Acoustical Parameters

Acoustical Instruments and Measurements July 2015, Argentina

1

ACOUSTICAL PARAMETERS

FEDERICO DAMIS1, NAHUEL CACAVELOS

2

Universidad Nacional de Tres de Febrero (UNTREF), Buenos Aires, Argentina.

[email protected], [email protected]

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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.

Figure 1. A measured impulse response in an

1800 m3 auditorium.

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.


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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.


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Figure 2. Decay curves based on filtering, one-

third-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-to-

late 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.

Hence:

( ( )

( )

) (4)

The relationship between C50and D50 is

then given by:

(

) (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-of-

eight 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-of-

eight 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:


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| ( ) ( )|

( )

(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 cross-

correlation 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:

( ) ( ) ( )

[ ( )

( )

] (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

obtain a single number the maximum value

of Equation 6 is taken:

| ( )| (9)

and is called the interaural cross-

correlation 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.

Figure 3. Correlation between %ALcons and

RT


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For calculating Ld-Lr, the formula to use

is:

(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)


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is calculated according to the following

expression:

(13)

Apparent global average SNR is then

calculated

(14)

Finally we get the STI value as:

(15)

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.

Figure 4. Subjective valuation of STI and

%ALcons

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:

(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:


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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 source-

listener 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


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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.

Figure 7. Direct sound and reflection paths

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.

Figure 8 Inverse source filter whit Kirkeby

correction.

Direct sound

2 m

Microphone

7,05 m

7,4 m

Reflected sound 14.24 m

Source


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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


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Figure 9. Sonometer positions

IACC, ACF, LF, LCF Equipment:

Tascam US1641

SoundField SPS200

Kemar Dummy-Head

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

Table 4 Music motifs

Figure 10. Positions of Soundfield Microphone

and Kemar Dummy-Head

EDT, RT, Clarity, STI, %ALcons

Equipment:

Tascam US1641

Omnidirectional Earthworks M-50

Omnidirectional DPA

Computer

Speaker Outline

Signal: Log Sine Sweep

Speaker

Ground floor

On axis First column Second column Third column Fourth column Fifth column

Speaker

First floor


Ground floor

1 2

3 4 5

6

First floor

7

8 9

Second floor

10

12 11

Speaker

Second floor



11

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

samples, using an ACF algorithm in

MATLAB software.

3. RESULTS AND DISCUSSION

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 EDT T20

100 Hz 1,91 2,57

125 Hz 2,05 2,11

160 Hz 1,90 2,03

200 Hz 1,95 1,82

250 Hz 1,74 1,75

315 Hz 1,78 1,63

400 Hz 1,56 1,48

500 Hz 1,50 1,44

630 Hz 1,40 1,45

800 Hz 1,32 1,43

1000 Hz 1,22 1,36

1250 Hz 1,19 1,27

1600 Hz 1,30 1,21

2000 Hz 1,20 1,18

2500 Hz 1,10 1,14

3150 Hz 1,09 1,06

4000 Hz 0,99 0,97

5000 Hz 0,97 0,91

6300 Hz 1,00 0,79

8000 Hz 0,79 0,64

10000 Hz 0,68 0,51 Table 5. EDT and T20 ground floor values.

Figure 11. Ground floor T20 values.

Ground floor

1

2

3

5

7

9

10

11

12

13

14

15

16

4

6

8

Speaker

5

7

8

9

10 12

1 2

3 4

6

11

First floor

Speaker

5

7

8

9 10 12

1

2 3

4

6 11

Second floor

Speaker


12

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 EDT T20

100 Hz 2,37 2,71

125 Hz 2,23 2,14

160 Hz 1,75 1,98

200 Hz 2,05 1,75

250 Hz 1,48 1,63

315 Hz 1,32 1,60

400 Hz 1,31 1,58

500 Hz 1,45 1,42

630 Hz 1,48 1,37

800 Hz 1,29 1,31

1000 Hz 1,20 1,29

1250 Hz 1,16 1,23

1600 Hz 1,21 1,22

2000 Hz 1,05 1,16

2500 Hz 0,97 1,09

3150 Hz 0,93 1,05

4000 Hz 0,91 0,96

5000 Hz 0,91 0,90

6300 Hz 0,94 0,77

8000 Hz 0,78 0,62

10000 Hz 0,69 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 EDT T20

100 Hz 2,07 2,21

125 Hz 2,16 1,91

160 Hz 1,81 1,91

200 Hz 1,65 1,81

250 Hz 1,44 1,66

315 Hz 1,40 1,58

400 Hz 1,51 1,51

500 Hz 1,41 1,33

630 Hz 1,34 1,33

800 Hz 1,17 1,37

1000 Hz 1,13 1,28

1250 Hz 1,08 1,24

1600 Hz 1,15 1,22

2000 Hz 1,10 1,19


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2500 Hz 1,00 1,11

3150 Hz 0,98 1,07

4000 Hz 0,91 0,98

5000 Hz 0,91 0,89

6300 Hz 0,98 0,79

8000 Hz 0,83 0,63

10000 Hz 0,72 0,47 Table 5. EDT and T20 second floor values.

Figure 13. Second floor 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 EDT T20

