Room acoustics modeling using the ray-

tracing method: implementation and

evaluation

Licentiate Thesis
University of Turku
Department of Physics
2005
David Oliva Elorza
2
3

A mi abuelo
 4

Abstract
This thesis discusses room acoustics modeling and the basics and drawbacks of different
methods. Then it concentrates on the ray-tracing algorithm, and offers the author's
implementation of the method and an evaluation of its accuracy when comparing
modeled sound decays and predicted acoustical parameters with the corresponding
measured data in three different enclosures.

First, a general introduction of the sound propagation principles and a discussion of the
state of the art in room acoustics modeling are given. How to implement the ray-tracing
algorithm is explained, not only the way adopted by the author but also other
possibilities which could improve accuracy. That is followed by a geometrical and an
acoustical description of the three studied enclosures, and the comparison of the
modeled and measured data. Sound pressure level is predicted with very good accuracy,
while reverberation time is more contradictory due to the negation of the wave nature of
sound which is inherent in the model. The paper ends with a discussion of the results’
accuracy and the reasons for similarities and/or differences.
 5

Preface
In August 2004 I started to write this thesis. In principle, it had to include the form of
the ray-tracing method that I had implemented during my time in the Finnish Institute of
Occupational Health (Työterveyslaitos), together with the evaluation of its accuracy in
some indoor enclosures, but it turned out to include more than that. The referred method
is good, although it is not perfect and it has several drawbacks. Hence, I finally ended
up writing down many things which in principle were outside the scope of the thesis,
but that constituted a part of the whole. That is, I discussed diffraction at great length
although I had not yet implemented it, but just because I thought I might do so some
day. Therefore, it became a very long paper, which could be hard to read for some
readers, useless until page 56 or 70 for others, and for the rest of them utterly pointless.
I do not know yet into which category my thesis reviewers belong, but maybe to one of
them. Their opinion made me work a little harder on the paper, delete about twenty
pages, and reorganize most of it. However, that was very positive, not only because this
thesis now looks better than it did before, but also because I have improved my
knowledge. A lot of time has passed since I typed the first lines of this paper, but on the
other hand, I have been doing some other things since then. For instance, the form of the
algorithm which is currently on my hard disk differs considerably from the one
described in this work, and my expected future work, and even more has already been
done. However, that is definitely another story about which I will perhaps still write
some day. Just one more thing: if you decide to go on with the reading of this thesis,
please, enjoy it!
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Table of contents
ABSTRACT.................................................................................................................................... 4

PREFACE....................................................................................................................................... 5

TABLE OF CONTENTS .............................................................................................................. 6

List of symbols and abbreviations .............................................................................................................8

INTRODUCTION.......................................................................................................................... 9

1. OVERVIEW OF SOUND ....................................................................................................... 13

1.1. Sound: Wave propagation.................................................................................................................13
1.2. Sound: Reflection ...............................................................................................................................16
1.3. Sound: wave effects............................................................................................................................18
1.3.1. Diffraction.................................................................................................................................18
1.3.2. Interference...............................................................................................................................19
1.4. Temporal distributions of reflections; the impulse response..........................................................22
1.5. Diffuse sound field..............................................................................................................................23
1.6. Acoustic levels.....................................................................................................................................24
1.7. Sound absorption ...............................................................................................................................25
1.7.1. Air absorption...........................................................................................................................26
1.7.2. Wall absorption ........................................................................................................................26

2. MODELING METHODS IN ROOM ACOUSTICS........................................................... 28

2.1. Modeling methods..............................................................................................................................28
2.1.1. Empirical and statistical methods...........................................................................................28
2.1.2. Wave-based methods................................................................................................................29
2.1.3. Geometrical acoustics methods ...............................................................................................30
2.1.3.1. Ray-tracing method.....................................................................................................31
2.1.3.2. Image method ..............................................................................................................33
2.1.3.3. Particles........................................................................................................................35
2.1.3.4. Fitting-zone method ....................................................................................................35
2.1.3.5. Cone or pyramids tracing method .............................................................................36
2.1.3.6. Hybrids.........................................................................................................................37
2.2. Diffusion and diffraction in the ray-tracing algorithm...................................................................38
2.2.1. Modeling diffusion in the ray-tracing method .......................................................................38
2.2.1.1. Statistical treatment of diffuse reflections. The scattering coefficient....................38
2.2.1.2. Reflection models with secondary diffuse sources....................................................43
2.2.2. Modeling diffraction in the ray-tracing method....................................................................46
2.3. Considerations about the ray-tracing algorithm.............................................................................49
2.3.1. Frequency range.......................................................................................................................49
2.3.2. Negation of wave nature, interference and diffraction .........................................................49
2.3.3. Angular distribution of reflected sound .................................................................................50
2.3.4. Geometrical description...........................................................................................................51
2.3.5. Frequency dependent factors. The absorption and the scattering coefficients ...................51
2.3.6. Number of rays .........................................................................................................................52
2.3.7. The detection problem .............................................................................................................52
2.3.8. Other problems.........................................................................................................................54
2.4. Round Robin.......................................................................................................................................54

3. IMPLEMENTATION OF THE RAY-TRACING ALGORITHM ................................... 57

3.1. Flowchart of the implemented ray-tracing algorithm ....................................................................58
3.2. Geometrical calculations ...................................................................................................................59
3.2.1. Geometrical description of the enclosure ...............................................................................59
3.2.2. Shooting rays ............................................................................................................................60
3.2.3. Detection of collisions...............................................................................................................61
3.2.3.1. Point-in-polygon test ...................................................................................................61
3.2.3.2. Visibility test ................................................................................................................63
 7

3.2.4. Modeling reflections.................................................................................................................64

3.2.4.1. Specular reflection.......................................................................................................65
3.2.4.2. Diffuse reflection .........................................................................................................65
3.2.5. The image method...........................................................................................................................66
3.3. Energy calculations............................................................................................................................66
3.3.1 The instant power of every ray.................................................................................................66
3.3.2. Reception of sound rays ...........................................................................................................67
3.3.3. Calculation of acoustic indices ................................................................................................68
3.3.3.1. Sound pressure level....................................................................................................69
3.3.3.2. Reverberation time......................................................................................................69

4. MATERIALS AND METHODS............................................................................................ 71

4.1. Description of the enclosures.............................................................................................................71
4.1.1. Reverberant room, Room 1......................................................................................................71
4.1.2. Industrial hall, Room 2a and Room 2b....................................................................................71
4.1.3. Auditorium................................................................................................................................72
4.2. Measuring methods............................................................................................................................73
4.2.1. SWL of the sound source .........................................................................................................73
4.2.2. Measurement of SPL and RT..................................................................................................73
4.2.3. Measurement and estimation of the absorption properties of materials.............................74
4.3. Modeling of the enclosures ................................................................................................................74
4.3.1. Reverberant room, Room 1......................................................................................................74
4.3.2. Industrial hall, Room 2a and Room 2b...................................................................................75
4.3.3. Auditorium, Room 3.................................................................................................................77
4.4. Validation methods ............................................................................................................................79

5. RESULTS ................................................................................................................................. 81
5.1. Reverberant room, Room 1 ...............................................................................................................81
5.2. Industrial hall, Room 2a ....................................................................................................................82
5.3. Industrial hall, Room 2b ....................................................................................................................85
5.4. Auditorium, Room 3...........................................................................................................................88

6. DISCUSSION ........................................................................................................................... 92
6.1. Reverberant room, Room 1 ...............................................................................................................92
6.2. Industrial hall, Room 2a ....................................................................................................................93
6.3. Industrial hall, Room 2b ....................................................................................................................95
6.4. Auditorium, Room 3...........................................................................................................................97
6.5. General discussion..............................................................................................................................99

7. CONCLUSIONS AND FUTURE WORK .......................................................................... 101

8. ACKNOWLEDGMENTS..................................................................................................... 103

9. BIBLIOGRAPHY.................................................................................................................. 104
Appendix 1.
Measurement of sound absorption in a reverberation room, ISO 354............................................ A1-1
Measurement of the random-incidence scattering coefficient of surfaces, ISO/DIS 17497-1........ A1-3

Appendix 2.
Survey of existing commercial software............................................................................................. A2-1
 8

List of symbols and abbreviations

α absorption coefficient 2D two dimensional space
δ scattering coefficient 3D three dimensional space
∆ path difference BEM Boundary Element Method
ϑi incidence angle EDT early decay time
ϑr reflection angle FEM Finite Element Method
FDTD Finite Difference Time Domain
Θ temperature Method
λ wavelength FIOH Finnish Institute of
Σ wave front Occupational Health
ϕ azimuth angle GTD geometrical theory of diffraction
ψi phase shift ISO International Organization for
ω angular frequency Standardization
A area RT reverberation time
Ai amplitude SPL sound pressure level
c speed of sound SWL sound power level
d distance TO transition order
dB decibel
D diffuse
f frequency
F false
h height
I intensity
IL insertion loss
k wavenumber
l height
L length
LI acoustic intensity level
Lp acoustic pressure level
Lw acoustic power level
m air absorption coefficient
M Maekawa function
N Fresnel number, number of rays
P source strength
P pressure, point
Q directivity factor
r distance, radius
r
r vector direction
R receiver
S specular
S sound source
S' virtual sound source
t time
T period
T true
V volume
W sound power
 9

Introduction
There are many reasons for the development and improvement of computer-aided
acoustical modeling of rooms. There are also many reasons to think that no existing
method is exact or 100 % accurate 1. This thesis is about the second of these topics, and
it presents the author's implementation of the ray-tracing algorithm and an evaluation of
its accuracy in the prediction of two acoustical parameters when comparing the
measured data of three different studied enclosures.

Acoustics is important in many different spaces, especially where noise exposure is high
and causes hearing loss or disturbance in the workers, and in places where music and
speech performance is very important, i.e. concert halls and auditoria. In the first case
we could refer to noise and in the second to sound, but the laws that govern both are the
same, and we define it as the problem of room acoustics.

The adverse effects of sound on human beings are numerous. For instance, high levels
of noise and long exposure to it cause hearing loss in industry workers 2. Noise also
affects the workers’ concentration, i.e. in hospitals, schools and in open-plan offices 3.
On the other hand, good acoustics are very important in concert halls, theatres and
auditoria, but also in public buildings. Probably we have all had the experience of being
in an airport or bus station where the announcements coming from loudspeakers are
difficult, if not impossible, to understand.

All these problems have a technical solution. Noise machines can be timbered and the
industrial hall can be acoustically treated 4. The ceiling of open-plan offices can be
made more absorptive, and the same solution can be applied in schools and hospitals. In
concert halls the solutions are normally more complicated, depending on the subjective
opinion of musicians and audience, but normally they consist of achieving relatively
long reverberation time and improving diffusion 5. Almost the opposite is normally
required in public buildings for message announcements. In general, correct acoustic
treatment makes a space more comfortable and desirable.

However, the problem in all of them is basically economic, although it is also a matter
of common sense. If an acoustic treatment costs money, to do it when the construction
is already finished will cost much more. And it is senseless to spend one year designing
a building and another building it, to finally obtain a bad environment. Not only
 10

acoustics but also lighting, humidity, ventilation, temperature, etc. are key elements that
6
affect occupant satisfaction . Acoustic modeling can give information before
construction about the acoustical problems or lucks that a building may have. However,
it seems that communication between architects and acousticians has not been very
good 7. In this thesis, I hope to shed some light on and bring some sense to the acoustic
phenomena involved in room acoustics. Not only for architects or acousticians, but also
for the general public who may find it interesting to know a bit more about it and about
the methods that are normally applied in room acoustics modeling.

Sound is wave motion within matter 8, and the physical phenomena involved in sound
wave propagation in enclosures are both numerous and complex, making overall
analytical modeling difficult, if not impossible, to perform 9. Because of the large
number of parameters to be taken into account for the description of a real situation,
only an approximation of it is possible. That is needed because the equation that
describes the propagation of sound, the wave equation, is not solvable in real enclosures
10
. So the problem is how to find a good approximation, which simplifies the problem
but not too much, as it seems, from the existing literature, that the greater the
simplification is, the less accurate the results become. The use of a digital computer in
room acoustics has opened possibilities for the prediction of acoustical behavior of
11
enclosures , so computer modeling of acoustic enclosures has developed in various
forms, although none of them has yet been demonstrated to have 100 % accuracy 1, 12.

The thesis is organized in four parts. In the first part, an introduction to the general
principles of acoustics is given. It is difficult to set the limits of this part, and some
readers may find no need to read it, as it might not tell them anything new. However, it
has been considered a fundamental part in this work, because many approaches and
their failure or success could not be understood without the knowledge of the wave
propagation principles in enclosures. How sound propagates and distributes, what kind
of movement it is, and the main effects due to the wave nature of sound, i.e. diffusion,
diffraction and interference, are explained. Moreover, the concept of frequency and
wavelength, and the description of the impulse response are given space in this chapter.

The second part of this work deals with existing modeling methods. Their principles,
limitations and accuracy will be the main topics. Existing methods can be separated into
three groups or categories: empirical, wave-based, and geometrical acoustics methods.
 11

The first are simple but highly inaccurate and are not able to represent many of the
effects that are present in wave propagation in complex rooms. The next type, the wave-
based methods, are undoubtedly the most exact, but are very slow and impractical for
most frequencies. The last group of methods studies the propagation of sound by
geometrical laws. Their complexity varies, as does their accuracy, but they have been
considered to be preferred by most of the acoustic community, and the author of this
thesis has implemented one form of them. For this reason, special attention is given to
them and to their limitations. To end the chapter, the results of two international round
robins about the accuracy of existing models are reported in detail.

The next and third part of this thesis offers a discussion about the algorithmic
implementation of the ray-tracing method. If the list of proposed and existing models
seems to be never ending, the way in which they are implemented is rarely described.
The reasons for such secretiveness remain unknown, but even when the description of
the algorithm is not given, their authors are in the habit of claiming their excellent
accuracy, their ability to overcome previous limitations, and the fast and interactive
possibilities of their models. The writer of this thesis has doubt about some
contributions, especially if they offer auralization based on databases or without taking
diffusion into account. The author of this work has developed a new form of the ray-
tracing method. So starting from zero, we have had to solve many problems in the
implementation of this new geometrical acoustics model. One example is the
approximation done to the point-in-polygon test (detailed later), which is repeated in the
innermost loop even millions of times, and which has been improved, giving more
competitive computation times. All the steps to implement the model are described,
with some paid attention to possible limitations or drawbacks.

The last and fourth part, which includes chapters 4, 5 and 6, offers a comparison
between modeling and measured data in three enclosures. These enclosures have been
chosen to cover some of the different types of indoor spaces that need to be modeled in
room acoustics. The first space is the reverberant chamber of our laboratory, which is no
more than a room where the wall absorption is extremely small, and therefore sound
remains for long time. The second space is an industrial hall with different kinds of
machines and with a special geometry. The last room is a small auditorium with ten
audience rows and some acoustic treatment in walls and ceiling. The geometrical and
acoustical descriptions of the enclosures, together with the way they have been
 12

modeled, are detailed in chapter 4. The results from the modeling and the comparison
with measured data are given in chapter 5. Finally, there is a profound discussion of the
accuracy, never perfect but satisfactory, and of the causes and reasons that conditioned
it.
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1. Overview of sound

1.1. Sound: Wave propagation

Sound is wave motion within matter, be it gaseous, liquid or solid, and it exhibits, in
many respects but not in all, behavior similar to other wave motions that we encounter
in nature, i.e. water waves and light waves, whose propagation phenomena are easily
observed 8. A fundamental property of all wave motions is that the matter itself does not
travel with the wave. Rather, the wave represents a disturbance or modification of the
matter, and it is this disturbance that propagates in periodic movements away from the
source (called wave front). In sound propagation, the air being the medium of interest in
room acoustics, the disturbance is an alteration of the atmospheric pressure over and
under its mean value, which produces a periodic movement of the air molecules back-
and-forth along the same direction in which the wave is propagating. These are called
longitudinal waves, while water waves are a case of transversal waves.

The propagation of sound through the air is illustrated in Figure 1.1. In the upper part
the alteration of the atmospheric pressure is illustrated, while in the lower part it is the
motion of the air molecules associated with the propagation of sound. If the intensity of
the sound grows, the gradient of the pressure does too, and as a result more air
molecules are in movement.

Figure 1.1: Sound motion through the air (longitudinal waves)

The velocity of the airborne sound does not depend on the size of the wave, as it does
with water surface waves, neither does it have an electro-magnetic nature as is the case
 14

with light. If the medium is assumed to be homogeneous and at rest the speed of sound
is constant with reference to space and time 13. For air, its magnitude is

c = (331.4 + 0.6Θ ) [m/s] (1.1)

Θ being the temperature in centigrade.

A general principle, first established by Fermat, states that every wave propagates from
the source to the receiver by way of the fastest path: not the shortest, but the fastest. If
the medium is homogeneous through the space of interest, as we consider it to be for the
air, then the speed of sound is uniform through it, and then the fastest path is also the
shortest. That is, the propagation occurs along a straight line.

The constant distance separating two points of equal phase on adjacent waves is called
the wavelength, λ [m], see Figure 1.1. The time elapsed for a point to return to the same
vibration state is the period, T [s]. Hence, the speed of propagation of the wave, c, is
given by the quotient of λ , the distance traveled by the wave between two instants of
equal phase, and T, the time required for this journey.

c = λ /T (1.2)

The reciprocal of the period is called the frequency, f, and it indicates how many periods
occur in a unit of time, or in other words, how many times the particles of air have
moved back-and-forth in a second. In honor of Heinrich Hertz, the frequency unity is
called the Hertz [Hz].

c=λ⋅ f (1.3)

As the speed of sound is constant, long waves have low vibration movement (low
frequencies), and short waves have high vibration movement (high frequencies). The
division of the frequency range into high, mid, and low frequencies is quite arbitrary,
especially their limits. But it is useful as it allows idealizing the wave nature effects
encountered in room acoustics.

14
Octave bands are used in acoustics rather than individual frequencies . When two
sounds have frequencies f1 and f 2 , they are separated by an interval f 2 / f1 and they
 15

define a frequency band of height ∆f = f 2 − f1 . When the relation, f 2 / f1 = 2 , one

frequency is double than the other, the interval is called octave. The acoustic octave
bands are centered in the frequencies that are listed in the first row of Table 1.1. Their
corresponding wavelengths are listed in the second row. The filters that are used to
analyze signals are able to eliminate the components which are outside the passing
band.

Table 1.1: Center frequencies of acoustic octave bands and their corresponding
wavelength.
f [Hz] 63 125 250 500 1000 2000 4000 8000
λ [m] 5.302 2.672 1.336 0.668 0.334 0.167 0.084 0.042

Mathematically the sound propagation in the space is described by the wave equation,
Eq. (1.4), also known as the Helmholtz equation, where p [Pa] denotes the sound
pressure, and x, y, z are the Cartesian coordinates. However, the wave equation can
10
seldom be solved in an analytic form , ant that is the reason for the existence of all
existing and applied approximations.

⎛ ∂2 p ∂2 p ∂2 p ⎞ ∂2 p
c 2 ∆p = c 2 ⎜⎜ 2 + 2 + 2 ⎟⎟ = 2 (1.4)
⎝ ∂x ∂y ∂z ⎠ ∂t

The propagation of sound through the air is explained by the Huygens propagation
principle, which establishes that a wave front ∑ progresses in the space as if every one
of its points would emit elemental spherical waves, being the position of the wave front
after a certain time ∆t the envelope of such elemental waves, see Figure 1.2.

Figure 1.2: Huygens principle. Propagation of a wave front.

 16

In this way, any point source, which spreads its influence equally in all directions
without a limit to its range, will obey the inverse square law, also called the divergence
law. This comes from strictly geometrical considerations, it applies to diverse
phenomena encountered in nature, and it is illustrated in Figure 1.3. The intensity I of
the influence at any given distance r is the source strength P divided by the area of the
sphere.

Figure 1.3: Divergence law for sound propagation.

