfer methods,
which are based just on economic factors involving socio-
economic aspects. The aim of this study is to show a new
approach which fits better with the soundscape paradigm
shift, highlighting the valuation of positive and negative
aspects of the evaluated soundscape, together with the
interaction of socio-economic and socio-cultural aspects, as
well as aesthetics, thermal-comfort and, air quality, which are
other stimuli who can influence our general environmental
perception.
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Soundscape Cognition for User Behavior in Urban Parks
Hyun In Jo, Jin Yong Jeon, Jean-Dominique Polack
[Abstract]
To investigate the cognitive components of urban park soundscape design, we evaluated soundscape preference based on visual object change factors for the background and people within the virtual reality (VR) environment. We investigated the changes in the sound pressure level and frequency characteristics among 18 evaluation points selected from three parks in Paris. For the investigation, sound environments of two minutes and 30 seconds during the daytime (10:00-14:00) were selected from each point using a 360-degree camera (Insta360) and SoundField (SPS200) microphone. To investigate the behavioral characteristics of urban users, the image of each selected point was divided into 30-second segments, and objective characteristics, such as activity, group, posture, and density, were analyzed. We also evaluated subjective responses, such as soundscape evaluation factors, landscape evaluation factors, and overall satisfaction, to investigate the differences in soundscape cognition as behavioral characteristics changed at the same point in the park. To realize an immersive environment, the subjects were allowed to move freely in VR by reflecting the direction of the sound source and head rotations. Assessments of the level of immersion and sense of reality were used to verify the experimental environment.
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Investigation on the restoration effect of soundscape in parks in high- density cities: Taking Lu Xun Park in Shenyang, China as an example
Yuan Zhang, Ruining Zhang, Ling Zhu
[Abstract]
Urban parks, in addition to providing a rest space for urban residents,
also provide a restorative environment for relieving stress and relaxing. As
the main environmental factor, the soundscape affects the
environmental restorative quality. Small parks in high-density cities have
special soundscape characteristic due to their crowded population. This
paper selected a small city park in the central area of a city - Lu Xun
Park(4,3000m2) and conducted soundscape survey, including sound
pressure level measurement, acoustic event recording, restorative
evaluation and tourist questionnaire; using ArcGIS for Interpolation
analysis and spatial analysis, and constructed a soundscape map to
explore the effects of soundscape on restorative effects.
The results showed that: 1) LAeq distribution inside the park: 42.9-
69.3dB(A), and exhibited obvious time and space laws; 2) sound
composition included rich citizen activity sounds, bird sounds and
surrounding road traffic sound. There are significant differences between
these acoustic events in terms of their value in restoration, sound
sensitivity, acoustic interference, range of transmission, etc; 3) sound
pressure level, acoustic event composition, spatial form, and landscape
features affect the environmental restoration quality of different patches;
4) individual factors such as gender, age, and especially visiting purpose
affect environmental restorative experience.
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The Synthesis of Soundscape using Genetic Algorithm and Popular Songs
Mia Suhanek, Sanja Grubeša, Ivan Djurek, Antonio Petošić
[Abstract]
This paper proposes the concept of soundscape synthesis
based on improving existing soundscapes with optimization
through Genetic Algorithms. As a goal function for optimization
the parameter of some acoustic environment called Total
Distraction Coefficient (which depends on loudness, sharpness
and loudness growth rate) is chosen by which one can
synthesize an improved soundscape. To verify the proposed
concept two representative (and familiar to most urban
residents) acoustic environments are further analyzed and
compared (namely, a children’s park and an expressway). Using
Genetic Algorithms (with previously known Total Distraction
Coefficient) two new soundscapes are synthesized from each
acoustic environment - one with lower and one with optimum
Total Distraction Coefficient (i.e. the environments supposed to
be perceived as ”worse” and ”better” than the starting one,
respectively). The obtained soundscapes are then tested for
subjective perception.
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Organization:
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D. Botteldooren, WAVES research group, Department of Information Technology, Ghent University |
L. Maffei, Università degli Studi della Campania |
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Dynamic approach for the study of the spatial impact of road traffic noise at peak hours
Arnaud Can, Pierre Aumond, Cécile Becarie, Ludovic Leclercq
[Abstract]
A dynamic modelling chain, coupling the Symuvia dynamic traffic model (LICIT, Ifsttar / ENTPE) and the Noisemodelling noise level prediction tool (UMRAE, Ifsttar / Cerema), is developed in this paper to assess the impact of road traffic on noise environments at the scale of an urban traffic network of about 10 km2 located in Lyon/Villeurbanne. The simulation, which reproduces 3 hours of a morning rush hour scenario, highlights the increase in noise levels associated with the increase in travel demands. It also highlights that the increase in noise levels is not uniformly distributed over the network and has a greater impact on the streets where traffic is reported. In addition, noise dynamics are modified: background noise is particularly sensitive to the increase in vehicle density on the network, so periods of calm on the network are rare. A more detailed analysis of this dynamic is possible locally through the analysis of the evolution of the LAeq,1s and the study of specific acoustic indicators. Desirable improvements to the model are discussed, with a view to assessing the currently unknown acoustic impact of road traffic management strategies at the scale of an urban network.
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Interactive soundscape augmentation of an urban park in a real and virtual setting
Timothy Van Renterghem, Kang Sun, Karlo Filipan, Kris Vanhecke, Toon De Pessemier, Bert De Coensel, Wout Joseph, Dick Botteldooren
[Abstract]
Inappropriate soundscapes are able to strongly deteriorate the user experience in parks. A possible remediation is adding positively perceived sounds. The case of an urban park, fully surrounded by road traffic noise sources, was studied to explore the potential of adding natural sounds in an interactive way. A preliminary test was conducted in the lab with Virtual Reality (VR) glasses and headphones. The audio-visual representation of the real environment was obtained by combining binaural recordings with first-order ambisonics and a 360-degree video camera footage. The users were allowed to mix in 8 types of natural sounds until their personal optimized soundscape was composed. This was done in a very similar setup as in the (real) park. The loudspeaker augmenting the sound environment in the park was steered with a smartphone application. This app ensured the user’s presence near the loudspeaker and allowed to gather more detailed assessments of the perceived sound environment through a small questionnaire. This combination of experiments allowed checking the validity of VR that is becoming increasingly popular in environmental perception studies. In addition, the most preferred natural sounds and their interaction with the changing road traffic noise exposure over time were determined.
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The Potential of Being the Quiet Place of the Khans Courtyards in Istanbul Historic Peninsula
Gülşen Akın Güler, Aslı Özçevik Bilen, Massimiliano Masullo, Luigi Maffei
[Abstract]
The importance of noise from transport and industrial sources
on human health were emphasized by Environmental Noise
Directive. As well as actions to minimize the impact of this
noise on human, the directive impresses on the need to
protect quiet areas having good environmental noise quality.
Therefore, studies on identification, determination and
utilization of quiet areas in urban and rural contexts have
increased in many countries in Europe. Sound environments in historic city centres consist of different
sounds from dense tourism, trade, entertainment, recreation
etc.; consequently, the noise problem is inevitable. Quiet
areas enable people get away from noise and reduce the
adverse effects of it by providing opportunities for rest,
relaxation and restoration to them. In this paper, the khans courtyards in Istanbul Historic
Peninsula (Turkey) were considered as potential Quiet Places.
Noise measurements, soundscape recordings with soundwalk
method, observations and surveys were realized to determine
quantitative and qualitative quietness in the courtyards. The
paper reports the preliminary results of the Quiet Place
perception in Turkey, and it reveals that the inner courtyard is
a protected place against the outside noise sources, but it is
necessary to take measure for the noise sources inside the
courtyard.
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Organization:
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S. Graetzer, Acoustics Research Unit, University of Liverpool |
P.N. Dokmeci Yorukoglu, Cankaya University |
F. Aletta, University College London |
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Designing spaces and soundscapes. Integrating sonic previews in architectural modelling applications.
Alessia Milo, Joshua Reiss
[Abstract]
Acoustics and architects are often engaged in design dialogues. While
architects work on the definition of the spaces, acousticians deal with
understanding in advance how these spaces will be sonically
perceived by the future inhabitants. Sometimes, communication
might be difficult, lacking a common vocabulary based on sound
energy descriptors, frequency distribution, and perceptual
dimensions. In this paper, we present the results of an online questionnaire
answered by 15 professional architects asked to consider the possible
integration of sonic anticipations in their design practice. The totality
of the participants declared to be interested in working with a
software environment enabling them to listen to (i) how changes in
the geometry would affect the diffusion of sound, (ii) how changes in
the materials would affect the diffusion of sound, (iii) simulate sound
sources which are likely to be present in the space to be designed.
Some of this percentage expressed the need to be further guided for
such activity.
A third of the respondents also completed an optional section which
required them to test three interactive online examples,
Playsound.space, Tranquil City pavement, and a javascript spatial
sound demo. In discussing the results, we provide our design
suggestions for the scenario depicted.
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The role of indoor soundscape methodology: From architectural design process to establishment of regulations
Ugur Beyza Ercakmak, Papatya Nur Dokmeci Yorukoglu
[Abstract]
One of the most important research field to enhance the
environment is ”soundscape” that concentrates on urban and
indoor sound environments, with a focus on improving the
quality and pleasantness. The ability of designing sustainable
soundscapes and the necessity of it, have led to a new
discussion in standardization of soundscapes. Most of the
detailed work of the working group on perceptual assessment
of soundscape quality (ISO/TC 43/SCI/WG 54) focuses on
soundscape quality, analysis methods and parameters on open
public spaces. However, indoor soundscaping, which is a more
recent research topic is as important as the urban soundscape
because people spent 80% of their time indoors. Therefore, its
standardization and application in the architectural design
process is very crucial to improve indoor soundscape quality. In
this study, present policies and applications about noise
management and soundscape are assessed and international
regulations are compared. In addition, deficiencies in both
national and international regulations regarding indoor
soundscaping, and promising leading role of standardizing
indoor soundscape methods and factors to be used in
architectural design process and the establishment of related
regulations are discussed.
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Characterization of the indoor kitchen soundscape
Pieter Thomas, Luc Dekoninck, Stephanie Van Hove, Anissa All, Peter Conradie, Lieven De Marez, Henk Huisseune, David Plets, Dick Botteldooren
[Abstract]
With a growing tendency for compact living, the kitchen space is typically at the center of the daily activity, combining different functionalities. In the context of the ”IoT chef” project, the cooking experience is optimized, with focus on the cooker hood noise.
In a first part of the study, a large scale online survey (N = 470) is conducted among Flemish home cooks in order to (1) map personal assessments of the sound environment during a typical cooking experience and (2) evaluate the indoor (kitchen) soundscape based on the ISO 12913-1/2 standard. The impact of the role of the observant (chef or bystander) on the subjective evaluation is investigated.
In a second part of the study, typical kitchen sound levels are monitored during a one-week period at 16 households, selected from survey participants. Additionally, cooker hood levels are measured at the cook’s ears to define the contribution of hood noises in the overall kitchen sound environment. Results from the measurement campaign are presented and discussed.
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Correlations between sound level and loudness, intimacy, reverberation, clarity, spaciousness in sequential spaces
Tingting Yang, Jian Kang
[Abstract]
Sequential space is a spatial form, which cannot be
considered as a single enclosure but rather as a series of
spaces connected by openings in between. The acoustic
perception of such spaces is important for indoor
soundscape. This study selected two sequential spaces as
sound-source-group (i.e. with a sound source at the end
room of the sequential space) and no-sound-source-group,
to explore the relationship between sound pressure level
and perception of loudness and reverberation. The
perception of loudness and reverberation was investigated
via questionnaire surveys. The results show that the sound
pressure level of sound-source-group attenuates
significantly, by about 10 dB(A), while that of no-sound-
source-group has no significant difference. For the sound-
source-group, while there is a significant correlation
between perception of loudness and sound pressure level,
the perception of reverberation has no significant
difference among rooms. For the no-sound-source-group,
the perception of loudness and reverberation has no
significant difference among rooms.
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Towards Understanding Healthy and Supportive Acoustic Environments: the Case of a Nursing Home
Paul Devos, Pieter Thomas, Francesco Aletta, Tara Vander Mynsbrugge, Patricia De Vriendt, Dominique Van De Velde, Dick Botteldooren
[Abstract]
In elderly care a nursing home is a facility where on a residential basis different caring functions are taking place in order to satisfy the specific care needs of the residents. As in many outdoor and indoor situations, the soundscape of such a space, as originated form its acoustic environment, is of crucial importance since it has the potential to affect the behaviour of the residents. Apart from sounds originating from the care-related interactions between staff members and residents and other typical indoor sound sources like equipment noise, a major contribution to the acoustic environment in a nursing home can be the shouting of residents themselves. It is known that in case of residents with dementia, shouting behaviour can be a prevalent element of BPSD (behavioural and psychological symptoms of dementia). Considering the supportive potential of a soundscape, its quality needs to be studied in view of these shouting events. Using signal processing techniques applied to datasets obtained in different nursing homes in Flanders these shouting events can be detected and allow to describe the shouting behaviour of a resident, the temporal pattern of which and its relation to the present soundscape is explored in this paper.
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Pleasantness of Typical Acoustic Environments inside a Living Room in a European Residential Context.
Romain Dedieu, Catherine Lavandier, Cédric Camier, Sylvain Berger
[Abstract]
Acoustic comfort defined in first approach as the absence of
unwanted sounds and the quality of desired sounds while doing
an activity arouses a growing interest in sociology, psychology
and building industry.
Comfort is influenced by the acoustic performances of building,
by the various types of sound sources, and by individual aspects
(sociological background, musical preferences etc.). A study
based on three perceptive tests aims to address these aspects.
The first perceptive test consists in the evaluation of
pleasantness for various sound sources according to their
semantic content and their origin (from the neighbor, the
outside or the room). Thus, 67 focused listeners rated the
pleasantness of every stimulus. Statistical analysis showed a
consensus on intrusive sounds coming from outside, people
preferring bird songs and rejecting site work sounds. For intrusive
sounds coming from neighbors, there is also a consensus,
people preferring classical music and rejecting human sounds.
Participants can be clustered in two groups, one preferring
nature sounds and the other preferring music sounds. Moreover,
the ratings of the human sound pleasantness (diner, laughs,
etc.) are influenced by the sound sensitivity of the participant,
the more sensitive the less pleasant the sound.
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Acoustic Comfort in Multi-group Conversation Space: a Field Experiment in an Actual Cafe with Different BGM Levels
Naoya Maruyama, Yasuhiro Hiraguri, Keiji Kawai, Mari Ueda
[Abstract]
We refer to ”Multi-group conversation space (MGCS)” as
a space where more than one group are conversing
within the group, such as restaurants or cafes, and have
been researched to find acoustic comfort in MGCS
considering speech intelligibility and speech privacy as
major factors. In this study, a field experiment was carried
out in an actual café, where a questionnaire was
performed on the customers for four days. The level of
BGM in the operating time was changed in each day with
none and three levels and the indoor noise levels were
measured by six microphones hung from the ceiling. The
questionnaire consisted of eight questions including
easiness of conversing, comfortableness,
atmosphere(lively/relaxed), and so on. The customers
were asked to answer the questions at each table and, at
the same time, noise level in the vicinity of them was
measured and the occupation of next seat were
recorded. As a result, easiness of conversing tended to be
evaluated better under the BGM of middle level than
under that of high or low levels. Also, the impression
evaluation varied by gender, age, or seat position and
occupation.
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Searching the Industrial Soundscape of The Urban Past of An Anatolian City: Eskişehir
Özlem Gök Tokgöz, Aslı Özçevik Bilen, Özlem Kandemir
[Abstract]
Sounds are inherent parts of cities. While any change occurs in
an urban environment, it effects the sounds thus its
soundscapes. Changes in urban soundscape with the industrial
revolution in Central Anatolia (Ottoman Period) dated back to
the construction of Bagdad Railway in 1894 and continued with
the industrialization process after the declaration of the
Republican regime. One of the leading cities got involved in this
process was Eskisehir which is a Central Anatolian city, on the
intersection of trade routes. Because of that locational
advantage, several factories had been built and had provided
the impetus to change and form a new urban identity, spatial
and social changes.
The aim of this study is to reveal the soundscapes of these
industrial buildings specifically at that period, and obtain
acoustical information of their environment. Therefore,
interviews were done with the factory workers of that period,
people who lived in the vicinity of the factories and experts who
studied the changes, as well as retrospective audio-video
records and photographs were collected and analyzed. As a
result, industrial soundscape of the urban past is concluded by
defining soundscape components and important role of the
industrial heritage in urban soundscape is revealed
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Can natural sound assist in the treatment of depression?
Yuzhe Ge, Hui Xie
[Abstract]
Music has been shown to be helpful in the treatment of patients
with depression. This article is to explore whether natural sound
also has such a promoting effect. The study was a 4-week
andomized controlled trial, in which we provided 6 types of
natural sound, including ocean sound, stream sound, bird
sound, insect sound, wind sound, and rain sound, for the listener
to choose freely. 60 depression patients were selected as
experimental subjects, half of them as experimental group and
the other half as the control group. The experimental group
listened to the natural sound for 30 minutes every day on the
basis of routine treatment, and listened for 7 days a week for 4
weeks, while the control group was only treated with routine
therapy. Depression was measured with the Beck Depression
Inventory before the study and then once a week after the start
of the treatment. After 4 weeks, the depression scores of the
patients in the experimental group were significantly lower than
those in the control group. Natural sound therapy may reduce
patients’ depression level and proved to be an effective
approach to public health and home care nurses in the
treatment of depression.
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The soundscape dimensions of third-class hospital ward in Indonesia
Anugrah Sabdono Sudarsono, Sugeng Joko Sarwono, Aisyah Shabrina, Laudita Natasha Tamrin
[Abstract]
In Indonesia, the categorization of hospital ward is determined based on the number of bed. Third class hospital ward (the lowest class) is the ward which consist of more than 4 beds inside the room. This type of ward is typical in developing country and might have different acoustic problems compared to the standard ward. The aim of this study is to understand the soundscape dimension in a third class hospital ward. Two experiments were conducted: the development of semantic scales and the identification of soundscape dimensions. The semantic scales were developed by asking the patient about the feeling in the ward. Most of the terms used to describe the environment are negative term such as annoying, boring, scary, upset, uncomfortable, etc. This result indicates the negative soundscape of the ward which need to be improved. The next experiment were conducted by asking the patient’s rating of 11 semantic scales. Principal component analysis is used to analyze the rating resulting four soundscape dimensions. The dimensions are Annoyance, Boredom, Privacy, and Dynamic. This dimensions indicates the important aspect which need to be improved in third-class hospital ward.
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Organization:
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S. Cooper, The Acoustic Group |
N. Timmermann |
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The use of Synthesised or Actual Wind Turbine Noise for Subjective Evaluation Purposes
Steven Cooper
[Abstract]
There are technical difficulties in producing an accurate wind turbine noise signal for subjective testing of the noise characteristics of the different operations of wind turbines. There are differences in the subjective response when limiting the test signals to infrasound only versus the use of full spectra. The concept of ”nocebo” effect that has been presented has relied upon the use of ”synthesised wind turbine infrasound” that does not reflect the signature or pressure pulsations observed in full-spectrum field measurements. The validity of ”synthesised wind farm infrasound signals” that have been used in such testing is examined and compared with full-spectrum signals.
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A Simplified Method for Determination of “Amplitude Modulation” of Audible and Inaudible Wind Turbine Noise
Steven Cooper
[Abstract]
The operation of a wind turbine results in a series of pulses where there is a significant increase in the amplitude of the pressure signal dependent upon the loading (power output and wind strength) of the wind turbine. Such amplitude variations can be significant. The modulation of the amplitude of the acoustic signature for wind turbines is often referred to as ”amplitude modulation”. One method of assessment of the degree of amplitude modulation for a wind turbine used in the UK is complex, time-consuming and expensive to undertake. A simplified method has been developed that is not just restricted to the dBA level and can be used to cover both inaudible and audible dynamically pulsed amplitude modulation. This simplified analysis method is not just restricted to wind turbine noise but has uses for other pulsating noise sources. Investigation of recreational music and industrial noise sources that give rise to the generation of pulsations occurring at an infrasound rate using the simple methodology is discussed.
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A comparison of Inaudible Windfarm Noise and the Natural Environment Noise whilst Monitoring Brainwaves and Heart Rate
Steven Cooper
[Abstract]
A pilot study undertaken in late 2017 using inaudible wind turbine noise found persons having a heightened sensitivity to turbine noise could detect the presence of the signal by way of feeling (rather than hearing) the test signal. A control group that had not been exposed to wind turbine noise was unable to detect or sense those operations. A pilot study utilising inaudible wind turbine noise, inaudible white noise and inaudible surf (ocean) noise was undertaken in a 126m3 reverberation room and also in an anechoic room, whilst monitoring of the test subjects heart rate and brainwaves was obtained. The results of that testing are discussed.
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Non-acoustic and acoustic variables associated with wind turbine noise annoyance
Jenni Radun, Valtteri Hongisto
[Abstract]
BACKGROUND. Previous residential surveys have been conducted in areas involving wind turbines (WTs) less than 3 MW. However, modern WTs are usually larger than 3 MW and the applicability of the previous results must be re-examined. Previous results show that sound level explains only a small portion of noise annoyance in WT areas. Here we will concentrate on examining the role of non-acoustic factors. AIM. We wanted to determine how various non-acoustic factors are associated with WT noise annoyance. METHODS. The survey was conducted around three areas involving WTs sizing from 3 to 5 MW in Finland. Altogether 318 permanent residents living within 2 km from the WTs answered the survey. The modelled WT noise outside their houses ranged from 27 to 44 dB LAeq. RESULTS. The concern for health effects of WT noise is the strongest associate of WT noise annoyance. Other factors were area, noise sensitivity, gender, and general attitude towards wind power as an energy production form. Sound level of WTs had only a weak association with annoyance outdoors but not indoors. PRACTICAL IMPLICATIONS. Provision of fact-based information about the health effects of wind turbines could reduce the health concerns and thereby noise annoyance.
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Organization:
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A. Preis, Adam Mickiewicz University |
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Audio/Visual Interaction in the Perception of Sound Source Distance
Pavel Zahorik
[Abstract]
The perception of sound source distance is known to exhibit
systematic biases. In general, the distances of far sources are
progressively underestimated, but near sources are
overestimated. Such biases are not typically observed in vision,
however. Under natural viewing conditions in which a variety of
visual distance cues are available to the observer, perceived
distance is highly accurate. Relatively little is known about how
distance information from both auditory and visual modalities is
combined in the perception of distance, however. This is
surprising, given that audio/visual aspects of directional
perception have been extensively studied, primarily in relation
to the ”ventriloquist effect”. Here, two experiments on
audio/visual distance perception are summarized. Both used
virtual auditory space techniques to simulate reverberant sound
field listening of a loudspeaker-produced broadband noise
signal. The results from both experiments suggest that not only is
perceived distance less accurate in the auditory modality than
in vision, but it is also considerably less precise. A computational
model was developed based on data from these two
experiments. Predictions from the model offer explanations as
to why visual information, when available, appears to dominate
auditory information in the perception of distance.
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Questions applied in audio and visual environment assessment
Jan Felcyn, Anna Preis, Marcin Praszkowski, Małgorzata Wrzosek
[Abstract]
The aim of the study was to examine how the way in which a question regarding audio, visual and audio-visual stimuli is formulated could influence environment assessment. To get quantitative ratings in both modalities, a numerical scale coupled with precisely formulated questions was used for all the investigated audio, visual, and audio-visual stimuli. The participants were asked either about the comfort or the discomfort caused by the perceived stimuli. The assessments of audio and visual comfort and discomfort were carried out in a laboratory settings. In all cases an 11-point numerical scale (from 0 to 10, derived from ICBEN recommendations) was used to rank the comfort and discomfort ratings. The results show that there are no statistically significant differences between the comfort and discomfort assessment of visual samples. Actually, the comfort and discomfort ratings are equivalent in such a way that a discomfort rating can be represented as the opposite of the comfort rating, i.e. discomfort rating is equal to 10 - the comfort rating. For the audio stimuli, whether the comfort and the discomfort scales are reversible depends on both the sound level at which the stimuli were presented and on the type of stimulus.
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The Human Perception based on Memory Recall of the Multi-sensory Stimuli in Outdoor Urban Space
Ni Putu Amanda Nitidara, Anugrah Sabdono Sudarsono, Ranti Dwi Tassia, Joko Sarwono, Fx Nugroho Soelami
[Abstract]
The human perception of environment condition is one of the aspects that greatly affect the use of open public space in a built environment. Studies related to the perception generally observe only one physical aspect in an isolated manner, in which the interrelationships between physical aspects are not considered. In the meantime, human beings are naturally multi-sensory and the human perception is exposed by multi-sensory stimuli that influence each other. This study tries to look at that relationship of multi-sensory stimuli to human perception, especially in the outdoor urban environment. In this study, we implement a semi-structured question to understand people’s perception based on their memory of four outdoor urban area in Bandung. The questionnaire explores the relationship between three physical aspects, aural, thermal, and visual. Three main perceptions were found, comfort, spatial, and age/era. There is also evidence showing that multi-sensory stimuli affect the same perception, e.g. thermal and visual affect comfort perception. The finding of this study shows a preliminary evidence of multi-sensory perception in an urban area. This knowledge will be a powerful tool for understanding the perception of the urban environment and help in the process of urban area design and enhancement.
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Organization:
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A. Chung, Macau Instituto de Acústica |
A. Radicchi, Technical University of Berlin |
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New Trends in Sound Planning using Immersive Virtual Reality Technology
Andy Chung, Terence Tsang, W M To
[Abstract]
Sustainable sound planning has been drawing more attention
as citizens are demanding higher levels of comfort and livable
environment while keeping the convenient access to
transportation and neighborhood in urban settings. Particularly
in Hong Kong, we see a lot of innovative building designs, such
as smart facades, are there improving the overall indoor sonic
environment of the premises. Apparently, different situations
and designs can result varying levels of costs and benefits. How
to balance these becomes a challenge for the acoustical
designers and sound planners. This paper presents a case study
revealing how virtual reality technology can be applied to let
the relevant stakeholders ”experience” the noise benefits that a
particular smart facade design can bring by ”perceiving” the
sonic environment and having it associated with different
design options and costs.
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Design for quiet living: tracing the development of projects in noise-affected areas with software-based data analysis and visualization
Cristina Calleri, Louena Shtrepi, Alessandro Armando, Arianna Astolfi
[Abstract]
The research adopts the Science and Technology Studies
view of architecture as product of collective action,
which develops through the translation of different
instances into common goals. In such context, it is crucial
to understand how codes and metrics impact on design
processes, to help designers and policy-makers make
informed preliminary choices.
With this perspective, we focus on noise mitigation
policies, as codes that can generate complex
environmental-spatial conflicts, to which designers are
required to find innovative solutions.
The case-study of a dwelling project in the Netherlands is
analysed through qualitative analysis of interviews and
documents, conducted through CAQDAS software
Atlas.ti.
Themes and codes are inductively derived from the words
of the actors and organized following concepts
developed by actor-network theory studies. Matter of
concerns that emerge during the process and actors
mobilized by them are therefore identified, together with
actions and design solutions that are implemented to
achieve a translation between noise mitigation policies
and other (potentially) conflicting requirements.
The outcomes are then visualized through Atlas.ti and
Node XL, with the aim of providing an intellectual
technology to make the project process legible and set
the basis to draw comparisons between different projects,
when analysed through the same methodology.
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Organization:
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J. Wiciak, AGH University of Science and Technology |
F. Zotter, Inst. f. Elektr. Musik u. Akustik, KUG |
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Analysis of ambient noise influence on recording of unique Arctic soundscape – Kapp Linné, Svalbard
Dorota Czopek, Pawel Malecki, Janusz Piechowicz, Jerzy Wiciak
[Abstract]
Soundscape interacts with landscape perception and affects the sense
of place and its cultural significance. Moreover, its quality is
important for human and wildlife wellbeing. Therefore scientists
emphasize that it should be considered natural resources worthy of
management and conservation.
Paper presents analysis of ambient noise influence on soundscape of
Kapp Linné Bird Sanctuary. This unique place is reserve located at
the south side of the outlet of Isfjorden on Spitsbergen, Svalbard.
The reserve covers also the sea and surrounding islands up to 300 m
from the shore with low tide. There is a Isfjord Radio at the northern
border of the reserve. It was build in 1933 and now it is used for
tourist purposes. During research expedition to Svalbard in July 2018
SPL measurements and ambisonic recordings were made in Isfjord
Radio and nearby. A generator that powers the station and the hotel
has the sound level 65 dB SPL (5 meters). Based on the designated
sound power levels of noise sources noise maps have been drawn.
The aim of the analysis is to illustrate that in nature reserve not only
landscape with wildlife, flora, fauna or features of geological but also
soundscape should be protected.
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Digitalizing the sounds of the past: the soundscapes of World Heritage rock art landscapes from Spain and Mexico
Angelo Farina, Margarita Díaz-Andreu, Enrico Armelloni, Mathieu Picas, Leslie Zubieta, Tommaso Mattioli
[Abstract]
In the last few years our team of archaeologists from the
University of Barcelona and acoustic engineers from the
University of Parma have collaborated in a series of projects
funding archaeoacoustics fieldwork in prehistoric rock art
landscapes. The projects, SONART, Palarq and the ongoing ERC
Advanced Grant ”Artsoundscapes -
The sound of special places”, aimed at investigating the
soundscapes of painted rock shelters, including some whose
Outstanding Universal Values have led to their inclusion on the
World Heritage list. Our tests have analyzed monaural (T20,
C50, G) and binaural (Lj, Jlf) parameters obtained by
Impulse Response analysis. Additionally, we have used
Transmission Loss analysis and 3D Ambisonic recording to
investigate two other types of acoustic
perceptions that the communities that produced the ancient
paintings made on rocks seem to have favored. These are a)
the augmented audibility of distant sounds and b) the arrival of
echoes originated in rock art sites. In this paper, the authors
discuss the results of fieldwork in two World Heritage landscapes,
the ”Rock Paintings of Sierra de San Francisco” (WHL property
714) in Baja California (Mexico) and the ”Rock Art of the
Mediterranean Basin on the Iberian Peninsula” (WHL property
874) in Spain.
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Organization:
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P. Lercher, TU-Graz, Austria |
P. Aumond, IFSTTAR, CEREMA, UMRAE |
A. Can, IFSTTAR, CEREMA, UMRAE |
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Exploring associations between soundscape assessment, perceived safety and well-being: a pilot field study in Granary Square, London
Francesco Aletta, Luca Molinero, Arianna Astolfi, Sonja Di Blasio, Louena Shtrepi, Tin Oberman, Jian Kang
[Abstract]
The number of soundscape studies is growing steadily over time
and the discipline as a whole is going through a process of
methodological standardization. Yet, more empirical evidence
is needed about the benefits of experiencing positive
soundscapes for health, well-being and Quality of Life. In this
study, a pilot questionnaire was distributed to 50 users of a
public open space in London (Granary Square), which
consisted of a standardized protocol for collecting soundscape
data (ISO 12913-2:2018, Method A), the World Health
Organisation-Five Well-Being questionnaire (WHO-5) and a
protocol to assess perceived safety. Results show that
associations between positive soundscape dimensions and
higher WHO-5 index scores exist at a statistically significant level
(p < .05) and offer further insights into the relationships between
sound perception, life satisfaction and perceived safety. By
making a stronger case for policy makers and planners, these
findings contribute to the discourse of approaching the
management of the urban acoustic environments in a
proactive rather than reactive way, to promote healthy and
supportive urban spaces.
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Relationships between noise annoyance, urban soundscape and acoustic indicators in the French city of Lorient
Pierre Aumond, Arnaud Can, Catherine Lavandier
[Abstract]
As part of the ANR CENSE project, launched in 2016, a questionnaire was sent to households in a 1 km2 study area in the city of Lorient in France. The main objective of this questionnaire was to collect information on the residents’ perception of the sound environments in their neighborhood (representation of the soundscape), their street (representation of the soundscape) and their home (noise annoyance). In the same study area, more than a hundred sensors were positioned to cross the perceptual variables collected by the questionnaire with acoustic measurements. The analysis of the relationships between perceptual variables and acoustical indicators will be presented.
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SiRENE-Survey Part 1: Exposure-effect relationships for transportation noise annoyance in Switzerland
Mark Brink, Jean Marc Wunderli
[Abstract]
The survey conducted as part of the SiRENE study aimed at establishing exposure-response relationships reflecting the percentage highly annoyed (%HA) as functions of road traffic, railway, and aircraft noise exposure, measured as day-evening-night level (Lden). We hypothesized that in addition to Lden, the effects of noise on annoyance can be better explained when also considering the intensity of short-term variations of noise level over time (see Part 2). We conducted a mixed-mode representative population survey in a stratified random sample of 5592 residents exposed to transportation noise all over Switzerland. Source-specific noise exposure was calculated for each floor and each façade based on comprehensive traffic data. Exposure assessment was validated with measurements in 102 survey respondent’s homes. The survey was carried out in 4 waves at different times of the year to account for potential temperature effects. For all noise sources, results revealed significant associations between Lden and %HA after controlling for confounders. Aircraft noise annoyance scored markedly higher than annoyance to railway and road traffic noise at the same Lden level. Railway noise elicited slightly higher percentages of highly annoyed persons than road traffic noise. Our results point to the conclusion that a ”railway bonus” is not warranted anymore.
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SiRENE-Survey Part 2: Effects of Intermittent versus Continuous Noise on Annoyance Reactions
Jean Marc Wunderli, Mark Brink
[Abstract]
The survey conducted as part of the SiRENE study aimed at establishing exposure-response relationships reflecting the percentage highly annoyed (%HA) as functions of road traffic, railway, and aircraft noise exposure measured as Lden. We hypothesized that in addition to Lden, the effects of noise on annoyance can be better explained when also considering the intensity of short-term variations of noise level over time. We therefore incorporated the acoustic indicator IR in the statistical models. Results show that for road traffic noise, IR has an additional effect on %HA and can explain shifts of the exposure-response curve of up to about 6 dB between low and high IR exposure situations. Interestingly, road traffic noise situations with high IR, i.e., situations with typically few vehicle pass-bys with high emergence from background noise, were thereby perceived as less annoying, possibly due to the effect of different durations of noise-free intervals between events. For railway and aircraft noise annoyance, the predictive power of IR was limited. This can be explained by the fact that these noise sources are eventful per se, producing in the vast majority of cases single pass-by events that clearly emanate from background noise with longer periods of calmness in between.
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Noise and hypertension: Testing alternative acoustic indicators
Peter Lercher, Reto Pieren, Jean Marc Wunderli
[Abstract]
Standard acoustic indicators are reasonable means for health impact assessment of average transportation noise exposure situations. However, in its Annex I.3, the environmental noise directive admits problem areas (low frequency, tonal or impulsive components, quiet areas), where the application of ”supplementary” indices may be necessary. Various indices accounting better for the traffic dynamics were proposed recently, with intermittency ratio (IR) being one of these. Hitherto, IR was mainly evaluated in the SIRENE study and the additional contribution was small. Our aim is to test, whether IR and another correlated indicator (Leq-L50) works similar in an alpine study, which covered main and side valleys with a different transportation mix.
In a secondary analysis (cross-sectional survey, N=2002, 80% participation) we were able to estimate IR and Leq-L50 for 1500 participants from the original data set. These indicators were tested against diagnosis and medication use of hypertension in multiple logistic regression analyses, adjusted for main confounders (age, sex, bmi, health status, sensitivity, education, area). The relationship with hypertension varied. Models explained 30 to 35% of the variance.
The contribution of both indicators differed between sources and whether they were used as additional covariate together with one of the standard indicators Lden or Lnight.
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Opportunistic In-Vehicle Noise Measurements assess Road Surface Quality to Improve Noise Mapping: Preliminary Results from the MobiSense Project
Luc Dekoninck, Wout Van Hauwermeiren, Joachim David, Karlo Filipan, Toon De Pessemier, Bert De Coensel, Wout Joseph, Luc Martens, Dick Botteldooren
[Abstract]
The quality of road pavements affects noise emission caused by tire-road interactions. This affects the health and well-being of residents near these roads. The quality of the third-party data to determine the road surface noise classification is poor to not existing. Road pavement quality degrades over time due to wear, accidents, and infrastructure works. These local features are not included in the current noise mapping methodology.
The aim is to assess the quality of the road surface by performing opportunistic noise measurements inside vehicles. In the demonstrator phase of the project, 20 vehicles collect data by random sampling. Each vehicle driver performs his own trips and all data is combined in an artificial intelligence implementation. Engine noise is removed, vehicle specific speed corrections are applied and a rolling noise proxy in third-octave bands is determined. The rolling noise provides the road surface quality correction including the degree of local road surface degradation. In the project area, this local road surface quality is included in the rolling noise component for the noise map calculations for Flanders (Belgium).
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Estimation of psychoacoustic indices and annoying auditory sensations from sound pressure levels of urban road traffic
Catherine Marquis-Favre, Pierre-Augustin Vallin, Laure-Anne Gille, Wolfgang Ellermeier
[Abstract]
The day-evening-night noise level (LDEN) is often used as an index of noise annoyance. However, various other acoustical and psychoacoustical indices have been shown to improve noise annoyance models in comparison with models solely based on LDEN. The difficulty is to determine such indices in the field, since that requires numerous in-field recordings and tedious calculations. Alternatively, it might be possible to infer such indices from the equivalent sound pressure level (LAeq) of the noise sources used in the production of European noise maps. In the current study, such alternative is assessed from a perceptual point of view. A listening test was carried out with 30 participants in order to assess annoying auditory sensations evoked by road traffic noise at different distances from the main street in urban environments. For each auditory sensation, i) the relationship between the calculated psychoacoustic index and the rated sensation magnitude, as well as ii) the relationship between the auditory sensation and the LAeq of the noise sample were determined. These two relationships permitted to reliably estimate psychoacoustic index values from the LAeq of the road traffic noise.
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Noise indicator evaluation of road and rail traffic noise for indoor- outdoor differences in acoustic parameters
Manuel Lienhart, Peter Lercher, Michael Cik, Martin Fellendorf
[Abstract]
The two main objectives of the extensive project
”INTRANOISE” were to study the effect of road and rail traffic
sound on human sleep patterns and explore the relationship
between the subjective perception of 100 test persons and
objectively measured acoustic and physiological parameters.
The free field measurement arrangement includes binaural
dummy head measurements at the façade of the test
subjects’ home and within the bedroom, thus providing a
solid basis for evaluation. Indoor-outdoor differences were
assessed for various sound indicators and their relation with
people’s subjective and objective responses.
People are able to distinguish between the investigated noise
sources. This ability contributes to their subjective traffic noise
annoyance and can be linked to acoustic parameters.
Indoor noise is the dominant factor for physiological reactions
during sleep, leading to observable cardiovascular reactions.
The link between reactions and acoustic measurements
varies depending on the health indicator. Further studies are
needed to determine, which noise indicator is more
appropriate. This paper aims to describe people’s subjective
and objective response to road and rail traffic noise using the
difference of indoor-outdoor sound pressure levels. Keywords: Naturalistic traffic noise, annoyance, physiology,
sleep, noise indicators
I-INCE Classification of Subjects Number(s): 52.3, 52.4
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Relative Duration of Quiet Periods Between Events Influences Noise Annoyance: a Laboratory Experiment with Helicopter Sounds
Armin Taghipour, Reto Pieren, Beat Schäffer
[Abstract]
Besides the commonly considered time-averaged noise exposure metrics such as the Lden, additional indicators (e.g., maximum level, eventfulness, or relative quiet time) might improve the prediction of noise annoyance. An experiment was carried out on the effects of relative duration of quiet periods on annoyance reactions to helicopter noise scenarios, with the relative duration being the ratio of total duration of quiet periods in a sound scenario to its total duration. Recorded flybys were spatially reproduced with a hemispherical loudspeaker array in the laboratory. Ninety-second noise scenarios were prepared, containing two, three, four, or five flybys. All scenarios exhibited the same Leq of 66.5 dB(A), however, differed in the number of flight events and the exposure levels of individual flybys. Subjects were asked to make paired comparisons (i.e., two-alternative forced choices) for three pairs of scenarios. In each pair, one scenario had a larger number of flybys than the other; i.e., 4 vs. 2, 5 vs. 2, or 5 vs. 3. An analysis of the data of 56 subjects was carried out to investigate the effect of the relative duration of quiet periods between events on annoyance. The results indicate that longer quiet periods may mitigate noise annoyance.
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Organization:
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M. Tsuruta-Hamamura, Utsunomiya University |
A. Fiebig, TU Berlin |
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Acoustically-friendly products - Sound quality as an emission related product feature
André Fiebig
[Abstract]
People are permanently exposed to noise emitted by technical products in all areas of life. It is very much known that harmful noise effects are likely to occur by lower sound pressure levels as well. Moreover, residents complain about noises caused by neighbours according to the German Environment Agency with an upward trend. Moreover, acoustics is effectively one of the top product features regarding purchase decisions and it seems that consumer journals increasingly ask for more information regarding acoustic comfort and sound quality. It is already well known that massive differences in the sound quality of products on the market within the same product category exist. So far, some eco-labels have been established in the market considering noise emission, but they often work with simple noise level indicators only. But, how to reliably determine the sound quality of products representing sound perception and to widely spread this information to consumer organizations and final consumers? The paper will present some case studies on product sound quality related metrics beyond sound power and will discuss options and difficulties to use sound quality classifications for recognizing and supporting sustainable acoustically-friendly products.
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Sound Quality Improvement of Operation Sounds Emitted by MFPs
Masao Yamaguchi
[Abstract]
Multifunctional peripherals (MFPs) integrating multiple functions including copying, printing, scanning, and faxing functions in one unit are now widely used in offices.
In the MFPs, there are many sound sources which generate steady sounds such as noises emitted by motors and fans, and transient sounds such as collision noises between parts. It is difficult to identify sound sources for an improvement in a sound quality, because of including many sounds in operation sounds.
In this study, first we conducted a subjective sound quality evaluation of the operation sounds using the SD method. Then, we developed the metrics for the sound quality evaluation, based on the subjective sound quality evaluation and the psychoacoustic analyses. We also examined an influence of each sound on an impression of the operation sounds. Finally, we conducted a sensitivity analysis by calculating a contribution of each sound to the sound quality improvement of the operation sounds, by using the metrics we developed. Moreover, we took measures of the sound sources which were high contribution to the sound quality improvement. As a result, we achieved the improvement for the sound quality of the operation sounds emitted by the MFPs.
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A Design Method for UI-sounds for Electrical Appliances
Sanae Wake
[Abstract]
This paper considers the importance of designing User
Interface
sounds for products and explains a method to design
these
sounds.
There are many kinds of artificial sounds from electric
products.
For example, a microwave may make button feedback
sounds,
and play a ”completion melody” or make an ”alert”
sound.
These sounds are all part of the user-interface(UI) of a
product,
and are therefore called ”UI sounds.”
UI sounds should be designed to be functional, in other
words to
clearly convey messages to users, as well as thoughtfully
designed to suit a particular product.
To design UI sounds, it is first necessary to determine the
purpose
of each sound. Then sounds should be designed logically
in
each element, such as timbre, pitch, on/off pattern etc.,
to be
functional. For example, an alert sound has to repeat in
short
intervals to let the user intuitively know there is danger.
On the other hand, sounds should be designed like art or
music.
Since UI sounds follow an operation sequence, each
sound is
one component of a total UI sound stream.
When UI sound in products are well-considered and
designed,
usability of the product goes up and the product becomes
more attractive to consumers.
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Organization:
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S. Töpken, Carl von Ossietzky University, Acoustics group, Oldenburg, Germany |
T. Masayuki, Kyushu University |
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Sound Quality Evaluation of Refrigerated Truck Noise
Weonchan Sung, Patricia Davies, J. Stuart Bolton
[Abstract]
Noise from refrigeration units on trucks can be a problem,
particularly when the trucks are parked near residences. The
development of a sound evaluation method that takes into
account the strength of all sound attributes affecting people’s
responses including level is described here. Such a model,
coupled with a sound prediction methodology, would be
helpful to unit designers who are concerned with optimizing a
unit’s acoustical performance. The analysis of human subject
responses to a variety of refrigerated truck sounds (in the third,
of a series of three subjective tests) is described. Sixty
participants rated 25 recorded and digitally modified
refrigerated truck sounds as well as 25 residential HVAC
equipment sounds. Fourteen sound quality metrics were
evaluated as potential variables in a model to predict the
average annoyance ratings of the refrigerated truck sounds. A
rate-change-of-level metric that captured the impulsive
character of the refrigerated truck unit noise was developed,
and the best performing model contained it, a level metric and
a spectral balance metric. The performance of the model when
used with signals and ratings taken from other tests is described.
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Quantification of the unpleasantness of fan noise in the form of preference-equivalent levels
Stephan Töpken, Steven Van De Par
[Abstract]
In a previous study of the authors, a semantic differential was used to rate a broad variety of fan sounds, which were equalized in A-weighted level. The factor analyses of the results indicated six perceptual dimensions and five groups of sounds whereby the pleasant and the unpleasant sounds differed mainly with respect to the first three perceptual dimensions. A loudness analysis of the signals showed systematic differences in the patterns of the specific loudness for the different groups of sounds. In the present study, loudness- and preference-equivalent levels were measured for typical sounds from the three major groups of sounds to enable an interpretation of the findings obtained for sounds with equal A-weighted levels in the form of level differences. It turns out that the group of unpleasant fan sounds needs to be reduced by 10 dB compared to the group of pleasant sounds in terms of median overall levels to become equally preferred. Furthermore, level differences of up to 8 dB were found between the preference- and the loudness-equivalent levels for the group of unpleasant sounds. This difference is smaller for the other two groups of sounds.
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“Humming” or “Hissing”? - Psychoacoustical investigation of sounds from heat pumps
Marc Schneider, Carolin Feldmann
[Abstract]
Heat pumps, which become increasingly popular as heating systems for houses, with the build-in fan as main acoustic source could be annoying for the neighborhood.
Previous jury tests with a semantic differential reveal that besides the psychoacoustic loudness and sharpness, temporal structures in a sound signal have a significant impact on the quality evaluation of fan noise. In this context, the blade passing frequency (BPF) and its harmonics play an important role. It emerges, that distinctive sound patterns rated ”helicopter-like” by acoustic experts especially when tonal signatures in the signal are amplitude modulated with BPF. From a further jury test based on synthetic generated sounds, which were derived from measured fan sounds in heat pump applications, two different evaluation cluster with opposing assessments were determined: one major group, which prefers the ”hissing of the fan” and a smaller group, which favors the ”humming of the electric motor”. In an attached interview some participants of the test described a ”harmonious interaction” of the two mentioned sensations as ideal. By changing the blade shape of an axial fan, design optimizations should be investigated psychoacoustically. Therefore the annoyance will be predicted based on the previous results and directly measured in a jury test.
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HVAC noise perception in car cabin: a preliminary comparison between ICEVs and HEVs
Massimiliano Masullo, Katsuya Yamauchi, Yumi Nakatani, Luigi Maffei
[Abstract]
Noise in car cabins is a key aspect of the drivers/passengers’ comfort experience. It is produced by the aerodynamic interactions of the vehicle and the rolling, by the engine and all the auxiliary systems installed in the engine compartment. Although cars manufacturers focused their efforts to reduce the noise transmission inside the cabin, for Internal Combustion Engine vehicles (ICEVs) the mainly noise source is still the engine noise. Nevertheless, with the advent of hybrid/full electric vehicles (HEVs) in the automotive industry, the engine noise has been reduced dramatically, especially at low rpm, leaving emerging the noise of auxiliary sources. One of them is the air conditioning noise.
Although doesn’t exist a consolidated model to describe HVAC noise perception inside the vehicles, the paper intend to explore the expected changes of the drivers/passengers’ auditory perception from ICEVs to HEVs.
The acoustic and psychoacoustic metrics of different binaural recordings, carried out inside ICEVs and HEVs, are showed and compared. Then, they were analysed according some existing sound quality indexes and commented.
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Improvement of Construction Machine Noise
Takeo Hashimoto, Shigeko Hatano
[Abstract]
Therre are several factors cotributing booming sensation of interior noise of a construction machine. These are fan noise, intake and exhaust noises. In order to see the effect of the treatment of these factors to the perception of booming sensation, subjective experimets were conducted. As a result, the reductoin of fan noise is effective counter measure for the reduction of booming sensation.
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Study on the Evaluation Index of the Tonal Noise Components Generated from Small Fan
Takefumi Nakano, Gaku Minorikawa
[Abstract]
In this study, the new evaluation indices are proposed. These are named Total Tone-to-Noise Ratio and Total Prominence Ratio based on tonal components evaluation indices Tone-to-Noise Ratio and Prominence Ratio defined in the annex such as ECMA-74. These evaluation indices are total values of TNR or PR of each tonal component included in the noise. Therefore, it is possible to evaluate multiple tonal components individually or comprehensively. In our previous study, it has been confirmed that Total TNR and Total PR are effective evaluation indices for subjective annoyance in small axial and centrifugal fan. This study is attempted to determine their subjective evaluation thresholds with the jury test.
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Acoustic attenuators for aggressive environments
Chris Van Dijk
[Abstract]
Alara-Lukagro has conducted research on the essence of
acoustic attenuators and in recent years on the possibility to
create broadband attenuators without absorbing materials. This
could be done using friction of the acoustic wave. An
experimental setup using micro-perforates shows how the
potential attenuation of systems without absorbing material
relates to attenuators with absorbing materials, quantitatively,
for both broadband attenuation, flow generated noise and
friction losses. The results will be presented, the principles
explained and the potential quantified.
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Organization:
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A. Fiebig, TU Berlin |
K. Ohtomi, The University of Tokyo |
S. Sack, KTH, Stockholm, Sweden |
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Acoustic validation of calculation software for ducts, panels and room acoustics
Chris Van Dijk
[Abstract]
Calculation methods for industrial noise calculation have been
around for decades. In order to predict such situations software
was developed completely in line with international noise
mitigation standards. Different acoustic phenomenon by
interaction can have a significant influence on the result, just like
accuracy of the methods, geometries and environmental
factors. Alara-Lukagro is an acoustic construction company and
specialized in noise control and is expected to give noise
guarantees. In a number of field measurements these methods
are evaluated. Methods that show high discrepancies were re-
evaluated, using physics, mathematics, and FEM simulations. This
lead to a number of conclusions and recommendations.
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Scenarios for embedding AI in Acoustic Design. Exploiting applications at several design stages
Andrea Giglio, Ingrid Paoletti
[Abstract]
Holistic methods for acoustic design project are increasingly developed, combining the knowledge gained from physical acoustic researches, early stage noise mapping, architectural parameters and psychoacoustic.
The aim of these methods is to combine subjective and objective parameters to achieve more performative outcomes to satisfy comfort needs.
These methods require a methodological instrument to structure the design problem and to develop an heuristic mechanism acquiring broad rules to handle the specific solution.
Today, digital tools handle large complex problems evolving from generative into intuitive tools such as the Artificial Intelligence (AI).
AI, through the deep learning, discovers intricate structure in large data sets, by using the backpropagation algorithm, to indicate machine’s way to change its internal parameters thus helping to develop unpredictable innovative solutions.
The paper intends to develop a theoretical background based on bibliographic survey of AI in architecture, with a focus on acoustics. The resultant reading grid will be implemented by a custom survey sent to main stakeholders of acoustic design aiming at understanding strength and weaknesses of its application.
Scenarios will finally be traced exploiting the use of AI at different stages in order to understand its level of usage and opportunity to integrate in acoustic design methods.
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Organization:
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A. Raake, TU Ilmenau |
N. Zacharov, FORCE SenseLab, Denmark |
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Methodologies for Assessment of Speech and Audio for Optimized Quality of Experience
Dan Darcy
[Abstract]
Optimized Quality of Experience (QoE) in audio depends on
preservation of creators’ artistic intentions, from dialog intelligibility
to spatial arrangement of sound. Numerous challenges exist in
measuring audio sensation over new or advanced systems such as
mobile audio, spatial audio playback systems, and augmented or
virtual reality environments. The impact of these systems on dialog,
which is critical to the success of creative storytelling, requires
strategies to measure and optimize characteristics such as clarity,
naturalness, and intelligibility of speech. We describe a range of
experimental approaches to assess and drive audio algorithm
development. First, we recognize the importance of and expand on
classical methodologies such as forced-choice signal detection,
methods of adjustment, and magnitude estimation to assess or
validate system performance, and to understand end user
preferences and the perceptual metrics underlying them. Second, we
have developed a framework for objectively capturing spatial and
timbral properties of audio used to optimize spatial audio systems
along salient perceptual parameters most critical for high QoE.
Finally, we utilize physiological measurements such as pupillometry,
EEG, thermal imaging, and electrodermal activity to make direct
estimations of the relationship between auditory perception and
biological and neurophysiological states, such as cognitive load during
speech comprehension tasks.
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Evaluation of an immersive audio experience using questionnaire and interaction data
Jon Francombe, Kristian Hentschel
[Abstract]
Device orchestration is the concept of using an orchestrated set of connected devices to reproduce or augment a media experience. To evaluate the concept applied to immersive audio experiences, an orchestrated audio drama was created and released on BBC Research & Development’s public-facing trial platform. The experience allowed users to easily connect personal devices, which reproduced sounds including speech and spot effects, ambient background sounds, and additional storyline content. The trial was evaluated through a simple questionnaire and user interaction logs. The 210 questionnaire responses suggested a positive reaction (mean rating of 4 out of 5 stars, 72% loved or liked using phones as speakers, and 80% would use the technology again). The interaction logs (for 2174 sessions) showed reasonable engagement, with 20% of sessions reaching the end of the content. For sessions lasting over 1 minute, over 60% had at least one extra device connected. The results suggest useful strategies for optimising engagement with orchestrated experiences. There are limitations and challenges to drawing strong conclusions from large-scale trials in uncontrolled conditions (including unrepresentative sampling of the population and lack of context for logged interactions). However, benefits include larger sample sizes and greater ecological validity than in controlled laboratory testing.
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A Methodology and a Tool for Interchangeable Reproduction of Sound Samples in Listening Tests Through Different Sound Reproduction Systems
Dominik Kisić, Marko Horvat, Kristian Jambrošić
[Abstract]
A common practice in the execution of listening tests is to choose the
sound reproduction system based primarily on its availability, with
little regard for its potential influence on the results of such tests. To
examine the magnitude of this influence, a methodology is proposed
and a tool is developed for presenting sound samples through
different sound reproduction systems. The tool is equipped with a
user interface that enables easy and seamless change of the sound
reproduction system, as perceived by the listener. The design of the
tool eliminates the need for any kind of intervention that would
interrupt the listening session, such as on-the-fly system
reconfiguration or calibration. The goal is to achieve
interchangeability between binaural reproduction using headphones
and 3D Ambisonics reproduction using a multichannel loudspeaker
system as typical spatial audio reproduction systems. However, the
extension to two-dimensional amplitude panning systems such as
stereo or 5.1 surround is considered as well. The software design of
the tool is based on Cycling 74 Max programing language. The tool is
to be used in listening tests focused on subjective evaluation of
overall acoustic comfort, in relation with the quality of sound
insulation in closed spaces and their room-acoustical properties.
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Parallel task contexts in QoE testing. Can EEG help to understand the results?
Jan Holub, David Panek
[Abstract]
The contribution deals with novel methods of subjective voice and audio transmission quality testing. The common denominator of these methods is to move subjective testing from traditional laboratory standards closer to the conditions of real use of tested technologies while maintaining the test repeatability and reproducibility. In particular, we will describe tests deploying parallel task (e.g., according to the new ETSI TR 103 503) where test subjects are asked to perform a secondary task during the subjective test.
Some achieved results are counter-intuitive and somewhat exciting (e.g., the intelligibility of specific speech samples is statistically significantly higher in the parallel-task testing than in regular intelligibility tests) but quite complex. To understand better the influence of parallel task on the human auditory processes in the brain, electroencephalography has been deployed during our experiments, and EEG signals of the test persons have been acquired. The results, illustrating the differences in human brain activity in regular subjective testing versus parallel task testing will be presented and analyzed.
