House of Acoustics Annual Report 2008

House of Acoustics
Department of Electrical Engineering
Technical University of Denmark
Annual Report 2008

ACOUSTIC TECHNOLOGY
&
HEARING SYSTEMS, SPEECH AND
COMMUNICATION
Department of Electrical Engineering
TECHNICAL UNIVERSITY OF DENMARK
Edited by Finn Jacobsen in February/March 2009
ANNUAL REPORT
2008

Front page: Predominant mode (2, 2) of a doubly curved plate with cross-stiffeners in the x- and ydirection.
See the description of the project ‘Modelling structural acoustic properties of loudspeaker
cabinets’ on p. 10.
Acoustic Technology and Hearing Research, Speech and Communication, Department of Electrical Engineering,
Technical University of Denmark, Building 352, Ørsteds Plads, DK-2800 Kgs. Lyngby,
Denmark
Telephone +45 4525 3930
Direct tel. +45 4525 39xx
Telefax +45 4588 0577
Web-server http://www.elektro.dtu.dk/English/research/at.aspx
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CONTENTS
3
Page
Chairmen’s Report 5
Staff 7
1 Research 9
Acoustic Technology 9
Structureborne sound 9
Acoustic Transducers and Measurement Techniques 11
Noise and Noise Control 16
Room Acoustics 17
Building Acoustics 20
MSc Projects 22
Publications 28
Hearing Systems, Speech and Communication 31
Auditory Signal Processing and Perception 31
Speech Perception 36
Audiology 40
Audio-Visual Speech and Auditory Neuroscience 41
Objective Measures of Auditory Function 42
MSc Projects 43
Publications 46
2 Teaching Activities 49
Introductory Level 49
Advanced Level 50
Lecture Notes Issued in 2008 52
Appendix A: Extramural Appointments 53
Appendix B: Principal Intramural Appointments 55

CHAIRMEN’S REPORT
In the middle of 2008 Department of Electrical Engineering was reorganised; the five sections
were replaced by eight research groups. A local consequence is that the section Acoustic Technology
was replaced by two groups, ‘Acoustic Technology’ and ‘Hearing Systems, Speech and Communication’,
where the latter essentially corresponds to Centre for Applied Hearing Research. At the same time
the Head of the Section, Mogens Ohlrich, resigned from the position he had held for seven years. Mogens
was succeeded by the two group chairmen Finn Jacobsen (Acoustic Technology) and Torsten Dau
(Hearing Systems, Speech and Communication). Seen from the outside it may be slightly confusing that
‘Acoustic Technology’ is now only a part of the former section Acoustic Technology. On the other
hand it might have been even more confusing if all names had been changed. Of course the two groups,
which share buildings and research facilities, continue to have a very close relationship; and evidently
we still offer the successful international MSc programme ‘Engineering Acoustics’ which involves
courses and projects given by both groups.
Another significant event in 2008 was that DTU’s former research dean Kristian Stubkjær replaced
Jørgen Kjems as Head of Department of Electrical Engineering by 1 December. Jørgen Kjems
was appointed as Head of Department ad interim in mid-2007; his task was to solve internal organisational
problems resulting from DTU’s merging with other research institutions in the beginning of
2007. There are reasons to believe that he succeeded, and that the resulting restructured organisation
with Kristian Stubkjær as the Head of Department will be thriving.
On the staff side there have been significant changes. We are in the middle of a generational
handover, with Mogens Ohlrich and Torben Poulsen now on part time. Moreover, as mentioned in last
year’s report our long time colleague Jens Holger Rindel left the University at the end of 2007 in order
to concentrate on Odeon, a spin-off company from his room acoustic research; and in September 2008
another long time colleague Anders Christian Gade also left us after a one-year leave of absence to concentrate
on work in a joint consulting company. On the other hand Jonas Brunskog was appointed associate
professor in the spring of 2008, and Cheol-Ho Jeong was appointed assistant professor at the end
of the year. Jonas and Cheol-Ho’s research activities are in architectural acoustics and structureborne
sound, and together they give an introductory course for students with a background in civil engineering,
‘Building Acoustics’ (given the first time by Jens Holger Rindel in 2007), the course ‘Architectural
Acoustics’, and, from 2009, ‘Environmental Acoustics’; they also contribute to ‘Sound and Vibration’.
At the end of the year Jörg Buchholz was appointed associate professor. Jörg’s research activities are in
communication acoustics, human sound perception and audiology. In terms of teaching, he will be taking
over Torben Poulsen’s course on ‘Acoustic Communication’, which is a compulsory part of the
MSc programme Engineering Acoustics.
Research grants received in 2008 include a three-year project on ‘Spectro-temporal processing of
complex sounds in the human auditory system’ supported by the Danish Research Council (FTP); a
five-year cooperation agreement between CAHR and the three Danish hearing aid companies GN Re-
Sound, Oticon and Widex that supports research and education in the field of human acoustic communication
by 12.4 million kr; and ‘A binaural master hearing aid platform’ supported by the Oticon foundation.
New research projects started in 2008 include Iris Arweiler’s PhD project ‘Processing of spatial
sounds in the impaired auditory system’, co-financed by DTU, the Graduate School on Sense Organs,
Nerve Systems, Behaviour, and Communication (‘SNAK’) and Phonak; Lu Yuan’s industrial PhD project
with Bang & Olufsen on the modelling of structural acoustic properties of loudspeaker cabinets;
and David Pelegrin García’s PhD project ‘Speech comfort in class rooms’. This project, which is financed
by the Swedish insurance and research organisation AFA, is a part of a larger cooperative project
together with the Logopedics, Phoniatry and Audiology group at Lund University, Sweden. By the
end of the year we received a PhD scholarship from Department of Electrical Engineering for Efrén
Fernandez Grande for a project on acoustic holography.
Two of our PhD students defended their theses in 2008: in November Torsten Elmkjær defended
his thesis on helmets with active noise control, and about one month later it was Gilles Pigasse’s turn to
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defend his thesis ‘Cochlear delays in humans using otoacoustic emissions and auditory evoked potentials’.
In the EU-funded PhD student exchange programme ‘European Doctorate in Sound and Vibration
Studies II’ we hosted two visiting PhD students in 2008: Eleftheria Georganti, University of Patras,
Greece, studied room transfer functions and reverberant signal statistics supervised by Finn Jacobsen;
and Lars-Göran Sjökvist, Chalmers University, Sweden, studied structural sound transmission and attenuation
in lightweight structures, supervised by Jonas Brunskog and Finn Jacobsen. Other guest PhD
students include Bastiaan Warnaar, University of Amsterdam, who spent three months studying computational
models of inner-ear hearing loss, and Yong-Bin Zhang, Hefei University of Technology, China,
who began a one-year stay in September in which he studies near field acoustic holography supervised
by Finn Jacobsen.
Once again Steven Greenberg, The Speech Institute, Berkeley, CA, USA, spent three months
working together with Thomas Ulrich Christiansen on speech research. Another guest researcher,
Arturo Orozco Santillán, Universidad Nacional Autónoma de México, worked together with Finn
Jacobsen on a project on sound intensity during the summer.
In October, CAHR celebrated its 5th birthday. Professor Brian Moore from Cambridge University
gave a guest speech, and most CAHR researchers presented their current projects with talks or
posters. Afterwards all guests, colleagues and friends were invited to test various special drinks at the
cocktail party in the ‘CAHR bar’.
The interest amongst foreign visiting students on short-term-stays continues to be high and we
have a satisfactory intake of foreign and Danish students on our international two-year MSc-programme
in Engineering Acoustics: this year we ‘house’ twenty-four foreign master students. We gratefully acknowledge
that three of the foreign students in our international MSc-programme in Engineering
Acoustics are receiving a DTU Student Sponsorship (DSS-sponsorship) financed by the companies
Widex, Oticon, and Oticon Polska.
Mogens Ohlrich Finn Jacobsen Torsten Dau
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Head of Acoustic Technology
Finn Jacobsen, MSc, PhD, Dr. Techn., Associate
Professor
Head of Hearing Systems, Speech and Communication,
and Head of Centre for Applied
Hearing Research
Torsten Dau, Dr. rer. nat. habil., Professor
Associate Professors
Finn T. Agerkvist, MSc, PhD
Hans-Heinrich Bothe, Dr. habil., PD
Jonas Brunskog, MSc, PhD, docent
Jörg M. Buchholz, Dr. Ing.
Anders Christian Gade, MSc, PhD
Mogens Ohlrich, BSc, PhD
Torben Poulsen, MSc
Assistant Professors
James M. Harte, BSc, PhD
Cheol-Ho Jeong, MSc, PhD
Scientists and Research Assistants on External
Grants
Salvador Barrera Figueroa, MSc, PhD (DFM)
Thomas Ulrich Christiansen, MSc, PhD
Dimitrios Christoforidis, MSc
Marton Marschall, MSc
Tarmo Saar, MSc
PhD Students
Iris Arweiler, Dipl.-Ing.
Torsten Haaber Leth Elmkjær, MSc (Terma)
Sylvain Favrot, MSc
Morten Løve Jepsen, MSc
Jens Bo Nielsen, MSc
David Pelegrin García, BSc, MSc
Tobias Piechoviak, MSc
Gilles Pigasse, MSc
Sébastien Santurette, MSc
Olaf Strelcyk, MSc
Eric Thompson, BSc, MSc
Sarah Verhulst, BSc, MSc
Industrial PhD Students
Lola Blanchard, MSc (B&O ICEpower)
Helen Connor, MSc (Widex)
Lars Friis, MSc (Widex)
Yu Luan, BSc, MSc (Bang & Olufsen)
Guilin Ma, BSc, MSc (GN ReSound)
STAFF
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Visiting PhD Students
Eleftheria Georganti, University of Patras,
Greece
Lars-Göran Sjökvist, Chalmers University,
Gothenburg, Sweden
Bastiaan Warnaar, University of Amsterdam, the
Netherlands
Yong-Bin Zhang, Hefei University of Technology,
China
Visiting Scientists
Steven Greenberg, The Speech Institute, Berkeley,
CA, USA
Arturo Orozco Santillán, Universidad Nacional
Autónoma de México, Mexico City
Emeritus Professors
Knud Rasmussen, MSc (DFM)
Administrative and Technical Staff
Nadia Jane Larsen, Secretary
Tom A. Petersen, Assistant Engineer
Jørgen Rasmussen, Assistant Engineer
Aage Sonesson, Assistant Engineer
Caroline van Oosterhout, Project secretary

1. RESEARCH
The groups ‘Acoustic Technology’ and ‘Hearing Systems, Speech and Communication’ are parts
of Department of Electrical Engineering at DTU and share research and teaching facilities. The research
comprises investigations of generation, propagation and effects of sound and vibration, as well as auditory
signal processing, speech, and perception of sound. The research may involve theoretical analyses,
numerical techniques, subjective experiments and advanced measurement techniques.
ACOUSTIC TECHNOLOGY
Acoustic Technology is concerned with structureborne sound, passive and active noise control,
outdoor sound propagation, transducer technology (loudspeakers, microphone calibration), and acoustic
measurement techniques (sound intensity, beamforming and acoustic holography), as well as room
acoustics and building acoustics.
Structureborne Sound
Minimisation of vibroacoustic feedback in hearing aids
Lars Friis
Supervisors: Mogens Ohlrich and Finn Jacobsen
Feedback problems in hearing aids are often caused by vibroacoustic transmission from the
loudspeaker to the microphones. The objective of this industrial PhD-project has been to examine these
mechanisms. This has been approached by developing a vibroacoustic model of a hearing aid. The
model combines finite element analysis and traditional methods for modelling acoustics and vibration
with an alternative, relatively new method, ‘fuzzy structures’, which is intended for predicting the vibrations
of a deterministic ‘master’ structure with attached complex ‘fuzzy’ substructures with partly
unknown dynamic properties. The main effect of the fuzzy substructures is to introduce high damping
in the response of the master structure. An important part of the fuzzy theory is the modelling of fuzzy
substructures attached to the master through a continuous interface. It has been shown that the continuous
interface reduces the damping effect of the fuzzy substructures. Furthermore, an experimental
method has been developed for estimating the fuzzy parameters of complex substructures. Simulation
results of the model of the hearing aid show a very good agreement with measurements.
The project is an industrial PhD project carried out in cooperation with the hearing aid manufacturer
Widex A/S, with Lars Baekgaard Jensen as the industrial supervisor. The defence takes place in
the spring of 2009.
Experimental arrangement for electro-acoustic measurements on a resiliently suspended ‘behind the ear’
hearing aid.
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Modelling structural acoustic properties of loudspeaker cabinets
Yu Luan
Supervisors: Mogens Ohlrich and Finn Jacobsen
The objective of this industrial PhD-project is to develop techniques for predicting the structural
acoustic response of loudspeaker cabinets of irregular shapes. The purpose is to minimise unwanted
audible vibration of the cabinet that may reduce the sound quality of the loudspeaker. The project attempts
to develop a model for simulating the structural acoustic properties in the low to mid-frequency
ranges. Initially, a simple approach has been pursued for examining cross-stiffened rectangular panels
and slightly curved shells. This involved the development of an improved ‘smearing method’ for predicting
the natural frequencies of simply supported cross-stiffened rectangular plates. In contrast to Szilard’s
original method, the improved method takes stiffeners in both the x- and y-direction into account
in calculation of the equivalent bending stiffness. With finite element calculations and measurement
results as a reference, the improved method has been found to give a higher accuracy of the natural frequencies
than Szilard’s method; the prediction errors are approximately halved. Moreover, it has been
found that the improved method can be applied for doubly curved rectangular panels with crossstiffeners.
The results show that engineering accuracy can be obtained with relatively limited computational
effort. As an example the figure on the front page shows the calculated modal deformation of a
(2, 2) mode of a doubly curved cross-stiffened plate.
The project is an industrial PhD project carried out in cooperation with Bang & Olufsen, with
Søren Bech, Mogens Brynning, Lars F. Knudsen and Gert Munch as the industrial supervisors.
Vibratory strength of machine sources and structural power transmission
Mogens Ohlrich and Tarmo Saar
This ongoing work focuses on a practical characterisation of vibrating machine sources and on
simple techniques for predicting the vibratory power that such sources inject into connected receiving
structures. The developed characterisation that specifies the ‘terminal power’ of a vibrating source, is
incorporated and further developed in the project ‘VibPower’ as the chosen technique to be used for
predicting vibratory power transmission from machinery. Vibration of complex machines is mostly
measured and the developed technique makes use of the fact that source characterisation and prediction
of power transmission can be simplified if all mobility cross-terms and spatial cross-coupling of source
velocities can be neglected in the analysis. For structurally compact machines, however, the influence
of cross-coupling may be important, at least at low frequencies. Such cross-coupling is examined in its
most simple form for a rotor-type source with two flange mounts; see the figure below.
This work is a continuation of Tarmo Saar’s MSc project.
X
impeller
front
terminals
Z
front
bearing
diffuser
stator
10
brushes
1 2
shaft
washer
Y
rotor
rear
bearing
rear
terminals
Schematic illustration of a rotor-type machine source in the form of a high-speed ‘vacuum-motor’ with flange
terminals at the front and rear ends.