1.2. Sound: Reflection

The Huygens principle is also applied to sound reflection as is illustrated in Figure 1.4.
Imagine an incident plane front ∑ that arrives at a surface with an angle of incidence θ i .
As before, every point in the surface after being illuminated by the sound wave front
will act like a secondary source radiating semicircular waves. The reflected sound wave
can be obtained by connecting the points that are in phase with each other. In Figure
1.4, only six points (P1, P2, …, P6) that act like secondary sources have been
represented. When the incident wave front arrives at P6, point P5 is already emitting
and its wave has spread the same length as d5-6. This is done in the same way for the
other points.

Figure 1.4: Huygens principle. Specular reflection of a plane wave from a smooth flat
surface.
 17

The reflection illustrated in Figure 1.4 obeys Snell’s law, i.e. the angle of reflection is
equal to the angle of incidence, θ i = θ r , like the mirror reflection of light. That is
considered as a special case of Fermat’s principle, where the reflection path between the
source and receiver via the panel is the one that is traversed in the least time. A specular
reflection of sound can be obtained approximately from a plane, rigid surface with
dimensions much larger than the wavelength of the incident sound 15. However, Snell’s
law fails in the reflections of wave fronts from everyday walls, and this is easy to
understand when we apply Huygens’s principle to the reflection of sound over rough
surfaces.

In the same way as before, we now study the reflection of a plane wave by a surface
whose irregularities are comparable to the size of the wavelength. The case is
schematically illustrated in Figure 1.5, where we observe how the wave front is
‘broken’ in pieces, which is a consequence of the irregularities of the surface, and a
result of the different delay times achieved by the different parts of the wave. This kind
of reflection is called a diffuse reflection and it occurs when the size of the irregularities
and the wavelength are similar. On the other hand, when the wavelength is very long
compared with the irregularities, the distance to the next wave front is so great that the
disturbance can be considered negligible. In the opposite case, when the wavelength is
very short, the distance between points with the same phase is smaller than the size of
the irregularities, and because the reconstruction of the wave front is done over points
with the same phase, the effect may not be noticeable. For a correct reconstruction of
the wave front of Figure 1.5, edge diffraction and interference effects should also be
taken into account, but to the author’s knowledge, there is no existing algorithm able to
solve it mathematically in an efficient and simple way.

Figure 1.5: Huygens principle. Reflection of a plane wave front from a surface, whose
irregularities are comparable to the wavelength of the sound wave.
 18

In a diffuse reflection the sound incident to a surface is reflected into a wider solid angle
12 16
than that of a specular reflection , and it is said that the sound has been scattered .
Scattering of sound inevitably occurs when sound is reflected from a surface, not only
by the fact that real life surfaces are not perfectly smooth nor uniform, but also because
of the effects created by their finite size. This last case corresponds to diffraction
effects, which are explained more deeply in the next section. The effect of a diffuse
reflection is simplified in Figure 1.6, where the arrows illustrate the direction of the
front waves. The ideal reflection follows Snell’s law, i.e. θ i ≡ θ rS , and a diffuse

reflection is such that θ i ≠ θ rD . A larger problem lies in the difficulty of determining the
direction of the sound wave after such a reflection, considering the possibility that there
can be more than one.

Figure 1.6: Reflection of sound by a rough surface.

1.3. Sound: wave effects

Effects due to the wave nature of sound are described in the following. These effects
depend directly on the wavelength of sound, i.e. diffraction and interference being more
pronounced at low frequencies, while scattering due to the roughness of surfaces is
more pronounced at high frequencies. This last case has just been discussed.

1.3.1. Diffraction
Diffraction is the local change in the direction of the propagation of sound waves
passing the edge of an obstacle. This is one of the most important acoustic phenomena
caused by the wave nature of sound 17, and also one of the most complex to solve. The
diffraction phenomenon depends significantly on the ratio of the sound wavelength to
the size of the obstacle. The longer the wavelength, the stronger the sound diffracts.
Diffraction theory has been extensively studied in optics. But, being an effect that
 19

depends basically on the wavelength of the wave, its application and representation in
acoustics are quite difficult.

When a sound wave encounters an obstacle, which is small in relation to its wavelength,
the wave passes round it almost as if it did not exist, forming very little shadow. But, if
the frequency of the sound is sufficiently high and the wavelength is therefore
18
sufficiently short, a noticeable shadow is formed . Due to the consequences for
acoustics and room acoustics, diffraction effects can be divided into three groups; by a
barrier, by an edge, and by an aperture, and are illustrated in Figure 1.7, where the ideal
cases for low and high frequencies are represented.

For instance, the most obvious effect of diffraction by a barrier is the presence of sound
behind it. On the other hand, diffraction by an edge or a corner causes diffusion or
scattering of sound. Diffraction by an aperture is a general cause of energy losses, and it
is not treated in greater detail in this study.

Figure 1.7: Diffraction by a barrier, by an aperture and by an edge. Upper cases for
ideal low frequencies. Lower cases for ideal high frequencies.

1.3.2. Interference
In room acoustics there are many important interference phenomena, and these can be
only simulated if reflections of pressure waves with amplitude and phase are considered
19
. Although, as we saw in section 1.1, sound propagation corresponds to the motion of
 20

longitudinal waves, we can visualize interference by the principle of superposition

applied to sinusoidal waves (however, the atmospheric pressure gradient corresponds to
such movement). The principle of superposition may be applied to waves whenever two
or more are traveling through the same medium at the same time. The waves pass
through each other without being disturbed. The net displacement of the medium at any
point in space or time is simply the sum of the individual wave displacements. For
instance, if two waves with the same amplitude ym and wavelength λ = 1 / k are traveling
in the same direction along the same line x, by applying the principle of superposition
the resulting medium displacement may be written as:

⎛φ ⎞ ⎛ φ⎞
y ( x, t ) = ym sin(kx − ωt ) + ym sin(kx − ωt + φ ) = 2 ym cos⎜ ⎟ sin ⎜ kx − ωt + ⎟
⎝2⎠ ⎝ 2⎠

(1.11)

which is a traveling wave whose amplitude depends of the phase φ . When two waves
are in-phase, φ = 0 , they interfere constructively and the result has twice the amplitude
of the individual waves. If the two waves have opposite-phase, φ = π , they interfere
destructively and cancel each other out.

However, if two waves are traveling in opposite directions they create a standing wave,
which appears to be still vibrating in place. So if two waves of the same kind are
traveling in opposite directions, the principle of superposition will lead to:

y ( x, t ) = ym sin( kx − ωt ) + ym sin( kx + ωt ) = 2 ym sin kx cos ωt (1.12)

This wave is no longer a traveling wave because the position and time dependence have
been separated. The amplitude of the wave, 2 y m sin kx , does not change position
(absence of propagation), but stands still and oscillates up and down according to
cos ωt . Characteristic of standing waves are locations with maximum displacement
(antinodes) and locations with zero displacement or nearly complete cancellation
(nodes). Nodes and antinodes, if present, have a fixed distribution in the space.

Standing waves are commonly set up in rooms by low frequency sound with a
wavelength being a multiple of half the length of one side, Eq. (1.13).
 21

2L
λn = n=1, 2, 3… (1.13)
n

To ensure that interference effects are stable in time the room walls need to be highly
reflective. Standing waves contribute to higher ambient noise levels, and therefore are to
be avoided in acoustic design, particularly in industry. They should also be avoided in
concert halls to ensure uniform frequency response and good diffusion. In typical
industrial spaces, interference effects are reduced by the fittings in the space and the
large number of sound sources which are likely to be non-coherent 20.

However, interference is not only observable between parallel walls, as there are also
19, 22
interference effects at reflecting boundaries of reverberant sound fields . Even in a
reverberant or diffuse field, where at all points there is equal mean energy flows in all
21
directions, Waterhouse stated that the sound energy is distributed into interference
patterns at the reflecting boundaries. Thus, the mean energy density is not uniform at all
points in the field. The phases of the wave trains near the reflecting surfaces are not
entirely random, and this lack of phase-randomness gives rise to the interference
patterns. More simply explained, the incident sound interferes with the reflected sound.
21 23
In their studies, Waterhouse and Waterhouse and Cook found the energy level at
the surface to be 2.2 dB higher than at points further away where the interference
patterns are negligible, with the largest departure from uniformity at the corners, with
levels 9 dB higher that at remote points.

In Figure 1.8a we observe a wave front arriving at a corner and the area where both
fronts, incident and reflected, will interact. In Figure 1.8b we observe the interference
pattern at a reflecting boundary when plane sound waves are randomly incident over the
positive quadrant. The numbers on the contours give the theoretical departure in dB
from the steady level, as a function of distance in wavelengths ( z / λ and y / λ ) from
two rigid reflecting planes xy and xz, intersecting at right angles, both planes being
perpendicular to the plane of the paper.
 22

Figure 1.8: a) Incident wave front to a corner. b) Interference pattern at the corner.
After Waterhouse and Cook 22.

1.4. Temporal distributions of reflections; the impulse

response
The impulse response is the time domain response of an enclosure to an idealized
infinitely short impulse (Dirac function). In room acoustics it is also understood as the
temporal distribution of reflections.

The sound propagation from a sound source to a receiver is characterized by the impulse
response, and the spatial information on all possible propagation paths between the two
positions is reflected in this signal. The first detection always corresponds with the
direct sound, and after it, first and multiple reflections are received. Their time delays
with respect to the direct sound are according to the lengths of the traveled paths, and
their pressure intensities depend on the air absorption of sound, and also on the
absorption characteristics of the surfaces involved in every path. Both ways of sound
attenuation are explained in section 1.7. For every frequency there is a different impulse
response, although it is possible to obtain them in octave bands. Every detection is also
characterized by its phase, and if that is neglected, then we are talking about an energy
response. In Figure 1.9, an idealized energy response is presented, which has been
separated into three parts: direct sound, early or first reflections, and late reverberation.
The selection of the limit between early and late reflections is quite arbitrary. In speech
it is normally 50 ms, while in music it is 100 ms.
 23

Figure 1.9: An idealized energy response.

The measured and/or modeled impulse responses give a lot of information about the
acoustic characteristics of the enclosure. From these, it is possible to calculate many
24
acoustic parameters, i.e. reverberation time RT and early decay time EDT . The
calculation of these parameters is based on the early-to-late energy ratio of the impulse
response.

It may be desirable, for instance, and especially in acoustics design, to listen to a virtual
environment before it has been constructed. To do that, the auralization process has to
be complete, and the impulse response is a crucial part of it. The process used to obtain
an auralized signal is not described in this thesis, but the reader can obtain basic
information about it from references 45 and 48.

1.5. Diffuse sound field

A sound field in an enclosure is composed of many single waves. Each of them has its
own particular amplitude, phase, and direction. However, when different conditions are
fulfilled it is possible to ignore these, and then we are talking about the existence in the
enclosure of a diffuse sound field.

Waterhouse stated in 195521 that a diffuse sound field is characterized by the following
three properties: 1. Uniform distribution of sound energy, i.e. equal mean energy density
at all points. 2. Equal mean energy flow in all directions at all points in the field. 3.
Random phase relation between the wave trains converging on any point. Forty years
25
later, Hodgson arrived at the same conclusion , and resumed that in a diffuse sound
field, at any position, sound waves are incident from all directions, with equal intensities
 24

and with random phase relations 26. Perfect diffusion is achieved in a box of volume d3,
when d << λ .

From another point of view, Beranek 27 considered that two requirements need to be met
for diffusion of the sound to be good: First, the reverberation time must be fairly long,
because the sound will have died out after relatively few reflections if the room is not
reverberant. Since sound to be diffuse, must undergo many reflections in a room, a long
reverberation time contributes to diffusion. Second, the ceiling and walls of the hall
must be irregular so that the sound waves are scattered when they are reflected from
them.

The role of sound diffusion in the good acoustics of an auditorium has been difficult to
prove, as diffusion has been difficult to measure, and the role of diffusing elements or
room surfaces difficult to quantify 28. It happens that while some acoustic engineers can
blame the disappointing acoustics of certain major halls on a lack of surface diffusion,
one eminent concert hall designer regularly claims that too much diffusion is
detrimental to the sound quality of the upper strings 28.

On the other hand, the existence of diffuse sound field in an enclosure allows the
application of simplified prediction models, and also of some important simplifications,
e.g. negation of interference.

1.6. Acoustic levels

In acoustics, logarithmic scales are used to express sound levels, as the human audible
range for the intensity varies between 10-12 and 10 watts per square meter, and as the
human hearing has not a linear but a logarithmic response to the change in sonic
perturbation 13. The unit used is the decibel, dB. A level is the magnitude of a signal in
reference to an arbitrary reference. In other words, a level is the logarithm of the
relation between a certain magnitude and a reference one of the same class. It is needed,
to express the base of the logarithm, the reference magnitude and the kind of level. One
thing worth mentioning about the utility of acoustic levels, is that changes of 5 dB give
the same change in the hearing feeling whatever the reference level, but a change of
0.01 Pa is a big change at low levels, while not perceptible at high ones.
 25

The acoustic power level Lw of a sound source is:

⎛W ⎞
LW = 10 log⎜⎜ ⎟⎟ where W0 = 1 pW (1.14)
⎝ W0 ⎠

The acoustic intensity level LI is:

⎛ I ⎞
LI = 10 log⎜⎜ ⎟⎟ where I0 = 1 pW/m2 (1.15)
⎝ I0 ⎠

The acoustic pressure level LP is:

⎛ p⎞
L p = 20 log⎜⎜ ⎟⎟ (1.16)
⎝ p0 ⎠

where p0 = 20 µPa , and the factor 2 in the exponent of the logarithm is obtained as the
sonic power is proportional to the square of the pressure.

The combination of two levels, i.e. LI1 and LI2, can be done graphically using a diagram
to add levels in dB 13, or analytically, by solving from Eq. (1.15) the intensities I1 and I2.
The total intensity is the sum of both, and from it we obtain the corresponding level.
That is:

I1 = I 0 ⋅10 0.1L I 1 and I 2 = I 0 ⋅ 10 0.1L I 2 (1.17)

(
I T = I1 + I 2 = I 0 10 0.1L I 1 + 10 0.1L I 2 ) (1.18)

( )
L IT = 10 log 10 0.1LI 1 + 10 0.1LI 2 = 10 log
IT
I0
(1.19)

1.7. Sound absorption

There are two mechanisms of sound absorption in room acoustics. The first is a
29
consequence of atmospheric absorption effects . The second is a consequence of the
absorbent properties of surfaces.
 26

1.7.1. Air absorption

The attenuation coefficient of the air, m, depends on the ambient atmospheric
temperature and pressure, molar concentration of water vapor, and frequency. While a
pure-tone propagates through the air over a distance L, its power W decreases
exponentially according to Eq. (1.20), where Wini is the initial power.

W ( L) = Wini e − mL (1.20)

The formula that allows the calculation of m is not considered in this work, but it can be
found in reference 100. In Figure 1.10, the values of m for a temperature of 20 0C and a
relative humidity of 60 % are given. Although we have illustrated the values of m in
decibels per meter, we can see how the absorption of sound increases as the frequency
does, and thus the exponent in Eq. (1.20) tends to zero.

Atmospheric Attenuation of Sound in dB/m

(Temperature 20°C, Humidity 60 %)
0.10

0.09

0.08

0.07
Attenuation, dB/m

0.06

0.05

0.04

0.03

0.02

0.01

0.00
1 10 100 1000 10000 Frequency, Hz

Figure 1.10: Atmospheric attenuation of sound in dB/m

1.7.2. Wall absorption

The second mechanism of sound absorption in room acoustics is caused by the
absorption characteristics of the surfaces. In this case, absorption is defined as the
dissipation of sound energy on striking a physical surface. At every reflection, a portion
α of its energy or power is absorbed as described in Eq. (1.21). This factor α is called
the absorption coefficient. The portion of sound energy not absorbed is either bounced
back as reflected sound, or transmitted through the material. However, when studying
the acoustics of an enclosure, the factor which is transmitted can be considered as if
being absorbed.
 27

W = Wini (1 − α ) (1.21)

30
As described by Yuzawa the sound absorption depends upon the angle of
incidence, α = α (θ ) . Figure 1.11 illustrates the results for feather glass. It is important
to notice how the values for this material remain more or less stable until angles
approach grazing incidence, and the extrapolation which is done between 70 and 90
degrees. At grazing incidence the value is zero.

Figure 1.11: Oblique incidence sound absorption of feather glass. After Yuzawa 30.

The standardized method to measure the absorption coefficients of materials is

described in Appendix 1.
 28

2. Modeling methods in room acoustics

Different modeling methods applied in room acoustics are discussed in this chapter.
Because the final algorithm used is based on the ray-tracing method, special attention is
given to geometrical acoustics methods. In Appendix 2, a review of different existing
commercial software for room acoustical modeling is given.

2.1. Modeling methods

The physical phenomena involved in sound wave propagation inside enclosures are both
numerous and complex, making overall analytical modeling difficult. Because of the
large number of parameters to be taken into account in the description of a real
situation, only an approximation of it is possible 9. There are several approximations,
some better than others, but still no one has been proved to efficiently solve both the
high and the low frequency problem. From the computational point of view, there are
three different approaches; Empirical methods, wave-based methods, and geometrical
10
methods . All of them are overviewed in this chapter, although only the last one is
treated in detail.

47
In 1968, Krokstad et al carried out the first application of ray-tracing computer
modeling. They studied the space and time distribution of the early reflections over the
audience area of a concert hall, and also stated that the use of the digital computer opens
up possibilities for the prediction of the acoustical behavior of concert halls. Since then,
many different approaches and implementations have been carried out, but although
computer modeling of acoustic enclosures has developed into various forms, none has
yet demonstrated 100 % accuracy 1, 12. The problem lies in the wave nature of sound and
in the complexity of its propagation and reflection in enclosures. Eq. (1.4) has a
different solution for each frequency.

2.1.1. Empirical and statistical methods

Empirical methods belong to the first group of approximations, the Sabine and Eyring
models being the most popular ones. Their methods offer a rough estimation in octave
bands of parameters like sound pressure level, SPL [dB], Eq. (2.1), and reverberation
time, RT [s], Eq. (2.2). They consider the existence of a diffuse sound field in the space.
In that way, only the size of the room and the importance of absorption, but not of the
location of each of the surfaces is considered 21. The Sabine and Eyring theories are still
 29

used to predict noise levels in factories 32. Diffuse field theory is used by practitioners to
predict sound fields in rooms of every type, but what is often forgotten is the fact that
the theory is based on assumptions which limit its applicability. If the theoretical
assumptions do not hold in the case of a particular room, the predictions may not be
26
accurate . For instance, diffuse field theory can not be applied in highly absorptive
rooms. Other methods and refinements have been proposed, e.g. by Hodgon 33, Kuttruff
34
, Thompson et al.35, Cotana 36
and others 37, 38
. However, those methods have their
own approximations and limitations, and have not proved to be efficient over a variety
of enclosures. Nor are empirical methods suitable for auralization.

⎡ 1 1−α ⎤
SPL (r ) = L w + 10 log ⎢ +4 (2.1)
⎣ 4πr
2
A ⎥⎦

V
RT = 0.16 (2.2)
A

with ASabine = 4mV + ∑α i Si (2.3a)

i

and AEyring = 4mV − ∑ Si ln (1 − α i ) (2.3b)

i

where Lw [dB] is the source power level (re. 1 pW), r [m] the distance from the source,
m [m-1] the air absorption coefficient, V [m3] the volume of the enclosure, α i the
absorption coefficient of the surface i, and Si [m2] its corresponding area.

