Acoustical Parameters

Acoustical Instruments and Measurements
July 2015, Argentina
ACOUSTICAL PARAMETERS
FEDERICO DAMIS1, NAHUEL CACAVELOS2
Universidad Nacional de Tres de Febrero (UNTREF), Buenos Aires, Argentina.
fede_damis@hotmail.com1, fnahuelc@gmail.com2
Abstract . Objective acoustical parameters were measured at Teatro 25 de Mayo in Buenos
Aires. Methods of measurement were fully described, and results were analyzed in relation with
its design. Thus, the hall was acoustically characterized and solutions were proposed for
determined issues found in the present work.
1. INTRODUCTION
1.1 OBJECTIVE ACOUSTICAL
PARAMETERS
The parameters used for assessing the
acoustic quality of a room obviously
depend on its intended use. Whereas the
reverberation time and/or the sound level
reduction by distance from the source may
be sufficient in an industrial hall, a more
comprehensive set of parameters must be
used in e.g. concert halls. It is
acknowledged that the reverberation time
has an important role and there is sufficient
background experience on how long or
short it should be depending on the size of
the room and related to the type of the
performance room; theatre, room for music
performance
etc.
As
for
music
performance, the type of music will be a
vital factor [1]. A number of other
parameters that correlates well with the
subjective impression are based on data
calculated from measured impulse
responses in the room; these parameters
are described in the ISO 3382 Standard [2].
An example of a measured impulse
response is shown in Fig. 1.
Irrespective of the intended use of the

room, whether for speech or music, it is
important to design the room in such a way
as to give a balanced set (in time) of the
early reflections onto the audience area.
Reflections following the direct sound
within a time span of approximately 50
milliseconds will contribute to the strength
of the direct sound. A listener will not
perceive these reflections as a separate part
or as an echo, but will if a strong reflection
has a longer delay. This phenomenon is
called the precedence effect or Haas effect,
the latter name in recognition of one of the
many researchers on the phenomenon [3].
Added to the time arrival of the
reflections, it is important for rooms for
music performances to know where the
reflections are coming from. The
directional distribution is critical for the
listeners feeling of spaciousness of the
sound field, i.e. lateral reflections are just
as important as reflections from the
ceiling. Added to this fact, there has in the
last 20 years been a growing awareness
that diffuse reflections are also very
important, again for rooms for music
performances. We shall therefore give
some examples of these other objective
acoustic parameters used for larger halls,
how they are determined and, to a limited
extent, on the underlying subjective matter.
Figure 1. A measured impulse response in an

1800 m3 auditorium.

1.2 REVERBERATION TIME AND
EARLY DECAY TIME
The reverberation time T is defined as
the time required for the sound pressure
level in a room to decrease by 60 dB from
an initial level, i.e. the level before the
sound source is stopped. This is not
necessarily coincident with a listeners
feeling of reverberation and in ISO 3382
one will find that measurement of the early
decay time (EDT) is recommended as a
supplement
to
the
conventional
reverberation time. Both parameters are
determined from the decay curve, EDT
from the first 10 dB of decay, and T
normally from the 30 dB range between 5
and 35 dB below the initial level. Both
quantities are calculated as the time
necessary for a 60 dB decay having the
rate of decay in the ranges indicated.
Throughout the time a number of
methods have been used to determine the
decay curves and thereby the reverberation
time. A common method is to excite the
room by a source emitting band limited
stochastic noise, which is turned off after a
constant sound pressure level is reached.
For historical reasons, we shall mention
the so-called level recorders, a level versus
time writer, recording directly the sound
pressure level decay, where the eye could
fit a straight line. Later developments
included instruments giving out the decay
data digitally, enabling a line fit e.g. by the
method of least squares.
Modern methods based on deterministic
signals such as MLS (Maximum Length
Sequence) or SS (Sine Sweep), however,
are superior in the dynamic range achieved
in the measurements and may well
measure over a decay range of 60 dB or
more. It may be shown that the decay
curve is obtained by a backward or
reversed time integration of impulse
responses. Normally as we are interested in
the reverberation as a function of
frequency, the impulse response is filtered
in octave or one-third-octave bands before

performing this integration. The decay as a
function of time is then given by [4]:
( )
( )
( ) (
(1)
where p is the impulse response. Certainly,

this equation was also utilized when
analogue measuring equipment was used
by splitting the integral into two parts as
follows:
( )
( ) ( )
( )
( ) ( )
(2)
The upper limit of the integration poses

a problem as the background noise
unrelated to the source signal will be
integrated as well. Different techniques are
suggested to minimize the influence of
background noise. One method is to
estimate the background noise from the
later part of the impulse response,
thereafter compensating for the noise by
assuming that the energy decays
exponentially with the same decay rate as
the actual one at a level 1015 dB above
the background level.
Such a technique [5] is used
calculating the decay curves shown in Fig.
2. The impulse response shown in Fig. 1 is
filtered by a one-third-octave band of
center frequency 1000 Hz and the decay
curves are calculated with and without
being compensated for background noise.
In one set of curves, the level of the
background is equal to the one present at
the time of measurement. In the second set,
the background noise is artificially
increased to show that also in this case one
will obtain a decay curve having an
acceptable dynamic range. Ideally, all the
solid curves should be coincident but this
will only be the case if the decay rate is
everywhere the same.
( )
( )
) (4)
The relationship between C50and D50 is

then given by:
Figure 2. Decay curves based on filtering, onethird-octave band 1000 Hz and reverse time
integration of an impulse response.
1.3 OTHER PARAMETERS BASED