Determination of structureborne sound power and reduction of vibrational power transmission
(VibPower)
Mogens Ohlrich
The purpose of this project is to develop a source strength characterisation method and to develop
design guides for reducing the transmission of structureborne sound from large machines to their foundation
structures. In the project a Round Robin Test has been carried out of the source-strength measurement
technique based on previous research at the Acoustic Technology (AT). The test arrangement
comprises an industrial gearbox and a lightweight foundation placed at the laboratory of Machine Design,
Tampere University of Technology (TUT). The Round Robin Test was carried out in succession
by AT, TUT, and VTT, and the developed source strength technique was used by TUT and VTT on
three industrial test cases. Power transmission in rotational coordinates can usually be neglected, and
this has been verified by simulation studies conducted by VTT using a large finite element model of a
complicated engine-foundation arrangement. It was found that the power in rotational coordinates is
less than one percent of the translational power. The project was finished in March 2008.
In addition to the mentioned research institutes the following industrial partners participate: ABB
Industry OY, Wärtsilä Marine Finland OY, and Moventas OY, all from Finland. The National Technology
Agency of Finland finances this research with contributions from the three companies.
Acoustic Transducers and Measurement Techniques
Time variance of loudspeaker suspension
Finn Agerkvist
The electrodynamic loudspeaker is conventionally described by a lumped parameter model.
However, the behaviour of the loudspeaker is in fact much more complicated than this model would
suggest. At high levels the loudspeaker becomes nonlinear, and the nonlinearity of the compliance of
the suspension is one of the major sources of distortion. Unfortunately the compliance is known to vary
with time depending on the signal fed to the loudspeaker. Three loudspeakers, identical except for the
type of surround, have been investigated with respect to changes in compliance. Measurement both in
the linear domain as well as in the nonlinear domain have been carried out At low levels the compliance
changes significantly with the level of the measurement signal, and very little memory is observed. In
the nonlinear domain compliance also increases with the level of the measurement signal, but previous
results suggesting that the shape of the nonlinearity also changes have not been confirmed by this investigation.
Creep, a viscoelastic effect in loudspeaker suspension parts
Finn Agerkvist
This project investigates the viscoelastic behaviour of loudspeaker suspension parts, which can
be observed as an increase in displacement far below the resonance frequency. The creep effect means
that the suspension cannot be modelled as a simple spring. The need for an accurate creep model is even
larger as it is attempted to extend the validity of loudspeaker models into the nonlinear domain. Different
creep models are investigated and implemented both in simple lumped parameter models and in
time domain nonlinear models, and the simulation results are compared with a series of measurements
on three version of the same loudspeaker with different thickness and rubber type used in the surround.
The project is carried out in collaboration with Tymphany A/S.
Error correction of loudspeakers
Bo R. Petersen
Supervisor: Finn Agerkvist
In order to apply nonlinear compensation in loudspeakers, the compensation algorithm needs
very accurate information about the loudspeaker parameter. Some of these parameters have a strong
time dependence, and it is therefore important to track them in real time. A system identification algorithm
has been developed and tested, and the convergence has been tested with musical signals. In order
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to help the algorithm tracking the changes in the suspension compliance an investigation has been carried
out on how the small signal compliance changes with the mechanical and electrical load of the
loudspeaker. Another effect in the loudspeaker is that the nonlinear force factor depends on the current
in the voice coil; this effect has been investigated in measurements and simulations.
Bo Petersen has been enrolled as a PhD student at Esbjerg Institute of Technology, Aalborg University,
with Jens Arnsbjerg as the main supervisor. The thesis was defended successfully on 13 October
2008.
Determination of microphone responses and other parameters from measurements of the membrane
velocity and boundary element calculations
Salvador Barrera-Figueroa, Finn Jacobsen, and Knud Rasmussen
Left: the measurement setup; right: acoustic centre of an LS1 microphone. Solid line: experimental estimate; line
with square markers: numerical estimate assuming a Bessel-like movement; line with circular markers: numerical
estimate obtained assuming a uniform velocity; and line with star-markers: estimate from the hybrid method.
Normalised amplitude and phase of the velocity of the membrane of a microphone at different frequencies. Solid
line: normalised amplitude; dash-dotted line: phase. Left: LS1 microphone; right: LS2 microphone.
Numerical calculations of the pressure, free-field and random-incidence response of condenser
microphones are usually carried out on the basis of an assumed displacement distribution of the microphone
diaphragms. The conventional assumption is that the displacement follows a Bessel function, and
this assumption is probably valid at frequencies below the resonance frequency. However, at higher
frequencies the diaphragm is heavily coupled with the thin air film between the diaphragm and the back
plate and with resonances in the back chamber of the microphone. A solution to this problem is to
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measure the velocity distribution of the membrane by means of a non-contact method, for instance laser
vibrometry. The measured velocity distributions can be used together with a numerical formulation
such as the Boundary Element Method for estimating the microphone response and other parameters as
e.g. the acoustic centre. The velocity distributions of a number of condenser microphones have been
measured using a laser vibrometer. This measured velocity distribution was used for estimating the microphone
responses and parameters. The agreement with experimental data is good. This method can be
used as an alternative for validating the parameters of the microphones determined by classical calibration
techniques.
Relation between the radiation impedance and the diffuse-field response of a condenser microphone
Salvador Barrera-Figueroa, Finn Jacobsen, and Knud Rasmussen
The relation between the diffuse-field response and the radiation impedance of a microphone has
been investigated. Such a relation can be derived from classical theory. The practical measurement of
the radiation impedance requires a) measuring the volume velocity of the membrane of the microphone,
and b) measuring the pressure on the membrane of the microphone. The first measurement is carried out
by means of laser vibrometry. The second cannot be implemented in practice. However, the pressure on
the membrane can be calculated numerically by means of the boundary element method. In this way, a
hybrid estimate of the radiation impedance is obtained. The resulting estimate of the diffuse-field response
is compared with experimental estimates of the diffuse-field response determined from measurements
in a reverberant room and with numerical calculations. The different estimates are in good
agreement at frequencies below the resonance frequency. Although the method may not be of great
practical utility, it provides a useful validation of the estimates obtained by other means.
Diffuse-field correction of microphones. The thick solid line is the diffuse-field correction determined using reciprocity;
the dash-dotted line is the random-incidence correction; the line with circular markers is the randomincidence
correction calculated using the boundary element method; the line with star markers is the diffuse-field
correction determined from the radiation resistance. Left: LS1 microphone; right: LS2 microphone.
High quality active earphone system
Lola Blanchard
Supervisors: Finn Agerkvist and Finn Jacobsen
In order to improve the sound quality of concha headphones, a study of their current limitations
has been carried out. One of the most used concha headphones, the Ipod ‘earbud’, has been studied, in
parallel to the Bang & Olufsen ‘A8’. The limitations of the measurement equipment have been looked
at as well. The target response has been chosen to be the response of a Sennheiser HD 650. The main
focus has been on measuring the leak and defining the coupling between the ear and the headphone by
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means of measurements and modelling. The influence of the back volume has also been investigated,
and it has been observed that the vented box headphone does not behave as a vented loudspeaker of
normal size. Further investigations should be carried out, especially on the coupling between the front
and rear of the loudspeaker.
This industrial PhD project is carried out in cooperation with B&O ICEpower, with Lars Brandt
Rosendahl Hansen and Anders Røser Hansen as the industrial supervisors.
New strategies for feedback suppression in hearing instruments
Guilin Ma
Supervisors: Finn Jacobsen and Finn Agerkvist
Audible feedback is among the most prominent problems with hearing instruments. Unchecked
feedback can lead to system instability and cause very unpleasant whistling or howling. The objective
of this industrial PhD project is to examine and develop new ways of improving feedback suppression
techniques. Two aspects have been investigated in 2008. The first is about modelling the physical feedback
path. The feedback path is subject to dramatic environmental changes, for instance when the user
picks up a phone and is therefore very difficult to model. A new model based on reflection assumptions
has been proposed, and this model shows much better accuracy in modelling the measured dynamic
feedback paths. The second aspect is about feedback adaptation. Feedback cancellation suffers from
correlation problems, and therefore on-line estimation of the feedback path is problematic. A new approach
for decorrelating the signals has been proposed. The proposed method shows an excellent performance
in the first test. The matter will be investigated further in 2009.
The project is carried out in cooperation with GN ReSound with Fredrik Gran as the industrial
supervisor.
Danish primary laboratory of acoustics (DPLA)
Knud Rasmussen and Salvador Barrera-Figueroa
DPLA was established in 1989 by the Agency for Development of Trade and Industry as a cooperation
between Brüel & Kjær Sound and Vibration and Acoustic Technology, DTU. DPLA is responsible
for maintaining and disseminating the basic unit of sound pressure (the pascal) and acceleration
m/s 2 ) in Denmark. The associated research and international cooperation is mainly performed by DTU.
International cooperation is accomplished through meetings within EURAMET, the International Electrotechnical
Commission (IEC), the Consultative Committee for Acoustics, Ultrasound and Vibration
(CCAUV) under BIPM, and by participating in common projects organised by these bodies. By the end
of 2005 the responsibilities and activities of Acoustic Technology were transferred to the Danish Fundamental
Metrology (DFM), located in building 307 on the DTU Campus. This was formally accepted
by DANAK-Metrology in 2007. However, close research cooperation is maintained between DFM and
Acoustic Technology on acoustic metrology and measurement techniques.
CCAUV.A-K4 key comparison
Knud Rasmussen and Salvador Barrera-Figueroa
An international key comparison on free field reciprocity calibration of type ½” laboratory standard
microphones in the frequency range from 3 to 31 kHz has been agreed on in 2006. DPLA acts as
the pilot laboratory with nine participating countries (Brazil, Denmark, France, Germany, Great Britain,
Japan, Korea, Mexico and USA). Two microphones of type B&K 4180 have been circulating among
the participants since the end of February 2007. During 2007 Great Britain has withdrawn its participation
and in 2008 also USA withdrew. A draft report was discussed among the participants during a
meeting in October at the CCAUV meeting in Paris.
Free-field comparison calibration of WS1 and WS2 microphones
Knud Rasmussen and Salvador Barrera-Figueroa
Free field reciprocity calibration of microphones is very time consuming, and thus a secondary
calibration technique based on comparison with the primary reference microphones has been developed.
A small broad band loudspeaker is used as sound source. A ¼” microphone placed close to the centre of
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the source is used to monitor the sound signal. The transfer function between this monitor microphone
and the microphone under test is determined, with the latter placed on axis about 70 cm from the
source. To improve the accuracy the standard procedure is based on comparison with three reference
microphones, all calibrated by the reciprocity technique. This technique has been approved by DANAK
and EURAMET in 2008. The uncertainty of the comparison calibration is estimated to be about 0.15 dB
in the frequency range 1 to 30 kHz.
The uncertainty in intensity-based sound power measurements
Arturo Orozco Santillán and Finn Jacobsen
The sound power emitted by a source provides the most practical and general description of its
acoustic radiation. Sound intensity measurements make it possible to determine the sound power of a
source in situ, even in the presence of other sources. However, the fact that intensity-based sound power
measurements can take place under widely different conditions makes it extremely difficult to evaluate
the resulting measurement uncertainty. The general objective of this work was to analyse the effect of
the most common sources of error on sound power determination based on intensity measurements. The
sources of error include the orientation of the measurement surface, the spatial sampling, phase mismatch,
the finite difference error, scattering and diffraction, the projection error, the presence of an extraneous
source, and reverberation. A theoretical analysis was been carried out based on computer
simulations, which showed that phase mismatch is one of the most significant sources of error. Experimental
data obtained under extreme measurement conditions supplement the theoretical results.
Near field acoustic holography based on the equivalent source method and pressure-velocity
transducers
Yong-Bin Zhang
Supervisor: Finn Jacobsen
The advantage of using the normal component of the particle velocity rather than the sound pressure
in the hologram plane as the input of conventional spatial Fourier transform-based near field acoustic
holography has recently been demonstrated. This investigation has examined whether there might be
a similar advantage in using the particle velocity as the input of near field acoustic holography based on
the equivalent source method. Error sensitivity considerations indicate that this method is less sensitive
to measurement errors when it is based on particle velocity input data than when it is based on measurements
of sound pressure data, and this is confirmed by a simulation study and by experimental results.
A method that combines pressure- and particle velocity-based reconstructions in order to distinguish
between contributions to the sound field generated by sources on the two sides of the hologram
plane has also been examined.
Near field measurement with pressure-velocity transducer in the large anechoic room.
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Noise and Noise Control
Combined health effects of particles and noise
Jonas Brunskog and Torben Poulsen
Air pollution is an important risk factor for the cardiopulmonary disease. There is also evidence
of a causal effect between noise exposure and health effects due to hypertension and ischemic heart disease.
However, although particles and noise originate mostly from traffic, no study has examined the
combined effects. The methodology of this project is based on controlled studies with well defined exposure
and medical examination combined with determination of measurable physiological response
and reported annoyance level. Healthy volunteers will be exposed to low particle concentration and
background noise, high particle concentration and low background noise, low particle concentration and
high traffic noise, and high particle concentration and traffic noise. Effects of particles, noise, and combined
particle and noise exposure on heart rate variability, stress hormones and inflammatory markers
in blood will be examined. Cognitive and psychological experiments will also be carried out.
Active noise cancellation headsets
Torsten H. Leth Elmkjær
Supervisor: Finn Jacobsen
Torsten Elmkjær’s PhD study has been motivated by the extremely high sound pressure levels
encountered onboard airborne military platforms. The suggested solution is hearing protectors with active
noise control based on a hybrid multi-channel combination of feedforward and feedback control.
Active noise reduction systems based on feedforward control are limited by lack of coherence between
the reference signals and the error signals to be minimised, and active noise reduction systems based on
feedback control are limited by time delays; therefore a considerable part of the research has been focused
on examining these limitations. Another issue studied in this work is the development of adaptive
algorithms that can handle non-Gaussian “heavy tailed” impulsive signals.
The project was successfully defended on 7 November 2008.
Sound quality evaluation model of air cleaners
Cheol-Ho Jeong
A comparison of tested and predicted subjective scores. (a) Performance; (b) annoyance.
People in a quiet enclosed space expect calm sound at low operational levels for routine cleaning
of air. However, in the condition of high operational levels of the cleaner, a powerful yet not annoying
sound is desired. A model for evaluating the sound quality of air cleaners of the mechanical type has
been developed based on objective and subjective analyses. Signals from various air cleaners were recorded
and edited by increasing or decreasing the loudness in three loudness bands. Subjective tests
using the edited sounds were conducted both by the semantic differential method and by the method of
successive intervals. Two major characteristics, performance and annoyance, were factored out from the
principal component analysis. The subjective feeling of performance was related to both low and high
16

