Acoustics

Knowledge booklet
Testing Knowledge Base
compilation

https://community.plm.automation.siemens.com/

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The content in this booklet and the Siemens PLM Community site
is the hard work and dedication of many individuals.

The Siemens PLM Community site can be found at:

https://community.plm.automation.siemens.com/t5/Testing-
Knowledge-Base/tkb-p/Simcenter_Test_tkb

Table of Contents
What is the acoustic quantity called Q? ....................................................................................................... 2
Sound Pressure, Sound Power, and Sound Intensity: What’s the difference? ............................................. 8
Sound Transmission Loss ............................................................................................................................ 13
What is Sound Power? ................................................................................................................................ 25
Sound Absorption ....................................................................................................................................... 33
Noise Level Certification: how to select the right standard?...................................................................... 49
Octaves in Human Hearing ......................................................................................................................... 53
Basics: What is a decibel (dB) anyway? Why is it used? ............................................................................. 63
Sound Fields: Free versus Diffuse Field, Near versus Far Field ................................................................... 68
Sound Pressure ........................................................................................................................................... 74
What is A-weighting? .................................................................................................................................. 78
All Articles - Siemens PLM Community Testing Knowledge Base ............................................................... 86

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What is the acoustic quantity

called Q?
What is Q? What is it used for?
The letter Q is often used to refer to VolumeAcceleration. VolumeAcceleration, or Q, can
be thought of as the equivalent of an acoustic force. It has units of m3/s2.

For a mechanical system, Newton's Second Law of motion is F=ma. F is the force input
into the system, a (acceleration) is the acceleration response output, and m is the mass.
For an acoustic system, the Volume Acceleration (Q) is equivalent to the force in a
mechanical system, and sound pressure (P) is the equivalent to the acceleration output
response.

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Volume Acceleration
At first, it may not be intuitive how “VolumeAcceleration” is equivalent to acoustic force.
The easiest way to visualize “VolumeAcceleration” is to think of it as Area (m2) multiplied
by Acceleration (m/s2)

If one visualizes a speaker, imagine dividing the speaker surface areas into a series of
small areas. In the center of each area, place an accelerometer. The total acoustic force
that the speaker produces would be the sum of individual areas times their respective
accelerations.

In practice, one must divide the surface into smaller and smaller areas as the desired
frequency for analysis increases.

A vehicle dash panel broken into small areas for measuring Q.

In other words, the higher the frequency, the smaller the area patches. This is due to
the change in the wavelength of sound.

What can Q be used for?

Utilizing a device called a Q-source, it is possible to take some interesting measurements.

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Mid-High Frequency Q-source: The tip of the nozzle is where a calibrated Q is generated.

Traditionally, when performing a measurement campaign to quantify the acoustic paths

of a structure, one might employ an impact hammer (which measures force) and
measure the acoustic response (sound pressure) with a microphone. This would result in
a Frequency Response Function, or FRF, between the input location (perhaps a motor
mount) and receiver location (a driver ear).
To quantify the sound contribution for a structure that had a motor attached to it at four
locations, this could require 12 separate impact measurements: applying a force via an
impact hammer at 4 separate mount locations in three different directions. The result is
12 separate Pressure/Force (P/F) transfer functions.

Using the Q-source, one could do these 12 measurements at a single time.

Putting the Q-source at the acoustic response location of interest, and accelerometers on
the mount attachment locations, the measurement is performed at one time.

This is because P/F = A/Q, as shown below:

P = Sound Pressure in Pascals (Pa)

F = Force in Newtons (N)
A = Acceleration (m/s2)
Q = VolumeAcceleration (m3/s2)

P/F = N/m2/N = 1/m2

A/Q = m/s2/m3/s2 = 1/m2

Both P/F and A/Q reduce to the same units of 1/m2, making them equivalent.

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Using a Q-source with Simcenter Testlab

To take a measurement with the Q-source using Simcenter Testlab (formerly LMS
Test.Lab), it can be attached to the source output of a SCADAS through a built-in stereo
amplifier cable. The Q-source will produce a sound field. There is a BNC connector on
the Q-source which outputs the Q signal.

The source output of the SCADAS is connected to the Q-source via an amplifier. The output of the Q-source is fed into
a SCADAS data channel.

In the "Channel setup" worksheet, select the following:

 Set "Measured Quantity" to "VolumeAcceleration"

 Set "Input Mode" to either ICP or Voltage, depending on the type of Q-source (will be
indicated on calibration sheet)
 Enter the calibration factor from the calibration sheet into the "Actual Sensitivity"
field. The field should be "mV/m3/s2".

Simcenter Testlab "Channel Setup" worksheet for Q-source

Much of the information to fill in comes from the Q-source calibration sheet.

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Q-source calibration sheet shows necessary information to use with now Simcenter Testlab.

The source can be setup as random in "Scope" worksheet of Simcenter Testlab Spectral
Acquisition.

Source setup in "Scope" worksheet

And select "FRF" as the measurement to save in the "Test Setup" worksheet of Simcenter
Testlab Spectral Acquisition.

In the Simcenter Testlab "Test Setup" worksheet, check on "Measure" and "Save" under the FRF tab.

After performing the measurement, and viewing the FRF in a display, the FRF can be
viewed as P/F rather than A/Q by right clicking on the Y-axis and selecting "Unit".

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Comparison of Impact Measurements vs Q-Source measurements

Taking reciprocal FRF measurements with Q-Source yields equivalent results to impact
hammer FRF measurements, but in less time.

Comparison of Impact Measurement vs Q-Source Reciprocal Measurement

Note: Even though the Q-source outputs Volume Acceleration, sometimes the sources
are referred to as Volume Velocity sources.

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Sound Pressure, Sound

Power, and Sound Intensity:
What’s the difference?
Confused by acoustic terminology?
This article highlights the main differences between the three acoustic terms: sound
pressure, sound power, and sound intensity. The article attempts to explain when to use
them, their units of measure, and how they relate to each other.

These three terms all measure different aspects of sound, but can all be expressed
in decibels as shown in Figure 1. A decibel is not unit of measure, but rather a
logarithmic ratio between two numbers (a measured quantity and a reference number).