ON THE IMPULSE RESPONSE
A large number of parameters
suggested in the literature and applied over
the years are listed and commented on in
ISO 3382. These are all derived from
measured impulse responses.
The balance between the early and late
arriving sound energy, which concerns the
balance between the clarity (and
distinctness)
and
the
feeling
of
reverberation, is important for music as
well as for speech. Several parameters are
suggested to cover this matter in room
acoustics. The simplest ones deal with the
ratio of the total sound energy received in
the first 50 or 80 milliseconds to the rest of
the energy received. We have an early-tolate index
defined by:
( )
( )
(3)
where te is 50 ms for speech and 80 ms for

music. An early variant of this parameter
was D50, which is denoted definition in
line with the original German notion of
Deutlichkeit. The difference from the
above is that, instead of the late energy,
one is using the total energy received.
) (5)
making it unnecessary to measure both

parameters. By way of introduction, we
pointed out that the direction of sound
incidence was important for the feeling of
spaciousness. Of special importance are
the lateral reflections, which also
contribute to an impression of widening a
source or a source area. Several early
lateral energy measures are proposed, one
being the lateral energy fraction LF based
on measured impulse responses obtained
from an omnidirectional and a figure-ofeight pattern microphones. It is defined as:
( )
( )
(6)
where pL is the sound pressure obtained

with the figure-of-eight microphone. This
microphone is intended to be directed in
such a way that it responds predominantly
to sound arriving from the lateral
directions and is not significantly
influenced by the direct sound.
Because the directivity of a figure-ofeight microphone essentially has a cosine
pattern and the pressure is squared, the
resulting contribution from a given
reflection will vary with the square of the
cosine of the angle between the reflections
relative to the axis of maximum sensitivity
of the microphone. An alternative
parameter is LFC, where the contributions
will be a function of the cosine to this
angle. This parameter, which is believed to
be subjectively more accurate, is defined
by:
Hence:
3

|
( ) ( )|
(7)
( )
In addition to the parameters given

above, there are others related to our
binaural hearing, based on measurements
using an artificial or dummy head. These
so-called inter-aural cross correlation
measures are correlated to the subjective
quality of spatial impression. Harold
Marshall (1967) and Veneklasen and Hyde
(1969) identified the importance of
reflections that surround or envelop the
listener with reverberant sound coming
from the side. Later authors quantified
these effects using the interaural crosscorrelation coefficient (IACC). Strong side
reflections are generated by narrow
rectangular rooms and from surfaces
placed close to the listener. Not
surprisingly Andos (1985) subjective
preference studies showed a strong
correlation between the IACC and room
width.
The interaural cross-correlation is a
measure of the similarity of sound arriving
at two pointsin this case, the two ears of
the listener. Mathematically it is based on
the interaural cross-correlation function
defined as:
( )
( )
( )
)
( )

obtain a single number the maximum value
of Equation 6 is taken:
|
( )|
(9)
and is called the interaural crosscorrelation coefficient. The integration

time can be varied with different results.
For t1=0 and t2=1000 ms, the term is
designated IACCA. The early IACCE (0 to
80 ms) is a measure of the apparent source
width (ASW) and the late IACCL (80 to
1000 ms) is a measure of listener
envelopment (Beranek, 1996).
While the C50 and C80 parameter
defined above is related to understanding
the spoken message, there are two others
that serve to quantify more precisely the
degree of speech intelligibility.
In the mid-70s era, the Dutch researcher
VMA Peutz conducted an exhaustive work
from which established the formula for the
calculation of intelligibility. Using
statistical theory, Peutz concluded that the
value of% ALcons at any given point you
could simply determine from knowledge of
the reverberation time (RT) and sound
pressure levels of direct field Ld and the
reverberant field Lr at that point. The law
in question is presented below in the form
of graph.
(8)
where the L and R refer to the entrances to

the left and right ear canals. The maximum
possible value for IACFt is one, occurring
when both signals are the same. The
integration is done beginning from time t1
measured from the arrival of the direct
sound at one ear and ending at time t2,
which is selected arbitrarily depending on
the period of interest. The variable
accounts for the time difference between
the two ears and is varied over a range
from 1 to +1 ms from the first arrival. To
Figure 3. Correlation between %ALcons and

RT

For calculating Ld-Lr, the formula to use
is:

The 14 Fm modulation
considered are the following:
frequencies
Table 1. Modulation frequencies
(10)
where,
log = logarithm
Q = directivity factor of the sound source
in the direction of interest (Q = 2 in the
case of the human voice, considering the
front direction of the speaker)
R = Constant of the room (in m2) R.
r = distance from the point considered to
the sound source (in m)
Typically, the %ALcons is calculated in 2
kHz band, because it is the band maximum
contribution to speech intelligibility.
From the observation of the above figure it
follows:
As closer is set the receiver to the
sound source (LD-LR higher), lower
values of %ALcons will be found, ie,
greater intelligibility.
As decrease lower RT values, also
decrease the %ALcons, ie, greater
intelligibility.
The value of %ALcons increases as
the receiver moves away from the
source, to a distance r = 3.16 Dc. For
distances r> 3.16 Dc, equivalent to
(LD - LR) <-10 dB, the value of
%ALcons tends to be constant. This
means that, from this distance, the
intelligibility of speech is no longer
worsening.
The STI index, defined by Houtgast and
Steeneken, quantify the degree of speech
intelligibility
between
0
(zero
intelligibility)
and
1
(excellent
intelligibility). The STI is calculated from
the reduction of different modulation
indices "m" of the voice due to the
existence of reverberation and background
noise in a room.
Each of these frequencies produces a

modulation effect on seven octave bands
most representative voice whose center
frequencies are listed below:
Table 2. Center frequencies
Steeneken and Houtgast presented a

specific calculation method for this
parameter. However in this practice it is
used alternatively calculation system
Fourier transform, which is equivalent to
Steeneken Houtgast and, with the
exception that the modulation transfer
function MTF is calculated from the
transformed Fourier square module of the
impulse response h (t, F0), divided by the
energy of the signal:
(11)
Because there are 14 Fm values F0 and
7, the total number of values m (F0, Fm)
is: 14 x 7 = 98.
Then signal noise ratios are calculated.
(12)
The 98 values obtained are truncated so
that all of them are between 15 dB and -15
db. For each octave band, the average
value of the 14 apparent signal/noise ratios
5

is calculated according to the following
expression:
(13)
Apparent global average SNR is then
calculated
(14)
Finally we get the STI value as:
(15)