2.1.2. Wave-based methods

Wave-based methods can not solve the wave equation, but they try to numerically
39
approximate its solution. These are, FEM (finite element method) , BEM (boundary
element method) 40, and FDTD (finite-difference time-domain method). Basically, they
divide the space into small elements or nodes. These elements interact with each other
according to basics of wave movement phenomena, for instance, as linear combinations
of a finite number of parameters. The size of these elements has to be much smaller than
the size of the wavelength (at least six nodes) for every particular frequency, and that is
a problem at high frequencies. The number of natural modes in a room increases
approximately with the third power of the frequency, which means that for practical use,
 30

wave models are typically restricted to low frequencies and small spaces 1. Wave
models are also characterized by their creation of very accurate results at single
frequencies. The most difficult part in their application is the definition of the boundary
conditions, as a complex impedance of the surfaces is required, and these are sometimes
very difficult data to find from the existing literature 10.

FEM is primarily used when only the interior acoustic field of an enclosure is to be
computed. In FEM, the complete space is represented by elements, creating matrices
that are large but sparsely filled. Each of these elements only interacts with the elements
directly adjacent to it. On the other hand, BEM can be used to compute the interior,
exterior, or both fields simultaneously. In BEM, only the boundaries of the space are
discretized, but all nodes are linked to each other, forming a dense coefficient matrix.
The main principle in the FDTD technique is that derivatives in the wave equation are
replaced by corresponding finite differences.

Those methods are only suitable for small enclosures and for low frequencies due to
their extremely long computation times. They are mostly applied in studies of the
automotive industry where control of vibrational sound is the main interest, although
they have also been applied, e.g. in studying the performance of noise barriers 41.

2.1.3. Geometrical acoustics methods

The next group of methods follows high frequency approximations based on
geometrical propagation paths 42. In principle, they consider the propagation of sound
through the air in straight lines, avoid the wave nature of sound, and model in one or
43, 16
another way reflections from boundaries. There are many of them . They can trace
rays, particles, cones, even pyramids, and model in one or another way diffraction
12
and/or diffusion. However, none has proved to be exact and some of them are not
even good enough. Those methods are also called energy-based methods, because in the
end, the only thing they consider is the energy which was assigned to every ray or
portion of wave (instead of amplitudes), and after all, they are also a statistical
approach. However, geometrical acoustics methods generally have an acceptable degree
of accuracy in the prediction of different acoustic parameters. They are also able to
achieve the temporal and spatial distribution of reflections, and so they are suitable for
44, 45, 46
auralisation . It has been said that geometrical room acoustics can reflect only a
partial aspect of the acoustical phenomena involved in a room, but that, however, is an
 31

aspect of great importance 13. In the following we will look at the principles of every
method, also at some of their drawbacks. Special interest will be paid to the ray-tracing
method, although many parts in the implementation are also common to the other
geometrical methods.

13
As mentioned in Kuttruff’s work , to apply wave theory in the resolution of sound
propagation is more than a difficult task. Instead, we can apply a greatly simplified way
of description -just as in geometrical optics- by employing the limiting case of
vanishingly small wavelengths, i.e. the limiting case of very high frequencies.
Therefore, the wave nature of sound is neglected, and the propagation of sound is
studied like the propagation of sound rays (portions of spherical waves with vanishing
aperture). This assumption is valid when the wavelength of sound is small compared to
the area of the surfaces of the room, and large compared with their roughness. This is
frequently met in room acoustics. At a medium frequency of 1000 Hz corresponding to
a wavelength of 34 cm, the linear dimensions of the walls and ceiling, and also the
distances covered by the sound waves are usually larger that the wavelength by orders
of magnitude. All phenomena due to the wave nature, such as diffraction and
10
interference, are ignored , since propagation in straight lines is its main postulate 13,
i.e., when several sound field components are superimposed their mutual phase relations
are not taken into account; instead, their energy densities or their intensities are simply
added.

2.1.3.1. Ray-tracing method

The concept of ray tracing is very simple and is represented in 2D in Figure 2.1. The
sound power emitted by a sound source is described by a finite number of rays, which
will be considered as carriers of power (or of energy or intensity). These rays travel
through space at the speed of sound, and are reflected after every collision with the
room boundaries. During that time, their energy decreases as a consequence of the
sound absorption of the air and of the walls involved in the propagation path. When the
rays cross predefined volumes, the receivers, an energy calculation process is
performed, and those data stored. Finally the impulse response at every receiver is
obtained, and with it all desirable acoustic parameters.
 32

Figure 2.1: 2D representation of the ray-tracing algorithm.

If the sound source has omni-directional characteristics, the directions of the virtual
sound rays are created from homogeneous random distributions, although special
directivity characteristics of sound sources can also be accounted for. The possible path
for every new ray emitted from the sound source needs to be found. Considering the ray
like a vector that changes its direction upon every reflection, it is possible to determine
which and only which surface will be collided with next and where. This is called the
point-in-polygon test, see section 3.2.2, and it is a process that is repeated many more
times. A wise implementation of this loop is recommended when complex room shapes
also involve a large amount of surfaces. After this the receivers are checked to see
whether they are crossed by the ray. The energy that the ray ‘impinges’ to a receiver
depends directly on the length that it has traveled inside it, and this energy is stored in
the time domain with its corresponding time delay. This is how the correspondent
energy response is obtained at that point. The attenuation of the energy is calculated
according to the absorption of the air and the length of the traveled path, and according
to the wall absorption properties. The collision with the surface can be considered
specular, following Snell’s law, or diffuse, following a direction other than specular.
Different ways to model diffusion are detailed in section 2.2.1. The sound ray now has a
new direction and the next surface to be collided with has to be found. Here the loop
goes back to the starting point and the process is repeated until the energy of every ray
has fallen under a certain limit. The selection of this limit is arbitrary depending on
which parameters are being sough. In the case of calculation of RT or detailed
echograms, the limit should be one millionth of the initial energy (a decay of 60 dB). If
only SPL is desired, the limit can be reduced to 20 or 30 dB. The overall process has to
be repeated again for every frequency band, unless the directional distribution of the
diffuse reflected sound is considered constant in the frequency range. In any case, the
air and wall absorption properties are frequency-dependent.
 33

We can see already that the divergence law (1/r2 law) is implicit in the ray-tracing
algorithm. It is taken into account by the constant area of the receiver, which comprises
a solid angle decreasing proportionally to 1/r2 as the distance to the source increases 47.

Although a description of the ray-tracing algorithm has been given, its detailed
implementation is described in Chapter 3. There, for instance, it is explained how to
detect the collided surfaces (the point-in-polygon test) and the receivers (detection
problem), how the sound is absorbed, or how the energy calculation is performed.

Ray tracing has been shown to predict noise levels with very good accuracy, although
32, 48, 49, 50
the accuracy is always low at low frequencies . That is logical since the wave
nature of sound is neglected, that nature being the reason for important effects like
diffraction or interference.

2.1.3.2. Image method

To model an ideal impulse response all the possible sound reflection paths should be
10
discovered and this is guaranteed by the image method 42. It can determine to a high
degree of accuracy in terms of level, arrival time and direction, the early reflections 51,
which are the most important in the subjective hearing feeling of listeners 52. However,
they could also create high order specular reflections which will be much more accurate
than would be possible with a real sound wave 51. A disadvantage of this method is that
it requires very long calculation times for higher order reflections in room geometries
that are not simple rectangular box shapes. Increasing reflection order leads to an
exponential increase in image sources, and therefore, an exponential increase of
computation time, which makes the method ineffective 10, 53.

In the image method, reflected paths from the real source are replaced by direct paths
from reflected mirror images of the source. This process is quite simple and it is
illustrated in Figure 2.2 (subscripts used in this figure are f: floor, c: ceiling, w: wall).
The sound source, S, is mirrored to the other side of every surface, so virtual sources are
created, e.g. through the ceiling S’c1 and S’c2, and through the floor S’f. The reflected
path is then replaced by a virtual path that goes from the virtual source to the receiver
location R. Then we need to find out if the path really exists, and that is done by the
visibility test, which has two parts. The first part checks whether the collision point of
 34

the virtual path with the surface is inside the surface or not, in a similar way as is done
in the ray-tracing method. For instance, in Figure 2.2, we observe how the virtual source
S’c1 can not directly see the receiver while S’c2 can. The second part of the visibility test
checks if there are any obstacles that could obstruct the path. This is done by a
backwards reconstruction of the propagation path, starting from the receiver location
and finishing in the sound source, via all the walls that have contributed in the
reflections. In this way, it would be found that the virtual source S’f can not contribute
to the impulse response obtained at R. Multiple reflections are found in the same way by
mirroring virtual images through other surfaces, i.e. the path ‘source-ceiling2-backwall-
receiver’ is obtained when mirroring the virtual source S’c2 through the back wall, from
where S’c2-w is obtained, and then tracing all the paths backwards. Descendants of
invisible sound sources must still be considered, as high-order reflections may generate
visible virtual sources. It is important to notice, finally, that the locations of the image
sources are not dependent on the listener's position, and only the visibility of each image
source may change when the listener moves.

In the same way as with ray-tracing, and due to the computational requirements for
finding high order images, the implementation of the image method needs to be as wise
13
as possible. There are simplified algorithms for simple-shaped spaces , and the
extension to arbitrary polyhedra was described by Borish 54.

Figure 2.2: 2D representation of the image method.

 35

Although it has been stated that the image method is more efficient than ray tracing 55,
this is not true and it is negated by several experimental studies 16, 49, 50, 56, 57.

2.1.3.3. Particles
The particle-tracing method is a simplification of the ray-tracing method. The procedure
is the same, but particles are traced instead of tracing rays. These also travel in straight
lines and are reflected by the room boundaries. The difference lies in the way sound is
attenuated. Instead of decreasing the energy of the particles at reflections, the particle
disappears or continues. To do this, a random number between 0 and 1 is simply created
and compared with the absorption coefficient of the corresponding surface. If the
number is smaller than the absorption coefficient the particle disappears, otherwise it
continues. The point of doing this could be to speed up the algorithm and save
computational time. But this method requires a very large number of particles to achieve
realistic results, and those come close to the results achieved by ray tracing when the
number of particles tends to infinite. However, the simplification is far from
representing the reality, and still has all the limitations common to ray-tracing models.

2.1.3.4. Fitting-zone method

The fitting-zone method is another feature of the ray-tracing algorithm. Instead of
carefully modeling the interior of a space, the space is subdivided into a number of sub-
volumes, and to each of them is assigned a fitting scattering cross-section density and a
32
fitting absorption coefficient . It is possible to consider the fitting density
homogeneous in the space (isotropic zoning), or to concentrate it close to the floor
surface or in certain areas (anisotropic zoning)58. Then, two kinds of approaches can be
used when the ray tracing is performed. The first possibility is to follow each of the rays
that emanate from the source until they strike a surface or obstacle. The ray is then
redirected according to the appropriate reflection law (specular reflection in the case of
a surface, diffuse reflection follows random or Poisson distribution in the case of an
obstacle). The other possibility is to compute random numbers every time a ray crosses
a sub-volume and compare them with the corresponding fitting density. When the
random number is smaller that the density the ray is randomly redirected, otherwise it is
just left to go through. In both possibilities the power of the ray decreases according to
the traveled path (air and fitting absorption).
 36

32
In 1989, Hodgson carried out a study on factory noise prediction with this approach,
comparing predicted and measured sound pressure levels. Although he found
differences at individual points in the range from –7 to +6 dB, with low accuracy at
lower frequencies, and with an equal tendency to overestimate and underestimate, he
concludes that the model is shown to give excellent prediction accuracy. A similar study
was published in the same year by Ondet and Barbry 9, who studied an application of
this method while studying sound propagation in fitted workshops. Although they
demonstrated the influence of fittings in noise levels and affirmed that predicted results
agreed closely with measured ones, the results were only obtained at 2 kHz and did not
58, 59, 60
prove the method to be efficient. Dance also studied the influence and
representation of fittings in a computer prediction of sound propagation in enclosed
spaces. He considered that computer models for the prediction of sound fields in rooms
were complex and required a skilled acoustician to use them effectively, and hence there
was a need for simpler models. Considering that, typically, a workroom is rectangular,
with uniform wall absorption and all fittings located in the floor, Dance suggested
models that require a minimum amount of input data, reducing the scope of the model to
only two or three acoustic parameters. From the data shown in his studies it appears that
the more detailed the description of the room, the more accurate and consistent results
were obtained. This implies that a large simplification of the environment leads to
inaccurate results.

2.1.3.5. Cone or pyramids tracing method

It is also possible to trace cones, approximate cones, pyramids, etc, in practice portions
of solid angles, with a sphere coverage that spreads from the sound source with
divergence. In this way, computation time can be reduced, although the implementation
can be very difficult.

The idea is illustrated in 2D in Figure 2.3. A sound source S and two receivers, R1 and
R2, are inside a five-side polygon abcde . The vertices of the cones are equivalent to
mirror sources (section 2.1.3.2) and mirrored in the familiar way, i.e. Sa. The cones are
hence, mirror image sources with built-in visibility borders 61. In Figure 2.3, we can see
how R1 is outside the illuminated region of Sa-d, while R2 is inside it. We also see how
the cone of Sa is clipped by side d.
 37

Figure 2.3: 2D representation of cone-tracing method in a five-side polygon. S is

mirrored through side ‘a’, giving Sa. Sa is mirrored through side ‘d’, giving Sa-d. Sa-d
illuminates R2 but not R1.

One of the first applications of this method was carried out by the commercial software
Ramsete 62, where triangular beams were generated at the sound source. The central axis
of each pyramid was traced as in ray tracing, being specular-reflected when a surface
was hit. The three corners follow the axis, all being reflected at the same plane. It
considers point receivers, and detection occurs when the receiver is inside the pyramid
being traced.

2.1.3.6. Hybrids
There is the possibility of combining different features of different methods, i.e.
hybrids. For instance, the image method could be applied to calculate very accurately
the first reflections, and then switch to ray tracing for the later part of the impulse
63
response. Another possibility includes the natural combination of a rigorous
numerical method for low frequencies, i.e. BEM, and a more efficient method for higher
frequencies, i.e. the ray-tracing method.
 38

2.2. Diffusion and diffraction in the ray-tracing algorithm

This thesis centers its attention now on the ray-tracing algorithm. In section 2.1.3.1 its
basic concept was introduced, i.e. how sound rays are shot from the sound source and
followed through the enclosure. Before we can describe the way to implement the
algorithm, two aspects have to be discussed, and they are the different ways to model
diffusion and diffraction of sound.

2.2.1. Modeling diffusion in the ray-tracing method

It is presently admitted that accounting for the scattering or diffusion of sound by
47
surfaces improves room acoustics predictions . The investigations carried out by
13 56, 57 16 49, 50
Kuttruff , Hodgson , Lam , and others , have shown that room acoustic
calculations which are based only on specular reflections overestimate reverberation
12
times in situations where diffusion is significant. For instance, Howarth and Lam
studied, in seven enclosures, the relation of the prediction of acoustic parameters with
the way diffusion was modeled. They found that a sufficient portion of reflected energy
needed to be modeled as diffuse for predictive accuracy, and that diffusion coefficients
had to be defined in the frequency domain. They stated: Accounting for scattering in the
modeling of all reflections orders (since the first reflection) would improve accuracy in
the time domain and assist in the prediction of acoustic parameters such as reverberation
time. Allowing the directivity of scattered energy to be modeled would improve spatial
accuracy.

2.2.1.1. Statistical treatment of diffuse reflections. The scattering coefficient δ

Diffuse reflections, if treated at all, are usually treated in a crude manner 43. The major
problem lies in the difficulty of knowing the angle diffusion properties of a surface
(directivity pattern of the scattered sound). Quantification of these would require very
precise laboratory measurements, and the fact that every frequency has a different
diffuse behavior for every angle of incidence, would make the amount of data very
difficult to handle. Therefore, simulating the scattering effects of diffuse surfaces by
simplified methods has been accepted.

A first approximation in ray-tracing models is to assume four kind of reflections

combinations 43. Those are; S-S, S-D, D-S, D-D, where S represents ‘specular’ and D
‘diffuse’, see Figure 2.4. Due to the wavelength of sound, perfect specular reflections
might not exist, and in this way, the approximation D-S may fail.
 39

Figure 2.4: Four reflection combinations possible when both specular and diffuse
reflections are taken into account by the ray-tracing method . S-specular, D-diffuse.
After Dalenbäck43.

The use of the scattering coefficient is the most popular method to weight the relation
between specular and diffuse reflections in geometrical acoustics. This coefficient δ is
defined as the ratio between reflected sound power in non-specular directions and the
total reflected sound power 16, 51, Eq. (2.4) and Figure 2.5.

Wref . Non − specular

δ= (2.4)
Wref . Non − specular + Wref .specular

Figure 2.5: Energy components of reflected sound.

Therefore, to every surface two octave band dependent factors can be assigned, an
absorption coefficient α , and a scattering coefficient δ . The relationship of these factors
is illustrated graphically in Figure 2.5, and it is described by Eq. (2.5), being on it the
terms, absorbed, scattered, and specular.

α + δ (1 − α ) + (1 − δ )(1 − α ) = 1 (2.5)

The selection of δ is made according to the geometrical properties of every surface,

and somehow, it is just a statistical way to define diffusion. When a ray encounters a
diffusing surface, a random number in the range [0,1] is generated. If the number is
smaller than δ the ray direction is randomized to simulate a diffuse reflection,
 40

13, 62, 64
otherwise the reflection is specular . In principle, the stochastic nature of the
algorithm needs one pass per frequency band, since the diffusion factor of a surface is
62
frequency dependent . However, today’s knowledge about which scattering
coefficients are realistic and how to determine them is very limited, and some authors
consider it sufficient to characterize each surface by only one scattering coefficient,
1, 15, 16, 51, 65
constant for all frequencies . The author of this thesis thinks that this
assumption can perhaps be accepted if both causes of diffusion are of similar
importance, as is illustrated in Figure 2.6. A positive point of this approximation is that
the computation time will improve when only one computation pass is needed.

Figure 2.6: Scattering coefficient as a function of two factors. Roughness and finite size
of surfaces.

Keränen et al. 4, 66 carried out a study of noise control in six industrial spaces. Based on
the idea of a single scattering coefficient for all frequency bands, they applied the
following approximation for the selection of the scattering coefficient, Table 2.1, which
combines both finite size and roughness of surfaces. Other studies have suggested the
advantage of weighting diffusion coefficients at low frequencies according to surface
dimensions 12, 67.

Table 2.1: Suggestion for choosing scattering coefficient accounting for surface size
and roughness. After Keränen et al. 4.
Scattering coefficient, δ Description of the surface
0.1,…, 0.19 Large, plain surfaces
0.2,…, 0.39 Large partially fitted surfaces
0.4,…, 0.59 Small or fitted surfaces
0.6,…, 0.89 Large densely fitted surfaces
0.9,…, 1.00 Small densely fitted surfaces
 41

Another aspect of the scattering coefficient δ is that it not only depends on the
frequency, but also on the angle of sound incidence. However, analogue to the random-
incidence absorption coefficient obtained in reverberant rooms (see Appendix 1), and
angular average of the scattering coefficient, i.e. the random-incidence scattering
coefficient, can be defined as well. Vorländer and Mommertz proposed an experimental
method to obtain δ 65, 68
, which has been accepted, adopted and standardized by the
15
International Organization for Standardization in ISO/DIS 17497-1 . The first
published application of this new method has already been carried out by Jeon et al 69.
The experimental method to determine δ is also explained later in Appendix 1.
Different methods for measuring the scattering coefficient of a surface have been
proposed by Farina 70, and by Haan and Kwon 71.