Figure 1: Sound pressure, sound power and sound intensity can all be expressed in decibels (dB) even though they
represent different measured quantities.

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The measurement units used in sound pressure, sound power, and sound intensity are
different. Often the measurement unit is omitted during discussions, and only the term
“decibels” is used. This can cause some confusion to arise.

Pressure, Power, and Intensity: Heater Analogy

Figure 2: Heater analogy for Sound Pressure, Sound Intensity and Sound Power

An analogy between a heater placed in a cold room, versus a sound emitting object in
quiet room, can be used to illustrate the differences between pressure, power and
intensity. There are several similarities between heat and sound as shown in Figure 2.

The heater creates heat, which spreads throughout the room. A noise emitting object
creates sound in a similar fashion. The following parallels can be drawn:

Sound Pressure and Temperature

At every position in the room, there is a specific temperature level, which is measured in
degrees. Likewise, at every position in the room with the sound source, there is a
particular sound pressure level, which is measured in Pascals. As heat is produced, the
temperature level is higher closer to the heater. Like temperature, the sound pressure
level is typically higher closer to the noise emitting object. Both the sound pressure level
and temperature level are dependent on the location and distance away from the source
object.

Sound Power and Heater Power

The heater generates a particular amount of heat per hour. The power required to
generate this heat is the same, no matter what the temperature in the room. The heater
power is measured in energy over time, or Watts. Sound power operates on the same
principal – the sound power of an object is solely a property of the object, and
is independent of the sound pressure levels in the room. Sound power is the rate at
which sound energy is emitted per unit time. Sound power is also measured in Watts.

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Sound Intensity and Heat Flow

Heat travels and flows throughout the room. The heat flow has a temperature level and
a direction. Sound intensity is the measure of the flow of sound and has a level and a
direction. This flow is observed over a specific area; hence the units of sound intensity
are W/m2.

*** In this analogy, it was assumed that both the heater and sound emitting object have a
constant output. The room was assumed to be free of reflections and other sources. ***

Relationship of Sound Pressure, Sound Power and Sound Intensity

Sound pressure, sound power, and sound intensity can be related to each other under
some specific circumstances.

Sound Pressure and Sound Intensity Relationship

In an acoustic free field, the sound intensity is directly related to sound pressure by the
following equation:

Sound Intensity = (Sound Pressure) x (Particle Velocity)

The particle velocity is the speed of which the air molecules vibrate back and forth while
transmitting a sound. Particle velocity is a vector quantity, while sound pressure is only a
scalar amplitude. The result is that sound intensity is a vector quantity.

At any given location around a sound source, either the sound intensity or sound
pressure can be measured, as shown in Figure 3.

Figure 3: Sound Intensity (left) versus Sound Pressure (right) around an electric motor

The sound intensity (left side) shows both amplitude (via color) and direction (with a
vector arrow). The sound pressure (right side) shows only amplitude with color:

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 Amplitudes the Same - Looking at the color distribution of sound intensity and
sound pressure amplitude levels, the same pattern is present in both images.
 Direction is the Difference - The sound intensity vectors on the left side of Figure
3 clearly indicate the direction of sound flow, making it easier to troubleshoot the
cause of the high sound levels. The sound pressure values on the right side do not
indicate flow direction or provide clues as to where the sound originated.

Sound pressure level can be measured with a single microphone, while sound intensity
is a more complicated measurement. A sound intensity measurement requires two or
more microphones in a specific arrangement. For example, the Simcenter
Soundbrush uses four microphones in a tetrahedral pattern to measure intensity.

Sound Intensity and Sound Power Relationship

It is easy to convert from sound intensity to sound power (and vice versa), if the area
over which the measurement was performed is known:

 Sound intensity is expressed in Watts/m^2.

 Sound power is expressed in Watts.

Multiply the sound intensity value by the area (in m^2) covered by the measurement to
calculate sound power.

In an acoustic free field, the sound intensity at a specific distance from a sound emitting
object can be calculated, if the sound power of the object is known. Take a printer that
has a sound power of 0.02 Watts. Sound intensity will be measured in a 2-meter hemi-
sphere around the printer as shown in Figure 4.

Figure 4: Object with known sound power in 2-meter hemi-sphere test

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To get the sound intensity, divide the sound power by the area of the hemi-sphere:

Sound power is 0.02 Watts

 Hemi-sphere surface area is 25 m^2 (2-meter radius)

 Sound intensity is 0.02W/25m^2 for a value of 0.0008 W/m^2
 Using Equation 1, the 0.0008 W/m^2 can be converted to decibels

Equation 1: Conversion of power measured quantities to decibels

Using the following values in Equation 1:

 Reference value for intensity (10e-12 W/m^2 from Figure 1) for Wref
 Sound intensity of 0.0008 W/m^2 for W

The sound intensity at two meters distance is 89 decibels.

Conclusion
Sound pressure, sound power, and sound intensity are acoustic quantities that can be
expressed in decibels. They describe different aspects of sound, and the decibels for
each represent different measurement quantities.

 Sound Pressure – Indicates the amplitude level of sound at a specific location in

space and is a scalar quantity. The level is dependent on the location and distance
the sound is observed relative to a sound source. Sound pressure is measured in
Pascals.
 Sound Power – The rate at which sound is emitted from an object, independent of
location or distance that the sound is observed. Sound power measurements are
often specified in the noise regulations of many different kinds of products, from
construction equipment to computer printers. Sound power is measured in
Watts.
 Sound Intensity – Sound intensity is sound power per unit of area. It indicates the
flow of sound through a specific area. Sound intensity is measured in Watts/m^2.

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Sound Transmission Loss

When sound reaches a barrier, three things can happen, as shown in Figure 1:

 Absorption – The sound is absorbed and dissipated as heat.

 Transmission – Sound can pass through the barrier.
 Reflection – Sound can be reflected off the barrier.

Figure 1: Sound at a barrier can be absorbed, transmitted, or reflected.

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Sound transmission loss (STL) is a quantification of how much sound energy is

prevented from traveling through an acoustic treatment. Transmission loss quantifies
the effectiveness of acoustic treatments for an engineering application.

Sound transmission loss can be defined as a ratio of the sound energy transmitted
through a treatment versus the amount of sound energy on the incident side of the
material.

