Sound Pressure Capacity Requirements for Monitoring
Immersive Audio Formats
Aki Mäkivirta, Juha Holm, Juha Urhonen, Ilkka Rissanen, Jussi Väisänen
Genelec, Iisalmi, Finland, email: aki.makivirta@genelec.com
Abstract
The statistical distribution of the sound event duration in a cinematic immersive audio track is considered as information todetermine the short and long term sound pressure output capacities needed in loudspeakers and subwoofers. The statistical
distribution of audio event duration in cinematic audio tracks is studied to evaluate the needed monitoring system capacity.
Monitoring system capacity requirement is also related to the monitoring room size, the listening distance, and the room
reverberation time. Impact of using bass management and selection bass management crossover frequency are notconsidered. An attempt is made to develop these factors into a design guideline enabling successful monitoring room designand selection of the acoustic and physical characteristics of the monitoring loudspeakers and subwoofers needed in the roomof a certain physical size.

Introduction
Modern immersive audio reproduction systems use a large
number of loudspeakers in geometrically predetermined
positions. Examples of these include the broadcastingrelated systems such as ITU recommendation for immersive
audio [1] and the NHK 22.2 system [7], as well as
commercial distribution and presentation systems such as
Auro 3D [9], Dolby Atmos [5,6], and DTS Neo:X [3]. ITU
recommendation BS.2159-4 for broadcast applications
covers both channel-based and object-based immersive presentation systems, including 10.2, 22.2, wavefield synthesis,
and object-based audio presentation formats [1]. International Electrotechnical Commission IEC 62574 describes
a three height layer reproduction system for consumer
applications with up to 32 speaker positions including anover-the-head speaker, able to support all current multichannel formats [1,2].

Use of channels
Object based audio typically uses a bed in a 7.1 presentation,
containing most of the audio presentation [11]. The bed
presentation is usually done with the monitors on the listenerear level, set according to the ITU recommendation at equal
acoustical distance and level at the listening position [12].
Object monitors or height layer monitors are usuallyassumed to be acoustically set to the same acoustic distanceand level as the bed monitors. However, it is not untypical tofind that the actual physical distance of the monitors may not
be the same, and that the acoustical distance equality has
been achieved with electronic delays. This produces amonitoring arrangement that performs perfectly from thesound design point of view but may require some of themonitors to work relatively harder, outputting a higher soundlevel, as they are located at a larger physical distances, andare operating at a higher output level after the level
calibration has been completed. This should also have an
influence on the monitor type choice if we know that certain

monitors are physically located at a larger distance from thelistening position.

Monitoring level
Level alignment practices use 18 or 20 dB digital audio
headroom. A permitted maximum signal level is usually set
to –9 dBFS, but the peak level may be finally scaled todigital full scale [10]. Maximization is done as the final
stage, when the print master mix is complete. During
recording of the individual tracks and objects sufficient
headroom is maintained so that signal fidelity can be
maintained.
EBU recommends the programme loudness instead ofcontrolling the permitted maximum (peak) level. The
loudness should be regulated to -23 LUFS so that the digitalaudio full scale (FS) is not exceeded [17]. However, thesefigures reference to the digital full scale, and do not say
much about the sound pressure level (SPL) used for monitoring the audio signal.
For the monitoring level EBU is recommending a scalingfrom an -18 dBFS pink noise input to a sound pressure level(SPL) at the listening position Lmon such that the monitoringlevel depends on the number of channel in the reproduction
systemLmon  85  10 log (n)

[dBc]

(1)

For example, the monitoring level becomes 78 dBc SPL foreach monitor in a 5.1 system, 75 dBc for a 10.2 system, and71.6 dBc for a 22.2 system [10,17,18].
This approach to setting the levels lowers the individual SPL
capability requirement for each monitor when the channel
count grows.
However, certain channels may need to have an output
capacity higher than this, for example when the environmentis intended for multiple types of productions, with varying
channel count. One such approach is presented by thestandard IEC 62574 [2] proposing a multipurpose monitor
layout.
While the broadcasters recognize the capacity of a highchannel count system to achieve a higher aggregate SPL, thisprinciple is not applied by the cinema industry.
SMPTE recommends aligning monitoring systems to 83 dBc
at a –20 dBFS input level [10] at the listening position,
implying a peak level of 103 dB SPL by an individual
monitor. Dolby Atmos specifies that each screen (front)monitor should have the maximum continuous output
capability of 92...105 dB SPL at the listening position,
depending on the frequency range, and 99 dB SPL for thesurround monitors [5].