Another approach was given in the modified scattered diffuse energy model (SU
model)16. Upon each reflection, the diffuse-reflection coefficient δ is used to define the
fraction of energy diffusely scattered into non-specular angles, while the remaining
energy is reflected specularly in the usual way. The diffuse energy is then assumed to
decay exponentially and contribution to the receivers is calculated. This method has an
important drawback, which is that the estimation of the decay constant for the diffuse
energy is estimated from Sabine or Eyring’s formula since the energy is assumed to be
diffuse, while it is possible for a room to have different decays throughout the space. A
72
different approach was suggested by Dalenbäck . This algorithm is based on
approximate cone tracing, where purely specular contributions are accounted for in a
primary pass. During the primary pass, the energy to be diffusely reflected is stored at
square wall patches and the energy to be specularly reflected continues. During the
secondary and subsequent passes, these patches in turn act as sources for diffusely
reflected sound, and energy that will be diffusely reflected is, once again, separately
stored at the patches. The problem is in principle recursive, but by using a finite number
of patches and exploring the properties of diffuse reflection, recursion is abolished. A
list is then assigned to every patch. When a primary ray encounters a patch, to that list is
added the arrival time and the diffuse energy fraction δ (1 − α ) . When a ray encounters a
receiver, the whole list of reflections associated with the patch, now acting as a diffuse
source, is added to the echogram. The algorithm contains some approximations that can
lead to errors. For instance, since the centre of the patch is used as secondary source
instead of the exact point of impact of the ray, the time origin of the diffuse reflection is
 42

somehow translated in time. The largest uncertainty will be for grazing incidence, but
this can be diminished for the most important surfaces by increasing the patch density.
Refinements are made, creating a higher number of rays for patches that are excited
earlier, and modeling the patch density dependent on the factor δ (1 − α ) (high density
for hard surfaces with strong diffusion). Dalenbäck established the long computation
times required by this model.

53, 62
Stephenson tries to avoid the explosion of computation time by applying the
72
radiosity method , where the energy flows are not only split but also re-unified. In
principle, the radiosity method is not consistent with ray tracing, as the coincidence
probability is very small, and the computation time will explode before recombination
occurs. The model traces pyramids, which can have many edges and can be clipped by
obstacles. A good point of the algorithm is the suggestion to model diffraction, where a
pyramid can be clipped by the edge of an obstacle into three new ones. Stephenson
admitted the difficulty of implementing the algorithm, recognized problems of
dynamical storage and access to pyramid data, and affirmed that the total algorithm was
yet to be tested.

A user-defined transition order, TO, has also been suggested, i.e. as in the commercial
software ODEON 2.5 73, 74. Reflections with lower orders than TO are considered purely
specular. For larger orders only the usual ray tracing is done with an evaluation of δ .
However, the concept of transition order is not physically satisfactory since diffuse
reflections should occur even at the very first reflection rather than suddenly being
switched on at the transition order 16, 43. Experimental studies do not support the use of
TO12.

Lam 16 compared the accuracy of three different computer models and their dependence
on δ in the prediction of the acoustical parameters of two rooms very different in shape
(cube and disproportionate), but composed of the same wall material. He found that the
three models need different values of δ to achieve convergence with the measured
values, and what is worst, that the selection of δ in all of them changed with the shape
of the room, even though they were made of the same type of material. That made him
conclude that there is much work still to be done on solving the relationship between the
physical scattering properties of real surfaces and the scattering coefficient used in
computer models. Moreover, as stated by Dalenbäck et al 43, a disturbing fact is that in
 43

some halls the choice of diffusion factors has a greater impact on the estimation of
acoustic parameters than the uncertainty in absorption factors.

A final possibility to model diffuse ray reflections would be to model the reflected rays
as having a certain dependence on the specular direction, i.e. forcing the direction of the
reflected ray to have a direction of ± 20° with the angle obtained by Snell’s law.

More information about different ways to account for diffusion in computer models can
be found in Dalenbäck et al 43, where a good overview is given.

2.2.1.2. Reflection models with secondary diffuse sources

A different way to look at a diffuse reflection process is to consider every point where
reflection occurs as a new secondary sound source, which emits the diffuse reflected
45, 62
sound in the form of new rays with some directional characteristics . However,
direct implementation of a split up ray algorithm would result in an exponential increase
in the calculation time, since even secondary rays, upon encountering a diffusing
surface have to be split up. Because the details of the sound become less audible with
time, it is perhaps not very useful to expend so much computational effort to follow an
exponentially increasing number of particles, even while the total energy remains
54, 61
constant . Instead, a secondary source can be created at each subsequent wall
reflection, which will radiate into a hemisphere inside the room as an elementary area
source. In this way, all visible receivers are irradiated with diffuse energy, but then the
energy is re-grouped into a ray and which is traced forward along a new direction, again
by creating a random number and comparing it with δ , to choose between specular or
43, 75
diffuse reflection. This is called the Naylor method . With this approach a high
density of reflections at a receiver is achieved without the need of tracing a very high
number of rays. The rays are only losing energy due to wall and air absorption, and their
reflected directions are randomized if the walls encountered are diffuse. Both the high
reflection density and the spatial information make the approach feasible for
auralization.

In an ideal diffuse reflection the sound is reflected in all directions over the hemisphere,
and all directions are equally probable, see Eq. (2.6). Integrating this equation into
spherical coordinates over the hemisphere we obtain 2π , this being therefore the
influence at any point a distance r away from that secondary source 1 / 2πr 2 .
 44

Ir
=1 (2.6)
I r , max

Many room acoustics prediction models consider the diffusion of sound to follow
13, 72
Lambert’s cosine law . Lambert’s law for the diffuse reflection states that the
directional distribution of the reflected intensity is proportional to cosθ r , Eq. (2.7).

Ir
= cos θ r (2.7)
I r , max

Integrating now Eq. (2.7) over the hemisphere we obtain π , being therefore the
influence at any point a distance r away form the secondary source cos θ r / πr 2 .
Although Lambert’s cosine law has been used in many room acoustics prediction
methods, its use has been debated since it originates from the reflection of light. On the
72
other hand, Dalenbäck considers his proposed diffuse model to be more efficient if
exit angles are independent of incidence angles as in the Lambert model. Some authors
no longer consider Lambert's law a valid approximation 76, 77.

76
Recently, Rindel has proposed a new approximation to model the directional
characteristics of sound reflections, which takes into account the wavelength and the
finite area of a rectangular reflecting area, and is valid for all angles of incidence
including the grazing 0° . The model is based on sound transmission theory and the use
of Babinet’s principle, which states that the reflection from a plane rigid surface is
equivalent to the transmission through an opening of the same geometrical shape
surrounded by an infinite rigid baffle. In Figure 2.7, the size of the surfaces is 2a ⋅ 2b ,
and the direction of the incident sound is defined by the angles α 0 and β 0 relative to
the x- and y-axis. In a similar way, the direction towards the receiver point is defined by
the angles α and β . The wavenumber of sound is k = 2πf / c [m-1], (f is frequency and
c the speed of sound). Using the result for sound transmission through a rectangular
opening 78, the intensity of the reflected sound relative to the maximum value is:

2
Ir ⎛ sin X sin Y ⎞
=⎜ ⋅ ⎟ (2.8)
I r , max ⎝ X Y ⎠
 45

where
X = ka (cos α − cos α 0 ) (2.9)

Y = kb(cos β − cos β 0 )

Figure 2.7: Definition of angles of incidence and reflection from a rectangular surface.
After Rindel 76.

The radiation function (sin X / X ) is valid for a single frequency, but averaging to the
2

frequency band, it is found that the centre frequency is not a bad approximation. Still it
has some drawbacks, e.g. X > 2π and X = 0. A mathematical approximation to the
octave band average, which does not have the same drawbacks was also suggested by
Rindel.

2
⎛ sin X ⎞
≈ (cosh(0.67 ⋅ X ))
−2
⎜ ⎟ (2.10)
⎝ X ⎠

The suggested approximate directivity function for different values of the relation ka is
plotted for an angle of incidence of 90° in Figure 2.8a, and for an angle of incidence of
30° in Figure 2.8b. In addition, Lambert’s law of diffuse reflection has been added for
comparison. It is observed that the reflections have high directivity in the direction of
the specular reflection for the high ka values, whereas the reflections are radiated with a
nearly uniform directivity for the low ka values. The Lambert model is not particularly
close to any of the examples.
 46

Figure 2.8: Directivity functions. a) for the angle of incidence α 0 = 90° . b) for the angle of

incidence α 0 = 30° . After Rindel 76.

2.2.2. Modeling diffraction in the ray-tracing method

Most of room acoustic modeling algorithms rely on geometrical acoustic techniques
which completely neglect the diffraction effects. This is because such methods assume
that sound propagates as light, i.e. that the wavelength of sound is much shorter than the
dimensions of the reflecting objects 17. In section 1.3.1, we introduced the dependence
of the wavelength of sound on diffraction effects, i.e. how low frequencies bend at the
edge of the barrier. As barriers are frequently used in industrial environments in order to
protect workers from noise exposure 20, diffraction effects should be correctly taken into
account in order to avoid an over-emphasizing of the barrier effects. Also diffraction
12
appears in auditoriums, especially in seating areas behind balcony fronts , or as the
79, 80, 81, 82
cause of the seat-dip effect . However, as we discuss below, diffraction effects
are very poorly considered by geometrical acoustics models.

According to the Geometrical Theory of Diffraction (GTD) proposed by Keller in 1962

83
, an acoustic field, incident upon an edge between two non-coplanar surfaces, i.e. a
wedge, forms a diffracted wave that propagates from the wedge in a cone-shaped
pattern, see Figure 2.9.
 47

Figure 2.9: Diffraction by a wedge.

Many methods, which are not discussed in this thesis, have been proposed to account
for the effect of barriers in a free field. Such methods are not applicable to room
acoustics, as also the sound energy reflected from boundaries has to be considered, i.e.
the sound reflected from the floor might also suffer diffraction, see Figure 2.10, and the
sound reflected from the ceiling obviously decreases the performance of the barrier 84.

Figure 2.10: Diffraction paths in a barrier geometry. The direct sound path is obstructed, and
the sound bends at the edge of the barrier.

85
To simplify this problem, Kurze suggested that the barrier attenuation due to
diffraction is exclusively effective in the case of direct sound. For reflected sound, the
barrier is just another obstacle in the room that causes diffuse reflections. Other methods
have been proposed to calculate only the effect of the diffracted beam in the shadow
region, with the discontinuities in the sound field considered not perceivable 86.

Haaren et al 87 validated a ray-tracing model applied in the modeling of noise barriers.

Their study can not be considered valid as the validation was done in open field, and
only at six points 25 meters away from the barrier. They did not explain how their
 48

algorithm works, nor how diffraction was modeled. On the other hand, Svensson et al 88
proposed an analytical method to obtain the impulse response behind a barrier. This
method applies the concept of secondary edge sources (every point at the edge of the
barrier is considered a sound emitter) and allows the derivation of directivity functions
for such edge sources (re-radiated wavefronts spread in all directions with different
amplitudes). The applicability of this method relies on the fact that the impulse response
inside an enclosure can be written as a sum of geometrical acoustics, hGA, and
diffraction components hdiffr. This is good news, but the theoretical development of the
method is not easy to follow and an application of it will require a deeper future study.

Still, known diffraction formulae assume that rays hit edges exactly, whereas sound rays
never do so 53. This problem could be solved as suggested by Dance 20, who, based on
an idea of Benedetto and Spagnolo 84, which at the same time was derived from Fresnel
and Kirchoff theory 53, proposed three versions of barrier models to apply in ray-tracing
methods, and compared predictions of each with measured values. According to the
published results, only two of them are interesting. These are: An imaginary plane
follows the perimeter of the barrier at a distance λ , where λ is the wavelength of the
mid-frequency of the octave band of interest. If a sound ray strikes that ‘diffraction
area’ it is randomly redirected propagating along its new path with no attenuation. The
second possibility (ten times slower in computation time) is to create a secondary source
at the incident point, which is assumed to be omni-directional and to contain all the
energy of the incident ray with no attenuation loss. Stephenson has also reviewed these
53
approaches . He said that staying with one particle (first approach) and taking the
directional distribution as a probability distribution would not save computation time to
the same extent as more sound rays had to be emitted from the source, requiring even
20
higher computational efforts. Continuing with the experience of Dance , the results
were better for the first approach, with good accuracy at low frequencies, although at 1
kHz, some deviations were observed and attributed to interference effects. Both models
failed to predict the negative IL in front of the barrier, and converged with the measured
IL at the furthest points, as the barrier effects becomes negligible in the reverberant
sound field. The first approach is considered interesting and should one day be
implemented by the author of this thesis, together with others, and compared with field
data.
 49

2.3. Considerations about the ray-tracing algorithm

Every model has a certain number of assumptions which might limit its applicability. If
the theoretical assumptions do not hold in the case of a particular room for which
26
predictions are to be made, the predictions might not be accurate . In any case,
16, 43
assumptions which are fulfilled in some space may fail in others , and here is where
validations of acoustical models should be checked carefully.

The ray-tracing algorithm has been shown to predict noise levels with good accuracy 32,
and it is considered one of the most elegant methods to represent diffuse reflections.
However, its limitations, restrictions and inherent problems are not few. In this section
we discuss and introduce some topics that should be considered when implementing or
evaluating the algorithm.

2.3.1. Frequency range

The wavelength or the frequency of the sound is not inherent in the ray-tracing model 51.
Normally, the only frequency-dependent factors that can be included are the absorption
coefficient and the statistical estimation of diffuse properties. Already in their
pioneering work Krokstadt et al 11 thought that the most severe restriction of the method
is the limited frequency range over which the results are valid. It is an underlying
assumption in all methods using sound rays that the wavelength corresponding to the
lowest frequency of the sound is small compared with the dimensions of the room and
its surfaces. This assumption, i.e. to neglect the undulatory nature of sound, is the
26, 32, 49, 50
reason why experimental studies based on ray tracing have been shown to
predict noise levels and acoustic parameters with good accuracy at high frequencies,
while not so good at low ones.

2.3.2. Negation of wave nature, interference and diffraction

Therefore, the wave nature of sound is not really considered in the algorithm, and
important effects like interference and diffraction are poorly represented or simply
ignored.

For instance, as we saw in section 1.3.2, in room acoustics there are two kinds of
interference effects that should be considered. These are interferences close to the
reflecting boundaries, and interferences between highly reflecting parallel walls. The
 50

modeling of those could be done only if the sound is considered as what it is; a pressure
wave which changes its amplitude and phase at every reflection. But then we would
need to use pressure reflection factors allowing a phase shift at reflection 19, and these
can only be obtained for certain materials in very ideal or controlled cases, but not in
real-life spaces. In this way when several sound field components are superimposed,
their mutual phase relations are not taken into account; instead, their energy densities
are simply added 13. However, in many cases, e.g. in industrial spaces, the interference
effects are negated by the fittings and bodies in the space and the large number of non-
coherent sound sources 20.

On the other hand diffraction phenomena are neglected in geometrical room acoustics
since the propagation in straight lines is its main postulate. In this way, we ignore the
effect of diffusion due to the finite size of surfaces, with the consequence of obtaining
stronger reflections by geometrical laws than those that occur in reality.

There is also one topic which is rarely found in the literature, namely, the seat-dip effect
81, 82
, i.e. the selective attenuation of low frequencies for sound passing over an audience
area at near grazing incidence. This effect, in which multiple diffractions and
interferences play a role, appears in the lower frequencies (~ 118 - 178 Hz), and
produces a variable attenuation of even 30 dB in magnitude 79, 80.

Due to the wave nature of sound, it would be very important for the quality of the
simulation model that diffusion and diffraction effects be included in a proper way, in
order to achieve more reliable computer models and, in consequence, better results. This
would extend the validity of ray tracing to the lower frequencies, where the usually
assumed ‘optical limiting case’ is no longer valid and the wavelengths are no longer
small compared to the room dimensions.

2.3.3. Angular distribution of reflected sound

Generally speaking, the most problematic matter in the geometrical acoustics methods is
how to model the angular distribution of reflected sound. It seems that only few options
are available, see section 2.2.1, be they specular, diffuse, or a combination of both.
However, we should wonder what 'diffuse reflection' really means or what it looks like.
Certainly, a lot of experimental work has already been done 69, but to measure and save
 51

the data of all existing construction materials has proved to be also a very laborious
task.

The existing literature 12, 49, 50 tells that models based only on specular reflections lead to
large errors and to too deep energy decays. Also, that they over predict the reverberation
time, especially in enclosed spaces where the absorption is concentrated on one surface,
or where the shape is highly disproportionate, such as large factories 16. On the contrary,
approaches that model diffusion by random distribution improve accuracy, but certainly
are far from correctly representing the sound distribution of sound after reflection from
uneven or finite surfaces.

2.3.4. Geometrical description

When modeling a certain enclosure, it is important to decide how precise the description
of the enclosures should be. This certainly affects accuracy but also computation time. It
has been said that a lot of acoustic experience is necessary to know how detailed a
50
geometrical model must be to provide a basis sufficient for the calculations . One
31
study suggested that to obtain a reasonable degree of precision, it is necessary to
49
avoid modeling too small surfaces. Another work that compared the efficiency of
different computer models concluded that better results were obtained with simple
geometrical descriptions. It a way it is true that the probability of an image of being
visible from the receiver decreases with the size of the surfaces involved in its
51
generation , and anyway, what is the point of representing every small detail of the
room when later the reflections are modeled by creating random numbers. However and
67
the other hand, Lisa in his recent work , which studied the acoustic prediction in
ancient open-air theatres, observed that the more complete the geometrical description,
the better the results obtained. In this kind of enclosure, where no roof or ceiling is
present, this has been previously observed 89.

2.3.5. Frequency dependent factors. The absorption and the scattering

coefficients
In section 1.7.2, I have introduced the absorption coefficient α , which describes the
absorption properties of materials and mentioned that it is an angle-dependent factor. In
section 2.2.1.1, I did the same with the scattering coefficient δ , which is used to
describe the diffusing capabilities of surfaces. The experimental methods used to
 52

measure random sound incidence in the laboratory as described in ISO are introduced in
Appendix 1. At this point, we mention that the main reason for avoiding angle-
dependent measurements is that they would need to be very precise, and thus to handle
such a large data base of all typical construction materials becomes very difficult.
Therefore, estimating the absorption characteristics of a surface by a form that averages
over all angles of incidence has been accepted90.

Sometimes it is not possible to measure anyone of those coefficients, and then their
16 12
estimation becomes a guess , which is a recognized source of inaccuracies . The
quality of a computer simulation is then strongly dependent then on the user’s feeling
about the absorption and diffusing properties of the modeled surfaces 50.

16
Still, as appointed out in one study differences up to 5 % or even 10 % can be
expected between the absorption coefficients of a certain material measured in a
reverberation chamber and the same material placed in the walls of an auditorium.

2.3.6. Number of rays

Uncertainties are also introduced in the results by the use of a limited number of
discrete rays to represent the sound field 31. As the angle between adjacent rays remains
roughly constant, and since all rays are radiated from a point source, this representation
gradually becomes less exact with increasing ray length. To achieve convergence
criteria (after all, this is a Monte Carlo method) the number of rays needs to be large
(even 100,000 in a large auditorium 47). The larger the number of rays, the smaller the
solid angle that an individual ray will cover. Again, this will affect the computation
time, but a too low number of rays can cause an insufficient number of detected
reflections 91.

2.3.7. The detection problem

Three kinds of receivers can be used in room acoustics simulation, point, plane and
volume, but one basic condition: The traced rays must meet a detection condition at the
receiver's position in order to contribute to the impulse response of the room at that
point 91. The probability of a ray being detected is a function of the size of the detector,
and point receivers are not adequate in ray tracing. An infinitesimally thin line
53
arbitrarily emitted by a source point almost never hits a receiver point . Plane
reception cells are also inadequate, as rays reaching the reception plane at grazing
 53

incidence can not be handled properly 9. Spherical receivers allow an omnidirectional

sensitivity pattern, as the cross-section of a sphere is a circle of constant size for all
10, 91
directions of incidence . Spatially anisotropic shapes, like cube receivers, show
91 9
more complex directional behavior . Ondet and Barbry considered that spherical
receivers are more difficult to implement and require longer computation time than
plane detectors. But this is not true, as a sphere is geometrically characterized only by
its center and its size (ratio), while a plane needs at least the coordinates of three points,
and the detection problem is surely more complex. In section 3.3.4, how to implement
the detection of sound rays by spherical receivers will be covered.