Equation 1: where W_i is the incident sound power, W_t is the transmitted sound power. These quantities are
pictorially represented in Figure 1 above.

Sound transmission loss is a function of frequency. The transmission loss performance of

a certain material will differ greatly with frequency (see Figure 2, below). The y-axis of a
plot represents how many dB the acoustic absorber reduces the incident energy.

Figure 2: Sound transmission loss as a function of frequency. At 2050Hz, the acoustic absorber reduces the incident
energy by 10dB. The data used in this plot was calculated using the matrix method on a muffler.

For example, in the plot above, it is clear that the transmission loss value is highly
dependent on frequency. At 2050Hz, the muffler reduces the incident energy by 10dB
(green dotted line). However, at 3500Hz, the muffler does not reduce the incident
energy at all (purple dotted line).

Effects on Sound Transmission Loss

When trying to reduce the sound passing through a barrier, it is common to apply a layer
of acoustic treatment material. Coverage of the barrier as well as any holes in the
acoustic treatment will affect the sound transmission loss

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Effect of coverage on transmission loss:

If trying to reduce the sound through an object (such as a square panel) one may apply
some acoustic material onto the panel. It is important to note that the percentage of
coverage of the object will have an effect on the sound transmission loss.

In the figure below, transmission loss is plotted for three scenarios: the square panel
being completely covered by acoustic material, 3% of the panel remaining exposed, and
25% of the panel remaining exposed.

Figure 3: When the area of the acoustic material does not completely cover an object, the sound transmission loss
decreases.

Notice that even with a small area of exposure (just 3%), the transmission loss can
decrease dramatically. Increasing the size of the exposure (even as large as 25%) does
not have as great of an effect as introducing a small initial hole. The effects are
especially notable at higher frequencies.

Effect of holes on transmission loss:

If there is a hole in the acoustic treatment, it will cause a decrease in the effective
transmission loss. Note that this effect is especially noticeable at higher frequencies.

In the figure below, transmission loss is plotted for three scenarios: full coverage, a small
1% leak, and a larger 5% leak.

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Figure 4: Introducing leaks and holes in the acoustic barrier reduces transmission loss, especially at higher
frequencies.

It is important to recognize that even small holes and small areas of missing coverage
can greatly reduce the transmission loss of a barrier. As the hole gets larger, the effect is
not as dramatic as introducing that first initial hole. This is especially apparent at higher
frequencies.
Applications and Measurements
Sound transmission loss can be a good metric for benchmarking the acoustic
performance of products.
Common testing applications include:

 Determining the effectiveness of a muffler in a system

 Determining the transmission loss of various ducting systems
 Determining how well a building panel or partition attenuates sound energy
 Determining how well an instrument panel insulates a cabin from engine noise

Knowing the sound transmission loss helps to determine and improve acoustic
properties of materials.
There are two major methods for determining STL:

 Impedance tube method

 Two room method

Transmission loss is independent of the source meaning that STL can be measured using
a source such as a loudspeaker (it does not need to be measured in-situ).

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Impedance tube method

The impedance tube method is useful for determining the STL of components like
ducting and mufflers as well as small material samples. Impedance tubes are typically
made of straight “sound proof” tubing (typically thick steel).

One end of the tube is connected to a sound source which outputs a broadband range of
sound waves. The element to be tested is mounted in the middle of the tube. The sound
waves approaching the sample are direct incidence and normal to the sample.

Figure 5: Top: Impedance tube. Bottom: Side view of impedance tube to show sample location in the tube.

Care must be taken to ensure a tight connection between the sample and the tube. For
example, if testing a muffler, there must be a tight connection between the outlet of the
first tube and the inlet of the muffler as well as the outlet of the muffler and the inlet of
the second tube.

Figure 6: The engineer ensures a tight fit between the ends of the muffler and the ends of the impedance tube.

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NOTE: Conical adaptors may be used to account for a difference in diameter between
the sample outlets and the impedance tube diameter. Conical adaptor corrections can be
factored in using Simcenter Testlab.

Figure 7: The cone correction screen in Simcenter Testlab.

There are a few methods to determine STL in an impedance tube. Simcenter Testlab uses
the four-microphone transfer matrix method which assumes a lumped parameter model
of a 3D acoustic cavity.

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In this equation set, p1, v1, p2, and v2 can be measured. The unknowns are T11, T12,
T21, and T22. Because there are four unknowns and two equations, there must be two
different loading conditions to create four equations to solve for the four unknowns.

To get the two different conditions, it is possible to either run the test under two loading
conditions or change the source location between test runs.

The two-load method is recommended over the two sources method because:

 It is easy and quick to change the end condition.

 There are no cables moved between conditions.
 Inexpensive objects like tubing is moved (not fragile/expensive objects like
sources).

Typically, the two loading conditions are a rigid termination and an anechoic
termination.

Simcenter Testlab offers a single page workbook for determining

Sound Transmission Loss.

Figure 8: The result of a sound transmission loss test in tube is the sound transmission loss of the material vs.
frequency.

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The above figure is taken from the Simcenter Testlab Sound Transmission Loss
Measurement Using Impedance Tubesoftware.

Room methods
It is also possible to calculate sound transmission loss using the two-room method. This
method is most appropriate for larger samples and even complete components like
doors, windows, and vehicle components like a dash panel.

There are a few methods using rooms:

 Sound intensity and sound pressure

 Sound pressure and sound pressure

Sound intensity and sound pressure

The two-room method using sound intensity and sound pressure uses one reverberant
room and one anechoic room.

Using this technique, the engineer not only learns the overall Sound Transmission Loss
of the component but also gains insight as to where most of the noise is coming through
the component. The intensity testing will result in an intensity map of the product in
which transmission paths are highlighted as well as a plot of STL vs frequency.

In the reverberant room, an omnidirectional source is used to create a diffuse field. The
anechoic room is used to create a free-field condition to avoid any reflections that would
show up as incorrect localization spots. The material sample is placed between the two
rooms.

In the anechoic room, the sound intensity can be measured either using a sound
intensity probe or the Simcenter Soundbrush.

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Figure 9: Rooms setup for a sound transmission loss test using intensity.