frequencies; whereas the annoyance was related to high frequencies. Annoyance and performance indices
of air cleaners were modelled from the subjective responses and the measured sound quality metrics:
loudness, sharpness, roughness, and fluctuation strength, using the multiple regression method.
This project has been carried out in collaboration with Korea Advanced Institute of Science and
Technology (KAIST) and Woongjin Coway co. Ltd.
Room Acoustics
Increase of voice level and speaker comfort in lecture rooms
Jonas Brunskog and Anders Christian Gade
Teachers often suffer from health problems related to their voice. These problems are related to
the acoustics of the lecture rooms. However, there is a lack of studies linking the room acoustic parameters
to the voice of the speaker. In this pilot study, the main goals were to investigate whether objectively
measurable parameters of the rooms can be related to an increase of the voice sound power produced
by speakers and to the speakers’ subjective judgments about the rooms. The sound power level
produced by six speakers was measured in six different rooms with different size, reverberation time
and other physical attributes. Objective room acoustic parameters were measured in the same rooms,
including reverberation time and room gain, and questionnaires were handed out to persons who had
experience talking in the rooms. Significant changes in the sound power produced by the speaker were
be found in different rooms. It was also observed that the changes mainly have to do with the size of the
room and to the gain produced by the room. To describe this quality a new room acoustic quantity
called ‘room gain’ is proposed.
Initial work in this project was carried out by two students, Gaspar Payá Bellester and Lilian Reig
Calbo, in special courses.
Statistical analysis and modelling of room acoustics
Eleftheria Georganti
Supervisor: Finn Jacobsen
The main concern of this work has been to examine the statistical properties of measured room
transfer functions across frequency and space inside typical rooms. The relationship between standard
deviation and the source-receiver distance has been also examined and it was shown that beyond room
critical distance, the spatial standard deviation approaches the result obtained for the deviation across
frequency. Simplified versions of transfer functions were obtained by complex smoothing using onethird
octave analysis. The idea was that complex smoothing might lead to simplifications that could be
useful for audio applications such as room compensation, auralisation and dereverberation techniques.
The sound power of a source in a reverberation room
Finn Jacobsen
Normalised spatial standard deviation
2
1.5
1
0.5
0
10 2
Small lightly damped room
Frequency [Hz]
10 3
Measured
Eq. (4)
Eq. (11)
17
Normalised space-frequency standard deviation
2
1.5
1
0.5
0
10 2
Small lightly damped room
Frequency [Hz]
10 3
Measured
Eq. (4)
Eq. (11)
Spatial standard deviation (left) and space-frequency standard deviation (right) of the sound power of a source.

It is well known that the sound power output of a source emitting a pure tone or a narrow band of
noise varies significantly with its position in a reverberation room, and even larger variations occur between
different rooms. The resulting substantial uncertainty in measurements of sound power as well as
in predictions based on knowledge of sound power is one of the fundamental limitations of ‘energy
acoustics’. The existing theory for this phenomenon is fairly complicated and has only been validated
rather indirectly. A simpler theory has been developed that gives predictions in excellent agreement
with the established theory, and the results have validated both experimentally in a number of rooms,
and from finite element calculations. The results confirm the phenomenon known as ‘weak Anderson
localisation’ – an increase of the reverberant part of the sound field at the source position. The numerical
calculations have been carried out by Alfonso Rodríguez Molares, University of Vigo, Spain.
The incident sound intensity in a reverberation room
Finn Jacobsen
The conventional method of measuring the insertion loss of a partition relies on an assumption of
the sound field in the source room being diffuse and the classical relation between the incident sound
power per unit area and a spatial average of the sound pressure in the source room; and it has always
been considered impossible to measure the sound power incident on a wall directly. Moreover, whereas
it has been established for many years that one should use the ‘Waterhouse correction’ for determining
the transmitted sound power in the receiving room, opinions vary as to whether one should use any correction
for determining the incident sound power in the source room. Theoretical analysis shows that
one should indeed use such a correction in the source room. In order to validate this theory a method of
measuring the incident sound power based on ‘statistically optimised near field acoustic holography’
and the sound pressure on the wall has been developed.
New weighting factors for estimating Sabine absorption coefficients
Cheol-Ho Jeong
In practice, a perfectly diffuse sound field cannot be achieved in a reverberation room, and therefore
measured absorption coefficients in such rooms deviate from the theoretical random incidence absorption
coefficients. The actual angular distributions of the incident acoustic energy should be taken
into account. This study tried to improve the agreement by weighting the random incidence absorption
coefficient according to the actual incident sound energy. The angular distribution of the incident energy
density is simulated using the beam tracing method for various room shapes and source positions.
The proposed angle-weighted absorption coefficients agree satisfactorily with the measured absorption
data determined using the reverberation room method. At high frequencies and for large samples, the
averaged weighting corresponds well with the measurement, whereas at low frequencies and for small
panels a relatively flat distribution agrees better.
Statistical absorption coefficient
1.2
1
0.8
0.6
0.4
0.2
10 2
0
e=2.4 m
Frequency (Hz)
A comparison of absorption coefficients. , Measurement; , absorption by the averaged weighting
function. , possible range of the weighted absorption coefficient; , the random incidence absorption.
18
10 3

Reconstruction of sound pressures in an enclosure using the phased beam tracing method
Cheol-Ho Jeong
Source identification in an enclosure is not an easy task because of wave interference and wall reflections,
in particular at mid and high frequencies. A phased beam tracing method has been applied to
the reconstruction of source pressures inside an enclosure at medium frequencies. First, surfaces of an
extended source were divided into small segments. From the each source segment one beam was projected
into the field, and all emitted beams were traced. Radiated beams from the source reach array
sensors after travelling various paths. Collecting all the pressure histories at the field points, sourceobserver
relations can be constructed in a matrix-vector form for each frequency. By multiplying the
measured field data with the pseudo-inverse of the calculated transfer function, one obtains the distribution
of source pressure. Omnidirectional spherical and cubic sources in a rectangular enclosure were
taken as examples in the simulation tests. The reconstruction error was investigated by Monte Carlo
simulation. When the source was reconstructed by the new method it was shown that the sound power
spectrum of the source in an enclosure could be estimated with precision.
p s,m
reflection coefficient, r 1
st
a 1
1 order reflection,
direct ray, a 0
rd
3 order reflection,
a3 reflection coefficient, r 3
19
nd
2 order reflection,
p n
a 2
Array of field points
A drawing for reconstructing source information using the phased beam tracing with an array of field points.
Sound intensity over an absorber in a reverberation room
Cheol-Ho Jeong
Normalized intensity
1.8
1.6
1.4
1.2
1
0.8
0.6
0.4
0.2
0
0 10 20 30 40 50 60 70 80 90
Incidence angle (deg.)
Distributions of the average incident intensity over the sample. , 250 Hz; , 500 Hz; , 1
kHz; , 2 kHz; , 4 kHz.
When measuring absorption coefficients of absorbers in reverberation rooms a perfect diffuse
field cannot be achieved because of the large absorbing area. Therefore measured absorption coefficients
in reverberation rooms deviate from random incidence absorption coefficients, which assume a
uniform intensity distribution over an absorber. If the actual distributions can be identified, it may be
possible to compensate for it. This study is concerned with sound intensity distributions over an absorber
sample by means of numerical simulations. A phased beam tracing has been used in order to estimate
sound pressures at two points near the sample, resulting in the incident sound intensity onto the
reflection coefficient, r 2

sample. The incident intensity distribution is affected by room properties such as total absorption area,
room geometry and volume, as well as the locations of a source-receiver pair. Assuming one surface is
fully covered by an absorber, the effects of observation position and absorption coefficient of the sample
on the incident intensity have been investigated. Near a corner of the sample the sound intensity at
grazing incidence is limited, and low absorption coefficients make the incident intensity uniform.
Speakers’ comfort and increase of their voice level in lecture rooms
David Pelegrin García
Supervisors: Jonas Brunskog and Torben Poulsen
Most studies of classroom acoustics have focused on the listeners’ point of view. In logopedics
and phoniatrics it has been stated that the teachers’ voice health problems is a major concern, and studies
have shown that every teacher on average has two days of absence every year because of voice
problems. To improve the teaching environment from an acoustic point of view it is necessary to get a
better understanding of the relationship between a speaker and the physical environment. It is assumed
that variations in the voice power level are a good indicator of the vocal effort experienced by the
teachers. An auditory virtual environment has been developed in order to carry the experiments regarding
the influence of the acoustic environment on the voice power. The setup consists of twenty-nine
loudspeakers placed in a sphere arrangement in a damped room. The system, shown in the figure below,
makes it possible to recreate the acoustics of any classroom in real time. First, the impulse response
with directional information is obtained. For each reflection the level, delay, spectral distribution and
incidence angle are specified.
Building Acoustics
Overview of the laboratory setup.
Flanking transmission in lightweight building structures
Jonas Brunskog
Lightweight building structures are of large interest in the building industry since the tendency is
toward a more industrialised building production, and since low weight means cheaper foundation.
However, the lightweight building structures are in many acoustic aspects different from traditional
building structures such as concrete and brick walls; they contain more layers and materials and are often
periodic in nature. This means that the usual building acoustic models based on statistical energy
analysis (SEA) do not work properly for such structures. Instead analytical wave or modal based methods
should be used to analyse the underlying physics. The purpose of this work is to improve the simplified
approaches such as SEA as specified in the standard EN 12354. An ad hoc working group has
been established, and this project should give input to the working group.
The work is carried out in cooperation with Lars-Göran Sjökvist, SP-Trätek, Sweden, and Hyuck
Chung, University of Auckland, New Zealand.
20

State of the art of acoustics in wooden buildings
Jonas Brunskog
A Swedish consortium was initiated by SP Trätek in 2007 in order to improve the general competence
in the acoustics of wooden buildings. The consortium consists of all national R&D performers,
leading companies in the building, building materials and wood sectors, and leading consultants. A report
is the first result. The report includes a literature survey, analysis and identification of industrial
needs for producing wooden buildings with good acoustic comfort, and a list of research needed to
reach that goal. For wooden constructions there are several features that differ from those in concrete
and other heavy constructions. The weight of a construction is an important parameter for the airborne
sound insulation at low frequencies, and therefore wood constructions may have poor sound insulation.
Impact sound from people walking is a common sound insulation problem for lightweight floors. Flanking
transmission is another problem. Noise from installations is often dominated by low frequencies,
and therefore special consideration is needed for wooden constructions. Existing models, e.g. the European
prediction standard EN 12354, are best suited for heavy constructions,. The costly process of using
test-buildings is common even though the results are not useful for slightly different building constructions.
Hence, there is a need for developing prediction tools.
This work was carried out together with a large group of Swedish scientist, acoustic consultants
and industry representatives.
Structural sound transmission and attenuation in lightweight structures
Lars-Göran Sjökvist
Supervisors: Jonas Brunskog and Finn Jacobsen
For some years it has been popular to build large houses with lightweight building techniques,
especially with wood. This project have examined features related to sound insulation in lightweight
buildings. The vibration pattern has been measured for a junction, and a simplified lightweight structure
has been analysed theoretically. Both studies focused on the attenuation rate, i.e. the rate at which the
vibrations decrease. The results show that the attenuation rate is high in the direction across beams; in
the direction along the beams there is not much attenuation. The attenuation rate depends on several
parameters, in particular the vibration frequency and structural stiffness. Very high attenuation is observed
for wavelengths longer than the distance between the beams. It had been implied that the vibration
level decreases fast away from the impact source in a wooden lightweight structure. The present
work showed that no such decrease could be proved with good significance in the direction along the
beams.
The plate seen from above. The position of the beams is marked with black lines and the excitation point is
marked with a white '+' sign. The vibration level for the 5000 Hz one-third octave band is displayed by grey scale,
where darker means more vibration; see the bar at the right side of the figure.
21

MSc Projects
The influence of material properties and receiver suspension shapes in hearing aids
Xinyi Chen
Supervisors: Finn Jacobsen and Mogens Ohlrich
Rubber is an important material with versatile engineering applications. When rubber is used for
vibration isolation the dynamic elastic modulus and damping of the material are of concern. This project
studied the rubber suspension in a hearing aid. The suspension holds the loudspeaker and should
prevent the transmission of vibrations to the shell of the hearing aid. The frequency range of concern
was broad (up to kilohertz). Neither the static modulus or damping nor the dynamic modulus or damping
of the material are usually known. The rubber suspension was examined both from a material point
of view and from a structural point of view. The dynamic elastic modulus (Young's modulus here) and
the damping were determined by different methods. Unfortunately, the results deviated from each other
because of an inherent feature of one of the methods and because of the uncertainty caused by the experiments.
However, a general conclusion is that both the Young’s modulus and the damping of the
rubber suspension are frequency dependent in the frequency range of concern.
The project was carried out in cooperation with Oticon with Martin Larsen and Søren Halkjær as
the local supervisors.
Properties of a bone conduction transducer
Gorm Dannesboe
Supervisor: Finn Jacobsen
This project examined a bone conduction microphone for use in headset applications for Bluetooth
and for radio. This was done both by theoretical modelling and through experiments. Simulations
were made by modelling the bone conduction microphone with electroacoustic analogous circuits. The
response of the head to vibrations generated by speech was determined experimentally. The bone conduction
microphone was improved by modifying several aspects of the design. Frequency response and
dampening of external noise were tested on prototypes of the new design.
The project was carried out in cooperation with Invisio.
Microphone calibration
Marta Díaz Ben
Supervisor: Finn Jacobsen
Working standard microphones are usually calibrated by comparison with a laboratory standard
microphone with a known sensitivity. A comparison calibration involves exposing the two microphones
to the same sound pressure either simultaneously or sequentially. The purpose of this project was to
examine the uncertainty of a calibration setup with the two microphones mounted face to face very
close to each other, and determine the influence of the distance between the microphone diaphragms,
the position of the source, possible reflections in the anechoic room etc. The theoretical part of the work
involved development of a boundary element model for calculating the sound pressure on the diaphragms
of the two microphones.
This project was carried out in cooperation with Danish Fundamental Metrology (DFM), with
Salvador Barrera Figueroa as the supervisor at DFM.
Sound radiation from a loudspeaker cabinet using the boundary element method
Efrén Fernandez Grande
Supervisor: Finn Jacobsen
Ideally, the walls of a loudspeaker cabinet are completely rigid. However, in reality, the cabinet
is excited by the vibration of the loudspeaker units and by the sound pressure inside the cabinet. The
radiation of sound caused by such vibrations can in some cases become clearly audible. The purpose of
this project was to provide a tool for evaluating the contribution from the cabinet to the overall sound
radiated by a loudspeaker. The specific case of a B&O Beolab 9 early prototype was investigated,
22