1.4 PARAMETERS BASED ON
RANDOM NOISE SIGNALS
1.4.1 G (strength)
The influence of the room on the
perceived loudness is another important
aspect of room acoustic. A relevant
measurement of this property is simply the
difference in dB between the level of a
continuous, calibrated sound source
measured in the room and the level the
same source generates at 10 m distance in
anechoic surrounding. This objective
measure called the (relative) strength G
can also be obtained from impulse
response recording from the ration
between the total energy of the impulse
response and the energy of the direct sound
whit the latter being recorded at a fixed
distance (10m) from the impulsive sound
source:
It can easily verify that STI values are

always between 0 and 1 because the values
of (S / N) ap is between -15 dB and +15
dB.
To obtaingood speech intelligibility the
following condition must be met:
(16)
It has been demonstrated that there is a
very good correlation between the values
of% ALcons and STI.
(17)
Here the upper integration limit in the
denominator tdir should be limited to the
duration of the direct sound pulse (which
in practice will depend on the bandwidth
selected). A distance different from 10 m
can be used, if a correction for the distance
attenuation is applied as well.
The expected value of G according to
diffuse field theory becomes a function of
T as well as of the room volume, V:
(18)
The subjective difference limen for G is
about 1.0 dB. The definition of G is
illustrated in the following figure:
Figure 4. Subjective valuation of STI and
%ALcons
Figure 5. Definition of G
1.4.2 DIRECT-TO-REVERBERANT
ENERGY RATIO (D/R)
Judgment of ego-centric distance to
nearby objects is an important human
sensory capability and is at times wholly and critically - dependent on auditory
input. Absolute sound energy at the
receiver is a function of intrinsic source
energy and source distance, both of which
may be time-varying, precluding the use of
energy alone as a cue to source-listener
distance. However, the combination of
energy received along the direct sourcelistener path with energy arriving
following reflections has potential as a
means of estimating source distance. The
direct-to-reverberant energy ratio (DRR)
has been suggested as part of the
mechanism for source distance judgments
in listeners [6]. Distance judgments are
more accurate in a reverberant space than
in an anechoic space, with small inter-test
variation in judgments in the same
environment.
Listeners
may
use
reverberation as an absolute distance cue
given that accurate distance judgments
were
obtained
at
first
stimulus
presentation. Zahorik [7] suggested that the
principal role of the DRR cue was to
provide absolute distance information
rather than support fine distance
discriminations and was poor as a relative
cue. Zahorik also suggested that DRR was
perceptually more salient than an intensity
cue, especially in a situation where prior
knowledge of natural speech level could
not be used due to other more variable and
complex acoustic information in the
surrounding environment.

The dependence of D/R on source
distance is caused by the fact that the
energy in the direct sound decays with
distance, while the energy of reverberation
is approximately constant throughout the
entire room. Conservation of energy
implies that direct sound level decreases by
6 dB for every doubling of the distance drs
between the source and receiver.
Therefore, D/R decreases by 6 dB for
every distance doubling, and we can write
[8]:
(
(19)
where rc is the critical distance of the

room, defined as the distance where D/R
equals 0 dB (equal energy in direct and
reverberant sounds).
Some recommended values
Chamber music Symphony
EDT
1.4s
2.2s
RT
1.5s
2.0 - 2.4s
C50
(+/-) 3 dB
C80
(+/-) 4 dB
STI y %Alcons
11,40%
LF
0.1 - 0.35
G
10 dB
3 dB
IACC
0.6
0.7
Table 3 Recommended values
2. MEASUREMENT PROCEDURE
AND POST PROCESSING
2.1 PERFECTING IMPULSE
RESPONSES
The source used has a pattern of almost
constant omnidirectional directivity. This
choice was made warning that the polar
characteristic of a source is not susceptible
of improvement made corrections with
post processing. We know that it is
possible to correct the measurements
obtained according to the impulse response
of the sound source used for the purpose of
repairing changes that could enter by the
source. This makes it possible to improve
7

the linearity in the transfer of a
measurement system. All this processing
was carried out in Audition 3.0 and Aurora
plugin. In this paper corrections are made
for that purpose.
Application of LSS and its inverse filter
under improved SNR.
Application of the inverse filter source
processing algorithm whit Kirkeby
Repeat 3 measurements in each position
to improve measurement uncertainty.
2.2 APPLICATION OF LOG SINE
SWEEP AND INVERSE FILTER
The sound stimulus for obtaining the
impulse response is performed using the
application of a further temporary LSS and
convolution with the inverse filter.
Figure 6 - LSS applied
2.3 APPLICATION OF INVERSE

FILTER
SOURCE
ALGORITHM
WHIT KIRKEBY PROCESSING
Parallel to the above measurements, the
measurement of the sound source was
performed at 2m distance as shown in the
following Figure.
By virtue of obtain only the direct sound

and eliminate possible reflections, a
truncation of the measured signal is
performed in the first 35 ms.
2.4 TRUNCATION OF DIRECT
SOUND Y KIRKEBY
It is important to clarify that the
microphone was placed on the ground so
the first reflective path was whit the side
wall. This had a length path of 12, 24
meters more than the direct sound path. By
virtue of obtain only the direct sound and
eliminate possible reflections, a truncation
of the measured signal is performed in the
first 35 ms.
This signal was then processed using
software Kirtkeby8 Aurora function,
performing a frequency filtering in the
range of temporal analysis of the selection
of the direct path. Thus, the convolution
with the inverse filter was made with each
of the measured signals.
While the improvement is observable to
the naked eye, the expensively waveform
is noticeably detectable to analyze their
spectrum. In the same is observed as the
application of the inverse filter processing
source Kirkeby, improve the timing of the
frequency content in the measured signals,
eliminating energy dispersion produced by
the source.
Reflected
sound 14.24 m
7,05 m
Source
Microphone
Direct
sound
2m
7,4 m
Figure 8 Inverse source filter whit Kirkeby