When measuring STL using the two-room intensity method, Equation 1 can be modified
such that STL is calculated as follows:

Equation 2: where L_Pi is the incident soun pressure, L_It is the intensity measured on the material sample in the
anechoic chamber, S_i is the area incident, S_t is the area transmitted, and 6.18 is a constant accounting for a few
things like air density and speed of sound.

When measuring flat samples, S_i = S_t ---> S_i/S_t = 1

Therefore, when measuring flat samples,

A common automotive STL application with this technique is measuring a vehicle dash
panel as seen in Figure 10, below.

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Figure 10: A sample being measured using the two-room intensity method. The sample is mounted between a
reverberant room and anechoic room.

An intensity map of the dash panel can be created with the resultant data which lends
information as to where the leaks are.

Figure 11: The intensity map of the object can be viewed to learn which locations are transmitting the most sound.

In addition, the standard STL value vs frequency plot is also generated.

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Figure 12: The sound transmission loss can be viewed on a per-octave basis.

The intensity method is common for investigating dash panel, doors, and instrument
panels.

Sound pressure and sound pressure

The two-room method is based on a pressure difference between a sending room and a
receiving room (both of which are reverberant rooms). The sample is placed in an
opening connecting these two rooms.

In both rooms there is either a microphone on a rotating boom or a collection of

microphones in various locations. The function of the rotating microphone / multiple
mics is to measure and calculate the average sound pressures of both rooms. An
acoustic source is placed in the sending room.

Figure 13: Setup for measuring sound transmission loss using two reverberant rooms.

The sound pressure must be measured in both rooms.

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Sound Transmission Loss (STL) is calculated as a pressure difference between the rooms
with a correction for the receiving room.

Equation 3: where L_1 is the average SPL in the sending room, L_2 is the average SPL in the receiving room, S is the
test object area and A is the equivalent sound absorption (A = 0.16V/T) where V = receiving room volume and T =
reverberation time of the receiving room.

Figure 14: A material sample is being prepared to be inserted in the opening between the two rooms.

Again, with this method the sound transmission loss is measured versus frequency. An
example of the results sheet in Simcenter Testlab from the two-room method is below.

Figure 15: The result of a sound transmission loss test using the two-room method.

It is possible to do this testing with Simcenter Testlab Sound Transmission Loss Testing
using Rooms software.

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What is Sound Power?

How to measure how much sound a product produces?
Consider the speakers in Picture 1. For the purposes of this article, we will assume that
the speakers are producing a constant, steady state noise that does not vary over time.

Picture 1: Speakers making sound

To measure how loud the speaker sound is, one might believe it is as simple as placing a
microphone close to the object and measuring the decibel level as shown in Picture 2.

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Picture 2: Measuring speaker sound with microphone

But where to place the microphone? How far away? The further distance a microphone
is away from a sound emitting object, the lower the decibel value will be
(Picture 3). Distance certainly affects the sound readings.

Picture3: Sound measurement microphones located 1 meter and 2 meters away from speakers

In fact, in an acoustic free field, the sound pressure level drops by 6 dB as the distance
from the sound emitting object is doubled. Going from 1 meter to 2 meters away would
decrease by 6 dB. If a product had a requirement to be 50 dB, but the microphone
distance was not specified, one could simply place the microphone far enough away to
meet the requirement!

Even if microphones were placed at the same distance away, the decibel reading could
vary depending on the location relative to the object, as shown in Picture 4.

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Picture 4: Sound measurement microphones at the same distance from sound emitting object, but at different
distances.

A microphone placed behind the speakers will not read the same decibel level as a
microphone placed in front of the speakers.

How can one quantify how loud an object is, but independent of the distance or location
of the microphone? The answer is Sound Power.

Sound power attempts to quantify the acoustic source strength of an object,

independent of the distance and location of the measurement.

How is this done in practice?

Sound Power Measurements
There are different methods used to quantify the sound power of an object. A common
method is to surround the object with sound pressure microphones (Picture 5).

Picture 5: Hemi-sphere arrangement of microphones around test object, above a reflecting plane. Left: Drawing with
red dots representing microphone locations. Right: Actual sound power test system.

For example, the microphones might be placed around the object in hemi-sphere, to
capture all the sound emitted by the object in all directions. By taking an energy

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average of the microphones, one gets a measurement of sound that is independent of

location. See the “Sound Pressure” portion of Equation 1.

To normalize the microphone readings over distance, the surface area of the hemisphere
is calculated and then converted into decibels. See the “Surface Area” portion
of Equation 1. By calculating the surface area of the hemi-sphere, one can also make
measurements independent of the distance.

Equation 1: Sound power based on Surface Area and Sound Pressure

Equation 1 is the basic formula for Sound Power (Lw), where L is the sound pressure
level and w stands for watts (the units in which sound power is reported). Sound power
is typically reported in decibels referenced to 1 Picowatt (1 pW).

The equations have two major parts:

 Sound Pressures - The sound pressure readings from the different microphones
(Lp) are averaged together and converted into decibels
 Surface Area – The surface area over which the microphones are arranged are
converted to into decibels. The reference surface area (So) is 1 m2.

The sound power of an object is always the same no matter what size hemisphere is
used to measure the sound power. The pressures and surface area work in conjunction
with each other to make the total sound power always be the same (Picture 6).

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Picture 6: Cross section of hemi-sphere-shaped sound power arrangement. Left shows microphones close to object
which increases sound pressure readings but has smaller surface area. Right shows larger surface area with
microphones farther from object.

As the surface area gets smaller, the microphones are at a closer distance to the test
object:

 The closer the microphones are to the object, the higher the sound pressure
readings.
 The higher sound pressure readings are offset in the equation by the reduction in
surface area.

So, the total sound power (Lw) remains the same!

Conversely, as the area increases, the microphones get farther from the test object. The
farther the microphones are from the test object, the lower their sound pressure
readings.

The sound power equation is setup so that any changes in sound pressures are offset by
equivalent changes in the surface area, so the total sound power remains constant.

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Picture 7: Sound power spectrum test result – A-weighted decibels referenced to Watts versus octave bands

The final results of a sound power test would be an A-weighted octave spectrum (Picture
7). It has units of decibels referenced to Watts.