ecause an influence of sound from the cabinet had been reported. The radiation from the cabinet was
calculated using the boundary element method. The analysis examined both the frequency domain and
the time domain. A significant influence of the cabinet was been detected, which becomes especially
apparent during the sound decay process.
This project was carried out in cooperation with Bang & Olufsen.
Room acoustic analysis of theatres with actors performing in the audience area
Berti Gil Reyes
Supervisors: Jonas Brunskog and Cheol-Ho Jeong
The purpose of this project was to investigate the optimum position and orientation of performers
in order to improve the acoustics of theatres using simulations based on geometrical acoustics. Five
types of actual setting of stalls, covering possible configurations of drama theatres, were investigated.
The Gladsaxe theatre was modelled using the Odeon room acoustic model. Investigation of the speech
directivity pattern lead to the conclusion that an optimum speech aperture angle for a speaker is 50°
with respect to the frontal direction, irrespective of frequency and elevation angle. Effects of actors’
movement and orientation were also examined through a number of computer models.
Jens Holger Rindel, Odeon, served as external supervisor.
Two configurations, a U-shaped and an arena setting for drama performances.
Acoustic conditions in small musical venues
Hallur Johannessen
Supervisors: Jonas Brunskog and Torben Poulsen
The acoustic properties of three small venues for performances of rhythmic music were investigated
using the Odeon room acoustic software. Sound samples from rock and jazz were used for assessment
of the simulated modifications of acoustic conditions of the venues. It was found that the subjective
parameter ‘clearness’ was the most relevant for prediction of preference. The sound level of the
rock music sample affected the assessment of the acoustic conditions. The overall satisfaction of the
acoustic condition increased with the level of the music, No correlation between the subjective ‘clearness’
and the objective ‘clarity’, C80, was found. A change in the integration time from 80 ms to 30 ms
in the calculation of clarity, called C30, revealed good correlation with the subjective ‘clearness’.
Investigation of parameter drift in microtransducers: a study of temperature dependence
Julien Jourdan
Supervisor: Finn Agerkvist
The purpose of this project was to investigate the electrical and mechanical parameter drift with
state variables in microtransducers in order to determine those that are important to include in an optimised
model. The investigation focused on the drift of linear parameters with temperature, and tried to
indentify the parameters that can be regarded as constant and the parameters that vary with the temperature.
The latter were modelled. Several heat transfer mechanisms occur in the transducer: conduction,
convection, radiation and heat storage. These mechanisms contribute to the heating of the microtransducer,
and therefore the temperature behaviour of the microtransducer with frequency was studied for
23

different input powers and voltages. Based on the results from temperature measurements performed on
a microtransducer, a thermal model was developed.
Loudspeaker suspension
Georgios Kostopoulos
Supervisors: Finn Agerkvist, Mogens Ohlrich and Finn Jacobsen
Many vibrating systems and actuators, including loudspeakers, are weakly nonlinear. Most studies
of nonlinearities in loudspeakers indicate that one of the important sources is the loudspeaker suspension.
The inner suspension (known as the ‘spider’) provides the main part of the restoring force on
the loudspeaker’s diaphragm. This project used a finite element model for studying the linear as well as
the nonlinear behaviour of a spider. The results were compared with experimental data. Satisfactory
agreement were obtained both for the linear and for the nonlinear behaviour, indicating that the model
can be used for predicting the properties of a spider. An investigation on the influence of the spider’s
geometrical parameters on its nonlinear behaviour showed that small changes in the construction of the
spider or even in the way that the voice coil is attached can cause significant changes in the stiffness
curve of this component.
Spherical near field acoustic holography
Guillermo Moreno
Supervisor: Finn Jacobsen
Spherical near field acoustic holography is a recently developed technique that makes it possible
to reconstruct the sound field inside and just outside a spherical microphone array. The purpose of this
project was to extend the existing theory for measurements with an acoustically transparent microphone
array to measurements with an array with the microphones mounted on a rigid sphere. A rigid sphere is
somewhat more practical and gives better defined boundary conditions, but in measurements very near
a source there is the potential problem of multiple reflections between the sphere and the surface of the
source modifying the incident sound field and therefore the reconstruction. Numerical simulations and
measurements were carried out. The results obtained from both simulations and measurements show a
minimum influence of the reflections on the accuracy.
This project was carried out in cooperation with Brüel & Kjær.
Pressure [dB re 20uPa]
70
65
60
55
50
45
40
35
30
0 500 1000 1500 2000 2500 3000
Frequency [Hz]
24
Pressure [dB re 20uPa]
75
70
65
60
55
50
45
40
35
0.1 0.2 0.3 0.4 0.5 0.6 0.7
Y axis [m]
Sound pressure level generated by an ‘experimental monopole’ at the centre of the sphere, which is 20 cm (left)
and 40 cm (right) from the monopole. Blue dotted line, ‘true’ pressure (measured without the sphere); red dashed
line, reconstructed pressure.
Determination of sound insulation based on sound pressure and sound intensity
Lluís Enrique Navarro Muñoz
Supervisor: Finn Jacobsen
The sound insulation of partitions is usually determined from sound pressure measurements but
can instead be based on sound intensity measurements. The purpose of this project was to compare
these two methods and examine their advantages and disadvantages. Another point in the work was to

examine the still unsettled issue of whether or not the ‘Waterhouse correction’ should be used not only
in the receiving room (as is well established in conventional pressure-based measurements) but also in
the source room. To examine the matter it was attempted to change the volume of the source room in a
scale model of two reverberation rooms. Unfortunately the possible effect of the volume of the source
room could not be detected because of unavoidable flanking transmission in the scale model rooms.
Damping of ultrasound for food production applications
Adrien Roux
Supervisor: Finn Jacobsen
The Sonosteam process combines the effect of vapour and ultrasound and is used for disinfecting
products such as food. However, long exposure to very high levels of ultrasound may be unhealthy for
human beings. Thus the purpose of the project was to study methods of protecting people from the ultrasound.
In particular sound absorbers for very high frequencies and satisfying the standards of the
food industry were examined. Because of the high absorption of air in this frequency range it turned out
to be rather difficult to measure the absorbers under test. Consequently, the focus of the project turned
to the measurement method for determining the absorption in the ultrasound frequency range. A test
chamber consisting of a small pyramid made of glass and filled with dry nitrogen was used as a reverberation
chamber, and the Schroeder method was used to determine the reverberant decay from measured
impulse responses.
The project was carried out in cooperation with SonoSteam.
Reduction of low frequency noise from ventilation systems
Semir Samardzic
Supervisor: Finn Jacobsen
The performance of low-frequency silencers can be predicted using a plane wave model, the
transmission matrix method. Because of their low (static) pressure loss concentric tube resonators are
regarded as the most suitable reactive element for use in ventilation systems. Improvement of the silencers’
performance at low frequencies was obtained by studying various configurations of concentric
tube resonators, such as partitioning of the cavity, partial perforation, and using various perforation degrees
of the inner tube. A number of configurations of concentric tube resonators were compared
through simulations and measurements. A simple and convenient laboratory technique for measuring
the insertion loss of a silencer with a high and with a low source impedance was developed. Fairly good
agreement between the predicted and measured results was obtained.
The project was carried out in cooperation with Lindab.
Musicians’ room acoustics conditions
David Santos Domínguez
Supervisors: Jonas Brunskog, Cheol-Ho Jeong and Anders Christian Gade
A 3D rendering system with eight loudspeakers for real time auralization.
25

Usually musicians have problems hearing themselves or other musicians during performances on
a badly designed stage. Different design recommendations and parameters as objective measures have
been proposed since the early 1970s, and it seems to be generally accepted that musicians have one
main concern: getting the right balance between hearing themselves (support) and hearing others. Because
it is not clear how to obtain this balance research is needed. This project attempted to investigate
musicians’ room acoustics conditions. An eight channel 3D rendering system based on Ambisonics was
deloped and adapted for use in subjective room acoustics experiments with musicians where the musicians
could play and hear the rendered room in real time. A set of pilot experiments with soloist musicians
based on rooms modelled using Odeon was conducted in order to test the influence of the quantity
and quality of the early reflections received by the musicians. The results indicated that more experimentation
and test subjects are needed before clear conclusions can be established.
A vacuum motor as a mechanical noise source
Tarmo Saar
Supervisor: Mogens Ohlrich
Sketch of vacuum cleaner shell-structure with resiliently installed vacuum motor.
Left: resilient rubber support element for vacuum-motor; right: symmetric test rig for determining the translational
dynamic stiffness of a resilient support element (i.e. the black vibration isolators in the figure) under different
static deformation.
The purpose of this project was to investigate the dynamic properties and vibratory source
strength of a typical vacuum motor; to develop a method for measuring dynamic stiffness of different
types of resilient rubber mounts; and to examine a technique for estimating the power transmitted from
a resiliently mounted motor to the plastic shell of a vacuum cleaner. Vibration isolation of the vacuum
motor is necessary since even a minor rotor-imbalance produces audible vibration and noise because of
the high revolution speed. Data and properties required for predicting the transmitted power are the free
velocities of the source at its connecting points, the mobilities at these terminals, and the isolator stiffness,
as well as the mobilities of the receiving shell structure. The complex dynamic stiffness of the isolators
was calculated from mobility measurements performed on specially designed test rigs that allow
26

for static compression, as shown in the figure. Calibration and prediction of the transmitted power have
revealed that cross-coupling between terminals plays a role for the examined case at frequencies below
100 Hz; this is the subject of further investigations.
The project was carried out in cooperation with Peter Nøhr Larsen from Nilfisk-Advance A/S.
Evaluation and development of simplified models for airborne and impact sound insulation of
double leaf structures
Irma Albolario Vedsted
Supervisors: Jonas Brunskog and Mogens Ohlrich
This study examined some of the available simplified prediction models for sound insulations.
The main focus was on double leaf lightweight structures. The purpose was to find the weaknesses of
the prediction models and try to improve the models. The evaluations were done on the basis of the result
of laboratory measurements which were compared with the evaluated and improved prediction
models. The light weight double leaf structures considered in the study were mainly of gypsum board.
The predicted models included in this study were those with almost the same parameters as used in
laboratory measurements, and the investigation was limited to the results of the laboratory measurements
and included sound transmission through the direct path and not flanking transmissions.
The acoustics of bullrings used for musical concerts
Raúl Zambrano Izcara
Supervisors: Cheol-Ho Jeong and Jonas Brunskog
Many popular concerts are given in large arenas or huge stadiums that are not built for musical
purposes. Thus in Spain concerts often take place in bullrings. Some of these bullrings are covered by a
concave dome in order to provide sound insulation. This project was focused on La Cubierta de Leganés.
Acoustic simulations using the Odeon model were carried out both for the uncovered and covered
bullring. The objective was to improve the acoustic conditions of these huge constructions. The
simulations concentrated on the room acoustic parameters reverberation time and clarity. Some focusing
problems were encountered because of the circular shapes of the venue. Moreover, for the covered
bullring, because of low absorption in the ceiling at low frequencies, long reverberation times and low
clarity were observed.
Jens Holger Rindel, Odeon, served as external supervisor.
A model of a bullring, and the distribution of the sound pressure level determined by Odeon simulations.
Compensation of loudspeaker nonlinearities - DSP implementation
Karsten Øyen
Supervisor: Finn Agerkvist
In this project compensation of loudspeaker nonlinearities was investigated. A compensation system
based on a loudspeaker model (a computer simulation of a real loudspeaker), was first simulated in
MATLAB and implemented on DSP for real-time testing. This was a pure feedforward system. However,
loudspeaker parameters are drifting because of temperature and aging. This reduces the performance
of the compensation. To improve the system online tracking of the loudspeaker linear parameters
27

is needed. The compensation system was tested without such parameter identification, using the loudspeaker
diaphragm excursion as the output measure. The loudspeaker output and the output of the loudspeaker
model were monitored, and the loudspeaker model was manually adjusted to fit the real loudspeaker.
This was done by real-time tuning on DSP. The system seemed to work for some input frequencies
and not for others.
PUBLICATIONS
Journal Papers
S. Barrera-Figueroa, K. Rasmussen and F. Jacobsen: A note on determination of the diffuse-field sensitivity of
microphones using the reciprocity technique. Journal of the Acoustical Society of America 124, 2008, 1505-1512.
L. Friis and M. Ohlrich: Vibration modeling of structural fuzzy with continuous boundary. Journal of the Acoustical
Society of America 123, 2008, 718-728.
L. Friis and M. Ohlrich: Simple vibration modeling of structural fuzzy with continuous boundary by including
two-dimensional spatial memory. Journal of the Acoustical Society of America 124, 2008, 192-202.
F. Jacobsen, X. Chen and V. Jaud: A comparison of statistically optimized near field acoustic holography using
single layer pressure velocity measurements and using double layer pressure measurements. Journal of the Acoustical
Society of America 123, 2008, 1842-1845.
A.I. Tarrero, M.A. Martín, J. González, M. Machimbarrena and F. Jacobsen: Sound propagation in forests: A
comparison of experimental results and values predicted by the Nord 2000 model. Applied Acoustics 69, 2008,
662-671.
G.B. Jónsson and F. Jacobsen: A comparison of two engineering models for outdoor sound propagation: Harmonoise
and Nord2000. Acta Acustica united with Acustica 94, 2008, 282-289.
J. Escolano, F. Jacobsen and J.J. López: An efficient realization of frequency dependent boundary conditions in
an acoustic finite difference time-domain model. Journal of Sound and Vibration 316, 2008, 234-247.
C.-H. Jeong, J.G. Ih and J.H. Rindel: An approximate treatment of reflection coefficient in the phased Beam tracing
method for the simulation of enclosed sound fields at medium frequencies. Applied Acoustics 69, 2008, 601-
613.
C.-H. Jeong and J.-G. Ih: On the errors of the phased beam tracing method for the room acoustic analysis. Journal
of the Acoustical Society of Korea 27, 2008, 1-11 (in Korean).
Y. Luan and F. Jacobsen: A method of measuring the Green’s function in an enclosure. Journal of the Acoustical
Society of America 123, 2008, 4044-4046.
M.C. Vigeant, L.M. Wang and J.H. Rindel: Investigation of orchestra auralizations using the multi-channel multisource
auralization technique. Acta Acustica united with Acustica 94, 866-882, 2008.
N. Stefanakis, J. Sarris, G. Cambourakis and F. Jacobsen: Power-output regularization in global sound equalization.
Journal of the Acoustical Society of America 123, 2008, 33-36.
Theses
L. Friis: An investigation of internal feedback in hearing aids. Department of Electrical Engineering, Technical
University of Denmark. PhD thesis, 2008.
T. H. Leth Elmkjær: Foundations of active control. Active noise reduction helmets. Department of Electrical Engineering,
Technical University of Denmark. PhD thesis, 2008.
28