Tests scenarios using virtual reality to generate repeatable and laboratory-feasible parallel tasks will also be quickly presented.
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Pupil Dilation Reveals Changes in Listening Effort due to Energetic and Informational Masking
James Woodcock, Bruno Fazenda, Trevor Cox, William Davies
[Abstract]
Pupil dilation has previously been shown to be a useful
involuntary marker of listening effort. An inverse relationship
between pupil diameter and signal to noise ratio has been
shown when speech is energetically masked by noise. The work
reported here aimed to investigate whether this relationship also
holds for informational masking. Informational masking is a
concept used in soundscape research to represent the
distraction from the target sound that comes from a masking
sound that is also highly salient. To investigate the effect of
informational masking on listening effort, eighteen normal
hearing participants completed a speech-in-noise task in which
they were asked to identify words in short sentences presented
in combination with four different types of masker (competing
speech, speech modulated noise, and urban and nature
soundscapes) at different levels of energetic masking set using a
distortion weighted glimpse proportion model. Time varying
pupil dilation was measured over the course of each sentence
presentation. Repeated measures ANOVA show a significant
main effect of both the level of energetic masking and the
masker type on the mean pupil dilation (p < 0.05). These results
suggest that pupil dilation reveals changes in listening effort due
to both energetic and informational masking.
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Prediction of speech and noise quality for super- wideband and fullband transmission
Jan Reimes
[Abstract]
ITU-T Recommendation P.835 provides a well-
established listening test procedure for the auditory
performance evaluation of devices containing signal
processing components for noise reduction. In contrast
to ”classical” listening tests according to ITU-T P.800, test
subjects are asked for independent votes for speech
quality (S-MOS), noise intrusiveness (N-MOS) and
overall/global quality (G-MOS). However, since auditory testing is time-consuming and
expensive, several instrumental models for quality
prediction according to ITU-T P.835 were developed
and standardized in the past. These models are already
commercially available and widely used in industry for
many years. However, so far only narrowband and
wideband speech signals were supported. As devices
with a larger acoustic bandwidth (e.g., terminals
supporting the EVS codec) are currently entering the
market, an extension to super-wideband and fullband
mode was necessary. The new prediction model was developed in close
interaction of several standardization committees. A
common auditory testing framework was specified and
a considerable number of auditory databases for
training and validation was created. This contribution
provides an overview of the final prediction model,
which was recently introduced and standardized as ETSI
TS 103 281. In addition, several prediction results from
validation databases are presented.
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A System for Instrumental Evaluation of Audio Quality
Magnus Schäfer
[Abstract]
An approach for instrumental assessment of audio systems is presented in this contribution, which is based on binaural recordings of real music signals as well as measurement signals. Any system for instrumental assessment of audio quality should ideally be able to replicate the perception of a human test subject. An auditory test design based on music signals was recently devised that was shown to lead to reliable test results even with naïve test subjects. The test utilizes three individual attributes (timbre, distortions and spaciousness) along with a judgement on overall quality to quantify the perceived quality. Several auditory tests were conducted with the proposed design to collect training material for the system that is presented in this contribution. The system consists of two main components: An analysis stage that contains specific components for the individual attributes and a trained regression that utilizes the results of the conducted listening tests to establish a relation between the analyses and human perception. This contribution presents an overview of the assessment system, highlights the interaction between analyses and quality perception, and makes a comparison with auditory results.
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Organization:
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S. Narayanan, University of Southern California |
H.-W. Gierlich, HEAD acoustics GmbH |
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RTF-Steered Binaural MVDR Beamforming Incorporating an External Microphone for Dynamic Acoustic Scenarios
Nico Gößling, Simon Doclo
[Abstract]
Noise reduction algorithms for head-mounted assistive listening devices are crucial to improve speech intelligibility and speech quality in noisy environments. A well-known binaural noise reduction algorithm is the binaural minimum variance distortionless response (BMVDR) beamformer, which can be steered using the relative transfer function (RTF) vectors of the target speaker. Aiming at improving the noise reduction performance, recently the usage of an additional external microphone has been explored. In this contribution, we consider the recently proposed spatial coherence (SC) method to estimate the extended RTF vectors, assuming that the noise component in the external microphone signal is uncorrelated with the noise component in the head-mounted microphone signals. Simulation results for a dynamic scenario with a moving speaker in a reverberant room with diffuse noise show that the SC method yields a slightly better performance than the widely used covariance whitening method at a much lower computational complexity. Moreover, simulation results show that the RTF-steered BMVDR beamformer filtering all microphone signals outperforms the RTF-steered BMVDR beamformer filtering only the head-mounted microphone signals, a fixed BMVDR beamformer steered towards the frontal direction as well as the external microphone signal. Funded by the Deutsche Forschungsgemeinschaft (DFG) - Projektnummer 352015383 - SFB 1330 B2.
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Comparison of Ideal Mask-Based Speech Enhancement Algorithms for White Noise and Low Mixture Signal-to-Noise Ratios
Simone Graetzer, Carl Hopkins
[Abstract]
The intelligibility of noisy speech can be improved by applying an ideal binary or soft gain mask in the time-frequency domain for signal-to-noise ratios (SNRs) that are typically between -10 and +10 dB. In this study, three mask-based algorithms are compared when applied to speech mixed with white Gaussian noise at much lower SNRs (from -29 to 0 dB). These comprise an ideal binary mask with a local criterion set to 0 dB or the mixture SNR, an alternative binary mask comprising [0.1,1] rather than [0,1] gains, and an Ideal Ratio Mask (IRM). The performance of Short-Time Objective Intelligibility (STOI) is compared with that of other monaural intelligibility metrics that can be used before and after mask-based processing. The research questions are, firstly, which of these enhancement algorithms results in the highest intelligibility scores in listening tests and, secondly, which metric best predicts intelligibility scores for noisy and mask-processed speech? The results demonstrate that IRMs can be used to obtain near maximal speech intelligibility (> 90% for sentence material) even at very low mixture SNRs. They suggest that caution should be taken when choosing intelligibility metrics to ensure that the metric has been validated for the mixture SNRs considered.
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Reverberation-induced speech improves intelligibility in reverberation: Effects of taker gender and speaking rate
Nao Hodoshima
[Abstract]
Humans adjust their speech when speaking in noise to enhance
their intelligibility (known as the Lombard effect). Speech spoken
through noise (noise-induced speech) is generally more
intelligible than that spoken in quiet when heard in noise. The
author has reported that ”reverberation-induced speech” also
improves speech intelligibility in reverberation, although the
masking of noise and reverberation differed. The present study
investigates whether reverberation-induced speech intelligibility
depends on the gender and speaking rate of talkers. Four
talkers (two males and two females) spoke under quiet (Q) and
reverberation (R) conditions. In R, the reverberant speech was
fed back to the talkers via headphones. Then speaking rate was
adjusted to 1.2 (fast), 1.0 (original), and 0.8 (slow). 18
participants performed word identification tests in reverberation
(reverberation time of 3.6 s). The results showed that speech was
significantly more intelligible under R than Q, male talkers were
significantly more intelligible than females, and slow speech was
significantly more intelligible than fast speech. And the
improvement in speech intelligibility under R compared with Q
was not significantly affected by the speaking rate. These results
suggest that reverberation-induced speech regardless of the
speaking rate might increase intelligibility of announcements in
public spaces such as airports.
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Madurese Speech Synthesis using HMM
Roudhotul Rouf, Dhany Arifianto
[Abstract]
This research is intended to study Madurese language which
may be the only local language in Indonesia classified into a
tonal language. However, the Madurese is not only under-
documented in term of phonetics but under-resourced as well.
The first step was developing the Madura language voice
database. The initial results are limited to the fundamental
frequency contour of native male and female utterances. We
used a well-known technique called Mel Frequency Cepstral
Coefficient (MFCC) to obtain acoustic cues, and the cues were
further processed by observing the delta, Δ Cepstrum for
velocity change in an utterance and the delta-delta Δ2, to
indicate the acceleration or deceleration of the acoustical
cues change over time, respectively. We used instantaneous
frequency tracking in STRAIGHT. We obtained results were
compared to the well-established TIMIT database in English and
Indonesian Speech Database. From the initial results, the Δ and
Δ2 indicated that the Madurese showed a rapid change in
both time- and frequency-domain cues. Although the findings
may be far from conclusive because the Madura island has four
different regions that have its own accent that slightly different
from each other. Currently, the on-going research aim is
towards a high-quality Madurese speech synthesis.
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HMM-Based Speech Synthesis System with Expressive Indonesian Speech corpus
Elok Anggrayni, Dhany Arifianto
[Abstract]
In this paper, we present a result of HMM-based speech
synthesis system applied to Indonesian expressive speech
corpus. The purpose is to observe speech quality of synthesized
speech. Firstly, we selected expressive Indonesian
conversation from movie, novel, and drama transcript. We
developed a speech database based on a phonetically
balanced sentence set in which consist of 33 Indonesian
phonemes and its IPA symbols and formed 500 sentences. Three
expressive styles were applied, namely happiness, sadness, and
anger. We hired two professional theater artist from Arts Council
of Surabaya, Indonesia to utter the sentences. We used 50
sentences of each expressive style for training data. The
expressive style-dependent decision trees achieve prosodic
conversion. In the objective test using MCD method showed the
best score for happiness expressive style with a score of 4.2, for
sadness with a score of 5.13 and 5.18 for anger in 50 training
data. Subjective test with Mean Opinion Score was collected to
measure naturalness for happiness, anger, and sadness with a
score of 3.51, 3.38, and 3,0, respectively. The result showed that
the quality of the synthetic speech was high in term of
naturalness.
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Organization:
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J. Jung, DTU Electrical Engineering, Acoustic Technology |
G.H. Yoon, Hanyang University |
J.S. Jensen, Technical University of Denmark, Department of Mechanical Engineering |
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Radiation optimization of piezoelectric plates
Olivier Doaré, Emil Garnell, Corinne Rouby
[Abstract]
The goal of the present work is to control the modal
dynamics of a radiating piezo-electric plate in order to reach
directivity or efficiency criteria. In this context, we study the
dynamics and radiation behaviour of a piezoelectric plate
biased by a static pressure on one side. This pressure
inflates the plate and its forced dynamics around this
equilibrium position are then studied. With the help of von Karman plate theory, the static
equilibrium position of a plate of any shape is sought for
before writing its linear dynamics around this equilibrium
position. This two-step problem is solved numerically using a
finite elements software (Freefem++). This results in a
series of eigenmodes and eigenfrequencies for each value of
the static pressure difference. Next the modal forces due to
the piezoelectric coupling are calculted. By considering a
baffled plate, the acoustical radiation is computed using the
Rayleigh integral. Based on the numerical results, the influence of the different
parameters of the problem are analysed. The numerical model is used to
perform a parametric optimisation of the system. A
prototype is currently under development, and some
comparisons between experiments and numerical results will
be presented.
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Optimization of Realistic Loudspeaker Models With Respect to Basic Response Characteristics
Daniel Gert Nielsen, Finn T. Agerkvist, Jakob S. Jensen
[Abstract]
A numerical model for optimization of loudspeakers mounted in infinite baffles are
presented in this paper. The optimization is carried out by using objective functions
based on basic response characteristics for loudspeakers such as the on axis frequency
response. In-order to create a realistic model of the speaker we include the excitation of
the speaker from the motor system. The interaction between fluid and structure is
modelled with the finite element method for exterior problems. As most loudspeaker
drivers are inherently symmetric, the presented model is therefore a 2D axisymmetric
finite element model. The exterior domain is modelled with perfectly matched layers,
which ensures free-field radiation conditions. Optimized designs for a selection of
objective functions are presented and discussed.
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Design of damping layer by topology optimization and Non-Negative Intensity
Wenchang Zhao, Haibo Chen, Steffen Marburg
[Abstract]
Non-Negative Intensity (NNI) is a quantity which could avoid the near-field cancellation effects in sound intensity and provide direct visualization of surface contributions to sound power. Hence, the NNI and its variants are implemented to be the objective function of topology optimization for damping layer design. Regarding vibro-acoustic systems, the structural vibrations are analyzed by the finite element method (FEM), and fast multipole boundary element method (FMBEM) is used for the acoustic analysis. A two-way coupling is established between the structural and the acoustic domains. By using the FMBEM and the implicitly restarted Arnoldi method (IRAM), the eigenvalue analysis for the symmetrized acoustic impedance matrix is performed with efficiency. Then, the NNI can be easily computed based on the eigen-solutions and the FEM-FMBEM analysis. Further, these eigen-solutions can be recycled in the optimization iterations since they are independent of the solutions of the coupled system. This reduces the computational efforts. For calculating the gradients of the objective function, the adjoint variable method is applied. With the evaluated gradients, the optimization problem is solved by the method of moving asymptotes (MMA) and the optimized distribution of damping layer is obtained.
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Reducing Offshore Pile Driving Noise: Shape Optimization of the Impact Hammer
Elin Klages, Jonas Von Pein, Stephan Lippert, Otto Von Estorff
[Abstract]
Offshore wind energy is an emerging source of electricity
generation and new offshore wind parks are under
construction around the world. The foundation of offshore
wind turbines is most often realized driving steel monopiles
into the sea bed using impact hammers, a technique that
causes high underwater sound pressure levels threatening
to harm marine mammals and other sea life. In order to protect the marine fauna, limiting values have
been introduced by several countries. Their fulfillment until
now has been possible applying state-of-the-art sound
mitigation systems. However, rapidly increasing
dimensions of wind turbines with accordingly increasing
pile diameters require an ongoing development of sound
mitigation systems. Nowadays applied sound mitigations systems, e.g. bubble
curtains, reduce the propagation of the emitted sound,
but do not affect its generation. Therefore, modifying the
sound source, i.e. the hammer and the pile, to reduce the
generation of the sound, has recently gained some
attention. This contribution focuses on the development of a
hammer that causes less noise but is still capable of
driving the pile. First results of optimizing the shape of the
impact hammer regarding its acoustic characteristics are
presented.
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Structural topology optimization for repeated eigenvalues with the adjoint sensitivity analysis
Gil Ho Yoon, David Ruiz, Alberto Donoso, Jose Carlos Bellido
[Abstract]
This research presents a new adjoint sensitivity analysis for repeated eigenvalues and topology optimization scheme. Despite the vast researches regarding the eigenvalue optimization problems, there are few examples considering mode shapes in topology optimization. For the structural topology optimization with the eigenmodes especially with the repeated eigenmodes, we propose to use a general adjoint variable approach developed for structural topology optimization for the repeated eigenvalue problem. Furthermore, this research presents a new method to track repeated eigenmodes during optimization. To validate the present approach, we compare the sensitivity values from the finite difference method, the direct method (the Dailey’s method), and the adjoint sensitivity analysis method. To show the validity of the present approach, some benchmark problems are considered and included.
This work was supported by the National Research Foundation of Korea (NRF) grant fundedby the Korea government (MSIT)(No.2018R1A5A7025522).
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Organization:
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S.-H. Lee, Korea Inst. of Machinery and Materials |
J.A. Cordioli, Federal University of Santa Catarina |
B. Van Der Aa, Royal SMIT Transformers |
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Measurement and Simulation of Vibration Acceleration of a High-Voltage Three-Phase Gapped-Core Reactor
Bart Van Der Aa, Bart Wijnhoven, Timo Overboom, Luc Dorpmanns
[Abstract]
In this work, vibration acceleration of a high-voltage three-phase gapped-core shunt reactor is studied. The active part is oil-immersed, has three wounded legs, and two flux return limbs. Measurements are done using fibre optical accelerometers, which allows one to install sensors at locations otherwise not accessible for conventional piezoelectric accelerometers due to presence of high electromagnetic fields. Measured results are compared to simulation results, where forcing of the structure is assumed to be dominated by Maxwell forces. The study shows good quantitative agreement at twice the power frequency. Results obtained give insights in the dynamic behaviour of the gapped-iron core, which can be used for optimization of noise and vibration behaviour.
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Increased Radiation Efficiency Using Band Gap Effect
Jaesoon Jung, Cheol-Ho Jeong, Jakob S. Jensen
[Abstract]
Periodic arrangements of elements, such as scatterers,
local resonators, and inertial amplifications can induce
bandgap phenomena, which result in vibration isolation in
mechanical systems due to a significant reduction in the
energy transmission from source to receiver. This effect has
led to vibro-acoustic applications focusing mainly on the
reduction of structure-borne noise. Here, we propose to
broaden the application range by utilizing the bandgap
effect to increase the efficiency of sound radiation from
vibrating mechanical structures. The increase in radiation
efficiency is caused by the enhanced vibration
localization near the excitation point. In order to illustrate
the phenomenon, a one-dimensional mass-spring model
with periodically placed local resonators is examined and
with this theoretical model vibration localization and
sound radiation efficiency are analyzed. An experimental
validation is carried out on a plate structure. For the
specific case, an increase in radiation efficiency up to 10
dB at frequencies above 2 kHz is demonstrated.
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Numerical simulation of vibration damping by granular materials
Masao Takeshima, Asakura Takumi
[Abstract]
In recent years, there have been some researches related
to damping of plate vibration using granular materials. On
the other hand, the numerical simulation scheme on the
damping effects of arranging granular materials on the
planar structure may be still on a development stage. In
this study, we tried to simulate the damping effect of
bending vibration of the plate by granular material using
following methods. One is the Discrete Element Method
(DEM) for simulating the motion of the granular materials,
where the other is the Finite-Difference Time-Domain
(FDTD) method for vibration analysis of the thin plate. Then,
by using a coupled DEM-FDTD scheme, we simulated the
influence of the granular material on the plate vibration,
confirming the validity of the proposed method by
excitation test on an acrylic plate model. In addition, we
have also confirmed that the dependency of the
damping effect of the plate vibration on the extent of the
external excitation force when the granular materials are
arranged on the plate.
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On the estimation of DLF of highly damped structures
Julio A. Cordioli, José V. Monteiro, Mathias Hinz, Guido T. Santos, Israel Pereira
[Abstract]
The damping loss factor (DLF) of a structure represents numerically its power
dissipation capacity under a given excitation field, and it is an important input
parameter for numerical models based on Statistical Energy Analysis (SEA). Different
experimental approaches are described in the literature to obtain the DLF from complex
structures. Nevertheless, the application of such methods to estimate the DLF of highly
damped structures, for example fuselage panels with viscoelastic materials, may be
questionable, With its focus on the DLF computation for highly damped structures, this
paper intends to present a broad overview through a numerical-experimental approach
on three of the main methods presented in literature: the Power Input Method (PIM), the
Periodic Structures Theory and a more recent approach known as Inhomogeneous Wave
Correlation (IWC). Numerical and experimental results are discussed, presenting the
nuances for each of the implementations, as well as their main advantages and
limitations under the given context.
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Assessing the similitude of vibrating plates
Christian Adams, Joachim Bös, Tobias Melz
[Abstract]
The similitude of vibrating structures allows for transferring vibration responses from a scaled structure to an original structure using scaling laws. Numerical calculations and experimental measurements show that complete similitude conditions such as equal damping are essential to accurately transfer vibration responses from the scaled structure to the original structure. However, complete similitude conditions can hardly be achieved in practice, which causes the mode shape order to change. Nevertheless, the scaling laws can transfer the vibration responses from the scaled structure to the original structure with a sufficient accuracy in some cases. In this paper, the vibration responses of rectangular plates with various sizes, free and simply supported boundary conditions, as well as shaker and hammer excitation are obtained from laser Doppler vibrometry. The vibration responses are transferred from the scaled rectangular plates to the original one using scaling laws, which are directly derived from numerical calculations. The accuracy of the transferred vibration responses is assessed by comparing them with the measured vibration responses of the original structure, where several a-posteriori error measures like MAC analyses or Mahalanobis distances are used. The error measures are analyzed with respect to the plates’ sizes, their boundary conditions, and the excitation method.
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Construction of a vibroacoustic test bench for electric motors and investigation of the noise emission of an electric motor
M. Ercan Altinsoy, Florian Spanka, Markus Klemm, Hong-In Won
[Abstract]
Recently, the number and the performance of installed electric
motors in cars and household appliances have been growing
fast. The integration of motors to the everyday devices causes
that the sound quality of electric motors gains more and more
importance. For the acoustical optimisation of an electric
motor, the sound generation and radiation mechanisms of
airborne and structureborne sounds are of crucial importance.
The aim of this study is to develop and construct of a
vibroacoustic test bench for electric motors. Particularly the
mounting and loading are very critical issues for the
construction of the proper test bench. This talk will present the
results of the noise emission investigations on the motor.
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Robust Virtual Sensing of the Exterior Noise Radiation from a Complex Structure in Different Acoustic Environments
Sjoerd Van Ophem, Elke Deckers, Wim Desmet
[Abstract]
By using a numerical model of a vibro-acoustic system in
combination with a small set of measurements in a
Kalman filter, it is possible to estimate the sound pressure
at locations where no microphones are present. This can
be achieved with increased accuracy as compared to
using only the numerical model due to the inclusion of the
expected process and measurement noise. This
procedure is also known as virtual sensing. In this paper, a
model-based virtual sensor is built for a complex, deep
drawn structure that radiates sound into the free field. A
finite element model is used that is extended with infinite
element to approximate the Sommerfeld radiation
condition. Since the original model is too large to be used
in a Kalman filter, it is reduced by a stable Krylov based
model order reduction technique. The performance of this
virtual sensor is tested in a semi-anechoic chamber, and
additionally in several acoustical environments that do not
conform to the Sommerfeld condition in order to validate
whether the Kalman filter can handle the additional
uncertainties. The obtained results show that the filter
performs well, which means that the required modeling
effort can be reduced significantly by assuming free-field
conditions.
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The Effect of the Weld Type on Ensemble Average in SEA
Paweł Nieradka, Sebastian Szarapow
[Abstract]
Statistical energy analysis (SEA) predicts the average response
of a population. This population consists of similar vibroacoustic
structures subjected to white noise forces at high frequencies.
However, usually in engineering applications, the SEA method is
used to determine the response of a particular structure. It
results from the assumption, that in the case of complex
structures the response of a single member from the population
will not significantly differ from the average. The aim of the
research was to check the above assumption on a simple
structures. Six similar mechanical structures were tested. Each
structure consisted of two plates welded together at right
angles. The vibration (velocity) level reduction between the
plates was determined. From the point of view of the SEA, each
of the six L-shaped structures is represented by the same SEA
system composed of two 2D subsystems connected by line
junction. However, the resulting line junctions have been
achieved by two different welding techniques. This detail is
omitted during SEA calculations. Vibration reduction obtained
on a single structure was compared with ensemble average
covering the entire six-element sample. The influence of the
weld type on transmission was considered. The obtained results
were compared with the SEA prediction.
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Vibration damping capabilities of treatments with frequency and temperature dependent viscoelastic material properties
Martin Gröhlich, Marc Böswald, René Winter
[Abstract]
Due to their low mass and stiffening components like stringers and panels, aircraft fuselage structures are prone to vibration. In order to improve passenger comfort inside a cabin, specific vibration damping of such components would be helpful. Most common damping layouts contain a viscoelastic material as the main source of vibration energy dissipation, which is placed at specific positions of a structure. However, the material behavior of viscoelastic materials varies considerably with frequency and temperature and needs to be considered during the design process.
In this paper, a Finite Element (FE) approach for local modelling of viscoelastic damping is presented. The damping properties of a viscoelastic material are investigated with regard to its temperature and frequency dependence. For the frequency dependence, the material follows the equations of a generalized Maxwell model. The influence of the temperature is described by William-Landel-Ferry and Arrhenius shifts and is also implemented in the FE calculations. Using the FE method, the consequences of temperature and frequency influence on the vibration damping capability of different damping concepts are presented. A conclusion concerning the usage of the viscoelastic material in a wide temperature range is derived.
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Organization:
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Yue-Ming Li, Xian Jiatong University, China |
G. Chen, Xi'an Jiaotong University |
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Vibro-acoustic Behavior of Plates Considering Static Load Effect
Yueming Li, Di Wang, Qian Geng, Xiongwei Yang, Lei Wu
[Abstract]
Hypersonic crafts are subjected to extreme heating, static load,
acoustic excitation and engine vibration during service. The
natural characteristics, vibration and acoustic radiation are
affected significantly hereof, which makes great challenges on
the reliability and safety of the crafts. In this paper, we present
the general form of governing equations of plates considering
the thermal/static load effect. The static state under
thermal/static load is obtained firstly, and then the vibration and
acoustic radiation are calculated at the reference static
position. Numerical and experimental approaches are also
displayed. Based on the above, the effect of thermal/static
load are further analyzed. The results show that both the inner
stress and geometry shape will be changed under
thermal/static load. With the temperature increasing, the
softening effect of thermal stress in the pre-buckling state leads
to the decrease of frequencies, while in the post-buckling state
the stiffening effect of thermal buckling deformation leads to
the increase of frequencies. The variation of vibration and
acoustic radiation responses are in accordance with that of the
frequencies. Furthermore, the vibro-acoustic behavior under
static load is also discussed in detail. The present research
provides a foundation for the vibration analysis of hypersonic
crafts under multi-environment.
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Mode conversion of elastic waves by using anisotropic metamaterials
Xiongwei Yang, Yueming Li, Gang Chen
[Abstract]
As a fundamental phenomenon in elastic fields, mode
conversion between longitudinal and transverse wave modes
can frequently occur when a wave is incident on an elastic
discontinuity or passes through anisotropic mediums. Due to the
generation of a different wave mode, the phenomenon is
widely used in industrial and medical applications for
nondestructive tests and ultrasonic inspections.
Metamaterials are men-made subwavelength composites and
could be designed based on the local resonance mechanism.
Due to local resonance, effective physical properties of the
metamaterials could be zero or negative in desired frequency
ranges, leading to novel wave phenomena, such as low
frequency bandgap. Metamaterials could also be designed
with specific macroscopic anisotropy in specific direction. Such
anisotropic metamaterials, if designed to be nonresonant,
barely exhibit material absorption losses, and could be more
suitable for wave manipulating, such as acoustic magnifying,
invisibility cloaking.
In this work, we are concerned whether the resonance
mechanism can be introduced into mode conversion. Because
of the resonance mechanism, the effective properties of the
metamaterials can be highly frequency-dependent, which
means vast possibilities in finding new phenomena and
obtaining new designs. Our final goal is to pave the way of using
the frequency-dependency concept for mode conversion.
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Assessing Engineered Materials via Non-Destructive Impact Acoustics
Yishan Dong, Shahram Taherzadeh, David Sharp, James Bowen
[Abstract]
Acoustic testing allows inspection of the quality of
engineered products without influencing their final use. It is an
example of non-destructive testing (NDT), a wide group of
analytical techniques used to evaluate the properties of a
material, component or system without causing damage.
Particularly popular across the civil engineering, healthcare,
automotive, and aviation industries. One early example
originated via the wheeltappers during the Industrial
Revolution when railways came into prominence. An impact between two solid bodies produces surface and
body waves, the velocity and characteristics of which
depends on material properties and the nature of the
contact. Parameters such as surface roughness, density, and
modulus all influence the properties of these waves.
Measurement of these waves can therefore be used in the
evaluation of material and surface properties. Techniques which are currently employed for acoustic NDT
are often expensive and designed to test cracks or system
flaws. This work reports on recent attempts to develop a low
cost, non-destructive method for assessing the structure and
quality of engineered materials. The method, which involves
delivering a low kinetic energy impact to a sample, permits
correlation between surface and/or material properties and
the acoustic output from the impact event.
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Acoustic cloud based approach for Corona early detection on Hydropower Equipment
Jose Manuel Nieto Diaz, Paulo Teixeira, Manuel A. Sobreira Seoane
[Abstract]
The Hydropower Generation market is facing globally high
operational flexibility, compelling the operator looking
forward maximum availability and performance. Early detection of incipient electrical failure sources at the
generator winding, transformers and conventional substations
will help reaching these targets. This paper presents a monitoring system which ”hears” the
acoustic emissions produced through small electrical
discharges called ”Corona”. Corona effect produces
frequencies in the range between 7 and 19 kHz which can be
characterized by its acoustic emission. Acoustic Corona
monitoring can also be used combined with Partial Discharge
Monitoring Systems, improving early detection of incipient unit
failure The approach allows to continuously monitoring Corona using
control system data -including historical machine behaviour-
and applying cloud based machine learning algorithms. This
allows to cluster machine ”normal” behaviour and to detect
corona anomalies, allowing early warning before failures. The output provides a web-based visualization to the
operator, indicating unusual behaviour, such that the support
engineer can investigate possible issues in an early phase. The
automated system monitors the equipment constantly and
just raises a warning or alarm in case unusual corona acoustic
behaviour. This solution allows easy retrofit of existing hydropower plants,
as it requires no physical modifications of the equipment
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Analysis of SPL Reduction Possibilities Inside the WFI ATHENA Filterwheel Assembly
Wojciech Binek, Adam Pilch, Szymon Polak, Mirosław Rataj, Tadeusz Kamisiński
[Abstract]
Ariane 5 rocket produces very high sound pressure levels
during launch, what can influence structures located inside
the fairing. In Wide FieldImager (WFI) project, the main part of
the filterwheel assembly (FWA) is an extremely thin (∼240 nm)
filter of large area (170 × 170 mm), very sensitive to noise and
vibrations.
The aim of this study was to minimize the sound pressure
affecting filter surface. Several modifications of the filterwheel
assembly including e.g. different filterwheel angular position
or height were analysed using finite element method (FEM)
simulations. The possible modification options were greatly
limited by non-acoustical design factors. The results from
computer simulations were compared with measurements of
the prototype FWA. The results will be used in further design
process of WFI filterwheel assembly.
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Organization:
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N. Totaro, INSA LYON -LVA |
J.-G. Ih, Acoustics Laboratory, Dept. of Mechanical Engineering, KAIST |
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Attempt to Classify the Microphone Array Deconvolution Methods in Aeroacoustics
Christophe Picard, Quentin Leclere
[Abstract]
In the field of experimental aeroacoustics in wind tunnel, the number of existing methods is important and is increasing continuously so that it becomes difficult for Industrials to have a clear view of the respective advantages of each of them to make appropriate choice. Benchmark works are also plentiful and often focus on the performance of these methods in terms of spatial localization and computing time, more rarely in terms of quantitative estimation. We are interested here by these deconvolution methods that we aim to classify, and even to show in what extent all of these methods are equivalent. The approach adopted is to define a common protocol for controlling and monitoring the different algorithms: define same initial conditions, same convergence parameters and same exit conditions. All of the considered deconvolution algorithms are implemented with respect to the same protocol. Convergence and calculation time are monitored at each frequency. Finally acoustic power of source area are extracted from the obtained maps and compared together. Indeed, it is observed that within the range of few dBs the results are quite similar and we propose to classify the algorithms with their equivalent Lp-norm minimization function.
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Extensions of the orthogonal beamforming to identify multiple multipole sources
Xingjian Pan, Wei Kang Jiang
[Abstract]
Some aeroacoustic sources are analogous to multipole
sources, such as dipoles or quadrupoles. For these sources,
the conventional beamforming methods may yield
questionable source locations and strengths due to the
monopole assumption. With the aim to identify sources with
various directivities, a normalized-and-square beamforming
combined with an inversed method is proposed. Multiple
sources are separated by the eigendecomposition of the
cross-spectral matrix of array signals. The influence of source
directivities can be eliminated by modifying the
eigenvectors. After the sources are located, the source
strengths can be estimated by an inverse method using the
spherical harmonic expansions. The proposed procedure is
verified by numerical simulations of various multipole
sources and an experiment of an Aeolian-tone dipole
combined with a loudspeaker. It shows the potential to
identify the airfoil noise or the airframe noise with complex
directivities.
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Acoustic source detection inside a pipe using vibroacoustic beamforming: assessment of the array gain from virtual experiments
Souha Kassab, Sanae Serbout, Frédéric Michel, Laurent Maxit
[Abstract]
A monitoring technique based on vibration measurements is developed for detecting a leak of water into a sodium flow. However, leak-induced vibrations could be smeared in the ambient vibrations. In order to increase the signal-to-noise ratio, the conventionnel and MaxSNR beamforming treatments have been considered. In order to study their efficiency for the present application, one has developed a laboratory mock-up composed by a pipe coupled to a hydraulic circuit through two flanges. The source to detect is a sound emitter introduced into the pipe whereas the background vibration noise is controlled by changing the speed of the flow. In parallel, numerical vibroacoustic tools were developed for predicting the vibratory response of the pipe excited by a monopole source or a turbulent boundary layer. These models allow us to realize virtual experiments mimicking the behavior of our laboratory test case. The virtual signals induced by the monopole source to be detected and by the turbulent flow are used for assessing the performances of the vibroacoustic beamforming treatments. After a brief presentation of the vibroacoustic models, we present the array gain estimated using the virtual experiments for different configurations.
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Blind identification using inverse Patch Transfer Function (iPTF) method
Emmanuel Manu Dabankah, Nicolas Totaro, Jerome Antoni
[Abstract]
The main work for vibration and noise reduction and
control is to identify and localise the vibration and noise sources.
Many inverse methods, like Nearfield Acoustic Holography,
have been developed in acoustics in the last decades. Among
others, the iPTF method, allows reconstructing the acoustic fields
(pressure, velocity, intensity) directly on the vibrating structure
surface even when it has a complex shape. In addition,
measurements can be done in non-controlled acoustic
environments. The concept of iPTF is based on the application
of Green’s identity on any closed virtual volume defined around
the source. The reconstruction of sound source fields combines
discrete acoustic measurements performed at accessible
positions around the source with acoustic impedance matrices.
In the present work, blind identification of the vibratory fields is
proposed. The ”blindness” has here two meanings:
- identification of the velocity field of a vibrating structure can
be blind if obstacles mask parts of the structure to characterise.
- identification can be blind if the velocity field is the result of the
combination of several unknown sources and if one wants to
separate the contribution of each source.
Some numerical and experimental results will be shown to
illustrate both aspects of the blind identification.
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Bayesian Inference to Damping Identification of Fiber-Reinforced Composites from Experimental Modal Data
Sourav Chandra, Kheirollah Sepahvand, Christian A. Geweth, Ferina Saati, Steffen Marburg
[Abstract]
To study the dynamic behavior of composite structures, the
identification of elastic and damping parameters is essential. A
significant amount of variability arises to the modal response of
such structures due to the uncertainty in the fiber orientation,
geometry parameters and inter-laminar stress during
manufacturing process. Furthermore, there are various sources
of modeling errors, such as inaccurate boundary conditions,
material constitutive behavior, simplification and linearization of
the numerical model. In this paper, the Bayesian inference
technique is used to treat the problem of uncertainty of the
responses by updating the damping parameters of the finite
element (FE) model based on experimental test data. Bayesian
updating aims to minimize the probability of the error associated
with the behavior of the FE model and the actual behavior from
experimental data. The sampling based Markov Chain Monte
Carlo method is used for the numerical Bayesian inference. This
methodology identifies the probability distribution of the best
possible distribution for the damping parameters to predict
uncertainty of the actual structural responses using the
computational FE model. The application of the method will be
examined to update the FE model performing the numerical
modal analysis of a twelve layers fiber-reinforced composite
plate from experimental modal data.
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Selective identification of structural force distribution
Nicolas Totaro
[Abstract]
On the one hand, in the field of structural source identification,
some methods have been developed during the last decades
like the Force Analysis Technique, the Virtual Fields Method, the
2D spatial Fourier transform or the structural holography. The
proposed method clearly falls in the same framework and is
close to the philosophy behind the Virtual Fields Method.
On the other hand, in the field of acoustic source identification,
the inverse Patch Transfer Function has been recently proposed
to reconstruct the acoustic fields on a complex shape source in
any (stationary) acoustic environment. This method is based on
the concept of virtual acoustic volume. The method proposed
here tries to adapt this concept to structural identification.
Therefore, the proposed method is based on the use of a virtual
testing structure solved by a Finite Element solver. It allows
selecting what kind of distribution force one wants to identify. In
the example of a plate with welded stiffeners, if one is only
interested in localizing and quantifying the applied force, the
virtual testing structure has to include the stiffeners. If one is
interested in forces due to the welded points, the virtual testing
structure has to exclude the stiffeners.
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High Frequency Structure-borne Sound Simulations using a hybrid Dynamical Energy Analysis / Advanced Transfer Path Analysis Approach
Gregor Tanner, Satoshi Morita, Timo Hartmann, Martin Richter
[Abstract]
Dynamical Energy Analysis (DEA) is a mesh-based high frequency method modelling structure borne sound for complex built-up structures. This has proven to enhance vibro-acoustic simulations considerably by making it possible to work directly on existing finite element meshes circumventing time-consuming and costly remodeling strategies. In addition, DEA provides detailed spatial information about the vibrational energy distribution within a complex structure in the mid-to-high frequency range. DEA was successfully applied and validated to a structure borne sound calculation of an assembled agricultural tractor. Modelling solid structures is still a challenge for DEA, however, as it is based on 2D wave transmission calculations.
We propose a novel method to generate DEA elements based on measurement data in order to model solid parts of a complex structure. Advanced Transfer Path Analysis (ATPA) is employed to extract energy transmission characteristics of a structure. First, Frequency Response Functions are measured between interface points on a structure. Then a direct transfer function between each interface points is calculated by using ATPA. Finally, DEA elements connecting interface points and representing energy transmission characteristics of the structure are created based on the ATPA result. Applications of the method will be presented.
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Estimation method of input power from road to tire based on experimental SEA
Hiroki Nakamura, Kaito Sawada, Atsushi Kitahara, Toru Yamazaki
[Abstract]
A tire is a composite structure and a viscoelastic body, and its
mode density is high. Then, strict methods such as Finite-
Element-Method are difficult to apply. Under the circumstances,
the authors are conducting research for predicting tire vibration
and noise using Statistical-Energy-Analysis based on the energy
propagation. The mechanism of tire vibration and noise goes
through three processes of input, propagation and radiation.
Initial examination of Experimental SEA model construction
method and input power evaluation methods is to be
conducted for propagation and input based on SEA theory.
In this report, the authors first propose a method to estimate
the input power from the road surface. For the problem of the
subdivision in estimation input power, we established a method
to make the experimental SEA models a total of two subsystems,
one for each tread and one for each sidewall. Furthermore, we
showed that the input power becomes constant regardless of
the subdivision angle. For the subsystem energy, a tri-axial
accelerometer was installed inside the tread and outside of the
sidewall, the response acceleration when rolling on a tire tester
was measured, and it was calculated by FFT.
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Vibration modelling of an elastic body of arbitrary shape subjected to mixed excitation
Goran Pavic
[Abstract]
Using the known analytical Green’s function for the displacements and stresses of a force-excited infinite, elastic, homogeneous solid the response of a finite body of arbitrary shape can be reconstructed. The response is obtained by applying to the infinite solid a distributed secondary force excitation in the exterior and interior of a virtual closed surface which coincides with the shape of the targeted finite body. The role of the secondary excitation is to re-create across the virtual surface the boundary conditions of the finite body. Such an approach permits one to obtain the response of the finite body under rather general excitation conditions. The latter may consist of a superposition of kinematic movements applied over one part of the body surface and the distributed force excitation within the body. The physical side of approach is discussed first, the corresponding mathematical procedure is then outlined and examples is provided to demonstrate the validity of the approach.
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Vibro-acoustic Rendering Methods to Radiate a Uniform Sound Field from a Panel Speaker
Jung-Han Woo, Jeong-Guon Ih
[Abstract]
The sound radiated from the usual panel speaker employing
the single point excitation is distorted due to the multi-modal
response of the regular panel, or the effective range of
radiation is limited to very low frequencies. Such problem
can be alleviated if an array of actuators are used in the
periphery of the panel to control the vibration response for
the uniform radiation in a wide frequency range. To achieve
the uniform acoustic field, the two inverse vibro-acoustic
rendering techniques can be adopted: first, a vibration
pattern composed of a virtual speaker and baffle is being
created, and, second, the spatial distribution of the
vibration is being controlled to radiate the desired sound.
Each method is called indirect or direct method. Numerical
simulations are conducted on a thin, simply-supported
plate, for which the high frequency limit is 300 Hz. The
indirect method results the rendering error of -20 dB and the
direct method -50 dB. Two methods are afflicted with the
instability, particularly in using the direct one. Regularization
technique is used to overcome the ill-posedness, that yields
the change of condition number from O(1017) to O(104).
Input gains are also reduced by 35 dB for both methods.
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HiFi Panel Speaker by Controlling the Vibration Field Using Array Actuators
Ki-Ho Lee, Jeong-Guon Ih
[Abstract]
Most of panel speakers adopt a thin rectangular plate, which is
excited by one or several actuators. Because of the multi-modal
characteristic of the plate, the vibration field composed of
many modes deteriorates the radiated sound quality inherently,
in particular at the low to medium frequency range. This study
deals with the control of a point-excited panel speaker to
radiate HiFi and powerful sound, and consuming small input
energy using an actuator array enclosing the main actuator
zone. The desired control is to confine the major in-phase
oscillation at around the main actuator and to suppress the
vibration in the remainder area, resulting a virtual circular
speaker and a baffle. The gain of actuators is obtained by
solving the inverse rendering problem derived from the transfer
function between the actuator input and the response of field
point on the plate. Two types of array configurations are
investigated by simulation, which shows that the adjacent
double circular array can produce the sound radiation for a
wide frequency band. The experiment conducted using the
double circular array reveals that the vibration field can be
controlled well as desired and the present control method can
improve the frequency response drastically to be flat.
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Bending Waves focusing in Arbitrary Shaped Plate-like Structures: Application to Spatial Audio
Nassim Benbara, Marc Rébillat, Nazih Mechbal
[Abstract]
Audio applications are more and more demanding with respect
to the visual impact of loudspeakers while still requiring more
channels for high quality spatial sound rendering. The use of
arbitrary plate-like structures driven by electromagnetic
actuators or by piezoelectric elements appears as a promising
solution to tackle those issues. However, to meet spatial
rendering audio constraints, the generated bending waves must
be focused to a certain extent within the host plate.
Theoretically, this means being able to invert the spatiotemporal
wave propagation operator for the generated bending waves
to fit a given target shape. Several methods are here
investigated to perform this task depending on the available
knowledge of wave propagation in the plate (theoretical,
partial spatial and full spatial knowledge) and using several
projection bases (modal basis and eigenvalue basis). The
various methods are presented in a unified theoretical
framework and their performances are compared by means of
three key performance indexes (focus localization error, width
error and spatial contrast) and with respect to various
parameters (damping in the structure, experimental noise level).
The study will involve numerical simulations and experiments to
compare them, firstly on a simple plate and after on a more
realistic structure as stiffened plate.
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Organization:
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H.P. Lee, National University of Singapore |
J. Bös, SAM, TU Darmstadt |
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Renaissance of the Structural Intensity Analysis
Heow Pueh Lee, Kian Meng Lim
[Abstract]
Structural intensity represents the flow of energy within a structure, subjected to external excitations. The interest in structural intensity comes from its practical applications such as damage detection, structure-borne noise control and structure optimization. By visualizing the flow of power, it is possible to identify the transmission paths and sources/sinks of mechanical energy within structure. It is a powerful tool for monitoring the structure behavior under different loading conditions which would result in altering energy flow path by using dissipative elements or mechanical modifications. We have extended the applications to not only structural mechanics, but also impact mechanics, fracture mechanics, rotating disk, thermal induced vibration as well as biomechanics. In recent years, Acoustic Black Hole (ABH) is developed for the flexural waves in the beam plate structures. It attracts lots of attentions in focusing wave energy and reducing vibration-acoustic level. It also enhances the efficient of the passive damping systems, reducing the weight of the damping layers. In this paper, we will first present an overview of the various applications of structural intensity as well as our more recent work on the use of structural intensity for guiding the design of ABH.
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Investigation towards an Active Barrier for Structure Borne Sound using Structural Intensity
Alexander Kokott, Thomas Haase, Hans Peter Monner
[Abstract]
Vibrations of aircraft or vehicle engines, for instance, are often transmitted by structure borne sound, leading to a significant radiation of noise inside passenger cabins. Current active vibration control concepts use either velocity or acceleration as a control source. However, these only lead to a local reduction in vibration and not necessarily to the reduction of the vibration energy flow. This study presents the implementation of current methods for structural intensity measurement with a real-time control. The work investigates one and two-dimensional structures. A reduction of energy flow in a beam structure is shown experimentally, as well as numerically for plates. The measurements are strongly influenced by theoretical simplifications concerning the composition of the structure borne waves and the quality of the sensor arrays used, i.e. the sensor spacing and the positioning accuracy. Though, sufficient accordance between numerically and experimentally estimated structural intensity can be found using methods with smaller sensor arrays. A barrier effect is shown by numerical investigations and is measured on a beam. Therefore, the control of vibration energy flow is a more effective method for a global reduction if vibration downstream the control area.
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Numerical study on energy transmission for soft materials and metamaterials structures by structural intensity method
Rong Huang, Ziqian Li, Yiheng Xue, Zishun Liu
[Abstract]
In this work, we studied the vibration performance of metamaterials structures from the aspect of energy flow by using structure intensity (SI) method. Compared with the traditional vibration analysis method, the SI method considers the internal force and displacement in the structure, and can accurately reflect the effect of the excitation force, the damper locations and the energy flow on the structures. In this study, numerical simulation system was built to represent the metamaterial structures under different loading and damping cases. We get the corresponding results of structural intensity distribution and plot the SI streamline for each case. We find that the structure intensity distribution can be significantly affected by adjusting the material parameters and setting damper in specific locations. Based on the SI energy distribution of the metamaterial structures, we present a method to reduce the vibration of the structure by blocking energy on its main propagation path. In addition, the effect of structural damping and constrained layer damping on the vibration performance of metamaterial structures is discussed, and a series of results are obtained. We hope that our works could provide researchers new ideas of vibration control for similar new smart structures in potential industry applications.
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Low vibration design for shell structure based on structural intensity distribution
Takeshi Miyama, Hiroki Nakamura, Toru Kikuchi, Toru Yamazaki
[Abstract]
Reducing structure-borne sound is an important issue for mechanical industrial products. This paper focuses on countermeasures at the vibration-energy transmission path as a better way to control structure-borne sound while the best approach has been applied at the source. This propagation path can be determined by measurements or calculations of the structural intensity (SI), which is defined as the instantaneous rate of energy transfer per unit area in a structure. There are many reports concerning SI, but we can find no such report of implementing structural design and modification based on the SI distribution.
This paper presents a methodology for structural design implementation by using the SI distribution. The controllability of vibration by SI is firstly discussed by comparing the kinetic energy and SI in wave solutions. We then design a shell structure using the proposed methodology. As a result, the SI can identify the location to be modified, and changing the structure at that location facilitates low vibration. Also, we present the experimental application results by using our developed measurement sheet sensor for visualizing structural vibration energy flow. it is shown that our measurement sheet sensor is useful to find the countermeasure areas on the shell-structure with ease.
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Structural intensity estimation via displacement and shape measurements of thin shells
Felipe Pires, Stéphane Avril, Steve Vanlanduit, Joris Dirckx
[Abstract]
The structural intensity analysis on plates is a well-documented approach to identify
energy paths taking place on a structure. By assuming that the sample behaves in
accordance with the Kirchhoff-Love postulates, the structural intensity can be estimated
by calculating spatial derivatives of the sample’s out-of-plane displacement and by
having a priori knowledge of its material properties. However, when shell-like
structures are analyzed, the in-plane displacements become non-negligible terms and
are critical inputs for the energy flow assessment. The purpose of this work is to present
a method that deals with this issue by interpolating full-field displacements on a finite
element mesh that resembles the analyzed sample. After approximating these measured
displacements on the nodes of that mesh and by treating the element’s assembly as a
Kirchhoff-Love plate model, the data differentiation via shape functions on predefined
local coordinates is carried out. By having all differentiated data at hand, the structural
intensity on thin shells is accessible. Due to the results that were obtained from this
approach, the method has shown itself to be a reliable tool to assess the vibrational
energy in irregular shells.
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Measurement of the structural intensity of curved shell structures by means of 3D laser vibrometry
Nikolai Kleinfeller, Joachim Bös, Tobias Melz
[Abstract]
The structural intensity (STI) describes the energy flow of structure-borne sound within a vibrating structure. For complex structures the STI is usually obtained from a finite element analysis. At the present time, a measurement procedure for validation purposes is not sufficiently developed if structures with arbitrary shapes are considered. For this reason, a measurement procedure for determining the STI of arbitrarily shaped thin-walled structures is developed within the framework of this research project. This paper deals with the measurement of structural intensity of shell structures with different shapes by means of 3D laser vibrometry. Therefore, a measurement procedure is set up that consists of measurement data acquisition, measurement data processing, and a general STI calculation. In this context, the challenges of the measurement procedure should be emphasized, if the structures are not flat but curved. In addition to the flexural vibrations, the extensional vibrations must also be recorded simultaneously. In the case of curved structures, knowledge of the curvature properties is necessary in addition to the structural vibrations. The results are compared with the structural intensity obtained from a finite element analysis and the accuracy of the measurement procedure is analyzed.
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Development of energy propagation analysis methods for low- frequency phenomena at BMW
Peter Groba, Johannes Ebert, Torsten Stoewer, Joachim Bös, Tobias Melz
[Abstract]
In recent years, a rise in computing power and an enhanced
understanding of energy propagation processes have led to an
increased use of the numerical simulation of energetic
quantities in vibroacoustical problems both including and
excluding the coupled fluids. These energy-based quantities
have proven themselves very useful in delivering information on
the behavior of vibrating structures. Over the last years the BMW Group has invested in research on
energetic quantities to evaluate their potential to support the
acoustic engineer in the early stages of the vehicle
development process. This paper presents an overview of the
development regarding the structural intensity (STI) and its
usage as a tool for the analysis of low-frequency energy
phenomena. It focuses on car structures in steady state
vibration with and without fluid coupling. Starting with the work of Stoewer, who conducted numerical
simulations in the time-domain to perform design modifications,
the overview continues with analyses conducted by Ebert who
linked structure-borne with air-borne energy propagation. The
paper closes by introducing the work of Groba who analyzed
energy propagation using power-based quantities calculated
from the STI, allowing for a simplified quantification and
description of energy propagation phenomena.
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Experimental Structure Intensity Analysis of an Airbus A400M fuselage structure using high-resolution vibration measurements
René Winter, Simon Heyen, Jörn Biedermann
[Abstract]
An ongoing challenge in the analysis of aircraft cabin
vibrations in the mid-frequency range is the analysis of
transfer paths through the complex fuselage structure.
Knowing these paths would allow for simpler and more cost
efficient counter measures to harmonic noise sources like
engines or the auxiliary power unit.
The A400M-MSN5 fuselage structure located at the University
of the Federal Armed Forces in Hamburg (HSU) is a
production line A400M fuselage repurposed as a test facility
for experimental studies allowing for full scale tests of novel
experimental methods for acoustics and structure-born noise.
Presented here are the results of the deployment of fully
automated and autonomous Laser-Doppler-Vibrometer
scanner used to measure the structures fuselage vibration
above cabin floor at a high spatial resolution. A total of more
than 17000 individual measurement positions spaced
approximately 9 cm apart was measured, using both a
loudspeaker system and a shaker as an excitation source.
Utilizing the measured high resolution deflection shapes
energy transfer paths were calculated using a hybrid
Numerical-Experimental approach to structural intensity
analysis.
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Organization:
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M. Scholz, TUM, Chair of Vibroacoustics |
M.E. Altinsoy, Technische Universität Dresden, Lehrstuhl für Akustik und Haptik |
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Relationship between Hand-arm Vibrations and subjective evaluation by a Magnitude Estimation Method : one or both hands excitation
Luc Laroche, Etienne Parizet, Damien Piranda, Frédéric Bornet
[Abstract]
Steering wheel vibration is one of the factors determining the
comfort in a driving car experience. This current study aims at
characterizing discomfort induced by steering-wheel vibration in
terms of frequency, level and direction. Measurements were
realized with seven cars of different sizes and motorizations. That
study showed that the maximum level of vibration was related
to the first steering column resonance, occurring between 30
and 50 Hz; vibration levels was determined between 0.3 and 5
m.s-2. Then a magnitude estimation experience was conducted
to relate the perceived intensity to the steering wheel physical
vibration. Two directions were investigated : along the arm and
in the normal direction of the palm. 34 subjects participated to
the experiment. The relation fits Steven’s law; the exponent
varied from 1.1 and 1.6, depending on frequency and direction
of the excitation. Part of the experience was also to quantify the
difference on perception of intensity introduce by one or both
hands on the steering wheel.
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A parameter study on measuring hand-arm vibrations of an impulsive vibrating tool applying the international standard
Magdalena Scholz, Leopold Winter, Steffen Marburg
[Abstract]
In daily life, many activities expose us to hand-arm vibrations
(HAV). The characteristics, such as frequency components or
level, of these vibrations vary substantially. At high levels
some can cause changes to the hand-arm system, which
are grouped under the term hand-arm vibration syndrome
(HAVS). Hence, there are multiple standards on the subject of
measuring, evaluating and simulating HAV. In these, many
cases are covered with specific instructions, but not all. This
leaves leeway in measurements.
Additionally, previous studies have shown largely varying
results, some even on the same device. As the assessment of
the risks for the hand-arm system depends both on accurate
measurements and knowledge of the influences here-on. The
leeway left in the specifications in the standards may be at
least part of the origin of the variance in the measured
results. In a measurement and comparison of previous
studies, influencing factors are to be determined. The goal is
an appraisal of the extent of their effect.
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Perceived discomfort for tri-axial helicopters vibrations
Laurianne Delcor, Etienne Parizet, Julien Caillet, Julie Ganivet-Ouzeneau
[Abstract]
Vibrations are an important contributor to perceived discomfort
in helicopters cabins. A test campaign was conducted in order
to understand how discomfort varies for helicopter tri-axial
excitations. Six experiments were conducted, which involved 30
participants. Subjects were seated on a six degree of freedom
vibration table. They had to evaluate discomfort of several
stimuli. The first three experiments investigated the influence of
vibrations due to the blade passage frequency of the main
rotor, to the tail boom modal behaviour and to a combination
of these two sources. The impact of a phase shifted vibrations
and of amplitude-modulated excitations were studied in a
fourth and fifth experiments. Finally, the last experiment focused
on real signals measured on several helicopters of Airbus fleet.
Results showed that 1) vertical vibrations represent the major
contributor to discomfort in helicopters; 2) for amplitude-
modulated excitations, discomfort only depends on the
amplitude of the modulation, not on the modulation frequency;
3) out of phase stimuli are perceived in the same way than
phased stimuli; 4) helicopter frequency content above 30 Hz
does not increase discomfort; 5) ISO2631-1 comfort index is a
good predictor of discomfort evaluations.
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Comfort Evaluation on the drivers using Transfer Path Analysis
Nihlatul Falasifah, Dhany Arifianto
[Abstract]
The vibration transmissibility to the seated passenger car has
important influences on the human body comfort, health,
quality, and safety. To establish the dynamic equation for
precondition of modal analysis, Newton-Euler method is usually
used as the common study. However, this method is difficult to
determine generalized coordinates belong to which natural
frequency for a linear system. On the other hand, by using
transfer path analysis (TPA) as a new technique, it is possible
to
determine generalized coordinates of the vibration for the car
machine affected the vibration of the human comfort seat. The
purpose of this research is to know the level of vibration on the
human body comfort by using TPA according to the ISO 2631-1
standard and also to characterize the vibration effect on
different sources (variation of the car machine) for seat
comfort.
This study involved human as the subject. The results of this
study
show that low-frequency vibration of the seat greatly affected
the comfort of the ride. Through TPA, the car machine with high
torque and power was found to have a high influence on the
vibration of the human body comfort.
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Organization:
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E. Sarradj, TU Berlin, Technische Akustik |
J.-G. Ih, Acoustics Laboratory, Dept. of Mechanical Engineering, KAIST |
D. Miguez, University of Salford - Farrat Isolevel |
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Design of lightweight skeletal structures for noise mitigation
Heow Pueh Lee, Sanjay Kumar, Thong Hoi Yong, Umeyr Kureemum, Xu Song
[Abstract]
Lightweight materials have often been used in the manufacture of transportation, giving rise to faster and more energy efficient vehicles. Apart from making the materials lighter and stronger, one consideration would be the improvement of acoustic rating of the material. Being able to build-in some extent of sound proofing within load bearing structures may eventually minimise the need for additional sound proofing padding. This may result in thinner walls while maintaining the effectiveness of the walls to bear load and block sound and further enable space-saving designs. In this paper, we would present the design of various light weight skeletal structure with auxetic behaviour for potential noise mitigation. The effect of introducing perforation holes in the structure on the acoustic behaviour will also be investigated.