Correction Factors
Sound power tests are run in a variety of facilities of differing quality and performance.
Correction factors can be used to remove some of the variation found between test
facilities.

In fact, the Sound Power equation that was presented in Equation 1 assumes that there
are no other sound sources nearby. It also assumes that there are no reflective walls near
the test, other than the reflecting plane of the ground.

Picture 8: A virtual CAE simulation of a sound power test does not have extraneous background noises

In a virtual CAE sound simulation, this could easily be the case (Picture 8). But in real-
world practice, there can be reflections and other sound sources. The correction factors,
K1 and K2, are used to remove the effects of reflections and other sources, within
certain limits. These corrections are performed on an octave band basis.

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K1 – Background Correction
When performing a sound power test, a measurement is made without the test object
emitting noise. This is called the background noise measurement. This correction is
done per octave band. Depending on the levels of the background noise compared to
the actual test, a few corrective actions might be made:

 If the background levels are higher than the noise emitting object under test, the
entire test is not valid.
 If the background levels are extremely low compared to the test object, no
correction is made. This is the case for background levels 15 dB below or less than
the measurement levels.
 If the background noise is within a prescribed range compared to the actual test,
the contributions of the background noise can be subtracted from the total sound
power.

K2 - Reflections
Some test environments are not perfectly anechoic. Sound reflects back from areas
other than the reflecting plane, causing the sound power levels to be higher than they
should be. The amount of reflected noise can be quantified and corrected.
To do this, a reference sound source is measured in the test environment. The reference
sound source creates a repeatable, known sound power level. For a given octave band,
if the reference sound source should be 90 dB, but 91 dB is measured due to extra
reflections, the increase can be corrected.
These two correction factors are subtracted from the sound power value. See Equation2.

Equation 2: Sound Power equation with sound pressures, surface area, and correction factors

With the addition of correction factors, the sound power equation is now complete.

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Conclusion
Sound power attempts to quantify the acoustic source strength of an object,
independent of the distance and location of the sound measurements.

There are four main factors taken into consideration when calculating sound power
(Equation 2):

 Sound pressure measurements – An array of several microphones is used to

measure sound pressures around the sound source test object. The density of the
microphone arrangement is to make sure the level is measured correctly without
any location dependency.
 Surface Area – The surface area used for the microphone arrangement is
measured and used to compensate for the microphone distance from the sound
source test object.
 K1 Correction – Takes into account background noise of the test facility. Under
certain circumstances, the calculated sound power can be corrected for the
background noise
 K2 Correction – Corrects for extra reflections that would make the calculated
sound power higher than it should be.

Sound power is often used in noise regulations and legal certifications because it is not
location or distance dependent. ISO 3744 and other standards have in-depth details on
how these measurements are to be performed.

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Sound Absorption
When sound reaches a barrier, three things can happen as shown in Figure 1:

 Absorption – The sound is absorbed and dissipated as heat.

 Transmission – Sound can pass through the barrier.
 Reflection – Sound can be reflected off the barrier.

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Figure 1: Sound at a barrier can be absorbed, transmitted, or reflected.

The amount of absorption, reflection, and transmission of the sound is different for
every frequency.

For example, a high frequency sound with a short wavelength can be absorbed by a
thinner piece of material, while lower frequency sounds are not absorbed, due to their
longer wavelength.

This article will explore what variables can affect sound absorption and how absorption
can be measured.

Absorption Quantification
Absorption is can be expressed via the “absorption coefficient” (Equation 1) which can
have a value between 0 and 1.

Equation 1: Calculating the absorption coefficient.

Alpha represents the absorption coefficient.

 When the absorption coefficient equals one, all the sound is absorbed
 When the absorption coefficient equals zero, no sound is absorbed

Typically, the absorption coefficient for a given material is plotted as a function of

frequency as shown in Figure 2.

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Figure 2: Absorption curve.

Between 50 and 100 Hz in Figure 2, 100% of sound is absorbed. Below 50Hz, the
material does not absorb well. Thicker material may aid in helping to absorb the lower
frequencies.

Some absorption values for different materials are below. All values are approximate.

Figure 3: Absorption values.

Note that the softer the material, the more absorption. The more dense and hard the
material, the less absorption.

What affects the absorption of a material?

There are several things that affect absorption. Some factors include: material
composition, humidity, material thickness, and material position.

Material Composition

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Porous materials typically absorb sound better than very dense materials. Examples of
porous materials include cloth, foam, fiber glass, and acoustic tiles. Porous materials
present a larger amount of surface area to the advancing sound waves. The fibers or
particles of the porous material can vibrate and dissipate the sound as heat. Very dense
materials (concrete, cinder block, glass) tend to reflect most of the incident sound.

Humidity
The speed of sound is affected by the humidity in the air. The speed of sound increases
as humidity increases. As the speed of sound increases, the absorption also increases, as
shown in Figure 4.

Figure 4: Effect of humidity.

When performing an absorption test, samples should be stored for several days in a
humidity-controlled room to ensure they are at the desired humidity for the test.

Material Thickness
With all else being equal, increasing material thickness increases the absorption
performance at lower frequencies (Figure 5). It has been experimentally determined that
peak absorption of a frequency occurs when the material thickness is about one-quarter
the wavelength of the wave.

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Figure 5: Absorption performance versus material thickness.

Low frequency sounds have longer wavelengths; therefore, the material must be thicker
to absorb lower frequency sounds.

Material Position
Sound absorption depends on angle of incidence ( ) of the incoming plane wave. Often,
an “angle-averaged” value of absorption is used for design purposes.

Figure 6: The angle of incidence of the incoming plane wave.

In addition to the angle of incidence, material position relative to the supporting wall
matters.

For example, mounting the absorptive material flush with the wall versus mounting the
absorptive material and leaving an air-gap between the material and the wall will affect
the absorption coefficient.

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Figure 7: The effect of air between absorptive sample and reflective wall.

Experimental data shows that leaving one quarter of a wavelength between the wall and
the absorptive sample maximizes absorption at a given frequency.

This is because there is an inverse relationship between the pressure of the sound wave
and the air molecule velocity.