Chapters in Books
F. Jacobsen: Intensity techniques. Chapter 58 (pp. 1109-1127) in Handbook of Signal Processing in Acoustics,
eds. D. Havelock, S. Kuwano and M. Vorländer. Springer Verlag, New York, 2008.
F. Jacobsen and Hans-Elias de Bree: The Microflown particle velocity sensor. Chapter 68 (pp. 1283-1291) in
Handbook of Signal Processing in Acoustics, eds. D. Havelock, S. Kuwano and M. Vorländer. Springer Verlag,
New York, 2008.
Edited Books
Acoustic Signals and Systems (ed. F. Jacobsen), Part I (pp. 3-144) of Handbook of Signal Processing in Acoustics,
eds. D. Havelock, S. Kuwano and M. Vorländer. Springer Verlag, New York, 2008.
Conference Papers
F. Agerkvist and B. Rohde Pedersen: Time variance of the suspension nonlinearity. Proceedings of the 125th Audio
Engineering Society, 2008
F. Agerkvist, K. Thorborg and C. Tinggaard: A study of the creep effect in loudspeaker suspension. Proceedings
of the 125th Audio Engineering Society, 2008.
B. Rohde Pedersen and F. Agerkvist: Non-linear loudspeaker unit modeling. Proceedings of the 125th Audio Engineering
Society, 2008.
S. Barrera-Figueroa, F. Jacobsen and K. Rasmussen: On determination of microphone response and other parameters
by a hybrid experimental and numerical method. Proceedings of Acoustics ’08, Paris, France, 2008, pp.
1519-1524.
S. Barrera-Figueroa, F. Jacobsen and K. Rasmussen: On the relation between the radiation impedance and the
diffuse-field response of measurement microphones. Proceedings of Inter-Noise 2008, Shanghai, China, 2008.
L. Blanchard: High sound quality and concha headphones: where are the limitations? Proceedings of Acoustics
’08, Paris, France, 2008, pp. 717-722.
J. Brunskog, A.C. Gade, G. Payà Bellester and L. Reig Calbo: Speaker comfort and increase of voice level in lecture
rooms. Proceedings of Acoustics ‘08, Paris, France, 2008, pp. 3863-3868.
A.C. Gade: Trends in preference, programming and design of concert halls for symphonic music. Proceedings of
Acoustics ’08, Paris, France, 2008, pp. 345-350.
E. Georganti, J. Mourjopoulos and F. Jacobsen: Analysis of room transfer function and reverberant signal statistics.
Proceedings of Acoustics ’08, Paris, France, 2008, pp. 5637-5642.
F. Jacobsen: Measurement of total sound energy in an enclosure at low frequencies. Proceedings of Acoustics ’08,
Paris, France, 2008, pp. 3249-3254.
F. Jacobsen, J. Hald, E. Fernandez and G. Moreno: Spherical near field acoustic holography with microphones on
a rigid sphere. Proceedings of Acoustics ’08, Paris, France, 2008, pp. 2869-2873.
F. Jacobsen and X. Chen: The incident sound power in a diffuse sound field. Proceedings of Fifteenth International
Congress on Sound and Vibration, Daejeon, Korea, 2008, pp. 690-695.
F. Jacobsen, G. Moreno, E. Fernandez Grande and J. Hald: Near field acoustic holography with microphones on a
rigid sphere. Proceedings of Inter-Noise 2008, Shanghai, China, 2008.
29

C.-H. Jeong and J.-G. Ih: Directional distribution of acoustic energy density incident to a surface under reverberant
condition. Proceedings of Acoustics’ 08, Paris, France, 2008, pp. 3077-3082.
G.B. Jónsson and F. Jacobsen: A comparison of two engineering models for sound propagation: Harmonoise and
Nord2000. Proceedings of Joint Baltic-Nordic Acoustics Meeting BNAM 2008, Reykjavik, Iceland, 2008.
J.D. Alvarez B. and F. Jacobsen: An iterative method for determining the surface impedance of acoustic materials
in situ. Proceedings of Inter-Noise 2008, Shanghai, China, 2008.
Y. Luan: The structural acoustic properties of stiffened shells. Proceedings of Acoustics ’08, Paris, France, 2008,
pp. 393-398.
L.-G. Sjökvist, J. Brunskog and F. Jacobsen: Parameter survey of a rib stiffened wooden floor using sinus modes
model. Proceedings of Acoustics ’08, Paris, France, 2008, pp. 3011-3015.
Other Papers
J. Brunskog: Att tala i en undervisningslokal. Bygg & Teknik 99 (3), 2008.
V. Tarnow and F. Jacobsen: Instruments de mesure en acoustique. Techniques de l’Ingenieur, Dossier R6010,
2008.
Reports
J. Forssén, W. Kropp, J. Brunskog, S. Ljunggren, D. Bard, G. Sandberg, F. Ljunggren, A. Ågren, O. Hallström, H.
Dybro, K. Larsson, K. Tillberg, K. Jarnerö, L.-G. Sjökvist, B. Östman, K. Hagberg, Å. Bolmsvik, A. Olsson, C.-G.
Ekstrand and M. Johansson: Acoustics in wooden buildings. State of the art 2008. SP Report 2008: 16, SP Trätek,
2008.
Abstracts
J.-G. Ih and C.-H. Jeong: Acoustic source identification in an enclosed space using the inverse phased beam tracing
at medium frequencies. Journal of the Acoustical Society of America 123, 2008, p. 3309.
30

HEARING SYSTEMS, SPEECH AND COMMUNICATION
The group Hearing Systems, Speech and Communication is concerned with auditory signal processing
and perception, psychoacoustics, speech perception, audio-visual speech, audiology and objective
measures of the auditory function. The objectives of the research is to increase our understanding of
the functioning of the human auditory system and to provide insights that can be useful for technical
applications such as hearing aids, speech recognition systems, hearing diagnostics tools and cochlear
implants.
Auditory Signal Processing and Perception
The research in auditory signal processing and perception is mainly concerned with understanding
the relation between basic auditory functions and measures of speech perception. Computational
models of auditory perception are developed based on the results from listening experiments both in
normal-hearing and hearing-impaired listeners. These models are integrated into speech and hearing-aid
signal processing applications.
Hearing aid amplification at low input levels
Helen Connor
Supervisor: Torben Poulsen
Persons with a hearing loss often have a loss of audibility of soft sounds, for example, distant
voices and birds. This lack of audibility can be alleviated by means of a compressor hearing aid. This
project considers some audiological factors and hearing aid parameters that influence the preference for
compression threshold. One experiment investigated the subjectively preferred compression threshold
with stimuli from real life everyday environments. The stimuli were compressed offline using six different
compression settings, and the compressed signals were presented to twelve subjects via the direct
audio input of binaurally fitted linear hearing aids. The results showed that the preference for low compression
thresholds increased with increased compression release time. A second experiment was a field
trial designed to validate the results from the first experiment when hearing aids are worn by the subjects
in their own daily listening environments. Twenty hearing impaired subjects compared two compression
threshold settings in two two-week trial periods. At the end of each trial, the subjects were interviewed
about their programme preference.
This industrial PhD project is carried out in cooperation with the hearing aid manufacturer Widex
A/S with Carl Ludvigsen as the industrial supervisor.
Estimating the basilar-membrane input/output-function in normal-hearing and hearing-impaired
listeners
Morten L. Jepsen
Supervisor: Torsten Dau
To characterise the function of cochlear processing it is desirable to estimate the basilar membrane
input/output function behaviourally in humans. Such estimates of compression are useful in adjusting
parameters of models of the basilar membrane to simulate individual hearing loss. In recent
studies, forward masking has been used to estimate the input/output function, showing a linear behaviour
at low input levels and a compressive behaviour above a certain input level (i.e., the knee-point). In
this study, an existing method is extended to estimate the knee-point and the amount of compression.
Data have been collected from seven normal-hearing and five hearing-impaired listeners with a mild to
moderate sensorineural hearing loss. Both groups showed large inter-subject but low intra-subject variability.
When the knee-point was estimated for the hearing-impaired listeners it was similar or shifted
up to about 30 dB towards higher input levels, and the amount of compression was similar or increased
compared to that of normal-hearing listeners.
31

The basic structure of a dual resonance nonlinear filter. The input is split into two paths: a linear path, and a
nonlinear path. The nonlinear path contains a single nonlinear element, a ‘broken-stick’ function that realises an
instantaneous compression above a threshold amplitude. The output of the system is the sum of the two paths.
Modelling auditory perception of individual hearing-impaired listeners
Morten L. Jepsen
Supervisor: Torsten Dau
In this work the perceptual consequences of hearing impairment in individual listeners have been
investigated within the framework of the computational auditory signal processing and perception model
of Jepsen et al. (2008). Several parameters of the model were modified according to data from psychoacoustic
measurements. Three groups of listeners were considered: normal-hearing listeners, listeners
with a mild-to-moderate sensorineural hearing loss, and listeners with a severe sensorineural hearing
loss. The simulations showed that reduced cochlear compression due to outer hair-cell loss quantitatively
accounts for broadened auditory filters. A combination of reduced compression and reduced inner
hair-cell function accounts for decreased sensitivity and slower recovery from forward masking. The
model may be useful for the evaluation of hearing-aid algorithms, where a reliable simulation of hearing
impairment may reduce the need for time-consuming listening tests during development.
Spectro-temporal analysis of complex sounds in the human auditory system
Tobias Piechowiak
Supervisor: Torsten Dau
This PhD project investigates the auditory processing of modulated sounds. This is relevant since
many natural sounds (including speech) contain amplitude and frequency modulations. One particular
feature of such sounds is that they exhibit temporally coherent amplitude modulations across a wide
range of audio frequencies. It has been proposed that the auditory system evaluates this acrossfrequency
coherence to facilitate signal detection (or speech recognition) in a noisy acoustic background.
In the modelling part of the project, an auditory mechanism for signal enhancement was suggested
that processes sound components across different spectral content (within each ear), in a similar
way as was earlier established for across-ear processing for improved localisation. In order to test the
proposed ‘equalisation-cancellation’ mechanism, various listening experiments were performed and the
results compared with predictions using a computational model of auditory processing.
Neural coding and perception of pitch in the normal and impaired human auditory system
Sébastien Santurette
Supervisors: Torsten Dau and Jörg Buchholz
The purpose of this PhD project is to investigate how the human auditory system extracts the
pitch of sounds. Pitch is an essential attribute of sound that contributes to speech intelligibility, music
32

perception, and sound source segregation. It is thus important to understand the mechanisms that underlie
pitch perception. In particular it would be crucial to determine whether the auditory system uses
mechanisms based on temporal cues, spectral cues, or both, for pitch extraction. In order to distinguish
better between possible theories, the project focuses on relating pitch perception outcomes to measures
of basic auditory functions in normal-hearing and hearing-impaired listeners. Such an approach could
reveal how and at which level of the auditory system pitch is represented, and tell us which modelling
approach to favour, with possible implications for hearing-aid and cochlear-implant signal processing.
Perception of pitch.
Relating binaural pitch perception to measures of basic auditory functions
Sébastien Santurette
Supervisor: Torsten Dau
Binaural pitch is a tonal sensation produced by introducing an interaural phase shift in binaurallypresented
white noise. As no spectral cues are present in the stimulus, binaural pitch perception is assumed
to rely on accurate temporal fine structure (TFS) coding and intact binaural integration mechanisms.
This study investigated to what extent basic auditory measures of binaural processing, TFS processing,
as well as cognitive abilities, are correlated with the ability of hearing-impaired listeners to perceive
binaural pitch. It was found that hearing-impaired listeners who could not perceive binaural pitch
neither had a general difficulty extracting tonal objects from noise nor showed reduced cognitive abilities.
Instead, they showed impaired binaural processing of TFS, reflected by reduced binaural masking
and binaural intelligibility level differences, as well as a loss of phase-locking occurring at much lower
frequencies than for other hearing-impaired subjects. These results suggest that the absence of binaural
pitch perception is a good indicator of a deficit in low-level binaural processing.
Measurement of binaural pitch.
33

Detection and identification of binaural and monaural pitches in dyslexic patients
Sébastien Santurette
Supervisor: Torsten Dau
Binaural pitch stimuli have been used in several recent studies to test for the presence of binaural
auditory impairment in reading-disabled subjects. However, the outcome of these studies is contradictory;
whereas some studies found that a majority of dyslexic subjects were unable to hear binaural pitch,
another study obtained a clear response of subjects in the dyslexic group to Huggins’ pitch (HP). This
work aimed at clarifying whether impaired binaural pitch perception is found in dyslexia. Results from
a pitch contour identification test, performed in 31 dyslexic listeners and 31 matched controls, clearly
showed that dyslexics perceived HP equally well as controls. However, nine of the dyslexic subjects
were found to have difficulties in identifying pitch contours independently of the stimulus used. The
ability of subjects to identify pitch contours correctly was found to be significantly correlated to measures
of frequency discrimination.
This work was carried out at Division of Experimental Otolaryngology, Department of Neurosciences,
K.U. Leuven, Belgium, with contributions from Hanne Poelmans, Heleen Luts, and Jan Wouters.
Frequency selectivity, temporal fine-structure processing and speech reception in impaired hearing
Olaf Strelcyk
Supervisor: Torsten Dau
Hearing impaired people encounter great difficulties with speech communication, particularly in
the presence of background noise. The benefit of hearing aids varies strongly among the individual listeners,
and whereas some show good performance in speech communication, others continue to experience
difficulties. The hypothesis of this project is that the variation in performance among listeners
can, at least partly, be traced back to changes in the perception of sounds well above hearing threshold.
Thus it has been hypothesised that reduced frequency selectivity as well as deficits in processing of
temporal fine structure are related to degraded speech reception. Perceptual listening experiments have
been performed on normal-hearing listeners, hearing-impaired listeners and listeners with an obscure
auditory dysfunction. The results gave insights about impairments in the peripheral auditory system,
which were related to speech reception deficits. They provide constraints for future models of impaired
auditory signal processing.
This PhD project is co-supervised by Graham Naylor, Oticon.
Auditory processing, based on the spatio-temporal cochlear response, may be degraded in hearing-impaired listeners,
due to reduced frequency selectivity, degraded phase locking or reductions in converging input to coincidence
detectors.
Objective and behavioural estimates of cochlear response times in normal-hearing and hearingimpaired
human listeners
Olaf Strelcyk
Supervisor: Torsten Dau
Auditory brainstem responses have been obtained in normal-hearing and hearing-impaired listeners.
The latencies extracted from these responses serve as objective estimates of cochlear response
times. In addition, two behavioural measurements have been carried out. In the first experiment, coch-
34