Effect of reverberation
Whilst the shot-term peak SPL of a monitor is determined by
the monitor design, the long-term SPL or the sound levelthat can be sustained by a monitor depends heavily on theroom acoustics, notably the reverberation time. Larger reverberation times are linked to smaller acoustic absorption in
the space, leading to higher steady state SPL.

1
Tm  0.25 (V / Vo ) 3 , Vo  100 m 2

[s]

(2)

Higher reverberation time is linked to more pronounced
room resonances, and these are detrimental to accuracy ofmonitoring. Because of this, high quality monitoring spaces
strive towards low reverberation time, and the targets are
given by several recommendations. For example ITU-T BS.
1116-1 recommends a midrange reverberation time Tm forhigh quality monitoring spaces. The typical reverberation
time in professional monitoring rooms has the median valueof 0.4 s in the mid frequencies (Figure 1).

places the height monitors above the respective ear-height
monitors [19,20] and this leads to a larger distance from thelistener to the higher-tier monitors. The voice-of-god or overthe head monitor distance can vary depending on the roomheight and other practical factors. The relative locations of
the ceiling object monitors result in varying distances to thelistening location.
The mean listening distance in a stereo configuration isabout 2.5 meters with the listening distances ranging from
1.2 to 4.2 meters [24]. Cinema standards require monitoring
rooms emulating the film theatre and require front monitor
distance of at least 5 meters, but there is also a
recommendation for a room size, monitor layout and setup
very similar to ITU recommendations for multichannel
music productions [25].

Distance dependency of sound level
The distance to the monitor and the room reverberation time
affect the sustained sound level in the monitoring room.
The sound output capacity of a monitor is usually given by
the manufacturer at a certain nominal distance. The standard
distance is one meter. Both the short-term peak sound level
and long-term sustained maximum sound level scale with the
distance.
Most monitors and subwoofers behave like point sources interms of sound level attenuation with increasing distance tothe monitor. This means that the sound level decreases by 6
dB for every doubling of the distance to the monitor until thereverberation in the room begins to dominate and the soundlevel reduction with distance becomes small [22].
In high quality monitoring spaces the reverberation time isrelatively low. It is usually safe to assume that the squarelaw of level reduction with distance is valid for usual
listening distances of the monitors in the monitoring room.
We can follow the principles given in [22] to estimate thesound level Lp on the acoustical axis of a monitor at thelistening position when the distance to the monitor is r.
For this, we assume that the monitor is placed near a wall, as
this is the typical installation case. The attenuation of thedirect sound from the monitors as a function of the listening
distance, the sound absorption in the room, and the
directivity of the monitor must be considered.
The sound absorption area of the room A is usually best
available after measuring the reverberation time RT60 in theroom, knowing the internal volume V of the room.

Figure 1: Percentiles of the sound reverberation time RT60 inprofessional monitoring spaces (N=372) [24]Distance from monitor to listening position
The monitor layout for the cinema reproduction systems is
not mechanically equidistant, so requirements for individual
monitors vary according to monitor location. The reference
listening location is typically closer to the back wall than to
the front, reducing distances to the rear monitors. Auro3DA

0.161 V
RT60

[m2]

(3)

Monitor directivity Q is either given by the manufacturer or
it can be estimated with measurements. Manufacturers
usually give this data in the form of a directivity index D.
D
Q  1010(4)

The radiating power Lw for a monitor is not given by the
manufacturer. However, the sound output level on the
acoustical axis at one meter Lp0 is usually given, and can beused to estimate the radiated acoustic power. Here we takethe usual case and assume the monitors are placed at or close
to a wall.

 2Q 
Lw  Lp0  10 log 4 

[dB PWL](5)

Directivity and absorption are frequency dependent, and for
precise calculations this should be considered. For
estimating the achievable sound pressure, the midrange
average value will be sufficient.
The sound level Lp at the listening position, on the
acoustical axis of a monitor, at distance r to the monitor isexpressed by 2Q   4 
Lp  Lw  10 log
 4 r 2   A [dB SPL](6)

Using these principles of calculation the peak short-term
output level and the sustained long-term output level can be
estimated based on data given by monitor loudspeaker
manufacturers once the room volume and reverberation time
have been modelled or measured.