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Application of Dynamic Substructuring and in situ Blocked Force Method for Structure Borne Noise Prediction in Industrial Machinery
Diego Miguez, Oliver Farrell, Mark A. Bannister, Ryan Arbabi, Andy Moorhouse, Andrew Elliott
[Abstract]
Prediction of structure borne noise from industrial machinery allows
not only to adopt mitigation measures to avoid excessive vibration at
the design stage but also for problem solving after the final
implementation. These measures also avoid potential damage to
machinery tooling and the amount of defective products, whilst
reducing the disturbance to operators within the working area and
people in nearby facilities. Different strategies for achieving accurate
predictions range from numerical and analytical methods, such as
FEA and SEA, to methods which use measured data. Dynamic
substructuring, an example of the latter, has been used for this
research to characterise vibration contributions of three key
assembly elements: source, isolator and receiver. The in-situ blocked
force method was used to extract intrinsic properties of an
operational vibration source as well as the mobility of the receiver
and the dynamic stiffness of the isolators. Dynamic substructuring was
then used together with the blocked forces data in order to predict
structure-borne noise at the source interface with the isolators and at
remote locations in the receiver. Predictions of source behaviour as
part of an assembly were then compared with actual measurements
made on the installation to serve as a validation of the method.
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Vibration transmission between two reinforced concrete beams with surface-to-surface contact conditions
Marios Filippoupolitis, Carl Hopkins
[Abstract]
To detect human survivors trapped in buildings after earthquakes by using structure-borne sound it is necessary to have knowledge of vibration transmission in collapsed and fragmented reinforced-concrete buildings. In this paper, Statistical Energy Analysis (SEA) is used to model vibration transmission between two reinforced concrete beams with two different contact conditions, surface-to-surface and edge-to-surface. Experimentally validated Finite Element Models (FEM) of two elastically connected beams were used to carry out a Monte Carlo simulation with 30 beam junctions in random formations for each support condition. Coupling loss factors were determined from the FEM data using Experimental SEA and these were compared against theoretical models based on a lump spring connector. Results are shown for bending modes, torsional modes and the combination of all modes.
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Organization:
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U. Steinmann, Otto-von-Guericke-Universität Magdeburg |
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Study of the sound field in thin polymer films induced by High-Intensity Focused Ultrasound
Coralie Koo Sin Lin, Marko Liebler, Klaus-V. Jenderka
[Abstract]
This contribution presents the study of acoustic effects occurring in polymer films irradiated by high-intensity focused ultrasound (HIFU), when the films are sandwiched between a solid waveguide transducer and a solid counterpart.
Whereas the heating of polymer plates induced by HIFU was already reported in the literature, the acoustic effects in thin polymer films remain not fully understood.
In order to investigate the sound field occurring in polymer films, measurements of the acoustic pressure in water between transducer and counterpart spaced from 1 mm are carried out using a membrane hydrophone. The measurements are compared to an acoustic finite element simulation modeling the sound field in water between the transducer and the counterpart.
In order to evaluate the analogy between the acoustic pressure measured in water and the effective acoustic pressure occurring in polymer films, a second acoustic finite element simulation is conducted to model the irradiation of polymer films with HIFU in dry environment and compared to the simulation and measurements of the acoustic pressure in water.
A direct correlation between measurements and simulation is found. The investigation reveals that complex interferences occur in polymer films. These findings are relevant for new joining processes of polymer based on HIFU.
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Acoustic design principles for energy efficient excitation of a high intensity cavitation zone
Örjan Johansson, Taraka Rama Krishna Pamidi, Vijay Shankar, Torbjörn Löfqvist
[Abstract]
Energy-efficient process intensification is a key aspect for a sustainable industrial production. To improve energy conversion efficiency high intensity cavitation is a promising method, especially in cases where the material to be treated is valuable and on the micro meter scale. Transient collapsing cavitation bubbles gives powerful effects on objects immersed in fluids, like cellulose fibers, mineral particles, enzymes, etc. The cavitation process needs optimization and control, since optimal conditions is multivariate challenge. This study focuses on different design principles to achieve high intensity cavitation in a specific volume in a continuous flow. This study explores some potential design principles to obtain energy efficient process intensification. The objective is to tune several different resonance phenomena to create a powerful excitation of a flowing suspension (two-phase flow and cavitation bubbles). The reactor is excited by sonotrodes, connected to two coupled resonant tube structures, at the critical frequency. Finally cavitation bubbles are initiated by a flow through a venturi nozzle. The acoustically optimised reactor geometry is modelled in Comsol Multiphysics®, and excited by dedicated ultrasound signals at three different frequencies. The effect of the high intensity cavitation is experimentally evaluated by calorimetric method, foil tests and degree of fibrillation on cellulose fibers.
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Small-Sized Acoustic Resonators for Ultrasound in Air
Tobias Merkel, Jürgen Harpain, Norbert Gorenflo, Jonas Stein
[Abstract]
A rigid-walled cavity which is connected with a neck to the surrounding air provides an acoustic system with significant resonance frequency, well known as Helmholtz resonator. By stimulating the system with an intense acoustic wave an air flow can be observed near the neck mouth, which gets directivity and differs from the expected periodically mass flow caused by oscillation.
For technical applications the resonance frequency must be shifted to ultrasonic range to maintain the well-being of humans and to avoid noise pollution. Based on studies with resonators at low kHz-range we perform a successive miniaturization to obtain well-working Helmholtz resonators in an inaudible area.
The measurement of acoustic and air-fluidic characteristics of those tiny devices is a high technical challenge, because of the small dimensions and wave length of air-borne ultrasound. Therefore we developed a special small-sized Kundt’s tube to estimate the resonance frequency and Q-factor of our samples. Another problem is to detect and characterize the estimated air flow near the resonator hole; especially the commonly used hot wire anemometers are too large for this measurement. Our investigations are assisted with numerical simulations based on linear wave acoustics as well as on fluid dynamical computations using the Navier-Stokes equation.
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Multifrequency ultrasonic transducers with spatially distributed stop band material
Johannes Henneberg, Simone Preuss, André Gerlach, Steffen Marburg
[Abstract]
Ultrasonic transducers are widely used for surround sensing applications in the automotive industry. A major challenge for today’s mobility solutions consists in the development of autonomous driving functions. Highly automated vehicle concepts require a comprehensive surround sensing performance, which leads to an increasing number of sensors. The applied ultrasonic sensors are desired to operate in parallel or in shorter intervals. Hence, transducers with two or more well discriminated operating frequencies are of great interest. Known multifrequency ultrasonic transducers use multiple electromechanical coupling elements resulting in more complex sensor electronics.
To overcome this issue, the authors investigate a method for controlling the operational deflection shapes. In finite element simulation, different resonator configurations, represented by a generic plate like structure, are studied. Spatially distributed stop band material is used to achieve two well-separated operating frequencies with similar and appropriate sound radiation. Furthermore, it is investigated whether the shape modification is caused by the resonant behavior or the added resonator mass. Based on the results, locally distributed resonators are identified as a suitable solution to modify the operational deflection shapes and the according sound radiation of the generic model. Thus, a multifrequency ultrasonic sensor with a single coupling element can be realized.
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Ultrasonic sensor based on phononic crystals
Paul Wasmer, Jannis Bulling, Jens Prager
[Abstract]
An essential task in many industries, e.g. food, petrol or chemical
industry, is the precise and accurate characterization of liquids.
Therefore, the development of innovative in-line sensors is of
great interest. New concepts based on periodic structures, so-
called phononic crystals (PnCs), are an interesting idea for the
design of innovative sensors.
A PnC-based sensor can be designed by introducing a
resonance inside a bandgap, a frequency region where no
wave propagation is allowed. High-Q measurement systems
using PnCs are already reported in the literature. However,
existing designs cannot be implemented into a piping system
directly, but need special fittings, openings or by-passes to be in
contact with the liquid.
To circumvent this issue, we develop a new sensor based on
PnCs, which can be directly implemented as part of the piping
system. For this purpose, we use a PnC consisting of hollow
cylinders with a periodic change of the outer diameter.
A bandgap could be found for the described geometry without
fluid in simulation and measurement. However, simulations show,
that a bandgap for fluid-filled cylinders can only be obtained
for quasi-longitudinal modes. Hence, we propose a mode
selective excitation for the sensor.
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Design of Resonant Vacuum Pressure Sensor with CMUT for High Sensitivity and Linearity
Xiaoli Zhang, Lu Yu, Haixia Yu, Dachao Li
[Abstract]
Vacuum pressure monitoring is mainly used for atmospheric pressure measurement and pressure measurement inside organisms, and is widely used in aviation, meteorology and medical diagnosis. This paper innovatively reports a vacuum pressure monitoring method using capacitive micromachined ultrasonic transducer (CMUT) array which is a small MEMS device manufactured by vacuum wafer bonding process. Both of the manufacturing process and detection circuits are simplified compared with those of the traditional MEMS devices used for vacuum pressure monitoring. By detecting the change of resonant frequency rather than capacitance of the CMUT array with varying air pressure, the anti-interference ability of vacuum pressure(less than 1atm) monitoring is improved. Simulation and experimental results show that the resonant frequency of the CMUT with 30V DC bias varies linearly with the applied pressure, and has high detection sensitivity of 438Hz/kPa and R-Square of 0.99335.
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Determination of frequency dependent Ultrasound Absorption by means of Radiation Force based Power Measurements
Tina Fuhrmann, Konrad Mehle, David Waltschew, Klaus-V. Jenderka
[Abstract]
The frequency dependent acoustic absorption of propagating ultrasound wave is one of the key quantities in acoustic material characterization. The standard approaches to determine the amplitude absorption coefficient are to analyze either the amplitude decay of subsequent rear surface reflections of a sample with known thickness (materials with low acoustic absorption) or the insertion loss of samples with different thicknesses (materials with higher acoustic absorption).
The new approach uses the total temporal averaged ultrasonic power determined by means of a Radiation Force Balance, by comparing the power values of excited bursts (number of cycles >100) with and without sample. The required information about the thickness of the sample as well as the samples speed of sound, respective the amplitude reflection and transmission coefficients, were determined in advance within the same measurement setup. The effects of half-wave layers on the reflection and transmission coefficients are incorporated in the analyzing routines. Additionally, the frequencies of the resulting maxima of transmitted power are used for refinement of the samples thickness and SOS values.
Measurement of ultrasound absorption by means of Radiation Force based power measurements takes advantage of the low uncertainty of the method, but requires an extended time frame for the measurements.
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Effective modeling of elastic waves for haptic surface interaction
Sebastian Wöckel, Michael Simon, Ulrike Steinmann
[Abstract]
Guided acoustic waves can be found in multiple
applications as SAW-devices, NDE, hearing aids, fluid
monitoring in tubes or resonance effects in structures. Among
those examples, the contribution addresses the sound field
design with linear elastic waves on thin display or
touchscreen surfaces. Based on the principle of reversibility
and the time reversal technique a simulation model was built
up to calibrate the refocussing of vibrations on certain area.
The main objective (of this work) is the haptic interaction of a
finger with virtual elements modelled by a local vibration.
The simulation bases on a finite element model of structural
dynamics. In detail, the influence of elasticity, plate thickness
in relation to the wavelength and the transducer positions on
the wave mode selectivity are discussed. Further the
numerical calibration is used to define the signals with
respect to the contrast and resolution of tactile sensation in a
focal point and the potential to selectively focus on multiple
points at the same time.
Finally two demonstrators with different frequency range are
used for evaluation of the consistency of simulated and
experimental excitation signals and the respective impacts
on the focal quality.
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Vibro-tactile displays for stimulating surface impressions
Andreas Sebastian Schmelt, Eike Christian Fischer, Viktor Hofmann, Jens Twiefel, Marc Christopher Wurz
[Abstract]
Nowadays, information transmission devices, which activate the human tactile sense, are limited on Braille generation to support the blind and on the enhancement of the gaming experience with a vibrating gamepad. The generation of real surface structures in a device, a so-called tactile display, has not been solved satisfactorily so far. A new approach is to generate a stress-strain-distribution in the finger similar to that, which occurs while moving a finger over a surface, so that the tactile perception is alike. Vibrations generate the stress-strain-distribution. To realize this in a tactile display, interdisciplinary knowledge is necessary. The actuators of the display have to address the different stimulation processes of the individual mechanoreceptors in the skin. In addition, every person has individual tactile impressions according to the fingers physiology. In order to satisfy this requirements, small, powerful and high dynamic actuators are needed. Furthermore, the current load characteristic of the actuator is always needed for control. In this work, our novel tactile displays to stimulate the mechanoreceptors of the human finger by suitable frequencies and amplitudes are discussed in detail. The self-sensing concept is presented to obtain online the individual actuators displacement and force.
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Acoustic Holograms for Particle Assembly and Fabrication
Kai Melde, Zhichao Ma, Korbinian Pöppel, Peer Fischer
[Abstract]
One established field of research for ultrasound is the contact-
free manipulation of particles and cells. Its non-toxic nature,
paired with the low absorption in aqueous environments, offers
potential avenues including in the biomedical field.
Conventional ultrasonic manipulation techniques have thus far
either relied on tweezing approaches or resonator structures. In
tweezers a single particle is trapped in a localized point, and the
trap is actively repositioned to move a particle, either by
moving the transducer or by reconfiguring a phased array
transducer. Assembly methods, on the other hand, rely on
resonant structures are inherently symmetric and thus limit the
attainable structural complexity. An alternative approach is
provided by acoustic holograms,
which allow one to control sound fields with high fidelity and
can be used to create both extended potentials and tweezer-
like traps. Holograms can be implemented on phased array
transducer or by using 3D-printed elements. The latter allows
several orders of magnitude more degrees of freedom in the
design at virtually no added cost, which enables more
advanced particle manipulation techniques. In this talk, I will
give an overview over acoustic holography and its use for
microparticle assembly, fabrication and I will highlight new
developments in our lab.
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Precise airborne sound field characterization using a miniaturized laser interferometer of 1 MHz bandwidth
Ryan Sommerhuber
[Abstract]
In order to determine the spatial pressure distribution of an
airborne sound field, idealized mathematical simulations are
often relied upon. However, these do not always account for
real life situations, for instance for emitter degradation over
time, or emitter surface damages. Thus actual measurements
of airborne acoustic fields can be of significant importance
for emitter manufacturers and users.
Special purpose applications such as air-coupled ultrasound
material inspection (NDT), gas flow rate measurement or
human gesture tracking require optimization of sound
pressure level, frequency matching or focal alignment.
With an effective sensing area of 0.6 mm2 and a frequency
bandwidth of 1 MHz, the presented optical microphone is
ideally suited for high-resolution scanning of ultrasound fields.
The sensing principle is based on measuring the pressure-
induced refractive index change using solely a fibre-coupled,
rigid Fabry-Pérot cavity, and does not feature any post-pulse
ringing of mechanically moveable parts. Contrary to laser
vibrometers, the detection laser remains confined to the
microphone capsule.
Measurement results will be presented, revealing nearfield
interference, side lobes and focal spots of resonant
piezoelectric transducers as well as broadband emitters.
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Organization:
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S. Dos Santos, INSA Centre Val de Loire |
A.S. Schmelt, Institut für Dynamik und Schwingungen |
L. Adler, ohio state university |
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Defect detection using the identification of resonance frequency by spatial spectral entropy for noncontact acoustic inspection method
Kazuko Sugimoto, Tsuneyoshi Sugimoto, Noriyuki Utagawa, Chitose Kuroda
[Abstract]
Our noncontact acoustic inspection enables detection of internal defects of composite material from remote location. The surface to be measured is excited with aerial sound waves. And a two-dimensional vibration velocity distribution is measured using a laser Doppler vibrometer (LDV). Then, signal analysis is performed. In addition to the conventional method, by introducing spatial spectral entropy (SSE), resonance frequency of internal defects can be detected. And at the same time, resonance frequency of a laser Doppler vibrometer can also be detected. By identifying them and choosing the resonance frequencies and visualizing, it is possible to visualize internal defects clearly. Actually, by experimental results using concrete specimens with a crack defect, it became clear that by SSE analysis it is possible to clearly distinguish the resonance of defective part and the resonance of LDV.
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Identifying objects in a 2D-space utilizing a novel combination of a re-radiation based method and of a difference-image-method
Andreas Sebastian Schmelt, Torben Marhenke, Jens Twiefel
[Abstract]
For identifying objects in a 2D space the feasibility of a novel attempt is investigated in this study. The acoustic fields are simulated with the 2D finite difference k-wave toolbox in MATLAB. The generated acoustic field is recorded at defined positions that can be understood as receiver positions. From these data, the acoustic field is recalculated using the 2D-reradiation-method. An object-free acoustic field serves as reference. The acoustic field (at the receiver) considering the object is determined. The full acoustic field is recalculated from the secondly recorded data. Using a difference-image-method, the calculated acoustic fields are processed and the position, size and geometric shape is identified from the remaining diffraction pattern. Different objects and object positions are studied. Furthermore, the influence of different excitation signals of the transducer for the identification are investigated.
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Guided Lamb and edge wave excitation by piezoelectric transducers in elastic plates
Mikhail Golub, Artem Eremin, Maria Wilde, Alisa Shpak, Inka Mueller
[Abstract]
Elastic guided waves (GWs) are widely employed in ultrasonic
NDT/NDE for damage characterization due to their capabilities
for long-range propagation and sensitivity to defects. For
excitation and sensing of GWs, thin piezoelectric transducers
adhesively attached to the structure are among commonly
utilized tools. The complexity of these mechanical processes
conditioned by, e.g., the multimodality of GW motion, dynamics
of piezotransducers and the quality of their contact with the
host structure should be properly addressed for better
performance of GW-based NDT/NDE systems.
In this contribution, the results of theoretical and experimental
studies of GW excitation by piezotransducers attached at an
edge or a face of an elastic plate are reported. The semi-
analytical methods relying on integral transform and modal
expansion techniques are applied for numerical analysis. Wave
energy distribution amongst excited GWs (Lamb and edge
waves) is carefully analysed and discussed to improve the
quality of damage identification with GW-based techniques.
The influence of partial debonding of a piezotransducer on GW
excitation is carefully analysed. The potential of edge waves for
practical NDT/NDE applications is discussed. Namely, it is
demonstrated that edge waves excited in plates and scattered
by surface-breaking cracks can be employed for crack
detection at the edges.
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Outer Wall Inspection by Noncontact Acoustic Inspection Method using Sound Source Mounted Type UAV
Tsuneyoshi Sugimoto, Kazuko Sugimoto, Ituski Uechi, Noriyuki Utagawa, Chitose Kuroda
[Abstract]
By using the noncontact acoustic inspection method that utilizes flexural resonance generated by acoustic irradiation induced vibration, crack defect existing near the surface of the measurement target can be non-destructively inspected from a long distance. It has been proved that the same test as the hammering test can be done with tunnels and bridges with a height of over 30 m and shotcrete with a rough surface.
This method has problems such as environmental noise and angle dependency. However, when a sound source itself is mounted on an unmanned aerial vehicle (UAV) which is becoming popular in recent years, it becomes possible to make the sound source position directly face the measurement target surface, and these problems are expected to be resolved.
Therefore, we investigated outer wall inspection by noncontact acoustic inspection method using sound source mounted type UAV. The inspection results using the outer wall specimen were good, and it became clear that peel defects of the tile can be detected even with a small sound source that can be mounted on UAV.
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Ultrasonic Spectroscopy to Characterize Flaws, Porosity and Adhesive Bonds
Laszlo Adler
[Abstract]
Ultrasonic Spectroscopy is the study of ultrasonic waves resolved into their Fourier frequency components. Ultrasonic spectroscopy is used in many applications of Nondestructive Evaluation. In this presentation three methods developed by the author to apply ultrasonic spectroscopy in NDE will be discussed.
1. Flaw characterization refers to the determination of size, orientation, and composition of discontinuities in solid materials. Multifrequency ultrasonic wave interaction with artificial and real defects measurements and subsequent theoretical analysis using diffraction of elastic waves provides these material discontinuities’ characteristics.
2. Gas porosity assessment. Measuring pore size and volume fraction from the spectrum of attenuation coefficient will be discussed. Theoretical analysis is based on the attenuation due to independent scatterer of spherical voids. The model prediction of the scattering of ultrasonic waves from porosities is in good agreement with measured values in aluminum casts.
3. Evaluation of Adhesive Layers. A system was developed to determine both adhesive and cohesive properties of thin bonded layers using multi transducer spectroscopy. Scattered longitudinal and transverse waves spectra is used in this method. The method was further enhanced by combining with a low frequency dynamic load acting as a nonlinear enhancement. Ultrasonic measurements are in good agreement with actual layer properties.
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The Effect of Water Saturation on the P-wave velocity of Sedimentary Rocks
Sair Kahraman
[Abstract]
Since ultrasonic methods are non-destructive and easy to
apply, they have been used widely in various fields such as
mining and civil engineering applications. The water content of
rock strongly influences seismic waves. Rock engineers
sometimes need to estimate the wet-rock P-wave velocity from
the dry-rock P-wave velocity. In this study, in order to investigate
the predictability of wet-rock P-wave velocity from the dry-rock
P-wave velocity, P-wave velocity measurements were
performed on 43 core specimens pertaining to 7 different
sedimentary rocks such as limestone, travertine, and sandstone.
The evaluation of the results showed that P-wave velocity
increment due to saturation ranged from 44.7 % to 99 %, with an
average of 66.7 %. The highest increment was observed in the
syenite specimens. The results were also evaluated using
regression analysis. The wet-rock P-wave velocity values were
correlated to the dry-rock P-wave velocity values. A very strong
linear correlation was found between the two parameters for all
tested samples. The regression analysis was also repeated for
each rock group, and found also strong correlations.
Concluding remark is that the wet-rock P-wave velocity values
of sedimentary rocks such as limestone, travertine, and
sandstone can be predicted from the developed empirical
equations.
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Baseline-Free Repetitive Pump-Probe Experiment for Structural Health Monitoring
Marina Terzi, Lynda Chehami, Emmanuel Moulin, Vladislav Aleshin, Nikolay Smagin
[Abstract]
In structural health monitoring, it is often necessary to make
reference measurements on a baseline healthy sample.
However, baseline parameters can be strongly affected by
environmental and other conditions and significantly change
over time. Here we propose a baseline-free damage
localization method consisting of a repetitive pump-probe
experiment in a metal plate and a backpropagation algorithm.
In this experiment, a PZT patch generates a high-frequency (HF)
probe wave at 20 kHz, while a continuous low-frequency (LF)
pump of 1 Hz is produced by a shaker. Signals propagated in
the plate are recorded using a set of PZT sensors glued at known
positions. We use a steel ball pressed against the plate to mimic
a nonlinear defect. The aim here is to produce solid-solid
contact that will be modulated by the LF pump, as would be
the case, for instance, in fatigue cracks. To enhance this effect,
we subtract signals recorded at different times (corresponding
to different loading states) and apply the backpropagation
algorithm that locates the origin of the modulation. The
procedure is repeated several times to enhance the localization
quality. In future we plan to use the method with a real defect,
thus exploiting more realistic contact acoustic nonlinearity.
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Organization:
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T. Leighton, ISVR, University of Southampton |
M. Ueda, Kanagawa Institute of Technology |
M. Kupnik, TU Darmstadt |
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Ultrasound in Air: New applications need improved measurement methods and procedures, and appreciation of any adverse effects on humans
Timothy Leighton
[Abstract]
In the growing topic of ultrasound in air, innovations in
powerful
sources and digital signal processing have produced a
wealth
of proposed devices and applications. However if we are to
realize the benefits of these for companies and consumers,
the
introduction of these new technologies must come in an
environment where we can properly conduct standardized
field
and source measurements with appropriate processes
calibrations, and set exposure limits based on knowledge of
the
potential for adverse effects on humans: these are difficult
areas
in which the research base is scarce. This paper covers new
devices and applications, the difficulties with measurements
and calibrations, and discoveries regarding the potential for
adverse effects (including discussion of whether claims of
ultrasonic attacks on embassy staff are credible).
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Airborne ultrasound noise at workplaces
Christoph Kling, Robert Schöneweiß, Andrea Wolff, Christian Ullisch-Nelken, Christian Koch
[Abstract]
Airborne ultrasound noise in the workplace is an increasing problem concerning occupational safety and health. Industry applications found most frequently are cleaning, cutting and welding. National regulations exist, non-consistently providing limit or guidance values and a few giving weak advise on the measurement procedure itself. But in general, ultrasound noise measurement struggles with several basic problems, e.g. lack of instrumentation, traceability, tailored procedure and even knowledge on the impact on humans. Among other issues, the EARS II project, funded by the European metrology programm for innovation and research (EMPIR) of EURAMET, addresses the development of an appropriate measurement procedure for the assessment of noise at the work site. In the daily business of occupational health and safety engineers, there is no time for intensive studies. Thus, a reference workplace has been set up in the laboratory to examine soundfields in detail. These studies give a detailed view into the specific characteristics of ultrasound fields and allow to derive an adequate and practicable measurement procedure for ultrasound noise assessment.
The talk will present the development of a revised measurement procedure for the assessment of ultrasound noise at workplaces and the findings on the influence of the worker’s torso on the result.
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Quantitative Characterization of High-Intensity Focused Airborne Ultrasonic Fields
Marko Liebler, Christoph Kling, Benjamin Best, André Gerlach, Christian Koch
[Abstract]
High-intensity focused airborne ultrasonic fields are increasingly applied in various technical fields, for example, to generate haptic feedback during gesture interaction. The sound fields used pose special challenges for a quantitative sound field characterization. The high sound pressure levels in combination with the higher harmonics generated by nonlinear effects require both, a wide dynamic range and a wide bandwidth of the measuring chain. Furthermore, the small wavelengths and the focusing result in spatially strongly varying sound fields. In the present case, a 40 kHz carrier signal was focused on a focal point using a transducer array with 256 elements. Different microphone types and analysis methods were investigated with respect to their suitability in measuring high-power airborne ultrasonic fields. Using the portal scanner of the PTB, a spatial characterization of the ultrasonic field was performed. The results for different microphones and measurement setups were compared and will be discussed. Due to the nonlinear propagation of the sound wave, peak sound pressure levels of more than 160 dB were measured in the focal area. To evaluate the exposure of potential users the measured sound pressure levels in the focal area were compared with currently existing limit values.
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Completing the Traceability Chain for Airborne Ultrasound
Salvador Barrera-Figueroa
[Abstract]
Measurements of sound should be supported by measurement references traceable to the International System of units. Traceability for measurements at frequencies in the audio range are well covered by existing standards, this means that traceability should be extended to cover measurements in the ultrasound region. Recent advances in free-field calibration of microphones have made it possible to determine the absolute sensitivity of working standard microphones up to 150 kHz. Microphones calibrated using this technique can be used as references to calibrating other types of microphones by direct comparison whenever possible. Most microphones used in airborne ultrasound measurements are free-field microphones, hence determining their free-field sensitivity is enough. However, most microphones are calibrated using a single (low) frequency measurement using a sound calibrator, their electrostatic actuator response and a model-based free-field correction. Traceability for the latter two is not always sufficiently documented. Absolute pressure calibration using the reciprocity technique at high frequencies can be challenging because it brings the technique to its limits. Direct comparison is limited to the frequency range in which references calibrated using primary methods are available. This paper presents an account of the current methods for calibration of microphones at ultrasound frequencies, their potential, and prospects.
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Understanding Extended High-Frequency Hearing Thresholds
Chris Plack, Hannah Guest, Samuele Carcagno
[Abstract]
The standard clinical test of hearing is pure-tone audiometry,
which measures the lowest detectable level of pure tones,
typically between 0.25 and 8 kHz. However, there is
increasing interest in extended high-frequency (EHF)
audiometry, above 8 kHz. Basal (high-frequency) cochlear
regions are especially sensitive to the effects of ageing, with
a rapid increase in EHF thresholds with age. However, even
across young adults, the variability in EHF sensitivity is
remarkable: EHF thresholds can vary by over 50 dB between
individuals with normal hearing in the standard clinical range.
The cause of this variability is not fully understood. While there
are calibration difficulties due to resonances in the ear canal,
this cannot explain the extent of the variability. Some studies
suggest that the EHF region is especially sensitive to noise
damage, and that EHF loss might be an ”early warning” of
the potential for losses at lower frequencies, but the
evidence is inconsistent. There is also evidence that EHF loss is
associated with speech perception difficulties, although this
may not be causal: EHF loss may instead be a marker for sub-
clinical damage at lower frequencies. Understanding the
causes and consequences of EHF hearing loss remains an
important challenge.
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Exposure levels for parametric arrays in light of guideline ambiguities
Craig N Dolder, Mark Fletcher, Sian Lloyd Jones, Ben Lineton, Timothy Leighton
[Abstract]
Recent revelations about ambiguities in guidelines for very-high
frequency sound and ultrasound require many everyday
devices to be revisited in terms of guideline compliance. One
device that has a high ultrasound component but is designed
for audio broadcast is a parametric array. Acoustic radiation for
commercial parametric arrays was measured at a distance of
3.5 m in an anechoic chamber. The output of the device is
compared against three international regulations and
guidelines for ultrasound exposure. The authors found that the
device was not compliant under the conditions tested in the
laboratory.
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Electrostatically Driven Airborne Ultrasound Transducer with Perforated Backplate for Nonlinear Acoustic Applications
Takaaki Kamigaki, Yuki Ninomiya, Hiroyuki Shinoda
[Abstract]
We propose a thin transducer driven by an electrostatic
force (ESF) for a high-intensity airborne ultrasound. A
prototype transducer with 334µm-thickness emitted 392Pa
(RMS) on the surface when driving at 51kHz. This value is
large enough to apply for nonlinear acoustic applications.
A generation of high-intensity ultrasound needs a several
µm displacement of a diaphragm. Conventional ESF-
driven parallel-plate transducers have not achieved it
because a thin air layer between electrodes prevents the
diaphragm from vibrating.
We solved the problem by providing perforations at a
backplate electrode. The air between the diaphragm and
the perforated backplate can travel freely through the
perforations when the diaphragm vibrates. In addition, we
drove the transducer by controlling the amount of
electrical charge accumulated in the diaphragm. This
driving method can prevent a pull-in phenomenon. In the
experiment, the diaphragm vibrated with an average
amplitude of 3µm (RMS) at 51kHz.
Our transducer can be fabricated by laminating flexible
materials and can be formed in a sheet-shaped phased
array device. Thus, our transducer can be an alternative
to the piezoceramic transducers applied for the nonlinear
applications.
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Auditory evaluation of very-high-frequency sounds radiated from the Japanese trains part 2
Mari Ueda, Shunsuke Hanazaki, Hironobo Takahashi, Masaaki Hiroe
[Abstract]
We have reported the results of measurement of high-
frequency sounds radiated from the Japanese railways, and
the auditory evaluation test of the railway noise with such
high-frequency components.
From the time histories of the short-time frequency spectrum
of wayside noise measured at inside of the train, we
acknowledged that the high-frequency sounds were
radiated when a train traveled round a curved track, and
the sound had dominant frequency components above 16
kHz and differed significantly in the frequency characteristics
from common railway noises.
At the same situations, the auditory evaluation tests were
performed for ten participants aged around 20. They
answered the following questions on 26 adjectives by 5-
point scale at every time Shinkansen train passed: a) timbre
and overall discomfort, b) psychological effects, c)
physiological effects. The auditory evaluation results showed
that high frequency sounds had a great influence not only
on their overall discomfort and psychological effects but
also on physiological effects.
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Organization:
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N. Declercq |
C. Koch, Physik.-Techn. Bundesanstalt (PTB) |
V. Sánchez Morcillo |
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Comparison of Cavitation Effect in Case of Fixed and Free Fibers in an Ultrasound Beaker
Taraka Rama Krishna Pamidi, Örjan Johansson, Torbjörn Löfqvist
[Abstract]
In this study, we investigate the impact of high-intensity
ultrasound treatment on the mechanical properties of
pulp fibers. The pulp fiber samples are sonicated in an
acoustically optimised beaker where high-intensity
ultrasound is generated using a tuned sonotrode device.
The idea is to create a controlled resonance to efficiently
enhance the sound pressure in the beaker. Input power is
90Watt. The objective is to define the difference between
freely suspended fibers in a beaker compared to keeping
fibers in a fixed position. The hypothesis is that fiber
treatment at a specific input power will be more efficient
in the case when fibers are kept in a high pressure zone.
Since the fiber wall is a layered structure, it is likely to
delaminate internally which will affect the mechanical
properties of the fiber. The effect on fiber properties is
verified by measuring the ultrasound attenuation spectra
for the treated fibers. The attenuation measurements are
based on measurements of a low-intensity ultrasound
pulse-echo technique. On a macroscopic scale, changes
in the attenuation spectra relates to a change in
mechanical properties of the fiber wall, since the
suspended fibers more or less retain their diameter and
length distributions.
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Time-resolved imaging of GHz acoustic waves in two-dimensional phononic crystals with an arbitrary-frequency technique
Osamu Matsuda, Hiroki Muramoto, Hiroki Nishita, Kentaro Fujita, Motonobu Tomoda, Oliver Wright
[Abstract]
By irradiating a medium with picosecond light pulses one can generate GHz surface acoustic waves. Their propagation can be monitored in the time domain with delayed probe light pulses. In addition, by spatially scanning the probe light focusing position, it is possible to monitor the spatiotemporal variation of the acoustic field. This can be used to study acoustic properties such as the dispersion relation of the surface acoustic waves. For such measurements, a periodic source of light pulses with a repetition rate of around 80 MHz is usually used, and so the frequency of the generated acoustic waves is limited to integer multiples of the repetition rate. We recently developed a technique to generate and detect arbitrary acoustic frequency components through the intensity modulation of the excitation, i.e. pump, light pulse train. In this paper, this arbitrary-frequency imaging technique is applied to study the acoustic properties of two-dimensional phononic crystals. The sample contains a square array of micron-scale holes on a Si substrate, and exhibits a phononic band gap around 0.5 GHz. We will present experimental results concerning acoustic wave propagation at frequencies around the phononic band gap.
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Attenuation Mechanisms of High-Frequency Acoustic Waves in Piezoelectric Cubic Crystals
Farkhad Akhmedzhanov, Sirojiddin Mirzaev, Ulugbek Saidvaliev
[Abstract]
Attenuation mechanisms of high frequency acoustic waves in
piezoelectric BSO and BGO crystals has been studied in the
frequency range of 0.41.6 GHz. It is shown that the main
mechanisms of attenuation are the Akhiezer mechanism and
the effect of dielectric losses due to the piezoelectric coupling
of deformation with polarization in piezoelectric crystals. To
describe the attenuation anisotropy of acoustic waves, an
acoustic anisotropy factor was introduced using imaginary
elastic moduli of these crystals. It is established that the most
contribution of dielectric loss to the attenuation of acoustic
waves is observed for the piezoactive transversal waves
propagating in the (100) plane. The dielectric loss can also
affect the frequency dependence of the attenuation
coefficient of piezoactive acoustic waves.
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Organization:
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A. Hurrell, Precision Acoustics Ltd |
G. Durando, INRIM -Istituto Nazionale di Ricerca Metrologica- |
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Absorbing target for radiation force measurements below 1 MHz
Megan Jenkinson, Andrew Hurrell
[Abstract]
Radiation force balance (RFB) measurements are a widely
accepted and standardised method of determining
ultrasonic power. Many RFBs incorporate an absorbing
target and at frequencies above 1 MHz there are a range of
ultrasonically anechoic materials that are available for this
purpose. Below 1 MHz wavelengths increase and the range
of candidate materials for use as absorbing targets is
dramatically reduced. This paper introduces a new anechoic target material
specifically designed for use in the frequency range 200 kHz
to 1 MHz. It is based upon a pre-existing and proven material
formulation but with modifications to its geometrical front
surface to enhance its echo reduction. Experimental results
are presented of both its performance in isolation, as well as
part of a lower frequency RFB system.
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Frequency and Power Limits of an Ultrasound Radiation Force Balance Brush Target
Subha Maruvada
[Abstract]
It is essential to know the acoustic power radiated by transducers used in high intensity therapeutic ultrasound (HITU) surgery devices for both safety and effectiveness considerations. The power radiated by medical ultrasound transducers usually is measured via radiation force balance (RFB) methods. However, for the high power, focused fields encountered in HITU applications, such measurements can be difficult due to the need for short measurement times to prevent transducer damage, RFB target heating, and bubble formation. A necessary criterion for accurate HITU power measurements is a robust absorbing target. We have previously established a custom-made brush target as effective for measuring HITU power levels up to 50 W and for frequencies between 1 and 3 MHz. In this work, the limits of our acoustic brush target have been further investigated over a wider range of frequencies and powers. This work demonstrates the usability of a RFB brush target over the frequency range 0.5 - 16 MHz and for acoustic powers in the clinical range of HITU treatment.
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Validation of Reflectance-Based Fiber-Optic Hydrophones
Sam Howard, Claudio Zanelli
[Abstract]
Reflectance-based fiber-optic hydrophones are frequently
used for high pressure ultrasound measurements. Often,
damage to the fiber forces the user to re-cleave the fiber-tip,
necessitating a means of re-validating the calibration over a
broad frequency range (often > 50 MHz). A previous study
proposed a validation method utilizing a stable, calibrated
pulsed ultrasound source operating at moderate diagnostic
levels (∼ 6 Mpa pk-pk). This presentation will briefly review this
approach, adding new results related to the overall
calibration scheme:
a) Comparison of the calibration from the pulsed source to
a conventional swept-frequency substitution method of
calibration
b) Recent experimental measurements of the effective
aperture from 2 to 15 MHz, fit to a model that indicates the
effective diameter is close to the nominal fiber core diameter
across a broad frequency range.
c) Recently compiled long-term stability data for a
proposed implementation of the pulsed check-source,
indicating a stability of better than 2% over the span of eight
years. 1.S. Howard, ”Calibration of Reflectance-based Fiber-
Optic Hydrophones,” IEEE_UFFC Ultrasonics Symposium, 2016 2.K. Wear and S. Howard, ”Directivity and Frequency-
Dependent Effective Sensitive Element Size of a Reflectance-
Based Fiber-Optic Hydrophone: Predications from Theoretical
Models Compared with Measurements,” IEEE -UFFC
Transacations, Oct. 2018.
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Measurement Parameters for the characterization of unfocused Extracorporeal Pressure Pulse Sources - Standardization of Biomedical Equipment
Friedrich Ueberle
[Abstract]
Extracorporeal Pressure Pulse Sources (EPPS) are in use in a variety of tissue and pain therapy applications, e.g. for the treatment of shoulder, heel spur and elbow pain, erectile dysfunction, cardiac pain, cellulitis etc. Today, a variety of second-generation devices, mostly using ballistic generation methods and claiming to provide ”radial shock waves”, is available. Nevertheless, a review of the publications on the EPPS shows that neither the characteristics of the soundfield nor the therapeutic effects in biological tissue are fully understood. Furthermore, clinical trial reports of EPPS therapies reveal contradictory results, mainly connected to a lack of knowledge on the physical properties of the pressure pulse EPPS field. As a proper measurement standard for unfocused EPPS devices is not in place, measurements for the certification process are carried out according to the IEC standard 61846, which was originally developed for the characterization of focused shockwave-lithotripters. Many of the parameters standardized in 61846 are not applicable or even misleading for the characterization of non-focused EPPS. A Chinese national standard YY 0950-2015 describes measurement methods for ballistic EPPS, mainly overall energy and impulse of the sources. This paper reports on the development of IEC 63045, the international standard on acoustic EPPS measurements.
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EURAMET EMPIR 18HLT06 RaCHy Project: Radiotherapy coupled with Hyperthermia (Induced by HITU)
Giovanni Durando, Piero Miloro, Volker Wilkens, Baki Karaboce, Jacco De Pooter, Gerard Van Rhoon, Gail Ter Haar, Barbara Caccia, Antonello Spinelli, Antonia Denkowa, Roeland Dijkema
[Abstract]
The main goal of external beam radiotherapy (EBRT) is to damage all cancer cells in a
tumour in such a way as to stop their proliferation. Unfortunately, normal tissue toxicity
can severely limit doses used, and thus it is useful to combine EBRT with other
treatments. An interesting approach to improving local tumour control is to combine
EBRT with hyperthermia.
The cell sensitisation induced by the heat is not completely understood, however
evidence suggest that it impairs cellular DNA repair, thus rendering radiation damage
more lethal.
During hyperthermia treatment the temperature distribution generated is crucial. High
Intensity Focused Ultrasound (HIFU) is a method that provides superior control of the
energy deposition and thus can homogenously increase the tumour temperature to the
target level. One of the main goals in the Radiotherapy coupled with Hyperthermia
(RaCHy) project is therefore to improve the efficacy, safety and range of applicability of
clinical HIFU treatments by providing validated methods for ultrasonic field
characterization, HIFU system performance testing, quality assurance and patient
exposure monitoring. The RaCHy project will address both the basic calibration and
specification of equipment (in particular characterisation of the acoustic pressure and
temperature distribution in water and in-vitro and in-vivo experiments).
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Organization:
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J. Raymond, University of Oxford |
S. Catheline, LabTAU, INSERM u1032 |
L. Verhagen, University of Oxford |
E. Konafagou |
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Local Phase Velocity Imaging (LPVI) as a New Technique for Shear Wave Elastography
Matthew Urban, Piotr Kijanka
[Abstract]
Shear wave elastography is used in many clinical settings to
measure the mechanical properties of soft tissues such as the
liver, kidney, breast, and thyroid. In most implementations, the
shear wave group velocity is measured using time-of-flight
methods in the time-domain. However, different factors related
to the estimation of the group velocity can cause variation in
the results. Factors such as acquisition parameters, processing,
tissue inhomogeneities and viscoelasticity can cause these
variations. We propose to use methods that rely on phase
velocities, shear wave velocities at specific frequencies, to
standardize measurements across different clinical applications.
We have developed methods for creating maps of phase
velocity using Local Phase Velocity Imaging (LPVI). The LPVI
method involves directional and wavenumber filtering before
applying various steps of Fourier-based operations to obtain
localized maps of phase velocity over large bandwidths. We will
demonstrate how LPVI works in numerical data, tissue mimicking
phantoms, and ex vivo and in vivo tissues such as liver and
kidney. The LPVI method has enormous potential because the
reconstruction frequency can be controlled, optimized, and
standardized for a wide array of clinical applications.
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Transcranial focusing of arbitrary ultrasonic fields using acoustic holograms
Sergio Jiménez-Gambín, Noé Jiménez, Jose M. Benlloch, Francisco Camarena
[Abstract]
We present ultrasonic phase plates that are able to
produce arbitrary field distributions, i.e., acoustic
holograms, inside the skull. Using 3D printing techniques,
we are able to produce lenses that correct the
aberrations of the skull and, simultaneously, produce
ultrasonic fields whose spatial distribution fits a target
organ at the central nervous system. In particular, using
experimental techniques on a human skull phantom and
full-wave simulations, several configurations are tested.
First, a multiple-point focusing lens is designed to
simultaneously focus at both human hippocampi. Second,
a beam following an arbitrary curved trajectory, i.e., a
self-bending beam is presented. Finally, we report a
holographic plate producing a broad focus that overlaps
with the left hippocampus. The results show that 3D
printed holographic lenses can be used to control the
spatial features of ultrasonic beams inside the skull in an
unprecedent manner using single-element ultrasonic
sources.
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High Frame Rate Ultrasound: Zooming into the Time Axis for Complex Flow Visualization
Alfred Yu, Billy Yiu, Jason Au
[Abstract]
With the advent of spectral Doppler and color flow imaging (CFI), ultrasound has been clinically assisting numerous vascular diagnoses in the past 30 years. However, spectral Doppler can only render such information at a single range gate, while CFI is known to be susceptible to numerous sources of estimation errors that hamper flow visualization quality, especially in complex flow situations. To overcome these limitations, our group has successfully harnessed the time-resolving power of high frame rate ultrasound (HiFRUS) to dynamically render complex flow dynamics over an imaging view at millisecond time resolution. Our HiFRUS technique, dubbed as vector projectile imaging (VPI), is capable of estimating flow vectors from multi-angle Doppler analysis and rendering these vector estimates as moving color-encoded projectiles. In a series of bifurcation phantom studies, VPI has vividly highlighted the flow jet emerging from the stenosis site and its trajectory. Also, it has enabled time-resolved observation of multiple flow disturbance zones. Moreover, in vivo investigations using VPI has revealed unique flow features in the carotid bifurcation of subjects who are experiencing lower-body negative pressure conditions. Taken together, VPI is poised to become a useful tool for obtaining new hemodynamic insights that are important to human physiology investigations.
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Viscoelasticity measurement comparison between microelastography and surface fluctuations
Pol Grasland-Mongrain, Stefan Catheline, Ali Zorgani, Ludovic Bellon, Thomas Gibaud, Sébastien Manneville
[Abstract]
Background: Microelastography is a recent technique
used to measure viscoelasticity of small objects at the
micrometer scale. A recent work has applied the
technique on a mouse oocyte, with a micropipette
inducing a 15 kHz shear wave, and the oocyte was
observed with a 200.000 fps camera through a
microscope. However, this technique lacks of gold
standard due to the high frequency of vibration. In the
other hand, another technique observing surface
fluctuations allows to measure sample viscoelasticity at a
wide range of frequencies. Aim: The objective was to compare both techniques on a
well-defined viscoelastic medium. Methods: A viscoelastic carbopol sample was built.
Microelastography used a 50 um diameter pipette to
induce the shear wave at various frequency. An high
speed camera tracked the shear wave. Elasticity
reconstructed with passive elastography. The second
technique was built using a 632 nm laser deviated towards
the object surface with a prism, and a CCD camera to
measure laser deviation Results and conclusions: Viscoelastic measurements were
comparable in both methods. Differences are supposed to
be due to elasticity variation between surface and bulk of
the medium.
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Organization:
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J. Fels, Teaching and Research Area of Medical Acoustics |
A. Raake, TU Ilmenau |
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Cognitive indicators for acoustic source localization and presence in a vivid 3D scene
Patrick Ruediger, Jan Spilski, Nûjîn Kartal, Sebastian Gsuck, Nils Ove Beese, Sabine Schlittmeier, Thomas Lachmann, Achim Ebert
[Abstract]
The easy access and availability of Virtual Reality (VR)
technologies open up more and more application fields.
Unfortunately, current VR applications hardly or only very
rudimentarily take acoustics into account, so that much of
the technology’s potential remains untapped.
In this paper, we present the results of our experiments for
localizing audio signals and presence using VR technologies.
We used the virtual representation of a realistic scene (360
movie shot) to accurately place sound signals in the three-
dimensional space using object-based audio. We manipulated
the volume and the setting of the environmental sound
(different positions and types of audio signals). Additionally,
we compared the effect of using simple stereo sound with
spatial stereo which uses the overlap of the field of hearing
(spatial sound) similar to the techniques used for
stereoscopic depths perception in 3D movie theaters. Depths
cameras and gyroscope, VR Glasses with Eye-tracker and
flying stick controllers were used by the participants to point
out the source of the audio signals. We tracked participants’
reaction time, accuracy, eye movements, and whole body
movements. In the light of our experimental results, we
discuss implications for practice, cognitive science and future
VR research.
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The Ability to Allocate Attentional Resources to a Memory Task Predicts Speech-on-Speech Masking for Older Listeners
Frederick Jerome Gallun, Kasey Marie Jakien
[Abstract]
To examine the degree to which attention and working memory
tasks predict performance in complex auditory environments, 54
participants (21-79 yrs) with a range of pure-tone hearing
thresholds completed both an auditory/visual working memory
(WM) span experiment and a complex speech task. WM tested
both modalities alone on a simple memory span task for strings
of letters or digits and both modalities together on selective and
divided attention span tasks. The speech task used three closed-
set sentences presented simultaneously via earphones using a
Virtual Spatial Array, such that the target sentence was always
at 0∘ azimuth angle and the maskers were either colocated or
positioned at +-45∘. Each condition was tested four times. WM
span under conditions of selective and divided attention were
highly correlated with speech performance, accounting for 30%
of the variance in the colocated condition 44% of the variance
in the separated condition. These results suggest that simple and
rapid tests of the ability to allocate attentional resources to
sensory stimuli can help explain some of why people of similar
age and hearing thresholds differ in their ability to understand
speech in complex acoustical environments.
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Examining auditory selective attention in realistic, natural environments
Josefa Oberem, Julia Seibold, Iring Koch, Janina Fels
[Abstract]
The topic of the present collaborative project (Medical Acoustics and Cognitive Psychology) is the exploration of cognitive control mechanisms underlying auditory selective attention. The aim is to examine the influence of variables that increase the complexity of the auditory scene with respect to technical aspects (dynamic binaural hearing with consideration of room acoustics and head movements) and that influence the efficiency of cognitive processing. Using a binaural-listening paradigm, the ability to intentionally switch auditory attention in various anechoic and reverberating setups was tested. The paradigm consists of spoken word pairs by two speakers which were presented simultaneously to subjects from two of eight azimuth positions. The stimuli consisted of a single number word, (i.e., 1 to 9), followed by either the direction ”UP” or ”DOWN” in German. Guided by a visual cue prior to auditory stimulus onset indicating the position of the target speaker, subjects were asked to identify whether the target number was numerically smaller or greater than five and to categorize the direction of the second word. Reproduction techniques and reverberation times were varied to analyze influences of the reproduction method in reaction times and error rates.
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Attentional mechanisms in static and dynamic cocktail-party listening
Hartmut Meister, Fabian Wenzel, Axel Gehlen, Martin Walger
[Abstract]
During daily-life communication multiple talkers frequently speak at the same time. This was coined by the term ”cocktail-party problem”. These situations pose high demands on both the auditory system as well as cognitive capacity of the listener. Particularly attentional mechanisms are crucial for cocktail-party listening: If listening is ”static” the aim is to focus attention on one talker. However, dividing and switching attention also play a role since the talker of interest may dynamically change from time to time in an unpredictable manner.
Since both hearing and cognition typically decline with age it is important to shed light on attentional mechanisms associated with cocktail-party listening. Therefore, we investigated performance of young and older listeners in a static and dynamic three-talker paradigm. Target sentences indicated by a particular call-sign were presented with (”static”) and without (”dynamic”) a priori knowledge about the talker of interest. Different target talker switches were considered in the dynamic situation. The listener’s task was to repeat back the target sentence and to ignore the competing sentences.
The presentation disentangles different attentional mechanisms, such as focused attention in the static situation and general as well as specific switching costs in the dynamic situation. Supported by DFG (ME 2751/3-1)
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Leveraging adaptation to study perceptual weighting of interaural time differences
Antje Ihlefeld, Nima Alamatsaz
[Abstract]
An important question in auditory cognition is how we perceive
the location of an object in space. Converging evidence from
animal models and humans suggests that when judging sound
direction, the central nervous system weighs the anticipated
reliability of binaural cues. Here, we used short-term adaptation
to bias normal-hearing listeners towards source direction
favoring either the left or the right frontal quadrant. Listeners
rated perceived laterality of tokens of band-pass filtered noise
(300 Hz - 1200 Hz) with interaural time differences that were
randomly selected from a uniform distribution spanning either
-375 to 0 µs or 0 to 375 µs. Using non-linear mixed effects
modeling of behavioral laterality reports, we tested how
exposure to source quadrant affects how listeners weigh the
reliability of interaural time differences. The cue reliability
hypothesis predicts that perceived direction should be skewed,
such that unreliable frontal source angles are more affected by
short-term adaptation than the more reliable lateral source
angles. Alternatively, short-term adaptation may affect all
source angles equally, predicting an overall shift in perceived
direction. Results show that frontal angles are more strongly
affected by short-term adaptation than lateral angles,
supporting the cue reliability hypothesis.
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Organization:
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D. Manocha, University of Maryland |
P.W. Robinson, Oculus & Facebook |
A. Plinge, Fraunhofer IIS |
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The Use of Inertial Measurement Units in Virtual Reality Systems for Auralization Applications
Kristian Jambrošić, Miljenko Krhen, Marko Horvat, Tin Oberman
[Abstract]
Auralisation finds increasingly more important applications in
everyday life with the development and broad availability of
virtual reality technologies and systems. This paper presents a
new research project focused on auralisation implementation in
different hardware systems, mainly by using binaural technology
with an added head tracking system. Beside commercially
available VR systems, the possibility of using inexpensive inertia
measurement sensors added to simple embedded systems will
be investigated as an alternative auralisation system for adding
aural experience in VR applications. Additionally, the possibility
of using widely available smartphones with already integrated
movement and direction sensors for auralisation purposes using
a binaural system will be examined. The listed platforms will be
measured and compared according to their precision, latency
and quality of the virtual sound field on practical auralisation
examples in laboratory conditions, and also in real life situation
in-situ. First measurements of sensor parameters and
characteristics, such as delay, sensitivity and accuracy, are
presented in the paper for both IMU sensors connected to an
embedded system and sensors integrated in smartphones.
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An Attention-Guided Algorithm for Improving the Performance of Acoustic Simulations
Hanna Autio, Delphine Bard-Hagberg
[Abstract]
When performing acoustic simulations with the purpose of auralization, there is a trade-off between accuracy and speed. In real-time simulations of virtual reality, finding the balance of this trade-off is paramount to achieving the desired result. If successful, the simulation speed is sufficient to provide a seamless acoustic experience as the agent moves around the space, while still being accurate enough to be realistic. It is generally accepted that a 20ms update interval for the impulse response is sufficient for achieving proper interactivity in most applications. However, reaching this threshold without degrading the quality of simulation too badly can be challenging indeed for complex scenes. In this paper, a compromise between interactivity (response time) and accuracy is suggested for raytracing simulations. This compromise mimics the behaviour of the listener or the agent, prioritizing speed or accuracy based on how the agent behaves. When the listener is actively moving around the space, interactivity is prioritized. When the listener stands still, fully immersing in the experience, accuracy is improved. This is achieved by exploiting a fundamental truth of Monte Carlo simulations: Convergence improves with more samples.
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Evaluation and comparison of novel music experiences in augmented reality
Arto Juhani Lehtiniemi, Jussi Leppänen, Henri Toukomaa, Antti Eronen
[Abstract]
This paper evaluates and compares different ways of presenting and experiencing music in six degrees of freedom (6DoF) augmented reality (AR). A musical work specifically composed and arranged for 6DoF AR consumption was presented to users in three ways; as a stereo downmix from virtual loudspeakers, as audio objects and as interactive (movable) audio objects. The audio objects and virtual loudspeakers were visualized as icons in the AR scene to help the user to move and interact with the objects. A qualitative and quantitative evaluation was conducted with 12 participants to evaluate the quality of each presentation and experiencing technique. The participants found the experiences with audio objects to be most innovative and most captivating. There was a clearly bigger effort and higher mental demand in accomplishing tasks with the interactive audio objects when compared to the virtual loudspeakers. Both of the audio object experiences were found to be more natural compared to the virtual loudspeaker experience. The overall results indicate that generation of rich and specifically tailored content for new realities is valuable and was perceived well by the users.
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Evaluation of the effect of head-mounted display on individualized head-related transfer functions
Maria Cuevas-Rodriguez, David Lou Alon, Samuel Clapp, Philip W. Robinson, Ravish Mehra
[Abstract]
Head Mounted Displays (HMDs) have become popular within
the research community for use in acoustic experiments, such as
perceptual evaluation of virtual spatial audio or Head Related
Transfer Function (HRTF) measurement processes. However,
wearing an HMD while listening to sounds can interfere with the
acoustic waveform that reaches the listener’s ears, modifying
the audio signals. In this work, we study the effect of HMDs on
individualized HRTF, comparing the individualized HRTFs distorted
by the HMD with a generic HRTF. We have collected acoustic
HRTF measurements, with and without the HMD, for 24 human
participants and a manikin head, over 612 directions (including
different azimuths and elevations). An objective evaluation
indicates that although there are frequency-dependent
directional regions where the distortion caused by the HMD is
higher, many of the individualized HRTF characteristics are
preserved. A MUSHRA test with 15 subjects compares
individualized HRTFs and a generic HRTF, both with and without
the HMD, for certain directions. The perceptual evaluation
reveals a significant perceptual effect of the HMD on the
measured individualized HRTF, but it also shows that this effect is
significantly smaller than the difference between individualized
HRTFs and generic HRTF, as indicated by the objective
evaluation.