lear response times were estimated, using the lateralisation of pulsed tones interaurally mismatched in
frequency. In the second experiment, auditory-filter bandwidths were estimated. The correspondence
between objective and behavioural estimates of cochlear response times was examined. An inverse relationship
between the auditory brainstem response latencies and the filter bandwidths could be demonstrated.
The results might be useful for a better understanding of how hearing impairment affects the
cochlear response pattern in human listeners.
LTFAT – The linear time-frequency analysis toolbox
Peter L. Søndergaard
This is ongoing work. The linear time-frequency analysis toolbox was started in 2004 and aims to
be a next-generation MATLAB signal processing toolbox focused on time-frequency analysis using
Gabor analysis. A major strength of the toolbox is to enable easy linear or nonlinear modification of the
time-frequency representation of a signal. The toolbox is Free Software and can be obtained from
http://ltfat.sourceforge.net. It is joint work with the Acoustics Research Institute, Austrian Academy of
Sciences, Vienna, and Laboratoire d’Analyse, Topologie et Probabilites, Université de Provence, Marseille,
but most of the work is carried out at CAHR.
Investigation of the independent manipulation of envelope and temporal fine structure in psychoacoustics
Peter L. Søndergaard
An important topic in current psychoacoustic research is the importance of envelope and temporal
fine structure (TFS) of complex signals such as speech. The envelope refers to the slow changes of a
signal at a given frequency, and the TFS are the fast changes (the information in the carrier wave). In
the cochlea the envelope and the TFS are naturally extracted from the input by the action of the basilar
membrane and the inner hair cells. In this project, it was investigated mathematically which information
is carried by the envelope and by the TFS. It was shown that certain methods used in the recent literature
to investigate the contribution of TFS versus envelope for speech and music perception are problematic
since the information carried by the TFS versus the envelope is not independent.
Spectrogram (left) and instantaneous frequency (right) of the Danish word ‘Stok’. Either representation of the
signal contain the same information.
Interactions of interaural time differences and amplitude modulation detection
Eric R. Thompson
Supervisor: Torsten Dau
Psychoacoustic measurements have been carried out to determine the effect of a perceived spatial
separation on amplitude modulation detection thresholds. Two temporally-interleaved transposed stimuli
were used as carriers for a narrowband-noise modulation masker and a 16-Hz sinusoidal modulation
probe. With these stimuli, the interaural time difference (ITD) of the masker and probe carriers could be
35

adjusted independently. In the first experiment, the listeners adjusted the interaural level difference of a
pointer stimulus to be aligned with the perceived lateral position of either the masker or the probe stimulus,
as a function of the masker and probe ITDs. The results showed that the listeners could lateralize
the two stimuli separately and robustly. The second experiment measured masked AM detection thresholds
as a function of the masker modulation frequency and masker ITD using a diotic 16-Hz AM
probe. These results showed modulation frequency tuning without a spatial release from modulation
masking even though the masker and probe were perceived to have a spatial separation. This suggests
that amplitude modulation cues and lateralisation cues are processed independently and in parallel in
the auditory system.
Speech Perception
The research in speech perception aims at describing the mechanisms underlying speech intelligibility,
i.e., how human listeners decode and integrate the information carried by the speech signal.
Models of speech intelligibility and models quantifying speech perception under challenging listening
conditions are under development. Like other areas in the Centre for Applied Hearing Research, the
research is interdisciplinary and relies on psychoacoustics, auditory signal processing and phonetics.
Processing of spatial sounds in the impaired auditory system
Iris Arweiler
Supervisors: Jörg Buchholz and Torsten Dau
A common complaint from people with hearing impairment is difficulty with speech communication,
particularly when background noise is present. The problem often persists even if hearing aids are
used. The overall goal of this project is to analyse how hearing-aid signal processing affects speech intelligibility
for listeners with different degrees and types of hearing impairments in complex listening
conditions. The first part focuses on the influence of early reflections on speech intelligibility. Early
reflections can improve speech intelligibility in noisy conditions. Normal-hearing and hearing-impaired
listeners will perform a monaural and binaural speech intelligibility task in a virtual auditory environment
where direct sound and early reflections can be varied independently. This will lead to a better
understanding of the underlying mechanisms involved in early reflection processing. In the second part
of the study hearing aids will be fitted to the hearing-impaired listeners and their influence on early reflection
processing will be investigated in the same conditions as before.
Direct sound
36
Early reflections
Ambient noise
Generating spatial sounds.

Building blocks of spontaneously spoken Danish—acoustically and perceptually
Thomas U. Christiansen, Steven Greenberg and Torsten Dau
The purpose of this project is to identify the basic articulatory, acoustic and auditory elements of
spontaneously spoken Danish. This is achieved by developing a quantitative description of the transformation
from linguistic representation via the acoustic signal to the auditory representation. The
analysis is based on new results from empirical investigations from phonetics and auditory models. A
great challenge lies in the explication of how large acoustic variation as a consequence of different talkers
in different listening environments can lead to a uniform linguistic percept. The results will impact a
variety of research areas such as linguistics, hearing science, cognitive psychology, computational neuroscience,
speech recognition and speech synthesis. Based on two observations new basic elements in
the production and perception of spoken language are proposed: 1) Slow energy variations in the
speech signal below 30 Hz are represented in the auditory cortex. It is conjectured that this representation
carries speech information under every day listening conditions. Realistic models of this modulation
representation have been proposed in the literature. 2) Certain articulatory and acoustic properties
vary systematically with stress and position in the syllable. These properties are therefore better suited
as candidates for building blocks of a quantitative description of spoken language than the traditional
units such as phonemes and diphones.
Phonetic flow of linguistic processing
Thomas U. Christiansen and Steven Greenberg
In spite of a substantial amount of research effort, the way in which speech is decoded is poorly
understood. In particular little is known about the process of decoding basic speech sounds such as consonants.
This is true both with respect to which underlying acoustic cues are used for decoding and the
stages the decoding process consists of. The aim of this project is to identify which acoustic cues are
most important for consonant identification, and to outline the ‘phonetic flow of linguistic processing’,
i.e., to describe a hierarchical process by which the speech signal is decoded. One of the ways to
achieve the project goals is to collect data from a consonant identification task. The results of such experiments
are represented in so-called confusion matrices. By analysing the asymmetries of these confusion
matrices it is possible to preclude certain decoding schemes, while providing evidence for other
decoding schemes. The results of the analyses are potentially important for signal processing in digital
hearing aids and in automatic speech recognition systems.
Objective evaluation of a loudspeaker-based room auralisation system
Sylvain Favrot
Supervisors: Jörg Buchholz and Torsten Dau
The setup with loudspeakers in a damped room.
37

A loudspeaker-based room auralisation system has been developed for studying basic human perception
in realistic environments. The system reproduces a controlled acoustic scenario designed with
the Odeon room acoustics software (state-of-the-art software developed at Acoustic Technology, DTU)
with the help of a loudspeaker array of twenty-nine loudspeakers placed on the surface of a sphere. An
objective evaluation has been carried out demonstrating the applicability of the system. Room acoustic
parameters (reverberation time, clarity, speech transmission index and inter-aural cross correlation coefficients)
of room impulse responses were compared at the input and the output of the system. The results
show that the involved signal processing preserves the temporal, spectral and spatial properties of
the room impulse response.
Predicting degraded consonant recognition in hearing-impaired listeners
Morten L. Jepsen
Supervisor: Torsten Dau
Reduced ability to recognise consonants plays a major role in general speech perception in noise.
Today little is understood about how our auditory system processes and decodes the speech information.
Which cues are the most salient and why is the normal-functioning system so robust in extracting
the right cues? These aspects are explored by using a model of the normal and impaired auditory system
to predict error patterns in consonant recognition. The idea is to use a simple binary consonant recognition
task with synthesised stimuli. This makes it possible us to use a simple speech recogniser such that
the predicted machine error can be associated with the degraded auditory processing. In the analysis of
the auditory representation of the consonants it may be possible to explore and define the salient cues in
consonant discrimination and thereby obtain a better understanding of general auditory processing of
speech.
This work was carried out at Boston University with Oded Ghitza as the local supervisor.
Perceptual compensation for effects of reverberation on speech identification
Jens Bo Nielsen
Supervisor: Torsten Dau
Reverberation can be viewed as a lowpass modulation filter that generally reduces speech intelligibility.
Recent research has shown that the auditory system appears to compensate for this effect. This
extrinsic compensation mechanism was demonstrated by asking listeners to identify a test-word embedded
in a carrier sentence. Reverberation was added to the test-word but not to the carrier, and the ability
to identify the test-word decreased because the amplitude modulations were smeared. When a similar
amount of reverberation was added to the carrier sentence, the listeners’ ability to identify the test-word
was restored. However, this study has not confirmed that such a compensation mechanism exists. In this
work the reverberant test-word was embedded in several additional carriers without reverberation. All
the carriers enhanced the ability to identify the test-word. This suggests that a listener’s perception of
the test-word is affected by other carrier characteristics than reverberation. It is proposed that an interfering
effect of the non-reverberant carrier in combination with the reverberant test-word can account
for the data that previously have been taken as evidence of extrinsic compensation for reverberation.
CLUE - a Danish hearing-in-noise test
Jens Bo Nielsen
Supervisor: Torsten Dau
A Danish speech intelligibility test called Conversational Language Understanding Evaluation
(CLUE) has been developed for assessing the speech reception threshold (SRT). The test consists of
180 sentences distributed in eighteen phonetically balanced lists. The sentences are based on an open
word-set and represent everyday language. The sentences were equalised with respect to intelligibility
to ensure uniform SRT assessments with all lists. In contrast to several previously developed tests such
as the hearing in noise test where the equalisation is based on objective measures of word intelligibility,
the present test uses an equalisation method based on subjective assessments of the sentences. The new
equalisation method is shown to create lists with less variance between the SRTs than the traditional
method. The number of sentence levels included in the SRT calculation has also been evaluated and is
38

different from previous tests. The test was verified with fourteen normal-hearing listeners; the overall
SRT lies at a signal-to-noise ratio of -3.15 dB with a standard deviation of 1.0 dB. The list-SRTs deviate
less than 0.5 dB from the overall mean.
list SRT re: overall SRT [dB]
2
1.5
1
0.5
0
−0.5
−1
−1.5
−2
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18
list number
The verification of the CLUE test with 14 normal-hearing listeners showed that the 18 sentence lists lead to
very similar SRT determinations. The mean SRT for each list lies within +/- 0.5 dB of the overall average. The
bars indicate one standard deviation.
A new speech manipulation framework
Peter L. Søndergaard
In the psycho-acoustic community there is a desire to gradually use more and more complex test
signals eventually leading to manipulated speech. Speech manipulation systems date back to Flanagan’s
‘Phase vocoder’ from 1966. Because of redundancy of information in speech, vocoders have never been
able to produce speech signals sounding perfectly natural. The goal of this project is to make a limited
system for speech manipulation that can change an input speech signal in a desired way in order to provide
a new suite of test signals for the work being done at CAHR and in other research groups. The
work is carried out in close cooperation with Thomas Ulrich Christensen to get input from the linguistics
community.
Monaural and binaural consonant identification in reverberation
Eric R. Thompson
Supervisor: Torsten Dau
Consonant identifications have been obtained monaurally and binaurally using vowel-consonantvowel
stimuli convolved with binaural impulse responses recorded from a concert hall. The impulse
responses had reverberation times of about 2.5 s and showed large interaural differences in modulation
transfer functions. The percentage of correct identifications were significantly higher when listening
binaurally than in either monaural condition, and the percent correct identifications were significantly
lower for one impulse response than for the other two, despite having similar speech transmission indices.
Not every stimulus that was correctly identified monaurally was also correctly identified binaurally,
indicating binaural interference. About 12% of the stimuli that were correctly identified binaurally
were not correctly identified with either monaural condition, which shows a binaural advantage beyond
simple ‘better-ear’ listening. The most frequent errors were voicing confusions, contrary to findings
from previous studies, which have found voicing to be relatively robust against reverberation. The data
will help in the development of models of binaural speech intelligibility in reverberant environments.
Binaural intensity modulation detection in reverberant environments
Eric R. Thompson
Supervisor: Torsten Dau
The speech transmission index (STI) uses the magnitude of the modulation transfer function
(MTF) in a single channel to predict speech intelligibility in rooms. This method often underestimates
speech intelligibility when binaural listening is possible. The two ears often have large differences in
the MTF, and interaural modulation phase differences can create perceivable interaural intensity fluctu-
39

ations that can be used to improve detection of intensity modulations. Modulation detection measurements
have been made monaurally and binaurally with three dichotic impulse responses and anechoically
at modulation frequencies between 6 and 24 Hz. The first impulse response consisted of the direct
sound and a single ideal reflection arriving at a different time in each ear, and the other impulse responses
were from a simulation of a classroom and a recording from a concert hall. Monaurally, the thresholds
measured with the impulse responses could be predicted within 1.5 dB from the threshold of the
anechoic condition and the magnitude of the MTF. However, with each of the impulse responses there
was a significant improvement in the modulation detection thresholds for binaural listening compared
with monaural listening at a modulation frequency where there was a large interaural modulation phase
difference. The results suggest a way for the STI to incorporate binaural cues and improve on predictions
of binaural speech intelligibility in reverberant environments.
Audiology
Audiology is concerned with the study of impaired hearing and the possibilities of compensation
by means, e.g., of a hearing aid. A major topic in this area is to develop new or better diagnostic tools in
order to distinguish between different kinds of hearing losses. Traditional descriptions in terms of the
audiogram, i.e., pure tone thresholds, have been shown to be far from satisfying. Better diagnostic tools
are essential for the fitting of advanced signal processing hearing aids, thus allowing the hearingimpaired
user to obtain the full benefit of the device. Identifying and quantifying factors that describe
hearing loss is therefore one of the main goals of the research in audiology.
Open-plan office environment; a laboratory experiment on human perception, comfort and office
work performance
Torben Poulsen
At International Centre for Indoor Environment and Energy, DTU, a laboratory investigation has
been conducted on the effect of office noise and temperature on human perception, comfort and office
work performance. The project is organised by Geo Clausen, Department of Mechanical Engineering,
DTU, and has been carried out in cooperation with Ivana Balazova, Indoor Air, and Jens Holger Rindel,
Odeon.
Hearing in the communication society (HEARCOM)
Torben Poulsen and Torsten Dau
The general focus of the HearCom project has been on the identification and characterisation of
limitations for auditory communication and on modelling and evaluation of ambient conditions that
limit auditory communication in everyday situations. In 2008 DTU has produced binaural room impulse
responses to be used by one of the other partners. Furthermore a review of various reports (deliverables)
from other partners has taken place.
Musicians’ health, sound and hearing
Torben Poulsen
A project on musicians’ sound exposure and their risk of hearing impairment takes place at Department
of Occupational and Environmental Medicine, Odense University Hospital. Torben Poulsen
serves as an external supervisor on a PhD study by Jesper Schmidt. The work is concerned with both
classical and non-classical musicians. The project is related to Centre of Musicians’ Health in Odense,
and is managed by Jesper Bælum and based on a grant from the Working Environment Research Fund.
Transparent hearing protector for musicians
Torben Poulsen
The idea of this work, supported by a grant from the Working Environment Research Fund, is to
develop a device that can prevent musicians from being exposed to excessive loud sounds but at the
same time allow them to hear sounds at lower levels without attenuation. The device is based on the
40