Monitor output capacity
As the thermal capacity and heat conduction of a dynamic
driver voice coil is not large, the long term sound output
level of an active monitor is normally limited by protection
systems to prevent the driver voice coils from overheating[13,14,15,16]. Because of this, even if monitor loudspeakers
typically have a very large short-term sound output capacity,
they have clearly lower capacity to sustain long-term soundoutput.
Activation of the protection lowers the highest sound level
and this can also change the tonal character or fidelity of thesound output.
One fundamental aim of active monitor design is to avoid
activation of the protection under all normal operating conditions. This design requires careful balancing of the shortterm capacity with the long term capacity. This balancing isbased on assumptions about the operating conditions as wellas the nature of the acoustic signals that are monitored.
The assumptions typically include (1) the sound pressure
normally used in monitoring, (2) overhead sound output
capacity needed for the maximums or peaks in the soundoutput, (3) distance from the monitor to the listening position, (4) reverberation time of the space where monitoringtakes place, and (5) typical dynamic level variability properties of the audio material.
In the following we discuss particularly the last item in thislist.

Temporal level variability in audio
Activation of the protection lowers the maximum soundlevel and this can also change the tonal character or fidelityof the sound output. The aim of an active monitor design isto avoid activation of protection under all normal operating
conditions. In order to do this, it is useful to study thecharacteristics of the typical audio signal.
Audio signal value distribution has been studied as change inthis value distribution over time, particularly for music
recordings. A typical statistical value analysis for film soundtracks shows the highest likelihood at –20... –15 dB of fullscale [26]. This coincides with the way the headroom istypically set for film sound.
The value distribution data does not give any information
about the temporal character of the audio data, i.e. for how
long large levels typically last and how long are the pauses
between high level peaks. This information would be very
useful for understanding the performance requirements for
active monitors and subwoofers.
It is important to note here that any length of audio signal isa valid audio signal, also a signal that has unlimited length intime. However, such signals are not very typical, particularly
at very high output level.
The purpose of the following discussion is to review thetypical temporal properties of audio in film sound tracks.
The film sound tracks used as material represent several
genres of films and are relatively recent multichannel productions (Table 1).
year

type

ori.

duration size
h:mm:ss i GB

no.
chap.

title

A

2004

actionUS

1:50:03

5.31

39

I RobotB

2006

animation

US

1:22:433.99

28Open Season

C

2009

drama

FI

1:12:03

3.47

12

D

2006

drama

FR

1:41:40

4.90

18

Postia pappi
Jaakobille
Science of
Sleep

Table 1: Material used in the study

The multichannel soundtracks were extracted from DVD
releases of the films. The soundtracks contain a 5.1 channelsound presentation with 48 kHz sampling frequency and 24
bit sample precision (a track sample is shown in Figure 2).
The data was analyzed at original, full dynamic range andoriginal sample rate. Also the LFE channel was included in
the extracted data. In total, the material includes 6.3 x 109(billion) samples.
The motivation for using the DVD release sound tracks isthat the word length and sample rate of this test material is
the same as the word length and sample rate of most film
releases with immersive 3D presentations. The test material
models well the bed audio presentation for 3D immersive
audio.
We can also assume that the audio signal levels seen in these
multichannel presentations are representative also of thelevels of the object channels or height channels in immersive
audio presentations. This is because, once the audio content

exists in the height channels and the object channels, therelative sound level that must be presented from there to thelistening location is bound to be similar.
Finally, actual 3D sound track material was not available forthe study.
Source materials A and B are action films with relatively
loud passages containing effects, Foley, and a significant
orchestral background besides the dialogue.
Source material C represents the Nordic drama style of film
making, with the dialogue as the central element, and minimal Foley and orchestral elements. This material contains a
lot of very low sound level passages, but the whole dynamic
range of the 24 bit data presentation has not been employed,probably due to reasons of the recording and production
technology.
Source material D represents a French drama film, with veryrich dialogue, Foley, and significant orchestral score.
While we wanted to include several types of film soundmaterial for the study, it was anticipated that the action typefilms would present the highest audio levels and widest
frequency range requirement.