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Spatial Upsampling of Sparse Head-Related Transfer Function Sets by Directional Equalization - Influence of the Spherical Sampling Scheme
Johannes M. Arend, Christoph Pörschmann
[Abstract]
Many immersive audio applications rely on a dense set of head-
related transfer functions (HRTFs). However, often only
measurements on a specific sparse grid are available. To obtain
dense HRTF sets from sparse measurements, one common
approach is to apply spatial interpolation in the spherical
harmonics (SH) domain. However, the SH representation of
sparse HRTF sets is order-limited, leading to spatial-aliasing
artifacts. In a recent publication, we presented the so-called
SUpDEq method (Spatial Upsampling by Directional
Equalization) for spatial upsampling of sparse HRTF sets. The
approach is based on a directional equalization of the sparse
set prior to the SH transform to remove direction-dependent
temporal and spectral components. This significantly reduces
the spatial complexity of the sparse set, allowing for an
enhanced interpolation at reduced SH orders. In this study we
investigate how different spherical sampling schemes for the
sparse measurements affect the performance of usual SH
interpolation and the SUpDEq method. For this, we compare
spatially upsampled HRTF sets originally based on sparse
Equiangular, Gaussian, Lebedev, and Fliege grids at various
spatial orders to a reference. The influence of the different grids
on the spatially upsampled HRTF sets are assessed spectrally,
temporally, and by means of localization models.
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Perceptual Relevance of Speaker Directivity Modelling in Virtual Rooms
Henning Steffens, Steven Van De Par, Stephan D. Ewert
[Abstract]
Sound sources like a human speaker, musical instruments, and
loudspeakers typically radiate sound with a frequency-
dependent directivity pattern. If a human speaker faces the
receiver, more high-frequency components reach the receiver
as when the speaker turns in the opposite direction. This is
perceptually easily detectable, particularly in anechoic
conditions. In rooms, early reflections and late diffuse
reverberation, which both can be differently affected by the
directivity pattern of the source, occur in addition to the direct
sound. While source directivity appears relevant for distinct early
reflections, it is unclear to which extent the perceptual effects
average out for later reflections and diffuse reverberation.
Here we assess to what extent directivity of a human speaker
should be modelled for early reflections and diffuse
reverberation. For this we used a hybrid approach to simulate
and auralize binaural room impulse responses [Wendt et al.,
JAES, 62, 11 (2014)], combining an image source model (ISM) for
early reflections with a feedback delay network for diffuse
reverberation. The approach was extended by a directivity filter
based on a spherical head model. Listeners distinguished
between source orientations solely based on ISM-directivity and
detected average source directivity filtering even in reverberant conditions.
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Subjective Performance Criteria for Mixed-Reality Immersive Audio
Jean-Marc Jot, Rémi Audfray, Justin Mathew, Dan Mauney
[Abstract]
An overall design objective of virtual, augmented or mixed-reality audio systems is to maximize the
plausibility of an atificial experience presented to a user via earphones or a head-mounted audio-visual
display apparatus. A number of metrics have been proposed for evaluating the subjective performance of
spatial audio rendering techniques in a variety of playback environments. They include, for instance,
positional audio rendering accuracy measures relevant for quantitatively evaluating the subjective
performance of virtual reality (VR) audio systems. The design objective of VR systems amounts to
suspension of disbelief, because the user is informed that every aspect of the experience is computergenerated
and presented via a display system that occludes and replaces the real world. Augmented or
mixed reality (AR/MR) audio systems, on the contrary, aim to display digital objects overlayed in the
world perceived naturally by the listener. A challenge of AR/MR applications is therefore to prevent
dissonance between the perception of computer-generated objects and the listener’s expectations (for
instance, avoid artifacts that might reveal that some of the perceived objects are artificial). We propose a
subjective-assessment rubric that we have begun employing in our AR/MR audio system performance
surveys, centered initially on loudness and positional audio rendering.
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Organization:
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L. Savioja, Aalto University |
D. Murphy, Audio Lab, Department of Electronic Engineering, University of York |
F. Wefers, International Audio Laboratories Erlangen, University of Erlangen-Nuremberg |
T. Kuhlen, RWTH Aachen |
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Reconstruction of binaural room impulse responses using spherical harmonics
Maarten Hornikx
[Abstract]
Dynamic virtual acoustic environments based on modelled signals can be based on
online or offline calculations of the binaural room impulse responses. For the former,
calculations have to be very fast for a real-time dynamic experience, while for the latter
the storage requirements may become huge. An approach to reconstruct binaural room
impulse responses based on offline calculated impulse responses is proposed here. It
relies on a dual-radius open sphere with receiver points around the listener’ position,
allowing to post-process the signals to any desired head-rotation within the sphere using
spherical harmonics. An inverse calculation is done as well, from the centre of the
receiver sphere to a dual-radius sphere around the source position. Based on these
calculations, a plane wave decomposition is applied in the post-processing algorithm to
compute the time-dependent angle(s) of incidence at the receiver and angle(s) of
departure at the source, such that the impulse responses at the receiver dual-radius
sphere can be reconstructed for a translated source position. As a result, based on two
room acoustic calculations, the post-processing algorithm allows for head-rotations and
source translations in the virtual environment. Prospects and limitations of this approach
will be discussed.
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Local Directional Source Modeling in Wave-based Acoustic Simulation
Stefan Bilbao, Jens Ahrens, Brian Hamilton
[Abstract]
Time-domain wave-based simulation approaches such as the
finite difference time domain (FDTD) method allow for a
complete solution to the problem of virtual acoustics over the
entire frequency range, in contrast with the methods of
geometric acoustics which are valid in the limit of high
frequencies. They also allow for flexible modelling of sources
and receivers, due to the inherently local nature of the
computation, and complete access to the computed acoustic
field over an enclosure. In this paper, a method for the
emulation of sources of arbitrary directivity is presented, framed
directly as an inhomogeneous wave equation. The additional
terms in the wave equation take the form of Dirac delta
functions and their distributional derivatives, and collections of
such terms may be associated directly with an expansion of
source directivity in terms of spherical harmonics. The local
nature of the model implies a) a locally-defined efficient
computational approach for wave-based methods defined
over a spatial grid, and b) the possibility of direct extension to
the case of moving sources. Numerical results are presented.
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Room acoustics modeling using a hybrid method with fast auralization with artificial neural network techniques
Roberto A. Tenenbaum, Filipe O. Taminato, Viviane S.G. Melo
[Abstract]
One of the objectives of the development of numerical
techniques in acoustic virtual reality systems and the production
of reliable and plausible auralizations is to reduce the
computational cost and, simultaneously, to guarantee the
sound quality. In this article, a new technique is presented to
model the head-related impulse responses and its filtered
versions, necessary for the binaural room impulse responses
computation. Artificial neural networks of the radial basis
functions kind are used. A set of such networks is trained and
tested to cover the entire auditory space around the human
head. Each neural network is associated to a given direction
and has, as input, the power spectrum of the sound wavefront
in octave bands that reaches the receiver and, as output, the
filtered head-related impulse response, in the corresponding
direction. With this strategy, the computation is performed
directly in time domain, bypassing the need of convolving each
wavefront with the head-related impulse response for the
considered direction. As it is shown, a reduction of the
computational cost of about 85% is achieved. The result
obtained with the proposed method is compared with that
computed with the classical convolution method, both in time
and frequency domain showing a negligible difference.
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Sound field reproduction with exterior field cancellation using variable-directivity loudspeakers
Bokai Du, Michael Kohnen, Michael Vorländer, Xiangyang Zeng
[Abstract]
Sound field reproduction aims to create or reproduce a
certain sound environment where not only the audio
content, but also the spatial property of the sound field
has to be preserved. For a real system which is usually
placed in a ”listening room”, the reproduction
performance is affected or even destroyed by
superimposing of reflections from walls. As a result, it is
important to control the sound energy which radiates to
the space outside the loudspeaker array. Compared with
omnidirectional loudspeakers, the use of variable
loudspeakers can eliminate the exterior field according to
the Kirchhoff-Helmholtz integral equation. This paper
investigates the performance of variable directivity arrays
using pressure matching-acoustic contrast control(PM-
ACC) and compared its performance with that of
omnidirectional one.
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Auralization of interactive virtual scenes containing numerous sound sources
Lukas Aspöck, Lucas Mösch, Jonas Stienen, Michael Vorländer
[Abstract]
Typical everyday situations usually contain a large number of sound sources. In virtual reality applications, where the processing demands for the acoustical rendering of a scene should be kept a low level, it is challenging to simulate and spatialize a high number of virtual sound sources. In extension of the state of the art, this work presents different solutions for rendering sound sources in virtual scenes with varying level of interactivity and complexity. For a binaural free-field auralization of up to hundreds of virtual sound sources, a model based on k-means clustering was recently developed, with the main objective to limit the number of required convolutions. To improve the perceptual quality of the rendering, the model was extended with an efficient correction of the interaural time difference of each virtual sound source. In addition to a brief benchmark analysis of the rendering module, it is also described how this clustering approach was integrated in the open source auralization framework Virtual Acoustics. Furthermore rendering configurations and combination possibilities with other rendering approaches are discussed.
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Rendering of scattering effects from finite objects using neural network-controlled parametric digital filters
Ville Pulkki, U. Peter Svensson
[Abstract]
The effect of scattering is implemented using parametric
filtering structures and the parameters for the filters are
estimated using artificial neural networks. The networks are
trained with computed scattering data of large numbers of
thick plate-objects, using an edge diffraction-based method.
The input data consists of a set of geometric features
describing a large amount of source-object-receiver
configurations, and the target data consists of the filter
parameters computed using measured or modelled data that
implement the spectral effect of scattering of the
configurations. In a dynamic test scenario with a 3D plate
object that reflects and diffracts sound,
the approach is shown to provide a similar spectrogram when
compared with a reference case, although some spectral
differences are present. A perceptual test shows that the
proposed technique produces a high level of plausibility,
which is comparable to the plausibility obtained with a state-
of-the-art model of scattering, in particular compared with
the modeling of reflections only.
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A real-time virtual reality building acoustic auralization framework for psychoacoustic experiments with contextual and interactive features
Anne Heimes, Imran Muhammad, Michael Vorländer
[Abstract]
The auralization of sound insulation between the adjoining rooms in virtual reality (VR) environments describes the sound transmission between enclosures, where sound insulation prediction methods are applied for a high quality auditory stimuli. In this context, a real-time building acoustic auralization framework in 3D audio-visual technology is developed to introduce more realism and contextual features into psychoacoustic experiments.
This paper describes the design and implementation of this framework that is a vital part of 3D-immersive sound rendering VR-Systems. This auralization framework relies on up-to-date knowledge of building acoustics techniques and enables a physically accurate airborne sound insulation auralization in virtually constructed environments, including important effects such as sound transmission, bending wave effects, source directivities, and source/receiver room acoustics. In spite of this realistic sound insulation rendering between adjoining rooms, not only the indoor sources are supported at runtime, but also outdoor moving sources such as vehicles.
All features are evaluated by investigating both, the overall accuracy of the building acoustics filters and the performance of implemented algorithms. Psychoacoustic experiments are conducted by presenting a dynamic virtual building scenario to the user with Head-Mounted-Display. First results of noise effects on the performance of humans are presented with contextual and interactive features.
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Rendering Environmental Noise Planning Models in Virtual Reality
Arne Nykänen, Roger Johnsson, Jonas Aråker
[Abstract]
In building and infrastructure projects sound design and requirement specification are often complicated by difficulties in understanding how the planned built environment will sound. Information about sounds is almost exclusively provided as sound pressure levels and sound reduction indices. It is difficult to understand how an environment will be perceived solely based on such data. VR models with sound give an experience much easier to comprehend. In this study, VR models were developed based on first order Ambisonics recordings. Such recordings provide spatial information and can be real time rendered based on the listener’s orientation. However, the recordings must be made in discrete points and therefore a model for cross-fading and mixing was developed. Recordings of road and railway sounds were made in a two dimensional grid and mixed and crossfaded based on the position of the listener. The sound levels were adjusted to match calculated levels from noise planning models. The spatial density of the grid of recordings, the cross-fading function, and the mix of recordings were varied and the realism of the models was assessed in listening tests. The results give guidance on how Ambisonics recordings could be mixed in order to achieve realistic sound in VR.
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Simplification of Reflection Orders in Virtual Soundscapes through a Subjective Evaluation
Chunyang Xu, Jian Kang
[Abstract]
Soundscape research stresses the interaction between
human beings and sound environments. Virtual reality (VR)
technologies provide a vivid approach to boost the virtual
soundscape experience for people. However, the real-
time VR audio simulation cannot process the complex
acoustic conditions (e.g., boundary conditions) due to the
high computational cost up to now. This study therefore
aims to investigate the applicability of less reflection orders
for reproduced sounds in virtual soundscapes. To test the
effect of reflection orders perceived by people, a series of
subject evaluations was carried out based on audio-visual
VR experience with different scale public squares
reproduced in Unity to provide visual scenarios for the
participants. The participants heard typical sounds (e.g.,
fountain and birdsong) existing in these squares, and gave
their perception evaluation to these sounds. The results
show that these reproduced sounds and typical building
spaces can employ less reflection orders which render
similar realism for participants in virtual soundscapes.
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Acoustic radiation modes and active structural acoustic control of coupled enclosure
Haichao Zhu, Rongfu Mao, Jinlong Liao, Changwei Su
[Abstract]
There are two problems in the active structural acoustic control (ASAC) of weak coupled enclosure. First, structural modal information is required to obtain the acoustic radiation mode (ARM) of coupled enclosure. Second, the control strategy of ASAC of coupled enclosure is not precise enough. To overcome these problems, by analogy to the ARM theory of free space, the acoustic potential energy (APE) is expressed in quadratic form of normal velocity on coupling surface and the ARM of coupled enclosure is redefined. Thus, the ARM theory of coupled enclosure which is consistent with that of free space and convenient for application is formed. Then, the control strategy based on the dominant acoustic radiation mode (DARM) is proposed, and corresponding ASAC model of the coupled enclosure is established. Simulation analysis and experimental research are carried out with elastic plate-rectangular coupled enclosures. Both simulation and experimental results show that better control effect can be obtained by controlling DARM contributed APE. Therefore, the feasibility and effectiveness of the proposed DARM control strategy are proved
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Nonlinearities in Sound Field Control Systems
Elias D. Lumpert, Finn T. Agerkvist, Franz Maria Heuchel, Jonas Brunskog
[Abstract]
If a loudspeaker is driven at high levels its output is likely to
be affected by nonlinearities. The result is a change of the
loudspeakers response for different levels and frequency
content. When looking at a sound cancellation {or sound field
control} system the introduction of such nonlinearities lowers
its performance. This paper investigates the extent of such
effects and looks at strategies to counteract the problem,
which has received limited attention so far. In the application
of sound field control for outdoor concerts the requirement of
high excitation levels and the usage of cost-effective
speakers makes this issue even more relevant. The goal of
the paper is to discuss the effects of nonlinearities on sound
cancellation systems and investigate measures to supervise,
analyse and optimise a running system. The paper
specifically focuses on quantifying the effect of compression
on the output of the system at the input frequency, finding a
model to represent the coherent distortion and counteract
accordingly. A comparison of different strategies to
counteract the effect of nonlinearities is presented.
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Acoustic Path Database for ANC In-Ear Headphone Development
Stefan Liebich, Johannes Fabry, Peter Jax, Peter Vary
[Abstract]
Active noise cancellation (ANC) for headphones is an emerging field with growing interest in consumer applications. Developing such an ANC system involves modeling the system from measurements, designing the algorithms and simulating the performance before building a real-time system. Modeling the system considers the transfer functions between loudspeaker and microphones, called secondary and feedback path, as well as the transfer function between the microphones, called primary path. These paths are modeled with varying degrees of accuracy in the development process: From coarse physical models, e.g., with pure delay; over measured paths under ideal conditions, called nominal; to considering uncertainty due to various operating conditions.
We saw of a lack of publically available measured paths of ANC headphones. Hence, this publication describes an alongside published database containing two measurement sets of the Bose QC20 in-ear headphone without ANC electronics. The first set contains primary, secondary and feedback paths of 23 subjects measured in an acoustic booth. The second set contains primary paths measured for equally distributed directions on a horizontal plane. Measurements were done in a semi-anechoic chamber with a dummy head. The database can be used to consider inter-person differences and direction-of-arrival dependency in algorithm design and simulation.
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Construction of IIR Filter for Adaptive Noise Control System of Boat Noise Reduction
Haruki Fukatsu, Kenji Muto
[Abstract]
Some boat noise is loud and transmitted to the room of the apartments or the companies near the canals in Tokyo. Our aim is the realization of the silence space using damping system with adaptive noise control. Here, we analyzed the characteristic of boat noise and investigated the optimal adaptive filter for the damping of the window vibration at acoustical path D. The investigation was the step size and the taps number of IIR filter to improve the performance with DSP(digital signal processor) of the ANC(adaptive noise control) system. The result showed that the boat noise had composed peak frequency and some dominant frequencies caused by the boat engine which was long noise continuously. Therefore we designed the IIR filter of the adaptive filter effectively. The system showed that the IIR adaptive filter became high performance for the boat noise and the reduction of the boat noise with computer simulation.
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Numerical acoustic modelling of a ventilation unit by 3D FEM and application to the design of an ANC feedforward system.
Stéphane Lesoinne, Jean-Jacques Embrechts, Guillaume Vatin, Bastien Ganty, Yves Detandt
[Abstract]
Strong insulation in modern buildings and housing requires efficient ventilation systems. A popular solution is a decentralized system with one ventilation unit per room, but the noise emitted by such a unit then has a major impact on the comfort. In this research, it is intended to equip an existing ventilation unit with an active noise control (ANC) system to reduce its low-frequency noise emissions. The ANC system will be hosted in an additional duct deporting the air inlet aperture approximately one meter away from its original position. The one-channel feedforward ANC system has a reference microphone located between the fan and the anti-noise loudspeaker and an error microphone at the duct’s end. The transfer functions have been computed by a 3D FEM solver, between the fan and several reference and error microphone locations. Several combinations of reference/error microphone positions were then tested regarding the possible active attenuation of a white noise emitted by the primary source. Optimal control by the ANC controller was assumed, the optimal filter being computed with the help of the corresponding transfer functions. Theoretical attenuations were obtained and optimal positions were finally defined for the different transducers of the ANC system.
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Design of Kronecker Product Beamformers with Cuboid Microphone Arrays
Xuehan Wang, Jacob Benesty, Gongping Huang, Jingdong Chen, Israel Cohen
[Abstract]
Microphone array beamforming has been widely used in a
wide range of acoustic applications. To make it effective in
suppressing noise yet able to preserve the fidelity and
quality of the broadband speech signals of interest, the
beamformer needs to be designed with high spatial gain,
consistent responses at different frequencies and high
robustness against array imperfections. A great deal of
efforts have been devoted in the literature to achieving this
goal, among which the Kronecker product beamforming
method developed recently has demonstrated some
interesting properties. This method decomposes the entire
beamforming filter into two smaller filters, each
corresponding to a virtual subarray. Then, each sub
beamforming filter is optimized so that the global
beampattern and directivity factor meet the design target.
This approach has been investigated for two dimensional
microphone arrays such linear and rectangular ones. In this
work, we extend this method to three-dimensional arrays.
Focusing on cuboid shape of microphone arrays, we discuss
how to decompose the global beamforming filter into a
Kronecker product of two to three sub-filters. Algorithms are
presented to design each sub-filter so that the global
beamformer has a high directivity factor and can be
steered flexibly in the three-dimensional space.
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Impulse source localization with background noise in a reverberant environment by multiple sensors
Tiangang Wang, Yat-Sze Choy, Jungang Zhang
[Abstract]
During vibration testing of space hardware, breaking sounds can be heard when cracks happen in the structure. However, much efforts are needed to determine the position of fault in a complex large-scale model such as a satellite with background noise in reverberant environment. This paper focuses on implementing non-contact localization of the sound sources which are attributed to structure failures. In order to achieve this goal, a system for fault detection by sound source localization with limited number of sensors under the moderate reverberant environment is established based on the TDOA technology. The low-quefrency filtering technique is adopted as a dereverberation pre-processing approach to alleviate the reverberation effects due to the echoes from the reflections of the laboratory. In order to reject the interference sound sources, two criteria, the geometrical criterion and the cyclical check criterion, are introduced. A series of experiments are conducted to verify the performance of this system. The results show that this system can localize the crack sources and hitting sources accurately in a short time.
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On the detection quality of early room reflection directions using compressive sensing on rigid spherical microphone array data
Frank Schultz, Sascha Spors
[Abstract]
The estimation of acoustic reflection coefficients from in-room measurements using one or more sources and microphone arrays has been addressed in various ways. One particularly illustrative method is the plane wave decomposition. Peaks in this representation can be assigned to the location and strength of mirror image sources. Traditional approaches employ rigid spherical microphone arrays together with modal beamforming. Compressive sensing (CS) techniques aim at undersampling by assuming sparsity of the soundfield in some given representation. By careful design of the sensing matrix, the number of required measurements can be considerably reduced compared to Nyquist sampling and reconstruction. However, often sparsity of the problem at hand is not given and a sufficient low mutual coherence of the sensing matrix is violated, yielding CS reconstruction with low robustness. We aim at robust detection of early, discrete reflections by means of CS using rigid spherical microphone array data. We discuss the interaction of room characteristics, sensing matrix and technical measures to qualify the detection performance. The influence of practical limitations, like the number of microphones and their self-noise is investigated, as well as the overall gain of applying CS to the problem compared to traditional modal beamforming techniques.
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Reducing Transfer Function Measurement in Local Sound Field Reproduction using Acoustic Modelling
Qiaoxi Zhu, Xiaojun Qiu, Philip Coleman, Ian Burnett
[Abstract]
Broadband local sound field reproduction over an extended spatial region is a challenging problem, when only limited transfer function measurement is available. In this paper, an acoustic modelling based approach is proposed to reduce the required transfer function measurements in the local sound field reproduction. The proposed method only requires measuring the transfer functions from each source to a few samples over the boundary of the controlled region, and the transfer functions to the samples inside the controlled region are estimated through an efficient acoustic modelling. The simulation demonstrates that the proposed method requires less transfer function measurement than existing methods such as the least squares and the spatial harmonic decomposition methods.
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Control of Sound Pressure in Audible Spot using Parametric Speakers
Takumi Hakamata, Hiroyoshi Yamashita, Keisuke Watanabe, Kotaro Hoshiba, Takenobu Tsuchiya, Nobuyuki Endoh
[Abstract]
We have studied about forming audible spot using parametric
speakers for playing an audible signal to particular person. In
previous work, the method to form audible spot using two
parametric speakers was proposed. In the method, different
tones of ultrasonic signals which have different frequency sent
by two parametric speakers are played at crossed point of
each signal. However, it has the problem that formed audible
area was not stable because frequency characteristics of each
speaker were not considered. This paper describes the
technique to form stably audible area. In this technique, sound
pressure of each speaker is controlled considering frequency
characteristics. The proposed method was evaluated by
experiments. Forming circular audible spot which diameter is 110
mm and sound pressure level is over 60 dB was tried. Although
audible area changed about 75% when sound pressure was
uncontrolled, stably audible spot could be formed by the
proposed method with the change less than 20%. As the result, it
is confirmed the usability of the proposed method.
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Left-right sound localization outside loudspeaker positions in stereo reproduction with parametric loudspeakers
Shigeaki Aoki, Kouki Ito, Kazuhiro Shimizu, Suehiro Shimauchi
[Abstract]
The parametric loudspeaker is known as a super directional
loudspeaker by using nonlinear interaction between
ultrasounds. At first, the listening tests in stereo reproduction with
the parametric loudspeakers and the ordinary loudspeakers
were conducted. These loudspeakers were set at angle of -30
and +30 degrees. The obtained test results were analyzed and
discussed. It was confirmed that when ILDs were used as
binaural information, the directions of sound localization with
the parametric loudspeakers were more outside than those
with the ordinary loudspeakers. The obtained results were
interesting especially when ILDs were minus infinity and plus
infinity. In using the parametric loudspeakers, the directions of
sound localization were outside the loudspeaker. On the other
hand, in using the ordinary loudspeakers, those were not
outside the loudspeaker but inside. Moreover it was confirmed
that in the sound reproduction with the right and left
parametric loudspeakers by using HRTFs of -90 and 90 degrees
in the horizontal plane, the sound localization was able to
control to localize outside the position of the loudspeakers.
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Pressure-matching-based 2D sound field synthesis with equivalent source array
Izumi Tsunokuni, Kurokawa Kakeru, Yusuke Ikeda
[Abstract]
Pressure matching(PM) Method is one of well-known effective
methods for physical sound field synthesis using massive loudspeaker
array. In PM method, to match a sound pressure at each matching
point as the desired one, all transfer functions between the
loudspeakers and matching points must be measured. Thus, as the
number of matching points and loudspeakers increases, it is more
difficult to measure the transfer functions by using a microphone. In
addition, it is also difficult to place a microphone with high accuracy
without a robot or massive microphone array. In this paper, we
proposed a method of local sound field synthesis with PM method
without large-scale measurements by estimating transfer functions
between the loudspeakers and the matching points from transfer
functions between the loudspeakers and small number of control
points. By sparse optimization, each equivalent source of the
loudspeaker is selected from a dictionary of mono pole sources placed
near each loudspeaker based on small number of transfer functions.
Driving functions of loudspeaker can be derived from transfer
functions between a lot of virtual matching points and all equivalent
sources. To evaluate our proposed method, two-dimensional
simulation experiments are conducted.
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Determination of Optimal Parameters Using Metaheuristics for the Sound Zone Generation by the Least-Squares
Kazuya Yasueda, Daisuke Shinjo, Akitoshi Kataoka
[Abstract]
We propose a method to determine optimal parameters by metaheuristics for sound zone generation using the least-square method. We realize reproduction with sound pressure difference in multiple areas using loudspeaker array. It is realized by controlling the sound field with inverse filter based on transfer functions between each loudspeaker and control point. At that time, it is necessary to determine the number of control points, control point arrangement and regularization parameter. In the conventional method, these parameters are experimentally and empirically determined. Particularly, the control points often are set with regularly form in the target control area. In this paper, we discuss that determine these parameters using by the metaheuristics. The metaheuristics is the solution of mathematical optimization. We studied on applying metaheuristics such as the genetic algorithm and the simulated annealing to determine these parameters. We propose an appropriate evaluation function for sound pressure in each zone. We evaluated performance with computer simulation and performed experiment in actual environment. As the result, we confirmed improvement of performance with relative sound pressure level, spectral distortion and frequency response.
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Single-channel signal Features for Estimating Microphone Utility for Coherent Signal Processing
Michael Günther, Andreas Brendel, Walter Kellermann
[Abstract]
Many microphone array signal processing techniques, e.g.,
for beamforming or localization, rely on coherent input
signals. However, low inter-channel coherence may result
from the occlusion of microphones, reverberation, or the
presence of undesired signal components, so that the
according signals contribute little to the overall algorithmic
performance. Thus, ranking the microphone channels by
their utility for subsequent coherent signal processing
schemes is of considerable interest. Direct estimation of the
pair-wise coherence is often straight-forward in compact
microphone arrays when all microphones share a common
sampling clock, while acoustic sensor networks require a
potentially costly time synchronization of the microphone
signals. In this case, estimating the channel utility ranking
from simpler, per-channel features, e.g., statistical moments
of the time-domain signal waveform or of the corresponding
magnitude spectrum, instead of the signal coherence
facilitates the clustering of useful sensor nodes for a particular
task. This approach further offers a way to determine whether
it is worth the effort to synchronize the sensor signals in a
sensor network, thereby saving computational power and
data rate. Therefore, in this contribution, we investigate the
efficacy of different signal features for the estimation of the
microphone channel utility ranking.
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On the Use of Spherical Microphone Arrays in a Classical Musical Recording Scenario
Johann-Markus Batke
[Abstract]
The usage of spherical microphone arrays in virtual reality production environments have been seen to be steadily increasing. This is understood to be a natural progression since the microphone arrayës output is easily converted to the Higher Order Ambisonics (HOA) sound field representation. However, for the recording of classical music, there is still much debate as to whether the application of spherical microphone arrays makes any sense at all. Depending on the targeted playback format, various established coincident single microphone setups, as well as spaced setups are preferred. Spaced microphone setups are particularly popular since they ensure a certain degree of decorrelation for the resulting loudspeaker signals. On the other side, microphone array recordings have a much higher degree of freedom with regard to spatial post processing. This contribution investigates how advantages of both recording strategies can be combined. The recording of a string quartet is used as a well known recording scenario for classical music. A spaced AB recording serves as the basis. A microphone array in a central position within the quartet another one distant from the quartet are used to enhance the AB signals. Different target formats including stereo, surround as well as 3D audio are discussed.
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A Target Direction Search Algorithm Based on Microphone Array
Xubin Liang, Difeng Sun, Tianqing Zhao, Liangyong Zhang, Houlin Fang, Fang Zhang
[Abstract]
The use of a microphone array to obtain the direction of a far-field
target is an important means of impacting-point detection. In this
paper, a new algorithm is proposed for the pulsed acoustic signal
orientation. By calculating the Euclidean distance between the
position vector projection and the acoustic path difference of each
microphone at different azimuth angles, the angle corresponding to
the minimum distance is the target azimuth. In the case of poor or
damaged signal quality of some microphones, the minimum distance
searched out will be relatively large. By setting the threshold value,
the participating operation can be eliminated, and the accuracy and
stability of the orientation can be improved. In this paper, the
principle of target direction search is introduced, the accuracy and
stability of orientation are simulated and experimentally verified, and
compared with the traditional algorithm, the algorithm is proved to
be not only high precision but also strong. It has important reference
significance for the practical engineering application of passive sound
detection such as impact point measurement.
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Automatic Choice of Microphone Array Processing Methods for Acoustic Testing
Ennes Sarradj, Gert Herold, Simon Jekosch
[Abstract]
If microphone arrays are to be used in acoustic testing, a
signal processing method must be applied to produce
measurement results such as spectra and location of sources
from the data. It is well know that different processing
methods may produce different results, so the practical
question arises which method to choose. We propose some
strategies that allow for a-priori choice of the most suitable
processing method from the raw measured data. One
method uses the eigenvalue spectrum of the measured cross
spectral matrix while another is based on the classical
beamformer output and a neural network. After estimating
the apparent number of sources and the dynamic range, the
best method is looked up based on the statistical analysis of
a large number of synthetic test cases. The procedure is
demonstrated using a practical example.
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Wind noise removal from mixture with speech: Using Wiener filter and invariant frequency beamforming
Fan-Jie Kung
[Abstract]
In this paper, we design a system to help the hearing-impaired
to hear clearly from a speaker in a noisy environment especially
in high winds. The amplitude and frequency of speech are
important for the hearing-impaired. They can hear better under
the speech of high amplitude and low frequency. So how to
design a system for them is a critical issue. First, we assume that
the wind noise mostly occurred in the low frequency so that we
can utilize the Wiener filter to reconstruct the clean
speech. Second, we assume that a listener and a speaker
always face to face to talk, so that we can utilize the
beamforming technique to enhance the speech. For
microphone array beam pattern, the beam width change with
the frequency variance. Therefore, some methods are used like
minimum variance distortionless response (MVDR) and linearly
minimum constrained variance (LMCV). Notwithstanding, these
two methods are computation complexity. Hence, we use the
multi-beamforming method to get the constant beamwidth with
low complexity. The simulation results show that this system
improve the quality of speech in high winds.
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A Maximum-Achievable-Directivity Beamformer with White-Noise-Gain Constraint for Spherical Microphone Arrays
Xi Chen, Gongping Huang, Jingdong Chen, Jacob Benesty
[Abstract]
In microphone array beamforming, it is desirable to achieve a directive gain as high as possible for maximum acoustic noise rejection. The well-known superdirective beamformer was developed for this purpose; but it is sensitive to array imperfections such as sensors’ self noise, sensor placement errors, mismatch among sensor responses, etc, which restrict its application in practical systems. To circumvent this lack of robustness, we recently developed a differential beamforming method that can achieve a flexible compromise between the directivity factor (DF) and the level of white noise gain (WNG) by adjusting the value of a control parameter. This principle is further extended in this work. We present a beamforming method with spherical microphone arrays. It first determines the order of the differential beamformer in an analytical way based on the minimum level of WNG that is tolerable by the array in the frequency band of interest. This order is then applied to design the beamformer to achieve the maximum possible DF. The advantage of this approach over the existing ones is that it is robust to implement and yet can achieve the maximum possible DF.
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Neural Network-based Broadband Beamformer with Less Distortion
Mitsunori Mizumachi
[Abstract]
Beamforming has been one of the important issues in the field of multi-channel signal processing including acoustic signal processing. A wide variety of beamformers have been proposed for each application. In general, acoustic beamforming deals with broadband signals such as speech signals compared to narrowband beamforming for antenna array and radar applications. Recently, neural network-based non-linear beamformers become popular but have a problem that causes an annoying non-linear distortion on the output signal. In the case of speech enhancement, it is a serious problem because our auditory system is highly sensitive to artificial non-linear distortion on speech signals. This paper proposes to solve the problem with the relaxed dual cost functions in the neural network-based beamformer for speech enhancement. The primary cost function aims at sharpening the beam-pattern, and the second cost function is introduced to achieve decreasing speech distortion. Those cost functions are alternatively used for optimizing the beam-pattern in the frequency range of speech signals. The feasibility of the proposed method is confirmed by computer simulation with a small amount of training data, which include sinusoidal signals, random noises, and speech signals.
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Estimating sound intensity from acoustic data captured by parallel phase-shifting interferometry
Fumihiko Imaeda, Risako Tanigawa, Kenji Ishikawa, Kohei Yatabe, Yasuhiro Oikawa
[Abstract]
Visualizing sound fields is important to understand them intuitively. Recently, sound field visualization using a parallel phase-shifting interferometer (PPSI) has been proposed. This optical method can observe a sound field instantaneously and quantitatively without placing any object inside the field. Thus, sound fields difficult for the ordinary instruments to measure can be investigated by PPSI such as the fields inside small cavity or air flow. After measurement, the observed data must be analyzed to obtain meaningful information in terms of acoustics. However, such analysis has not been studied much as PPSI itself is a newly developed method. In this paper, we estimate sound intensity from the data captured by PPSI. The number of the observation points of our PPSI system is up to 262,144, where the interval between the adjacent points is 0.22 mm. Therefore, sound intensity can be estimated densely at quite a lot of points. In addition, noise reduction is considered to aid the estimation. The accuracy of estimated sound intensity is investigated through numerical experiments, and then real data observed by PPSI are analyzed for visualizing the sound intensity.
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Investigation into Transaural System with Beamforming Using a Circular Loudspeaker Array set at Off-center Position from the Listener
Yu Ito, Yoichi Haneda
[Abstract]
A transaural system based on a crosstalk canceller is effective
for virtual acoustic imaging when two loudspeakers are
arranged in front of the listener. However, when the two
loudspeakers are arranged off-center from the listener, the
crosstalk canceling performance declines drastically, especially
the stereo dipole system. For example, when both loudspeakers
are located on the left front of the listener, the sound pressure
level in the left ear is higher than that in the right ear. In such a
situation, it is difficult to cancel sound in the left ear and
reproduce the desired sound in the right ear. In order to
reproduce equal sound pressure levels in both ears, we
introduce a method that forms two directivity beams using a
circular loudspeaker array.
Here, we consider two beams from the array to the listener’s
left and right ear as being equal to left channel and right
channel loudspeaker in the stereo dipole. To confirm the
performance of the proposed method, we compared the
stereo dipole and the proposed method through computer
simulations and subjective evaluation. The proposed method
improved the sound pressure level difference between the left
and right ear.
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Subjective evaluation of Head-Related Transfer Functions reconstructed with Spatial Principal Component Analysis and their domain dependency
Shouichi Takane, Keisuke Sakamoto, Koji Abe, Kanji Watanabe, Masayuki Nishiguchi
[Abstract]
It is well known that data amount for spatial variation of
Head-Related Transfer Functions (HRTFs) can be compressed
by using Principal Component Analysis (PCA) with less
perceptual influence. The PCA of the HRTFs is called Spatial
PCA (SPCA). The author analyzed the effect of domain
selection for SPCA in objective aspect [S. Takane, Appl.
Acoust., 101, 64-77(2016)], but the effect is not analyzed in
subjective aspect. In this paper, the effect was investigated
using non-individual HRTFs with a 3 alternative forced choice
(3AFC) experiment. The domains for the SPCA were HRIR
(Head-Related Impulse Response), HRTF, amplitude of HRTF,
and log-amplitude of HRTF. In the latter two domains,
minimum phase was assumed for generating the phase
components of the HRTFs reconstructed from the SPCA
results. As a result, it was found that number of the
principal components required to reconstruct the HRTFs with
perceptually undetectable difference is small for all domains,
and this corresponds to the results of many researches.
Moreover, the HRTFs reconstructed from the results of the
SPCA in log-amplitude domain achieved the relatively low
number of the principal components, differed from the
results of the objective evaluation done by the authors
previous research.
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Adhoc method to Invert the Reassigned Time-Frequency Representation
Shristi Rajbamshi, Peter Balazs, Nicki Holighaus
[Abstract]
Time-Frequency Representation (TFR) technique has been
extensively used for past many years to analyze non-stationary
signals. There have been several TFR methods, out of which TF
reassignment is one such method which was primarily developed to
improve the readability of existing TFRs, thereby proving itself to be
an efficient tool to analyze signals. Despite this, it has not gained
much popularity because of its non-bilinear nature and lack of
invertibility. Therefore, as a way to address this shortcoming, we
present an ad-hoc approach to invert the reassigned TFR.
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Detection of clean time-frequency bins based on phase derivative of multichannel signals
Atsushi Hiruma, Kohei Yatabe, Yasuhiro Oikawa
[Abstract]
In this paper, a method for evaluating the cleanness of
each
time-frequency bin of multichannel spectrograms is
proposed.
When observing acoustical signals with noise and/or
interference, the degree of noisiness is usually
different for each
bin in the time-frequency domain. Therefore, array
signal
processing techniques should be possible to be
improved by
choosing only ”cleaner” bins, which contain less noise
and/or
interference, for extracting the spatial information. The
proposed method aims to distinguish such clean bins
from noisy
ones. To do so, the similarity of phase derivative
among
channels is considered since phase is sensitive to
noise and
interference. Constant phase is removed by derivative
so that
convolutive mixtures can be handled without care on
spatial
condition. The proposed method is applied to
direction-of-
arrival estimation (MUSIC method) and blind source
separation
(independent vector analysis) for demonstrating the
possibility
of the proposed measure.
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Column-Wise Update Algorithm for Independent Deeply Learned Matrix Analysis
Naoki Makishima, Norihiro Takamune, Daichi Kitamura, Hiroshi Saruwatari, Yu Takahashi, Kazunobu Kondo
[Abstract]
In this paper, we propose a robust demixing filter update algorithm for audio source separation. Audio source separation is a task to recover source signals from multichannel mixtures observed in a microphone array, which can be applied to, e.g., speech recognition and music signal analysis. Recently, independent deeply learned matrix analysis (IDLMA) has been proposed as a state-of-the-art separation method. IDLMA utilizes deep neural network (DNN) inference of source models and blind estimation of demixing filters based on sources’ independence. In conventional IDLMA, iterative projection (IP) is exploited to estimate the demixing filters. Although IP is a fast algorithm, when the specific source model is not accurate owing to the bad SNR condition, the successive update of filters will fail hereafter. This is because IP updates the demixing filters in a source-wise manner where only one source model is used for each update. In this paper, we derive a new microphone-wise update rule which exploits all information of the source models simultaneously for each update. Moreover, we propose a method to select the appropriate source- or microphone-wise update rule depending on the source signal’s pseudo SNR estimated via the DNNs. Experimental results show the efficacy of the proposed method.
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Deep Clustering for Single-Channel Ego-Noise Suppression
Annika Briegleb, Alexander Schmidt, Walter Kellermann
[Abstract]
In the context of audio signal processing for microphone-equipped robots, the robot’s self-created movement noise, so-called ego-noise, is a crucial problem. It massively corrupts the microphone signal and degrades the robot’s capability to interact intuitively with its environment. Therefore, ego-noise suppression is a key processing step in robot audition, which is commonly addressed using learning-based dictionary or template approaches. In this contribution, we introduce a deep-learning framework called Deep Clustering (DC) for ego-noise suppression in a single microphone channel, which was initially introduced by Hershey et al. for the task of speech separation. In DC, a bi-directional recurrent neural network is trained to embed each time-frequency bin of a mixture, containing ego-noise and speech, to a higher dimensional domain under the constraint that embeddings of bins dominated by ego-noise have maximal distance to those dominated by speech. During testing, clustering is performed in the embedding domain to assign each time-frequency bin uniquely to one of the two signal components and thereby allowing the estimation of both. We demonstrate that DC allows a significant reduction of ego-noise in the reconstructed signal. Additionally, we investigate the influence of the embedding size and the amount of training data on the suppression performance.
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A Study on the Data Augment Method considering Room Transfer Functions for Acoustic Scene Classification
Minhan Kim, Seokjin Lee
[Abstract]
Acoustic scene classification (ASC) is a recognition problem
using a machine-based approach of sound caused by a daily
life. People can recognize an event by using a sound
information without seeing the event, so the sound-based
awareness algorithms have been researched recently. Recently,
the deep-learning-based algorithms to solve the ASC problem
have been researched mainly, but the algorithms require
massive number of data to achieve a good performance.
Therefore, the data augment method is important to apply the
deep-learning-based algorithm to the ASC problem. In this
paper, data augmentation method considering the acoustic
propagation in a room is researched, and it is applied to a
convolutive-neural-network-based ASC algorithm. A simulation
was performed with the dataset from DCASE 2018 challenges to
evaluate the proposed data augment algorithm, and the results
show that the proposed algorithm can enhance the state-of-
the-art method.
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Real-Time Audio Processing on a Raspberry Pi using Deep Neural Networks
Fotios Drakopoulos, Deepak Baby, Sarah Verhulst
[Abstract]
Over the past years, deep neural networks (DNNs) have quickly grown into the state-of-the-art technology for various machine learning tasks such as image and speech recognition or natural language processing. However, as DNN-based applications typically require significant amounts of computation, running DNNs on resource-constrained devices still constitutes a challenge, especially for real-time applications such as low-latency audio processing. In this paper, we aimed to perform real-time noise suppression on a low-cost embedded platform with limited resources, using a pre-trained DNN-based speech enhancement model. A portable setup was employed, consisting of a Raspberry Pi 3 Model B+ fitted with a soundcard and headphones. A (basic) low-latency Python framework was developed to accommodate an audio processing algorithm operating in a real-time environment. Various layouts and trainable parameters of the DNN-based model as well as different processing time intervals (from 64 up to 8 ms) were tested and compared using objective metrics (e.g. PESQ, segSNR) to achieve the best possible trade-off between noise suppression performance and audio latency. We show that 6-layer DNNs with up to 200.000 trainable parameters can successfully be implemented on the Raspberry Pi 3 Model B+ and yield latencies below 16-ms for real-time audio applications.
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Underwater Acoustic Recognition System for Detection of Low-altitude Moving Source
Tianqing Zhao, Xubin Liang, Difeng Sun, Fang Zhang, Deyu Sun
[Abstract]
Air targets detection underwater is an important part of
marine safety management. For the difference of physical
properties between water and air, there are many difficulties
for air sound detection launched underwater. In this paper,
we present a recognition system using both time and
frequency domain characteristics for low-altitude moving
source detection. In the detection phase, a short-to-long-
term energy ratio (SLR) is first obtained by analyzing the
coupling acoustic data. This initial SLR indicates the possibility
of target appearing. For signals exceeding SLR threshold, the
cepstral coefficients are extracted by a 4-Gammatone filter
with cochlea membrane nature. The validity of the
algorithm is proved by a test adopting BP network with both
synthetic and real data. The results indicating that the joint
detection algorithm can detect low-altitude moving targets
such as helicopters, and has a good anti-noise
performance.
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A Study on Separation Method Combined Gamma-Process Non-negative Matrix Factorization and Deep Learning.
Jomae Satoru, Kenko Ota, Hideaki Yoshino
[Abstract]
Accurate analysis of fundamental frequency and chord constitutive notes is a hard problem. However, to solve this problem is important for similar music retrieval and music arrangement etc. Development of an accurate method for sound source separation is required to analyze the fundamental frequency etc. accurately. In this research, we propose a method of sound source separation that combines gamma-process non-negative matrix factorization (GaP-NMF) and Deep Neural Network (DNN).In the proposed method, we first estimate the basis with GaP-NMF. Then, DNN classifies the estimated basis according to musical instruments.The basis estimated by GaP-NMF is emphasized by multiplying with the spectrum template of musical instruments which is specified by DNN. We conducted a sound source separation experiment to verify the performance of the proposed method. Sound sources are composed of multiple musical instrument sounds. As a result of separating a single musical instrument sound from the sound sources, we confirmed that the proposed method improved the SNR by 1.4 dB over the conventional method depending on the data.
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Detection of Boat Noise by a Convolutional Neural Network for a Boat Information System
Haruki Yamaguchi, Kenji Muto
[Abstract]
Some of boat noises cause noisiness and annoyance to a
resident near a canal. We proposed an information system to
notice about coming noisy boat by a cellphone using an
audiovisual effect. However, the system had a problem to
detect of boat coming at night by camera. Then we
investigated the train data to detect a boat noise in
environmental sound by a convolutional neural network
(CNN). In the detection of boat noise, we used the train data
using the spectrogram of the environmental sound. We
investigated the spectrogram configuration for improvement
of the boat noise detection. As a result, when the
spectrogram configuration was the time axis of 5 seconds
and the frequency axis between 10 and 750 Hz, the
detection performance became the highest accuracy of
over 95 %.
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Effective Method for Screening Discharged Battery Using Support Vector Machine and High- Resolution Acoustic Analysis
Tomoaki Magome, Kan Okubo
[Abstract]
Alkaline dry batteries and nickel - metal hydride (NiMH) rechargeable batteries are
used worldwide for various portable devices that require continuous current.
Nevertheless, visual verifying such a battery as discharged or not remains as difficult
checking a watermelon for ripeness. Although one can detect a dead battery
using a battery indicator, such a useful tool is not always available. Therefore, a
simple and intuitive means of ascertaining whether a battery is dead or not must be
found to avoid problems such as battery leakage. In our previous work, we
proposed an acoustic analysis based method for estimating the discharge state of
an alkaline dry battery: a hammering test method can screen dead batteries by
analyzing the tone color of the tapping sound. In this report, we propose a more
effective method and apply it to NiMH rechargeable batteries. To improve the
decision accuracy, we also employ a support vector machine (SVM) and super
high-resolution recording system, which can obtain sound up to 100 kHz. Our
experimentally obtained results suggest that the proposed method can provide
effective screening.
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Gated convolutional neural network-based voice activity detection under high-level noise environments
Li Li, Kouei Yamaoka, Yuki Koshino, Mitsuo Matsumoto, Shoji Makino
[Abstract]
This paper deals with voice activity detection (VAD) tasks
under high-level noise environments where signal-to-noise
ratios (SNRs) are lower than -5 dB. With the increasing needs
for hands-free applications, it is unavoidable to face
critically low SNR situations where the noise can be internal
self-created ego noise or external noise occurring in the
environment, e.g., rescue robots in a disaster or navigation
in a high-speed moving car. To achieve accurate VAD
results under such situations, this paper proposes a gated
convolutional neural network-based approach that is able
to capture long- and short-term dependencies in time series
as cues for detection. Experimental evaluations using high-
level ego-noise of a hose-shaped rescue robot revealed
that the proposed method was able to averagely achieve
about 86% VAD accuracy in environments with SNR in the
range of -30 dB to -5 dB.
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Acoustic Remote Sensing for Irrigation Systems Control in Agriculture
Anna Radionova, Chandra Ghimire, Laura Grundy, Seth Laurenson, Stuart Bradley, Valerie Snow
[Abstract]
The paper addresses the problem of measuring free water
on the
surface of agricultural soils by an accurate real-time
acoustic
method. The generation of free water is the result of a fine
balance between the irrigation rate and the rate at which
the
soil can transport water away from the surface and is the
primary
cause of inefficient and environmentally-harmful losses
during an
irrigation event. The innovative component of the project
is
vested in the development of directional acoustic arrays
and
sophisticated signal processing which can remotely detect
the
onset of free water in the soil via changes in reflectivity.
The
proposed method estimates the amount of free water on
the soil
surface based on the changes in the amplitude of the
reflected
sound waves of the soil surface at different moisture levels.
Our
results show that sound wave reflectivity depends on the
proportion of the soil surface covered by water. The
presented
results are based on both laboratory and field
measurements, and
therefore form the basis of an inexpensive and accurate
free
water sensor for irrigation systems.
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Sound Capture from Rolling-shuttered Visual Camera Based on Edge Detection
Koichi Terano, Hiroki Shindo, Kenta Iwai, Takahiro Fukumori, Takanobu Nishiura
[Abstract]
Recently, several new studies for recording sound have been
conducted on extracting sounds from images of an object
surface vibrated by sound waves. This method can capture the
sound with a high-speed camera instead of the air conduction
microphone. However, it is unpractical due to the high cost of
the high-speed camera. In this paper, we propose a method to
capture sound from the image of the object surface vibrated
by sound waves with a rolling-shuttered visual camera. This
camera uses a CMOS image sensor and writes in order from the
top row of the image. Therefore, when a moving object is
photographed, a rolling-shutter distortion in which the object is
distorted occurs due to the different writing time for each row.
The proposed method uses the edge detection to emphasize
the edge of the rolling-shutter distortion caused by
photographing the vibrating object, and the method resamples
the edge as the amplitude of the sound wave. As a result of the
sound capturing experiment, we confirmed that the proposed
method can capture pure tone from the image by using the
CMOS image sensor.
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Designing Nearly Tight Window for Improving Time-Frequency Masking
Tsubasa Kusano, Yoshiki Masuyama, Kohei Yatabe, Yasuhiro Oikawa
[Abstract]
Many audio signal processing methods are formulated in
the time-frequency (T-F) domain which is obtained by the
short-time Fourier transform (STFT). The properties of the
STFT are fully characterized by window function, number
of frequency channels, and time-shift. Thus, designing a
better window is important for improving the performance
of the processing especially when a less redundant T-F
representation is desirable. While many window functions
have been proposed in the literature, they are designed
to have a good frequency response for analysis, which
may not perform well in terms of signal processing. The
window design must take the effect of the reconstruction
(from the T-F domain into the time domain) into account
for improving the performance. In this paper, an
optimization-based design method of a nearly tight
window is proposed to obtain a window performing well
for the T-F domain signal processing.
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Noise-reducing Sound Capture Based on Exposure-time of Still Camera
Hiroki Shindo, Koichi Terano, Kenta Iwai, Takahiro Fukumori, Takanobu Nishiura
[Abstract]
A visual microphone has been proposed to capture the distant sound. This microphone captures the sound from high frame-rate video by using the pixel difference. It is expected to be applied in surveillance cameras because it is robust to be affected by the distance attenuation. However, this is an unpractical capturing method due to expensive-equipment requirement. In this paper, we aim to realize inexpensive visual microphones focused on the CMOS image sensor with rolling-shutter distortion. An image shot with this sensor has time information because the sensor writes image to each line of the element in sequence. Therefore, it is possible to capture sound from an image without the high frame-rate video if acoustic signals can be extracted from the time information included in the image. However, lower frequency noise is mixed in the captured sound using the sensor due to the inclination and distortion of the object to be photographed. This noise depends on the exposure-time of the sensor. We thus propose a method of suppressing the noise by designing a digital filter based on exposure-time. The experiment results show that the proposed method can suppress the noise compared with the original captured sound.
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Time-Variant Acoustic Front-End Measurements of Active Noise Cancellation Headphones
Johannes Fabry, David Hilkert, Stefan Liebich, Peter Jax
[Abstract]
A robust design of active noise cancellation (ANC) headphones by means of digital signal processing requires a deep understanding of the underlying acoustic front-end. Particularly the primary path, which is defined as the transfer path between the outer and inner microphone of the headphone, and the secondary path, which is the transfer path between the loudspeaker and the inner microphone, are of interest. These paths may vary due to e.g. fitting of the headphone, direction of arrival or the physique of the user. In this contribution, we present the results of a measurement series with the objective to examine acoustic paths for their intra-person variances with influences such as jaw movement, head rotation and refitting of the headphone, as well as inter-person variance between different subjects. The measurements were conducted in an anechoic chamber with 25 participants from age 21 to 61. Furthermore, the implications on the performance for time-invariant feed-forward ANC solutions will be considered and the benefit of calibrating the secondary path investigated.
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DFT-Filterbanks with Spectral Refinement and its Comparison with Polyphase Filterbanks
Mohammed Krini
[Abstract]
The most popular uniform analysis scheme applied for speech
enhancement periodically performs DFTs of overlapping and
windowed signal segments. However, due to the windowing of
successive signal segments, often a significant frequency
overlap arises among neighboring subbands. These overlapping
effects are undesirable as they limit the performance of
adaptive filters and the feature estimation in the subband
domain. In order to reduce this overlap without increasing the
DFT order, the so-called spectral refinement (SR) can be utilized.
The SR is based on a linear combination of weighted and shifted
speech segments and can be applied as a post-processing
stage after a DFT-based analysis filterbank. In this contribution
the SR is used as a predecessor to a DFT. It can be shown that
the resulting SR structure in the time-domain for DFT-based
analysis flterbanks looks similar to polyphase filterbanks. For
enhanced frequency selectivity of the analysis, preceding
weighted blocks need to be added before performing the DFT.
A window function of higher order has to be defined that covers
the current as well as previous input segments. In case of SR, a
set of shifted low-order window functions are linearly combined
and transformed into a desired window function of higher order.
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Optimal Design of Symmetric and Asymmetric Beampatterns with Circular Microphone Arrays
Xudong Zhao, Gongping Huang, Jacob Benesty, Jingdong Chen
[Abstract]
This paper is devoted to the study of the beamforming
problem with circular microphone arrays (CMAs) and
presents an approach to the design of beamformers with
asymmetric and symmetric frequency-invariant
beampatterns. We first discuss how to express a desired
target directivity pattern, either symmetric or asymmetric,
into a linear weighted combination of sine and cosine
functions as well as circular harmonics of different orders.
Using the hypercardioid pattern as an example, we show
how to determine the weighting coefficients by
maximizing the directivity factor (DF) with different
constraints. Then, by using the Jacobi-Anger expansion, an
approximation of the beamformer’s beampattern is
presented. A linear system is subsequently formed by
forcing the approximated beampattern to be equal to a
target asymmetric or symmetric directivity pattern. The
optimal beamforming filter is finally determined by
identifying the linear system. Simulations demonstrate the
properties of the proposed beamforming approach.
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New Aspects in Birdsong Recognition utilizing the Gabor Transform
Sven Heuer, Pavel Tafo, Hajo Holzmann, Stephan Dahlke
[Abstract]
In this paper we will be concerned with mathematical methods for birdsong recognition and classification. Current approaches compute the spectrogram of an audio recording using the Gabor transform, which is then used as input for a convolutional neural network (CNN) to classify the recording. While recent work is dedicated to finding the best hyperparameters for training the CNN and data augmentation, the parameters for the Gabor transform receive less attention. We aim to close this gap by evaluating the effect of different window functions and window lengths on the overall classification accuracy.