Lehnert 91 studied the systematic errors of the ray-tracing algorithm due to the detection
problem, concluding that one of them was due to the finite size of the receivers
(independently of their shape). In order to achieve a constant density of rays across the
solid angle of the receivers, he derived a formula, Eq. (2.11), to obtain the size of the
receivers r, for a certain number of rays N and the maximum length of a ray, lmax (which
is the largest path a ray can travel within the room). However, large detectors can cause
more errors as they may detect some rays too far away from the point of interest.

r = l max (2π / N )
1/ 2
(2.11)

Another possibility that has been suggested is Eq. (2.10), where VR and V are the
volume of the receiver and of the enclosure, respectively 92:

1/ 3
⎛ 15V ⎞
V R = 10V / N so r = ⎜ ⎟ (2.12)
⎝ 2πN ⎠

91
Lehnert also called attention to the fact that due to the finite size of the detectors,
sound paths may appear which are invalid in terms of the geometrical model, see Figure
2.11. Some of them (first and second) can be interpreted as rough models of diffraction
and diffuse reflections, and can as such be tolerated. Others (third) are definitely
intolerable.
 54

Figure 2.11: Examples of invalid detections; a) The sound source is shaded. b) The
reflection point is not located within the wall. c) The receiver is on the back side of the
wall.

2.3.8. Other problems

Ray-tracing programs execute in ‘batch’ mode, taking seconds or minutes to update new
locations, and therefore are not suitable for real-time auralization. Nor can they scale to
support large 3D environments, and only do an inspection of certain predefined
propagation paths, as a function of the established sound source and receiver locations
42
.

2.4. Round Robin

As we have seen throughout this chapter, many approaches have been suggested to
model room acoustics, and therefore many commercial softwares have also appeared in
the international market. In Appendix 2 a review of software is given. Those programs
are in use today for the purpose of research, training and consulting. Different
approaches may lead to different results, and then it is necessary to know, which ones
are accurate enough to be accepted as able to represent the sound distribution in
enclosed spaces.

The PTB institute published the results of the first international round robin on room
49
acoustical computer simulations in 1995 . Not only could the accuracy of different
models be checked, but also the ability of researchers, acousticians and/or architects to
make use of them. Sixteen participants from 7 countries, and 14 different programs for
room acoustical computer simulations were compared concerning the results of 8
different acoustical criteria (RT, EDT, D, C, TS, G, LF and LFC) in a speech
 55

auditorium. Two source positions and 5 receiver positions were chosen and the criteria
were calculated only for the 1 kHz octave band.

The project was run in two steps: first the room geometry and the material data provided
on the basis of construction plans and material descriptions in words. No information on
the measured results was given. In the second step, an estimation of the absorption and
diffuse properties was given to the participants. The results of the first phase showed a
surprisingly large scatter with a strong tendency to underestimate the absorption
coefficients, and thus to overestimate the reverberation time. In the second phase, the
agreement was, of course, better, but even with harmonized input data some results
varied considerably from others. Only three of the softwares were considered accurate
enough. One interesting point for further investigation was the increasing difference
between the results from simulations and measurements at increasing source-receiver
distances. The reason could be that in none of the programs is the attenuation of sound
waves at grazing incidence over seat rows or the audience (seat-dip effect) taken into
account. All programs neglected the wave nature of sound. Programs which modeled
only specular reflections were the most outlying. The three ‘winners’ were: The image
source method (5th order) plus random tail (35 minutes in processor 486), the hybrid
model plus random tail (4 hours in processor 386), and ray tracing (200,000 rays) (140
minutes in processor 486). It is curious to note that these three programs did not make a
very detailed description of the room, and nor did they need very long computation
times, compared with the other participants.

In 1996-98, the PTB organized another international ‘round robin 2’ with 16

50
participants from 9 different countries and using 13 different computer programs .
This time the object under examination was a concert hall in Jönköping/Sweden named
the ELMIA hall, which has an approximate volume of 11,000 m3 and 1,100 seats. The
results of this round robin were much better than those of ‘round robin 1’ but the
complex design of this hall generated many problems. The ground plan of the hall with
the source and receiver locations is given in Figure 2.13, also a photo of it.
 56

Figure 2.13: a) Ground plan of the Elmia hall with source and receiver locations. b)
Photo of the Elmia hall.

Again, the round robin was done in two phases. In the first one, only a number of
photos, drawings, and a description of the surface materials and the climatic conditions
were given to the participants. Thus, not only the quality of the simulation software is
relevant, also the skill of the user as an acoustician. The amount of results data is
extremely high, and there is not room for it in this thesis, although it can be obtained
from the net 94. As expected, the weak points of the programs are concentrated on the
low frequency calculations as diffraction effects have been neglected in the geometrical
calculations. In the second phase, the estimated absorption and diffusion were given by
the coordinator of the study, so some real differences in the programs should be
revealed. For instance, even with the same kind of computer processors, the
computation times varied from 42 seconds to 30 hours. In Ref. 94, all the information
about the hall geometry and the measured results can be found. This will allow us in the
nearest future to know more about the accuracy of our model.
 57

3. Implementation of the ray-tracing algorithm

In this chapter we present the actual and complete form of our ray-tracing algorithm.
From all the methods presented in the previous chapter, the ray-tracing method has been
selected not only because it has been shown to predict acoustics with considerably good
accuracy 32, but also because its concept is feasible and it has a simple implementation.
Some author may claim that his/her proposed way is the best among others, or that
he/she already did that already twenty years ago. That is good news, although it has
been noticed that normally more time is spent on describing the virtues and accuracy of
an algorithm than on the algorithm itself. It was not the aim of this thesis to present an
ultimate implementation of the ray-tracing algorithm, nor to state that it is the best
method among all the approaches. The propagation of sound may be well defined, but
not its representation in room acoustics. Every method has a certain degree of
limitations and accuracy, and we are content to comment on the method we chose. It has
43
been also noticed that it seems tempting to extend a method that one already knows
rather than to incorporate a completely new one (or even to rewrite the complete
algorithm). So far that is not our case, and we expect that many changes will be made in
the future to this research tool. However, the time spent on the development of the
model has been very illustrative, and it is time to write down some impressions.

The ray-tracing algorithm is an application of geometrical acoustics. The sound power

emitted by the sound source is described by a finite number of rays, which are
considered as carriers of energy. These rays travel through the air at the speed of sound
and are reflected from room boundaries, until their energy, due to the air and wall
absorption, has been reduced below a certain limit. During this time, the rays might
cross predefined volumes, the receivers, where an energy calculation process is
performed. These data build up the energy response, from which desired acoustic
24
parameters can be obtained . Geometry laws are used to follow the paths of all the
rays, and therefore they must be repeated a considerably large number of times, thus
causing great computation efforts. To achieve competitive computation times, all the
operations involved in the inner loops should be as efficient as possible. A deeper
description of the ray-tracing algorithm has already been given in section 2.1.3.1.

First, we will look at the schematic flowchart of the model. Then, the algorithm
implementation is described. This will be done in two parts, a geometrical and an
energetic part, because they are the two main groups of operations involved. In the first
 58

part, a description of the enclosure from the geometrical point of view is given, e.g. the
surfaces or locations of the sound source and receivers. Also the way the sound
distributes, that is, how the rays interact with the surfaces, i.e. the point-in-polygon test,
the visibility test, or the way collisions are modeled, i.e. specularly or diffusely. In the
second part, the instant energy of every ray is described as a function of the
characteristics of the sound source and of their interaction with the medium, i.e. air and
walls. Also how those rays ‘impinge’ energy to the receivers and how the energy
response and the desired acoustic parameters are obtained. Comments will also be made
on which operations are performed in a logarithmic scale while using sound levels or in
linear scales.

3.1. Flowchart of the implemented ray-tracing algorithm

First, all the input parameters have to be introduced. These are the geometry and
acoustic properties of the enclosure, and the locations and characteristics of the sound
source and receivers. Then, one by one, create new ray with a direction and an initial
power. We trace the ray from the sound source or from the previous collision point, as
long as its power Wray is larger than a certain limit Wlimit. Then, we have to check all
surfaces, and by applying the visibility test we find out which of them all is collided.
Then, we check receivers to see if the ray crosses any along its actual path. If that
happens, the energy that the ray ‘impinges’ to it is calculated, therefore calculate Erec(t).
Then, the sound power of the ray is attenuated according to the air and wall absorption,
calculate absorption, to finally model reflection by a diffuse or specular reflection.
When all rays have been shot, the impulse response of the receivers is obtained, and
from it the calculation of acoustic parameters, SPLRec & RTRec. It should be noticed that
the repetition of the process for every octave band has not been represented in the
flowchart.
 59

Figure 3.1: Flowchart of the implemented ray-tracing algorithm

3.2. Geometrical calculations

All the operations and mathematical steps which are related to the geometrical part of
the model are described in the following subsections.

3.2.1. Geometrical description of the enclosure

Cartesian coordinates are used. A point in the space is then described by its coordinates
x, y, z, a line by its canonical form, Eq. (3.1), and a plane by its general equation, Eq.
(3.2).

x − x1 y − y1 z − z1
= = (3.1)
l m n

Ax + By + Cz + D = 0 (3.2)

The canonical form of a line has all the information that we need. It contains a point,
(x1, y1, z1) (departure point of the ray) and a vector, (l, m, n) (direction of the ray). At
every collision with a surface the direction of the ray has to change, also the departure
point, which will be the previously calculated collision point at that surface. How to find
the collision point of a ray with a surface is discussed in the next section. But first we
 60

have to notice that a plane extends to infinite, while a surface does not. A surface then
has to be described by a plane and by its corners, or by the segments that define its
r
edges. The normal vector N ( A, B, C ) is orthogonal to the plane, and if D = 0 the plane
lies over the origin of coordinates. In Figure 3.2a, a plane obtained from three points
(P1, P2 and P3) and its normal are represented. All the equations in the following use
the same nomenclature, even though the given subscripts.

Figure 3.2: a) A plane and its normal obtained from three points. b) Collision point P
between a line r and a plane A.

3.2.2. Shooting rays

10
In principle, the way sound rays are emitted can be either randomized or predefined
as a function of the directional characteristics of the sound source. Normally, the sound
sources used in acoustic measurements have omnidirectional characteristics, which is
our case. Therefore, it must be ensured that the generation is almost uniform on the
surface of a spherical source 62. The simple assumption of three random generators for
the three components of the vector direction of the ray is not completely correct, as that
produces a “cube of rays” instead of a sphere. It is possible to ‘cut away’ the corners of
the cube discarding each vector with a modulus greater than one. Instead, we have
applied a semi-probabilistic generator, in which the sphere surface is mathematically
divided into a large number of equal areas (actually 400=20x20)62. This is done by
creating two random numbers, rnd1 and rnd2 in the range [0, 1], and projecting their
values over the sphere to obtain the components of the ray, as mathematically expressed
in Eq. (3.3). The result is equivalent to cutting the sphere with 20 iso-z planes, equally
spaced along the z-axis, and then dividing each circle again into 20 parts, as shown in
Figure 3.3. Although the single facets have a very different shape, all of them have the
 61

same area. In this way, with just two random numbers, rnd1 and rnd2, the directions of
400 rays can be created, each of them discretely shot through one of these sub-areas.

2
i + rnd 1 ⎛ i + rnd 1 ⎞ ⎛ j + rnd 2 ⎞
l=2 −⎜ ⎟ cos⎜ 2π ⎟ (3.3)
20 ⎝ 20 ⎠ ⎝ 20 ⎠
2
i + rnd 1 ⎛ i + rnd 1 ⎞ ⎛ j + rnd 2 ⎞
m=2 −⎜ ⎟ sin⎜ 2π ⎟
20 ⎝ 20 ⎠ ⎝ 20 ⎠

i + rnd 1
n =1− 2
20
for i = 0…19, and j = 0…19, with rnd1∈ [0,1] and rnd 2 ∈ [0,1]

Figure 3.3: Subdivision of the omnidirectional properties of the sound source in facets
of equal area. After Farina 62.

3.2.3. Detection of collisions

Once the ray has been shot in a certain direction, we have to find out the one and only
surface which will be collided with. After that, the ray is reflected and again the same
calculation process is performed. How to detect the true reflection points is described in
this section.

3.2.3.1. Point-in-polygon test

Mathematically, the collision point xP, yP, zP, between a line, Eq. (3.1), and a plane, Eq.
(3.2), is calculated by Eq. (3.4).

⎡ x P ⎤ ⎡ x1 ⎤ ⎡l ⎤
⎢ y ⎥ = ⎢ y ⎥ − x1 A + y 1 B + z 1 C + D ⎢ m ⎥ (3.4)
⎢ P ⎥ ⎢ 1⎥ lA + mB + nC ⎢ ⎥
⎢⎣ z P ⎥⎦ ⎢⎣ z1 ⎥⎦ ⎢⎣ n ⎥⎦
 62

Because every surface is bounded, the next question to answer is whether the collision
point is inside or outside the geometrical perimeter of the surface. This is called the
point-in-polygon test. In principle, this test has to go over every surface after every ray
reflection, so it should be meticulously implemented as it is repeated many times in the
inner loop.

One definition of whether a point is inside a polygon is the Jordan curve theorem.
Essentially, this says that the point is inside the polygon if, for any line that we trace
from it, there is an odd number of crossings of the line within the polygon’s edges, see
Figure 3.4a. This definition also solves the problem for areas that are inside the
polygon.

Figure 3.4: a) Jordan Curve theorem. b) Resolution of Jordan theorem in the plane oxy.

On the other hand, the polygon and the sound ray are naturally defined in three
dimensions, and this can lead to a problem that is now discussed. The problem is related
to the floating point precision of the used programming language, i.e. with the
determination of the decimal digits. As every rational number is rounded, results can
appear which differ from the analytical results. For instance, when applying the Jordan
curve theorem to perform the polygon test, a line is traced from the collision point
towards one of the polygon edges. This line and the collision point should lie over the
surface, but perhaps no longer do because of the rounding. In this way, the computer
program fails when it just lightly considers that two lines are not in the same plane and
therefore never intersecting with each other. All this leads to likings, sound rays that
never hit a surface and escape from the room.

To avoid this problem and also to simplify computation, it is worthwhile to project the
polygon and the test point into two dimensions. One way to do this is to simply ignore
one component. The best component to ignore is usually that which, when ignored,
 63

gives the largest area polygon. This is easily done by taking the absolute value of each
component of the polygon plane’s normal and finding the largest. The corresponding
coordinates of the polygon and test point are then ignored. If the polygon is not parallel
to the coordinate planes, oxy, oyz, oxz, in practice, it does not matter which component
is ignored. Precomputing this information once for a polygon uses more memory but
increases the speed of the intersection test itself.

For instance, when we project the line described in Eq. (3.1) over the plane oxy, the z
components disappear, see Figure 3.4b. Then we can express that line in a more
convenient way, as in Eq. (3.5). Note: Do not confuse with Eq. (3.2).

Ax + By + C = 0 (3.5)

The coordinates (x0, y0) of the intersection point between two lines in 2D are obtained
by solving their equations simultaneously. If these lines are the line that we trace from
the collision point to perform the Jordan theorem and the line defined in one of the
edges, which are given by A1 x + B1 y + C1 = 0 and A2 + B2 + C 2 = 0 , then the
intersection point is

B1 C1 C1 A1
B2 C2 C2 A2
x0 = and y0 = (3.6)
A1 B1 A1 B1
A2 B2 A2 B2

A1 B1
and if = 0 , the lines are parallel.
A2 B2

We should repeat this process with all the edges of the polygon and count how many are
crossed within the corner limits to complete the point-in-polygon test.

3.2.3.2. Visibility test

It has been explained how to obtain the collision point between a line and a plane, and
then how to know if the point is inside the surface’s polygon or not. That is actually the
second of the three Boolean functions that compose the visibility test. However, the
remaining two are even simpler. The three functions are represented in Figure 3.5,
where T represents true and F false. The first ensures that the collided surface is in
 64

accordance with the direction of the ray and not in the opposite direction, see Figure
3.5a. The second has already been explained, and the last selects from all still possible
solutions only the closest, see Figure 3.5c.

Figure 3.5: The three steps of the visibility test. a) Right direction of the ray. b) Point-
in-polygon test. c) Only the closest of all the possible collided surfaces. T represents
True and F represents False.

3.2.4. Modeling reflections

As we saw in chapter 2, the reflection of sound from real surfaces does not generally
follow Snell’s law. This means that the reflections are never perfectly specular, instead
they are partially diffuse. This is a consequence of the natural roughness of construction
materials and of the finite size of surfaces. Also we said in section 2.2.1.1 that the most
popular method for modeling reflections in the ray-tracing algorithm is the use of the
scattering coefficient δ , and this is the method we use. When a ray encounters a
diffusing surface, a random number in the range [0, 1] is generated. If the number is
smaller than δ the ray direction is randomized to simulate diffusion, otherwise the
13, 72
reflection is specular . The problem is; how well will the real situation be
reproduced by just a normal randomization, Figure 3.6a, or by a weighted
randomization closer to a specular reflection, Figure 3.6b. We have simply chosen the
first possibility where all directions are equally probable in the case that reflection is
diffuse.

Figure 3.6: a) Diffuse reflection by normal randomization. b) Diffuse reflection by

weighted randomization.
 65

Just to illustrate the consequences of the applied method we have presented in Figure
3.7 the possible path followed by a ray. The energy of the ray has been represented by
the thickness of the line, which decreases at every reflection, jn. The first three
reflections, j1, j2, j3 are specular, while the others are diffuse.

Figure 3.7: Representation of specular and diffuse reflections in a 2D enclosure.

3.2.4.1. Specular reflection

The three directions of the new vector direction of a ray after a specular reflection from
a surface are calculated from:

⎡ l´ ⎤ ⎡ l ⎤ ⎡ A⎤
⎢m´⎥ = ⎢m ⎥ − 2 lA + mB + nC ⎢B⎥ (3.7)
⎢ ⎥ ⎢ ⎥ A2 + B 2 + C 2 ⎢ ⎥
⎢⎣ n´ ⎥⎦ ⎢⎣ n ⎥⎦ ⎢⎣C ⎥⎦

3.2.4.2. Diffuse reflection

To model a diffuse reflection, some alternatives are available. One way is to make use
of Eq. (3.7), and check what components have changed their sign. The sign of those
ones should be kept, but the values could be randomized. This is what we have done,
although another way would be to also make use of Eq. (3.7), and then introduce some
randomness to every one of the components, i.e. by creating random numbers in the
interval [-1, 1], dividing them e.g. by 20, and add these values to the components of the
vector (this method is closer to what was illustrated in Figure 3.6b). There are some
other possibilities, however, a simple randomization of the vector components can lead
to obvious errors, and it is important to ensure which components have changed their
sign upon reflection. One last thing to be mentioned is that if the scattering coefficient is
constant for all the frequencies, the tracing rays process can be done in a single pass.
However, the absorption of sound is not constant along the frequency range.
 66

3.2.5. The image method

If, for the early reflections the image method is desired, its implementation is still
simple. Its results can be superimposed on those from ray tracing in order to see
similarities or differences between both in the early reflections. Having a point in the
space, P(x, y, z), its image, PIm(x’, y’, z’), is calculated by Eq. (3.8), which translates the
coordinates of the point to the other side of the plane. We remember that the concepts of
the image method were introduced in section 2.1.3.2, as well as the backwards visibility
test, to find out whether an image is real. Second and higher order images, e.g. S’c2-w in
Figure 2.2, are obtained by successive applications of Eq. (3.8).

⎡ x´⎤ ⎡ x ⎤ ⎡ A⎤
⎢ y´⎥ = ⎢ y ⎥ − 2 xA + yB + zC + D ⎢ B ⎥ (3.8)
⎢ ⎥ ⎢ ⎥ A2 + B 2 + C 2 ⎢ ⎥
⎣⎢ z´ ⎦⎥ ⎣⎢ z ⎦⎥ ⎣⎢C ⎦⎥

3.3. Energy calculations

Once the implementation of the geometrical part has been discussed, it is time to
introduce the part that includes the energy calculations. Here we notice already that
instead of energy, we could talk about power or intensity, but not about levels as they
are a logarithmic and not a linear scale, as is discussed in the following.