correction.
Figure 7. Direct sound and reflection paths

It should be clarified that this process
will be satisfactory for the correction of the
nonlinearity in the transfer of both the
sound source and the omnidirectional
microphone used for measurement. In the
case of measurements made with different
polar pattern microphones it is considered
that they have a linear transfer and said
correction only influence the improvement
of the sound source.
2.5 MICROPHONES
Most parameters were calculated out of
impulse responses taken at the theater. The
responses were taken by setting up an
omnidirectional sound source (an Outline
Globe Source Radiator and Subwoofer)
on the stage area and recording using four
Earthworks
M50
omnidirectional
microphones simultaneously. A total of
forty positions were used in the case of
these microphones: sixteen positions on
the ground floor, twelve positions on the
first floor and twelve positions on the
second floor as well. The sound source
reproduced three continuous logarithmic
sine sweeps (separated from each other
with four seconds of silence) with duration
of fifty seconds, and ranging in frequency
from 80 Hz to 15 kHz. LSS audio files
were generated with a 32 bits resolution
and 48 kHz of sampling frequency, using
Aurora 4.3 software. The recorded files
were then deconvolved using an ILSS
filter(Inverse Logarithmic Sine Sweep) and
an Inverse Kirkeby filter to improve the
impulse response. The sound source level
was set so that the signal would reach all
microphones with a decent amount of S/R
ratio (Sound-to-Noise ratio). For that
reason, sound source level was fixed, while
the preamp levels for each microphone was
adjusted at each position. The height of all
microphone positions was 1,2 m to
simulate the height of a subjects ears.
Besides, six anechoic samples were
recorded to compare and contrast the
variation of ACF and emin as a function of
seat position.

Binaural impulse responses were
measured as well, using a Kemar HATS
(Head and Torso Simulator), but at
different
positions
than
monaural
responses. On these same positions, a
Soundfield microphone (SPS200) was used
to record impulse responses for the
determination of Lateral Fraction. Both the
dummy head and the Soundfield
microphone were calibrated prior to the
measurements. This process was critical to
the Soundfield microphone accuracy, since
it was compulsory that all four
microphones in the array were matched in
phase and amplitude to obtain trustworthy
results, so a four-input audio interface was
used and calibration was made by means
of adjusting the preamp levels.
Sound level meters were used to
determine G (Strength) values. The source
emitted pseudo-random pink noise and
sound pressure levels were measured at
various positions. The G parameter
requires an anechoic measurement at 10 m
of the sound source, which was not
possibly at the place. However, ISO 3382
[2] recommends that this measurement can
be replaced by a measurement of the sound
pressure level at approximately 3 m and to
extrapolate a 10 m measurement
considering attenuation by spherical
waves. Sound pressure level measurements
were carried out considering positions
within an axis from the stage to the back of
the floors.
2.6
MICROPHONE POSITION
G AND S/R
Equipment:
Sound Level Meter SVANTEK
959
Speaker Outline
Signal: Pink random noise
Ground
floor
Ground
floor
Speaker
On axis
First column
Second column
Third column
Fourth column
Fifth column
6
5
First
floor
First
floor
Speaker
7
On axis
First column
Second column
Third column
Fourth column
Fifth column
8
Second
floor
Second
floor
Speaker
10
On axis
First column
Second column
Third column
Fourth column
Fifth column
12
11
Figure 10. Positions of Soundfield Microphone

and Kemar Dummy-Head
Figure 9. Sonometer positions
IACC, ACF, LF, LCF

Equipment:
Tascam US1641
SoundField SPS200
Kemar Dummy-Head
Computer
Speaker Outline
EDT, RT, Clarity, STI, %ALcons

Equipment:
Tascam US1641
Omnidirectional Earthworks M-50
Omnidirectional DPA
Computer
Speaker Outline
Signal: 6 different motifs

Motif 1
Michael
Motif 2
Soprano
Motif 3
Organ
Motif 4
Piano
Motif 5
Violin
Motif 6
Pulse
Signal: Log Sine Sweep
Table 4 Music motifs
10
15
Ground
floor

samples, using an ACF algorithm in
MATLAB software.
Speaker
16
5
9
10
11
7
6
3. RESULTS AND DISCUSSION
14
13
8
12
First
floor
Speaker
11 7
8
5
3 4 10
12
Second
floor
Speaker
6
9 10 11
12
3
7
5
8
Figure 10 Microphone positions
2.7
PARAMETER PROCESSING
Parameters denoting clarity (C7, C50,
C80), speech intelligibility (STI, %Alcons)
and reverberation time (T20, EDT) were
calculated in third-octave bands using
EASERA software. IACC was also
calculated in third-octave bands using
EASERA, but by means of the binaural
impulse responses taken by the dummy
head. LF and LFC were calculated by
analyzing the responses by the Soundfield
microphone. Since the Soundfield system
has an A-format output, it was converted to
B-format using a VST plugin. Then, only
two of the audio wave files (w and x
directions) present in the B-format were
considered to calculate LF and LFC in
third-octave bands using EASERA as well.
G was calculated using the sound pressure
level values and ACF was determined by
comparing the Auto-correlation Function
of anechoic recordings to the measured
3.1 RT AND EDT ANALYSIS

By analyzing the impulse responses at
various positions in the concert hall, T20
and EDT values were calculated in third
octave bands, using EASERA software.
The results were averaged for each floor
and then for the whole theater.
Frequency
100 Hz
125 Hz
160 Hz
200 Hz
250 Hz
315 Hz
400 Hz
500 Hz
630 Hz
800 Hz
1000 Hz
1250 Hz
1600 Hz
2000 Hz
2500 Hz
3150 Hz
4000 Hz
5000 Hz
6300 Hz
8000 Hz
10000 Hz
EDT
1,91
2,05
1,90
1,95
1,74
1,78
1,56
1,50
1,40
1,32
1,22
1,19
1,30
1,20
1,10
1,09
0,99
0,97
1,00
0,79
0,68
T20
2,57
2,11
2,03
1,82
1,75
1,63
1,48
1,44
1,45
1,43
1,36
1,27
1,21
1,18
1,14
1,06
0,97
0,91
0,79
0,64
0,51
Table 5. EDT and T20 ground floor values.
Figure 11. Ground floor T20 values.
11
Figure 12. Ground floor EDT values.
Figs. 9-10 present the values for the