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Applying Convolutional Neural Networks to the Analysis of Mouse Ultrasonic Vocalizations
Reyhaneh Abbasi, Peter Balazs, Anton Noll, Doris Nicolakis, Maria Adelaide Marconi, Sarah M. Zala, Dustin J. Penn
[Abstract]
Many species in diverse taxonomic groups, including
rodents, bats, and insects, communicate with complex
ultrasonic vocalizations (USVs) (>20 kHz). Two main
components of processing and analyzing USV recordings
include detection and classification of syllable types.
Recently we developed an efficient algorithm for
detecting mouse USVs (Automatic Mouse Ultrasound
Detector (A-MUD)). The main challenge is detecting USVs
under conditions with a low signal-to-noise ratio, which
results in high rates of false positives (FP). Mice produce
many short USVs (< 10 ms), which especially inflate FPs. We
aimed to improve the detection of mouse USVs with A-
MUD by classifying vocalizations into three discrete syllable
types (with 0, 1, or ≥2 frequency-jumps) or FPs. Supervised
Convolutional Neural Networks (CNNs) were fed by 2D
Gammatone Filtered Spectrograms (GFSs) adapted to the
frequency range of mice. Evaluation of performance
shows that CNNs yielded an overall accuracy of 95±1.2%
and macro-F1 score of 90±2.7%. In contrast, Multilayer
feed-forward neural networks provided an overall
accuracy of only 85.4±1.9% and macro-F1 score of
75.4±2.9%,
which indicates that CNNs outperformed this conventional
classification method.
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Acoustic Performance of Soundproof Ventilation Units installed in Dwelling Walls
Sohei Nishimura, Yuya Nishimura, Thulan Nguyen
[Abstract]
Large number of houses in developing Asia countries are being
affected by road traffic noise emission from motorcycles which
create frequent klaxon sounds. On the other hand, the use of
ventilation holes for providing healthy indoor air is commonly
found in those countries. However, these ventilation holes also
provide direct paths for traffic noise to enter a residence. This
paper describes the design of the soundproof ventilation unit for
installation inside the wall of a house. For this purpose, the
rectangular cube or elliptic cylinder form is reasonable as the
unit shape. The sound pressure propagating in the unit has a
standing wave and higher-order mode wave components. The
latter occurs in the high-frequency range. The Noise tends to
increase in this frequency range due to higher order mode
resonance. In this paper, at first, the theoretical calculations of
the sound pressure inside the rectangular cube and elliptic
cylinder are carried out by solving the wave equation. Next, the
resonance frequencies of the higher-order mode wave
component are calculated for rectangular cube and elliptic
cylinder of the same volume. Some comparisons between the
experimental results and the estimated ones by the proposed
method are discussed.
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Acoustic regulations and design of the multipurpose hall and exhibition halls of the new Munch museum in Oslo
Jannicke Olshausen
[Abstract]
The presentation will deal with room acoustical challenges that comes up with a multipurpose hall and high ceiling exhibition halls for monumental works by Edvard Munch. The new museum for the Norwegian painter Edvard Munch drawn by estudio Herreros architects will open next year in 2020 if everything goes by the plan. The building is located close to the Oslo opera house (opened 2008). The capacity of the multipurpose hall is about 277 to 380 seats, depending on the seating arrangement. The Odeon models used for the acoustic design of the halls will be presented and discussed. The calculation results and the geometry of the multipurpose hall will be presented in context with the Norwegian standard NS 8178 ”Acoustic criteria for rooms and spaces for music rehearsal and performance”. The calculation results of the exhibition halls will be put up against the Norwegian acoustic regulations (NS 8175) to reverberations times in museums. The requirements are difficult to fulfill when wall absorbers are not desirable because they are conflicting with the paintings. Acoustic capacities are calculated for two different exhibition halls with the given reverberation times. Compensating actions are presented for exceeding the reverberation time requirements.
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Evolution of the Italian regulations on acoustic classification of buildings
Patrizio Fausti, Antonino Di Bella, Andrea Santoni, Fabio Scamoni, Simone Secchi
[Abstract]
In Italy there are two national standards which deal with the
procedure of acoustic classification of buildings. The UNI
11367:2010 gives a detailed description of the classification
scheme, including sampling procedures and calculation of
uncertainty and is useful for buildings with a large number of
homogenous elements. A different procedure is given in the
standard UNI 11444:2012, involving the selection of the
”worst case” in a sample of measurable building elements in
order to identify the building units to be completely
measured. In this case, the calculation of the uncertainty is
not required. Both standards are based on voluntary
application, except for the specific case of public buildings,
for which there are some enforced requirements. In fact, in
2017, a new legislation introduced, for public buildings, the
minimum requirement of class II of 11367. The law also refers
to UNI 11532 for the requirements of indoor environments
(reverberation time and speech transmission index). This
paper describes the classification procedures and reports
the possible impact of the aforementioned law on the
debate concerning new legislation for all buildings.
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Speech privacy as a harmonizing factor in rating the sound insulation between dwellings
Miomir Mijić, Dragana Šumarac Pavlović, Miloš Bjelić, Tatjana Miljković
[Abstract]
Recent consulting practice has shown an increase in the
number of complaints concerned with insufficient sound
isolation between dwellings. In some of them the sound
reduction index value was above the value proposed in
legislation, but the neighbours’ conversation was heard too
clear. Comprehensive analysis has shown that in new buildings
there is an extremely low ambient noise level, often below 20
dBA. One can guess that energy efficiency introduced
extremely good windows on the facades, and in the same time
home appliances and equipment have become too quiet.
Such environment enable clear recognition of the neighbours’
speech even with enhanced sound insulation. The conclusion is
that in evaluation of sound insulation between dwellings it is
suitable to include somehow, besides R or D, the ambient noise
level as an embedded factor. A parameter called Speech
privacy class as a measure of speech privacy had been
introduced in the literature. It is defined as the sum of frequency
restricted values of D and ambient noise level. The paper
discusses the possibility of introducing such information
regarding privacy in assessment of the sound isolation between
dwellings. Such approach promises to provide harmonisation of
sound insulation assessment and the subjective experience.
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Derivation of Frequency Dependent Time-Domain Boundary Conditions Based on In-Situ Surface Measurements and Model Fitting
Baltazar Briere De La Hosseraye, Huiqing Wang, Fotis Georgiou, Maarten Hornikx, Philip W. Robinson
[Abstract]
In order to achieve accurate time-domain wave-based simulations of a real room, the acoustical properties of the locally reacting materials within the room have to be implemented as time-domain impedance boundary conditions (TDIBC), in such a way that the behavior of the materials in well-simulated within the wave-based solver. This paper presents the implementation of such TDIBCs of two materials: a porous absorber and an acoustic carpet. Firstly, the material properties were measured both in the impedance tube and in-situ with a pressure-velocity sensor. Advantages and drawbacks experienced with both methods in this context will be presented. Next, the measurement results were fitted to broadband impedance models to extend impedance data in the lower frequency range (20 Hz - 300 Hz), resulting in broadband range impedance data (20 Hz - 4000 Hz). Finally the TDIBCs, in the form of complex reflection coefficients, were fitted as discrete sums of rational functions.
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Sound Absorption of Brazilian wooden panels and their use as Building Components
Rodrigo Scoczynski Ribeiro, Márcio Henrique Avelar Gomes, Rosemara Santos Deniz Amarilla, Fernando Jun Hattori Terashima, Luis Henrique Santana, Rodrigo Catai, Adalberto Matoski
[Abstract]
In this study, the normal incidence sound absorption coefficients
were obtained from three types of wooden panels with different
thicknesses: Oriented Strand Boards (OSB); Medium Density
Fiberboards (MDF) and Cement-bonded wood particleboards
(CBWP). The panels were manufactured by industries of the
sector, using Brazilian woods and reaching lower levels of CO2
emission in its Life-cycle Assessment when compared to the
Brazilian conventional building techniques for walls. The panels
are mostly used in the furniture market, but the results show that
these materials can be used as an option for internal walls
instead of the traditional masonry ones when the acoustic
comfort is a goal. They can also be part of a room as acoustics
panels, a ceiling or floor, leading to the enhancement of the
sound absorption.
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Sound absorption provided by an impervious membrane/cavity/activated carbon arrangement
Veronica Marin, Jorge P. Arenas
[Abstract]
A tensioned impervious membrane at the entrance of an air
cavity which is backed by a reflecting end provides high
and narrow sound absorption peaks at certain frequencies.
In the absence of flexural bending, the sound absorption is
mainly due to the vibration energy dissipation in the
membrane. The frequencies at which maxima of sound
absorption occur are dependent on the membrane
resonances and the geometry of the cavity. Inclusion of a
porous material inside the air cavity broadens the sound
absorption curve at low frequencies making the absorber
more efficient. It has been shown that, due to adsorbing
and desorbing physical mechanisms, granulated activated
carbon exhibits unusually very good low-frequency sound
absorption properties. This article presents some
experimental and theoretical results of a circular tensioned
membrane clamped in a cylindrical cavity that is closed
with a reflecting backing. The effects on the normal-
incidence sound absorption coefficient of adding a porous
layer of activated carbon inside the cavity are reported. The
use of activated carbon layers with different thicknesses
showed better performance in the low frequency range
than traditional fiberglass.
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Acoustic properties of façade fragments of historical monuments
Lukas Vargic, Jana Gregorova, Monika Rychtarikova
[Abstract]
Retrofitting of historical monuments in terms of their
improvement in energy performance and overall
sustainability is one of the frequently discussed questions
in building sector and architecture. However, the
renovation of monuments is highly interdisciplinary task
and therefore, the global assessment of proposed solutions
is needed.
In this article we discuss the impact of additional thermal
insulation in historical buildings while keeping the
reversibility of the action, on its acoustic performance.
Typical cases of energetic retrofit in Slovakia are shown
and sound transmission properties are calculated. The
solutions are discussed in term of its thermo - acoustic
performance.
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Analysis of society response to the value of sound insulation in dwellings
Monika Rychtarikova, Andrea Vargová, Vojtech Chmelík, Daniel Urbán
[Abstract]
When speaking about the acoustic comfort in dwellings, the sufficient sound insulation of building structures from internal and external noise is without any doubt an important issue. A lot of research is performed, that concerns the improvement of sound insulation.
However, when it comes to the global real estate value of dwellings, acoustic comfort in Slovakia is in the audit of buildings highly underestimated. This paper presents the methodology used in development of a questionnaire that helps us to get an important response from society. Presented results are based on the analysis of on-line questionnaires performed in Slovakia that investigates the sensitivity of dwellers to noise and their satisfaction about the acoustic comfort at their homes and awareness about noise as such. Discussion is led on a fraction of the budget (reserved to buy an apartment) would people potentially consider on spend on improvement of sound insulation.
Questionnaire with 74 questions have been optimized and finally over 100 respondents have indicated answers to the questions.
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Experimental study on sound absorption characteristics of granular material: Influence of lateral constraints of casing
Tsuruha Takumasa, Yoshinari Yamada, Makoto Otani, Yasushi Takano
[Abstract]
The effect of friction between the fine granular material and
the partition wall on the sound absorbing characteristics was
investigated. The normal incident sound absorption
coefficient of the hollow glass beads was measured using an
acoustic tube. As the thickness increased, the frequency of
absorption peak tended to converge exponentially. As the
diameter of an acoustic tube expanded at same thickness,
the frequency of absorption peak shifted to the lower
frequency side. The normal incident sound absorption
coefficient is calculated using the Biot model. The density
and elastic modulus of hollow glass beads in the Biot model
are modified by considering the friction between hollow
glass beads and the tube wall. As a result, the validity of
calculation is shown because the calculation is good
agreement with the experiment. Subsequently, the random
incident sound absorption coefficient of hollow glass beads
was measured by the reverberation chamber method. When
the hollow glass beads was divided by partitions laterally at
certain intervals, the random incident sound absorption
coefficient changed according to the interval. It is shown
that the sound absorbing characteristics of fine granular
materials are controlled by changing the intervals of
partitions.
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Cellular automata modeling of propagation and absorption of acoustic waves in impedance tube
Meng Wang, Bo Zhang, Qiqi Chen, Liheng Wang, Yutian Bai
[Abstract]
A one-dimensional cellular automata (CA) model for acoustic
wave’s propagation and absorption in impedance tube with
and without porous metals is studied to explore acoustic
absorbing properties of porous metal in multiple physical fields.
In cellular space, wave motion equation for small amplitude
travelling waves at room temperatures and normal sound
pressure levels is reconstructed by using finite differential
method; accordingly, local evolutionary rules of each cell are
deduced out. Provided that a specific sound source is definitely
given, the sound pressure distribution in impedance tube can
be obtained by solving this CA model. Based on above CA
theory and classical acoustic models of porous materials, the
evolutionary rules of cell for the case that porous material is
placed on the end of impedance tube are formulated. And
sound absorption coefficient of porous materials may be
calculated as well. Finally, the effects of temperature and sound
pressure level on nonlinear propagation properties of acoustic
travelling waves with finite-amplitude are analyzed by rewriting
Westervelt wave motion equation in the form of finite
differential. In this way, one may predict sound absorption of
porous material at different temperatures and sound pressure
levels. The results from CA model agree well with experimental
data.
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Acoucou Platform to Acquire Proffesional Skills and Knowledge in the Field of Acoustics
Karolina Maria Jaruszewska, Filip Baranski, Magdalena Piotrowska, Manuel Melon, Olivier Dazel, Michael Vorländer, Lukas Aspöck, Marko Horvat, Kristian Jambrošić, Monika Rychtarikova, Léopold Kritly, Andreas Herweg
[Abstract]
Today, in a digital era, education must meet the needs and
adapt to modern forms of knowledge transfer.
The Acoustic Courseware Platform http://acoucou.org/ is a
space for new, innovative and multidisciplinary approaches to
teaching and e-learning acoustics (e.g. gamification, VR,
web-based solutions, simulator). It is a place where
professionals, educators, students and other people
interested in acoustics could find educational materials of
different types concerning various branches of acoustics. Specification of the platform allows to go beyond the
framework of a typical MOOC online course. It is a unique
combination of e-learning and hands-on simulation. Utilizing
interactive examples and problem-solving tasks, courses
explain the role of acoustics in engineering and industry. The
language of visualizations is comprehensible, universal, and
easy to share with a multilingual team. Platform is designed
to serve as modern self-development tool for engineers as
well as comprehensive solution for professional education in
work environment. ACOUCOU is created as a result of international partnerships
between several expert teams in the field of acoustics,
industry experts, digital designers and researchers.
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Construction noise management and control policy in Hong Kong and innovative quieter methods for renovation work
Joe Siu Cheong Mok, Cheung-Lam Wong, Chi-Wing Law, C.K. Lee
[Abstract]
Many concerns revolve around construction noise particularly in a densely populated metropolis as Hong Kong. The statutory permit system for construction work during restricted hours and labeling system for powered mechanical equipment (PME) have been well established in Hong Kong to manage and control construction noise. However, renovation work within high-rise residential buildings remains one of the most challenging noise sources as the renovation cycle is relatively short and it usually involves noisy PME. Common noisy activities in Hong Kong, such as demolition of concrete partitions, reformation of flooring and various types of fitting-out work, produce intrusive noise disturbing quite a number of neighboring flats through both air-borne and structure-borne transmissions. Structure-borne noise is difficult to be mitigated without substantial treatment in the noise transmission path which is almost impossible in a dense residential setting. Owing to rising public aspiration for a quieter living environment, it is necessary to address the concerns associated with renovation work within multi-flatted residential buildings. This paper will give an account of the statutory and administrative noise control framework in combating noisy construction and explore the effectiveness of innovative quieter construction methods to further mitigate renovation noise at source to achieve a quieter living environment.
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Influence evaluation of infrasonic/audible noise environment by using both of biological information and infrasound sensors in the vicinity of wind turbine facilities
Megumi Nagamatsu, Masa-Yuki Yamamoto
[Abstract]
In recent years, wind power generation attracts attention. However, problem of low frequency sound generated by wind turbines and its influence on human beings have been gradually taken up in society. We investigated the influence of ultra-low frequency sound generated from wind turbines on human subjects/testers from both aspects of measurement of audible/infrasonic sound and multiple biometric information sensing systems. In the experiment outside, portable electroencephalographs and pulse wave sensors are used because only such small equipment can be used to perform biological information sensing outdoor, in the vicinity of a wind turbine facility. In order to calculate relax trends from the obtained biological information, a method based on previous analyses for obtaining relaxation degrees R is used. It is coming from the fact that content rate of each frequency band included in the electroencephalogram waveform varies depending on the human mental state in the brain wave. Also, in analyzing the heart beat wave, we used an analytical method to find stress degrees S by using the balance of autonomic nerve that can be calculated from the heart beat signal fluctuations. Here we would like to introduce the result using the above analyses methods for almost 30 examples.
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Investigation of the unpleasantness of infrasound combined with audio sound using psychoacoustic scaling methods
Elisa Burke, Euginia Stederi, Stefan Uppenkamp, Christian Koch
[Abstract]
At many immission sites, human exposure to infrasound (f < 20 Hz) is usually accompanied by sound in the audio-frequency range (audio sound, 20 Hz < f < 20 kHz). This gives rise to the question of whether the interaction between infrasound and audio sound affects the quality of auditory perception. Psychoacoustic experiments were performed within the framework of the European project EMPIR 15HLT03 Ears 2. Recent results in this project had already shown that detection thresholds for infrasound were increased when simultaneous audio sound is present. The current study deals with the hypothesis that the unpleasantness related to infrasound is changed when infrasound is presented along with audio sound. A rating task on a numerical scale and a pairwise comparison task were conducted to quantify and to compare the unpleasantness of (1) isolated infrasound (sinusoid), (2) isolated audio sound (sinusoid and broadband), and (3) the combination of both, at different sound pressure levels. Normal hearing listeners aged 18 to 30 years participated in the hearing tests. The results should be of use to improve the understanding of the impact of combined noise on humans and their well-being in the vicinity of potential noise sources.
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A study on the influence of noise and vibration on the living environment along the Hokuriku Shinkansen railway
Takashi Morihara, Shigenori Yokoshima, Yasunao Matsumoto
[Abstract]
One year after opening of the Hokuriku Shinkansen (high-speed) railway, we conducted a social survey targeting detached houses along that rail in 2016. Noise and vibration exposure level were estimated at outdoor points closest to the noise source side of the house. Of the 1,980 people contacted, there were 1,022 valid respondents. The purpose of this research is to investigate the relationship between noise and vibration exposure and social responses. Regarding the noise annoyance and some living activity disturbances, it was shown that the responses of residents living in areas without the conventional railway are higher than those of existing areas. The same trend was observed for the reactions in areas with high vibration levels. This tendency was remarkable especially in areas with high vibration exposure caused by Shinkansen railway. High vibration annoyance was shown in areas without the conventional railway, while there was no difference by the degree of the noise exposure.
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Influence of nocturnal noise on non-restorative sleep: Gender effects
Daniel Fong, Sha Li, Janet Wong, Bradley McPherson, Esther Lau, Lixi Huang, Mary Ip
[Abstract]
Background: Nocturnal noise exposure may affect the sleep of
men and women differently. However, evidence remains
limited, especially for non-restorative sleep (NRS). Therefore, this
study aimed to assess the moderating effect of gender on the
effect of nocturnal noise exposure on NRS. Methods: This was a household survey of Chinese adults.
Participants completed the Chinese Non-restorative Sleep Scale
(NRSS), standardized on a 0-100 scale, with a higher score
indicating less NRS. Nocturnal noise level was measured by a
dosimeter for one week. Results: We recruited 90 subjects (58% female) with average
age 37.0 years (range: 18-65). The mean nocturnal noise level
was 55.5dBA (SD: 13dBA). The mean NRSS domain scores were
57.3 (SD: 17.6) for Refreshment from sleep, 67.0 (SD: 16.7) for
Physical/Mental symptoms, 60.5 (SD: 15.4) for Daytime function,
and 62.6 (SD: 21.6) for Affective symptoms. The gender-by-
nocturnal noise exposure interaction was significant (p=0.043)
for the Refreshment from sleep domain. The influence of
nocturnal noise exposure on Refreshment from sleep was more
profound in women (-1.00, 95% CI: -1.87 to -0.13) than in men
(0.22, 95% CI: -0.57 to 1.01). Conclusions: Women may be more vulnerable than men to the
adverse impact of nocturnal noise exposure on NRS.
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Finger pulse wave amplitude response during sleep to environmental noise
Branko Zajamsek, Gorica Micic, Kristy Lee Hansen, Peter Catcheside
[Abstract]
Changes in finger pulse wave amplitude (PWA) as measured by photoplethysmography might be a sensitive marker of sleep fragmentation caused by an environmental noise. This marker has been shown to be a sensitive marker of autonomic arousals with great clinical potential due to its simplicity and ease of measurement. In this study we investigated the finger PWA in response to several types of environmental noise up to 48 dBA sound pressure level. The study was done on 20 young participants over two consecutive noise exposure nights. We found no significant differences between sound pressure level and finger PWA size with respect to the control for any noise type. These results thus suggest that environmental noise at SPL of 48 dBA or lower does not evoke autonomic system arousals above the spontaneous rate and is hence unlikely for that noise to have any long-term effects on cardiovascular health.
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Studying Individual Noise Disturbance Using Long Term Ear-EEG (Electroencephalography) Recordings In Everyday Life
Martin Georg Bleichner
[Abstract]
Understanding how people are affected by noise on an individual level can improve our understanding of the non-auditory effects of noise on public health. In my research, I work towards a perception based noise dosimetry using mobile ear centered electroencephalography (EEG). We have shown in the past that we can study auditory perception and auditory attention using wireless ear-EEG. For this, we combine small, lightweight, wireless EEG amplifiers, with ear-EEG electrodes and use a smartphone for signal acquisition and signal processing. This setup allows us to do long term EEG recordings beyond the lab, either at work, at school or at home. Over extended periods of time, we can monitor the perception of sounds/noise on an individual level at those locations the person is exposed to noise. We then relate the EEG to the environmental sounds and to the subjective level of noise disturbance/annoyance. In a step by step procedure, we will move from well-controlled auditory perception experiments in the lab to the study of environmental noise in everyday situations. This work will advance the field of mobile ear-centered EEG for long term EEG recordings in everyday life and will provide new insights on dealing with individual noise exposure.
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Examination of the Causal Relationship between Aircraft Noise Exposure, Noise Annoyance and Diagnoses of Depression Using Structural Equation Modelling
Sarah Leona Benz, Dirk Schreckenberg
[Abstract]
Environmental noise is ubiquitous in daily life and a massive environmental burden, though the effects of transportation noise on mental illnesses are sparsely investigated to date. The recently published WHO review on the impact of transportation noise on quality of life and mental health shows inconsistent findings.
Noise annoyance, as an approved psychological stress response to noise, is identified as one of the health impacts of environmental noise. There is some evidence that the impact of noise on health is at least partly mediated by noise annoyance.
A reanalysis of data is conducted using data from the socio-acoustical longitudinal survey NORAH (Noise-Related Annoyance, Cognition, and Health) that has been performed mainly at Frankfurt Airport (and three other airports) in Germany from 2011 to 2013. Residents older than 18 and living in the 40 dB-noise contour around Frankfurt Airport were randomly selected to participate in the study. Telephone interviews assessing diagnosed depressions and aircraft noise annoyance among others were conducted annually from 2011 to 2013. Aircraft sound levels were calculated for the exact address of every participant.
The present work is an approach to examine the causal relationship between noise exposure, noise annoyance and diagnoses of depression using structural equation modelling.
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Do ultrafine particles confound studies on noise and cardiovascular disease?
Anna Hansell, Anja Tremper, Jamie Soussan, Paolo Vineis, Gary Fuller, John Gulliver
[Abstract]
Ultrafine particles (UFPs) are emitted by both road traffic and jet
engines and may potentially confound associations with noise.
For example, it was suggested that UFPs from aircraft could be a
possible explanation (Corbin, BMJ 2013) for the higher rates of
cardiovascular hospital admissions and mortality found near
Heathrow airport (Hansell et al, BMJ 2013). UFPs (particulate
matter <0.1mcg/m3 diameter) penetrate into the bloodstream
and have been associated with cardiovascular disease in short
term studies (>80) and the small number of long term studies
available, but UFPs are not routinely monitored. We conducted a monitoring campaign with 483 thirty-minute
periods of co-located measurements for LAeq noise and UFP at
roadside and background sites in Norwich, a medium size city in
the south-east of England. We found a weak to moderate
(R2=0.23, equivalent to Pearson’s R of 0.48 ) between noise and
UFPs, suggesting that road noise is unlikely to be strongly
confounded by UFP exposure. We have recently concluded (December 2018) a measurement
campaign for noise and ultrafine particles near and under flight
paths to and from London Gatwick airport, with co-located
monitors at nine sites for >500 thirty-minute periods and will
present preliminary analyses from this work.
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Numerical simulation of in-vehicle sound field under conditions of open or closed car windows
Liang Linda, Guangzheng Yu, Le Yu
[Abstract]
As computation technology develops, in-vehicle sound field
numerical simulation has attracted more and more attention.
However, it is difficult to simulate sound field in-vehicle at high
frequency by using wave-equation-based numerical methods,
such as finite element method (FEM), finite difference in time
domain (FDTD), etc. To extend the region of effective
frequencies, Aretz used a combined method consisting of finite
element method (FEM) and ray-tracking method to simulate
the sound field in a car up to 12kHz, in which the segmentation
frequency is 800 Hz. (Aretz M,2014) Even so, the combination of
the spectrum at the segmentation frequency between two
methods still remains a well-known problem and is worthy of
investigating. In this work, an improved FEM was adopted to
simulate the in-vehicle sound field. Through an optimized car
model with smoothing fine structures and non-uniform meshing
grids, the computation was optimized and the effective
frequency was extended up to 4 kHz. Further, sound field
distributions under conditions of open or closed car windows
(the condition of opening windows is approximated by an
absorptive boundary) were obtained, to analyse and validate
the optimized car calculation model and numerical method.
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Sound field reproduction in a cabin using loudspeakers
Dong Wang, Guosong Feng, Gongbo Ma, Xinming Li
[Abstract]
To evaluate the interior sound quality of an aircraft during flight, multiple loudspeakers are used to reproduce the sound field inside a mock-up cabin which is used for the passengers to experience on ground. The reproduction is based on the principle of the frequency-domain pressure matching approach. To reduce the influence of measurement errors in acoustic transfer functions, the case of more loudspeakers than the matching points is studied, which is different from most studies before. Some important aspects, including the number of loudspeakers and the loudspeakers placement are discussed. Both numerical and experimental examples are both described in this paper.
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Artificial Neural Network predicts noise transfer as a function of excitation and geometry
Dimitrios Ernst Tsokaktsidis, Timo Von Wysocki, Frank Gauterin, Steffen Marburg
[Abstract]
The Noise, Vibration and Harshness development of vehicle components still heavily relies on full vehicle tests. To reduce costs, the number of simulations is increasing. Commonly used simulation methods (Multi-body-Simulation, Finite-Element-Method and Statistical-Energy-Analysis) are only valid within a limited frequency range and need high computational resources. The manually created models require validation through measured data. Holistic digitalization is therefore not achievable with today’s simulation methods. Machine learning as a different approach lets an algorithm compute the physical relation between input and output. Once this relation is found, the trained neural network can predict the output to any new given input. Time- and frequency-domain input data show different suitability. Network architecture and hyper-parameters are essential for the outcome. The network training is supported by Finite-Element computations.
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Effect of temperature variation on the perceived annoyance of rattle sounds in the automotive industry
Mohsen Bayani, Casper Wickman, Rikard Söderberg
[Abstract]
Product quality improvements and electrification in automotive industries, denote the growing need for squeak and rattle free cars. Studying the relation between main sources of rattle sound, like temperature changes and the emitted sound can help to improve the design robustness. In this research, the effect of temperature on the perceived annoyance level of rattle sounds that are generated from selected material pairs from the car cabin is studied. The sound stimuli were collected from a rattle test apparatus by binaural technology in laboratory condition, by a parameter study on temperature, gap and material pairs. Estimated annoyance levels, using the psychoacoustic metric developed through a subjective listening survey, show that perceived annoyance of rattle sounds varies differently for various materials in different ambient and boundary conditions. Employing this approach can lead to a rattle sound database for material pairs to be incorporated with geometry variation results for requirement setting and design robustness improvements in the early development phases of passenger cars before the physical prototypes are available.
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Noise Distribution Change of the Backside Urban Blocks depending on the Plans of the Roadside Buildings
Chan Hoon Haan, Yong Seong Kim
[Abstract]
The noise of the backside roads is affected by the following
factors including the height of the roadside buildings, the
distance between the road and the side streets, distance
among adjacent roadside buildings, and the difference of
the adjacent building heights.
The present paper aims to investigate the influence of these
factors on the noise of the backside area of the roads. In
order to this, the noise levels were measured in 16 urban
blocks of a busy city. The both noise levels on the road and
the backside street were measured simultaneously at the
designated points of each urban block.
As the results, It was found that the average noise reduction
was 12dB on the basis of the building height of 4m. Also, it
was revealed that for each 4m increase in the building
height, noise reduction is occurred by 2dB.
Also, it was proved that the noise of the back streets is mainly
affected by the lowest height of the roadside buildings. The
noise is increased by 0.5dB for each 4m increase of the
height difference between adjacent buildings. Also, for each
0.5m increase in the distance between roadside building,
noise reduction is decreased by 1dB.
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Estimation of Infrasound Source Positions using Multipoint Observation
Zhenglie Cui, Ryosuke Sasahara, Ryouichi Nishimura, Yôiti Suzuki
[Abstract]
Infrasound can be excited by dangerous global
phenomena such as tsunamis and volcanic eruptions.
Because infrasound travels in the atmosphere significantly
faster than tsunami gravity waves, early detection of
tsunamis can be achieved by monitoring this infrasound. To
accurately identify a tsunami source (i.e., the position of the
infrasound source), it is important to use numerous sensors
distributed with long baselines. We developed a low-cost
infrasound observation device using micro-electro-
mechanical system based atmospheric pressure sensors. To
test its usability, indoor and outdoor tests were conducted
using multipoint observations. In the indoor experiments, the
infrasound was excited by opening and closing a door, and
by bursting a large balloon. The accuracy of time
synchronization was examined by positioning infrasound
observation devices at different points. The results showed
that the time errors were within one sampling time (10 ms) in
nearly all the indoor conditions. In the outdoor field test,
infrasound excited by the explosion of fireworks during a
festival was observed at multiple points in Sendai, Japan.
The results showed that the source positions can be
estimated with reasonable accuracy. This suggests the
suitability of the device to estimate the source position of
tsunami-scale earthquakes.
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Urban canyon noise assessment: sound incidence on façade and mitigating actions
Jessica De Almeida Xavier, Aline Lisot, Paulo Fernando Soares
[Abstract]
The urban noise problems aggravate when roads and buildings
configurate urban canyons, especially due to quasi-
reverberation phenomenon. This study objective was to propose
mitigating measures for urban canyon situations. To know sound
behavior in these places, it was carried out sound pressure levels
monitoring in an urban canyon, in 4 characteristic rush hours, to
evaluate the variation of sound levels during the day. On
morning rush hour, the monitoring was done on 10 floors of a
building to evaluate sound levels variation with increasing
altitude, along the façade. The noisiest periods of the day are
late afternoon (18:00 - LA,eq=73,8 dB) and early in the morning
(08:00 - LA,eq=72,8 dB). There was a predominance of low
frequencies at all times and floors measured. Sound levels do
not decrease with increasing height uniformly, being the third
floor the noisiest (LA,eq=69,3 dB). The mezzanine causes an
acoustic shadow on the leisure pavement, therefore the noise
levels on this floor are the lowest (LA,eq=62,4 dB). The monitoring
results, both on the floor and in height, exceeded the
acceptable levels of Brazilian technical standards by 8 dB to 24
dB, depending on the situation. Measures have been proposed
to mitigate noise pollution.
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Immission measurements of wind turbine noise with a microphone array to set up a sound database
Sebastian Mellert, Dagmar Rokita, Friedrich Ueberle
[Abstract]
Today there are several databases available on the internet where different sounds for many purposes can be found. Especially common sounds are available in these collections. Other sounds such as sounds of wind turbines are rarely found. For this reason in science, sounds of wind turbines are mainly artificially generated. These simulations are based on complex models that imitate the reality very well, but do not fully reflect it. In order to close this gap of real recordings, the research project X-Eptance Imulse is carrying out sound measurements with an acoustic camera in wind farms. From these measurements, not only the real sounds of a wind turbine can be determined. Through analysis in frequency domain, it is possible to generate a specific signature for a respective type of a wind turbine.
In a first field campaign, measurements were taken on the same type of wind turbine in two wind farms. The analysis of these measurements has led to the first indicators for the signature of this type. Presently, the analysis is performed manually. Later on, it is planned to perform the analysis automatically and to store the sounds and signatures in public accessible database.
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Analysis of Low-Frequency Noise from Wind Turbines using a Temporal Noise Code
Franck Bertagnolio, Helge Madsen, Andreas Fischer
[Abstract]
The growing demand for clean energy has led to the installation of large wind farms worldwide. This is often met by criticisms from nearby communities due to noise annoyance issues. Even at large distances, wind farms may occasionally be heard due to specific atmospheric conditions. Low-frequency noise has been mentioned as a possible source of annoyance for neighbors. The aim of this work is to use an existing aeroelastic code commonly used for wind turbine design. The atmospheric inflow turbulence is carefully modeled in the time domain and the loading on the blade can be used as an input for low-frequency noise generation. It is found that, in certain conditions, the previous mechanism (blade-turbulence interaction noise) can dominate the ambient low-frequency noise when confronting the model results with actual field measurements. Furthermore, a study of the characteristics of low-frequency noise is conducted. A series of parametric sensitivity analyses are conducted to identify the main parameters driving low-frequency noise levels, at least those that do have an influence on the proposed model.
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Environmental noise event classification based on self-organizing map using psychoacoustic features and spatial filtering
Jure Murovec, Luka Čurović, Tadej Novaković, Jurij Prezelj
[Abstract]
One of the main challenges of environmental noise monitoring
is classification of noise events. This is critical for assessment of
individual noise source contribution to the overall noise level.
Numerous attempts have been made to provide algorithms for
such classification, however with low certainty. The main reason
for poor results is lack of proper noise features. For this purpose a
new feature ”noise source direction” has been applied, based
on a small horizontal microphone array. It uses a never seen
before algorithm which is computationally inexpensive and
provides exceptional beam-pattern directivity. The array is
implemented as a part of autonomous environmental noise
source classification system. Some of the features are provided
directly from the array itself and are combined with
psychoacoustic metrics. Artificial Neural Network (Self
Organizing Map) performs the classification based on extracted
spatial and acoustical features of environmental noise sources.
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Prediction of quiet side levels in noise map calculations – an initial suggestion of methodology
Jens Forssén, Andreas Gustafson, Laura Estévez-Mauriz, Marie Haeger-Eugensson, Meta Berghauser Pont
[Abstract]
Urban morphology, i.e. shape and position of buildings in relation to
streets, affects the distribution of noise and air pollution and can
thus, through urban planning and design, be used to improve urban
areas. This is of interest in an ongoing project where both air
quality and noise are in focus. The present paper concerns the
prediction of noise levels at positions of non-direct noise exposure
such as noise levels at inner yard facades and values of noise
contour maps at inner yards. With increasing densification, housing
is built closer to the roads and the directly exposed facades receive
higher noise exposure levels. The balancing effect of access to a
quiet side is often counted on, supported by regulation, whereas the
noise levels predicted using commercially available software are
often not correct at those points. A suggested methodology is
described for how to combine the previously developed Qside model
with a commercial noise mapping software to improve the
prediction for non-direct noise exposure situations like essentially
enclosed inner yards. We present also initial noise level results as
outcome of a model study of varying building morphologies.
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Effect of handy microphone movement in Mixed Reality visualization system of sound intensity
Yuta Kataoka, Wataru Teraoka, Yasuhiro Oikawa, Yusuke Ikeda
[Abstract]
We proposed a system of real-time measurement and visualization
for three-dimensional (3D) sound field by using Mixed Reality (MR)
device. MR device enables effective visualization of information with
depth perception using the 3D computer graphics (3DCGs)
superimposed on the real world. Our system can superimpose 3D
sound intensity data on the measurement points in the see-through
head mounted display (STHMD). Visualizing the 3D sound intensity
map by the MR visualization system helps us to design the sound
field and to detect the source of noise. We measure sound intensity
in the measured area by scanning a handy 4-ch microphone freely
and slowly. The camera of STHMD recognizes an AR marker installed
at the microphone and the measurement point is detected form the
position and rotation of the AR marker. However, the measurement
results of sound intensity would be affected by the doppler effect
because of microphone movements. In this paper, we investigate the
effect of microphone movements using a simulation. In addition, we
obtain the moving speed of microphones with AR marker in the MR
visualization system and compensate the value of sound intensity by
the measured microphone speed.
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Spatial Principal Component Analysis of Head-Related Transfer Functions using their complex logarithms with unwrapping of phase
Shouichi Takane
[Abstract]
Head-Related Transfer Function (HRTF) changes depending
on sound source position and has strong individuality. In
order to compactly represent the spatial variation of the
HRTF in acceptable accuracy, the Spatial Principal Component
Analysis (SPCA) has been adopted in many researches.
Accuracy of the reconstructed HRTFs with the SPCA is
different depending on which domain is used for the SPCA.
In this paper, a new procedure for the SPCA of the HRTFs
using complex logarithm of the HRTFs with unwrapping of
phase is proposed. Moreover, the performance of the
procedure is evaluated via the SPCA of HRTF dataset that
involves the HRTFs of multiple subjects with sound source
direction in upper hemisphere.
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Variability of Head Related Transfer Functions across subjects
Maciej Jasiñski, Jan Zera
[Abstract]
The paper discusses across subject differences of measured Head Related Transfer Functions (HRTFs) in horizontal and vertical planes. The comparison was also done with reference to the results obtained with the use of an acoustic manikin. The work was done for an exemplary database available on-line and for the HRTFs measured with the use of a specific measurement setup in our laboratory. In the latter case, acquisition of HRTF pairs (left and right ear) was done at 370 sound source positions in the surrounding space with the use of two miniature microphones placed at the entrance to the blocked ear canal. Across subject differences of measured HRTFs were assessed in horizontal planes at various elevation angles. The results showed that HRTF is nearly symmetrical in the 300 Hz to 1 kHz frequency band. For frequencies above 1 kHz (more so above 5 kHz), as expected, the effect of interaural asymmetry increases due to filtering effect of complicated anatomical structures of pinnae. The influence of these effects differed among 15 subjects who participated in the study. The results obtained with the use of the acoustic manikin were not representative (as an average) for results with subjects.
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Active Control of Scattering Effects in 2.5D Multizone Reproduction
Junqing Zhang, Wen Zhang, Lijun Zhang, Mengyao Zhu
[Abstract]
Multizone reproduction has gained significant interest in recent
years for its advantage that listeners can enjoy their personal
audio without physical isolation or wearing headphones.
However, study of multizone reproduction has been focused on
reproducing sounds in an empty listening space. The scattering
effect of the human and objects in sound zones is often
neglected, causing degraded system performance. In this work,
we focus on reproduction on a 2D horizontal plane using 3D
secondary sources, i.e., 2.5D reproduction. Through experiments
we show how scatters such as human head and body in sound
zones influence reproduction results. A modal-domain method
for 2.5D multizone reproduction with an active control strategy
to correct the scattering effect is proposed. The method is
based on modelling the human head as a rigid sphere, based
on which analytical expressions are developed to represent the
scattered sound fields. We then proposed an active control
strategy to correct the scattering field. The method is validated
through simulations and experiments using a 48-element circular
loudspeaker array. Experimental results especially show the
improved ACC performance using the proposed method.
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Dynamic local sound field synthesis with multi-channel 1-bit signal reproduction system
Kurokawa Kakeru, Izumi Tsunokuni, Yusuke Ikeda, Yasuhiro Oikawa
[Abstract]
Local sound field synthesis (Local SFS) with virtual loudspeakers,
which are placed densely around a local listening area, has
been proposed to reproduce a desired sound field with higher
accuracy.
However, as it is necessary to limit the spatial extent of a listening
area in Local SFS, there is a risk that a listener’s head moves out
of the listening area as a result from the listener’s natural
movements.
In this paper, we propose a dynamic sound field synthesis
system where the sound field synthesis area moves so as to track
the listener’s head, thereby allowing body movements.
To facilitate the construction of a massive loudspeaker system,
we have adopted a playback system based on a high-speed 1-
bit signal that can directly drive a loudspeaker.
The proposed system can move a synthesized area by loading
the Local SFS signals corresponding to listener’s head position
from a database which stored the signals that convolved with
the driving functions for multiple local areas.
In the experiments, we discussed a time discontinuity of the
wave front that occurs at updating sound field synthesis area
based on simulation and experimental results obtained with the
proposed system.
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Design of a Constant Beamwidth Beamformer for the Parametric Array Loudspeaker
Chuang Shi, Ruyu Bai
[Abstract]
The parametric array loudspeaker (PAL) is a directional sound
device that transmits inaudible ultrasonic frequencies in air to
reproduce audible sound based on the parametric array
effect. The convolution model describes the directivity of the
PAL by convolving the product directivity with the
Westervelt’s directivity. Owing to the contribution of the
Westervelt’s directivity, the beamwidth of the PAL gets
narrower when the difference frequency increases. A
change of timbre is therefore perceived when people stand
off the axis of the PAL. Hence, a constant beamwidth
beamformer is required. This paper works on the beamformer
design to have a product directivity that is getting broader
when the difference frequency increases in order for the
Westervelt’s directivity to be compensated for. In the end, a
constant half-power-beamwidth (HPBW) is achieved for the
typical frequency range of the PAL.
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Spectral-change Enhancement with prior SNR for the Hearing Impaired
Xiang Li, Xin Tian, Henry Luo, Jinyu Qian, Xihong Wu, Dingsheng Luo, Jing Chen
[Abstract]
A previous signal processing algorithm that aimed to
enhance spectral changes (SCE) over time showed benefit for
hearing-impaired (HI) listeners to recognize speech in
background noise. In this work, the previous SCE was
manipulated to perform on target-dominant segments, rather
than treating all frames equally. Instantaneous signal-to-
noise ratios (SNRs) were calculated to determine whether the
segments should be processed. Initially, the ideal SNR
calculated by the original signals before mixed was
introduced to the previous SCE algorithm (SCE-iSNR).
Speech intelligibility (SI) and clarity preference were
measured for the speech with steady speech-spectrum noise
(SSN) and six-talk speech (STS) maskers across 12 HI
listeners. Results showed the SCE-iSNR processing improved
SI significantly for both maskers at high signal-to-masker
ratios (SMRs) and for STS masker at low SMRs, while
processing effect on speech quality was small. In addition,
the estimated SNR obtained from real mixtures was used
(SCE-eSNR). SI and subjective rating on naturalness and
speech quality were tested for 7 HI subjects. The SCE-eSNR
algorithm showed benefit on SI for SSN masker at high SMRs
and for STS masker at low SMRs, and on naturalness and
speech quality for STS masker. The limitations of applying
the algorithms are discussed.
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Impact of Amplification on Speech Enhancement Algorithms Using an Objective Evaluation Metric
Zhuohuang Zhang, Donald S. Williamson, Yi Shen
[Abstract]
Hearing loss is prevalent among elderly adults, which leads
to speech-understanding difficulties in noisy environments.
Speech enhancement algorithms are thus proposed to
alleviate this speech-in-noise problem. However, most of
these algorithms have not been evaluated for hearing-
impaired people either with or without the use of hearing
aids. In this study, we evaluated the performance of
several speech enhancement algorithms [i.e., non-
negative matrix factorization based, deep-neural-network
based and long short-term memory (LSTM) based
algorithms] for hearing-impaired listeners using an
objective speech quality metric, namely the Hearing-Aid
Speech Quality Index (HASQI). The HASQI is based on a
physiologically inspired model of auditory processing,
which also allows the simulation of hearing impairment.
The evaluation was repeated separately for the typical
hearing characteristics of different genders in various age
groups. For the aided condition, linear amplification was
implemented using the NAL-R prescription formula. The
benefits from the speech enhancement algorithms
decrease with increasing degrees of hearing loss. With
amplification, the benefit diminishes for the listener group
with the most severe hearing impairment. Among the
various algorithms, the LSTM-based structures exhibit
superior performance with and without amplification.
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De-reverberation using CNN for Non-Reference Reverberant Speech Intelligibility Estimation
Kazushi Nakazawa, Kazuhiro Kondo
[Abstract]
Reverberation distorts speech, leading to degradation of
speech intelligibility and speech quality. We have been
investigating on methods to predict the speech intelligibility of
reverberant speech. This normally requires clean speech, which
is often not available. In a past paper, we attempted to use
Deep Neural Networks to reconstruct clean speech from
distorted speech. We predicted a frame of clean spectrum from
11 frames of reverberant short-time Fourier transform (STFT)
magnitude, and evaluated the clean speech that is
reconstructed from the predicted clean spectrum. The
segmental SNR (SSNR) improved by 3.5 dB in reverberant
environment, with reverberation time from 0.2 to 1.0 second.
However, with this method, the features were limited to 11
frames, shown to be inadequate to extract severe reverberation
feature. To improve SSNR further, we need to extract adequate
number of frames of STFT magnitude, and furher exploit the
temporal characteristics. For this reason, we adapt
Convolutional neural networks (CNNs), where we predict 257
frames of clean STFT magnitude from 257 frames of reverberant
STFT with 4×4 size filters. However, the SSNRs of predicted clean
speech remains unchanged from DNN so far. we will attempt
different size filters for improvement.
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On Non-Reference Speech Intelligibility Estimation Using DNN Noise Reduction
Hiroto Takahashi, Kazuhiro Kondo
[Abstract]
Methods for estimating speech intelligibility are classified into
two types. One is the full reference method that estimates the
subjective evaluation values using both degraded speech that
passed through the evaluation system, and the original speech
before degradation. The other is the non-reference method that
estimates intelligibility from only degraded speech. In speech
intelligibility estimation, it is assumed that the original speech
can not be obtained. Therefore, from the viewpoint of
practicality, a non-reference method is required. In this
research, we consider a method to apply the full reference
method on degraded speech, and original speech estimated
from degraded speech. The model used here is an intelligibility
estimation model using Deep Neural Network (DNN) which
shows higher estimation accuracy than other methods such as
logistic regression and random forest. In this paper, we
compared the estimation accuracy between estimation using
original speech and estimation using estimated speech. In the
closed test, the correlation coefficient was 0.9721 and the RMSE
was 0.0743, showing the same degree of accuracy as when the
original speech was used. However, in the open test, the
correlation coefficient and RMSE was 0.8307, 0.1839,
respectively, indicating room for improvement.
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Reverberant speech recognition with actual cochlear implants: verifying a pulsatile vocoder simulation method
Fanhui Kong, Xianren Wang, Xiangbin Teng, Nengheng Zheng, Guangzheng Yu, Qinglin Meng
[Abstract]
Classic channel vocoders have been widely used for
simulating cochlear implants (CIs). However, some important
features of state-of-the-art CI strategies were usually
ignored, such as pulsatile current, broad current spread, and
low quantization resolution of intensity. To solve this problem,
a novel vocoder was proposed, which is combined from
Gaussian-shaped acoustic pulses each corresponding to an
electric pulse and can directly transformed CI-
electrodograms to sounds. Compared with the classic
channel vocoders on a reverberant speech recognition
task, this new vocoder was considerably more sensitive to
the reverberant condition (published in APSIPA-2018).
Moreover, its performance is comparable to actual CI users
in previous CI-reverberation studies. Here, the same
paradigm as in the APSIPA-2018 paper was tested on actual
CI users. We found that under reverberant conditions the CI
users got comparable scores to the simulated performance
from the new vocoder simulation in the NH cohort, while the
classic vocoder predicted significantly much higher
performance This study demonstrated that the classic
vocoders overestimated the task performance and further
lent great support to the usage of the novel vocoder for the
simulation of reverberant speech recognition in adult post-
lingually deafened CI users.
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Relationship between Drive Signal and Stability in MIDS Modulator
Shigeto Takeoka
[Abstract]
We proposed a matrix driving method using a Multi-Input Delta-
Sigma (MIDS) modulator as a method of driving a very large
number of speakers. The MIDS modulator shares a quantizer in a
plurality of Δ∑ modulators and by using the multiplication output,
it is able to drive the speaker array individually and at the same
time though it is a common wiring. In other words, it is possible to
individually drive N x Nch speaker arrays with 2 x N wirings,
instead of using time division as in image devices.
In this paper, we consider the relationship between input signal
and stability in these MIDS modulators. In general, it is known
that the amplitude of the input signal of the Δ∑ modulator is
limited in relation to the peak value of the transfer function of
the quantization noise. On the other hand, in the MIDS
modulator, a common quantizer is used for a plurality of signals,
and quantization noise viewed from each input signal is related
to other input signals. Therefore, focusing on the entire input
signal, we confirmed that the stability of the system in the MIDS
modulator depends on the total energy of the input signal by
simulation.
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Measurements of Current Noise and Distortion in Resistors
Youhei Miyaoka, Minoru Kuribayashi Kurosawa
[Abstract]
In RF telecommunication systems, the passive intermodulation
has become a significant problem to get stable radio
communication. The passive elements’ distortions are actual
problems in telecommunication systems due to multi-carrier
frequencies. In audio systems, much more frequency
spectrum peaks are existed and the spectrum peaks change
in time and time. If a resister generates some distortions, the
distortions are significant problems in elector-acoustic systems.
In the resistor, there is current noise generated by the current
flow. Due to the influence of the current noise, the current flow
is not always constant, so this may be nonlinear response.
Then, it is thought that the resistor has nonlinearity, and the
resistors’ nonlinearity causes the signal to be distorted. In this
study, we first measured the current noise of several types of
resistors at different parameters; the resistance value, the
applied voltage and wattage. Then we designed a measuring
circuit for the distortion and measured the distortion in the
resistors. Finally, we try to find the relation between the current
noise and the distortion.
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Optimization of vented earpiece designs using the finite element method
Reinhild Roden, Nick Wulbusch, Florian Denk, Henning Schepker, Alexey Chernov, Simon Doclo, Matthias Blau
[Abstract]
Vented earpieces equipped with multiple receivers and microphones operated using suitable algorithms offer the possibility to achieve acoustically transparent sound presentation by conserving the acoustic properties of the open ear, see also Schepker et al. (this conference). Acoustic transparency should ideally hold for all external sound sources, which translates into the requirement that the relative transmission paths between microphones located in the earpiece and the eardrum should be independent of the positions of external sound sources. Since measurements made on various prototypes showed different degrees of dependence on external source positions, numerical simulations using the Finite Element Method (FEM) were performed to better understand the impact of the microphone positions, with the ultimate goal to optimize the earpiece design for acoustic transparency. Both commercial and custom implementations of the FEM were used, considering generic as well as individual 3D geometries of the outer ear. For the existing prototypes, the dependence on external source positions observed in the measurements could be reproduced by the numerical simulations, indicating the suitability of the method. A strategy to optimize the earpiece design is proposed.
Funded by the Deutsche Forschungsgemeinschaft (DFG) - Projektnummer 352015383 - SFB 1330 A4 and C1.
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Robust Long-distance Aerial Audio Data Hiding: Comparison between Amplitude Modulation-based Hiding and Bilateral Time-spread Echo Hiding
Akira Nishimura
[Abstract]
Voice evacuation and mass notification systems that
use outdoor
loudspeaker systems play an important role in
propagating emergency
information in Japan. However, there is a severe
problem of informing
the hearing impaired and the elderly who have
difficulty in hearing as
well as informing tourists and foreigners who do not
understand the
Japanese language. Therefore, aerial data transmission
combined with
voice evacuation messages should be realized with the
aid of
information hiding technology. We proposed the
bi-lateral time-spread
echo hiding method for an aerial audio data hiding in
which speech
signals are broadcast by the outdoor loudspeakers of a
voice
evacuation and mass notification system. Also,
amplitude
modulation-based watermarking has been shown its
robustness against
reverberation and background noises. Evaluations of
the both methods
for speech signals were conducted using computer
simulations including
several disturbances caused by the long-distance (from
70 to 800
meters) aerial transmission of sounds. The frequency
response of a
distant horn-array loudspeaker system, the absorption of
sound by the
atmosphere, reverberation and a single long-path echo
and additive
background noise are simulated as disturbances. The
results showed
that the bi-lateral time-spread echo hiding was more
robust against the disturbances than
the amplitude modulation-based method.
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Comparison of Acoustic and Laryngeal Parameters between Healthy and Disordered Subjects
Patrick Schlegel, Stefan Kniesburges, Michael Döllinger, Stephan Dürr, Anne Schützenberger
[Abstract]
In voice research and clinics various mathematical parameters are in use to describe different features of the recorded data. However, little is known about norm values of parameters and how they are affected by different pathologies such as functional dysphonia.
In this work we investigate the expressiveness of in total 59 glottal area waveform (GAW) parameters (from which 33 were also calculated for the synchronously recorded audio signal). The parameters were computed for females (162 healthy, 95 disordered) and males (67 healthy, 31 disordered) whereas all disorders were functional dysphonia related. For each subject laryngeal high-speed videoendoscopy was performed applying the clinical prevalent Photron Fastcam MC2 camera (frame rate: 4000 fps, spatial resolution: 512×256 pixels). From each high-speed recording a section of 1000 Frames (=0.25 s) of preferably constant phonation with synchronously recorded audio was chosen for parameter calculation.
Data analysis revealed that most parameters, for audio and GAW signal, did not show clear differences between the healthy and the pathologic groups. However, it may be possible that a more complex, nonlinear multi-parametric based approach could allow for a differentiation between healthy and disordered subjects. This project is funded by the Deutsche Forschungsgemeinschaft (grant no. 323308998).
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Characterization of turbulence noise in breathy human phonation
Philipp Aichinger
[Abstract]
Perceived breathiness is a voice trait contributing to the assessment of the human phonatory function in the clinical care of voice disorders. A vocal dysfunction requiring clinical treatment such as logopedic therapy of phonosurgery is often associated with increased breathiness. Breathiness is provoked by turbulence noise, which is generated at the glottal constriction of the larynx. Pulsatile noise may be generated owing to the cyclic opening and closing of the vocal folds, whereas a quasi-static noise component may be generated when a glottal chink is present, i.e., a residual glottal gap.
Clinically observed voice signals are selected from an existing database of audio signals recorded during high-speed videolaryngosopy. Only sustained vowels with temporally homogenous voice quality are analysed. Harmonic modelling of the audio signal enables segregation of the noise component by subtraction. Harmonic modelling using Fourier synthesis considers that the glottal cycle length and peak cycle amplitude may be modulated. Statistical properties of the noise such as energy and spectral features (static/pulsatile) are reported. The results presented improve the current understanding of production, acoustics, and perceptual attributes of dysphonic breathy voices.
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Aerodynamic and aeroacoustic optimization of a centrifugal fan with backward-curved blades by means of inverse design
Chris Eisenmenger, Stefan Frank, Hakan Dogan, Martin Ochmann
[Abstract]
Centrifugal fans with backward-curved blades are characterized by a relatively high efficiency compared to other types of centrifugal fans. In contrast, there is an overall louder, tonal sound and higher rotational speeds. Therefore, in home appliances such as clothes dryers, rather squirrel cage fans are used. However, the influence of the fan unit increases with the need of a continuously rising efficiency of the overall device. Hence, the use of centrifugal fans with backward-curved blades for operation in commercial tumble dryers with respect to aerodynamic and aeroacoustic properties will be examined. In the first steps, a reference fan has been designed using classical design guidelines. The fan is numerically studied with a hybrid Computational Fluid Dynamics/Computational Aeroacoustics approach and the results are experimentally validated with the in-duct method according to ISO-5136. The reference fan will be further optimized in terms of its aerodynamic and aeroacoustic properties. For this purpose, the ”inverse design” method is used to extend the scope of classical design rules. Using design of experiments (DoE) and genetic algorithms, various parameters and their influence on fan efficiency and sound radiation are examined. In further project steps, promising fan designs will again be investigated numerically and subsequently validated experimentally.