As the wave hits a wall, the molecule velocity slows down because the wall resists the
motion of the air molecules. At the wall, there is a high pressure, low velocity location.

 The high pressure, low velocity location is called a node.

 The low pressure, high velocity location is called an anti-node.

In the areas of low pressure, the molecules are less densely spaced. Therefore, in the
low-pressure areas, the molecules can move more freely and at higher velocities.

Figure 8: The inverse relationship between pressure and molecular velocity. In areas where the pressure is high, the
molecular velocity is low. In areas where pressure is low, molecular velocity is high.

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Sound is absorbed when there is friction between the air molecules and the absorptive
material causing the sound energy to dissipate as heat.

 The higher the velocity of the air molecules, the more friction and the more
absorption.
 The lower the velocity of the air molecules, the less friction and the less
absorption.

Looking at the graphic, you can see that the area of low pressure, high velocity, is one
quarter wavelength away from the wall. Therefore, the absorptive material should be
placed one quarter wavelength away from the wall to maximize the absorption of the
wave.

Figure 9: Absorptive material (pink) is placed one quarter wavelength away from the wall to maximize absorption.

The concept of placing the absorptive material one quarter wavelength away from a wall
is sometimes referred to as the “Quarter Wavelength Rule”.

Note that placing the material one quarter wavelength away from the wall maximizes
absorption for that corresponding frequency. If it is desired to absorb a broadband range
of frequencies, a thick piece of material must be used so that there is material “one
quarter wavelength” away from the wall at many frequencies. There is a trade-off
between material cost and the range of frequencies targeted with the “quarter
wavelength rule”.

Measuring Absorption
There are two methods to measure a material’s absorptive properties: the tube method
and the room method.

 The tube method uses direct incidence. In this case, the sound waves all approach
the sample at the same angle. See Figure 10.

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 The room method uses random incidence. In this case, the sound waves approach
the sample at random angles. See Figure 10.

Figure 10: The blue arrows represent sound waves. Direct incidence: all sound waves approach sample at same
angle. Random incidence: sound waves approach the sample at random angles.

Measuring Absorption with an Impedance Tube: Direct Incidence

Hardware
Measuring a material’s absorption can be done with an “impedance tube”. These tubes
are typically made of straight “sound proof” tubing (typically thick steel).
One end of the tube is connected to a sound source which outputs a broadband range of
sound waves. The other end of the tube holds the sample to be tested. Therefore, the
sound waves approaching the sample are both direct incidence and normal to the
sample.
The tube ends in a rigid termination. A pair of microphones are positioned just before
the sample. See Figure 11 below.

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Figure
11: Top: Impedance tube. Bottom: cross section of the impedance tube to show sample location within the tube.

Sample Preparation
Samples of the test specimen must be cut to fit within the tube. Care must be taken
when cutting the samples as the boundary conditions between the sample and the tube
can greatly affect the resulting measurements.

Some impedance tubes come with sample cutters that are designed to cut samples to
the exact diameter of the tube. But, these material cutters can slightly compress the
sample during cutting, altering the samples absorption properties.

Figure 12: Material cutter and some cut samples.

A better alternative to a sample cutter is using a waterjet to cut the material. A CNC
waterjet will produce an accurately cut, repeatable sample.

A well-cut sample will fit flush with the tube and will neither compress nor leave a gap
between the tube and the sample diameter.

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Figure 13: Example of a well-cut sample.

An improperly cut sample could be cut too large and compress when placed in the tube
or be cut too small and leave a gap between the sample and the tube wall.

Figure 14: Two examples of improperly cut samples.

In the above figure, the first sample was cut too large: the edges of the sample are
compressed in the tube. The second sample was cut with too small: there is a gap
between the tube and the sample.

Test
To test the absorption of the sample, a broadband signal is output by the speaker.
Multiple averages are taken to ensure that random noise on the measurement is
averaged out.

The result of the measurement is a graph of the absorption coefficient vs. frequency. See
Figure 15 below.

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Figure 15: The result of an absorption test in tube is the absorption coefficient of the material vs. frequency (bottom
graph).

The above figure is taken from the Simcenter Testlab Absorption Testing using
Impedance Tubesoftware.

Measuring Absorption with a Reverberation Room: Random Incidence

It is not always practical to test a material’s absorption in an impedance tube. The
impedance tube only tests a material’s absorption in a direct field (Figure 10). This does
not accurately represent real world situations in which the sound is often random
incidence (Figure 10). This is where testing using a reverberation room is valuable.

A reverberation room is designed to have hard, reflective walls built at oblique angles, so
no walls are parallel to each other. This causes the sound waves to be reflected a
maximum number of times around the room to help create a diffuse field.

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Figure 16: Reverberation room are built with walls at oblique angles.

Often, hemi-spherical features are added to large walls to increase wave diffraction
(spreading out), adding to the diffusivity of the chamber. One of these hemispheres can
be seen in Figure 17.

Figure 17: Hemispheres are added to increase wave diffraction.

Common items to test using the reverberation room method are material samples
(usually about 1m x 1m), acoustic panels, and even furniture.

The test requires two measurement conditions:

 Step 1: Measuring the reverberation time in the room with no test sample
 Step 2: Measuring the reverberation time in the room with the test sample

It is recommended to take multiple measurements for each test condition and average
the results to remove the effects of uncorrelated noise on the measurement.

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Step 1: Characterize the absorption of the empty reverberation room

The first step is to characterize the reverberation room.

We need to test the room with and without the sample. The difference in reverberation
time between the two conditions gives an indication of the absorption of the sample.

The absorption value is closely related to the reverberation time. See ISO 354:2003 for
the exact equations and specs.

To determine the reverberation time of the room, a T60 decay time is used. This is the
time in seconds it takes for the sound pressure level (SPL) to decrease by 60dB after the
sound source is stopped.

The sound source is played for a few seconds, then stops. After there is 5dB of
attenuation, the decay time measurement starts (green dot in Figure 18). After 60dB of
additional attenuation, the decay time measurement stops (red dot in Figure 18).

Figure 18: T60 decay time.