signal processing capabilities of a modern hearing aid. The work is carried out in cooperation with Ture
Andersen, University of Southern Denmark.
Music and hearing loss
Torben Poulsen and Anders Christian Gade
Three short guidelines have been produced for the music and entertainment sector. The guidelines
aim at hearing conservation and limitation of sound exposure for symphony orchestras, pop-jazz
groups and music clubs/discotheques. The guidelines may be downloaded from http://www.barservice.dk/.
The work is organised by Per Møberg Nielsen, Akustik Aps.
Guidelines (in Danish) for symphony orchestras, pop-jazz groups and music clubs/discotheques.
Audio-Visual Speech and Auditory Neuroscience
The research in the field of audio-visual speech is concerned with the integration process of visual
cues and auditory cues performed in the brain. It includes synchronised audio-visual speech generation
and investigations on neural synchronisation in the brain. The central goal is to design a computerbased
‘language laboratory’ for lip-reading that can be used by people with a sudden profound hearing
loss to quickly re-gain communicational skills. Visual speech information is also useful for humanmachine
communication in noisy environments. One important aim of computational auditory neuroscience
is to model human auditory perception with the help of biological neural networks.
Audio-visual speech analysis
Hans-Heinrich Bothe
The work is part of a framework to design and implement a multilingual language laboratory for
speech- or lip-reading. The goal of the project was to model the dynamics of mouth movements during
the speech articulation process for an open vocabulary. Video films with one speaker and emotion-free
articulation were analysed and the lips movements were modelled in the temporal context of English
speech production. The dynamic model is related to characteristic single images of the video clip that
represent the sounds and on transitions between neighbouring images. A general dynamic model employing
parameters that state the importance of specific changes of the lip contours for the correct articulation
of the sounds was used; it is, for example, necessary to close the lips completely in order to
articulate a /p/, /b/, or /m/ correctly, whereas correct articulation of a vowel as /a/ or /e/ is ambiguous
and depends on the context. The project is ongoing. The momentary results can predict the courses of
the feature sets with specific artifacts when only the text input is given. In the future, an improved
model will be developed, implemented and evaluated in audio-visual speech intelligibility tests.
41

Objective Measures of the Auditory Function
The field of objective measures of the auditory function comprises studies of auditory evoked potentials
and otoacoustic emissions. Auditory evoked potentials represent electrical fields recorded from
the surface of the head in response to sound. Otoacoustic emissions are low-level sounds generated by
the inner ear in response to sound or spontaneously and can be recorded in the ear canal. Otoacoustic
emissions and evoked potentials can provide insights about the physiological state of the ear and the
brain. Research projects range from basic scientific interpretations of the objective measures themselves
to applied technical work on improving signal quality and robustness under noisy conditions.
Cochlear delay obtained with auditory brainstem and steady-state responses
Gilles Pigasse
Supervisors: James Harte and Torsten Dau
A great deal of the processing of incoming sounds to the auditory system occurs within the inner
ear, the cochlea. The cochlea has different mechanical properties along its length. Its stiffness is at
maximum at the base and decreases towards the apex, resulting in locally resonant behaviour. High frequencies
have maximal response at the base and low frequencies at the apex. The wave travelling along
the basilar membrane has a longer travel time for low-frequency stimuli than for high-frequency stimuli.
In order to obtain an objective estimate of the cochlear delay in humans, the present project investigated
several non-invasive methods. These methods included otoacoustic emissions, auditory brainstem
responses and auditory steady-state responses. A comparison between the three methods was made. Below
2 kHz the otoacoustic emission delay was twice the cochlear delay, as if the travelling wave went
back and forth in the cochlea. However, this relation did not hold for higher frequencies, calling into
question the physical relation between the delay estimates obtained from otoacoustic emission and those
obtained from the evoked responses.
This PhD project was defended successfully on 3 December 2008.
Optimised stimuli for auditory evoked potentials and otoacoustic emissions
James Harte and Torsten Dau
Spectrogram for (a) click- and (b) chirp-evoked otoacoustic emissions. The stimulus can be seen in the left
part of each spectrogram, recognisable by its high level. The otoacoustic emissions occur later and have a level of
about 40 dB below the stimulus level. The chirp was designed to compensate for cochlear travel times across frequency.
As a result, when analysed in time and frequency, the pattern of the chirp evoked emission was close to a
straight line between 1.6 and 4.5 kHz, thus indicating stronger synchronisation.
Auditory evoked potentials and otoacoustic emissions (OAEs) are routinely used as tools for
audiometric investigation. This project attempted to improve recording techniques by optimising the
stimuli. In terms of evoked potentials, rising broadband frequency chirps have traditionally been inves-
42

tigated to compensate for the dispersion along the cochlea. Here, narrowband chirp stimuli were considered
to obtain frequency specific responses. A classic transient-evoked otoacoustic emission is relatively
widely dispersed over time, has a chirp-like structure starting at high frequencies, and rings down
to low-frequencies. Chirp stimuli were investigated with the opposite direction frequency glide to produce
emissions recorded in the ear canal that should be more ‘click-like’ than the classical recordings.
The results were interpreted in terms of a model for OAE generation.
Characterising temporal nonlinear processes in the human cochlea using otoacoustic emissions
Sarah Verhulst
Supervisors: James M. Harte and Torsten Dau
Otoacoustic emissions are measured to investigate cochlear nonlinearities in the human inner ear.
Temporal suppression, which is linked to nonlinear active processes in the inner ear, was examined by
use of click-evoked otoacoustic emissions. The temporal suppression-effect is created when a suppressor-click
is presented close in time to a test-click. However, the methods used to obtain temporal suppression
are highly critical if one wishes to link this phenomenon with changes in cochlear compression.
This project has demonstrated the existence of a phase and magnitude component in the suppression
measure, a finding that has consequences for the interpretation of this suppression. The data also
show that the mechanisms underlying this suppression can work either in a compressive or expansive
state. These findings point to a temporal adaptation mechanism in the human outer hair cells. Results
for four subjects show that compression as well as augmentation are subject dependent. The next stage
of the project involves nonlinear time domain modelling of the cochlea.
MSc projects
Modelling the complex transfer function of human auditory filters
Leise Borg
Supervisors: Torsten Dau and Morten Jepsen
Many auditory masking phenomena in human listeners can be described on the basis of the
processing of sound through the inner ear. This project investigated the properties of cochlear filtering,
such as tuning characteristics, phase response, and compressive behaviour, using perceptual masking
experiments. Tonal signals and specific complex maskers, called Schroeder phase tone complexes, were
used as the stimuli. The auditory processing of these stimuli was investigated using several cochlear
models and a modelling framework that qualitatively simulates masked signal thresholds. It was concluded
that the curvatures of the filter phase responses in a state-of-the-art cochlear model should stay
negative above the centre frequency and that the amount of compression is too little in the model compared
to the compression in the ‘real’ system to account for the dynamic range in the thresholds of the
human masking data.
Speech intelligibility prediction of linearly and nonlinearly processed speech in noise
Claus Christiansen
Supervisor: Torsten Dau
In this study, a speech intelligibility measure based on a model of auditory perception was developed
to predict the intelligibility of linear and nonlinearly processed speech in noise. The predicted
speech intelligibility is based on the perceptual similarity between a reference signal and the processed
signal. This perceptual similarity is determined by applying the auditory pre-processing to both signals
and calculating the linear cross-correlation. The performance of the proposed model was compared to
the classical speech intelligibility method (SII) and the speech transmission index method (STI). The
three models were evaluated for speech in additive noise and speech-in-noise mixtures processed by a
nonlinear technique termed ideal time-frequency segregation (ITFS). In ITFS processing, a binary mask
is applied to the time-frequency representation of a mixture signal and eliminates the parts of the signal
that are below a local signal-to-noise-ratio threshold. All three models were adequate for predictions of
speech-in-noise, whereas the results for the ITFS processed mixtures were mixed. The SII method was a
43

poor predictor and completely failed to predict the noise vocoding ability of the binary mask. The STI
method worked well for most of the binary masks, except for sparsely distributed masks where speech
intelligibility was strongly overestimated. The proposed model was able to account for both the noise
vocoding effect and the rapid drop in performance for the sparsest masks.
The project was carried out at Oticon with Michael Syskind as the local supervisor.
Schematic of the model. First the clean and processed speech is transformed to an internal psychological representation
by an auditory model. The cross- correlation between the two representations is calculated in short
frames, which all are level classified based the RMS level. A final score is calculated for each level by averaging
the cross-correlation values of the corresponding level.
Audio-visual speech analysis
Torir Hrafn Hardarson
Supervisor: Hans-Heinrich Bothe
This work is a part of a framework to design and implement a multilingual language laboratory
for speech- or lip-reading. The goal of this project was to model the dynamics of mouth movements
during the speech articulation process for an open vocabulary. Video films with one speaker and (relatively)
emotion-free articulation were analysed, and the lip movements were modelled. The dynamic
model is related to characteristic single images of the video clip that represent the phonemes and on
transitions between neighbouring images. A general dynamic model employing parameters that state the
importance of specific changes of the lip contours for the correct articulation of the sounds was used.
The momentary results can predict the courses of the feature sets with specific artifacts when only the
text input is given.
Modelling the sharpening of tuning in nonsimultaneous masking
Marton Marschall
Supervisors: Jörg Buchholz and Torsten Dau
Frequency selectivity refers to the ability of the auditory system to resolve sounds in the spectral
domain. In humans, frequency selectivity can be measured behaviourally with masking experiments.
Results from such experiments show greater frequency selectivity when nonsimultaneous masking
techniques are used as opposed to simultaneous ones. The differences in tuning have been suggested to
occur because of suppression, a nonlinear interaction between the masker and the signal, whose effects
are only observed when the two stimuli overlap in time. Suppression is thought to be a consequence of
the compressive behaviour of the cochlea. As a result of suppression, the increased frequency selectivity
in nonsimultaneous masking may enhance the perception of dynamic signals, such as speech. This
project investigated the relationships between frequency selectivity, compression, and suppression using
a modelling approach. A computational auditory signal processing model was used to simulate two behavioural
measures of frequency selectivity: psychophysical tuning curves and the notched-noise method.
As important part of the signal processing, the model included a dual-resonance nonlinear filterbank
to mimic the processing in the cochlea. An ‘optimised’ version of the model was able to account
44

for instantaneous changes in frequency selectivity due to suppression effects.
Illustration of a model of suppression. Suppression is seen as the difference between the case when only a tone (S)
is introduced (1), and the case when a masker (M) is added (2). The bars correspond to the powers of the stimuli
indicated.
Comparison of speech intelligibility measurements with Odeon model simulations
Sebastian Alex Dalgas Oakley
Supervisor: Torben Poulsen
This project compared speech intelligibility measurements made in real rooms with speech intelligibility
measurements made in equivalent Odeon-simulated rooms. No such direct comparison between
speech intelligibility in real rooms and in the corresponding simulated rooms has been made. Fifteen
normal hearing test subjects participated using the DANTALE 1 and DANTALE 2 speech material.
A small and a larger lecture room were used. Comparisons of speech transmission index values in
the real rooms and the Odeon-simulated rooms were also made. Good agreement was found for most
listening positions, but for positions close to the sound source the intelligibility in the real room was
better than in the simulated room
Left: individual intelligibility results in a real room (blue lines, black: mean) and in the same room simulated
in Odeon (red lines, green: mean). Right: psychometric functions fitted to the data.
Effects of compression in hearing aids on the envelope of the speech signal
Justyna Walaszek
Supervisor: Torsten Dau
The influence of the compression in hearing aids on speech intelligibility was investigated.
Speech signals were presented together with either a speech-shaped noise background or another speech
interferer. Compression in the hearing-aid signal processing affects the temporal envelope of the signals
45

in different ways depending on parameters such as the attack and release time constants of the compressor.
The effects of compression on speech intelligibility were measured in normal-hearing listeners. The
measured data were compared with simulations based on (i) the amount of correlation between signal
and interferer as well as (ii) the change of the signal-to-interferer ratio at the output of the compression
system. The main results of the perception experiments were well correlated with the results of the
above ‘signal-based’ predictors. Specifically, it was found that the effect of compression on speech intelligibility
was largest in the case of fast-acting compression, leading to the worst speech intelligibility
both in the measurements and the predictions. The results could be useful for further investigating the
effects of compression in state-of-the-art multi-channel hearing aids on speech intelligibility in challenging
acoustic environments.
The project was carried out at Oticon Research Centre Eriksholm with René Burmand Johannesson
as the local supervisor.
PUBLICATIONS
Journal Papers
A. Rupp, N. Sieroka, A. Gutschalk and T. Dau: Representation of auditory filter phase characteristics in the cortex
of human listeners. Journal of Neurophysiology 99, 2008, 1152-1162.
M.L. Jepsen, S.D. Ewert and T. Dau: A computational model of auditory signal processing and perception. Journal
of the Acoustical Society of America 124, 2008, 422-438.
T. Poulsen and S.V. Legarth: Reference hearing threshold levels for short duration signals. International Journal
of Audiology 47, 2008, 665-674.
T. Poulsen and H. Laitinen: Questionnaire investigation of musicians’ use of hearing protectors, self reported
hearing disorders, and their experience of their working environment. International Journal of Audiology 47,
2008, 160-168.
E.R. Thompson and T. Dau: Binaural processing of modulated interaural level differences. Journal of the Acoustical
Society of America 123, 2008, 1017-1029.
S. Verhulst, J.M. Harte and T. Dau: Temporal suppression and augmentation of click-evoked otoacoustic emissions.
Hearing Research 246, 2008, 23-35.
Theses
G. Pigasse: Deriving cochlear delays in humans using otoacoustic emissions and auditory evoked potentials. Department
of Electrical Engineering, Technical University of Denmark, ISBN 978-87-911-8489-5. PhD thesis,
2008.
Chapters in Books
T. Dau: Auditory processing models. Chapter 12 (pp. 175-196) in Handbook of Signal Processing in Acoustics,
eds. D. Havelock, S. Kuwano and M. Vorländer. Springer Verlag, New York, 2008.
Conference Papers
H.-H. Bothe: Human computer interaction and communication aids for hearing-impaired, deaf and deaf-blind
people. Proceedings of 11th International Conference ICCHP 2008 (Computers Helping People with Special
Needs), Linz, Austria, 2008, Springer-Verlag, Berlin-Heidelberg, 2008, pp. 605-608.
T. Hardarson and H.-H. Bothe: A model for the dynamics of articulatory lip movements. Proceedings of International
Conference on Auditory-Visual Speech Processing 2008, Tangalooma, Australia, 2008, pp.209-214.
46