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Predicting Cooling Fan Noise by Numerical Conditions Using Compressible Large Eddy Simulation
Kimihisa Kaneko, Tsutomu Yamamoto
[Abstract]
This paper describes predicting motor fan noise using direct
computational aeroacoustics (CAA) methods. In the direct CAA
method, both turbulent flow and acoustic phenomenon are
solved at the same time with compressible Navier-Stokes
equation. Large Eddy Simulation is applied for solving turbulent
flow. One of the issues for using direct CAA method is obtaining
the numerical simulation accuracy for 10 times difference
between fluid velocity and sound velocity. High order numerical
schemes are applied to obtain high accuracy solutions. The
purpose of this study is developing simulation technology for
using in the low noise product development.
Therefore, in this study, we investigated the accuracy effect of
numerical simulation conditions (e.g., numerical schemes, time
integral discretization, and computational grid size). And, to
verify our direct CAA model, pressure fluctuation of fluid
measurement and sound pressure level measurement were
carried out with the experimental structure. It was found our
proposal numerical simulation model can apply to the product
development stage.
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Experimental Study on BTI Broadband Noise Reduction with Wavy Leading Edge for Sweep Blade
Weiyang Qiao, Xin Guo, Wenhua Duan, Fan Tong, Weijie Chen
[Abstract]
Experimental studies are performed to investigate the Blade-Turbulence Interaction (BTI) noise reduction of a sweep blade using wavy leading edge(WLE) configuration. The noise radiation from the blade is generated by the interaction of blade and incoming anisotropic turbulence which is produced by a rod whose wake impinges onto the downstream blade. The BTI noise of the blade with swept angle of 30 degrees and constructed by NACA0012 airfoil with and without wavy leading edge were measured with the flow speeds of 30m/s to 70m/s. A linear array of 31 microphones was used to determine the BTI noise. The clear and quantitative sound radiation results of the leading-edge noise source of swept blade are obtained. Parametric studies of the WLE amplitude and wavelength are conducted in order to understand the effect of WLE on BTI noise reduction. It is observed that the OASPL reduction is sensitive to both the amplitude and wavelength of the WLE. The WLE with the largest amplitude and smallest wavelength can achieve the most considerable OASPL reduction up to 2dB. It is found in this study that the spectra of BTI noise reduction with Strouhal number is almost the same for same WLE under different airflow velocities.
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Study on relationship between the geometry of Leading and Trailing Edges of Airfoil and their Aeroacoustics Parameters at low Reynolds number
Joanna Kopania
[Abstract]
In last few year’s there has been considerable interest in the use of edge modifications for the reduction of airfoil noise. One of the most commonly used methods is sawtooth serration edge inspired by the owl’s wing. The edge serration is known to be quite effective in reducing aerodynamic drag and self noise radiation. But the correlation between edges modifications of airfoil and noise reduction was studied also for other forms such as wavy or M-shape. Results of an experimental study on turbulent flow over the two types of NACA airfoil, symmetrical and unsymmetrical, with a serrated and wavy trailing and leading edge are presented in this paper. Detailed aeroacoustic measurements are presented of the noise radiated by study airfoils in a low to moderate speed flow under acoustical free field conditions. Parametric studies of the amplitude and wavelength characteristics are conducted to understand the effect of edges modifications on noise reduction. The influences of geometric parameter applied tooth and waves, free-stream velocity and angles of airfoils on the noise reduction effect of are observed.
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Methods of estimation of frequency spectrum and power of the sound generated by an unsteady flow through a sonic crystal
Viktor Hruska, Michal Bednarik, Milan Cervenka
[Abstract]
Although the importance of sonic crystals serving as sound barriers is growing, only a little interest was given to the sound emission from the sonic crystal due to the windy weather conditions. A computational framework for a future extensive parametric study is proposed in the article. Three different methods are compared on a representative test case: a) the large eddy simulation, b) a hybrid calculation combining Reynolds averaged Navier-Stokes equations (RANS), Curle’s analogy and the boundary element method and c) a hybrid calculation consisting of RANS simulation and subsequent calculation by means of Howe’s analogy making use of a suitable Green’s function. Comparison of the characteristic frequency, the overall spectrum shapes and radiation patterns is given.
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An effective method for measuring the thickness of Cobalt-rich Manganese Crust based on the neighborhood information and dual-channel information
Feng Hong, Haihong Feng, Minyan Huang, Binxian Wang
[Abstract]
The thickness of Cobalt-rich Manganese Crusts (CRC) is one
of the important information to map and estimate the
volumetric distribution of deep-sea mineral. In this paper, we
propose an improved method for measuring the thickness of
Cobalt-rich Manganese Crust based on the neighborhood
information and dual-channel information. The high-
frequency signal and low-frequency parametric signal is
obtained by feeding the signal received by the transducer
centered at the array to two designed filters. The reflection
instant of the top surface is determined by the high-
frequency signal, while that of the bottom surface is
determined by rule based on the neighborhood information
assuming that the CRC has the local thickness invariability.
Finally, the demonstration experiment is conducted in the
tank to validate the effectiveness of the method.
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A Study of Acoustic Characteristics at Sea Bottom Sediment Including Organic Matter
Hanako Ogasawara, Kazuyoshi Mori, Hiroshi Yagi
[Abstract]
At shallow area, organic rich sediment is piled up and run away
according to the tidal stream or upper sea water movement.
Authors measure the acoustic characteristics of organic rich
sediment to monitor the movement of those sediment at the
sea bottom. The bottom statement environmental changes
affect benthic activities.
Sample sediment was obtained from Ariake Sea in Japan,
which consist with cray and many organic matter, was
measured its acoustic speed and attenuation with the
frequency of 2 MHz and 5 MHz in laboratory. At the same time,
the grain size distribution was also measured. This result will help
to understand the reflected waves from the sea bottom and
sound propagation simulation in models.
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Snapping shrimps sound in Black sea
Nikolay Grigorievich Bibikov, Andrey Ninelovich Serebryany, Michail Pavlovich Ivanov, Oleg Borisovich Ovchinnikov
[Abstract]
We recorded signals of snapping shrimp (Alpheus dentipes)
using high-frequency hydrophones in a number of coastal
areas in the eastern part of the Black Sea. These signals
completely determined the noise background in these areas.
In all registered cases, a relatively low-frequency precursor and
a powerful short pulse characterized each individual signal,
the spectrum of which included frequencies exceeding 100
kHz. The relative rarity of powerful clicks, the level of which
reached several hundred Pa, made it possible sometimes to
identify the individual click by analyzing its waveform. It has
been found that in some cases one object can emit almost
identical clicks at intervals of about one second. In one of the
recording sites, we carried the observations out for several
years, which made it possible to relate the activity of the
emitters to the daily cycle, seasons and weather conditions. At
this registration point, we performed a statistical analysis of the
point process of generating such intense clicks. It was shown
that the process undergoes slow changes over time and can
be interpreted as a fractal point time series. The study was
supported by Grant RFBR-abh 19-52-40007.
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Time Arrival Structures of the Empirical Green's Function Extracted from Ambient Noise in Shallow Water
Xishan Yang, Fenghua Li
[Abstract]
Theoretical and experimental results show that the empirical
Green’s function(EGF) can be extracted by correlating the
ambient noise between two sensors. Ambient noise was
recorded by bottom mounted horizontal line arrays(HLAs)
deployed in shallow water with different horizontal separations.
Stable EGFs were extracted from ambient noise correlations
between the HLAs using array signal processing which can
accelerate the convergence rate of EFGs significantly. The EGFs
extracted from the short-distance hydrophone separation have
three distinct envelopes which correspond to the head wave,
the direct wave and the surface-reflected wave. The EGFs
extracted from the long-distance hydrophone separation reveal
the multipath arrival structures. The fluctuations of travel times of
EGFs are shown to be in agreement with the variations of sound
speed profiles, which can be used to estimate the sound speed
fluctuations.
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Upwelling Effect on Acoustic Propagation in the Cabo Frio Island Bay
Vicente Junior Barroso, Rogério Calazan, Marcus Simões, Sérgio Jesus
[Abstract]
Coastal upwelling is an oceanographic phenomenon that
challenges the understanding of acoustic propagation,
especially when it extends to shallow water areas with complex
geographic configuration. The data used in this paper was
collected in a semi-enclosed bay filled by the upwelling stream
through an inlet off the island of Cabo Frio, in the coast of Brazil.
The experiment was carried out in January 2019, under
BIOCOM, a joint initiative between the Brazilian Navy and the
University of Algarve (Portugal). Broadband acoustic signals
were transmitted over a 1.6 Km-long
transect across the upwelling flow during five days.
These signals were superimposed in a soundscape formed by a
strong biological acoustic signature and boat engine noise due
to close by recreation boats. Signals were received in
two hydrophone arrays: one vertical and another horizontal,
close to the bottom. The comparison of received data with that
generated with a 3D acoustic propagation model shows good
prospects for water column properties inversion. The ability to
track temperature changes of the highly dynamic system
formed by the conjunction of the upwelling stream, the tide and
the wind, allows for improving the performance of vertical line
arrays for other tasks such as source detection and localization.
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DOA Estimation of Underwater Targets via Improved Monopulse Method with Sonar Array
Jiani Wu, Changchun Bao
[Abstract]
In the field of sonar detection, the most commonly used
method for DOA estimation of underwater targets is the
beamforming algorithm. However, due to the Rayleigh limit
of resolution, this method cannot effectively resolve and
locate multiple targets within one beam. In this paper, In
order to resolve two targets in a single beam, we frst
establish an echo model of two unresolved targets with
sonar array. Then we derive an improved monopulse
method to estimate the DOA of the targets according to
the maximum likelihood estimation principle. The core idea
of the proposed method is that, two monopulse systems are
formed and each of them points at one of the targets, and
iterations are then carried out to reduce the DOA estimation
errors. The method is expected to gain advantages in DOA
estimation accuracy and moderate requirement on the
system complexity. The simulation results indicate that, this
method performs very well in many aspects, including less
computational complexity and faster calculation speed
compared with the maximum likelihood angle estimation
algorithm and smaller estimation error compared with the
single snapshot MUSIC algorithm. The estimation error is close
to the CRLB. Moreover, this method enhances adaptation to
inter-target amplitude difference.
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Characterising Nonlinearities in Acoustic Transfer path Measurements
Samira Mohamady, Allahyar Montazeri
[Abstract]
The purpose of this study is to characterize the measurement errors in impulse response measurement in room acoustic analysis. An auditor room is considered as a case study. A repeated measurement is carried out to measure 8 transfer function from a loudspeaker in the corner of the room and two microphone installed in the ear position. Here we use multi-sine excitation signal for measurement and compare the results with other type of excitation signals such as sweep sine signals. As the next step all 8 transfer functions are estimated with only one experiment using multichannel excitation and multi-sine signal. This will give not only a better signal to noise ratio but also reduce the experiment time as well as a more accurate transfer path in case of coupling between transfer functions. In this case, we are able to quantify various sources of uncertainties such as measurement noise, leakage, and nonlinear distortion by deeper analysis of the results. The approach relies on robust local polynomial method as an advanced technique for non-parametric measurement of the room impulse responses. The analysis shed light on the measurement technique and factor influencing to reduce the measurement errors and produce a high quality audio system.
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Rotor Dynamics Analysis under Uncertainty in Lubricant Film
Xiaodong Sun, Kheirollah Sepahvand, Zhe Liu, Steffen Marburg
[Abstract]
Predicting vibrations in rotor dynamics under the impact of
uncertainty is still a challenging issue. Research work
published in
this area is usually focused on the stochastic dynamics of
rotor-
bearing systems by considering the stiffness and damping
coefficients of simplified bearing models as random
parameters
rather than the original characteristics of the oil film. In this
paper,
the characterizing parameters of the lubricant film such as
the
dynamic viscosity are considered as uncertain parameters to
investigate their influences on the rotor dynamics response. To
this
end, the modified Jeffcott rotor model with offset disc is
established
considering the gyroscopic effect. At the same time, the
parameters
of the bearing are obtained by solving the dynamic
characteristics of
oil film. Then, the dynamic response of the rotor is numerically
calculated. In order to investigate the influence of the
parameters of
the lubricant film, the response is modeled as a stochastic
process
with unknown coefficients by using the generalized
polynomial chaos
expansion. The coefficients are then determined from some
realizations of the response on a set of collocation points.
Finally, the
accuracy of the method is compared with Monte Carlo
Simulation.
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In-plane Vibration and Tire Force Transmissibility owing to Tire Non-uniformity Defects
Zhe Liu, Kheirollah Sepahvand, Yintao Wei, Steffen Marburg
[Abstract]
Tire non-uniformities give rise to rolling force variation at the
spindle during steady state rolling. These kinds of characteristics
are to some extent unavoidable. Therefore, limiting and
managing the non-uniformities is of great economic importance
to tire manufacturers. In this paper, an analytical model for
predicting the high-speed uniformity and the force
transmissibility using low-speed uniformity data is presented. The
model is established based on a three-dimensional ring on an
elastic foundation model. Considering the tire mass unbalance
and the non-uniformity of the geometry, the variations of the
force on the tire-road contacting area and the length of the
footprint are analyzed. The mass and the geometrical non-
uniformities are transformed into generalized forces and the
equilibrium equations are solved in the space coordinate
system. The length of the footprint caused by the non-
uniformities is expanded by Fourier series, and the deformation is
integrated to obtain the variation of the axial force. The method
provides a possibility for the diagnosis and evaluation of high-
speed tire dynamic characteristics.
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Calculation of wave propagation characteristics in pre- deformed periodic lattice frame structures via Spectral Element Method
Marius Mellmann, Chuanzeng Zhang
[Abstract]
In recent years, the design, optimization and application of
periodic materials and structures, such as phononic and
photonic crystals, are becoming popular in many engineering
fields. For instance, the phononic crystals can manipulate the
elastic and acoustic wave propagation characteristics.
Consequently, it is feasible to create the so-called band-
gaps, which are certain frequency ranges, in which the
propagation of elastic or acoustic waves is prohibited.
Accordingly, such structures may have many promising
applications in sound and vibration insulation as well as wave
filters.
The vibration and wave propagation characteristics of a
periodic structure can be obtained by considering a
representative unit-cell with imposed Bloch-Floquet boundary
conditions and calculating the corresponding dispersion
curves. The calculation can be carried out by different
numerical methods such as the FEM, FDM, Plane Wave
Expansion (PWE) Method and Layered Multiple Scattering
(LMS) Method. In this paper the dispersion curves of pre-
deformed periodic lattice frame structures are calculated
using the Spectral Element Method (SEM), which is based on
the exact solution of the elastic wave equations. Numerical
examples will be presented and compared with the FEM
calculations. Keywords: Vibration, elastic wave propagation, band-gaps,
periodic lattice structures, phononic crystals, Spectral Element
Method (SEM)
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Metaheuristic optimisation of sound absorption performance of multilayered porous materials
Vivek Thaminni Ramamoorthy, Ender Ozcan, Andrew Parkes, Luc Jaouen, François-Xavier Bécot
[Abstract]
In multilayered-sound-packaging optimization, often the
search space becomes very large to optimise by brute
force as the number of acoustic parameters is increased.
We present the application of different metaheuristics for
the optimisation of multi-layered configurations involving
porous materials for maximizing the sound absorption
coefficient. The multi-layered system is composed of a
stack of facing screens (i.e. a woven or non-woven textile,
a perforated plate) and/or thicker porous material
combinations, chosen from a database of common
acoustic materials. In this case, the total thickness of the
system is kept constant while the acoustical parameters of
the layers are explored to obtain the parameters for the
highest sound absorption coefficient. We study different
representations of the structure. We also study different
algorithms, including evolutionary and stochastic local
search metaheuristic methods, and hybrids of these. We
will report results comparing the performance of different
combinations of representation and algorithms. This will be
extended to include more layers and materials, to obtain
improved sound packaging.
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Study on Simulation of Steady-state Sound Field Matrix in Complex Undersea Environment and Detection Sensitivity
Jun Suo, Yangyang Zhang, Simi Tang
[Abstract]
It’s crucial to simulate the acoustic detective effect timely and accurately between any two positions in the complex undersea environment in a real-time simulation platform. A numerical analysis method of sound ray propagating in complex undersea environment based on the eikonal equation, non-coplanar sound ray and sound velocity gradient vector is presented in this paper. The total undersea concerned information under certain accuracy requirement can be obtained in the algorithm. The multidimensional sound pressure array is established to simulate the whole sound space. The calculated amount for the real-time simulation of the acoustic detective effect is decreased from O(mnu) by the traditional eigenray solution to O(1) (m is the number of the non-layered structures, n is the grid quantity of the local layered structure, u is the number of the iterations in Newton method or Runger Kutta method). The real-time capability is highly improved by the method mentioned above. By comparison, the results of the simulation and the marine measurement are in good agreement. The concept of detective sensitivity is proposed to quantitatively describe the relation among the marine environment, the detectors and the sound sources. The concept can give good expression of the relation among the three factors.
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A new local boundary integral equation method for meshless acoustic computations
Hakan Dogan, Martin Ochmann
[Abstract]
The local boundary integral equation (LBIE) method has been introduced approximately three-decades ago, and has been successfully used in engineering problems such as elasticity, acoustics and electrodynamics. The method is based on writing the local weak form of the governing equation. If an appropriate form of the Green’s function can be found considering the governing equation in subdomains, the local integral equation may have a convenient form, e.g. without domain integrals or without the flux term on the subdomain boundaries. Considering the application of the methods to acoustic problems, four different test functions have been employed so far in the literature. All of the resulting formulations contain domain integrals. In this paper, we present a new form of Green’s function to be used in LBIE method for acoustic calculations, which yields a simple form of the local integral equation without domain integrals. This feature provides significant improvement in terms of the computational time (CPU) requirements. The efficiency and the accuracy of the new method is presented and compared with a Finite Element Method.
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Physical realization of the radiation of complex multipoles
Rafael Piscoya, Martin Ochmann
[Abstract]
Complex multipoles are sound sources located at positions
having
an imaginary component. They are also solutions of the
Helmholtz
equation, but their radiation patterns are different from
those
of their real counterparts. If rq is the complex source
position,
the source preferably emits the sound in a direction
indicated by
Im(rq) and the degree of focusing depends on the
amplitude of
Im(rq). A loudspeaker in a box behaves at low frequencies
like a
monopole, a loudspeaker without a box like a dipole and
a tuning
fork like a longitudinal quadrupole. In contrast, complex
multipoles do not have simple physical equivalents, since
they
are pure mathematical constructions. In the present work,
the
sound radiation of complex multipoles will be modelled by
an
array of loudspeakers whose amplitudes are determined
so, that
their radiation pattern is reproduced. To obtain the
amplitude of
each loudspeaker, a combination of the Boundary
Element Method
(BEM) and the Equivalent Source Method (ESM) is applied.
With the
BEM the transfer function of each loudspeaker is
calculated and
with the ESM their amplitude by minimizing the error at
certain
control points.
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The SPECFEM package : an open-source and scalable wave propagation solver based on a spectral element method
Etienne Bachmann, Jeroen Tromp
[Abstract]
The SPECFEM package is an open source wave propagation solver based on a spectral element method and a time domain explicit scheme, which permits a fast wave modeling in heterogeneous media without accuracy compromise. This method has been combined with a specific Gauss-Lobatto-Legendre integration rule, to minimize hardware requirements. Initially developed for geoscience purpose such as earthquake modeling, its use is also compatible with acoustic problems. It can perform 2D, 2.5D and 3D simulations, in acoustic, viscoacoustic, elastic, anisotropic, viscoelastic or poroelastic materials, and any combination of these materials. Simulations can run on multiple CPU cores, or on multiple GPUs, while achieving optimal strong and weak scaling efficiency for large problems. SPECFEM also features adjoint capabilities and sensitivity kernel computations, which enables reconstruction of velocity, density or attenuation maps from real data, using adjoint tomography techniques.
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Comparison of characterization methods for rigid porous materials
Ferina Saati, Christian A. Geweth, Steffen Marburg
[Abstract]
Theoretical and empirical models for the acoustic properties of
porous or granular media allow inverse determination of
properties such as permeability, tortuosity, and pore size
distribution. Granular materials are typically characterized using
acoustic models. However, the accuracy of the prediction
depends heavily on the input parameters and the chosen
method. In order to macroscopically characterize granular
media by means of impedance tube measurements, acoustic
inversion models are available assuming homogeneity as well as
a rigid framework. These models efficiently predict the behavior
of the materials and are often used to derive the intrinsic
properties of the media. However, it is often unclear how
precise such models represent the individual material properties.
In this work it is shown how the prediction of key pore
characteristics of granular media by means of acoustic
characterization methods is comparable to the prediction with
previous methods.
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Spatial sound made easy: metamaterial prisms
Thomas J. Graham, Chinmay Rajguru, Arash Pouryazdan, Gianluca Memoli
[Abstract]
This work demonstrates an application of an acoustic metamaterial prism in combination with a frequency limited musical piece, to create spatially relevant sound experiences. Specifically, we play an original composition made up of tones within two octaves through the prism and show that each note is redirected in a different position. We underpin our conclusions by measurements and a user study, inspired by soundscape techniques. This work concludes that these devices are not as frequency limited (i.e. single tones or buzzers) as previously thought, leading to applications in sound delivery for cinemas, concert venues and home entertainment.
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Influence of orthotropic properties on vibration of violin top plates
Masao Yokoyama, Riccardo Roberto De Lucia, Fabio Antonacci, Augusto Sarti
[Abstract]
In this paper, the mode vibrations of violin plates are
investigated by numerical simulations. The plate geometry is
scanned by means of a 3D laser scanner. Numerical
simulations are performed using Finite Element Methods, in
order to retrieve eigenmodes and mode shapes. The
orthotropic properties of wood such as maple and spruce are
introduced in the numerical simulation. We show that shapes
of the main modes calculated by numerical simulations are in
good agreement with those of a real violin. We discuss how
material properties influence the eigenmode frequencies and
shapes. Moreover, the acoustic radiation patterns of the plate
vibrational modes are obtained and compared with real data.
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Direct and inverse Hopf bifurcation in a neutral delay differential equation model of reed conical instrument
Tom Colinot, Louis Guillot, Jean Kergomard
[Abstract]
In conical musical instrument, the self-sustained oscillations appear at a blowing
pressure threshold. At this threshold, the mathematical model encounters a so-called
Hopf bifurcation. The bifurcation can be either direct or inverse which determines the
stability of the arising oscillation. It is well known that for cylindrical instruments
(clarinet-like instruments), the bifurcation is direct most of the time. In the case of
conical instruments, the nature of the bifurcation has been studied analytically before,
for instance using geometrical approximations based on cylinders. It is expected to find
an inverse bifurcation in some cases. The proposed study addresses the bifurcation of
an idealized model of reed instrument with a lossless conical resonator and a compliant
mouthpiece, using two methods : a direct time-integration algorithm and a Taylor
series based continuation method of periodic solutions, based on the harmonic balance
method. It allows to draw the whole branch of periodic solutions and to deduce the
type of Hopf bifurcation from a full diagram. Both direct and inverse bifurcations can
be found, depending on the value of the geometrical parameters of the instrument.
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On the cutoff frequency of conical woodwind instruments
Jean Kergomard, Erik Petersen, Tom Colinot, Philippe Guillemain, Michael Jousserand
[Abstract]
Since Arthur Benade, the cutoff frequency of woodwind instruments is a well-known concept that is a result of the acoustic regularity of the tonehole lattice. The distinction between the global cutoff frequency and the local cutoff frequencies was recently introduced. The first one can be approximately defined from the input impedance curve, while the second are related to the different cells of the lattice and can be theoretically determined from their individual geometries. From the local cutoffs, it is possible to check the acoustic regularity of an instrument, as was done for the clarinet, even if the geometry is irregular. The present work aims at finding a generalization for conical instruments. The existence of a global cutoff is evident in the impedance curves of oboes, bassoons, and saxophones published by Benade. However, the existence of periodicity is not obvious for a conical geometry because the cells are a priori very different, due to the taper. The present work uses a change of variables to investigate an acoustic periodicity for conical resonators and, for the example of a saxophone, determines the local cutoffs, compared to the global one.
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Reconsiderung the method of evaluation for tempo estimation
Madoka Okemoto, Masanobu Miura
[Abstract]
Tempo estimation for musical audio has been intensively
conducted in the field of MIR(Musical Information Retrieval).
Most popular methods for tempo estimation uses frequency
analysis for envelope of the audio to obtain tempo from
power spectrum of the Fourier transform of the envelope. A
method is implemented to estimate local tempo for the
audio of tempo-variated music. To estimate it, the authors’
former study employs conventional Zero-padding, in order to
obtain high resolution of tempo estimation. The resultant
accuracy of tempo estimation is, however, not so high
(37.6%) compared to their global tempo estimation’s result
(83%). Possible reason of it is the difficulty to have
appropriate length of time window, since the desired length
of the window depends on the cycle of beats. Thus, this
report proposes a novel method for local tempo estimation
for tempo-variated musical audio. The authors propose to
use a certain set of time window length for the musical
audio, to obtain frequencies from power spectrum, and then
estimate tempo by integrating the results of the Fourier
analysis for the signals from different time windows. The
authors will share the result of their proposing method for
tempo-variated music with audience of conference.
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Differences and similarities in the production mechanism of reeds, brass, and voice: the source-filter viewpoint
Bernd Kröger
[Abstract]
The acoustic sound production mechanisms of reed instruments,
brass instruments as well as of the human voice (singing and
speech) are already well investigated. In this paper we try to
compare these mechanisms from three viewpoints: (i) sound
source (reeds, lips, vocal folds), (ii) filter, i.e., sound transmission
and sound radiation via tube or pipe and (iii) aeroacoustic
interaction mechanisms between source and filter. It has been
found that the aeroacoustic interaction decreases from reeds
via brass to voice. Thus, as a first goal of this study the resulting
influence of this interaction on pitch as well as on sound quality
(spectral envelope) is investigated by using a computer
simulation model for sound production. Another important
feature by comparing a reed instrument like a bassoon with
voice is that reeds have a completely different mechanism in
producing so called formants in comparison to voice. Thus it is
the second goal of this study to investigate the differences in
formant generation between reeds and voice using our
computer simulation model.
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Sparse Modeling of Musical Instruments Sounds in Time-frequency Domain
Hiromu Ogi, Kohei Yatabe, Yasuhiro Oikawa, Yusuke Miyagi, Koji Oishi
[Abstract]
Sample-based synthesis is one of the most common synthesis methods of the digital synthesizer. It can synthesize musical instrument sounds well because it plays recorded sounds instead of generating sounds such as additive and frequency modulation syntheses. The recorded sounds are usually stored in the time domain. If the recorded sounds are stored in the time-frequency domain, sound processing can be simplified. Users can process audio signals intuitively by operating the time-frequency bins of the spectrograms. However, the representation of sounds in the time-frequency domain usually requires more memory than that in the time domain. In synthesizers, if much data is needed, hardware costs are increased correspondingly. Therefore, a technique to reduce the amount of data is required. In this paper, to reduce the amount of data, we introduce a sparse modeling technique for musical instrument sounds in the time-frequency domain. We propose an optimization algorithm of sparse modeling with four shrinkage operators. Numerical experiments show that the data quantity of the signals reduced by over 95 percent by using the technique.
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Effects of classical, film and video game music on creativity
Tomohiro Murata, Nao Hodoshima
[Abstract]
Classical music is commonly thought to enhance work performance.
Previous studies have reported that, compared to silence, listening to
classical music with a positive mood and high arousal enhances
creativity, but the effects for other music genres are unknown. Therefore,
in addition to classical music, we measured the effects of film and video
game music on creativity using the Alternative Uses Test (AUT). We also
measured mood changes throughout the experiment and the Affective
Value Scale of Music (AVSM) which is used to measure what type of
affective values music has. The mood changes and AVSM results were
used to analyze the effects of music on creativity. Twelve participants
were subjected to two conditions for 15 minutes each, silence and one
of the three types of music. The results showed that there was no
significant difference between with music and silence, as well as among
the three music types. Therefore, it is difficult to say that creativity can be
enhanced by listening to music. However, the number of samples for
each condition is insufficient to draw such a conclusion, and further
studies are needed.
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Numerical Approach for Aerodynamics around a tone hole with a moving pad: an example solving moving boundary problems with topological change
Taizo Kobayashi, Daiki Wakasa, Sho Iwagami, Toshiya Takami, Kin'Ya Takahashi
[Abstract]
In this paper, we discuss the numerical reproducibility of the
compressible fluid behavior around a tone hole of woodwind
instruments by using compressible Large Eddy Simulation
(LES). In particular, we focus on the situation that the tone
hole is opened and closed with moving a pad above the tone
hole and the change of the pitch occurs. From the numerical
point of view, this is a moving boundary problem with
topological change. Our numerical models are of 2D and 3D
analogs of Keefe’s experimental model, for which a tone hole
is placed at the center of the resonance tube. To reproduce
the opening and closing the tone hole, the pad should be
moved continuously and we developed a method solving this
problem. As a result, the process of ”open - close - open” in
order is completely reproduced. Namely, the change of the
pitch with opening and closing the tone hole is well
reproduced, and the details of complex vortices near the tone
holes can be clarified by these numerical calculations. Our
numerical results are in agreement with Keefe’s experiment
results. Therefore, we solved the moving boundary problem
with topology change under the situation of acoustics of
fluid-structure interactions.
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Mridangam shell-cavity analytical acoustic model for shape sensitivity studies
Harikanth Mohandas, Chandramouli Padmanabhan
[Abstract]
Mridangam is a double-sided percussion instrument used in
South Indian classical music concerts as an accompaniment.
It consists of a smaller, centrally loaded membrane on the
right and a larger membrane on the left. These are
assembled on a double-conical wooden shell which encloses
an acoustic cavity. The effect of the shape of the cavity on
the music generated is of interest. To achieve this, a one-
dimensional analytical model based on a diverging-
converging conical horn with short cylindrical ends is
developed. The first six axial natural frequencies and modes
are computed from this model and validated by comparing
with a 3D finite element model. The effect of the horn shape
on the acoustic characteristics is shown to be not significant
for moderate changes to the geometry. Further, the cavity
response to membrane vibration is obtained by applying a
boundary excitation to the analytical model.
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Reproduction of shakuhachi from X-ray CT images by additive manufacturing
Arisa Kuramoto, Yoshinori Takahashi, Akisato Mizuno
[Abstract]
The shakuhachi that is Japanese traditional wind instrument is made
of bamboo, and it has a different timbre for each instrument. There
are valuable shakuhachi created over 500 years ago and appropriate
protection is desired. The purpose of this study is to make it possible
to remake the shakuhachi from the shape information stored in
advance as a preparation for the situation where the instrument is
lost due to unexpected disasters.
Restoration a shakuhachi by additive manufacturing (3D printing)
has been attempted based on the X-ray images taken from two
directions. It has been confirmed that a shakuhachi with the feeling
of playing the original shakuhachi can be remade. However, there
was a problem of the appearance was quite different, since the
irregularities on the surface of bamboo could not be reproduced with
images only two directions. This study created a 3D model
introducing the adjustment processing of the contour of the X-ray CT
image, and attempted to remake a shakuhachi with the shape even
closer to the original shakuhachi using the 3D printing.
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The use of Nested Helmholtz Resonators for Broadband Low Frequency Sound Absorption
Alexander James Dell
[Abstract]
A theoretical and numerical study of sound propagation in periodically spaced, nested, systems of Helmholtz resonators is conducted. Incrementally varied geometrical properties of the nested resonators makes it possible to generate low frequency resonances. Using the transfer matrix method and the JCA approach to the modelling of visco-thermal losses, an analytical formulation for the acoustic properties of the ënested’ configuration is proposed. An optimization method is developed to tune the system for the absorption of sound waves at specific frequency ranges. Through this study it is shown that using an optimized configuration of nested resonators at the low frequency range, broadband sound absorption can be achieved at sub-wavelength sample thicknesses. The results of the analytical study are validated through numerical and experimental means.
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Investigation of Elastic Surface and Edge Modes in Finite-Size Three-Dimensional Phononic Crystals
Tian-Xue Ma, Quan-Shui Fan, Chuanzeng Zhang, Yue-Sheng Wang
[Abstract]
Three-dimensional (3D) phononic crystals exhibit the so-called
3D phononic bandgaps, which are frequency ranges where the
propagation of elastic or acoustic waves is prohibited along all
spatial directions. Phononic crystals have various potential
applications, such as vibration suppression, noise isolation,
waveguiding and filtering, etc.
By appropriately terminating the 3D phononic crystals, the
elastic waves can be guided on the surface and/or the edge of
the finite-size crystals. In this work, the 3D-printed phononic
crystal samples are experimentally investigated to assess the
transmission spectra of the surface and edge modes in the
finite-size phononic structures. And the experimental results show
a good agreement with the theoretical predictions. The surface
and edge modes may open up new avenues for the
manipulation of elastic waves in periodic structures, and exhibit
great potential for designing complex and compact phononic
circuits.
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Chessboard Design of Pyramid-Core Sandwich Structures
Zhengyang Li, Tian-Xue Ma, Fengming Li, Chuanzeng Zhang
[Abstract]
Lightweight materials and structures having high stiffness
and strength but low density inspired by nature have been
developed and applied since many years. Sandwich
structures are one special kind of the most famous
lightweight structures, which have been widely used in air
crafts, automobiles and civil engineering. Despite the high
stiffness-to-density ratio, it is still rather difficult to control
acoustic or elastic waves in sandwich structures. Recently, investigations on wave propagation in periodic
structures, known as the phononic crystals, have been
received considerable attention. The most important
characteristic of periodic structures is the so-called band-
gap, the frequency range in which the propagation of
elastic or acoustic waves is prohibited. The generating
mechanisms of bandgaps are Bragg scattering and local
resonances. Thus, numerous efforts have been made by
using both mechanisms of the bandgaps to efficiently
control the elastic wave propagation in the sandwich
structures. In this study, we develop a chessboard design
of pyramid-core sandwich structures which improves the
wave attenuation property significantly but maintains the
high stiffness-to-density ratio.
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A monostable acoustic metamaterial for broadband low frequency sound absorption
Xianhui Li, Tuo Xing, Junjuan Zhao, Xiaoling Gai
[Abstract]
This paper presents a monostable membrane-type acoustic
metamaterial (MAM) for achieving a broadband sound
absorption in low frequencies. The proposed monostable
MAM is
realized by placing a prototypic MAM with a magnetic proof
mass in a symmetric magnetic field. An equivalent circuit
model
of such a system is presented. It is shown that the monostable
MAM exhibits rich nonlinear dynamic features. The sound
absorption peak significantly shifts downwards with the
increasing magnetic field. Frequency sweeping test clearly
shows the softening hysteresis phenomenon. To quantify the
sound absorption performance of the nonlinear system, an
energy index is proposed and indirectly obtained from the
velocity of the proof mass measured by a laser vibrometer.
Experiments demonstrate that the proposed design can
realize a
broader sound absorption bandwidth than its linear
counterparts
in the low frequency range.
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Validity Assessments of Optical Transfer Function Measurements for Technically Premixed Flames
Audrey Blondé, Giacomo Bonciolini, Nicolas Noiray
[Abstract]
When analysing thermo-acoustic systems, flame transfer-
functions are an essential tool for modelling the feedback of a
flame to acoustic perturbations. This paper compares optical
and fully acoustic methods for measuring such transfer
functions, when progressively switching from fully- to
technically-premixed combustion in an atmospheric lab-scale
combustor. Optical methods relate the chemiluminescence of
the flame to the acoustic forcing in order to extract the flame
response at different frequencies. In the technically-premixed
scenario, the validity of the integrated chemiluminescence as
a proxy of the heat release rate is jeopardized and it is usually
recommended to adopt the fully acoustic method. The
present paper illustrates the progressive deviation of the two
approaches when the degree of unpremixedness is increased
and provides a systematic comparison of the two approaches.
This assessment is of utmost importance as flame transfer
functions are the core element of linear models used for
thermoacoustic stability assessment, such as lumped network
models or Helmholtz solvers.
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Meniere's Disease Prognosis by Learning from Transient- Evoked Otoacoustic Emission Signals
Sheng-Lun Kao, Han-Wen Lien, Tzu-Chi Liu, Hau-Tieng Wu, Te-Yung Fang, Pa-Chun Wang, Yi-Wen Liu
[Abstract]
Accurate prognosis of Meniere’s disease (MD) is difficult.
The aim of this study is to treat it as a machine-learning
problem through the analysis of transient-evoked (TE)
otoacoustic emission (OAE) data obtained from MD
patients. Thirty-three patients who received treatment
were recruited, and their distortion-product (DP) OAE,
TEOAE, as well as pure-tone audiograms were taken
longitudinally up to 6 months after being diagnosed with
MD. By hindsight, the patients were separated into two
groups: those whose outer hair cell (OHC) functions
eventually recovered, and those that did not. TEOAE
signals between 2.5-20 ms were dimension-reduced via
principal component analysis, and binary classification
was performed via the support vector machine. Through
cross-validation, we demonstrate that the accuracy of
prognosis can reach >80% based on data obtained at the
first visit. Further analysis also shows that the TEOAE group
delay at 1k and 2k Hz tend to be longer for the group of
ears that eventually recovered their OHC functions. The
group delay can further be compared between the MD-
affected ear and the opposite ear. The present results
suggest that TEOAE signals provide abundant information
for the prognosis of MD and the information could be
extracted by applying machine-learning techniques.
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Tonotopic Sensitivity to Supra-Threshold Hearing Deficits of the Envelope Following Response Evoked by Broadband Stimuli
Sarineh Keshishzadeh, Viacheslav Vasilkov, Sarah Verhulst
[Abstract]
Scalp-recorded electrophysiological Envelope Following Responses (EFRs) evoked by modulated sounds have recently gained popularity as an objective measure of supra-threshold hearing deficits (e.g., as caused by synaptopathy). To explore the frequency specificity of EFRs in diagnosing deficits, we recorded EFRs to sinusoidally amplitude-modulated (AM=120Hz) white noise carriers with bandwidths of [0.25-22], [0.5-22], [1-22], [2-22] and [4-22] kHz in two participant groups: A young normal-hearing control group (N=16) and a group with self-reported hearing difficulties in noisy listening environments (N=9). Despite individual variability in EFR strength, the group-means revealed minor differences across the first three bandwidth conditions. The limited contribution of frequency channels below 2kHz to the EFR was confirmed through EFR simulations from a biophysically-inspired model of the human auditory periphery. Further investigation showed that the minor low-frequency contribution stems from the lack of AM coding in individual low-frequency channels, although the overall spike-rate increases during stimulation. In conclusion, despite the broadband cochlear excitation, the broadband EFR mostly reflects summed AM coding strength in frequency channels above 2kHz. In addition, the model-based approach provides a promising way to further explore the influence of different stimulus characteristics in capturing individual differences in low-frequency supra-threshold hearing deficits.
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Influence caused by placement of a bone-conducted vibrator on sound transmission
Xiuyuan Qin, Sho Otsuka, Seiji Nakagawa
[Abstract]
Bone conduction (BC) is another way of sound transmission for our
hearing perception besides air conduction. It conducts sound through
skin and skull, bypassing part of the ear canal. The position where
the vibrator is put on one’s head makes a difference in BC
transmission. Although standards of bone conduction were
established on the position of the mastoid (the prominence of the
skull behind the ear), nowadays manufacturers favor bone-
conducted headphone whose vibrator is put on the position of the
condyle (the articular process of the ramus of the mandible bone, in
front of the ear). However, there are not enough data in details for
the condyle. In this research, hearing thresholds and ear-canal sound
pressure of bone conduction on the condyle and the mastoid were
measured, and a comparison between them was made to investigate
the difference of BC transmission between the two positions, followed
by a discussion on the influence of placement on bone conduction.
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The Pitch of Synchronized Spontaneous Otoacoustic Emission Does Not Sound Familiar to Ears that Emit It
Tsung-Tai Liao, Ching-Yun Hsu, Han-Wen Lien, Yi-Wen Liu
[Abstract]
Spontaneous otoacoustic emissions (SOAEs) are sounds that
originate from the cochlea and escape the ear. Computer
simulation predicts that, given the typical sound pressure level
of SOAE measured in the ear canal, the vibration on the basilar
membrane should be in the audible range. The aim of this study
is to find out whether humans can hear this vibration in quiet.
Sixty-three subjects were recruited and synchronized SOAE
(SSOAE) were found in 50 ears. These subjects were instructed to
spend extra time quietly while paying attention to “sounds in
the ear” (SinE). Next, each subject listened to five randomized
tones, and was forced to select the one that sounded closest to
the SinE. The frequencies of the tones were 88%, 94%, 100%,
106%, and 112% times the frequency of the largest peak in the
SSOAE spectrum, respectively. If a subject had SSOAE in both
ears, the procedure was repeated for the two ears. Results show
that the five choices received 11, 12, 8, 9, and 10 votes,
respectively, so none of them was significantly higher than by
chance. Hence, the present study falls short of supporting the
theoretical prediction that SOAE is audible to the ear that emits
it.
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Assessment of adjustment to electrical threshold (T) level and electrical stimulation rate on IDL and AM detection at soft presentation level in adult CI users
Terry Nunn, Tim Green, Dan Jiang, Patrick Boyle, Deborah Vickers
[Abstract]
Cochlear Implant (CI) users often show poorer speech
perception at low intensity levels.
Very little is understood about the influence of sound
processor fitting parameters on the discrimination of low
intensity acoustic cues. A pilot study was conducted to evaluate the influence of
electrical stimulation rate and the setting of electrical
threshold (T) Level on intensity difference limens (IDLs) and
amplitude modulation (AM) discrimination.
Participants were thirteen adult experienced Advanced
Bionics CI users. The frequency of the acoustic sinusoidal
stimuli was set to 1076Hz with the intention of stimulating
electrode seven. Nine experimental maps were created with
T levels set at: the perceptual threshold of audibility;
perceptually equivalent to ëvery soft’ and 10% of the
perceptual equivalent to ëmost comfortable loudness’. Rates
of electrical stimulation used were 900, 1800 and 2700 pulses-
per-second.
Adaptive 2I-2AFC IDL and AM discrimination tasks were
presented via headphones at soft presentation levels (48 and
54dBSPL for IDL and AM tasks respectively).
The influence of T Level and stimulation rate adjustments on
IDL and AM detection have been explored with the intention
of understanding the impact of altering individual parameter
settings to optimise detection of acoustic cues at low
presentation levels.
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Binaural Sound Localisation Directly From The Raw Waveform
Ning Ma, Paolo Vecchiotti, Guy Brown
[Abstract]
It is well known that binaural sound localisation is largely based on
cues such as the interaural time difference (ITD) and
level difference (ILD). These cues encode a location in terms of the
difference between the left and right ears in both arrival time and
magnitude. In addition, the outer ear, together with the head,
shoulders and torso, form direction-selective filters, whose
resonances impose direction-specific patterns into monaural spectral
cues. Conventional machine systems for sound localisation attempt to
explicitly extract ITD and ILD features. An alternative approach is to
estimate the azimuth of a source directly from the waveform arriving
at each ear. Here, a convolutional neural network (CNN) is employed
to implicitly extract features from the waveform that are suitable for
localisation. A convolutional layer with 2-dimensional kernels extracts
binaural features that are similar to ITDs and ILDs, but have
advantages in terms of robustness under reverberant conditions.
More importantly, the CNN allows binaural cues and monaural
spectral cues to be combined effectively. Our evaluation shows that
the system is able to accurately estimate the location of a sound
source directly from the raw waveform, even in challenging
reverberant conditions, and significantly improves upon the
performance of previous approaches.
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Predicting binaural effects in perceived listening effort
Jan Rennies-Hochmuth, Christopher Hauth, Thomas Brand
[Abstract]
Recent studies measured perceived listening effort in
addition to speech intelligibility. The main interest in listening
effort is because it allows assessing differences between
listening conditions at which intelligibility is at ceiling. In this
light, it would be desirable to have quantitative prediction
models to instrumentally compute listening effort. However,
previous studies investigating predictors of perceived
listening effort have focused on monaural conditions, thus
neglecting the important advantage humans can obtain
when target and maskers are spatially separated. This
contribution therefore exploits a recent data set of binaural
listening effort as a basis for extending and testing a binaural
prediction model. The front-end of the model is the
equalization-cancelation stage to model binaural
processing. The back-end is the speech intelligibility index
(SII). The presented model extension consists of deriving a
mapping function relating SII and perceived listening effort
for a diotic, anechoic reference condition, and then using
the same mapping function to predict all other conditions.
The results show that this relatively simple approach can
predict the experimental data with good accuracy. In
particular, it generalizes well to reverberant conditions as
well as binaural unmasking conditions, indicating that this
model approach may be a useful tool for assessing speech
perception.
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Spatial cue distortions within a virtualized sound field caused by an additional listener
Sergio Luiz Aguirre, Lars Bramsløw, Thomas Lunner, William Mcallister Whitmer
[Abstract]
Realistically, we are rarely alone in a central position with
respect to our acoustic environment, yet virtual sound fields are
usually evaluated in this manner. Sound presentation with more
than one person present using sound source virtualization can
be useful to invoke natural behaviors in auditory research.
Interaural time and level differences (ITDs and ILDs, respectively)
were measured on a symmetric mannequin (HATS). Sound
sources were virtualized using vector-based amplitude panning
and presented from a horizontal ring of 24 loudspeakers. The
influence of a second listener was simulated by positioning a
second mannequin (KEMAR) along HAT’s midcoronal plane (i.e.,
shoulder-to-shoulder). ITDs and ILDs were measured with HATS
centered and KEMAR 0.5-1.0 m to the right of center, or with
HATS and KEMAR 0.25-0.75 m to the left and right of center.
Results were compared to HATS alone. When HATS was
centered in the soundfield, the spatial cue distortions are small,
independent of KEMAR’s position. When both listeners are off-
center, there are large distortions in cues. These results confirm
difficulties in virtualizing sound sources for listeners outside of the
”sweet spot.” However, for a listener in the center, presentation
with an additional listener present is feasible for future hearing
research.
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Evaluation of Auditory Reality and Hearing Aids Using an Ecological Momentary Assessment (EMA) Approach
Niels Søgaard Jensen, Ole Hau, Dina Lelic, Petra Herrlin, Florian Wolters, Karolina Smeds
[Abstract]
Knowledge about the variety of acoustic environments
experienced by an individual (the auditory reality) and how the
individual listening experience is affected by hearing loss and
hearing aids is important when developing hearing solutions for
people with hearing loss. Acquiring this type of knowledge calls
for an assessment method that can be applied in real time while
a listener experiences a specific listening situation in real life. This
ability is a characteristic of the Ecological Momentary
Assessment (EMA) methodology. In this presentation, we report
on a study in which 16 participants with hearing loss used a
newly developed EMA smartphone app to answer questions
about the current listening situation (location, listening intention
and task, background noise, importance, frequency, difficulty)
and to compare two different hearing-aid settings in that
situation. The participants answered the questions when
prompted by the app (according to a predetermined time
schedule) or when initiating the app themselves. The
participants also evaluated the EMA app. Besides responses
from the participants, the app also collected data from the
hearing aids during the assessments. We will present the main
results from the study and discuss potential benefits as well as
issues of using the EMA approach in hearing research.
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3D Localization of Speech by Mildly and Moderately Hearing-Impaired Persons in Ecological Environments
Laurent S. R. Simon, Andrea Kegel, Hannes Wüthrich, Norbert Dillier
[Abstract]
Studies about 3D audio perception with hearing devices are sparse. There are a number of studies on localization on the horizontal plane, but none about localization in elevation. In this study, 20 hearing-impaired subjects took part in a 3D localization test. Participants had to point their head in the direction of various target stimuli. These stimuli were generated by convolving a speech signal with 4th HOA room impulse responses recorded with an Eigenmike. 4th order HOA background audio recordings were played in addition to the target in order to reproduce realistically difficult environments. In total, 32 different directions were generated for both a noisy restaurant and a busy street background sounds. Three different hearing device beamformers were compared along with an unaided condition.
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Integration of speech information (or not) across electric and acoustic modes in hearing impaired listeners.
Bob McMurray, Michael Seedorff
[Abstract]
Many people with hearing loss use two cochlear implants (bilateral CIs), or combine a CI with a hearing aid (bimodal). While the addition of a second CI or acoustic input improves speech perception (particularly in noise). However, it is not clear how listeners fuse these disparate inputs. We used a duplex perception paradigm to investigate this issue. Listeners heard the four corner vowels from six talkers. In the duplex condition, the first formant was presented to the CI, and the second to the other CI or hearing aid (and vice versa). In normal listeners (N=14), accuracy under duplex presentation (across ears; M=.80) did not differ from combined presentation (both formants, both ears, M=.76, p=.14), but performance dropped to near chance with isolated formants (M=.40, p<.001). CI users (Bilateral: N=7; bimodal: N=11; single-side-of-deafness [SSD]: N=7) also performed well in combined listening (Bilateral: M=.72, Bimodal: M=.70, SSD: M=.83), and poorly for isolated formants (Bilateral: M=.42, Bimodal: M=.39, SSD: M=.41). However, duplex presentation was much poorer than combined (Bilateral: M=.54, p=.0078; Bimodal: M=.55, p<.001; SSD: Duplex: .48, p<.001). Thus, CI users may not fuse inputs well at an auditory level, and the benefits of acoustic+electric hearing may derive from central integration.
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Glimpsed Periodicity Features and Recursive Bayesian Estimation for modeling attentive voice tracking
Joanna Luberadzka, Hendrik Kayser, Volker Hohmann
[Abstract]
Computational models are a way of approaching research
questions related to auditory perception. One relevant
question is how we are able to follow and understand speech
in complex acoustic scenes. Previous studies suggest that for
tracking a speaker in such conditions, humans use (1)
sparse, speaker-related bits of robust information - ’auditory
glimpses’ and (2) a mechanism of predictive coding with a
movable locus of attention. The goal of the present study is
to develop a computational model for attentive tracking of
voices, which takes these two aspects into account. We
model auditory glimpses using Glimpsed Periodicity Features,
and predictive coding using Recursive Bayesian Estimation.
We assume that perception is organized into an attended
foreground and unattended background. We propose parallel
particle filters - one for each category - to track the
concurrent events. In this approach, each incoming glimpse
is associated with either foreground or background based on
accumulated evidence.
Simulations with artificially generated data of a ’glimpsing’
nature (sparse, robust) showed that this approach is suitable
to track multidimensional parameter trajectories of two
competing sources. This suggests the potential of the
method to track simultaneously active voices based on the
Glimpsed Periodicity Features.
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Prediction of Speech Intelligibility Based on Deep Machine Listening: Influence of Training Data and Simulation of Hearing Impairment
Jana Roßbach, Birger Kollmeier, Bernd T. Meyer
[Abstract]
An accurate prediction of speech intelligibility (SI) is a useful tool for the development of speech enhancement algorithms; if a model is blind it could also serve as real-time SI monitor in future hearing devices. A blind processing strategy implies that speech or noise references are not used for predictions. Previous work [Spille et al. (2018), Comp. Speech & Lang. doi:10.1016/j.csl.2017.10.004] introduced an accurate SI model using a deep neural network (DNN), which however had access to a priori information about the noise signal. This current work investigates model predictions for mismatched noise with the speech reception threshold (SRT) as target measure. Moreover, we simulate hearing impairment to explore if SI can also be predicted for hearing-impaired listeners. For similar but mismatched noises, we obtain an SRT RMSE of 1.5 dB in contrast to 1.9 dB obtained with the previous model. Further, several baseline models (SII, ESII, STOI, and mr-sEPSM) are outperformed. The hearing-impaired model reached an RMS error of 4.2 dB which is less accurate as for normal-hearing listeners. This result has been expected because of the larger variety in the SI of hearing-impaired listeners.
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Spatial Speech Intelligibility Map Rendering for Hearing Device Users with TASCAR, openMHA, and FADE
Marc René Schädler
[Abstract]
Listeners interact with a surrounding sound field by turning their heads to optimize its perception according to complex criteria. With impaired hearing, they perceive the sound field differently. In addition, hearing devices might continuously take decisions for their users to assure sufficient speech recognition performance by providing level and spatially dependent amplification. To build a model of this complex situation and better understand the interactions between the involved entities, sound fields are modeled with the Toolbox for Acoustic Scene Creation And Rendering (TASCAR), hearing devices with the open Master Hearing Aid (openMHA) and listeners with the Framework for Auditory Discrimination Experiments (FADE). The model simulates the transmission of speech information though sound field, hearing device, and impaired auditory system with the German matrix sentence test and an automatic speech recognizer. For a given position and orientation of a listener, speech reception thresholds (SRTs) for virtual talkers at arbitrary positions in a room are predicted and depicted as a map. The current implementation shows better-ear binaural integration patterns. The results suggest to interpret the SRTs relative to speech levels that normal-hearing interlocutors would choose in conversations, which in turn depend on the sound level at the position of the interlocutor.
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Effects of Audiovisual Expression of Emotion on Age Perception
Sumi Shigeno
[Abstract]
In this research, we examine the effects of emotional expression (neutral emotion, happiness, and sadness) on the perception of a speaker’s age in a multimodal condition. Although many researchers have explored the effect of aging on cognitive mechanisms, effects of emotion on a speaker’s age perception have not been explored in the view of the differences among the modalities. In auditory perception of age, Shigeno (2016) revealed that the perception of age is biased by the emotional state of speaker. On the other hand, in visual perception of age, the results are not uniform; a smiling face photo makes a speaker look younger (Wang et al., 2015) or look older (Ganel, 2015). The discrepant findings among researchers might be because facial expression photographs capture just the moment of emotional climax of a face. The present study used emotional videos instead of photographs. An experiment was conducted to explore whether audiovisual happy expressions by face and voice give younger impression to perceivers than neutral emotion or sadness. Sixty participants identified speakers’ ages who were 24-75 years. The results showed that facial expression did not promote the effect of vocal emotion. We discussed the integration process of vocal and facial emotions.
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Reinforced statistical learning of auditory categories: A preliminary report of cognitive, cortical and computational mechanisms
Bob McMurray, Samantha Chiu, John Freeman
[Abstract]
Learning to categorize speech sounds is crucial for language development, and a persistent struggle for second language learners. Most research has examined supervised auditory category learning in which learners receive feedback. This offers a good model for second language acquisition. However, work on first language acquisition suggests people also acquire auditory categories by tracking the statistics of the input without feedback. However, few studies examine how these forms of learning work together and little cognitive neuroscience examines unsupervised learning. We present results from a new paradigm in which learners are first exposed to statistically structured stimuli in an unsupervised paradigm and then receive supervised training. Results of 136 listeners suggests the unsupervised phase benefits later supervised learning, but only in limited conditions, which are not predicted by work on auditory plasticity in animal models or by some models of human learning. I also present data from electro-corticography recordings made directly from the brains of epilepsy patients. A machine learning analysis on these recording suggest some surprising ways in which auditory cortical representations change with learning. Finally, I present a new theoretical approach that suggests unsupervised learning may have varied effects on the auditory processing system beyond simply learning categories.
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Effects of vibrator placement in the auricle on cartilage- conduction hearing: Assessments of detection threshold and ear canal sound pressure
Toru Jitsukawa, Sho Otsuka, Seiji Nakagawa
[Abstract]
Bone-conduction has been used as hearing aids for the
conductive hearing impairment, but there was a problem in the
wearability of the transducer. To solve the problem, cartilage-
conduction (CC) that presents the transducer to the auricular
cartilage has been proposed and applied to hearing aids and
smartphones. In the CC, the hearing changes depending on
the presentation site within the auricle, the contact area of the
transducer and the contact pressure, but effects of such
conditions has not been studied. In this study, we observed
variations of the threshold and the sound pressure in the ear
canal (ECSP) depending on the position of the transducer within
the auricle. Tone bursts with frequencies from 250 to 8,000 Hz
were presented to the upper, middle and lower parts of the
auricle. Both measurements were also conducted under
conditions that the transducer did not contact the auricle. The
results showed that he threshold decreased and the ECSP
increased when presented at the middle and lower auricles at
lower frequencies (250 and 500 Hz). These results provide useful
information not only to optimize the devices using CC, but also
to elucidate the peripheral mechanism of the CC perception.