It’s important to note that it is not always to possible to play a sound loud enough to get
60 dB of attenuation. Therefore, T20 or T30 decay times are often used. These work the
same way at the T60 decay time except they only require 20 dB or 30 dB of attenuation
respectively. Even though the decay is only measured over 20 dB or 30 dB of
attenuation, the decay times are projected for a 60-dB decay. So even if the T20 or T30
methods are used, times are still reported as T60.

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Figure 19: In Simcenter Testlab, it is possible to calculate the T20, T30, and T60 reverberation times in the Online
Processing workbook.

Of course, the decay time can change with frequency, and is therefore measured for
each octave band (see Figure 20).

Figure 20: The decay time is plotted for each octave band. This graph shows the decay time per octave band.

The reverberation time is closely related to the absorption of the empty room. After
calculating reverberation time for each octave band, the absorption of the empty room
is then calculated for each octave band.

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Step 2: Characterize the absorption of the test sample

After characterizing the room, we now need to characterize the material sample.
A common test sample is a large material sample (typically about 1 meter x 1 meter).

Figure 21: Setup to measure absorption of a material sample (purple). Multiple microphones (grey circles) and
sources (red boxes) are used in the reverberant room.

Now the reverberation time of the room with the sample in it is calculated.

The reverberation time of the room with the sample should be shorter than the
reverberation time of the room without the sample. This is because we expect the
sample to add absorption to the room.

After calculating reverberation time for each octave band, the absorption of the empty
room is then calculated for each octave band.

Steps 1 and 2 are now completed:

 Step 1: Measuring the reverberation time in the room with no test sample.
Calculate absorption of empty room.
 Step 2: Measuring the reverberation time in the room with the test sample.
Calculate absorption of room with sample.

To determine the absorption of just the test sample, the absorption of empty room is
subtracted from the absorption of the room and sample.

Equation 2: the absorption of the sample is calculated by subtracting the absorption of the empty room from the
absorption of the room with the sample in it.

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When testing an absorber in a reverberant room, the results of the test is the absorption
coefficient with respect to frequency.

Figure 22: The calculated absorption coefficient vs. frequency (bottom right).

The above is a screenshot of Simcenter Testlab Sound Absorption in Room Software.

Conclusion
There are two common methods to measure sound absorption: in an impedance tube
and in a reverberant room. Measuring in an impedance tube results in the absorption
coefficient vs. frequency. Measuring in a reverberant room results in the absorption
coefficient vs. frequency or the equivalent absorption area of the test sample vs.
frequency.

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Noise Level Certification:

how to select the right
standard?
Siemens PLM software has created a new Guide to measuring Sound Power to provide
assistance to manufacturers who need to do noise certification testing for their products.

Background
Within the European Union (EU), there are explicit limits on how much noise outdoor
machinery can produce, putting pressure on manufacturers and end users.

Producers of equipment both made outside and inside of the EU are affected by this
legislation if they wish to sell products to EU countries. Anyone who wants to sell certain
equipment on the European market has to measure its emitted sound power level and
stay below the specified target set by national legislations.

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Picture 1: European energy label including sound power declaration (lower right)

Other types of equipment do not need to meet specific targets, but the standards and
directives require manufacturers to label (Picture 1) their products with noise
declarations; their guaranteed sound power levels.

Standards that specify measurement procedures for a particular type of machinery or

equipment are called the noise test codes, or C-standards. Following these standards
ensures reliable and repeatable results.
Standards

Choosing the correct standard for qualifying an acoustic source can be a challenge. The
decision must consider a number of factors: Type of noise source, available equipment
and testing environment, regulations, etc.

Standards on specific products include:

 EC/2000/14 directive on noise emission by equipment for outdoor use

 ISO 15744: Hand-held nonelectric power tools
 ISO 7779/ECMA-74: Information technology and telecommunications equipment
 ISO 9296/ECMA-109: Declared noise emission values of computer and business
equipment
 IEC 60704: Household electrical appliances
 DIN 45635 standards
 ISO 6393: Earth-moving machinery measured in stationary test conditions
 ISO 6395: Earth-moving machinery measured in dynamic test conditions

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Measuring of Sound Power

Sound power is expressed in units of watts [W]. It is a measure of the acoustic power of
a product that is independent of the environment, distance or direction. Measuring
sound power is not a straightforward process.

Sound power can be determined either through the measurement of sound pressure
[Pa] or sound intensity: the rate of energy flow through a unit area, expressed in watts
per unit area [W/m2].

Sound pressure-based sound power measurements (Picture 2) can only be performed in

very specific environmental conditions, usually only met in acoustic rooms (anechoic or
reverberant chambers).

Picture 2: Fixed array microphone method of sound pressure-based sound power measurements. Microphones are
arranged in a fixed arrangement around the object under test.

The sound pressure-based approach is most commonly used when performing

certification measurements. A set of ISO standards governs these requirements and
indicates measurement procedures that are necessary to obtain quality results.

Sound pressure-based sound power standards include 2000/14/EC, ISO 3744, ISO 3745,
ISO 6394, ISO 6395, and ARI 260.

Sound intensity-based sound power can be measured in any sound field, but certain
requirements regarding the type of sound must be met. The measurements can be
performed on individual machines or sound sources in the presence of other
components radiating noise, because steady background noise does not contribute to
the measured sound intensity (Picture 3).

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Picture 3: Sound intensity-based sound power measurements

However, this approach has a limited usable frequency bandwidth, time demanding
measurement procedures and limitations on the characteristics of the noise source.

ISO standards govern sound intensity-based measurements and describe the procedures
that are required to obtain quality results. ISO standards for intensity-based sound power
include ISO 9614-1 and ISO 9614-2.

Conclusions
Choosing the correct standard to follow can be a challenge. The decision must consider
several factors: Type of noise source, available equipment and testing environment,
regulations, etc.

This Guide to measuring Sound Power provides an overview of the existing standards as
well as the applicable regulations and noise codes, but the decision to execute which
tests which tests and measurements ultimately rests with the product manufacturer.

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Octaves in Human Hearing

Octaves are groups of frequencies that help quantify how humans distinguish between
frequencies.

Octaves represent the overall level of energy over a specific frequency range.

Figure 1: An octave map. Each vertical block is an octave and represents the overall level of sound energy over that
range of frequencies.

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How are octave bands determined?