The two Hollywood productions (materials A and B) use
high levels and hit close to the full scale more frequently
than the European productions (C and D). Material C has thelargest headroom, and consists dominantly of dialogue,
containing very little effects and music.
While in material A all three front channels are used equallyin terms of level, other productions use the centre channel
more than the left and right front channels. This is possibly asign of the dialogue placed mainly in the centre channel
having a higher level in materials B, C, and D. The LFE
signal seems to be usefully exploited also at low level inmaterial A whereas the other materials use the LFE muchless.
The full scale of the dynamic range is hit in material A so
precisely that it must be assumed that the sound track has
been digitally maximized to the full scale, also evidenced
with a tiny amount clipping in the all the audio channels.
The other materials do not show this. Other materials rarely
show values in the last 10 dB below the full scale, with thecentre channel content in the highest level. The least loudmaterial C has a 6 dB headroom in the front left and rightchannels, and a 2 dB headroom in the centre channel.
Surround maximum levels remains systematically 5...10 dBlower than the front triplet level. When used, the LFE levelpeaks typically to the full scale. The LFE is usually playedback 10 dB higher than the main channels.

Peak level distribution
The aim is to evaluate the frequency and length of peaks
exceeding a set detection level L. In this way it is possible toevaluate (1) how likely a peak exceeding the detection levelis to occur, (2) what is the expected length of the peak, (3)what is the expected time between repetitions of peaks.
Audio signals can be short or infinitely long but in typicalaudio material, high level peaks have finite length. Whenevaluating the signal level, time windowing has been used to
extract an average of level over a certain period of time, forexample in [26]. Applying a time window tends to reduce
the level of short duration peaks, and particularly considering the high frequency radiator channel, accurate estimatesof the signal maximum level are not obtained.
Figure 2: Example of the extracted multichannel sound track; from
top to bottom: left, right and centre front channels, LFE channel,
and left and right surround channels

Value distribution
The value ranges in audio tracks were studied by measuring
the histograms of the signal envelopes (Figure 4), enabling
precise envelope estimation also at high frequencies, fasttransitions and short peaks. The envelope has been extracted
as the magnitude of the analytic signal [27]. The analytic
signal is a complex-valued signal consisting of the originalreal-valued signal and its Hilbert transform (equations 7-9).
The histogram was measured in bins having one dB width
across the dynamic range of the 24-bit PCM presentation.
The histograms contain the whole audio track of the film.

In order to avoid attenuating short peaks, we can look at themagnitude y(n) of the signal x(n) directly, or the envelopeh(n) of the signal (Figure 3).

y ( n)  x ( n)(7)h(n)  x(n)  j xˆ (n)(8)

j  1(9)

Figure 3: A sample burst signal x(n) (left) and its envelope h(n)(right)Figure 4: Value histograms for materials A (top left), B (top right), C (bottom left) and D (bottom right). Each histogram contains thewhole duration of the audio track in the material. The bin spacing is 1 dB. Each histogram contains more than 109 samples.

Figure 5: Lengths of peaks higher than L= –10 dBFS in the full bandwidth signal envelope. All audio channels in the material A are
included. Material A has the highest likelihood of peaks values of the studied materials.

Figure 6: The observed frequency versus the length of peaks higher than L= –10 dBFS, in envelopes of driver-specific bandwidth signals.
All audio channels in the material A are included. Material A has the highest likelihood of peaks values of the studied materials. Bandwidth
limitation is explained in the text.

The duration of the output peak levels was studied for peaks
exceeding the level L = –10 dBFS in the full bandwidthenvelope of the signal (Figure 5). The data is presented ashistograms. The likelihood of peaks increases with decreasing peak length. The gap between peaks follows similar
statistic. Looking at the likelihoods of peaks and the gaps
between peaks, the expectation is that a 99% of the peaks
have a length less than 4 ms. Peaks longer than 10 ms are
infrequent in our material but do occur.
The immediate conclusion is that high level long duration
peaks are unlikely to occur. While a continuous signal is aperfectly normal audio signal, it is not very likely to occur at
a very high level, close to the maximum output level.
The peak statistics changes after the signal has been filteredwith bandpass filters being a part of an active monitor or
subwoofer crossover filtering.
The crossover filters were modelled with fourth-order
Linkwitz-Riley frequency responses. The crossover cut-off
frequencies were modelled at 90 Hz for the subwoofer, 500
Hz and 3 kHz for the three-way monitor model, and 3 kHz
for the two-way monitor model. These frequencies represent
well the typical monitoring applications seen on the market.