ground floor, as well as their
corresponding expanded measurement
uncertainty. As it can be seen, the
uncertainty remains high at low
frequencies, which could represent the low
amount of diffusion in the enclosure. A
low amount of diffusion in this frequency
range would explain an uneven spatial
distribution of the reverberation time.
Frequency
100 Hz
125 Hz
160 Hz
200 Hz
250 Hz
315 Hz
400 Hz
500 Hz
630 Hz
800 Hz
1000 Hz
1250 Hz
1600 Hz
2000 Hz
2500 Hz
3150 Hz
4000 Hz
5000 Hz
6300 Hz
8000 Hz
10000 Hz
EDT
T20
2,37
2,23
1,75
2,05
1,48
1,32
1,31
1,45
1,48
1,29
1,20
1,16
1,21
1,05
0,97
0,93
0,91
0,91
0,94
0,78
0,69
2,71
2,14
1,98
1,75
1,63
1,60
1,58
1,42
1,37
1,31
1,29
1,23
1,22
1,16
1,09
1,05
0,96
0,90
0,77
0,62
0,47
Table 4. EDT and T20 first floor values.
Figure 11. First floor T20 values.
Figure 12. First floor EDT values.
In the case of the ground floor, sixteen

positions were evaluated, all of them on
the right side of the floor. Since the theater
was symmetrical on the horizontal plane,
this distribution of microphones was
enough to acoustically characterize the
hall. On the first and second floor, only
twelve positions were analyzed (also
taking into account the symmetry) because
the amount of seats was reduced in these
zones.
Frequency
100 Hz
125 Hz
160 Hz
200 Hz
250 Hz
315 Hz
400 Hz
500 Hz
630 Hz
800 Hz
1000 Hz
1250 Hz
1600 Hz
2000 Hz
EDT
T20
2,07
2,16
1,81
1,65
1,44
1,40
1,51
1,41
1,34
1,17
1,13
1,08
1,15
1,10
2,21
1,91
1,91
1,81
1,66
1,58
1,51
1,33
1,33
1,37
1,28
1,24
1,22
1,19
12

2500 Hz
3150 Hz
4000 Hz
5000 Hz
6300 Hz
8000 Hz
10000 Hz
1,00
0,98
0,91
0,91
0,98
0,83
0,72
1,11
1,07
0,98
0,89
0,79
0,63
0,47
Table 5. EDT and T20 second floor values.

400 Hz
500 Hz
630 Hz
800 Hz
1000 Hz
1250 Hz
1600 Hz
2000 Hz
2500 Hz
3150 Hz
4000 Hz
5000 Hz
6300 Hz
8000 Hz
10000 Hz
1,47
1,46
1,41
1,26
1,19
1,15
1,23
1,13
1,03
1,01
0,94
0,93
0,98
0,80
0,69
1,52
1,40
1,39
1,38
1,32
1,25
1,22
1,18
1,12
1,06
0,97
0,90
0,78
0,63
0,49
Table 6. EDT and T20 total averaged values.

Figure 13. Second floor T20 values.
Figure 15. Total averaged T20 values.

Figure 14. Second floor EDT values.
By comparing the three tables it can be

seen that the values obtained are very
similar in most cases, except for the low
frequency range where the uncertainty for
all measurements is higher. The mean
reverberation time is insignificantly higher
in the case of the ground floor, and it
decreases as the floors rise by a very small
amount. Improved speech intelligibility
can be expected at higher floors for this
reason.
Frequency
100 Hz
125 Hz
160 Hz
200 Hz
250 Hz
315 Hz
EDT
T20
2,10
2,14
1,83
1,89
1,57
1,53
2,51
2,06
1,98
1,80
1,69
1,61
Figure 16. Total averaged EDT values.
Total averaged values for all three floors

are shown in Table 6. The total mean value
for T20 was 1,34 while the total mean value
for EDT was 1,32. These values could be
optimal for a theater, and speech-oriented
activities, but should be somehow
preferably higher in the case of music
being played at the venue. The low relative
amount of reverberation time in this case is
highly determined by the total enclosure
volume, since it is a small venue with a
13
low amount of seating capacity, even

though it consists of three floor
3.2 C7, C50 AND C80 ANALYSIS
It is necessary to clarify first as to be
understood clearly values a room. If the
clarity is too low, the fast parts of the
music are not "readable" anymore. C80 is a
function of both the architectural and the
stage set design.
If there is no reverberation in a dead
room, the music will be very clear and C80
will have a large positive value. If the
reverberation is large, the music will be
unclear and C80 will have a relatively high
negative value. C80 becomes 0 dB, if the
early and the reverberant sound are equal.
For orchestral music a C80 of 0dB to 4dB is often preferred, but for rehearsals
often conductors express satisfaction about
a C80 of 1dB to 5dB, because every detail
can be heard. For singers, all values of
clarity between +1 and +5 seem
acceptable. C80 should be generally in the
range of -4dB and +4dB. As the same way
the factor C50 is used for speech analysis.
The results of clarity of the word
analysis for different times and for
different areas of the audience, is shown in
the graph below. The analysis is done by
thirds of octave band. Uncertainty levels
are clarified, having taken three
measurements per position.
Figure 18. C50 Ground floor.
It can be seen that clarity is

significantly higher for high frequencies,
taking fairly constant values for
frequencies above 800Hz, which is also
reflected in reduced uncertainty. This is
largely due to the existence of resonance
modes in the room and the variation of RT.
For such low frequency values, the clarity
is very low to the floor but if kept in
respectable values for C80, finding very
high levels of clarity even for high
frequencies. Similarly it happens for values
of between 8k and 10k where clarity is
greatly increased. While low frequency
values are high uncertainty, all respond to
a change in expected for these situations. It
is obvious that the clarity for the C80 used
for musical reasons are higher that C50
used for word. The first floor balcony will
be presented below.
14
Figure 20. C7 First floor.
Figure 23. C7 Second floor.
In this case the clarity values not differ

heavily of was found on the ground floor.
This
demarcates
that
acoustics
characteristics of the rooms maintains the
clarity even on the top floor. However it is
possible to analyze some particular areas
where clarity was affected. It can be seen
for greater dispersal in C50 and C80
analysis. This as seen, it is not determined
by an increase in RT, but considering that
it is more unstable in closed areas of the
side balconies, which presents a
considerable width. Subsequently the top
floor was analyzed.
In this case it is possible to note that in

all graphs the clarity values are decreased.
In the RT analysis, the values decreased in
the upper floors, so it would be expected
that clarity grows since it is directly related
to the RT. But this event is mainly due to
the decreased of the ratio between direct
sound and reverberant sound, which we
discuss later whit the parameter D/R.
Anyway we can see that the C80 values are
maintained even within the permissible
limits, maintaining higher values for high
frequencies, as is expected. Finally total
results of clearly is shown.
15
Figure 26. C7 Total values.
As it can be seen, the final clarity

values correspond with those expected for
this type of rooms. Increasing the clarity
values for high frequencies and with
relatively low uncertainty. A striking factor
is the high uncertainty in the C7 for high
frequencies, which increase significantly
with values above 800 cycles.
3.3 STI - % ALCONS ANALYSIS
Speech intelligibility analysis was made
for the same positions as with the TR/EDT
analysis. This sums up a total of forty
positions in which STI (Speech
Transmission Index) and %Alcons
(Percentage
Articulation
Loss
of
Consonants) were measured.