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Reversing Degraded Auditory Processing Using Targeted Plasticity
Yuko Tamaoki, Jonathan Riley, Michael Borland, Seth Hays, Crystal T Engineer, Michael Kilgard
[Abstract]
Hearing loss is a very common and often debilitating
neurological disorder, estimated to affect over 5% of the
population. A major source of disability related to hearing loss
stems from deficits in speech processing. Assistive devices, such
as hearing aids, can yield modest improvements in speech
perception, but these devices often provide inadequate utility
in complex, challenging environments. We have developed a
new approach to drive robust, specific plasticity that
substantially enhances recovery after neurological damage.
This strategy uses brief bursts of vagus nerve stimulation (VNS)
paired with sound presentation. Like humans, rats are
significantly impaired in their ability to accurately discriminate
speech sounds following intense noise exposure. Additionally,
noise trauma results in substantial maladaptive plasticity in
multiple auditory structures. In these experiments, we test the
hypothesis that VNS paired with sound therapy, 300 times per
day for 20 days, will reverse maladaptive plasticity and restore
auditory processing. Following the last day of VNS-sound pairing,
the reversal of maladaptive plasticity is tested through
behavioral discrimination ability, auditory brainstem responses
(ABR), and auditory cortex responses. The successful
development of adjuvant therapies to restore speech
processing has real potential to yield tangible benefits for
millions suffering from hearing loss.
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Demodulation Characteristics in Propagation Process of Amplitude-modulated Bone-conducted Ultrasound Presented to the Neck, Trunk and Arms
Koichiro Doi, Ogino Riki, Sho Otsuka, Seiji Nakagawa
[Abstract]
Bone-conducted ultrasound (BCU) is can be heard clearly and
can transmit speech information using amplitude-modulation
(AM). Additionally, BCU is perceived even when presented to
body parts distant from the head, like the neck, trunk, and arm.
It is expected that demodulated low-frequency components
appears by the nonlinearity existing in the human body when
amplitude-modulated BCU (AM-BCU) were presented to such
distal locations.
First, to elucidate demodulation characteristics in the
propagation process of such distantly-presented BCU, vibration
at the external auditory meatus was measured when AM-BCU
was presented to the neck, shoulder, upper limb, breastbone
and backbone. The result showed spectrum peaks
corresponding to the carrier frequency (30 kHz) and the
modulation frequency. The level of the peak of the carrier
tended to decrease depending on the distance between
stimulation and measurement points, however, the peak of the
modulator did not change by the distance.
Second, vibrations around the cartilages of the auricle, tragus
and articulations, that have strong nonlinearity, were measured.
The demodulation components were larger for the auricle and
tragus than the peripheral articulations.
These result suggests the demodulation mainly occurs in the
vicinity of the ear even when BCU is presented to distal body
parts.
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Targeted Neuroplasticity in Rat Primary Auditory Cortex with Vagus Nerve Stimulation and Near-Threshold Tones
Alan Michael Carroll
[Abstract]
Vagus nerve stimulation (VNS) is a method for driving
therapeutic, targeted neuroplasticity in clinical populations
suffering from tinnitus and stroke. VNS facilitates specific cortical
changes through the phasic release of plasticity promoting
neuromodulators simultaneously paired with delivery of a
sensory stimulus, such as tones or speech. Recent clinical
evidence and ongoing pre-clinical experiments in rats show that
VNS paired with near-threshold somatosensation of the
hand/paw can significantly reduce elevated sensory thresholds
resulting from neural injuries after only one week of therapy. A
possible explanation for this quick and robust recovery is that
VNS is more effective at driving neuroplasticity in cortical circuits
when paired with stimuli just above the response threshold of
neural receptive fields. To date, all auditory VNS therapies have
used stimuli considerably above auditory thresholds, potentially
diminishing the therapeutic effect of VNS-paired treatments. To
test the effectiveness of VNS-paired, near-threshold stimuli in
driving auditory neuroplasticity, unimpaired adult rats will
receive VNS repeatedly paired with the brief presentation of a
10 dB SPL 9 kHz tone for one week. A cortical map of receptive
field properties in primary auditory cortex will be made one day
later and compared to the maps of naïve rats.
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Evaluation of Propagation Characteristics of Bone- conducted Ultrasound Presented to the Neck, Trunk and Arms
Ogino Riki, Sho Otsuka, Seiji Nakagawa
[Abstract]
High-frequency sound above 20 kHz can be heard clearly
via
bone conduction (Bone-conducted ultrasound: BCU).
Additionally, BCU is perceived even when presented to
body
parts distant from the head, like neck, trunk, and arm, and
is
expected to be applied for novel devices that transmit
sound
selectively to persons who touch a vibrator by the arm or
so.
First, to elucidate propagation characteristics of such
distantly-
presented BCU, vibration in the ear canal was measured
using
an acceleration sensor when a 30-kHz tone was
presented to
the neck, arm, and trunk. A prominent spectrum peak
corresponding to the stimulus frequency was obtained.
Further,
the vibration tended to attenuate when the stimulus
points were
moved away from the head, whereas some exceptions
existed.
Second, vibrations of the upper limb were measured at 9
surface points lined up straight from the forearm to
shoulder
when BCU was presented to nearby the wrist. The
vibration
tended to attenuate depending on the distance from the
vibrator basically, however, some points with swollen
upper-arm
muscles showed larger amplitudes.
These results suggest that efficiency of the propagation
varies
depending on the stimulus points, and the upper-arm
muscles are suitable also for stimulus location of BCU.
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Correlative changes of medial olivocochlear bundle reflex and electroencephalographic activity during sleep
Yuto Suzuki, Sho Otsuka, Seiji Nakagawa
[Abstract]
Sensory systems through the ascending auditory pathway undergo changes that depend on the sleep or waking state of the brain. By contrast, it is not well understood how the efferent systems change their responses from wakefulness to sleep. The present study examined the effect of sleep on the medial olivocochlear reflex (MOCR), which is an a sound-activated feedback response that controls the input to the auditory system. The MOCR was assessed non-invasively in terms of the suppression of otoacoustic emissions (OAEs) induced by a contralateral acoustic stimulation. The amount of OAE suppression was monitored during sleep, and the sleep stage was assessed simultaneously by electroencephalographic (EEG). For a two-hour measurement period, the amount of the OAE suppression significantly fluctuated over time, and its pattern varied across individuals. There was a significant positive correlation between the time fluctuation of the OAE suppression and alpha-band EEG activity, which dramatically decreases during the transition between wakefulness to light sleep. In contrast, there was no link between the OAE suppression and delta-band EEG activity, which specifically appears during deep sleep. The results indicate that MOCR dynamically changes its activity during the transition from wakefulness to sleep but does not depend on sleep depth.
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On the frequency limit of interaural time difference sensitivity for pure tones
Helen Theresa Heinermann, Jonas Klug, Sven Herrmann, Go Ashida, Jörg Encke, Mathias Dietz
[Abstract]
Pure tones with different interaural time differences (ITDs) can be discriminated due to their differently lateralized perception. For young normal hearing listeners, previous studies have identified a frequency limit very close to 1400 Hz where an abrupt cliff-like increase in threshold ITD is observed. Published data only reports thresholds at certain correct rates but does not report the correct rates decline with increasing frequencies. Here, we report threshold ITDs as well as correct rates for a broader range of age and hearing status. We investigate if individual high-frequency limits much below 1400 Hz are still cliff-like or more gradual. Most existing models cannot account for the abrupt sensitivity decline without an explicit change of parametrization between 1400 and 1500 Hz. A physiologically inspired model of the auditory pathway suggests that the duration of the excitatory input to the medial superior olive causes the steep upper frequency limit in young normal hearing listeners. Individual data can be fitted by varying the excitatory post synaptic potential duration along with peripheral parameters.
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Multivariate Statistical Analysis for Acoustical Characteristics of the Onomatopoeic Expression on Tinnitus
Takahiro Tamesue
[Abstract]
It is difficult to measure tinnitus objectively and quantitatively. Subjective expression with onomatopoeia is adopted as one of the Japanese standard test methods used to assess tinnitus. In this test method, whether tinnitus resembles pure tone or narrow band noise, and, frequency of the pure tone and bandwidth of the noise, are determined. This study focused on the subjective expression with onomatopoeia about the tinnitus the patient complained. Techniques of multivariate statistical analysis are used for evaluation of timbre on the subjective expression with onomatopoeia about the tinnitus. Specifically, first, the Semantic Differential (SD) method with 5-point bipolar adjectival scale is employed to evaluate timbre on the subjective expression with onomatopoeia about the tinnitus. Next, 3 factors related to timbre on the subjective expression with onomatopoeia were extracted by use of factor analysis. Then, subjective expression with onomatopoeia about the tinnitus was classified by factor score.
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Effects of a preceding sound on medial olivocohlear bundle reflex as a function of the preceding time interval
Sho Otsuka, Seiji Nakagawa, Shigeto Furukawa
[Abstract]
Our previous study reported that a preceding sound, regardless whether it induced an medial olivocochlear bundle reflex (MOCR) by itself, expedites MOCR, and a 60-dB-SPL preceding-noise, which itself induces an MOCR, not only expedites but also enhances the MOCR. The current study compares the dependency of the enhancing and expediting effects on the preceding time interval. The MOCR is induced by sounds and exert an inhibitory effect on the outer hair cells. The suppressive effect was assessed non-invasively in terms of the suppression of otoacoustic emissions (OAEs) induced by a contralateral acoustic stimulation (CAS). A 60-dB-SPL noise was used as a preceding sound, and the inter-stimulus interval (ISI) between the preceding sound and the CAS was varied from 0.5 to 2 s. The MOCR strength decreased as the ISI became larger and reached the same level as without a preceding sound within 2 s of ISI. By contrast, the onset delay of the MOCR did not depend on the ISI and was still significantly smaller than without a preceding sound even at 2 s of ISI. The difference in the dependency of the preceding time interval implicates that the preceding-sound-induced enhancing and expediting effects are underpinned by different mechanisms.
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Voice conversion model for estimation of transmission characteristics in auditory feedback
Shota Morita, Daiki Kawamoto, Teruki Toya
[Abstract]
We propose a voice conversion model for estimation of transmission characteristics in auditory feedback. Speakers feel sense of discomfort when they listen recorded speech sound of themselves. The phenomenon is caused by different transmission pathways. The recorded speech sound contains just air conducted sound from the speakers. On the other hand, the speech sound perceived by the speaker contains both air- and bone- conducted sounds in auditory feedback. A method for measuring transfer function and transmission characteristic in auditory feedback has never proposed, since direct observation of the sound in the inner ear is impossible. In addition, there is a problem that the sound source and listening positions cannot be underspecified. In this paper, we propose a voice conversion model from recorded speech sound for estimation of transmission characteristics in auditory feedback. The transmission characteristic in auditory feedback will be estimated by subtracting air conduction sound from converted voice. The converted voice was obtained by synthesizing air- conducted sound and bone- conducted sound with time delay, sound pressure level, and reverberation as parameters. Bone- conducted speech was obtained by filtering from air conducted sound.
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Spatial sound segregation in monaural listening condition
Daisuke Morikawa, Daiki Kojima, Tatsuya Hirahara
[Abstract]
A sound image flies around the head with the head rotation in monaural listening condition. The sound image moves greatly when the sound source moves from the open-ear side to the closed-ear side, and vice versa, with the head rotation. Actually, sounds presented at certain direction jumps from rear to front or from center to either side, and vice versa, and a listener perceives segregated sound images at certain head rotation angle. In this paper, we describe this perceptual phenomenon in detail based on a series of psychophysical experiments.
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Acuity of Spatial Stream Segregation along the Horizontal Azimuth with Non-individualized Head-Related Transfer Functions
Martha M Shiell, Elia Formisano
[Abstract]
Auditory spatial cues help the nervous system segregate
features of a soundwave into distinct streams. Although this
process benefits hearing, most basic research into auditory
spatial processing has focussed on understanding the neural
correlates of single sound locations rather than spatially-
separated concurrent sounds. To study the latter, we must allot
for changes in auditory spatial acuity along the horizontal
azimuth, and some evidence suggests that this relationship
differs for concurrent versus consecutive sounds. Here, we aim
to measure the change in spatial stream segregation along the
horizontal azimuth and evaluate the effectiveness of non-
individualized HRTFs, the most easily-accessed form of auditory
spatialization, in this procedure. We tested 16 normal-hearing
adults using anthropometrically-matched non-individualized
head-related transfer functions. We applied a spatial stream
segregation task where participants identified the rhythm of a
target stream presented with a spatially-separated masker.
Spatial separation varied according to an adaptive staircase
procedure, and thresholds were calculated both near the
midline and in the far left periphery. Thresholds were stable over
two separate days of testing, and individual participants each
showed the expected positive relationship between azimuth
and threshold separation. These results will help scientists to
design and implement future investigations of spatial stream
segregation.
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HRTF and panning evaluations for binaural audio guidance
Sylvain Ferrand, François Alouges, Matthieu Aussal
[Abstract]
We develop a device to guide blind people using binaural sound obtained by HRTF convolutions and reproduced by headphone. A sound source is placed virtually in front of the person then allows him/her to be directed along a predefined path. This device combines a high-performance real-time localization device, a head tracker and a binaural engine. In order to obtain the best accuracy and user experience, we design different binaural filters (HRTFs) on the criteria of the azimuthal localization accuracy. We are doing it for both static and moving sound sources. Indeed, the question of the ability to quickly identify changes in the position of moving sounds sources has been little studied in the literature. We focus on the optimization of filters in the context of guidance (i. e. on the azimuthal plane and in the frontal zone), with various strategies: remapping of the HRTFs, reverberation optimization for externalization and HRTF selection. We subjectively evaluate these different filters on groups of subjects, both for static and moving sound sources and by combining sources and moving subject. On these experiments, users are equipped with headphones and tracking equipment. We also compare these results with those obtained with natural sounds emitted by loudspeakers.
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Feature Analysis of Sound Direction Perception Using Frequency Band-Limited Stimuli: Extension of a Directional Band Model
Michika Yamada, Fumikazu Saze, Toshiharu Horiuchi, Kan Okubo
[Abstract]
A directional band model of median-plane localization
has been developed. Perception of sound direction for a
narrow band noise in the median plane depends mainly
on the center frequency and bandwidth. In median-
plane localization, a frequency band limited stimulus gives
perception associated with a given direction. Earlier
studies were commonly implemented for the median
plane and binaural hearing. No directional band model
of monaural hearing, however, has been examined.
Moreover, none for another plane has been examined
sufficiently. For this research, we make an extension of a
directional band model: we examine features in a
different plane (e.g., the horizontal plane) and monaural
hearing. For this study, stimuli were applied from
loudspeakers on the upper hemisphere surface. Results of
experiments conducted for direction recognition
demonstrate that some frequency bands in monaural
hearing correspond to specific directions. Additionally, we
clarified features of the directional band model for
binaural hearing aside from the median plane, which
suggests the possibility of a directional band model of 3-D
space localization.
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Parametric measurement of the effects of relative loudness on the relative weights
Alexander Fischenich, Jan Hots, Jesko Verhey, Daniel Oberfeld-Twistel
[Abstract]
Loudness judgments of sounds varying in level across time show
a non-uniform temporal weighting, with increased weights
assigned to the beginning of the sound (primacy effect). In
addition, higher weights are observed for temporal components
that are higher in level than the remaining components
(loudness dominance). Two experiments were conducted in
which sounds consisting of 100-ms Gaussian wideband noise
segments with random level variations were presented and
either none, the first, or a central segment was amplified or
attenuated. In Experiment 1, the sounds consisted of four
segments that were separated by 500-ms gaps for which
previous experiments showed no primacy effect. In Experiment
2, four- or ten-segment sounds without gaps between the
segments were presented to examine the interaction between
the primacy effect and level dominance. As expected, for the
sounds with segments separated by gaps, no primacy effect
was observed, but weights for amplified segments were
increased and weights for attenuated segments were
decreased. For the sounds with contiguous segments, a primacy
effect as well as effects of relative segment level (similar to
those in Experiment 1) were found. The data indicate no
substantial interaction between the primacy effect and
loudness dominance.
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Evaluation of Frequency Resolution Characteristic of Cartilage- conduction Hearing using Physiological and Psychological Measurement
Gaik Sean Yap, Sho Otsuka, Masato Yumoto, Seiji Nakagawa
[Abstract]
In bone-conduction, a transducer needs to be clamped
strongly against the mastoid process of the skull and this has
severely caused pain and discomfort on users. To solve such
defects of bone-conduction, ”cartilage-conduction” has been
proposed and applied to several devices such as hearing aids
and smartphones. As the name suggests, cartilage-
conduction uses ear cartilage tissue as the main sound
transmission medium, together with skins and bones, by
bringing a transducer into contact with the ear pinna.
However, because of the nonlinearity of cartilage tissues,
cartilage-conduction is thought to have perceptual
characteristics dissimilar to the ordinary air-conduction and
conventional bone-conduction. Moreover, studies regarding
cartilage-conduction have been extremely limited and
quantitative understanding of its perceptual characteristics
have yet to be discovered. In this study, the frequency
resolution of cartilage conduction was evaluated, using a
neuromagnetic measurement (mismatch field measurement)
and a psychological measurement (frequency difference
limens measurement). The results were then compared with
air-conduction and conventional bone-conduction hearings.
Furthermore, in order to investigate the contribution on
perception of each propagation component in cartilage-
conducted sound, measurements under condition in which
the air-conducted component was blocked by an earplug
were also conducted.
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A preliminary study of desirable sound environments in hospital wards
Junko Matsumoto
[Abstract]
This study explored desirable sound environments of hospital wards. Participants were 111 chief nurses in hospital wards in Japan. They were asked about problems concerning sounds in hospital wards for patients and staffs, trials of improvement for problems of sounds in hospital wards, and desirable sound environments in hospital wards. It was found that staffs annoyed by alarms of medical equipment and noises when staffs carried medical equipment or a wagon, voices of staffs, and patients’ snores or some noises patients occurred. Participants felt that patients annoyed by alarm of medical equipment, television, voices and footsteps of staffs, and other patients’ snores or some noises they occurred. With respect to trials of improvement to sound environments in hospital wards, staffs had tried to stop alarms as possible, had recommended that patients used an earphone when watching television, had used background music, and had paid attention to voices and footsteps on their own. As a desirable sound environment of hospital wards, it was quiet at midnight but at daytime not too quiet environment and patients and staffs heard background music. These results suggest that staffs’ attention to sounds may be useful to make sound environments in hospital wards comfortable.
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Fourier Spectral Method for Acoustic Simulation with Domain Enclosed by Curved Boundary
Yu Kohase, Tsubasa Kusano, Kohei Yatabe, Yasuhiro Oikawa
[Abstract]
Curved geometries often make sound propagation complex. Such
complexity may cause acoustical problems including flutter eco and
sound focusing. When designing a room, acoustic simulation can be
helpful to prevent such problems. Fourier spectral method (FSM) is a
simulation method by approximation using the Fourier basis.
Although FSM has many advantages such as its high convergence
rate, the application range of the conventional FSM is limited to a
simply-shaped domain with a specific boundary condition.
In a realistic acoustics room setting, there exist a lot of room shapes
beyond the scope of FSM.
At the same time, the Fourier extension has been proposed for
approximating a function on a complicated domain by the Fourier
basis. It can be expected that the Fourier extension technique
expands the application range of FSM.
In this paper, we introduce the Fourier extension technique into FSM
for extending it to make a computational domain enclosed by curved
boundaries tractable.
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Differences in Sound Absorption of Samples with Periodic Porosity Produced using Various Additive Manufacturing Technologies
Tomasz G. Zielinski, Kamil C. Opiela, Piotr Pawlowski, Nicolas Dauchez, Thomas Boutin, John Kennedy, Daniel Trimble, Henry Rice
[Abstract]
With a rapid development of modern additive manufacturing
technologies it seems inevitable that they will sooner or later serve
for production of specific porous and meta-porous acoustic
treatments. Moreover, these new technologies are already being used
to manufacture original micro-geometric designs of sound absorbing
media in order to test microstructure-based effects, models and
hypothesis. In the view of these statements, this work reports
differences in acoustic absorption measured for porous specimens
which were produced from the same CAD-geometry model using
several additive manufacturing technologies and 3D-printers. A
specific periodic unit cell of open porosity was designed for the
purpose. The samples were measured acoustically in the impedance
tube and also subjected to a thorough microscopic survey in order to
check their quality and look for the discrepancy reasons. Finally, the
corresponding microstructure-based numerical analyses on the
periodic unit cell are also briefly discussed.
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Drinking Straws as a Broadband Sound Absorber
Ioanna Christia, Martin Tenpierik, Foteini Setaki, Marcel Bilow
[Abstract]
The range of products developed by the most common
absorption principles that fulfil high performance tuned to
specific frequencies, low price, sustainability and aesthetic
quality is often limited to porous materials. The quarter
wavelength tube principle may perform well, even in low
frequencies, but in a narrow frequency range. To widen the
range, multiple tubes can be tuned to multiple frequencies by
choosing combinations of lengths and diameters. If designed
properly, less material is needed thus lower price and tubes can
be easily re-used. Based on a research through design method
a more broadband sound absorber using drinking straws has
been developed. Measurements of the normal incidence sound
absorption coefficient were performed in an impedance tube
and compared to an analytical model based on the theory of
viscothermal wave propagation in prismatic tubes. Different
tube materials and geometries were examined regarding the
influence on their performance. The outcomes of this study led
to a sound absorber based on 19 different straw lengths that
absorbs well in the frequency range from 400Hz till 900Hz. The
importance of these absorbers can be high when it comes to
selective frequency performance that needs to be cost-
effective, sustainable and aesthetically pleasing.
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Study on the Acoustic Environment of Tibetan Buddhist Temples in Han Region
Ziqing Tang, Hui Xie, Fenghua Lu
[Abstract]
Buddhism, especially Tibetan Buddhism, has an important influence on Chinese social folk culture and particularly on people’s life. Their monasteries are mainly divided into plain monasteries and mountain monasteries according to different geographical features, but the type of acoustic landscape is still unknown. In order to explore the acoustic landscape of Tibetan Buddhism in the city, mining variety of sound sources, this paper takes two Tibetan Buddhist monasteries, Wutai mountain in Shanxi Buddha temple and Lama Temple in Beijing as the research objects, making investigation and analysis on the acoustic landscape elements of representatives positions, analyzing the field testing data, and the relationships between their typical spatial characteristics and the present situation of acoustic environment are discussed, besides, several reformative design suggestions are proposed. At last, the key nodes are modeled, calculated and simulated through the software Cadnaa, then some conclusions are put forward to.
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Crossover Frequency Estimation From Statistical Features of a Room Transfer Function
Marcio Henrique De Avelar Gomes, Paulo Bonifacio, Eric Brandão, Luis Henrique Santana, Elvis Bertoti, Rodrigo Catai, Hilbeth Azikri De Deus
[Abstract]
In certain practical situations one faces the problem of
choosing the most suitable approach to analyze the sound field
in a room. For acoustically small rooms, a modal approach is of
higher importance, as opposed to acoustically large rooms,
when statistical considerations provide more useful models. In
the context of reverberation rooms, an important issue to be
dealt with relates to the diffusion of the sound field. Although
the evaluation of how close the field is from being diffuse is
under discussion, it seems to be accepted that such a condition
isn’t reached under a crossover frequency, also referred to as
the ”Schroeder Frequency”. This paper presents the investigation
of procedures based on statistical features of rooms’ Transfer
Functions aiming the experimental determination of such a
frequency. Real and imaginary parts of transfer function are
analyzed with Kurtosis, and the phase with a proposed measure.
Results from measurements performed in a large (108.5 cubic
meter) and in a scaled (1.48 cubic meter) reverberation rooms
are presented, as well as results from analytical solution from
three shoebox rooms with the same volume, but different
symmetry degree. Both measures presented comparable
results, showing possible, and potentially more precise practical
approaches.
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The effect of reverberated speech on working memory: Toward an optimal balance of calmness and liveliness in libraries
Kazuma Shamoto, Hiroko Terasawa, Hiroshi Itsumura
[Abstract]
Although quiet sound environments have been idealized as
learning space,
the function of modern libraries include an active, social, and
interactive learning place for old and young people including
children.
We focus on reverberation and working memory and
investigate the effect of children’s conversation sound under
reverberation on working memory and the impression of the
sound.
We measured impulse responses at 2 libraries: a traditional
library (RT: 0.3s) and a modern library (RT: 1.0s).
Working memory capacity (WMC) was measured by Operation
Span Task, and compared for the following 4 sound conditions:
children’s conversation sound (anechoic condition), it
reverberated in a traditional library (traditional condition),
it
reverberated in a modern library (modern condition), and
silence.
Perticipants were 20 university students.
The result shows WMC is better in silence condition than in
anechoic condition.
Also, WMC tends to be better in modern condition than in
anechoinc condition and there were no significant differences
between modern condition and silence condition.
In addition, WMC is improved in conditions which perticipants
feels comfortable.
This result may means choosing learning space with our own
instincts can result in choosing learning space with good
working efficiency.
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Acoustic simulation of the sound quality of standardized classrooms in higher education institutions in Natal / RN
Luciana Da Rocha Alves, Bianca Carla Dantas De Araújo
[Abstract]
In architectural design of schools, architects usually
replicate one or more types of classrooms throughout the
building. However, few studies refer to the verification of
acoustic quality of classrooms, which can cause damages
in the teaching-learning process, in case there is low
quality, commonly found in Brazilian schools. This study
aimed to propose basic guidelines for architectural
acoustics of classrooms in two universities in Rio Grande do
Norte, Brazil, considering the replication of solutions in
standard rooms. The methodology was a simulation of
acoustic quality of standardized classrooms with
alterations on ceiling and posterior wall through the
software EASE® in comparison with in situ measurements
results, used as calibration control of the computational
model. The best solution, among the proposals, depended
on the shape and proportion of each classroom, varying
between different percentages of absorbent material
applied in the ceiling and in the posterior wall. The
Definition (D50) was better distributed in all rooms with a
100% absorbent ceiling, however, the other parameters of
intelligibility presented variables according to the shape of
the room and the application of absorbent material. From
the simulations, some guidelines for the architectural
design of these classrooms were purposed in terms of
acoustic quality.
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Audible room acoustic differences of public preschools in the Gothenburg area
Julia Winroth, Denice Perkhed, Erling Nilsson, Kerstin Persson Waye, Ann-Charlotte Thysell
[Abstract]
Previous investigations have found high equivalent and maximum noise levels in preschools which raises concerns about children’s hearing, well-being and learning environment. A number of factors affect the sound environment; activity, room acoustics, number of children and personnel, pedagogic methodology, noise awareness among others. This study focus on the effect of room acoustics on the resulting sound environment. We measure room acoustic parameters, monaural and binaural impulse responses in up to 30 public preschools in the Gothenburg area in Sweden. Variation of the room acoustic character is sought by choosing objects built in the span between 1980 and 2006. With anechoic measurements of preschool children the aim is to simulate the resulting sound in different preschools. Are there audible differences between preschools when virtually using the same group of children as sound sources? Swedish building regulations use reverberation time as the main target when planning preschools, how does it and other room acoustic parameters compare with the audible impression of the room? The study is a part of the SPACE research project where we want to find the acoustic quality criteria that result in a Supportive Preschool ACoustic Environment.
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On the production mechanisms of the singer’s formant
Bernd Kröger
[Abstract]
A small extra tube directly located above the glottis (laryngeal
tube), which is about six times smaller in cross-sectional area
than the pharyngeal tube section located above this laryngeal
tube and which has a length of about 1/6 of the overall length
of the vocal tract tube seems to be responsible for producing
a broad (hyper-)formant at about 3 kHz, called the singer’s
formant. From an acoustic viewpoint the singer’s formant has
been described as a clustering of the third, fourth and fifth
formant. Using a computer simulation model for sound
production, which includes a self-oscillating vocal fold model
(source model), an acoustic wave propagation and radiation
model (tube model or filter model) as well as source-filter
interaction mechanisms we were able to show that a further
mechanism is responsible for increasing the amplitude of partials
in the frequency region of 3 kHz. Our simulations indicate that in
the case of mid to high subglottal pressure (loud singing voice)
specific source-filter interaction mechanisms are important
contributors for increasing partials in the region of 2 kHz to 5 kHz.
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Effect of a training program of voice use on voice functioning of college professors
Ángela Patricia Atará Piraquive, Pasquale Bottalico, Lady Catherine Cantor-Cutiva
[Abstract]
BACKGROUND: Occupational voice users present different
vocal loads depending their daily activities during a working
day. College professors may have different vocal load
depending their working conditions. Training programs have
been widely used to promote healthy voice use and prevent
the occurrence of voice disorders due to overuse or misuse of
the voice by teachers. OBJECTIVE: To determine the effect of a
training program of voice use on voice functioning of college
professors. METHODS: A group college professors will participate
in a training program on healthy voice use at work. The program
will be developed in 4 sessions. Pre and post training assessment
will be implemented including a questionnaire and voice
samples. In addition, voice dosimetry will be used to evaluate
the vocal load during the classes. RESULTS: Expected results are
a decrease on phonation time percentage during classes due
to the adjustments on the teaching-learning strategies discussed
during the training program. We expect also to identify change
on voice parameters associated with decreased vocal fatigue
and voice complaints. In addition, we expect to find evidence
that supports the hypothesis that training programs on healthy
voice use at classrooms are a good approach to decrease the
occurrence of voice disorders.
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Audio-visual Mutual Effect of Spatial Impression in Architectural Space
Chiharu Hiraoka, Takane Terashima, Ayumi Ishikawa, Yasunobu Tokunaga, Xue Gong
[Abstract]
The purpose of this study is to clarify the influence of visual
information on subjective impression for room sound field.
Authors have carried out subjective experiments on it using
room impulse response signal as auditory stimulus and interior
panorama picture as visual stimulus, and found that
reverberance/clarity for room sound field is affected by
information sensible from room volume, dimensions, geometry
or finishing etc. And then in this report, experiments about visual
effect on spatial impression (ASW and LEV) in auditorium were
conducted to clarify the influence of room volume, size or
geometry on them. Auditory stimuli were made in anechoic
chamber by binaural microphone recording under the
conditions that directions of early reflections and late
reverberation were varied, and were presented to subjects
through headphone. Visual stimuli were made of panorama
pictures captured around room interior or drawn by CAD, and
were presented by HMD. ASW and LEV were measured under
single-mode or multi-mode presentation of visual /auditory
stimuli, and mutual effect of visual and auditory sense were
examined in architectural space.
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Experiment on Audio-visual Mutual Effect on Subjective Impression in Architectural Space by HMD VR Display
Takane Terashima, Chiharu Hiraoka, Xue Gong, Yasunobu Tokunaga, Ayumi Ishikawa
[Abstract]
The purpose of this study is to clarify the influence of visual
information on subjective impression for room sound field.
Authors have carried out subjective experiments on it using
room impulse response signal as auditory stimulus and interior
panorama picture as visual stimulus, and found that
reverberance/clarity for room sound field is affected by
information sensible from room volume, dimensions,
geometry
or finishing etc. Visual stimulus of interior panorama picture is
projected on flat screen in these experiments, but visual
experiences were not always same as those in real room. In
this
report, new experiments using head mounted display instead
of
flat screen in presenting visual stimulus were conducted.
Results
were compared with previous experiment of flat screen, and
the presentation methods of visual stimuli were inspected. It
was
found that subjective dimensions for room were affected by
reverberance of auditory impression in the new experiment
unlike previous one. Influence of visual information was
clarified
in the close condition to reality.
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Sound Field Evaluation by using Closely Located Four-point Microphone Method and Mixed Reality Technology
Wataru Teraoka, Yuta Kataoka, Yasuhiro Oikawa, Yusuke Ikeda
[Abstract]
The closely located four-point microphone method was proposed by Yamasaki in 1989.
The method can grasp the spatial and temporal structures of sound reflections by
estimating an image source distribution from four measured impulse responses. In
recent years, mixed reality (MR) technology has rapidly developed and is now more
familiar. Many sensors, display devices, and ICT technologies have been implemented
in MR equipment, which enable interaction between real and virtual worlds. And we
proposed an MR display system for image sources distributions and directivity patterns
of sound reflections, which are obtained by the closely located four-point microphone
method.
In this paper, we measured impulse responses at few points using closely located four-
point microphone and display image source distributions at these points by using the
MR display system. The user can view a real room and the data representation at user’s
point simultaneously. In addition, the data at user’s position is displayed while the user
is moving in the room. Thus, the user can observe the spatial and temporal structures of
sound reflections at user’s position while maintaining a relationship between the
positional information of the real room and the data and can evaluate whole sound field
of the room.
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A Study on Discrimination of Acoustic Conditions in an Audience Area of an Auditorium
Yasunobu Tokunaga, Ryo Inoue, Takane Terashima
[Abstract]
The purpose of this study is to clarify the acoustical difference depending on the seat position in an auditorium, particularly the level of difference perceivable by the audience and acoustical cues for perception of difference.
Subjects were presented through headphones with pairs of impulse responses measured in different positions of an auditorium, and then answered for questions whether two impulse responses sounded the same or not.
Answers of the subjects were subjected to cluster analysis and multidimensional analysis so that the audience area, which was previously divided into several zones, was grouped according to acoustic similarity.
As a result of comparison between the subject’s answer and objective measurement obtained by the impulse responses in respective seats, it is likely that the subjects perceived the difference in the acoustic conditions on the basis of not only the subjective level of sound but also the perceived clarity of sound and perceived reverberance.
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Comparison between sound absorption coefficients of resonant membrane panels laminated bamboo tested in reverberant chamber and virtual simulation
Brunno Guilherme Barbosa De Sa, Jaime Gonçalves De Almeida, Maria Luiza De Ulhôa Carvalho, Rafael Caetano Cardoso Boaventura
[Abstract]
This work reviews the master’s research that evaluated the
possibilities of using laminated and glued bamboo panels
(BaLC) from the species Dendrocalamus giganteus in the
construction of a resonant membrane material for low
frequency sound absorption. The devices were
acoustically
tested with internal air cavities of 100mm, 75mm, and
50mm
thickness with and without glass wool filling. The procedure
for
obtaining the sound absorption coefficient followed the
ISO 354
international standard with a sample of 12m2 of BaLC in a
reverberant chamber of 207m3. The results showed the
resonance frequency at 100Hz and the maximum sound
absorption coefficient at the same frequency for all
sampling
configurations. In the analysis of the sound-absorption
coefficients, a variation at 100Hz was observed from 0,387s
for
empty cavity with 50mm thickness to 0,778s for 50mm
cavity
filled with glass wool. Furthermore, results were compared
with
the theoretical prediction in computational modelling to
verify if
the coefficients obtained in the laboratory coincide with
those simulated virtually. This research represents an
evidence-
based contribution to reveal the benefits of bamboo
technology
for the built environment given that it can be an
accessible
resource in tropical countries due to fast growth and
easy adaptation to local soil.
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Acoustic adjustment of large sports halls - case study: correction of the acoustic response of an Ice Arena for uses other than skating
Alberto Piffer, Elena Resenterra
[Abstract]
In 2016, the municipality of Folgaria - North Italy - decided to
using its ice
arena for various activities due to the difficulty in the
economic management
of a structure exclusively dedicated to ice skating.
This is a structure with a playing field of 2,500 m2 and 1,000 m2
for the
grandstand. The roof has two flaps with a variable radius
curved course and a
maximum height of 19 metres. The total volume, equal to
53,000 m3, was
initially characterised by a reverberation time of about 8
seconds at the
middle frequencies.
The author has been commissioned for the feasibility study
aimed at verifying
the possibilities for intervention. The second step involved the
detailed
design of the interventions and their cost for the preparation
of the
competitive tender. At the beginning of 2018 the building
process were
contracted and the works, of which the author was the
Construction
Supervisor, were completed in November. The reverberation
time is now equal
to 2.2 seconds and allows the use of the environment for
concerts, meetings,
parties and sporting events of various kinds.
This contribution reports the whole history of the project, going
from the
construction phases untill the final approval tests.
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The acoustical assessment of the commercial buildings - design assumptions
Elżbieta Nowicka
[Abstract]
Issues of interior acoustics include shaping appropriate
acoustic conditions necessary to receive music or text
spoken in the rooms.
The purpose of the interiors is related to the relevant criteria
of acoustic quality, which can be divided into subjective
and objective ones. Subjective criteria are based on the
opinions of the listeners and are very important for the
assessment of the acoustic values of the room. Many
authors of research works on this subject strive to find a
connection between subjective assessment of acoustic
properties of rooms and an objective one in order to
connect the latter with methods of designing interior
acoustics.
The issues of the acoustic climate of commercial buildings
include, among others, shaping the appropriate acoustic
conditions necessary to receive the spoken text in these
rooms, especially from sound alarm system.
Designing acoustics must be considered at an early stage of
the design process and re-analyzed and managed
throughout the investment process. The paper discuss the
acoustical assumptions important during designing proces
of the commercial buildings.
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Visualization of Distribution of Room Acoustic Parameters by Using Mobile Robot
Masashi Uehara, Naoki Ishikawa, Shigeki Okawa
[Abstract]
In this paper, we design a robot system to generate a map
which shows acoustic information in a room. A distribution of
room acoustic parameters helps us understand acoustic
environment; nevertheless it needs measurement of impulse
responses at multiple points that are time-consuming. To
visualize a distribution of room acoustic parameters with
efficiency, we utilized a mobile robot with SLAM (which
estimates the self-localization and environment map
simultaneously). The mobile robot leverages self-localization to
record measurement points and utilizes an environment map as
a base to be plot acoustic information on. The robot repeats
measurement of impulse responses and moving all over the
room. We extract parameters from each impulse responses then
conversion this parameters to color. Finally, we plot the color
information at measurement points on the environment map.
We examined how visualized a distribution of room acoustic
parameters in an actual meeting room with sufficient accuracy.
In result, the robot system generated maps of room acoustic
parameters that are related to reverberation and speech
clarity.
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Reverberation Time Estimation through Neural Networks
Antonio Petraglia, Rodrigo Prates, Mariane Rembold Petraglia, Julio Cesar Boscher Torres
[Abstract]
In this paper we investigate a procedure for blind
estimation of the reverberation time of a room, that is,
without the a priori knowledge of its impulse response. A
neural network based method is proposed, which is robust
to noise and to abrupt variations over time of the
acquired signal. The input features of the neural network
are calculated in the frequency domain, where training
samples are generated in mel scale. A competent
network architecture (i.e., with appropriate number and
type of hidden layers and number of neurons comprising
each layer), as well as the best training algorithm and
regularization technique, are searched. The use of cross-
entropy error during training gives better results that the
use of mean squared error. Comparative results are
presented, considering two previously proposed methods:
a blind neural network technique and a semi-blind power
spectral density approach.
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Fluttering Reverberance: Real Life Examples of Chaotic Billiards with Convex Sections
Trevor Cox, Richard Hughes, Dominic Parry-Merrell
[Abstract]
Billiards are used to explore dynamical systems. The
mathematical literature focusses on idealised physics and
simple shapes like the Bunimovich stadium. But real
enclosures
where sound waves created by such shapes exhibit strong
audible artefacts are uncommon. Also, billiard modelling is
a
high frequency approximation and diffraction is often
important. Notwithstanding, some rare real-life acoustic
billiards
where the effects of closed-orbits are clearly audible have
been
found and examined with simulations and measurements.
The
abandoned Thurgoland railway tunnel near Penistone, UK
has
extraordinary metallic flutter in its long reverberant
decay.
The booking hall in the National Theatre subway station in
Oslo
has a very strong warble, and sounds different to the flutter
echo heard in a simple cylindrical space. Both spaces
feature
concave geometries: the Thurgoland tunnel has a
horseshoe-
shaped cross-section, whereas the Oslo station is a
distorted
cylinder in plan with a domed roof. Ray-tracing reveals
closed
non-isolated orbits, which lead to repeated reflections and
marked flutter echoes in both spaces. Examining the
angles of
the rays arriving at the receiver, and the autocorrelation of
the
impulse responses, provides further evidence of non-
diffuseness.
FDTD simulations are used to reveal the role that diffraction
plays
in the spaces.
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In-situ Sound Absorption Measurement Method of Materials Using Ensemble Averaging - Comparison of Proposed Method with Tube Method or Reverberation Room Method -
Noriko Okamoto, Otsuru Toru, Reiji Tomiku, Kaho Ito
[Abstract]
The absorption characteristics of building materials are
generally measured by the reverberation room method or the
tube method. However, these methods are known to show
large deviations of measured absorption coefficients due to the
diffuseness of sound fields or the installation condition of
materials to the tube, respectively. On the other hands, the
authors have recently proposed the acoustic absorption
measurement method using ensemble averaging technique,
i.e., EA method. The method carries out a simple and efficient
in-situ measurement method of surface normal impedance of
materials at random incidence. In the previous study, the
reproducibility of the method was confirmed through the round
robin test, in which the absorption characteristics of some
building materials were measured in three different rooms.
However, few studies have focused on the relationship between
value measured by the EA method and those of the
conventional methods. In this paper, in order to clarify those
relationships, sound absorption characteristics obtained by the
EA method are compared with those by the tube method or
the reverberation room method for several kinds of materials
with different sound absorption performances.
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A Ray tracing algorithm developed at the Acoustical engineering course of UFSM in Brazil
Eric Brandão, Rodrigo Dal Fiume, Gonçalo Morgado, William D'Andrea Fonseca, Paulo Mareze
[Abstract]
This paper presents a ray tracing algorithm developed as a research and
teaching tool at the acoustical engineering course of the Federal University of
Santa Maria in Brazil. The motivations to pursue this task and some novel
features of the code presented. The results obtained so far were compared to
some ODEON simulations. Calculations on the Elmia Hall, extensively covered
on the 2nd round robin of room acoustics inter-comparison software were
also performed.
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The Importance of Several Room Acoustic Descriptors in Operation Rooms
Mai-Britt Beldam
[Abstract]
Sustainable working environments include good room acoustics but operation rooms (OR) can be a challenge. Due to hygiene regulations and demands the surfaces are often hard and reflective and this leads to high sound pressure levels, bad speech clarity and long reverberation times. Furthermore, guidelines and building codes for healthcare facilities normally only include reverberation time.
At Hvidovre Hospital in Copenhagen, three identical ORs were investigated. One room was used as a control room, one went through an acoustic intervention with added acoustic absorbers in the ceiling, and in the third room both an acoustic ceiling and wall absorbers were installed. Questionnaires about the sound environment in general and the effect on wellbeing, mistakes and misunderstandings were completed by all staff members (nurses, doctors, etc.). Room acoustic measurements were done in all three rooms.
Several acoustic descriptors were evaluated and the trend was clear. The results showed that reverberation time alone does not show the full picture. Small differences in other acoustic descriptors affect perception and performance. Staff questionnaires revealed that even when an OR lives up to local guidelines mistakes are more likely to happen in comparison to an OR where improvements in several descriptors are investigated.
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Acoustic conditions of clinic rooms for sound field audiometry
Valentina Zapata-Rodriguez, Cheol-Ho Jeong, Ida Hoffmann, Wan-Ho Cho, Mai-Britt Beldam, James Michael Harte
[Abstract]
The British Society of Audiology (BSA) recommends a maximum reverberation time of 0.25 s for sound field audiometry in clinic rooms. The present study measured room impulse responses from 36 rooms where sound field audiometry is conducted in Denmark, Germany and the Republic of Korea. The volumes of the rooms measured ranged from 14 m3 to 60 m3, and the reverberation times ranged from 0.1 s to 0.65 s. Their background noise levels varied from 20 dBA to 30 dBA. Office type rooms where other medical tasks also take place generally lack sound absorption particularly at low frequencies. Other acoustical parameters such as clarity and definition are also presented.
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On the Dimension and Scaling Analysis of Soundscape Assessment Tools: a case study about the “Method A” of ISO/TS 12913-2:2018
M Lionello, Francesco Aletta, Jian Kang
[Abstract]
Soundscape assessment methods are currently being standardized
and semantic differentials with bipolar pairs of adjectives are a
common method used, as reported for instance in the Method A of
the ISO/TS 12913-2:2018. However, when applying this method
during surveys with non-trained participants, a deviance in scaling
adjectives has been observed. This study aims at analysing the
distribution of these deviances and possible systematic errors in the
scaling. Data were collected through surveys in public open spaces in
the UK, Italy and China. Statistical analyses were conducted over the
pairs ”pleasant-annoying”, ”vibrant-monotonous”, ”calm-chaotic”, and
”eventful-uneventful” with corresponding 5-point Likert scales. The
results show a scaling bias effect over adjectives that belong to the
same semantic dimension. In particular, participants who gave a
neutral evaluation of the adjective ”pleasant”, show a systematically
greater disagreement with ”annoying” whereas the agreement is
significantly lower. Similar patterns are observed across other pairs of
adjectives. A further analysis revealed that only a small percentage of
participants gave an evaluation for each pair of adjectives within a 1-
point scale distance.
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Sound and Weather - A Complex Relationship
Jan Lordieck, Bryce Timothy Lawrence
[Abstract]
The emergence of soundscape ecology focused attention on the composition of sounds in an area and its influencing factors. Many studies have investigated the influence of the natural and built environment on soundscapes, but less attention has been paid to atmospheric influences on the sound environment at a specific moment - even though the immediate sound environment determines human perception of an area at a specific moment. This poster presents a short study on the influence of weather and landscape position on the metrically quantified sound environment in a public park in Gelsenkirchen, revealing that cloud coverage has a significant influence on NDSI, AEI, and ADI values, as well as human and avifauna observations. The results inform the conclusion that dense low-hanging cloud coverage may influence the sound environment by dampening soundwaves and masking background anthrophonies, in a type of ëbackground noise absorber’ effect, constituting a higher fidelity soundscape. A poster presents these results and invites discussion around the idea of the ëbackground noise absorber’ effect and the potential for quantification of other psychoacoustic effects of weather on soundscape perception.
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The effect of spectral centroid on perceived birdsong in urban forests
Xinchen Hong, Chen Yan, Guangyu Wang, Siren Lan
[Abstract]
This study explored the relationship between soundscape and
birdsong and the factors that influence perceived birdsongs in
urban parks. Three soundscape parameters were conceived:
the soundscape harmonious degree (SHD), the number of
perceived birdsongs (NPB) and the accuracy of perceived
birdsongs (APB). These were collected to find an effect of
geophony, biophony and anthrophony on perceived birdsongs.
Results indicate that there is a positive correlation between SHD
of geophony, biophony and birdsongs, and the sound
frequency is an important factor for NPB in the urban parks. In
addition, this study found group birdsong play a key role in APB
in urban parks.
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Soundscape engineering of a promenade on Namsan Mountain Park by noise, sound and preference maps
Jisu Yoo, Jae Kwan Lee, Seo Il Chang
[Abstract]
Namsan Mountain is a 262m-high mountain located in the
center of Seoul. It has a 7.5km-long promenade around it with
a moderate slope. Some sections of the promenade are
accompanied by trees and rills, which provide a good sonic
environment, but the other sections are influenced
acoustically by shuttle buses, road-traffic and various noise
sources. Noise map for the road-traffic was generated to find
the spatial distribution of the noise over the mountain
including the promenade. It shows that the promenade of
Namsan Mountain is exposed to the noise levels ranging from
55dB(A) to 75dB(A). The natural sound which was identified
most frequently by visitors was rill sound and rill-sound map
was generated. Individual responses from 4 acousticians and
6 non-acoustic people were collected from the on-site
questionnaire about sonic environment along the
promenade at 37 evaluating spots. The results of the
questionnaire were used to generate a preference map.
Separate and combined analyses of the three maps were
performed and the addition of the rill sound was proposed to
improve the sonic environment. To determine the levels of the
rill sound a listening experiment was conducted and the
adjusted levels of rill sound were incorporated into the noise
map.
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Using 3D City model platform for the analysis of Andorra's Soundscape
Rosa Ma Alsina-Pagès, Marc Vilella, Marc Pons, Robert Garcia Almazan
[Abstract]
In a SmartCities world, with the citizen in the center of all the
decisions, the measurement of environmental noise cannot
finish by providing an equivalent level captured over a certain
time in a place of interest. By means of WHO reports and other
studies, we know that the type of noise (traffic, anthropogenic,
industrial, etc.) has a different impact on people and that it is
important to distinguish the source of the measured noise levels.
In order to facilitate this work for the competent authorities, in
Andorra a 3D City model platform is used to visualize acoustic
measurements carried out at three strategic points in the
country. The platform represents the measured noise
differentiating the type of noise, the acoustic level it is providing
and the salience of the event with respect to background noise,
providing a wider and more complete amount of information to
allow to decide policies to minimize the impact on people.
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Noise and Soundscape Mapping in a Taebaeksan Mountain National Park of South Korea
Kyong Seok Ki, Jisu Yoo, Hunjae Ryu, Seo Il Chang
[Abstract]
The conservation of sound environment in national parks is
required in order to secure habitat quality. However, many
national parks in South Korea are exposed to various noise
sources such as roads and airplanes and etc. Therefore, it is
necessary to improve the sound environment in order to
preserve the environment of national parks. The purpose of this
study is to establish management plan through analysis of noise
and sound environment for Taebaeksan Mountain National
Park. The sound environment analysis was divided into four
categories: road-traffic noise mapping, aircraft noise analysis,
sound environment analysis, and sound resource analysis. The
noise map modeling result showed that 91.5% of the total area is
exposed to noise levels below 40dB(A). The aircraft noise
analysis shows that daily average aircraft noise frequency was
54 times per day. Aircraft noise occurred not only during the day
but also during the night. Aircraft noise was also affecting the
entire area of the park. Sound resource mapping shows that the
valley area has a lot of wildlife. In the ridge area, there was a lot
of wind sound. For the noise management of Taebaeksan
Mountain National Park, it was recommended to manage
aircraft and road noise.
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Investigation on the impact of traffic noise on the acoustic environment of urban parks in high-density cities: Taking Shenyang City, China as an example
Yuan Zhang, Xinhao Yang, Siyang Guo
[Abstract]
Urban parks provide residents with space for relaxation, fitness and
recreation, while the surrounding traffic noise has a negative impact on
their environmental quality. This paper conducts a survey of surrounding
road conditions in 49 urban parks in the typical high-density city of
Shenyang, China. 2 parks were selected for traffic condition
measurement and noise simulation with Cadna/A; Making spatial
analysis with ArcGIS.
The results showed that: (1)57.6% of the 205 borders in 49 urban parks
directly adjacent to urban roads, including expressway (6.8%), main road
(25.4%), secondary road (41.5%) and branch road (26.3%); (2)Based on
the actual traffic measurement, traffic flow is significantly positively
correlated with road level; (3)The traffic sound LAeq of the adjacent
roads of the two parks was distributed at 59.0-70.9dB(A), and there is a
certain distribution law in the time period and whether it is a working day;
(4)The relatively clear area of traffic sound perception in the two parks is
54.8% and 88.6% respectively; As it goes deeper into the interior space,
the significance exhibits a different gradient of attenuation. The
attenuation gradient are affected by factors such as spatial
characteristics, landscape features, and sound composition.
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Spatial Analysis of the Impact of Urban Forms to Road-traffic Noise in a Highly Populated City
Taeho Park, Minho Kim, Phillip Kim, Seo Il Chang
[Abstract]
Road traffic noise is widely distributed in the cities and is
more closely related to the urban resident people than other
noise sources. To understand the relationship between road-
traffic noise level and urban form indicators such as ground
space index, floor space index, road density, road length,
traffic volume and so on, spatial analysis was performed on
highly populated city in South Korea. A system of 125m X
125m grid cells is placed on the city. And the urban forms
and noise levels were calculated in each grid. Noise levels
were calculated as energetical averages on each grids based
on the noise map. In addition, the relationship between the
area exceeding noise standard and urban forms was
analyzed. Finally, it was simulated for changes in road-traffic
noise levels due to urban forms change situations such as
land development, road construction, and land use change.
This study can be a policy tool to consider residents’ noise
exposure or area exceeding noise standard in the design
process such as the redevelopment of highly populated urban
city or the policy decision making process for urban planning.
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The Importance of An Urban Interpretation of Environmental Noise. The case of Mexico City
Fausto E Rodriguez Manzo, Elisa Garay-Vargas, Laura Lancon-Rivera, Dulce Ponce-Patron, Silvia Garcia-Martinez
[Abstract]
Cities are the space where environmental noise finds its most
evident manifestation. For this reason it can be considered that
environmental noise is a problem that concerns urbanism and in
many cases is also a reflection of poor urban planning. This
paper shows a methodology that incorporates urban analysis to
find the relationships that exist between urban aspects and
environmental noise in public space, in representative areas of
the city. The urban environment is analyzed through statistical
and cartographic data and noise maps to show the impact of
environmental noise with greater definition. The analysis of the
sound environment is also incorporated by means of binaural
sound registers and acoustic measurements. Some specific
examples of the interpretation of data and contributions to the
study of environmental noise are shown through the joint
analysis of the various urban, social and demographic factors
that make it up. The sound environment of cities, perhaps the
most important element in the perception and definition of the
character and urban image of cities is still absent from the
concepts of urbanism and what this study presents is a possible
approach to incorporate it.
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Artificial neural network analysis of the relationship between road-traffic noise and air pollutants and urban form indicators
Phillip Kim, Hunjae Ryu, Jong June Jeon, Seo Il Chang, Nokil Park
[Abstract]
Road-traffic noise and air pollutants have adverse effect to
people’s health and life quality. For management of the noise
and air pollutants, noise and air pollution maps can be used to
provide quantitative information of exposure levels. In this study,
the more efficient method of noise and air pollution mapping
was developed statistically. The relationship between road-
traffic noise level and air pollutants and urban forms was
analyzed by artificial neural network analysis. The selected
representative urban form indicators are road-related (traffic
volume, speed), building-related (floor space index, ground
space index), and land-use-related indicators. The artificial
neural network model was optimized by adjusting the number of
hidden nodes and layers. In the optimizing process maximum
steps for the training, number of repetitions and algorithm type
can be adjusted. The 2/3 of data sets for a region was used for
the model development to select the model with the least
prediction error. The selected model was applied to the
remaining 1/3 of data sets for verification. The result from the
artificial neural network model was compared with that from
engineering model.
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Sustainable Resources to Improve on Living Green Walls Acoustics: Supply Chain Study
Ghofran Salah, Anna Romanova
[Abstract]
Living Green Walls (LGWs) or Vertical Gardens are built and
deigned using a selection of plants, growing medium, frames,
supports, and irrigation system. LGWs are considered as a
natural solution of controlling noise especially in urban
environments. A better sound absorption can be achieved
through replacing the currently used growing medium by a new
material with a higher porosity. A high porosity material allows
more sound to enter and dissipate in the material matrix.
Coconut fibre can be a considerable alternative for the
currently used growing medium, due to its high porosity. It has
proven that natural fibres have good sound absorption
coefficient at medium and high frequencies (500-2000HZ). The
study works on improving the LGWs acoustic using sustainable
resource (Coconut fibre). In order to increase the sound
absorption over the current LGWs and utilise the coconut waste
obtained from grinding the coconut waste which naturally
exists. Moreover discussing the impact of introducing the
coconut to the LGWs Supply Chain. The paper presents analysis
based on real data collected from LGWs companies.
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Perception of auditory-visual simultaneity changes by ambient illumination
Masaaki Tezuka, Mariko Tsuruta-Hamamura, Hiroshi Hasegawa
[Abstract]
In this study, we investigated changes in the point of
subjective simultaneity (PSS) between auditory and visual
stimuli by changes in ambient illumination. The test stimuli
were a pure tone sound of 1000 Hz at 80 dB SPL and white
light of LED at 21.9 cd/m2. The illuminance levels at around
the light source and at the subject’s eyes were adjusted
at the following 4 conditions;(100 lx, 100 lx), (100 lx, 1000 lx),
(1000 lx, 100 lx), and (1000 lx, 1000 lx). Nine people in their
early 20s participated in the experiments. Synchronized
sound-light stimuli, which were the same as the test stimuli,
were presented as the preceding stimuli, followed by the
test stimuli at an interval 1000 ms. The test stimuli had a
stimulus onset asynchrony (SOA) of 0, ±20, ±40, ±80, or ±160
ms. After each presentation of the test stimuli, we asked
subjects to answer which of the stimuli - sound or light -
they perceived first. Then we evaluated the PSS of the test
stimuli and considered the influence of ambient
illuminance on the PSS. As a result, the PSS shifted toward
positive direction. We showed that the PSS shift depended
by changes in ambient illumination.