As expected, the frequency bandwidth affects the duration
distribution of the peaks (Figure 6). While the peaks in thetweeter channel are typically 40 s in length and seldom
more than 1 ms long, the peak length in the subwoofer
channel is about 10 ms and can be several hundred milliseconds in length. If we take the 1% occurrence level as thecriterion, meaning that 99% of peaks are shorter in duration
than a limit, then these limits are given in Table 2. The
figures given do not imply that runs longer than the limits
could not occurs or that one cannot produce a sound trackthat would have a higher incidence of long runs, but thelimits given do give an idea of the typical situation we can
expect.
crossover channel
modelpassband(Hz)

tweeter
midrange
two-way woofer
three-way woofer
subwoofer>3k
0.5 k - 3 k<3k< 0.5 k< 90

run length
1% limit
(ms)
0.4
5
15
40
300

relativeoccurrence(percent)
7.7
11
100
16.4
0.8

Table 2: Lengths and relative occurrences of peaks in driverspecific signals for material A

Discussion
While it is not a great concern for theatrical presentations
where the listener’s distance to the loudspeakers is large, thephysical size of the loudspeaker becomes a serious consideration for the compact monitoring rooms. The roomdesigner must consider the loudspeaker’s influence to theacoustic performance of the room particularly for immersive
audio reproduction systems where a large number of loudspeakers can create acoustically significant reflecting area inthe room. While avoiding the acoustic reflection andworking in smaller-sized rooms imply using physically small
loudspeakers, high maximum sound pressure requirement
combined with the requirement that the frequency bandwidth
extends to low frequencies leads to selection of physicallylarge monitoring products. Bass management can offer at
least a partial solution to this, particularly in small monitoring rooms, if the use of bass management is acceptable.
Certain cinema standards present requirements on the maximum sound level capacity of the audio reproduction system.
An example of these is the Dolby Atmos, specifying the SPL
at the reference listening location.
In this paper, an attempt has been made to describe themethods to consider the variables in monitoring room design
when selecting the acoustic performance of the monitoring
loudspeakers and subwoofers. Calculation methods toestimate the sound pressure level at the listening location
have been given for a case where the physical size andacoustical characteristics of a room are known or have been
estimated using simulations. These have been described forthe typical case where the monitors are located close to one
acoustically hard surface, typically a wall or ceiling. The
method to estimate the sound level at the listening location
has been provided as a collection of formulas. The use of theformulas requires some data describing the characteristics ofa particular monitor. This data can typically be obtained
from the manufacturers or with measurements.
The material studied demonstrates that the surround channels are operating at about 5...10 dB lower maximum level
than the front monitor triplet. Most likely this also applies tothe height output channels in immersive audio presentation
systems, too, due to the tendency to concentrate a lot of theaction in the front at the screen in cinema mixes.
The realistic duration at a high output level was studied
using the multichannel sound track material on DVD
records. It can be assumed that the characteristics of thismaterial can be extended realistically also to 3D immersive
audio systems because the method of building the 3D
presentation uses heavily the standard multichannel audio
presentation on the ear height of the listener.
The statistical distribution of the high level sound event
duration in a full bandwidth cinematic immersive audio trackenvelope is less than 4 ms when all audio channels are considered collectively. In a multiway monitor and subwoofer
the high sound level event duration depends on thefrequency band, with the highest occurrence of high levels in
the two-way woofer channel. The benefits of a three-way
design over a two-way design were evident as a lower
occurrence of high levels in each driver channel.
The high level audio event duration was estimated using
modelled Linkwitz-Riley crossover filter bands at the typicalpassbands for active monitors and subwoofers. After crossover filtering the envelope peak level durations change toreflect the passband of the crossover filter. The statisticallimits found in this work do not mean that longer duration
maximum output levels could not occur, as any duration of
the audio signal at any given frequency is completely normal
and can be produced at will. The statistics merely describe
the typical case in real material, and therefore can be taken
as an indication of what typically will be required of amonitoring loudspeaker or subwoofer.
When evaluating the maximum SPL capacity of the reproduction system it would make sense to use test signals that
resemble the typical signals that are reproduced. Attempts
towards this direction have already been made, for example
the Jiffy signal by Dolby and the use of non-continuous
pulsed pink noise [5]. Signals with realistic characteristics
should be preferred over using continuous technical test
signals, such as sinusoids or uninterrupted pink or white
noise, as the continuous signals will produce misleading
estimation of the systems real-life performance and arelikely to lead to severely over-specified and expensive system implementations not necessarily delivering better
performance in real life.

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