Floor
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
First
First
First
First
First
First
First
First
First
First
First
First
Second
Second
Second
Second
Second
Second
Second
Second
Second
Second
Second
Second
Position
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
STI
0,58
0,57
0,61
0,60
0,60
0,58
0,61
0,65
0,60
0,60
0,58
0,59
0,54
0,59
0,53
0,59
0,56
0,59
0,57
0,59
0,57
0,59
0,62
0,63
0,60
0,61
0,64
0,60
0,54
0,53
0,60
0,63
0,65
0,66
0,65
0,64
0,63
0,62
0,57
0,56
%Alcons
7,29
7,93
6,28
6,62
6,65
7,38
6,35
5,03
6,70
6,64
7,50
6,85
9,35
6,84
9,58
7,01
8,35
6,90
7,66
7,12
7,83
6,85
5,80
5,71
6,73
6,37
5,44
6,44
9,22
9,85
6,59
5,66
4,98
4,82
4,99
5,38
5,57
5,88
7,78
8,39
Table 6. Speech intelligibility values as a

function of position in the room.
16

Table 6 shows the different values
obtained for each position. The values for
STI range between 0,53 and 0,66 while
%Alcons fluctuates between 4,81 % and
9,84 %. It is possible to interpret the STI
values in a simple way in terms of a
proposal made by Barnett. [5] According
to his scale (shown in Fig. 29) the values
obtained are relatively fair for nearly all
positions measured.

This has direct relation with the T20 values
obtained for these positions, since higher
T20 values were obtained over the ground
floor. Nevertheless, %Alcons is not as
evenly distributed on the ground floor as
STI. Variations of around 5% are present
between positions, mainly due to early
reflections which could contribute to the
masking of consonants.
Figure 29. Barnett qualification of STI.
As regards %Alcons, it is considered

that values over 10 % represent a poor
intelligibility (higher values represent a
loss in the perception of consonants, and
thus, lower intelligibility). Therefore,
values obtained for %Alcons are within a
fair range but with the exception of some
positions near critical values.
Figure 30. STI distribution over seats for the

ground floor.
Figure 31. %Alcons distribution over seats for

the ground floor.

first floor.

the first floor.
On the first floor, the opposite situation

is given: the STI values show great
variation while %Alcons values display an
even analysis curve and better spatial
distribution. STI values remain relatively
high (in relation to the mean values) for
determined positions while lower for the
rest of them (see Fig. 32). The positions
which show greater values for STI
correspond to the lower values of %Alcons
which is reasonably expected.
Figs 30-35 show the intelligibility

values for each floor. On the ground floor,
it can be noted that STI values are the
lowest values in the theater but are evenly
distributed among the seating positions.
17

second floor.

the second floor.
As stated above, since T20 values were

the lowest for the second floor positions,
STI was the highest in this case. Its
distribution as well, is the most optimal as
it can be seen on Fig. 34. Again, the
%Alcons values strongly correlates with
the STI ones, being both inversely
proportional.
Floor
Ground
First
Second
Total
STI
0,59
0,60
0,61
0,60
%Alcons
7,12
6,77
6,59
6,86
Table 7. Total averaged speech intelligibility

values.

doing the same with the seats. Considering
the reverberation time is higher at the
ground floor, which is the place that has
the majority of seats, it could be assumed
that seat absorption is not enough to
provide high intelligibility. The setup of an
electroacoustic amplification system can
be considered an option, but it would
bypass the acoustic effect imposed by the
theater, which may not be desirable and
could lower the subjective preference since
listeners would not be hearing the acoustic
response of the room. Having said that,
speech intelligibility could greatly benefit
from the installment of speakers on the
stage and on the balconies.
3.4 LATERAL FRACTION EARLY
AND
LATE.
LATERAL
FRACTION COSINE ANALYSIS
The results for lateral fraction are
presented below.
Normal values should be between 0.1 and
0.35, but as is possible to seen in the
graphs, the values reach de 3 and does not
became slower than 1 for LF. There are
better results for LFC, but anyway still
being out of the normal scale.
Since the resulting values defer much from
the normal scaling parameters for this
room, it was felt that there were process
failures of measurement or post processing
with the plugin which converts format A to
format B. That is why it was resolved not
to make assumptions about the results
obtained.
Total averaged values can be seen on

Table 7. Considering the venue is mostly
used for stage plays, STI values should be
higher in order to assure optimal
comprehension of the actors speech. A
practical alternative to accomplish this,
would be to introduce acoustical absorbent
panels on the lateral and back walls. A
most complex solution could also be to
change the carpet material for another one
with a higher absorption coefficient, or
18

frequency spectrum. However during the
measurement measurement system moved
several times, what could have caused the
change in the parameters of the interface.
3.5 G
(STREGHT)
ANALYSIS.
DISTRIBUTION
OF
SOUND
PRESURE LEVEL
G parameter indicates the relation
between the sound pressure levels
compared to fix distance to the source. In
this case, the values were normalized with
the lowest noise level.
This will intimately related with the
reverberation time in the room. The
following graphs show the parameter for
measuring different orientations (in the
axis and in a different columns separated
from the axis) is displayed. The analysis
contains three different settings: Midrange
(500Hz to 1kHz), Low frequency (125-250
Hz) and the total values.
Figure 36. G factor in axis normalized (total,

low and mid)
Figure 37. LF and LFC, total early and late
values
The causes of error in the measurement

can be given by the difficulty presented at
the time of calibration. It is very important
that the gains of the sound card to be
adjusted accurately. In this case I did not
count with digital potentiometers as
generally used for such measurements.
Instead, a reference channel which was
inserted at all entrances of the interface
was used. Subsequently, a transfer curve
was performed to see the differences
between the two channels, both gain as
Figure 37. G factor first column normalized