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Digitization of cultural heritage of the UNESCO site - Lipnica Murowana
Pawel Malecki, Jerzy Wiciak, Katarzyna Sochaczewska, Natalia Krygowska
[Abstract]
Lipnica Murowana is a unique, although a small town in the
centre of the Małopolska Region. It is known of the 15th century
St. Leonard wooden church that was entered on the UNESCO
World Heritage List in 2003. The article shows the activities of THE
MULTISENSORY UNESCO project that was aimed at digitalizing
and sharing the cultural and national heritage of Lipnica
Murowana. The paper focuses on acoustical layer digitalization.
It shows soundscape analysis and recording and the acoustics
of the St. Leonard church and other interesting historical
monuments.
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Acoustic quality and health in urban environments - First methodological experiences of the pilot study SALVE
Robynne Sutcliffe, Bryce Timothy Lawrence, Salman Ahmed, Timo Haselhoff, Dietwald Gruehn, Susanne Moebus
[Abstract]
In cities, sound can characterize urban environments
(soundscapes) just as land-use types and patterns, which can
impact well-being and health of the population. To identify
which criteria are important for health-promoting soundscapes,
a two year interdisciplinary MERCUR Research Center funded
pilot study ”acouStic quAlity and heaLth in urban enVironmEnts
(SALVE) - analysis of the interrelationships between soundscapes
and health” was launched in October 2018. SALVE aims to
establish one of the largest multi-seasonal urban soundscape
datasets in Germany by making year-long direct and
automated auditory measurements. The calculation of a wide
array of metrics (WAM) over a large heterogeneous urban area
will enable analysis of associations to health outcomes from the
long-term, population-based Heinz Nixdorf Recall study.
SALVE is a starting point for an innovative approach to
understanding the effects of sound quality on urban public
health - beyond noise mitigation. Here we will present
methodological aspects of study design, like the operation of
direct and automated measurement devices in urban
environments and spatial epidemiological analyses.
Furthermore, we will give first results, as well as insights into
practical experiences of acceptance, data protection and into
opportunities as well as pitfalls which arise during a large
soundscape data collection and analysis project.
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Comparison of Direction of Arrival Methods for Separation of Vehicle Sound Sources
Gabriela Dantas Rocha, Felipe Petraglia, Julio Cesar Boscher Torres, Mariane Rembold Petraglia
[Abstract]
Road traffic is the main component of urban noise, considered
by the World Health Organization as the third type of pollution
that most affects the population of large urban centers.
The auralization technique stands out as an intuitive tool for
assessing environmental noise, but does not yet explore the
sound characteristics to provide an accurate hearing
experience. In this context, it is desirable for a database
containing vehicle sounds to be provided to the auralization
systems, but these signals must be previously located and
isolated from undesirable sources of noise. This paper presents a
performance evaluation and comparison of five direction of
arrival (DoA) algorithms to locate vehicular noise sources. Four
of the algorithms are based on estimates of the time difference
of arrival (TDoA), whereas one employs the minimum variance
distortionless response (MVDR) beamformer. The estimated
source locations obtained by the five algorithms for data
acquired by eleven circularly arranged microphones are
compared with the correct positions of the noise sources.
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Relationship between Subjective Responses and Physical Parameters of Air-Conditioner Noises in a Car
Yoshiharu Soeta, Hiroko Kagawa, Yoshisada Sakamoto
[Abstract]
Sound pressure levels (SPL) of air-conditioner noise are
decreasing due to the developments of noise control
techniques. However, people can still feel uncomfortable by
certain aspects of the sound quality, even when the actual SPL
is low. Not only quantity aspects of the noise, such as SPL, but
also quality aspects of the noise can affect the subjective
evaluation of air-conditioner noises in a car. Therefore, both the
SPL and the sound quality of an air-conditioner noise are
important for the user’s acoustic comfort. The aim of this paper
is to clarify the characteristics of air-conditioner noises in a car
and determine the factor that is most influential on the
subjective responses caused by the noise. Sound quality can be
characterized by factors obtained from autocorrelation
function (ACF) and interaural cross correlation (IACF) of a
sound. Subjective annoyance was evaluated using a paired
comparison method. Multiple regression analyses were
performed using linear and nonlinear combinations of SPL, the
ACF, and IACF factors, and their standard deviations (SDs). The
subjective annoyances were formulated by the ACF and IACF
factors.
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Psychoacoustical assessment of thermal impression of automotive HVAC noise
Seiji Nakagawa, Takuya Hotehama, Masaru Kamiya
[Abstract]
Two subjective assessments were performed focusing on the
auditory impression of automotive HVAC noise concerning
coolness and warmness. First, paired comparison tests were
carried out under various conditions of room temperature. Five
stimuli were synthesized by stretching the spectral envelopes of
recorded automotive HVAC noise to assess the effect of the
spectral centroid. Twelve normal-hearing subjects were asked to
rate the auditory impression of the stimuli for each pair on a
seven-point scale according to how much the latter is warmer
(for the winter tests) or cooler (for the summer tests) than the
former. Results show that the spectral centroid significantly
affects the auditory impression concerning coolness and
warmness; a higher spectral centroid induces a cooler auditory
impression regardless of the room temperature. Second, effects
of HVAC noises on the subjects’ sensation of coolness and
warmness were evaluated by using a method of continuous
judgment by category. Room temperature were controlled to
increase/decrease linearly, and HVAC noise, having warm/cool
auditory impressions, were presented. Subjects had to answer
their sensation of coolness/warmness at regular time intervals.
The results showed that HVAC noise had significant effect on the
time change rate of sensation of coolness/warmness.
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Active control and passive consequence of vowel devoicing in Japanese: Evidence of highspeed movies and PGG
Masako Fujimoto, Seiya Funatsu, Ichiro Fujimoto
[Abstract]
Vowel devoicing is a phenomenon that is found in many
languages including Japanese, Korean and Montreal
French. Japanese is one of the most explored languages on
the mechanism of vowel devoicing (VD). The typical case of
VD in Japanese is that high vowels /i/ and /u/ are
surrounded by voiceless obstruents. However, the frequency
of occurrence of VD differs depending on consonant types,
accent, dialects, and so on. Previous physiological studies
have shown that VD in Japanese involves both active re-
organization of the glottal opening pattern during /ÇVÇ/
and passive overlapping of glottal opening gesture for each
voiceless consonant. The former is regarded as phonological
and the latter, phonetic. This paper demonstrates the rich
variety of the glottal opening pattern during VD using a
high-speed movie of the glottis and Photoglottography
(PGG). Demonstration covers the difference between stop
and fricative consonants and that between Tokyo and
Osaka dialects, as well as the pattern of consecutive
devoicing and the low vowel devoicing.
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Investigation of Acoustic Models for Emotion Recognition using a Spontaneous Speech Corpus
Tetsuo Kosaka, Yuka Haneda, Daisuke Makabe, Masaharu Kato
[Abstract]
In this study, we aimed to improve emotion recognition
performance for spontaneous emotional speech. While several
studies on emotion recognition have utilized acting speech, few
have focused on spontaneous emotional speech because their
speech corpora are not well developed. Recently, the online
gaming voice chat corpus (OGVC) consisting of spontaneous
emotional utterances was developed. We conducted emotion
recognition experiments to classify four emotions (anger, joy,
sadness and neutral) using this corpus. In our recognition system,
we used the features standardized in the Interspeech 2009
emotion challenge as speech features, and a deep neural
network which has a feed-forward type of architecture as an
acoustic model. Subsequently, through recognition experiments,
we examined the hyperparameter of the acoustic model, and
the amount and quality of training data. Findings pertaining to
the amount of training data revealed that the recognition
performance did not improve even when the amount of data
was simply increased. Generally, the intensity of emotional
expression is low in spontaneous emotional speech, while it is
high in acting emotional speech. We found that it is effective to
select utterances with low emotional intensity from the acting
speech corpus and add them to the training data.
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Opening the Black Box: Real-Time Speech Perturbation Experiments Reloaded
Bahne Hendrik Bahners, Sebastian Heidelberg, Joseph Baader, Ruben Van De Vijver, Markus Butz, Julian Rohrhuber
[Abstract]
In neuroscience, speech perturbation experiments are a well-studied and useful mean to assess speech pathologies and neuronal integration mechanisms. These experiments often depend on complex signal processing technology. In order to enable replication and modification of experiments as well as the interpretation of their results, technological details need to be accessible. However, in practice the experiment’s actual mechanism often remains hidden in black boxes like digital signal processors or other audio equipment. We conducted a study with 20 Parkinson patients, assessing vocal responses to pitch shifted auditory feedback. While we could replicate earlier experimental findings of Parkinsonian speech pathologies, we introduced an open-source and easy-applicable setup for real-time speech perturbation experiments. It runs on standard audio interfaces, allows researchers to interactively reprogram the signal flow at runtime and can be applied both inside and outside the laboratory.
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Detection of Anchors’ Utterances in Broadcast News Using I-vector-based Speaker Similarity and Temporal Information
Daichi Nozaki, Masaru Yamashita, Hiroyuki Takada, Matsunaga Shoichi
[Abstract]
Accurate speech recognition of anchors’ utterances in broadcast news programs is essential for indexing tasks such as topic extraction, summarization, and so on. Consequently, the number of anchors and all utterances for each anchor are important aspects that must be evaluated correctly. This paper discusses the effectiveness of these two aspects based on speaker similarity using i-vector. In our proposed approach, first, each input audio source was segmented into four types of sound: speech, noise, silence, and music. Next, a bottom-up clustering was performed using all speech segments to collect utterances from the same speakers. In this clustering, cosine similarity from i-vector was used. However, it was difficult to extract sufficient speaker characteristics from short utterances. To address this problem, we considered the time intervals between successive pairs of utterances and the existence of background noises, adding to the information obtained from i-vector. Among the detected speaker clusters, anchor clusters were obtained based on the heuristic that the ratio of the number of utterances from anchors was much higher than that of other speakers. It was experimentally shown that the detection method of anchors’ utterances using the proposed approach outperformed the method using conventional k-means clustering.
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Using partial a priori knowledge of Relative Transfer Functions to design an MVDR beamformer for a Binaural Hearing Assistive Device with External Microphones
Randall Ali, Toon Van Waterschoot, Marc Moonen
[Abstract]
This paper considers a binaural hearing assistive device
(HAD)
equipped with a separate local microphone array (LMA)
for the
left and right ear, as well as single external microphone
nodes
that may be located within the vicinity of this HAD. For
such a
system, an MVDR beamformer may be used for noise
reduction,
and for the preservation of the relevant binaural speech
cues,
provided that a reliable
estimate of the left and right ear relative transfer function
(RTF)
vectors pertaining to all the microphones can be
obtained. In
this paper, as opposed to estimating the entire RTF vectors,
another approach is followed, which considers the case
when
there is partial a priori knowledge of these RTF vectors,
specifically when
the separate left and right ear RTF vectors for the LMA on
the
HAD
are known. The procedure for this approach will be
discussed,
which requires the computation of an
appropriate scaling between the left and right ear RTF
vectors,
and
the missing part of these RTF vectors pertaining to the
external microphone nodes. Experiments involving a
dummy
head, two behind-the-ear dummy hearing aids, and
external
microphone nodes are also performed in order to
evaluate the
benefit of the proposed approach.
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Discrimination of Mono-syllables in Sentence Context: the Case of Japanese Listeners’ Perception of /ba/-/da/ Continuum
Kanako Tomaru
[Abstract]
It is widely known that native speech sounds are perceived
according to the phonetic categories of the native
language. Perception according to the phonetic categories
is called ”categorical perception.” The hypothesis of
categorical perception assumes that two stimuli are
perceived to be different only when these are identified as
different. Therefore, the best discrimination performance
(discrimination peak) is observed when the paired stimuli are
members of different categories. Traditionally, categorical
perception has been tested using mono-syllables presented
in isolation. However, my recent research shows that the
discrimination peak is not observed when the mono-syllabic
stimuli are embedded in a sentence. The preceding study
investigated the categorical perception in a sentence using
an AXB paradigm with the stimulus interval of 300ms; the
relatively short interval may be one of the factors that
suppressed the discrimination peak. To clarify this point, the
current study conducted a perceptual experiment on
Japanese listeners using an AXB with the 1s stimulus interval.
The results suggest that, in the case of perception of a
Japanese /ba/-/da/ continuum by Japanese listeners, the
length of the interval has only a small impact.
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Speech Enhancement by Bit-Rate Extension Based on Time-Frequency Simultaneous-Constrained Griffin-Lim Algorithm
Haonan Wang, Takanobu Nishiura
[Abstract]
Different from the conventional approach of bandwidth extension
(BWE), we propose an algorithm to enhance speech by bit-rate
extension (BRE). Due to the different resolution of amplitude, low bit-
rate (LB) signals contain much more quantization noise than high
bitrate (HB) signals. Simply applying a speech-denoising algorithm
can enhance LB signals. Even if LB signals are corrupted by
quantization noise, the quantization mechanism makes the remaining
upper bits of LB signals identical to the corresponding HB signals, and
a conventional denoising algorithm will corrupt the ”correct” upper
bits of LB signals. We found that the ”correct” upper bits of LB signals
will determine a unique limited region in the time domain, where the
waveform of a real HB signal will be strictly inside. Therefore, after
general magnitude-spectrum estimation to suppress quantization
noise, we also designed a time-frequency simultaneously constrained
Griffin-Lim algorithm (TFC-GLA) to ensure that the results of HB
signal estimation will be inside the limited region while maintaining
the estimated clean magnitude spectrum. Objective and subjective
evaluations showed that our proposed BRE algorithm is effective in
enhancing speech.
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Comparative Acoustic – Phonetic Analysis of Retroflex Consonants of Some Indian Languages
Shyam Sunder Agrawal, Shweta Bansal, Shweta Sinha
[Abstract]
Retroflexion is an important phonological category in many
Indian languages. In this paper an effort has been made to
present those findings of acoustic characteristics of retroflex
sounds which significantly describe and differentiate them from
other sounds. The retroflex sounds acoustically analyzed for 3
Indian languages i.e. Hindi, Nepali and Punjabi are /t./, /t.h/,
/d./, /d.h/,/ŋ./ /sh/ /r./ /r.h/. Selected words containing each of
these consonants in different positions were recorded by 50
native Male and Female speakers of these three languages. Parameters such as spectral properties of burst and formants
transition, durational measures etc. were analyzed. It has been
observed that the formants F2, F3 & F4 of the burst and
adjoining transitions, particularly the rising of F2 and falling of F3
& F4 are significant. The minimal difference in F3 and F2, strong
release of burst, shorter duration of silence and VOTs etc. are
noteworthy. The retroflex sounds /r/ and /r.h/ which appear in
medial and final positions of a word are mostly heard as rhetoric
/r/. Some of the linguistic differences observed include /ŋ./
pronounced as /ŋ/ and /r.h/ as /rh/.in Nepali and /d.h/ as /dh/
in Punjabi. Detailed results are discussed in the paper.
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Acoustic Evaluation of Simplifying Hypotheses Used in Articulatory Synthesis
Ioannis Douros, Yves Laprie, Pierre-André Vuissoz, Benjamin Elie, Angélique Amelot
[Abstract]
Articulatory synthesis allows the link between the temporal evolution of the Vocal Tract (VT) shape and the acoustic cues of speech sounds to be investigated unlike other approaches of speech synthesis.
A number of simplifying assumptions have to be made to enable the speech signal to be generated in a reasonable time. They mainly consist of approximating the propagation of the sound in the vocal tract as a plane wave, approximating the 3D VT shape from the mid-sagittal shape, and simplifying the vocal tract topology by removing small cavities.
This work is dedicated to the evaluation of these assumptions. For this purpose 3D MRI VT shapes for sustained vowels and fricatives were acquired together with the speech sound recorded via an optical microphone. Vocal tract resonances were evaluated from the 3D acoustic simulation computed with the K-wave Matlab package from the complete 3D VT shape and compared to those of real speech, those provided by simplified 3D VT shapes (near epiglottis), and those provided by either the electric simulation from the mid-sagittal shape, or the 2D K-wave package from a mid-sagittal VT shape. We will exploit these results to improve the simplified acoustic simulations used in articulatory synthesis.
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Inverse Estimation of the Vocal Tract Shape from Speech Sounds Including Consonants Using a Vocal Tract Mapping Interface
Kohichi Ogata, Takayuki Tanaka
[Abstract]
This paper describes the inverse estimation of the vocal tract shape from speech sounds including consonants using a vocal tract mapping interface. In the interface, a vocal tract shape is determined based on formant frequencies of speech sounds via the interface window that maps pixel points onto assigned vocal tract shapes. A change in the vocal tract shape for a sound sequence can be observed as a trajectory of the pixel points on the window that correspond to estimated vocal tract shapes. In this study, an attempt was made to inversely estimate the vocal tract shapes from speech sounds of VCV sequences and estimated trajectories on the window during the VC transition were evaluated to investigate the articulatory behavior. The results showed that the voiced consonants had longer trajectory patterns on the window than the unvoiced consonants. It was also shown that the difference of the successive vowel, i.e., the final vowel V affected the route of the trajectories for the VC transition. These results suggest that the interface can be a useful analysis tool to investigate the behavior of the vocal tract for speech sounds including consonants using its inverse estimation function.
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Acoustic and Subjective Evaluation of Brazilian Portuguese Speech Recordings Made in Critical Listening Environments
Ticiana Matar De Lello, Stephan Paul, Luiz Wagner Pereira Biscainho
[Abstract]
The present work aims to characterize Brazilian Portuguese speech in realistic environments by means of acoustical parameters and subjective perception, hoping to provide data for research on auditory prostheses. Speech signals were recorded in several non-controlled environments such as restaurants, classrooms and stores. Sound pressure level, loudness and spectral content were extracted for speech and background noise separately by using recording segmentation and spectral subtraction. Subjective perception was analyzed by applying listening tests with samples from the recordings. Data analysis showed an increase of 0.6 dB in the speech’s SPL per each dB increase in the background noise’s SPL and a great difference in overall loudness between the quietest and the noisiest environments. SPLs were also mostly coherent to those previously reported in the literature for the same type of environment. The listening tests’ results, while also greatly influenced by personal taste, showed that the annoyance caused by background noise, impairment in speech intelligibility, and the effort to understand speech were correlated to the background noise level and the SNR. In general, the environments perceived as the most critical for spoken communication were those in which background noise presents the highest energy concentration in the spectral region of speech.
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A Simple Strategy for Natural Mandarin Spoken Word Stretching via the Vocoder
Yi-Jhe Lee, Ting-Chun Liao, Yi-Wen Liu
[Abstract]
In Mandarin, when a word is spoken with a longer duration,
different regions of it are not stretched uniformly. Moreover,
transition between consonants and vowels may be hard to
define. Therefore, it is challenging to find the stretchable regions
of words automatically. In this paper, we explore the idea of
parsing a Mandarin word into the part that should be played
with the original speed followed by a uniformly-stretchable
region. Then, the optimal dividing point to start stretching could
be determined automatically by minimizing the distance
between the stretched version (generated by computer) and a
ground truth (spoken by human). A database of 42 pairs of
regular-speed and slow utterances were created. The dividing
points on the regular-speed utterances were found as
proposed. The points could be aligned to words with the same
pronunciations in full sentences by dynamic time warping, and
the sentences could be synthesized with arbitrary tempo and
rhythms. The naturalness of stretching by three methods was
evaluated subjectively: uniform stretching by waveform-
similarity-based synchronized overlap-add (WSOLA), uniform
stretching based on linear interpolation in the vocoder domain
(LI-VD), and the proposed strategy. 74% of answers chosen by
41 subjects show that our method outperforms.
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A Study on English Speech Acclimatization Based on Accent Conversion for Non-native Speaker
Yutao Zhang, Takuro Sasaki, Yukoh Wakabayashi, Takahiro Fukumori, Takanobu Nishiura
[Abstract]
It is considered that listening and understanding English speech of
native speakers is a hard task for many Japanese. Therefore, we
carry out a research on converting the English speech spoken by
native speakers based on emphasis of the feature which is familiar to
listening to Japanese. We believe the converted speech could be
understood more easily by Japanese listeners for English education.
We called the research English speech acclimatization. In this paper,
we proposed an accent conversion method as an English speech
acclimatization method. Specifically, we convert the accent of native
speakers into the accent of Japanese speakers. Experiments are
conducted to evaluate the performance of proposed method based on
the English speech before conversion and after. To confirm the
effectiveness on words recognition, Japanese volunteers are required
to answer dictation questions. Moreover, the questions of selecting
the best response to the question or statement are answered by
Japanese volunteers as a subjective evaluation experiment to confirm
the effectiveness on comprehension of the sentence. As a result of
the evaluation experiments above, the effectiveness of proposed
method was confirmed.
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Piecewise Bilinear Characteristics of Acoustic Mode In Dual Pulse Solid Rocket Motor Combustion Chamber
Hao Le, Yingchen Wang, Jun Chen, Xiaoming Shi
[Abstract]
Recently, dual pulse solid rocket motor has gradually become
the main power device of spacecraft. The motor consists of two
burning chambers, separated by pulse separation device which
achieves the restarting operation. Based on the flight tests, the
axial abnormal high frequency oscillation occurred at the end
of first pulse operation while it was normal during the second
pulse. However, little insight had been gained concerning the
mechanisms involved. Therefore, a method was set up here to
solve this problem. Firstly, the function of chamber’s cavity
volume and time was proposed with the law of grain motion,
the characteristic time points were determined by the ”state
frozen” method. Moreover, instantaneous propellant grain
shapes of the motor were obtained by the specialized software,
so the 3D model of the acoustic cavity were rebuilt accurately.
Then, the chamber’s acoustic mode under the state of the two
pulse’s termination were obtained by FEM. Meanwhile, the first
three mode were obtained by acoustic testing. With the
comparison, simulation agreed well with test results, which
proved that the model was correct.Consequently, acoustic
mode’s piecewise bilinear characteristics during motor’s whole
operation was proposed completed, which has great
significance on the cooperative design of motor and
spacecraft.
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Verification of Accuracy Using Measured Values of Low Frequency Noise Numerical Analysis Generated from Expressway Bridge
Hiroshi Iwabuki, Osamu Funahashi, Masayuki Shimura, Noboru Kamiakito
[Abstract]
In Japan, road managers including expressway may respond to complaints such as road traffic noise, vibration, low frequency noise, and so on from residents along the road. Among these countermeasures, noise and vibration and so on complaints are identified and countermeasures are further examined and constructed, but on the other hand, in the case of low frequency noise caused by bridges, many examinations on the occurrence mechanism are required. Therefore, in this study, unsteady analysis was proposed by three-dimensional finite element method on responses of a bridge due to vehicle load running on a bridge. The numerical simulation of the low frequency noise field was carried out by calculating the three dimensional wave equations with this unsteady analysis result as an unsteady boundary condition. By this method, it became possible to quantitatively investigate to which part of the bridge the cause of the low frequency noise originates. In this paper, identification of the cause of occurrence and simulation accuracy of the propagation path were verified by using the measured results of low frequency noise and vibration acceleration.
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Resonance modes for exterior vibro-acoustic problems, application to a dielectric elastomer loudspeaker
Emil Garnell, Olivier Doaré, Corinne Rouby
[Abstract]
Dielectric elastomers are soft active materials capable of
large deformations . They are made of a thin
elastomer membrane, sandwiched between two flexible
electrodes. When a voltage is applied between the
electrodes, the membrane thins down, and because of the
elastomer incompressibility this results in an area
increase. Loudspeakers using this electro-activation principle have
been studied and tested by several research
groups. Sound radiation over the whole audible frequency
range has been demonstrated. Models of the
electro-activation principle have also been developed.
However, no complete model coupling all the different
physics involved in such a loudspeaker system has been
presented. Any advanced optimisation of the different
parameters of such a device (geometry, material
parameters, inflation pressure, etc) requires a validated
model. In the present research, a complete finite element model is
set-up. Acoustics inside the cavity on which
the membrane is inflated, free-field acoustics, electro-
statics and membrane dynamics are computed by a
fully coupled model. Acoustic radiation in free-field is
computed using Perfectly Matched Layers (PMLs).
Heavy fluid effects that arise because of the membrane
light weight are studied. The model is validated by
an experimental study of the dynamics and of the sound
radiation of this device.
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Study on the influence of thrust bearing isolation on the acoustic and vibration characteristics of submarine
Yangyang Zhang, Jun Suo
[Abstract]
Abstract: In order to minimize the underwater acoustic radiation of submarine caused by longitudinal unsteady excitation of propeller, vibration isolation design of thrust bearing is proposed. Firstly, the design method of thrust bearing isolator based on transfer matrix method is studied. Two sets of vibration isolation schemes of thrust bearing are designed for a submarine’s propulsion shafting. The influence of thrust bearing vibration isolation on shafting stability, acoustic and vibration transmission characteristics of submarine and underwater acoustic radiation of submarine under longitudinal unsteady excitation of propeller are studied. Results show that thrust bearing vibration isolation can obviously minimize the underwater acoustic radiation of submarine caused by longitudinal unsteady excitation of propeller.
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Study on the sound transmission loss of a truncated conical shell excited by an incident plane acoustic wave
Masoud Golzari, Ali Ashgar Jafari
[Abstract]
Conical shells are extensively used in many practical applications such as aircrafts, submarines, rockets, and tanks. The interaction of these structures and surrounding fluids mostly induces transmission of undesirable vibroacoustic energy, which can lead to structural fatigue and noise pollution in the systems. In this work, a theoretical model is proposed to study on the sound transmission loss of a thin-walled truncated circular conical shell. The shell motion is governed by Love’s theory. The model predictions are firstly validated against those of the experimental and analytical works reported in the literature. Parametric study is then conducted to investigate the effects of several important design parameters including different boundary conditions at the ends of the shell, cone angle, thickness and material properties of the shell, and incidence sound wave angle on the characteristics of the sound transmission loss. The presented model can be useful and effective in the acoustic design stage of the truncated conical shells.
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Study on the influence of kinetic parameters of propeller on acoustic and vibration characteristics of propulsion shafting
Yangyang Zhang, Jun Suo
[Abstract]
Propeller in most research on the vibration characteristics of propulsion shafting is simplified, in order to study the influence of kinetic parameters of propeller itself on acoustic and vibration characteristics of propulsion shafting, the propulsion shafting model which contains precise three-dimensional elastic propeller is established, focusing on the mass effect and elastic effect of propeller on shafting longitudinal vibration based on analytical method and finite element method. Results show that the simplified method of propeller has great influence on the propulsion shafting vibration and the influence of propeller on the acoustic and vibration characteristics of propulsion shafting includes not only mass effect, but also obvious elastic effect. From the point of view of shafting vibration control, increasing propeller mass and blade elasticity are beneficial.
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Investigation of the applicability of acoustic emission and vibration analysis to describe the thermo-mechanical mechanism during ultrasonic metal welding
Elie Abi Raad, Isabel Balz, Uwe Reisgen, Michael Vorländer
[Abstract]
Ultrasonic Metal welding (USMW) is particularly suitable for connecting electrotechnical components. Despite industrial spread, unexplainable process fluctuations may occur in USMW. This largely originates from the lack of profound scientific knowledge and understanding of the interdependency between tools and joining members. The aim of this research is to derive characteristic variables which correlate with the mechanisms during bond formation and joint properties, to achieve a more profound understanding, and in the future to open up new possibilities for online process control. In this study, aluminum sheets (AW1050A) and copper sheets (CW008A) were ultrasonicly welded at 20kHz to study their effects on the thermo-mechanical mechanism during welding. Two laser vibrometers recorded the tool vibrations of the anvil and horn, directly connected to the sheets, and a 1/4’ microphone recorded the airborne sound. Additional T-peel tensile testing and laser scanning microscopy of the fractured surface at different welding stages are used for validation.
The results of this paper provide a more profound understanding of the thermo-mechanical mechanism during USMW and will introduce a prospective improvement in quality and process stability.
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Influences on structural intensity for injection-moulded thermoplastic parts
Dennis Netzband, Andreas Ujma, Elmar Moritzer
[Abstract]
In product development such as electromobility engineers have to place special emphasis on NVH. A very large number of automotive interior components consists of injection-moulded thermoplastic parts. If required, these lightweight components are reinforced with fibres or fillings. This reinforcement leads to a higher vibration sensitivity and higher natural frequencies. In order to influence the acoustic behaviour in early development stages, the structure-borne sound - including its source and propagation paths - has to be analysed. A suitable tool to visualise these elements is the structural intensity. In addition, the anisotropic material behaviour of reinforced thermoplastics offers the possibility to conduct the structure-borne sound. In this paper a method is presented to examine the influence of fibres, fillings and geometric modifications on the structural intensity. Based on an integrative simulation method (coupling of process and structural simulation) an anisotropic, realistic material model will be generated.
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Field test of resonant frequency monitoring utilizing background vibration in various buildings
Yoshinori Takahashi, Naru Sato, Yasutaka Nakajima
[Abstract]
A number of big intermittent aftershocks often follows a great earthquake. Even if a
certain building escapes from the collapse due to the first main quake, there is no
guarantee that the building can survive from the aftershocks. If it is possible to monitor
the condition of the buildings constantly using some faint random vibration caused by
background vibration such as wind, weak earthquakes and traffic vibration, the safety
will be more secured.
The authors have proposed cumulative harmonic analysis that emphasizes the hidden
resonances in a time-invariant transfer function, and conducted the experimental
monitoring of the resonant frequency in an actual building from the analysis of the faint
random vibration on the building caused by the background vibration. From the results
of comparison with the measured values of the resonant frequency by an active
diagnosis it is confirmed that the estimation of the resonant frequency by the proposed
method obtains an accurate value only including few percent error. This report describes
about the results of field tests on various buildings using the proposed method.
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Identification of dominant noise sources in a diesel power group
Hasan Hassoun, Jaafar Hallal, Mohammad Hammoud, Denis Duhamel
[Abstract]
Reduction of noise and vibration from the diesel power group
has drawn significance attention from manufacturers and
researchers to achieve global specifications. To effectively
reduce the noise level especially at a specific frequency band,
it is essential to identify the sources of noise in addition to the
contribution of various components in the total radiated noise.
The aim of this paper is to investigate and identify the dominant
sources of noise in a diesel group using airborne acoustic
measurements. Analyzing of airborne acoustic signals provides a
promising tool because of its rich information contained within
the acoustic waveform. The presented work first describes the
theoretical sources of noise and the benefits of the time-
frequency domain for the analysis of airborne acoustic signals.
An experimental approach was investigated to identify the
major sources and the contribution of dominant sources in the
total radiated noise upon analysis of the measured data.
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An adaptive structural excitation system as a tool for structure-borne noise research
Marco Norambuena, René Winter
[Abstract]
Engine vibration has been identified as one of the main sources of structure-borne cabin noise. Researchers could benefit from having the ability to reproduce operational vibro-acoustic excitation on the ground instead of requiring flight testing in order to investigate its generation and transmission paths. While realistic excitation generated by engine operation can be obtained through simulations the injection of these forces is not a trivial task. The excitation system needs to compensate for all modifications introduced not only by the amplification chain but also by the interaction of injected forces in different directions to achieve precise excitation in each individual direction. The development of a multi-channel controller able to perform all required compensations can enable a new approach for vibro-acoustic characterization of aircraft fuselages. The adaptive controller presented here is able to provide all counteractions required in order to replicate translational and rotational forces at the injection points on the structure. The system capabilities are demonstrated on a laboratory structure that resembles typical fuselage subcomponents as skin fields, stringers and frames. The following work describes the adaptive MIMO controller, presents results and discusses its applicability.
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Research on Structural Vibration Control Based on Local Stiffness Reinforcement and Local Constrained Damping
Simi Tang
[Abstract]
Structural strengthening and restrained damping have been
widely used for vibration control, but some negative effects
was brought such as weight increase. Stiffness and damping
are both important measures for structural vibration control.
Based on modal and harmonic response analysis, the
technology of design with stiffness and damping for vibration
control is studied in this paper. Under the condition of little
weight gain, the wonderful performance of vibration control
was obtained. This method has significance for structures
with high requirements for weight and vibration control in
engineering.
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Classification of operating conditions of machinery combined with transmissibility function method
Peidong Jia, Jing Wang, Mingmei Han, Xun Wang, Xiaobin Cheng, Jun Yang
[Abstract]
Condition monitoring of machinery is concerned for the purpose of maintenance, productivity and avoiding potential downtime. In the trend of smart factory, the virtual representation of machining process can be built by measuring the vibration. Different working conditions and fault conditions should be identified. Different working conditions have different characteristics with the general signal processing parameters in time and frequency domain. The dimension reduction method like PCA, and the cluster method like k-mean can be used for further classification. In order to accurately classify different operating conditions, the feature extracted from the measured signal should be distinct enough. In this case, multiple vibration sensors have to be implemented, and transmissibility function between different sensors of the dynamic system can be considered as the characteristic parameters. This paper combines transmissibility function and signal processing parameters for classification of different operating conditions to extract more information about operating process.
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Non-contact Thickness Gauging Method based on Frequency Analysis using Underwater Ultrasonic for Steel Structures
Hiroyoshi Yamashita, Takumi Hakamata, Kazuki Abukawa, Tomoo Sato, Sayuri Matsumoto, Kotaro Hoshiba, Takenobu Tsuchiya, Nobuyuki Endoh
[Abstract]
Periodic measurement of thicknesses of steel plates used at
port structures is important to keep their durability. Non-contact
thickness gauging system using ultrasonic for underwater steel
structures has been proposed. However, high accuracy
gauging is difficult under low signal-to-noise ratio (SNR) because
received signals are analyzed in time domain.
In this paper, the novel non-contact measurement method
based on frequency analysis is proposed for solving above
problem. In this method, the thickness of a steel plate is
calculated by analyzing the period of multiple reflection
occurred in a steel plate in frequency domain. Because the
power of noise is averaged in frequency domain, influence of
noise such as a sparse signal could be reduced. Moreover,
because frequency resolution could be determined arbitrarily, it
is expected to perform high performance measurement. The
performance of the proposed method was evaluated using
experimental data. As the result, the thickness of steel plate
could be measured with the error less than 1 mm in SNR of 0 dB,
and it is confirmed the availability of the proposed method.
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3-D numerical simulation of nonlinear elastic wave propagation in locally damaged materials
Benjamin Ankay, Chuanzeng Zhang
[Abstract]
In recent years, nonlinear ultrasonic technique has proven to
be a promising non-destructive testing method for early
damage detection. The sensitivity of this method to
microstructural changes and damages is considerably higher
than the conventional method based on linear ultrasound. In
construction materials, the underlying constitutive relations
can be diverse and complex. Usually, nonlinear ultrasound
experiments are based on an evaluation of the higher-order
harmonics content in the frequency spectrum of the
corresponding ultrasonic signals.
In order to investigate the nonlinear wave propagation in
locally damaged structures, it is often suitable to use
numerical simulations, because analytical solutions are
generally not available and experiments are expensive. In this paper, the three-dimensional (3-D) ultrasonic wave
propagation problem in an elastic half-space with a localized
damaged zone is numerically simulated by a mapped
staggered Chebyshev pseudo-spectral method. To simulate
the semi-infinite elastic half-space, convolutional perfectly
matched layers (CPMLs) are implemented at the artificial
boundaries of the half-space. Classical nonlinear elastic and
non-classical hysteretic material laws are used to take into
account of the localized damage zone within the structure.
The potential applications of the nonlinear ultrasonic
technique are discussed based on the numerical results. Keywords: Ultrasound, Nonlinear elastic wave propagation,
Chebyshev pseudo-spectral
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Temperature dependence of a propagation speed of longitudinal waves in different solids for use as a wedge material in an extreme-temperature-resistant ultrasonic transducer
Oleg Shapovalov, Mate Gaal, Gerald Hönig, Thomas Gradt, Sabine Weiß
[Abstract]
In special cases of an angle beam ultrasonic testing - i. e.
defect detection on hot objects as well as flow measurement
of liquid gases or energy storage mediums - the applied
transducer has to withstand extreme temperatures. Since the
irradiation angle is determined not only by a wedge design,
but also by a sound speed in both the wedge material and
tested object, the developer must take into account the speed
of wave propagation in a wedge material over the whole
temperature range of transducers application.
In this study we investigate the temperature dependence of
speed of longitudinal wave propagation in 10 different
materials in the range from -200 to 400 ∘C. The investigated
materials belong to different material classes (ceramics, glass,
as well as ferrous and non-ferrous alloys) and are all
temperature-resistant in the atmosphere up to at least 600 ∘C,
and therefore can be used as a wedge in an ultrasonic
transducer for use at extreme temperatures.
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Use of Higuer-Harmonic Generation To Detecting Cracks due To Steel Corrosion In Reinforced Concrete
Jaime Ramis, Marina Miró, Jesus Carbajo, Pedro Poveda, Guillem De Vera, Miguel A. Climent
[Abstract]
Reinforced concrete is of vital importance in many civil and
industrial structural applications. Corrosion or other interface
debonding in steel-concrete is a typical failure mode during the
long service period of the structures, which can severely reduce
the load-bearing capacity.
Higuer_Harmonic generation have been used to used to
detecting the onset of microcracking due to steel corrosion in
model reinforced concrete elements. The specimens were of
prismatic shape with a single steel rebar. The corrosion was
forced by admixing an appropriate amount of sodium chloride
at the moment of preparing the concrete mix, and by the
application of an electric field, using a constant current density
power source. The results indicate that the onset of cracking
seems to be accompanied by the appearance of higher-
harmonic generation at the output signal, when the system is
excited by means of an ultrasound wave with a burst central
frequency
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Numerical Analysis of the Nonlinear Restoring Force Based on Near- field Acoustic Levitation
Yuanyuan Liu, Minghui Shi, Kheirollah Sepahvand, Steffen Marburg
[Abstract]
The stability of the floating object is a key factor in near-field
acoustic levitation, which has an extensive application prospect in
manufacturing of micro-electromechanical system parts. This study
presents a numerical study of the restoring force based on the theory
of gas film lubrication. Owing to the eccentricity of the levitator, the
gas film in the radiator coordinate is discontinuous. Thus, a finite
difference scheme is used to solve the Reynolds equation considering
the movement of levitator and to obtain the air pressure distribution.
After a coordinate transformation, the restoring force which acts on
the levitator can be acquired. An experimental rig is constructed to
measure the restoring force with different eccentricity. The
experimental results show that the restoring force increases with the
increase of eccentricity, which shows very good agreement with
numerical results. The predictions indicate that with the increase of
the amplitude and the weight of the levitator, the restoring force also
increases, which leads to a higher stability of the system.
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3D Imaging Method for an Air-Coupled 40 kHz Ultrasound Phased-Array
Gianni Allevato, Jan Hinrichs, Dominik Großkurth, Matthias Rutsch, Jan Adler, Axel Jäger, Marius Pesavento, Mario Kupnik
[Abstract]
We present a three-dimensional far-field imaging method using a 40 kHz 8x8 ultrasound phased-array in air. A waveguide is attached to the array which reduces the aperture to 3 cm x 3 cm and the inter-element spacing to λ/2 for grating lobe free beamforming.
Using transmit beamforming, a pulse echo detection is performed sequentially for each direction of the rasterized region of interest. All transducers are used for pulse transmission and echo reception. The echo signals are FIR band-pass filtered and sinc-interpolated. The conventional Delay & Sum beamformer generates a spatial filtered and summed echo signal. The envelopes of the summed echo signals extracted by Hilbert filters are used for generating B-Scans and 3D C-Scans. The signal processing and image generation is performed by a GPU using OpenGL.
The imaging properties are characterized in an anechoic chamber. The localization of objects is validated for a range of 0.5 to 6 m in a vertical and horizontal field of view of +-50∘. The system is able to detect sound-absorbing objects and objects hidden by obstacles. The implementation has limitations in frame rate and lateral resolution. In this context, future optimization options are outlined.
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The Effect of 40 kHz Ultrasonic Noise Exposure on Human Hearing
Andrew Di Battista
[Abstract]
Purpose: High-Intensity ultrasound is being used for various
applications in the latest consumer technology, including mid-air
haptic feedback. International guidelines concerning maximum SPL
levels for 40 kHz noise are contentious. This pilot audiometry study
aimed to determine the immediate effects that noise exposure at 40
kHz has on human hearing and to help inform concerns of noise
exposure levels in future investigations. This study was conducted in cooperation with the NHS Occupational
Health Service (UK) and was ethically approved by the Health
Research Authority (HRA). Methods: 16 people (10 experiment, 6 control) (ages 24-67 years)
were exposed to high intensity ultrasound emitted from a haptic
device (40 kHz, SPL values ranging from 100-120 dB SPL) for 5
minutes. Pure Tone Audiometry (PTA) was conducted prior to and
immediately after exposure. Results: PTA tests indicate no significant change to hearing sensitivity
at any of the tested frequencies (statistical power >80% in all cases,
with 5dB threshold shift set as a clinically significant risk factor). Conclusion:
Further studies are required to consider different age groups and
further consolidate appropriate sensitivity levels to high frequency
sound. Initial results suggest levels as high as 120 dB SPL could be
used in future investigations.
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Agglomeration of aerosol using intense standing wave field of cylindrical shape
Rintaro Motoi, Takuya Asami, Hikaru Miura
[Abstract]
An ultrasonic dust collector is one of the devices aimed at dust
collection of PM 2.5. Ultrasonic dust collector utilizes a
phenomenon called agglomeration which induces vibration of
fine particles by strong aerial ultrasonic waves to cause collision
between particles to increase particle diameter of fine particles.
In this research, in order to improve the cohesion efficiency, a
standing wave field is formed by using a cylinder and a parallel
reflector plate for an aerial ultrasonic sound source with a rigid
wall, and aerosol is agglutinated. In this paper, we investigated
the change in the particle size and the total number of particles
when irradiating microparticles with ultrasound and changing
the input power to the sound source. As a result, by irradiating
ultrasonic waves to the microparticles, the total number of
microparticles decreased as compared with before ultrasonic
irradiation. Also, when determining the ratio of distribution by
range in the whole area, the central diameter was about 1.50
µm when no ultrasonic irradiation was applied, but the central
diameter was about 2.65 µm when input power was 20 W. From
this fact, it was confirmed that there is an agglomeration effect
of fine particles by ultrasonic irradiation.
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Development of aerial ultrasonic source with cylindrical radiation surface for ultrasonic agglutination
Takuya Asami, Hikaru Miura
[Abstract]
Abstract: We developed a high-power aerial ultrasonic source
with cylindrical radiation surface for using ultrasonic
agglutination. The developed ultrasonic source consists of a
bolt-clamped Langevin-type longitudinal vibration transducer,
an exponential horn, a resonance rod, and a cylinder typed
vibrating plate. Both ends of the cylinder typed vibrating plate
have flanges that do not vibrate. The vibration source resonates
at around 28 kHz. The feature of the developed ultrasonic
source can be used for ultrasonic agglutination by preventing
air leakage by connecting directly to the pipe, and can obtain
a high sound pressure in the space inside the cylinder typed
vibrating plate. In this presentation, we describe the design
method of the cylinder typed vibrating plate and the vibration
characteristics of the developed ultrasonic source. In the
design, the dimensions of the flanges that do not vibrate, and
the dimensions that the cylinder part vibrates and generated
high sound pressure were obtained using finite element method
(COMSOL). In the vibration characteristics, we found that the
designed flanges almost no vibrate. In addition, the obtained
sound pressure was around 7 kPa (171 dB) at an input electrical
power of 9 W in the space inside the cylinder typed vibrating
plate.
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Removal of extraneous matter by ultrasonic washing in running water
Hosaka Hidenobu, Takuya Asami, Hikaru Miura
[Abstract]
Among the fabrication methods of additive manufacturing
technology is the fabrication method called fused deposition
mod (FDM). In FDM, support materials that act as splints are
required depending on the shape of the shaped object, and
the support material must be removed after the model has
been completed. Removal of support material can be done
cleanly by using a water-soluble support material, but it takes a
long time. In order to remove those support materials, a washing
method using running water type ultrasonic washing as a
washing method that can be performed locally in a short time
by using cavitation or acoustic flow generated when strong
sonic waves are irradiated in running water. In this study,
washing effect of running water type ultrasonic washing was
investigated using polyvinyl alcohol as a model of soiling. As a
result of the experiment, when ultrasonic waves were irradiated
in the running water, a high washing effect was obtained as
compared with the case which ultrasonic waves were not
irradiated in running water.
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DNA amplification by sound and ultrasound frequency vibration
Ryu Kobayashi, Seiji Yoneda, Shigeo Yamaguchi
[Abstract]
In the Polymerase Chain Reaction (PCR) method, which is a
strong DNA amplification technology, a step with a high
temperature of 94 ∘C is inevitably included to denature
double-stranded DNA to single-stranded DNA. However, the
thermal denaturation has problems causing DNA damage
and enzyme deactivation. To overcome them, we have
proposed and developed a new technology for DNA
denaturation and amplification, namely, vibration-driven
DNA amplification method (the frequency ranges from 100
Hz to 25 kHz). In our method, DNA is denatured and amplified
by vibrating a micro tube with at sound to ultrasound
frequencies using a wave transducer. In the micro tube,
DNA, enzyme, and other materials are mixed and dissolved
in water. Using the DNA amplification system we developed,
we examined the optimum condition for DNA denaturation
and amplification by changing amplitude, frequency and
time of vibration. Maximum amplification rates were
obtained at frequencies from 200 Hz to 500 Hz. The analysis
through the electrophoresis with a microchip demonstrated
that DNA amplification rate was achieved to 1.7 for a cycle
(2 is the ideal value). Moreover, through the agarose gel
electrophoresis, we found that no DNA was damaged by
vibration.
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Light scattering observation of oscillation of a bubble in acoustic cavitation cloud
Takanobu Kuroyama
[Abstract]
High intensity ultrasound generates oscillating fine bubbles
called acoustic cavitation. The oscillation behavior of the
bubbles is important in the application of the acoustic
cavitation, such as degradation of pollutants and production of
chemicals. Although the spatially and periodically averaged
oscillation of the bubbles is observed by optical holography and
light scattering method, the oscillation characteristics of a
bubble in acoustic cavitation cloud is not clear because of the
difficulty of measurement. The single bubble behavior and the
variation of the oscillation are desired to be measured for the
construction of the bubble cloud model. To overcome the
problem, we propose an improved light scattering
measurement technique employing a confocal optics. In this
system, a laser beam irradiates the bubble cloud. The light
scattered by a bubble in a tiny measurement volume, which is
formed by the confocal optics, is measured. The detected
scattered light intensity is proportional to the square of the
bubble radius. Thus, the oscillation of a bubble in the cloud can
be observed with the scattered light. The measurement
principle and the limitation of the method is described. The
transient behavior, the intensity, and the variation of the bubble
oscillation are discussed with the experimental result.
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Development of longitudinal-torsional vibration source with a helical slits transmission rod
Shinya Oishi, Takuya Asami, Hikaru Miura
[Abstract]
It has been reported that in ultrasonic welding using planar
vibration by a two-dimensional vibration locus, the joining
performance such as an increase in joining area and
improvement in welding strength is improved. Therefore, the
authors focused on generating planar vibration by
simultaneously driving one-dimensional vibration loci with
different directions by an ultrasonic vibration source using a
transmission rod having helical slits. In this study, an analysis
model of the ultrasonic vibration source having the helical slit
was created by the analysis using the finite element method
(FEM method). Then, by changing the depth of the slits, the
depth of the slits in which the ratio of the longitudinal vibration
displacement and the torsional vibration displacement at the
tip of the transmission rod coincide with each other in the
longitudinal vibration mode and the torsional vibration mode.
From the analysis results, a transmission rod having helical slits of
the selected depth was prototyped, the resonance
characteristics of the ultrasonic vibration source using it, and the
vibration trajectory at the tip of the transmission rod were
studied fundamentally. As a result, planar vibration was able to
be generated by an ultrasonic vibration source having helical
slits.
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Intracranial Nonthermal Ablation Mediated by Transcranial Focused Ultrasound and Phase-Shift Nanoemulsions
Tyrone Porter, Chenguang Peng, Tao Sun, Natalia Vykhodtseva, Chanikarn Power, Yongzhi Zhang, Nathan McDannold
[Abstract]
One of the major challenges of using transcranial focused
ultrasound for ablating brain tumors is that the pressure and
time-averaged power required to generate lesions is relatively
high and could induce skull heating and pain. In this study, we
investigate the utility of phase-shift nanoemulsions (PSNE), lipid-
coated nanodroplets composed of perfluorobutane, to serve
as cavitation nuclei and enable focal ultrasound-mediated
nonthermal ablation in the brain using transcranial focused
ultrasound. A transcranial focused ultrasound transducer
operating at 740 kHz and 12 CD-1 mice were used to test the
vaporization of PFB PSNE and the corresponding damage. The
PNP was increased stepwise up to 1.8 MPa and the broadband
emission from the inertial cavitation nucleated by vaporized
PFB-based PSNE were recorded to quantify the inertial
cavitation level. A significant elevation of broadband emission
was noticed first at PNP = 1.25 MPa and increased dramatically,
indicating strong inertial cavitation. The corresponding
biological damage was analyzed with H&E staining and
exhibited confined ischemic and hemorrhagic lesions. The
damage was found to correlate with the acoustic pressure and
depended upon the administered PFB PSNE concentration.
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The use of magnetic nanoparticles to increase the efficiency of magneto-ultrasonic heating.
Katarzyna Kaczmarek, Milan Timko, Tomasz Hornowski, Arkadiusz Józefczak
[Abstract]
Hyperthermia treatment is a medical procedure that involves
generation of heat between 41∘C and 45∘C. This temperature
rise leads to several physiological changes within the tumor
tissues causing their weakening. Weaken cells are more
susceptible for other cancer therapies. Hyperthermia within the
tissue can be applied through different mechanisms. It can be
induced by ultrasound sonication, radiowaves or magnetic
field. In this work magnetic, ultrasound and novel magneto-
ultrasonic source of hyperthermia are discussed. To investigate
the magneto-ultrasonic thermal effect tissue mimicking
phantom, doped with magnetic nanoparticles, is simultaneously
irradiated with the focused ultrasonic wave and the alternating
magnetic field. During this magneto-ultrasonic heating even
more effective thermal effect can be observed. This novel
simultaneous magneto-ultrasonic heating has a big application
potential in treating cancer at a lower magnetic particle
concentration and in shorter exposure time. The coupling of
magnetic and ultrasound hyperthermia gives possibility for
developing the new, innovative thermal therapy.
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A study on the relationship between speed of sound image and evaluation value of moving sound image
Mika Kurabayashi, Kenji Muto
[Abstract]
The purpose of this study is to develop a high presence sound
system with a personal computer for an emergency drill
simulator using a virtual reality exhibition system. Some studies
have reported the systems to provide acoustical high presence,
however, the method used in these studies is needed powerful
computers to calculate a tremendous of processing. In order to
simplify these systems, we propose the method to reduce the
processing of the moving sound image. The moving sound
image was given by the discrete coordinates of not-
continuously sound image with the proposed method. We
conducted the experiment to compare with the pairs of stimuli
for the evaluation of the reality of the moving sound image. In
the experiment, some moving sound source with different speed
were compared to obtain the threshold number of the discrete
coordinate of the moving sound image for acoustical high
presence. We analysed the relationship between the
evaluated value and the speed with the pair comparison
method. The result showed that the evaluation of the reality
depended on the speed of the sound image.
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A Physically Motivated Approach for Binaural Simulation of Moving Sound Sources and Receivers
Christoph Urbanietz, Gerald Enzner
[Abstract]
In this paper we derive a physically motivated approach to handle the simulation of moving sound sources and receivers including phenomenas like the Doppler effect in audio signal processing by using the concept of retarded time in a very rigorous way.
After recapitulating the physical basis of moving point-like sound sources and receivers we transfer an alternative formulation of the received signal based on superposition of Green’s functions to a system theory point of view. A first proof of the validity of this formulation is shown for continuous-time systems. Afterwards we consider the discrete-time version and reveal the strengths and the limits of this approach. Finally a binaural receiver is added to the model.
The proposed formulation is anchored on a physical basis and therefore it handles physical effects inherently. Beside the correct reproduction of the Doppler shift it also includes the physically attested amplitude shift of the received acoustic signal for a moving sound source and mislocalization due to limited speed of sound.
Nevertheless the specific implementation for discrete-time signal processing can be challenging. We present possible implementations and investigate the limits on some examples. The simulated results are measured against physically predictable features.
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The open-source Virtual Acoustics (VA) real-time auralization framework
Jonas Stienen, Frank Wefers, Michael Vorländer
[Abstract]
Virtual Acoustics (VA) is a free and open-source real-time auralization framework developed for scientific research in acoustics. Its main focus lies on the computer-aided generation of virtual acoustic scenes that can be explored interactively. By design, VA is modular allowing for high flexibility, fine-grained control and universal customization options. Audio rendering and audio reproduction is strictly separated. The combination of available modules and the possibility to use them simultaneously and with multiple instances make it applicable for comparative studies and rapid design processes. Solutions are only limited by the available hardware and computational resources at hand. VA natively provides a C++ interface and binary packages can be downloaded for Windows as a standalone application with remote control access via network enabled. It is readily employable for audio-visual Virtual Reality systems driven by commercial designers like Unity and Unreal, but can also be operated from Matlab and Python environments. The presented poster gives an overall view of the framework, outlines the modular concept and lists the interfaces provided.
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Tournament Formats as Method for Determining Best-fitting HRTF Profiles for Individuals wearing Bone Conduction Headphones
Tray Minh Voong, Michael Oehler
[Abstract]
An approach is presented how to practically determine best-fitting HRTF profiles for individuals wearing a bone conduction headphone. Such headphones may be particularly useful for visually impaired people (e.g., for navigation applications), as the outer ear is not covered and the perception of environmental sounds is not affected. For a fast and user-friendly identification of best-fitting HRTFs, different tournament methods are compared. Relevant validation parameters are localization accuracy, coherency of virtual auditory images, envelopment and externalization. There are several studies that investigate (a) aspects of spatial sound perception with bone conduction headphones and (b) tournament systems as selection procedure for best-fitting HRTFs, but to our knowledge there is no study that analyzes the interdependence of (a) and (b). In a mixed factorial design, different tournament methods were compared for two headphone types (bone conduction headphone vs. open circumaural headphones). Compared with other tournament formats, the Swiss-system tournament produced more accurate results than the knockout tournament in less time than the robin-tournament system. There was no significant difference for the headphone types regarding the tournament formats. Hence, tournament methods (particularly the Swiss-system tournament) seem to be suitable to determine best-fitting HRTF profiles for the use with bone conduction headphones.
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Spatial Upsampling of Individual Sparse Head-Related Transfer Function Sets by Directional Equalization
Christoph Pörschmann, Johannes M. Arend, Fabian Brinkmann
[Abstract]
Determining full-spherical individual sets of head-related transfer functions (HRTFs) based on sparse measurements is a prerequisite for various applications in virtual acoustics. However, when applying HRTF interpolation in the spatially continuous spherical harmonics (SH) domain, the number of measured HRTFs limits the maximal assessable SH order. Such a limited SH representation shows restricted spatial resolution and causes perceptual artefacts related to coloration and source localization, among others.
In a previous publication we presented the SUpDEq method (Spatial Upsampling by directional equalization), which reduces these artifacts by a directional equalization based on a spherical head model prior to the SH transform. This removes direction-dependent temporal and spectral components and thus reduces the spatial complexity of the HRTF set enabling improved interpolation of HRTFs already at low SH orders. A subsequent de-equalization recovers energy in higher spatial orders that was discarded in the sparse HRTF set.
In this study we analyze 96 individual HRTF sets and investigate to what extent the performance of SUpDEq determined for dummy heads can be transferred to individual sets. We show that the SUpDEq method clearly outperforms common SH interpolation of HRTFs with respect to the spectral and temporal structure as well as modeled localization performance.
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