J.M. Buchholz and P. Kerketsos: Spectral integration effects in auditory detection of coloration. Proceedings of
DAGA’08, In Fortschr. Akust., Deutsche Ges. Akust., Dresden, Germany, 2008, pp. 177-178.
I. Balazova, G. Clausen, J.H. Rindel, T. Poulsen and D.P. Wyon: Open-plan office environments: A laboratory
experiment to examine the effect of office noise and temperature on human perception, comfort and office work
performance. IndoorAir 2008, DTU, Lyngby, Denmark, 2008.
O. Strelcyk and T. Dau: Impaired auditory functions and degraded speech perception in noise. Proceedings of
DAGA’08, In Fortschr. Akust., Deutsche Ges. Akust., Dresden, Germany, 2008, pp. 179-180.
Abstracts
J.M. Buchholz and P. Kerketsos: Across frequency processes involved in auditory detection of coloration. Journal
of the Acoustical Society of America 123, 2008, p. 3867.
S. Ewert, J. Volmer, T. Dau and J. Verhey: Amplitude modulation depth discrimination in hearing-impaired and
normal-hearing listeners. Journal of the Acoustical Society of America 123, 2008, p. 3859.
S. Favrot and J. Buchholz: A virtual auditory environment for investigating the auditory signal processing of realistic
sounds. Journal of the Acoustical Society of America 123, 2008, p. 3835.
M. Jepsen and T. Dau: Estimating the basilar-membrane input/output-function in normal-hearing and hearingimpaired
listeners using forward masking. Journal of the Acoustical Society of America 123, 2008, p. 3859.
E.R. Thomson and T. Dau: A binaural advantage in the subjective modulation transfer function with simple impulse
responses. Journal of the Acoustical Society of America 123, 2008, p. 3296.
O. Strelcyk and T. Dau: Fine structure processing, frequency selectivity and speech perception in hearing impaired
listeners. Journal of the Acoustical Society of America 123, 2008, p. 3712.
N.W. Adelman-Larsen, E.R. Thompson: The importance of bass clarity in pop and rock venues. Journal of the
Acoustical Society of America 123, 2008, p. 3090.
S. Verhulst, J. Harte and T. Dau: Temporal suppression and augmentation of click-evoked otoacoustic emissions.
Journal of the Acoustical Society of America 123, 2008, p. 3854.
47

2. TEACHING ACTIVITIES
The Danish system for tuition at a university level prescribes close connections between research
and teaching. Consequently, university scientists must devote time for teaching students at undergraduate
and graduate level. As a rule, their teaching will be related to their personal research activities; thus,
nearly all the research projects described in chapter 1 are reflected in the basic and advanced tuition
described below.
At the Technical University of Denmark, BSc, MSc and PhD degrees will be awarded to successful
students after nominal periods of study of three, two and three years. Undergraduate courses are
scheduled to fit into the University’s existing tuition modules, thus permitting the students to compose a
sensible curriculum encompassing their professional requests. DTU also offers a BEng degree, and an
increasing number of BEng students follow the courses on acoustics.
In 2000 DTU launched a series of two-year international MSc programmes. Together, Acoustic
Technology and Hearing Systems, Speech and Communication, Department of Electrical Engineering,
offer an international MSc programme in Engineering Acoustics, coordinated by Mogens Ohlrich. See
www.msc.dtu.dk for information about the application procedure.
Tuition related to preparation for academic degrees (MSc and PhD) is individual, based on the
expertise of an academic member of the staff as adviser to each student, often supplemented with assistance
from industry or other research institutions. Twenty-three students carried out their MSc thesis
work in 2008. Supervision was received by a total of seventeen PhD students enrolled at DTU and four
visiting PhD students from other institutions. These projects have been presented in chapter 1.
SCHEDULED COURSES
Acoustic Technology and Hearing Systems, Speech and Communication offer a series of courses
every year. Most courses are supported by written material prepared for the purpose by staff members.
Each course comprises a series of lectures and laboratory exercises and sometimes excursions. The
courses are listed and briefly described below. Detailed descriptions of all the courses on acoustics are
available on the internet (http://www.elektro.dtu.dk/English/research/at/at_courses.aspx). 1
In addition ‘special courses’ may be given to individuals or small groups of students.
INTRODUCTORY LEVEL
Fundamentals of Acoustics and Noise Control (Finn Jacobsen)
The purpose of this course is to introduce the students to fundamental acoustic concepts (e.g., the
sound pressure and sound power), simple sound fields, wave phenomena such as reflection and interference,
acoustic measurements, fundamental properties of our hearing, room acoustics, building acous-
1 Enquiries from students interested in the courses are encouraged. It can be recommended to contact the relevant
teacher directly (email addresses are available at http://www.elektro.dtu.dk/English/research/at/staff.aspx). Rules
for admission are available at the address http://www.elektro.dtu.dk/English/education/master_programmes
/engineering_acoustics/admission_requirements.aspx. A completed Guest Student Form should be sent to DTU’s
International Affairs:
International Affairs,
http://www.dtu.dk/English/education/International_Affairs.aspx
Technical University of Denmark, Building 101A,
DK-2800 Kgs. Lyngby, Denmark
Tel.: +45 4525 1023
Fax: +45 4587 0216
E-mail: intcouns@adm.dtu.dk
49

tics, and structureborne sound, and thus to give the necessary background for the more advanced and
specialised courses in acoustics. There is one single laboratory exercise.
ECTS credit points: 5.
Building Acoustics (Jonas Brunskog and Cheol-Ho Jeong)
The goal of this introductory course specially designed for students with a background in civil
engineering is to provide the participants with knowledge about the effects of noise on human beings,
the principles of noise control in buildings and in the environment, and the acoustic requirements that
must be fulfilled in a typical building design project.
ECTS credit points: 5.
Environmental Acoustics (Torben Poulsen)
This 3-week course is intended for students with an interest in sound as an environmental factor,
wishing to become able to solve problems as they appear in traffic, in industry and under domestic conditions.
Course work includes noise predictions and field measurements. From 2009 the course will be
given by Cheol-Ho Jeong and Jonas Brunskog.
ECTS credit points: 5.
Signals and Linear Systems in Discrete Time (Hans-Heinrich Bothe)
This course provides fundamental knowledge of digital signal processing including classical and
adaptive system analysis and filter design techniques. It is elective for students of electrical engineering
or acoustics in the fifth semester of the bachelor study.
ECTS credit points: 5.
ADVANCED LEVEL
Acoustic Communication (Torben Poulsen, Jörg Buchholt and Thomas Ulrich Christiansen)
The purpose of this course is to provide the students with a fundamental knowledge about the
elements of acoustic communication. The course comprises topics such as hearing (anatomy, physiology),
hearing loss, perception of sound (psychoacoustics), speech, and speech intelligibility. The aim is
to provide knowledge and comprehension of these topics and their mutual relations for both normal
hearing and hearing impaired persons. The special psychoacoustic measurements methods are applied
in five laboratory exercises. Group work and writing of reports are part of the course objectives.
ECTS credit points: 10.
Technical Audiology (James Harte, Torsten Dau and Torben Poulsen)
The purpose of this 3-week course is to train the students in experimental methods used in hearing
research and audiology. The goal is also to train the participants in oral presentation of scientific
material and to produce, present and discuss a poster.
ECTS credit points: 5.
Auditory Signal Processing and Perception (Torsten Dau)
The purpose of this course is to obtain an understanding of the processing mechanisms in the auditory
system and the perceptual consequences, to learn about functional relationships between the
physical attributes of sound and their associated percepts using a system’s approach, to study sensory
and brain processing and their locations using objective methods such as auditory evoked potentials,
and to learn about clinical and technical applications by applying auditory-model based processing
techniques.
ECTS credit points: 10.
50

Advanced Topics in Hearing Research (Torsten Dau, Jörg Buchholz, James Harte, Thomas
Christiansen, Hans-Heinrich Bothe)
The purpose of this PhD course is to develop the students’ self-learning skills, and to increase
awareness of the recent literature within their own research areas and related topics. The course also
attempts to improve the students’ scientific communication skills and promote timely discussions of
current topics in the field of auditory and audio-visual signal processing and speech processing and perception.
From Biology to Technical Neural Systems (Hans-Heinrich Bothe)
This course gives an introduction to the signal processing of biological systems and to biologically
inspired models for neurons, muscles fibres, and the functionality of neural networks. The topics
include neuroprotheses, computer-brain-interfaces and electronic technologies for artificial hearing and
vision.
ECTS credit points: 10.
Architectural Acoustics (Jonas Brunskog and Cheol-Ho Jeong)
The purpose of this course is to give student a profound knowledge of the theories and methods
of room acoustics and sound insulation. The topics include reflection and absorption of sound, design of
absorbers and of rooms for speech and music, structureborne sound and sound insulation of building
constructions, building codes and test methods.
ECTS credit points: 10.
Electroacoustic Transducers and Systems (Finn Agerkvist)
This course provides the student with knowledge of the fundamentals of acoustic transducers.
Analogies between mechanical, acoustic and electrical systems are considered and applied to loudspeakers,
microphones and communication systems. The student will also gain insight into the fundamental
principles of sound recording and reproduction, as well as audio coding techniques. Laboratory
exercises are a substantial part of the course.
ECTS credit points: 10
Advanced Loudspeaker Models (Finn Agerkvist)
This 3-week course introduces advanced elements in loudspeaker models in order to improve the
validity of the models at high frequencies and/or high levels. The main part of the course is related to
measurement and modelling of the dominant sources of distortion in the loudspeaker and also to introducing
methods for compensation of the distortion. The course also deals with micro-loudspeakers as
used in mobile phones and headsets. The final element in the course is the modelling of ‘break-up’ vibrations
in the loudspeaker diaphragm, which are modelled by finite element calculations.
ECTS credit points: 5
Advanced Acoustics (Finn Jacobsen)
This course is intended to give the student an insight into the fundamental methods of theoretical
acoustics. The topics include sound fields in ducts, modelling of silencers, the Green’s function, radiation
of sound, classical room acoustic modal analysis, statistical room acoustics, numerical methods of
sound field calculation (the finite element method and the boundary element method), fundamentals of
active noise control, outdoor sound propagation, and sound intensity and other advanced acoustic measurement
techniques such as beamforming and near field acoustic holography. Laboratory exercises and
simulation studies are a substantial part of the course.
ECTS credit points: 10.
Sound and Vibration (Mogens Ohlrich, Jonas Brunskog and Cheol-Ho Jeong)
By introducing the principles and laws governing the generation, transmission and radiation of
structureborne sound, the course will enable the student to analyse technical noise and vibration problems,
design practical measures for reduction of structureborne noise in machines, vehicles and build-
51

ings, and apply advanced measurement techniques in the field. Mini-projects, laboratory exercises and
reports are a substantial part of the course.
ECTS credit points: 10.
LECTURE NOTES ISSUED IN 2008
M. Ohlrich: Structure-borne sound and vibration: Introduction to vibration and waves in solid structures at audible
frequencies, Part 1a. Acoustic Technology, Department of Electrical Engineering, Technical University of Denmark,
Note no 7016, 5 th print, 2008. (80 pp.)
52

Appendix A: Extramural Appointments
Finn Agerkvist
Member of the Board of Danish Acoustical Society as well as the its Technical Committee on Electroacoustics
Salvador Barrera-Figueroa
Contact person of EURAMET’s Technical Committee of Acoustics, Ultrasound and Vibration.
Member (via DFM) of CCAUV, Consultative Committee for Acoustics, Ultrasound and Vibration (of
the BIPM)
Hans-Heinrich Bothe
Member of the International Academic Advisory Council, Natural and Artificial Intelligence Systems
Organization (NAISO), Canada
Member of the Editorial Board of International Journal of Computer Research, NOVA Science Publishers,
Commack, NY, USA
Senior member of Institute of Electrical and Electronics Engineers (IEEE)
Torsten Dau
Member of the Technical Committee for ‘Hörakustik’ in the German Acoustical Society (DEGA)
Member of the Scientific Advisory board of the ‘Hanse Institute for Advanced Study’ in Germany
Member of the board of Danavox Jubilee (Organiser of ISAAR symposia, International Symposium on
Auditory and Audiological Research)
Editor of EURASIP Journal on Advances in Signal Processing (Special Issue on Digital Signal Processing
for Hearing Instruments)
Anders Christian Gade
The Rockwool Price, Member of the Board
Finn Jacobsen
Member of the Editorial Board of International Journal of Acoustics and Vibration
Section leader of Handbook of Signal Processing in Acoustics, Springer Verlag, ed. by D. Havelock, S.
Kuwano and M. Vorländer
Member of the Scientific Committee for Fifteenth International Congress on Sound and Vibration,
Daejon, Korea, July 2008
Member of the Scientific Committee for Sixteenth International Congress on Sound and Vibration,
Krakow, Poland, July 2009
Member of the Board of Directors of International Institute of Acoustics and Vibration
Reviewer of proposal for the Czech Science Foundation
Midterm reviewer of EU project
Mogens Ohlrich
Member of ISO/TC 43/SC 1/WG 22, ‘Characterisation of machines as sources of structure-borne
sound’
Torben Poulsen
Member of ISO/TC 43/WG l, ‘Threshold of Hearing’
Convener of ISO/TC 43/SC 1/WG 17, ‘Methods of measurements of sound attenuation of hearing protectors’
Chairman of the board of Danavox Jubilee Foundation (Organiser of ISAAR symposia, International
Symposium on Auditory and Audiological Research)
53

Knud Rasmussen
Chairman of IEC Technical Committee 29, ‘Electroacoustics’
Convener of IEC TC29/WG5, ‘Measurement microphones’
Member (via DFM) of CCAUV, Consultative Committee for Acoustics, Ultrasound and Vibration (of
the BIPM)
Contact person of EURAMET’s sub Technical Committee ‘Acoustics’
Technical Manager of DPLA, Danish Primary Laboratory of Acoustics
Chairman of S529, ‘Electroacoustics’, Danish Standards Association
Technical assessor for United Kingdom Accreditation Service (UKAS)
Technical assessor for Swedish Board for Accreditation (SWEDAC)
Technical assessor for National Agency for Testing and Accreditation (NATA), Australia
Life member of IEEE (Institute of Electrical and Electronic Engineers)
54

Appendix B: Principal Intramural Appointments
Torsten Dau
Head of Hearing Systems, Speech and Communication, one of the eight groups at Department of Electrical
Engineering (from 1 August 2008), and head of CAHR
Member of DTU’s PhD Programme Committee Electronics, Communication and Space Research
Finn Jacobsen
Head of Acoustic Technology, one of the eight groups at Department of Electrical Engineering (from 1
August 2008)
Mogens Ohlrich
Head of Acoustic Technology, one of the sections of Department of Electrical Engineering (until 1 August
2008)
Coordinator of the International MSc Programme in Engineering Acoustics
Torben Poulsen
Coordinator of pedagogical education at Department of Electrical Engineering
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