(total, low and mid)
19
Figure 38. G factor in second column

normalized (total, low and mid)

axis, while for the side columns the effect
decreases. This is so because the energy
product of the reverberation time of the
hall takes an important place in this regard,
avoiding the fall of the sound level.
It is important to remember that this
parameter does not indicate the level of
auditory perception that can have an
spectator, but only a reference sound
pressure level present in the room after
stimulate with continuous pink noise.
Another striking aspect is that all the
graphics, the shape has less level for lower
frequencies than mid frequency, this is
because the porous absorbent materials
commonly attack resistance more often
this frequency range.
3.6 IACC AND ACF ANALYSIS
Figure 39. G factor in third column normalized

3.6.1 ACF
An analysis of the behavior of te for
different positions in the room with 6 types
of musical programs performed. Thus
different analyzes were carried out in order
to understand the spatial behavior of the
room.
Figure 40. G fourth column normalized (total,

low and mid)
Figure 41. G factor fifth column normalized

As can be observed, in all cases G is

directly related to the distance, such that
when the distance increase the parameter G
falls. This phenomenon becomes more
marked for measurements made on the
Figure 42 - te average
As we can see the motif 3 (Organ) has
the highest average for all positions of the
room and motif 1 (Michael Jackson) as the
20

lower motif. This is mainly due to an
intrinsic characteristic of the musical
program analyzed and its temporal
variation. We can also observe that the
position directly modifies the te average for
all kinds of motifs.
This effective length will bring with it a
change of Loudness in the listener so as the
motif that plays in the room will get
different sensations of loudness.

Either way it can identified that in some
particular situations the te minimum
presented in the room can be less than
anechoic audio, although it is not the most
probably, it should be take into account the
spatial variation circumstantially modifies
all acoustic characteristics and thus the
signal dynamics. It is curious to note that
this property does not hold for all
positions, but the position in which this
phenomenon occurs is also a function of
the musical program.
Figure 43 - te min
On the other hand it can see that there are
no major changes to the te min beyond the
different positions and different motifs,
saving the case of motif 5 position 9. This
becomes more obvious for 7, 8, 9, 10
positions in 1, 2, 3 motif, where there is
almost no variation between them, and its
total variation is less than 10ms.
te min variation depending on the room
Then various graphics corresponding to
the difference between te measured in the
room and te inherent to each music
program, may then detect the color
introduced into the room sound for each
type of program are presented.
It is possible to see how practically in
all positions expected the te min is greater
for measured in the room that the inherent
anechoic audio.
Figure 44 - te min difference between

measured in the room and the inherent to
each music program.
On the other hand is possible to identify

the musical program of motif 1, which will
suffer major alteration to be played in the
room, because the auditory perception will
be more influenced by the chamber music
program as for the same musical program.
Instead the musical motif program 3 does
not suffer major changes in the t min, so
21

that perception if it is controlled by the
musical program.
3.6.2 IACC
To understand the laterality of the room
response and psychoacoustic parameters a
study of signals obtained with a Kemar
head with standard HRTF filter was
performed.
In order to analyze the involvement of
both the source and the feeling of
involvement in the room, the results for the
IACC early and beats are presented.

of the sound direction. Mostly this is given
by the constancy and low variability signal
as shown in the graphs above of te min.
On the other hand it can be seen a
decrease in the IACC for some specific
positions (Position 5 and 6), they warn that
in that area there are a lot of diffusion in
the field, which transduce a low correlation
between the inter-aural cross correlation
signals. It is hoped that these positions
have a high sense of the apparent source
width.
An analysis is shown to IACC late is
showed in the following graph.
Figure 45 - IACC Early.

Figure 46 - IACC late.
In this graph the early IACC parameter

is shown in different positions in the room
for 5 of the music motifs presented above.
This parameter remains an interesting
dependence on the ASW (apparent source
width).
It can be notice that the case Motif 3
exceeds most of the samples analyzed,
except for overcoming the motif 5 in a few
moments. That is why you can clearly infer
about the dependence on the musical
program used, being the motif 5 and motif
3 the most difficult to interpret the origin
In the above graph we can detect the

prevalence of motif 3 whit high values in
all positions. This is related to the musical
program used (organ), where monotony
does not allow discretize easily the direct
sound to the reflective sound. The
IACClate has great relationship with the
psychoacoustic parameter LEV, which
refers to the sound produced by
envelopment room. So if a high magnitude
IACClate is presented, the result will be
difficulty to identify the direct sound from
22
the reverberant field as the same way that

will be difficult to identify the
envelopment produced by the acoustics of
the room. If is easier to notice changes in
the Motif 1 and 2, which have a high te,
giving the possibility of energy discretize
late reflections of early and that is why we
find lower values of IACC Late.
It can be notice that for certain
positions there are a tendency to decrease
or increase the IACC. At position 5, a
decrease for different musical programs
was noted. This refers to that in this place
the inter-aural cross correlation is
aggravated by some reason. This may be
produced by scattering surfaces that
provide or by proximity to the stage.
IACC late percentile 1%.
Finally 1% and 10% percentile values are

presented in graphs.
Figure 50. IACC late percentile 10%
Figure 51 IACC late percentile 10%.