3.3.1 The instant power of every ray

In section 1.6 we introduced the concept of sound levels. But the problem in the
operations that involve logarithmic scales is that they are more time-consuming.
Certainly all the calculations could be performed in decibels, but to add two levels
requires more computation effort than adding two powers. Therefore, it is wise to just
convert the sound power level of the sound source into power, then perform all the
operations in a linear scale, and finally change back to decibels if needed. For instance,
if we know the sound power level of the sound source for a certain frequency band, we
can obtain the corresponding power using Eq. (1.14), and obtaining Eq. (3.9), where W0
= 1 pW.

W = W0 10 0.1Lw (3.9)
 67

Once we have the sound source described by its power and not by its sound power level,
we can assign to every ray its initial power just by dividing the total of the sound source
by the number of rays, N.

Avoiding all explanations that have already been given, the i’th ray that arrives at a
receiver carries with it a certain power Wi , which will be its initial value minus the part
that has been absorbed in all previous paths, Eq. (3.10) 92.

e ∏ (1 − α j )
W0 Q −mL J
Wi = (3.10)
N j =1

where W0 is the power of the sound source, Q is the directivity factor of the sound
source (if omnidirectional, Q = 1), N is the total number of rays, and m is the air
absorption coefficient, L is the total distance traveled by the ray from the sound source
to the receiver, J is the number of reflections so far involved in the path, and α j the

absorption coefficient of the wall on which the j’th reflection has occurred.

3.3.2. Reception of sound rays

Suppose the energy contributed by the ith ray is Ei , and its sound power is Wi , we can
get the following formula:

E i = Wi Ti (3.11)

where Ti is the time that the ray has spent on crossing the receiver, which can be
obtained from the velocity and the distance that the ray has crossed within the receiver.
Let us illustrate how to calculate this distance by looking at Figure 3.8a, where a
spherical receiver is crossed by one ray. Three quantities are illustrated there; dri the
distance traveled through the receiver, d the shortest distance from the ray to the center
of the receiver, and r the radius.

Because the distance d from a line to a point can be calculated from Eq. (3.12) 95, and
because the radius of the sphere receiver is known, we can apply the Pythagoras
 68

theorem and obtain the desired distance as d ri = 2 r 2 − d 2 . If the distance dri is

smaller than the radius of the receiver, the ray has crossed it.

⎡ [(a − x1 )m − (b − y1 )l ]2 + [(b − y1 )n − (c − z1 )m]2 + [(c − z1 )l − (a − x1 )n]2 ⎤

1/ 2

d =⎢ ⎥
⎣⎢ l 2 + m2 + n2 ⎦⎥
(3.12)

Figure 3.8: a) Sphere receiver. b) Cube receiver.

Finally, it can be derived that the relationship between the sound energy of a ray and the
sound intensity ‘impinge’ to the sphere receiver is 92:

E i ⋅ c Wi ⋅ Ti ⋅ c Wi ⋅ d ri
Ii = = = (3.13)
Vr Vr Vr

The corresponding arrival time of every ray at detection will determine where the
intensity is allocated in the intensity response.

3.3.3. Calculation of acoustic indices

Once all the rays have been traced, we can consider the impulse response obtained at
every receiver almost complete. The last thing that we have to do is to change back to
decibels by the use of Eq. (1.19). The reason for doing this is because the different
acoustic indices that we can calculate are obtained using sound levels 24.
 69

The impulse response, see Figure 1.9, could be saved on disk in two different ways. The
first includes two columns of data: one for the intensity or energy of every pulse, and
the other for their corresponding time delay. Another possibility is to divide the impulse
response into certain ∆t steps, e.g. 1/44.1 kHz. Thus, only one column (or a row of
cells) is needed and that is how we have done.

Two acoustic parameters are introduced, the sound pressure level, SPL, and the
reverberation time, RT. These parameters provide information about the acoustic quality
of an enclosure, and also allow a good comparison of measured and modeled data. Both
can be obtained directly from the impulse response, see chapter 5 and Figure 5.1. Other
parameters can be obtained from the impulse response and are defined in Ref. 24.

3.3.3.1. Sound pressure level

Sound pressure level, SPL, is the experimental measurement of the amount of pressure
in the air being sensed at a given location. In the ray-tracing algorithm, it can be
obtained from Eq. (3.14), although it has the same value as the total intensity or energy
expressed in dB.

SPL = 120 + log10 ⎛⎜ ∫ I (t )dt ⎞⎟

∞
(3.14)
⎝ 0 ⎠

3.3.3.2. Reverberation time

Reverberation time, RT, is the time expressed in seconds, that would be required for the
sound pressure level to decrease by 60 dB after the sound source has stopped. The
accuracy in reverberation time measurements is limited due to random fluctuations in
the decay curves, which are produced as a result of the randomness of the excitation
signal. The same effect occurs in modeled impulse responses. A popular method
minimizes the effect of the fluctuations in decay curves by repeating the experiment
many times and then averaging. This is inefficient, especially when modeling. Instead,
we can apply a method proposed by Schroeder 96, the Reverse integration of the squared
impulse response, also called the integrated impulse response method, which is more
efficient than spatial averaging and forms the basics of the standard ISO 338224. For
each octave band a backward integration of the squared impulse response is performed.
In an ideal situation with no background noise the integration should start at the end of
 70

the impulse response (t → ∞ ) and proceed to the beginning of the squared impulse
response. Thus the decay as a function of time is:

∞ t ∞ t
E (t ) = ∫ p (τ )dτ = ∫ p (τ )d ( −τ ) = ∫ p (τ )dτ − ∫ p 2 (τ )dτ
2 2 2
and /
t ∞ 0 0

or

∞ tn ∞ tn

E (t ) = ∫ p 2 (τ )dτ = ∫ p 2 (τ )dτ + ∫ p 2 (τ )dτ = ∫ p 2 (τ )dτ + C (3.15)

t t tn t

In chapter 5, i.e. Figure 5.1, we will see how the Schroeder method applies in modeled
impulse responses.

The reverberation time is then obtained by a least-squares fit line to the decay obtained
by Schroeder over the range from 5 dB to 35 dB below the initial level, represented as
T30, or from 5 dB to 25 dB, represented as T20.
 71

4. Materials and methods

Three different enclosures have been studied. Measured data and modeling results of
two acoustic parameters, sound pressure level SPL and reverberation time RT, have
been compared in order to evaluate the accuracy of our algorithm. In this chapter the
geometrical and the acoustical descriptions of the enclosures are given. Also the way the
measurements were performed and how the modeling was carried out are described.

4.1. Description of the enclosures

The enclosures have been chosen to cover three different types of indoor spaces. A
reverberant room, an industrial hall, and an auditorium. Those spaces are referred in the
following as Room 1, Room 2a, Room 2b and Room 3. Previous studies of Room 2 and
Room 3 have already been reported 97, 98.

4.1.1. Reverberant room, Room 1

A reverberant room has highly reflecting surfaces (low absorption characteristics), and
therefore its reverberation time is fairly long. The dimensions of the enclosure were 2.9
x 7.6 x 3.6 m. The bounding surfaces were of concrete. There were three wooden doors,
and a large opening which was also covered with wood. On the ceiling there was an
acoustic resonator.

4.1.2. Industrial hall, Room 2a and Room 2b

The laboratory of acoustics of the FIOH studied the hall in 1999, during a noise control
project 4, 66. For that project, the acoustic characteristics of the enclosure were modified,
so there are data available before and after the acoustic treatment was performed. The
hall before the acoustical treatment is named as Room 2a and after it as Room 2b.

This space was an industrial textile hall with dimensions 40 x 20 x 8.5 m. It had a
saddle roof with 5 baulks, 16 weaving machinery pieces of an approximate height of 2
meters, and some shelves. The floor was made of concrete, also the front wall and the
two side walls. Special characteristics were a corridor that connected to another room
through the front wall, and opposite it, an opening that led to another large hall, which is
not included in the model. The acoustic treatment that was carried out consisted of
covering the two side walls and the ceiling baulks with 50 mm mineral wool. A photo of
the industrial hall is shown in Figure 4.1.
 72

Figure 4.1: View of the industrial hall, Room 2b.

4.1.3. Auditorium
Photos of the studied auditorium are shown in Figure 4.2. It had an approximate volume
of 1050 m3, and a rectangular ground area of 4.6 x 13 m. The seating area was inclined
by approximately 20 degrees with a capacity of 180 seats distributed in 10 rows. The
walls were of concrete and the floor was covered by wool carpet. There were absorbent
elements in the sidewalls. The same material was suspended from the ceiling by 90-200
cm. Tables were located in the ground area. A strong flutter echo was observed between
the side walls using handclaps.

Figure 4.2: Views of the auditorium, Room 3. Omnidirectional loudspeaker at location

S1 and receiver at location R1.
 73

4.2. Measuring methods

The measurements in Room 1, Room 2a and Room 2b were done in 1999 by Hongisto
and Keränen. The author of this thesis was directly involved in the measurements of
Room 3 in 2004.

4.2.1. SWL of the sound source

The sound power level of the sound source, B&K OmniPower 4296, was measured
99
before every study according to ISO 3741 in a reverberant chamber . The source is
proved to be omnidirectional. The obtained values for the six studied octave bands are
listed in Table 4.1.

Table 4.1: Measured sound power level of the sound source (ref. 1 pW)
125 250 500 1000 2000 4000
Source at Room 1 92.8 93.7 93.7 92.4 92.2 84.4
Source at Room 2 105.1 106.1 106.1 104.2 103.8 96.2
Source at Room 3 103.1 111.1 107.2 102.9 103.0 96.6

4.2.2. Measurement of SPL and RT

In Room 1, Room 2a and Room 2b, an omnidirectional loudspeaker (B&K 4296), a
power amplifier (QSC 1300 W USA) and a pink noise signal generator (Behringer DSP
800) were used as the sound source. Sound pressure levels, SPL, were measured using a
precision sound level meter (B&K 2260A) equipped with a ½” microphone (B&K
4189). For the reverberation time measurements a digital tape recorder (Tascam DA-P1)
was used to record pistol shots. The shots were recorded via the sound level meter and
the recordings were analyzed with a real time analyzer (B&K 2133) using Schroeder’s
backward integration method. The RTs were determined using a decay of 20 dB. The
final RT values are arithmetic averages of several RT measurements.

In Room 3 the sound pressure level was measured in 1/3 octaves with an Investigator
B&K 2260A and a calibrated microphone (B&K 4189). The sound source was the same
as in previous rooms. The height of the receivers was the same height as the ears of a
possible listener sitting in the audience area, 1.2 meters above the floor. At the same
locations, room responses were measured with WinMLS2000 using a sound level meter
(B&K 2231 with microphone B&K 4165). From the room responses the RTs were
calculated using a decay of 20 dB.
 74

4.2.3. Measurement and estimation of the absorption properties of

materials
As we have mentioned above, it is not always possible to reliably know the absorption
properties of materials, and then their estimation becomes a guess. That was the case in
Room 1 and Room 2, where a library database was used. On the other hand, we had the
chance to study in the laboratory the most important materials present in Room 3 (wall
and ceiling absorbent, and audience chairs). The principles and measuring procedure of
ISO 354 are described in Appendix 1, although they are introduced now. The
reverberation time of a reverberation chamber is measured. Later, the specimen of the
material is placed in the room and the reverberation time is measured again. The
difference between the two reverberation times allows the calculation of the absorption
coefficients of the surface, and these are called random incidence absorption
coefficients.

4.3. Modeling of the enclosures

The studies were done only for six octave bands, from 125 Hz to 4000 Hz. We did not
model the 63 Hz and 8 kHz octaves because loudspeakers are normally inefficient at
those frequencies. As mentioned above, the absorption coefficients were estimated or
measured, and the scattering coefficients were always estimated to be constant through
the frequency range. The absorption coefficients of the air were taken from ISO 9613-
1100, and the temperature and humidity conditions of the enclosures were modeled as
they were when measuring. Because the effect on accuracy of the number of rays and of
the size of the receivers was unknown, different numbers of these were used.

4.3.1. Reverberant room, Room 1

In Figure 4.3, some views of the modeled room can be seen. Receivers with a radius of
40 cm were located at the same locations as measured, see Table 4.2. In Figure 4.3, the
sound source has been represented by a point at a certain height, and the spherical
receivers by a point and a circle. The sound power level of the modeled sound source
corresponds with the first row of Table 4.1. Four different construction materials were
used in the room. Their absorption characteristics, which are listed in Table 4.3, were
estimated from the material database. In Table 4.3, the estimation of the scattering
coefficient (Scatt.) for every surface has been included. The last column in Table 4.3
 75

illustrates the total area covered by each material in m2, and the last row represents the
average modeled absorption and scattering distributed throughout the room in m-2.

Table 4.2: Locations of source, S1, and receivers, R1, R2,…,R6, in Room 1 (in meters).
S1 R1 R2 R3 R4 R5 R6
X 1.45 1.45 1.45 1.45 1.45 1.45 1.45
Y 1.5 2.03 3.04 4.05 5.07 6.08 7.09
Z 1.50 1.50 1.50 1.50 1.50 1.50 1.50

Table 4.3: Description of modeled surfaces in Room 1. Absorption and scattering

(Scatt) coefficients, area in m2, and average in m-2.
125 250 500 1000 2000 4000 Scatt. Area
Walls, Floor, Ceiling 0.01 0.01 0.01 0.02 0.02 0.02 0.2 84.34
Doors (wood) 0.14 0.1 0.06 0.08 0.1 0.1 0.2 6.32
Acoustic resonator 0.15 0.25 0.4 0.55 0.6 0.6 0.2 0.96
Opening (wood) 0.15 0.11 0.1 0.07 0.06 0.07 0.2 10.16
Average 0.03 0.03 0.03 0.03 0.03 0.04 0.2

Figure 4.3: 3D and 2D front and side views of the modeled reverberant room, Room 1.
(Dimensions in meters).

4.3.2. Industrial hall, Room 2a and Room 2b

The appearance of the room model is shown in Figure 4.4. The locations of the sound
source and of the receivers are listed in Table 4.4. The radius of the receivers was 50 cm
and their centers were at the same level as the top of all the machinery fittings. For
simplicity, all the machines were considered to have similar geometry and absorptive
properties. The absorption characteristics of all the modeled surfaces were estimated
from the database. The acoustic treatment, which consisted of 50 mm mineral wool, was
 76

mounted on the sidewalls and in the ceiling baulks. These data are illustrated in Table
4.5 for Room 2a (before acoustic treatment) and in Table 4.6 for Room 2b (after
acoustic treatment). The estimation of the scattering coefficient (Scatt.) was done
according to Table 2.1. The scattering coefficient of the floor has been modeled high
due to the amount of fittings lying over it. The scattering coefficients of other surfaces
have been modeled with a value of 0.5 for simplicity. In Table 4.5 and Table 4.6, the
last column represents the total area covered by each material in m2, and the last row
represents the average modeled absorption and scattering distributed throughout the
room in m-2. The sound power of the sound source corresponds with the second row of
Table 4.1.

Table 4.4: Locations of source, S1, and receivers, R1, R2,…,R6 in Room 2a and Room
2b (in meters).
S1 R1 R2 R3 R4 R5 R6
X 8.5 27.0 9.2 2.5 10.0 27.0 29.0
Y 6.0 6.0 1.4 6.0 18.0 13.5 18.8
Z 1.5 1.5 1.5 1.5 1.5 1.5 1.5

Table 4.5: Description of modeled surfaces in Room 2a (before acoustic treatment).

Absorption and scattering (Scatt) coefficients, area in m2, and average in m-2.
125 250 500 1000 2000 4000 Scatt. Area
Floor 0.01 0.01 0.01 0.02 0.02 0.02 0.7 800
Walls, Bricks 0.02 0.03 0.03 0.04 0.05 0.07 0.5 860
Ceiling, Baulks 0.01 0.02 0.02 0.02 0.02 0.05 0.5 1101
Corridor, Machinery 0.50 0.50 0.50 0.50 0.50 0.50 0.5 649
Open wall 0.99 0.99 0.99 0.99 0.99 0.99 0.5 165
Average 0.15 0.15 0.15 0.16 0.16 0.17 0.54

Table 4.6: Description of modeled surfaces in Room 2b (after acoustic treatment).

Absorption and scattering (Scatt) coefficients, area in m2, and average in m-2.
125 250 500 1000 2000 4000 Scatt. Area
Floor 0.01 0.01 0.01 0.02 0.02 0.02 0.70 800
Bricks 0.02 0.03 0.03 0.04 0.05 0.07 0.50 56
Ceiling 0.01 0.02 0.02 0.02 0.02 0.05 0.50 801
Walls, Baulks 0.20 0.70 0.95 0.95 0.95 0.95 0.50 1104
(acoustic treatment)
Corridor, Machinery 0.50 0.50 0.50 0.50 0.50 0.50 0.50 649
Open wall 0.99 0.99 0.99 0.99 0.99 0.99 0.50 165
Average 0.20 0.36 0.44 0.44 0.44 0.45 0.54
 77

Figure 4.4: 3D and 2D ground and side view of the modeled industrial hall, Room 2a
and Room 2b. Dimensions in meters.

4.3.3. Auditorium, Room 3

The measurements and modeling in Room 3, Figure 4.5, were done over ten points, and
repeated for two different locations of the sound source, altogether twenty measuring
points, see Table 4.7. The size of the receivers was conditioned by their proximity to the
audience chairs, as they should not overlap such planes. Their ratios were 38 cm and
their height corresponded with the height of a listener in the audience, 120 cm above the
floor. Ten different types of materials were used to model the auditorium. Their
estimated absorption and scattering coefficients are listed in Table 4.8. This time we had
the chance to study, in our laboratory, the absorptive performance of the most important
materials, according to ISO 35490. These data correspond with the first three rows of
Table 4.8. The wall absorber was measured with a gap of 7 cm above the floor, and the
ceiling absorber with a gap of 90 cm. The rest of the materials were estimated from the
material database. Again, the last column in Table 4.8 represents the total area covered
by each material in m2, and the last row represents the average modeled absorption and
scattering distributed throughout the room in m-2. The sound power of the sound source
corresponds with the last row of Table 4.1.
 78

Table 4.7: Locations of sources and receivers in Room 3 (in meters).

S1 S2 R1 R2 R3 R4 R5 R6 R7 R8 R9 R10
X 2.85 1.01 4.15 4.25 5.20 5.97 8.05 8.12 9.01 11.07 12.01 13.12
Y 11.30 7.19 10.12 4.56 3.06 11.26 9.60 3.58 7.41 3.55 10.80 7.98
Z 1.50 1.50 1.20 1.20 1.65 2.10 3.00 3.00 3.35 4.35 4.80 5.20

Table 4.8: Description of modeled surfaces in Room 3. Absorption and scattering

(Scatt) coefficients, area in m2, and average in m-2.
125 250 500 1000 2000 4000 Scatt. Area
Wall Absorber 0.09 0.35 0.85 0.63 0.38 0.47 0.4 32
Ceiling Absorber 0.45 0.45 0.52 0.60 0.50 0.60 0.6 192
Chairs, soft side 0.13 0.45 0.75 0.94 0.94 0.91 0.8 68
Chairs, hard side 0.09 0.11 0.09 0.07 0.06 0.07 0.8 116
Concrete Walls 0.01 0.01 0.02 0.02 0.02 0.03 0.1 210
Hard Parts in Ceiling 0.10 0.10 0.10 0.10 0.10 0.10 0.4 12
Stairs (soft carpet) 0.03 0.04 0.09 0.20 0.34 0.45 0.7 157
Floor (soft carpet) 0.03 0.04 0.09 0.20 0.34 0.45 0.2 94
Tables 0.09 0.11 0.09 0.07 0.06 0.07 0.4 28
Doors 0.09 0.11 0.10 0.07 0.06 0.07 0.2 6
Average 0.13 0.17 0.24 0.29 0.30 0.35 0.48

Figure 4.5: 3D and 2D ground and side view of the modeled auditorium, Room 3.
Dimensions in meters.
 79

4.4. Validation methods

The validation of any acoustic modeling method has to be done by a comparison with
measured data, and by an evaluation of its accuracy. However, it should be pointed out
that there are no perfect measurements, and that three types of factors affect the
accuracy of a measurement, i.e. human, device and external factors. The first type
occurs when two different persons make a set of measurements or when one repeats
his/her own ones. It might be that the measurement locations are not the same, or
neither is the orientation of the microphones. The second type is due to the devices. The
accuracy of the sound source SWL, and the sound level meter accuracy, are aspects to
be considered. For instance, the microphone used in Room 3 was a free field
microphone that is optimized to sound arriving from directions straight to it. On the
other hand, the modeled spherical receivers are considered to have omnidirectional
sensitivity. Finally, there are external factors like changes in temperature and humidity
conditions, or changes in the background noise. Normally, it is assumed that the
accuracy of acoustic measurements in 1/1 bands is within ± 3 dB in a reverberant room
97
, and ± 5 dB in the open field 100, 102.