100 Hz 2,10 2,51

125 Hz 2,14 2,06

160 Hz 1,83 1,98

200 Hz 1,89 1,80

250 Hz 1,57 1,69

315 Hz 1,53 1,61

400 Hz 1,47 1,52

500 Hz 1,46 1,40

630 Hz 1,41 1,39

800 Hz 1,26 1,38

1000 Hz 1,19 1,32

1250 Hz 1,15 1,25

1600 Hz 1,23 1,22

2000 Hz 1,13 1,18

2500 Hz 1,03 1,12

3150 Hz 1,01 1,06

4000 Hz 0,94 0,97

5000 Hz 0,93 0,90

6300 Hz 0,98 0,78

8000 Hz 0,80 0,63

10000 Hz 0,69 0,49 Table 6. EDT and T20 total averaged values.

Figure 15. Total averaged T20 values.

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


14

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 17. C7 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.


15

Figure 20. C7 First 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.

Figure 23. C7 Second floor.



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.


16

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 Position STI %Alcons

Ground 1 0,58 7,29

Ground 2 0,57 7,93

Ground 3 0,61 6,28

Ground 4 0,60 6,62

Ground 5 0,60 6,65

Ground 6 0,58 7,38

Ground 7 0,61 6,35

Ground 8 0,65 5,03

Ground 9 0,60 6,70

Ground 10 0,60 6,64

Ground 11 0,58 7,50

Ground 12 0,59 6,85

Ground 13 0,54 9,35

Ground 14 0,59 6,84

Ground 15 0,53 9,58

Ground 16 0,59 7,01

First 17 0,56 8,35

First 18 0,59 6,90

First 19 0,57 7,66

First 20 0,59 7,12

First 21 0,57 7,83

First 22 0,59 6,85

First 23 0,62 5,80

First 24 0,63 5,71

First 25 0,60 6,73

First 26 0,61 6,37

First 27 0,64 5,44

First 28 0,60 6,44

Second 29 0,54 9,22

Second 30 0,53 9,85

Second 31 0,60 6,59

Second 32 0,63 5,66

Second 33 0,65 4,98

Second 34 0,66 4,82

Second 35 0,65 4,99

Second 36 0,64 5,38

Second 37 0,63 5,57

Second 38 0,62 5,88

Second 39 0,57 7,78

Second 40 0,56 8,39

Table 6. Speech intelligibility values as a

function of position in the room.


17

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.

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.

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.

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.


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.


18


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 STI %Alcons

Ground 0,59 7,12

First 0,60 6,77

Second 0,61 6,59

Total 0,60 6,86 Table 7. Total averaged speech intelligibility

values.

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

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.


19

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

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. G factor first column normalized

(total, low and mid)


20

Figure 38. G factor in second column

normalized (total, low and mid)

Figure 39. G factor in third column normalized


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

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

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 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


21

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.

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.

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 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


22

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.

Figure 45 - IACC Early.

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

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 46 - IACC late.

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


23

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.

Finally 1% and 10% percentile values are

presented in graphs.

Figure 47. IACC early percentile 1%

Figure 48. IACC early percentile 10%

IACC late percentile 1%.

Figure 50. IACC late percentile 10%

Figure 51 IACC late percentile 10%.

Figure 52. IACC late 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


24

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 Position DRR

Ground 1 -7,1

Ground 2 -3,6

Ground 3 -4,7

Ground 4 -0,7

Ground 5 -5,1

Ground 6 -3,2

Ground 7 -3,4

Ground 8 -1,8

Ground 9 -2,3

Ground 10 -3,8

Ground 11 -4,4

Ground 12 -5,2

Ground 13 -4

Ground 14 -6,1

Ground 15 -1,6

Ground 16 -0,3

Ground Total -3,58

First 17 -4,5

First 18 -2

First 19 -1,4

First 20 -1,4

First 21 -2,5

First 22 -7,3

First 23 -4,3

First 24 -3,4

First 25 -6,7

First 26 -6,3

First 27 -1,9

First 28 -7,2

First Total -4,08

Second 29 -5,2

Second 30 -6,2

Second 31 -1,1

Second 32 -6,1

Second 33 -1,9

Second 34 -3,8

Second 35 -2,5

Second 36 0,5

Second 37 -12

Second 38 -5,4

Second 39 -2,8

Second 40 -5,8

Second Total -4,36

Average Total -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.


25

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.


26

[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

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Reverberation Time RT in Concert Halls.

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Documents

Acoustical Parameters