Figure 47. IACC early percentile 1%
Figure 52. IACC late percentile 10%

Figure 48. IACC early percentile 10%
Figure xx Percentile 1 and 10 for

IACC early and IACC late
3.7 D/R ANALYSIS
Direct-to-reverberant
energy
ratio
(D/R) measurements were made at the
23

venue in order to estimate the influence of
the reverberant field in the perceived
sound. Table X shows the values obtained
for each position as well as the (sub)total
averages.
Floor
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
Ground
First
First
First
First
First
First
First
First
First
First
First
First
First
Second
Second
Second
Position
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
Total
17
18
19
20
21
22
23
24
25
26
27
28
Total
29
30
31
DRR
-7,1
-3,6
-4,7
-0,7
-5,1
-3,2
-3,4
-1,8
-2,3
-3,8
-4,4
-5,2
-4
-6,1
-1,6
-0,3
-3,58
-4,5
-2
-1,4
-1,4
-2,5
-7,3
-4,3
-3,4
-6,7
-6,3
-1,9
-7,2
-4,08
-5,2
-6,2
-1,1
Second
Second
Second
Second
Second
Second
Second
Second
Second
Second
Average
32
33
34
35
36
37
38
39
40
Total
Total
-6,1
-1,9
-3,8
-2,5
0,5
-12
-5,4
-2,8
-5,8
-4,36
-3,96
Table 8. Measured D/R ratio.
Positions whose value approximates 0

dB can be interpreted as positions at
critical distance from the sound source,
that is, farther from that the reverberant
field will start to influence the total sound
field. As it can be seen, only a few
positions are located at critical distance,
and most of them display a partial amount
of direct sound only, with an average of -4
dB of D/R ratio approximately.
Figure 53. D/R ratio at the ground floor for

all seats.
Figure 54. D/R ratio at the first floor for all

seats.
24
Figure 55. D/R ratio at the second floor for

all seats.
On Figs 53-55 the distribution of D/R

ratio for all floors is shown. This parameter
is heavily influenced by position, since a
lower distance to the source allows for an
increased direct sound energy. Therefore,
its spatial distribution is highly variable,
but it is worth noting that higher floors
show higher dispersion of values. This
parameter is relevant to the perceived
distance to a sound source, and this can
enhance the listening experience by adding
depth to the sound perceived. In an
orchestra, for example, where every
instrument is at a determined distance from
the subject, the D/R ratio of the room
could naturally mix the sound in terms of
depth. Ergo, D/R ratios in concert halls
could take up values from -6 to -12 dB, but
as stated below, since this particular hall is
more speech-oriented, the actual values
given by the present measurements
correspond to reasonable ratios. Positions
with a lower D/R ratio would greatly
benefit from musical performances within
the theater, and vice versa in the case of
plays.
4. CONCLUSION
The work of fully characterizing a
theater/concert hall can be a very complex
task and also a very time demanding one.
The amount of parameters needed in order
to describe the acoustic behavior of such a
room can be very high, and therefore, leads
to a great amount of measurements and
post-processing of information. Also, the
amount of systematic errors can grow very
fast, because of the nature of the

measurements. In view of the fact that
many people are needed so as to carry out
the assessment of acoustical parameters
easily, each one of them is a source of
potential
systematic
errors
(since
measurements require a certain degree of
silence), and the proper organization of
tasks is a key to successfully analyze the
room. The measurement pre-evaluation
and distribution of responsibilities within
the theater/concert hall is therefore a
crucial element to maximize time and
minimize errors.
5. REFERENCES
[1] Kuttruff, H. (1998). Sound in
enclosures. Encyclopedia of Acoustics,
Volume Three, 1101-1114.
[2] ISO 3382. Acoustics - Measurement of
room acoustic parameters.
[3] Haas, H. (1951). Uber den Einuss
eines Einfachechos auf die Horsamkeit von
Sprache. Acustica, 1, 4958.
[4] Vigran, T. E. (2008). Building
acoustics. CRC Press. [5] Lundeby et al.
(1995)
[5] Barnett, P. W. and Knight, R.D.
(1995). The Common Intelligibility Scale,
Proc. I.O.A. Vol 17, part 7.
[6] Lu, Y. C., & Cooke, M. (2010).
Binaural estimation of sound source
distance via the direct-to-reverberant
energy ratio for static and moving
sources. Audio, Speech, and Language
Processing, IEEE Transactions on, 18(7),
1793-1805.
[7] P. Zahorik. Direct-to-reverberant
energy ratio sensitivity. J Acoust Soc Am,
vol. 112, no. 5, pp. 2110-2117, Nov., 2002.
25
[8] Larsen, E., Iyer, N., Lansing, C. R., &

Feng, A. S. (2008). On the minimum
audible difference in direct-to-reverberant
energy ratio. The Journal of the Acoustical
Society of America, 124(1), 450-461.
[9] Barron M. Objective (2006)
Assessment of concert hall acoustic.
Proceedings of the Institute of Acoustics.
Vol 28.
[10] Leo L. Beranek (2010) Listener
Envelopment LEV, Strength G and
Reverberation Time RT in Concert Halls.
International congress of Acoustic.
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Acoustical Instruments and Measurements

July 2015, Argentina

Irrespective of the intended use of the

Figure 1. A measured impulse response in an

Acoustical Instruments and Measurements

July 2015, Argentina

where p is the impulse response. Certainly,

The upper limit of the integration poses

Acoustical Instruments and Measurements

July 2015, Argentina

The relationship between C50and D50 is

1.3 OTHER PARAMETERS BASED

where te is 50 ms for speech and 80 ms for

making it unnecessary to measure both

where pL is the sound pressure obtained

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In addition to the parameters given

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and is called the interaural crosscorrelation coefficient. The integration

where the L and R refer to the entrances to

Figure 3. Correlation between %ALcons and

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Table 1. Modulation frequencies

Each of these frequencies produces a

Table 2. Center frequencies

Steeneken and Houtgast presented a

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It can easily verify that STI values are

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where rc is the critical distance of the

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Figure 6 - LSS applied

2.3 APPLICATION OF INVERSE

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By virtue of obtain only the direct sound

Figure 8 Inverse source filter whit Kirkeby

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Figure 10. Positions of Soundfield Microphone

IACC, ACF, LF, LCF

EDT, RT, Clarity, STI, %ALcons

Signal: 6 different motifs

Signal: Log Sine Sweep

Table 4 Music motifs

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3. RESULTS AND DISCUSSION

Figure 10 Microphone positions

3.1 RT AND EDT ANALYSIS

Table 5. EDT and T20 ground floor values.

Figure 11. Ground floor T20 values.

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Figure 12. Ground floor EDT values.

Figs. 9-10 present the values for the

Table 4. EDT and T20 first floor values.

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Figure 11. First floor T20 values.

Figure 12. First floor EDT values.

In the case of the ground floor, sixteen