Sound pressure levels SPL and reverberation times RT were measured. In modeling an
impulse response was obtained for every receiver and at every frequency octave. From
these SPL and RT were calculated. The SPL at a point is calculated by adding all the
energy pulses of the corresponding point response, Eq. (3.14). Both measured and
modeled data can be compared. Their difference at a certain receiver location i and
frequency f is obtained from Eq. (4.1). This can be averaged over all the receivers to
obtain the mean value of the deviation, Eq. (4.2). Applying Eq. (4.3), we can obtain the
standard deviation of the difference between what is predicted by modeling and what is
measured at any frequency octave.

Ai , f = SPLi , f , Modeled − SPLi , f , Measured (4.1)

N
1
ASPL f =
N
∑A
i =1
i, f (4.2)
 80

2
N
⎛ N ⎞
N ∑ A − ⎜ ∑ Ai , f ⎟
2
i, f

STD f = ±
i =1 ⎝ i =1 ⎠ (4.3)
N ( N − 1)

The comparison between modeled and measured RT is done visually from the plots.
 81

5. Results

5.1. Reverberant room, Room 1

In figure 5.1, four modeled point responses in the reverberant room, Room 1, are
plotted. Those plots represent the sound decay of the enclosure at a certain position.
They have been obtained at two different receivers, R2 (left) and R6 (right), for two
different frequencies, f = 125 (upper) and f = 2000 Hz (lower).

SPL and RT can be calculated directly from the data of the plot. SPL is obtained by
direct application of Eqs. (1.18) and (1.19) over all the intensity pulses. RT is calculated
by applying its Schroeder backwards integration method and fitting a line to it by the
least-squares method over the range 5 to 35 dB below the initial level, see section
3.3.3.2.

Figure 5.1: Point responses obtained in Room1 in receivers, R2 (left) and R6 (right), at
the frequency octaves f = 125 Hz (upper) and f = 2000 Hz (lower). x-axis represents
time with a time interval of ∆t = 0.01 seconds, while the y-axis represents SPL in dB.

The difference between modeled and measured SPL in Room 1 has been illustrated in
Figure 5.2, where the middle point represents what was obtained by Eq. (4.2) and the
 82

error bars what was calculated by Eq. (4.3). The same calculations were repeated to
compare also the prediction by the Sabine theory, Eq. (2.1), with the measured data.

Figure 5.2: Room 1, averaged difference in SPL between modeled and measured, and
between Sabine and measured. (mean value by Eq. (4.2) and error bars by Eq. (4.3)). N
= 1000 rays and r = 40 cm.

5.2. Industrial hall, Room 2a

In Figure 5.3, six modeled sound decays of the industrial hall, Room 2a (before acoustic
treatment) are plotted. They correspond with the point responses obtained at receiver R4
with r = 50 cm. The three sound decays on the left in Figure 5.3 correspond with the
125 Hz octave band. The three on the right correspond with the 2000 Hz octave band.
For both octave bands, and from top to bottom, they illustrate the cases N = 3000, N =
10000 and N = 30000 rays. SPL and RT can be calculated in the same way as described
in section 5.1 for Room 1.

Figure 5.4 represents the difference between modeled and measured SPL by the use of
(4.2) and (4.3) in Room 2a. The modeled results correspond to the case where N =
30000 rays and r = 50 cm. As in figure 5.2, the difference between predicted SPL by the
Sabine theory and the measured SPL has also been illustrated.

Figure 5.5 represents the difference between modeled and measured reverberation time,
RT, in Room 2a. The modeling parameters are the same as in Figure 5.4, N = 30000
rays and r = 50 cm.
 83

Figure 5.3: Six point responses obtained in Room 2a at receiver R4 with r = 50 cm. On
the left for frequency f = 125 Hz, and on the right for frequency f = 2000 Hz. Three
cases, N = 3000, N = 10000, N = 30000 rays. x-axis represents time with a time
interval of ∆t = 0.001 seconds, while the y-axis represents SPL in dB.
 84

Figure 5.4: Room 2a, averaged difference in SPL between modeled and measured, and
between Sabine and measured. (mean value by Eq. (4.2) and error bars by Eq. (4.3)). N
= 30000 rays and r = 50 cm.

Figure 5.5: Modeled, measured, and Sabine reverberation time in Room 2a. N = 30000
rays.

It is possible to compare the predictions made by our model in Room 2a with the
predictions made by the commercial software ODEON v3.1. This is done very lightly
simply with the idea of comparing our model, which is nicknamed dBWorks, with any
other existing model. Such a comparison is only possible for enclosures, Room 2a and
 85

Room 2b, and it is based on the work of Keränen 4. The prediction of SPL by both
methods is compared with the measured data, in Table 5.1 for Room 2a. The
comparison is illustrated by the results of the arithmetic mean Eq. (4.2), and the
standard deviation Eq. (4.3).

Table 5.1: Difference in Room 2a between modeled SPL by dBWorks and by ODEON
with measured SPL. Arithmetic mean (ASPL) by Eq. (4.2) and Standard deviation by Eq.
(4.3).
125 250 500 1000 2000 4000
dBWorks Mean(ASPL) 1.6 1.7 1.5 1.0 0.8 0.7
STD 1.4 1.3 0.9 0.6 1.2 1.0
ODEON Mean (ASPL) 1.2 1.7 1.3 0.7 0.2 -0.6
STD 1.1 0.8 0.9 0.7 1.1 1.1

5.3. Industrial hall, Room 2b

In Figure 5.6, six modeled sound decays of the industrial hall, Room 2b (after acoustic
treatment) are plotted. They correspond with the point responses obtained at receiver R4
with r = 50 cm. The three sound decays on the left in Figure 5.6 correspond with the
125 Hz octave band. The three on the right correspond with the 2000 Hz octave band.
For both frequency octaves, and from top to bottom, they illustrate the cases N = 3000,
N = 10000 and N = 30000 rays. Note that the time axis has changed with respect to
Figure 5.3.

Figure 5.7 represents the difference between modeled and measured SPL in Room 2b,
where the middle point is the difference between both data, Eq. (4.2), and the error bars
its standard deviation, Eq. (4.3). The modeled results correspond to the case where N =
30000 rays and r = 50 cm. As in Figure 5.2, the difference between predicted SPL by
the Sabine theory and measured SPL has been also illustrated.

Figure 5.8 represents the difference between modeled and measured reverberation time,
RT, in Room 2b, when N = 30000 rays and r = 50 cm.
 86

Figure 5.6: Six point responses obtained in Room 2b at receiver R4 with r = 50 cm. On
the left for frequency f = 125 Hz, and on the right for frequency f = 2000 Hz. Three
cases, N = 3000, N = 10000, N = 30000 rays. x-axis represents time with a time
interval of ∆t = 0.01 seconds, while the y-axis represents SPL in dB.
 87

Figure 5.7: Room 2b, averaged difference in SPL between modeled and measured, and
between Sabine and measured. (mean value by Eq. (4.2) and error bars by Eq. (4.3)). N
= 30000 rays and r = 50 cm.

Figure 5.8: Modeled, measured, and Sabine reverberation time in Room 2b. N = 30000
rays.

In the same way as for Room 2a, the SPL prediction by our model, dBWorks, and by the
commercial software ODEON v3.1 have been compared in Table 5.2.
 88

Table 5.2: Difference in Room 2b between SPL modeled by dBWorks and by ODEON
with measured SPL. Arithmetic mean (ASPL) by Eq. (4.2) and Standard deviation by Eq.
(4.3).

125 250 500 1000 2000 4000

dBWorks Mean (ASPL) -0.6 -1.3 -1.1 -0.4 1.2 -6.7
STD 2.7 2.0 2.1 2.1 2.2 1.9
ODEON Mean (ASPL) 1.3 -1.4 -1.5 -1.2 -1.1 -2.5
STD 2.0 1.3 1.6 1.4 1.5 1.5

5.4. Auditorium, Room 3

Figure 5.9 represents the difference between modeled and measured SPL and between
Sabine and measured SPL in Room 3. The upper plot represents the case when the
sound source was at location S1, and the lower plot when the sound source was at
location S2. Both cases for N = 10000 rays and r = 38 cm.

Figure 5.10 represents the difference between modeled, measured and Sabine
reverberation time, RT, in Room 3. The upper plot represents the case when the sound
source was at location S1, and the lower plot when the sound source was at location S2.
Both cases for N = 10000 rays and r = 38 cm.
 89

Figure 5.9: Room 3, averaged difference in SPL between modeled and measured, and
between Sabine and measured. (mean value by Eq. (4.2) and error bars by Eq. (4.3)). N
= 10000 rays and r = 38 cm. Top: source S1. Bottom: source S2.
 90

Figure 5.10: Modeled, measured, and Sabine reverberation time in Room 3. N = 10000
rays, r = 38 cm. Top: source S1. Bottom: source S2.
 91

Figure 5.11: Sound decays at receiver R4 with source S2 in Room 3. Left: f = 125 Hz.
Right: f = 1000 Hz. x-axis represents time with a time interval of ∆t = 0.001 seconds,
while the y-axis represents SPL in dB.
 92

6. Discussion

6.1. Reverberant room, Room 1

As a consequence of the low absorption properties of this enclosure, the sound decays
are very smooth with fairly long reverberation times, see Figure 5.1. The decays are
steeper for higher frequencies. The gradient of the sound decays are independent of the
location. According to these data and with the concepts that were introduced in section
1.5, this enclosure fulfils the first requirement for the existence of a diffuse sound field,
according to Beranek 27. The second requirement is about the irregularities of the walls
for a good scattering of sound. This is not met at all because all surfaces were quite
smooth and because there was no furniture. That may be the reason, as will be discussed
in the following, for the inaccuracy in the estimation of the SPL at low frequencies.

According to Figure 5.2, the mean value of the difference between modeled and
measured SPL, Eq. (4.2), is smaller than ± 1.6 dB, and the standard deviation, Eq. (4.3)
is within ± 1.5 dB. In the same figure we can see how at low frequencies (125 and 250
Hz) the model underestimates the SPL. One of the sources of potential errors is found to
be in the estimation of absorption coefficients, and that underestimation could be due to
an overestimation of the absorption properties of the walls. However, this may not be
the reason in this case, since the absorption characteristics of the most important walls
are already very low. Instead, we could consider the reason to be in interference effects
as a consequence of the low diffuse properties of Room 1. It was introduced, in section
1.3.2, that standing modes may appear between highly reflective parallel walls.
Although the interference effect was lightly simulated in this case by considering the
low scattering coefficient, see Table 4.2, its consequences are not correctly represented.
In section 1.3.2, we also introduced the existence of interference effects at reflecting
boundaries in reverberant sound fields. That is another reason for the discrepancies
between modeled and measured SPL at low frequencies. Due to the dimensions of
Room 1, the receivers are 1.45 meters away from the side walls, which is not enough to
ensure that the receivers are inside a node or an antinode. For a frequency of 125 Hz
with a corresponding wavelength of 2.6 meters, y / λ = 0.56 . For this relation, the
increment of SPL due the presence of standing modes can be up to 3 dB, see Figure 1.8.
 93

It is observed in Figure 5.2 how both modeled and Sabine estimation of the sound
pressure level are within 0.3 dB. This is obvious since the Sabine theory considers the
existence of a diffuse sound field, and this room partly fulfils these requirements.

Figure 5.2 referred to the case where the number of rays N was 1000 and the receivers
had a ratio r of 40 cm. Topics of interest in room acoustics modeling are; what is the
optimum number of rays, or how big the receivers should be. For this enclosure the
running process was repeated with N = 3000 rays obtaining the same values within 0.1
dB. Using same amount of rays, but reducing the size of the receivers from 40 cm to 20
cm, again produced results within 0.1 dB of the results illustrated in Figure 5.2. This
suggests that in similar enclosures (small, without barriers or furniture, with low
absorption and diffuse properties, and with non-locally reacting surfaces) the number of
rays does not need to be very high to achieve convergence, and nor does the size of the
receivers.

Still further, a final test was carried out with N = 1000 rays and receivers with r = 40
cm. But this time the scattering coefficients of all surfaces were changed from 0.2 to
0.05. These results are again within 0.1 dB of the results of Figure 5.2, suggesting that
in this type of enclosure, where the sound remains for a considerably long time, the way
reflections are modeled is not a determinant factor.

6.2. Industrial hall, Room 2a

Figure 5.3 presents six modeled sound decays at receiver R4 in the industrial hall, Room
2a. The three sound decays on the left of Figure 5.3 correspond to the 125 Hz octave
band. The three on the right correspond to the 2000 Hz octave band. For both octave
bands, and from top to bottom, they correspond with the cases N = 3000, N = 10000 and
N = 30000 rays. Some comments can be made, and we start by comparing the slope of
the sound decays. It is observed that the upper slopes are not as smooth as the others,
especially after one second. In section 2.3.6 the importance of the number of rays for
convergence in prediction was discussed. A too low number of rays may cause an
insufficient number of detected reflections, and/or also, the importance of the energy of
every detected reflection may be too high, as the energy of every ray depends on the
total amount of rays, N, Eq. (3.10). Those two factors produce inaccuracies in the
determination of the sound decay, and therefore also in the estimation of the
 94

reverberation time. However, we have to wait for results in Room 2b to see how these
inaccuracies increase when the total absorption increases. Also, we can compare how
the gradient of the sound decays in the 125 Hz frequency band are not as steep as in the
2000 Hz frequency band. This difference is the result of a slightly larger air and wall
absorption at the higher frequency.

According to Figure 5.4, the mean value of the difference between modeled and
measured SPL in Room 2a is smaller than 1.7 dB. The standard deviation is within
± 1.5 dB. SPL values are normally overestimated by the model, and larger departures
from measurements are found in the lower frequencies, probably as a consequence of
neglecting the wave nature of sound. We should remember that the height of the
receivers is the same as the height of the fittings, see Figure 4.4. Important diffraction
effects, which have not been modeled, may exist as a consequence of the multiple
obstacles in the line of sight between source and receiver, i.e. R4 and R5. It has been
previously stated 51 that reflections from small surfaces (diffraction due to the finite size
of surfaces, sections 1.3.1 and 2.2.2) are generally much weaker than calculated by the
laws of geometrical acoustics.

In Figure 5.4, we can also observe the difference between Sabine and measured SPL.
The mean value of the difference is within ± 2 dB, but the standard deviation is almost
± 4 dB. The reason for this large difference between predicted SPL by the Sabine theory
and measured SPL is due to the fact that the Sabine theory, see section 2.1, takes into
account only the distance from the sound source to the receiver, but not the exact
location of the receiver throughout the room, as the Sabine theory assumes the existence
a of diffuse field, see section 2.1.1.

Going back to the discussion about the importance and effect of the number of rays in
the calculation, we point out that in the case N = 3000 rays the main value of the
difference between modeled and measured SPL, calculated by Eq. (4.2), was ± 1.9 dB,
and the standard deviation, Eq. (4.3), was ± 1.5 dB. In the case of 10000 rays, the
values were 2.2 and ± 1.7 dB. In the three cases, the trends were similar along every
frequency. It seems that the number of rays is not a determining factor in the evaluation
of SPL. The answer is yes and not. In cases like Room 1 and Room 2a, the sound
absorption properties of the enclosures are low, this allowing the sound rays to travel for
a long enough time, and therefore, to achieve a sufficient number of detected collisions
 95

without a too large number of rays. Room 2b, where absorption properties are higher
illustrates the opposite case, see section 6.3.

Figure 5.5 presented the difference between modeled and measured reverberation time,
RT, in Room 2a. It can be observed that the measured data have greater values than the
others, but the trend of all of them is the same. A description of this enclosure was given
in section 4.1.2. It was commented there, how this room led to another hall by one wall,
and that this other hall was not modeled. Instead, the wall was modeled as almost totally
absorbent. It may be, that part of the sound that reaches that second hall comes back
attenuated up to some level and with a certain time delay. This effect may make the
reverberation time of the hall to longer. In consequence, the comparison between
measured and modeled RT does not tell much about the model’s accuracy. Another
reason could be suggested for the general underestimation of the RT, being an
overestimation of the absorption properties of the hall. However, that might not be the
reason as the prediction of SPL was indeed very good. Anyway, it should be
commented now that the measurements of the RT in that room (year 1999) were taken
only in three locations, and the locations were different from the locations where SPL
was measured, and from the locations that we have simulated. From Figure 5.5, it is also
observed that the prediction of RT by the Sabine theory and by the model lead to similar
values. In this respect, it can be stated that the Sabine theory has proved to be efficient
only in enclosures where absorption is low 25, 26.

6.3. Industrial hall, Room 2b

It is important to notice that in Figure 5.6, the time axis has changed with respect to
Figure 5.3. This had to be done, as in Room 2b the absorption was higher than in Room
2a. Thus, a visual comparison between the gradient of the sound decays obtained in
both rooms as a function of the amount of absorption has been sacrificed. However, an
evaluation of the acoustic treatment effect can be carried out by comparing the
corresponding SPL and RT values, which are listed in both figures.

From Tables 4.4 and 4.5 it can be seen how the absorptive treatment carried out in the
balks and side walls of the industrial hall was not very effective in the 125 Hz octave
band. This is why a reduction of ‘only’ 2 dB in SPL but of 0.5 seconds in RT was
obtained at that frequency. Instead, in the 2000 Hz octave, the corresponding reduction
 96

is of about 7 dB in SPL and 1.4 seconds in RT, as a consequence of the high absorptive
properties of the used materials at that frequency.

In Room 1 and Room 2a, the number of rays was not a strongly determining factor for
convergence in the prediction of SPL. In this case, Room 2b, the effect is bigger, and as
we discuss, it is directly related to the amount of absorption throughout the hall. For
instance, in Figure 5.7, which referred to the case N = 30000 rays, we can see that the
model has predicted, with high accuracy, the SPL with -1 dB, with a standard deviation
of ± 2.2 dB. A special case was found in the 4000 Hz, with a large underestimation of
the SPL. It is obvious that, at this frequency, the estimation of the absorption properties
of the material introduced in the acoustic treatment is greater than what it really was.
We no longer consider that frequency, and we concentrate the discussion on the others.
When N = 3000 rays was used, the prediction of SPL was underestimated by the model
by –2.8 dB, with a large standard deviation, i.e. of ± 8 dB in the 2000 Hz octave. When
the running process was repeated with N = 10000 rays, the prediction was better, with –
2.5 dB and ± 2.5 dB, respectively.

From all these data, it can be concluded that in enclosures where absorption is
considerably high, the number of rays for convergence in the prediction of SPL needs to
be large. In Room 2b, 30,000 rays are considered to be good enough.

In Figure 5.8, the prediction of RT in Room 2b by the model and the Sabine theory and
the corresponding measured data, have been illustrated. A large underestimation is
observed in the model prediction. Again, we could think the reason to be in the
connection of that enclosure through an opening to another hall. We remember that the
other hall has not been modeled and instead the opening has been modeled as an almost
totally absorbent surface. It may be that the sound going through the opening to the
other hall, comes back attenuated after a certain time, affecting the sound decay.

In Figure 5.8, a large standard deviation in the prediction of RT is found in the 4000 Hz
octave. From the results of Figure 5.7 and the values in Table 4.5, it was concluded that
a large overestimation was made of the absorptive properties of the acoustic treatment at
that particular frequency. But checking the individual sound decays obtained at every
receiver, it is found that in receiver R6 a too low number of detected reflections was
found, producing a poor estimation of the rate of the sound decay. It could be that the
 97

used number of rays, N = 30000, was not enough to accurately predict the reverberation
time in this enclosure with considerably high absorptive properties. However, before
increasing the number of rays it should be confirmed that the receiver detects the most
important reflections, which are the first ones. Because diffuse reflections have been
modeled since the first reflection of every ray, it could be that some important
propagation paths have been missed. This problem could have been solved by the image
method.

6.4. Auditorium, Room 3

According to Figure 5.9, the mean value of the difference between modeled and
measured SPL in Room 3 when source is at location S1, is smaller than ± 1.4 dB, and
the standard deviation is within ± 1.8 dB. When the sound source is at location S2, these
values are ± 1 dB and ± 2 dB. Those deviations are acceptable and also smaller than
expected, seeming to confirm the reliability of the estimated absorption coefficients.

In a diffuse room, the sound is homogeneously distributed in the space and the
reverberation time is constant in the room volume. As flutter echo was observed in this
auditorium, the room could not be considered as a diffuse space, even when the
audience chairs, the stairs, and the wall and ceiling absorbent material have diffusing
properties. The small standard deviation of measured reverberation time seems to
confirm the partially diffuse properties of the room. However, it must be noticed that to
have a large standard deviation in the prediction of RT does not necessarily mean that
the accuracy of the algorithm is bad. It is normal that the sound decay in an enclosure
depends on the place where it is measured.

The ray-tracing algorithm is a statistical method and can not represent correctly the
wave nature of sound. In Figure 5.11, it was observed for the frequencies 125 and 250
Hz, that the difference between modeled and measured data becomes large, probably
because the ray-tracing algorithm is not able to represent standing modes and other
effects of an undulate nature. Such effects are easily observed at low frequencies, and
can make the reverberation time to be large.

The uncertainty in the estimation of the reverberation time is smaller at low than at high
frequencies. The key to this problem is found by looking at the individual sound decays,
i.e. the sound decay of R4 with the sound source at location S2, Figure 5.11. First it can
 98

be seen that the shape of the decay depends directly on the amount of absorption
distributed in the room. In the 1000 Hz octave sound decay, with a time delay of 0.26
and 0.40 seconds, two very high peaks have appeared and affected the Schroeder curve.
This receiver is in the lower part of the room, between the concrete walls that were
modeled as non-diffuse. It can be suggested that some rays were trapped between both
walls without being attenuated, until they finally crossed the receiver. In this case,
according to sound speed, the high peaks would correspond to 6 and 10 collisions
between both walls since they were shot. At 125 Hz, the decay is not so steep as at 1
kHz and the peaks are not so clearly observed, even when they also exist, like one with
a time delay of 0.37 seconds. This effect may also help to explain the differences in
results between the two locations of the sound source. Although the results in both
cases, S1 and S2, seem very similar, it could be that the accuracy in the determination of
SPL and RT has been somewhat conditioned by the location of the sound source
throughout the room. As we can observe in Table 4.6 and Figure 4.5, S1 is close to the
left wall of the auditorium, being at the same time between parallel walls. The other
source, S2, is more in the front space and has some fittings around it. In the first case, it
seems that a larger amount of rays have been trapped between the parallel walls, which
were highly reflecting and non-diffuse, affecting the determination of the acoustic
parameters, especially in the closest receivers.

As an experiment, the running process was repeated considering all collisions to be

purely diffuse (Scatt = 1). In this way, with no specular collisions, the effect of some
rays being trapped between sidewalls was avoided. This made the high peaks disappear,
which led to a larger underestimation of the reverberation time. In consequence, the
standard deviation in the prediction of RT throughout the hall became smaller.

Another thing that should not be forgotten and could explain the underestimation of the
reverberation time at low frequencies is the size and height of the receivers (0.38 and
1.2 meters). In Refs. 97 and 98, it was stated that the size of the receivers was an
important factor. Smaller sizes could produce greater standard deviations if there is not
a large enough number of detected reflections, see section 2.3.8. In this study, the
receivers were placed at the same height of a possible listener sitting in the audience
area, and their size was affected by their proximity to the chairs. The effects at low
frequencies that occur at such close distances to obstacles might not be reproducible at
all by simple ray tracing.
 99

Over a certain limit, the number of rays, N, did not play a significant role in the obtained
results. 5000 rays produced the same deviations as 10,000 and 15,000.

6.5. General discussion

The repeatability of calculations led to the same results and deviations. This could
indicate that the deviation of the created algorithm is small, and that all created random
numbers, i.e. for shooting rays or evaluating diffusion, correspond to homogeneous
random distributions.

The computation time is not the most important factor, although it has to be taken into
account. Obviously, it depends on the used computer processor, so it changes a lot
between one computer and another. Parameters that affect computation time are number
of rays, number of surfaces, and amount of absorption. As we described in Chapter 3,
the rays are traced up to a certain limit of energy. The faster the energy decreases due to
absorption, the sooner this limit is reached. It has been interesting to see how the
computation time for Room 1 with N = 3000, has been larger than for Room 2b with N
= 10000, even though the number of surfaces is larger in the second case. The
computation time in the first case was 1 1/2 hours, and in the second case 25 minutes,
with a Pentium4 © processor.

Further comments can be made when comparing the predictions done by our model,
dBWorks, with the predictions done by the commercial software ODEON v3.1. For
Room 2a, Table 5.1, the deviations of both methods with respect to measured SPL are
very similar. In Room 2b, Table 5.2, the predictions by ODEON are slightly better than
the predictions done by our model. However, from these results we could think that our
model is quite close in terms of accuracy to existing models. ODEON has proved to be
efficient 49, 50, 73, and it is the most popular prediction method in the Nordic countries.

11, 26, 32
It has been frequently stated in previous research works , and observed in this
study, that perhaps the most severe restriction of the ray-tracing method is the limited
frequency range over which the results are valid. It is an underlying assumption in the
ray-tracing algorithm that the wavelength corresponding to the lowest frequency of the
sound is small compared to the linear dimensions of the room and its surfaces. Although
at low frequencies the estimation of SPL has been satisfactory, the estimation of RT has
 100

been more problematic. But here, there is also a conflict with previous researches. For
12
instance, Howarth and Lam studied the prediction by ray tracing of SPL and RT in
seven different enclosures. The volumes of the enclosures varied between 2,000 and
110,000 m3, and their shapes were rectangular, asymmetric and fan-shape. When
averaging over all the enclosures, reverberation time was over-predicted at low
frequencies. Interference effects due to the wave nature of sound are negated by the ray-
tracing algorithm. Such interference effects, i.e. standing modes, contribute to
increasing the reverberation time in any enclosure where they are present. From that
point of view, the underestimation of RT by our model seems more logical than their
overestimation.

Another interesting topic deals with the way diffusion is modeled. It is presently
admitted that accounting for the scattering of sound by surfaces improves room
13, 16, 47, 49, 50, 56, 57
acoustics prediction . However, it seems that the way diffusion is
modeled is not very clear, and many authors 12, 43, 51, 53, 76, 77 have suggested that there is
more work to do in this field. This thesis has described how to implement the ray-
tracing algorithm, and has discussed its accuracy. We should be critical with ourselves,
and admit that the way we have implemented diffusion has been very simple and rough.
First, we have considered a constant scattering coefficient for all the frequencies.
Second, the directivity of the reflected sound has been modeled from random numbers
distributions. As will be mentioned in the next chapter, the future work of the author of
this thesis is directly involved with the way diffuse reflected sound should/could be
modeled. Nor has diffraction been modeled, although it might be present in enclosures
Room 2 and Room 3. In Room 2, sound arriving at the edge of all machinery fittings and
all ceiling baulks may be diffracted. In Room 3, diffraction effects may appear at the
edges of the audience chairs and seats. Not to consider diffraction in that room may
have affected the accuracy of the determination of RT at low frequencies.
 101

7. Conclusions and future work

This thesis has offered the mathematical implementation of the ray-tracing method
applied in room acoustics modeling. A discussion about its weak points has also been
presented. The prediction by the model of two acoustic parameters, SPL and RT, has
been compared with the corresponding measured data in three enclosures. Two main
conclusions can be outlined. First, the ray-tracing algorithm can represent room
acoustics with acceptable accuracy. Second, the inaccuracies might be due to negation
of the wave nature of sound at low frequencies. Diffraction, diffusion and interference
should be better taken into account. We have not discovered very much new, but now
we have our modeling tool ready and we are about to achieve the actual state of the art
in room acoustics modeling by ray tracing. It is also a tool to test new inventions, ideas,
and approaches.

The future work of the author involves three main lines. First, different series of
laboratory measurements would be carried out, in order, as introduced in the previous
chapter, to obtain more clues about the directional characteristics of reflected and
diffracted sound. These results will be applied in our model in order to obtain a better
representation of the phenomena involved in room acoustics. Second, the development
of the model should continue. The results of the mentioned measurements and other
suggested methods, e.g. Svensson 88, Rindel 76 and Metchel 101, should be applied in the
model. Other acoustic parameters can also be obtained from the impulse responses and
should be implemented, i.e. speech transmission index STI, and early decay time EDT.
Another very interesting topic for further development is the concept of auralization,
listening in a virtual environment. Although it has been mentioned throughout this
work, the concept of auralization has not been detailed as it was outside the scope of
this thesis, but the reader is referred to Refs. 56, 57 and 58 for an overview and main
principles. The implementation of auralization, and we are not far from it, can open very
big doors for work-place and auditoria design, if the acoustics of an environment can be
listened to before construction. Third, a proper validation of the model will be carried
out. It would be desirable to compare the prediction of our model with measurement
data in a wider group of enclosures. Such indoor spaces could be two auditoria, two
speech halls, two industrial halls, and an open-plan office.
 102

The future work of the author is directly related to this topic, and some approximations
should be studied, e.g. the recently presented findings by Rindel 76 described in section
2.2.1.2, or the corner source model suggested by Mechel 103.
 103

8. Acknowledgments
The Finnish Work Environmental Fund is endlessly appreciated for the financial
support throughout this work, project No. 103464. Virtual Space 4D and Tekes are also
appreciated. The author would like also to thank Dr Valtteri Hongisto, Mr Jukka
Keränen and all the personnel of the Laboratory of Ventilation and Acoustics of the
Finnish Institute of Occupational Health (FIOH) for their encouraging and intellectual
support.
 104

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 A1-1

Appendix 1

The absorption and scattering properties of materials are needed in room acoustics modeling.
Standard methods are available and they are briefly introduced in the following.

Measurement of sound absorption in a reverberation room, ISO 354A1

When a sound source operates in an enclosed space, the level to which reverberant sound builds up,
and the subsequent decay of reverberant sound when the source is stopped, are governed by the
sound-absorbing characteristics of the boundary surfaces and objects within the space. In general,
the fraction of the incident sound power absorbed at a surface depends upon the angle of incidence.
In order to relate the reverberation time of an auditorium, office, workshop, etc. to the noise
reduction that would be effected by an absorbing treatment, a knowledge of the sound-absorbing
characteristics of the surfaces, usually in the form of a suitable average over all angles of incidence,
is required. Since the distribution of sound waves in typical enclosures includes a wide and largely
unpredictable range of angles, it is convenient, for the purposes of standardization, to take a uniform
distribution as the basic condition. If, furthermore, the sound intensity is independent of location
within the room, such a distribution is called a diffuse sound field, and the sounds reaching a room
surface are said to be at random incidenceA1.

Reverberation time is the time that would be required for the sound pressure level to decrease by 60
dB after the sound source has stopped. The quantity is denoted by T and is expressed in seconds.

The calculation of the absorption coefficient of a certain material, A [m2], is based on the difference
between the reverberation time measured in a reverberation room with a sample of the material, T2,
and without it, T1. The equivalent sound absorption area of the test specimen is calculated by the
following formula

V ⎛1 1⎞
A = 55.3 ⎜⎜ − ⎟⎟ (A.1)
c ⎝ T2 T1 ⎠

where V [m3] is the volume of the empty reverberation room, c [m/s] is the speed of sound in air, T1
[s] is the reverberation time of the empty reverberation room.
 A1-2

The reverberation time of the room in each frequency band is expressed by the arithmetic mean of
the total number of reverberation time measurements made in that frequency band, for example, two
each of six sound source/microphone combinations.

Finally, the sound absorption coefficient α of the test specimen is calculated using the formula

A
α= (A.2)
S

where A [m2] is the equivalent sound absorption area calculated by Eq. (A.1), and S is the area [m2]
of the test specimen.

A photo of the test arrangement is shown in Figure A1, when the absorption coefficient of a
material present in Room 3 was measured in the laboratory of acoustics of the Finnish Institute of
Occupational Health, see sections 4.1.3 and 4.2.3.

Further information about the test arrangement, i.e. size and diffusion characteristics of the
reverberation room, size of the sample, and receiving equipment, is given in more detail in ISO
354A1.

Measurement of the random-incidence scattering coefficient of surfaces, ISO/DIS 17497-1A2

The principle of the measuring technique takes advantage of the fact that an impulse response,
measured over a diffuser in the same place but in different orientations, presents differences in
amplitude and phase in its late part. When impulse responses measured for different orientations of
the diffuser are averaged, the late non-correlated parts will cancel each other out, resulting in a
specular impulse response, whose decay is smaller when compared to a single measurement, see
Figure A2A3.
 A1-3

Figure A2: Impulses responses measured in a scale model reverberation chamber. After Vorländer
et alA3.

The scattering coefficient is defined as the ratio of the scattered energy to the total reflected energy.
In Figure 3, the energy components of the reflected sound are illustrated. From the total incident
sound, one part α , is absorbed by the material. The rest is partly scattered and partly specularly
reflected. This definition agrees very well with the model of diffuse reflections used in our ray-
tracing program.

Figure A3: Energy components of the reflected sound.

The method goes as follows. Impulse responses are measured without and with the sample
following ISO 354 and giving the reverberation times T1 and T2, respectively. The random
incidence absorption coefficient α of the material is then calculated by Eq. (A.1). Then, a rotating
 A1-4

base plate is placed in the reverberation room, and T3 is measured. Finally, the sample of the
material is placed on the rotating base, and n measurements of the reverberation time are performed,
with the sample rotated between each measurement by ∆ϕ = 360° / n . A value of n = 72 is preferred,
corresponding to angular steps of ∆ϕ = 5° . The mean value of all these measurements is T4. The
specular absorption coefficient, α SPEC , is calculated by Eq. (A.3)

V⎛ 1 1 ⎞
α SPEC = 55.3 ⎜⎜ − ⎟ (A.3)
S ⎝ c 4T4 c3T3 ⎟⎠

Under ideal conditions, the reverberation times T1 and T3 should be equal.

The random-incidence scattering coefficient s is finally calculated using the formula

α SPEC − α
s= (A.4)
1−α

More details about the test arrangement are given in ISO/DIS 17497A2.

Bibliography:

A1
ISO 354. Acoustics - Measurement of sound absorption in a reverberation room. International
Organization for Standardization, Genève Switzerland, 2003.

A2
ISO/DIS 17497-1 Acoustics - Measurement of the sound scattering properties of surfaces.
International Organization for Standardization, Genève Switzerland, 2002

A3
M. Vorländer, J-J. Embrechts, L. Geetere, G. Vermeir, M. Averlar Gomes. Case studies in
measurements of random incidence scattering coefficient. Acustica - acta acustica. Vol. 90 (2004)
858-867
 A2-1

Appendix 2

Survey of existing commercial software

A survey of existing commercial acoustic software has been carried out, in order to find out some
more about the current state of the art in room acoustics modeling. A total of 18 programs have
been considered. In Table 2.2, their name, price, internet address and home origin are listed. In
Table 2.3, their properties are summarized. It is important to notice the large range of prices. The
software referred to in Tables 2.2 and 2.3 by the number 17, dBWorks, is the acoustic model
developed by the author of this thesis.

Table 2.2: General information on commercial acoustic software.

Name Price Net address

1 ODEON 6.5 5080-14400 € http://www.dat.dtu.dk/~odeon Intotel, Finland
2 CATT v8 ? http://www.catt.se Catt, Sweden
3 RayNoise 24 650 € http://www.lmsintl.com LMS, Belgium
4 ULYSSES 995 € http://www.ifbsoft.de IFBsoft, Germany
5 SoundPlan 6 000 € http://www.soundplan.com SP, Sweden
6 PlantNoise 2100 $ http://www.soeh.ubc.ca/hodgson_research/plantnoise.htm UBC, Canada
7 ClassTalk 652 $ http://www.soeh.ubc.ca/hodgson_research/classtalk.htm UBC, Canada
8 RAMSETE 2 3 750 € http://www.ramsete.com Parma YO, Italy
9 Ease 4.0 ? http://www.olsonsound.com/EASE/index.html Olson Sound Design
10 VisualEars 49 € http://www.kbacoustics.com/visualears/ KB Acoustics
11 CARA 2.2 49-75 € http://www.cara.de/ Scanteknik, Finland
12 NEMPEE 155 € http://www.nempee.com/AcouSoft.htm Nempee, India
13 Acoustic-X 286 € http://www.pilchner-schoustal.com/old/acoustic-x Pilchner Schoustal
14 Acousalle 245 € http://lesowww.epfl.ch/anglais/Leso_a_frame_sof.html LESO-PB, Switzerland
15 Sopran Not in market …
16 RaySoft ? http://www.oe.com.tw/01/prode2.htm 01dB, France
17 dBWorks in progress …
18 MagiCAD 2600-3500€ http://www.progman.fi Progman, Finland
 A2-2

Table 2.3: General information on commercial acoustic software. Commercial name. Applied method.
Application field. Calculated acoustic parameters.

Name Method Application Acuoustic parameters

1 ODEON 6.5 Image + Ray-tracing Auditorium, industrial halls SPL, RT, EDT, STI,…, Auralization
2 CATT v8 Image + Cone-tracing Auditorium, industrial halls SPL, RT, EDT, STI,…, Auralization
3 RayNoise Image + Cone-tracing Auditorium, industrial halls SPL, RT, STI,…, Auralization
4 Ulysses Image + Ray-tracing Loudspeaker arrangement SPL, RT, STI,…, Auralization
5 SoundPlan Image + Sabine diffuse field Industrial spaces SPL, RT
6 PlantNoise Empirical (Heerema & Hodgson) Industrial spaces SPL, RT, Auralization
7 ClassTalk Empirical (Heerema & Hodgson) Speech and classrooms RT, STI, Auralization
8 RAMSETE Pyramid-tracing Industrial spaces SPL, RT, Auralization
9 Ease 4.0 Image + Ray-tracing Loudspeaker arrangement SPL, RT, EDT, RASTI, Auralization
10 VisualEars Mode analysis Listening rooms SPL
11 CARA 2.2 Mode analysis + diffuse field Listening rooms SPL, RT, EDT, Auralization
12 NEMPEE Diffuse field theory General spaces RT
13 Acoustic-X Mode analysis + Ray-tracing Listening rooms SPL, RT
14 Acousalle Mode analysis + diffuse field Auditorium, industrial halls RT
15 Sopran Particle-tracing Auditorium, industrial halls SPL, RT, EDT,…
16 RaySoft Ray-tracing Industrial spaces SPL, RT
17 dBWorks Image + Ray-tracing Auditorium, industrial halls SPL, RT, EDT, STI,…, Auralization
18 MagiCAD Unknown (Sabine?) Ventilation noise SPL