The term “octave band” is borrowed from music theory where there is a doubling of
frequency between notes of the same name.

Figure 2: In music, there is a doubling of frequency between notes of the same name.

Octave bands in human hearing are developed in the same manner: the range of human
hearing (20-20kHz) is divided into eleven octave bands, each band having double the
frequency span of the previous band. These are called the 1/1 octave bands.

Figure 3: The lower, upper, and center frequencies of the 1/1 octave bands over the human hearing range.

To more closely match how humans distinguish frequencies, each 1/1 octave band can
be split into three bands. These are called the 1/ 3 octave bands. These smaller bands
more closely represent how humans distinguish between frequencies.

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.
Octaves are not equally spaced
Octaves are often displayed so that the center frequencies
of the octave bands are linearly spaced. This is called octave
format. This causes the octaves to appear equally spaced in
frequency even though they are not.

In Figure 5 below, white noise is expressed in both narrow

band (blue) and octave format (green). The x-axis is in
octave format (described above). Notice that the octave
bands at lower frequencies have less data filling them
compared to the octaves at higher frequencies. This is
because octave bands at higher frequencies cover a broader
frequency range than octave bands at lower frequencies.

Figure 4: The upper, lower, and center frequencies for the 1/3 octave bands

Figure 5: X-axis in octave format. The same random white noise is expressed in octaves and as an autopower.

The same data is plotted below but the x-axis is in linear format. Notice that an
autopower (blue) has the same density at all frequencies. Notice that the range of
frequencies each octave band (green) covers gets larger and larger.

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Figure 6: X-axis in linear format. The same random white noise is expressed in octaves and as an autopower.

Octave bands help determine how the human ear distinguishes between frequencies. At
lower frequencies, the ear can more easily distinguish between frequencies. Thus, the
octave bands are narrower. At higher frequencies, the ear has difficulty distinguishing
between frequencies (even frequencies that are somewhat far apart). Thus, the octave
bands are wider.

How are octaves calculated?

Octaves are calculated using a series of time-domain based, band-pass filters.
Time data is fed into a series of narrow-band filters. Each filter band represents the
frequency range of each octave. In Simcenter Testlab, the filter shapes conform to the
ANSI and IEC standards (ANSI S1.11-2011, IEC 61261:2014).

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Figure 7: Time data is fed through narrow band filters. Each filter represents an octave band. The overall level of the
time data is plotted on a per-octave basis and then plotted.

Both octave maps (like in the figure above) and octave sections can be calculated.

 Octave maps plot the octave bands vs. the overall level of energy in those bands.
 Octave sections track how a single octave band behaves against a tracking
parameter.

For example, the figure below shows how the 12.5 Hz, 20.0 Hz, and 25.0 Hz octave
bands change with time.

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Figure 8: An octave section tracks how an octave behaves against a tracking parameter. In this case, it is tracking the
octave against time.

Calculating Octaves in Simcenter Testlab

To calculate filter-based octaves in Simcenter Testlab, go to “Tools -> Add-ins” and turn
on “ANSI-IEC Octave Filtering”. This add-in requires 23 tokens.

Figure 9: Turn on “ANSI-IEC Octave Filtering”.

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In Time Data Processing, open the “Acquisition parameters” settings button (“Change
Settings”).

Figure 10: The change settings dialog for octave averaging.

Select an averaging method: exponential or linear. The exponential average weights the
acquisitions taken later in time more heavily than earlier acquisitions. The linear average
calculates the arithmetic mean of all the values.

Select a sound level type: fast (0.125 sec), slow (1.000 sec), impulse (0.035 sec), or user
(custom). This parameter changes the length of the averaging frame. The averaging
frame is the length of time that is used to calculate an average at every increment.
Check out the Simcenter Testlab Throughput Processing Tips article for more information
about increment and frames.

In Time Data Processing, open the “Section” settings button by clicking “Change Settings”
under “Section. To calculate an octave map, go to the “Octave Maps” tab. Check on the
octave band type that needs to be calculated.

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Figure 11: Calculating octave maps. In this case, 1/3 octaves will be calculated.

A result of this calculation would look something like Figure 12 below.

Figure 12: The result of an “octave map” calculation.

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To calculate a single octave section, go to the “Octave Sections” tab. Type in the center
frequency of the octave section(s) that is desired to be calculated.

Figure 13: Octave sections to be calculated.

The octave section calculation will calculate how an octave behaves vs. a tracking
parameter. Figure 14 below gives an example of the result from this calculation.

Figure14: Example of how various octaves behave vs. time.

NOTE: There are a few additional tabs with the RTO (Real Time Octaves) suffix. Anything
with the suffix RTO will used filter-based octaves to do processing. For example,
“Psychoacoustic Metrics RTO” uses time-based filters to calculate sound metrics.

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Octave displays in Simcenter Testlab:

There is a special display in Simcenter Testlab called the “Octave” display. The octave
display automatically puts the x-axis in octave format. It also automatically puts data into
a “block display” format (as octaves are traditionally displayed in). The icon for this
display is enlarged in Figure 15 below.

Figure 15: The octave display icon overlaid on an octave display.

When data is dropped into an octave display, it will look similar to the data in Figure 12
above.

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Basics: What is a decibel (dB)

anyway? Why is it used?
Background
The decibel (abbreviated as ‘dB’) seems to be everywhere in the world of NVH
measurements. It may seem like a unit of measurement (as it is typically shown on the
Y-axis) but really, it’s not, it is unit-less. We see it used in acoustics, vibration,
electronics, telephony, audio engineering & design…. But what is this unitless-unit, why
is it used, and how do we use it correctly?

The decibel was originally developed and used by the telephone industry to quantify
power loss in telegraph and telephone signals when sent through long cables. It is
named in honor of Alexander Graham Bell, a pioneer in the field of
telecommunication. While a decibel is defined as one tenth of a Bel, the Bel unit is rarely
used.

Formulation
The decibel is really nothing more than a logarithmic ratio between two numbers – a
measured value and a reference value. It is shown in two forms below: Equation 1 for
POWER quantities, and Equation 2 for field AMPLITUDE quantities.

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Examples of POWER quantities: Sound Power (Watts), Sound Intensity (Watt/m2),

electrical Power, electrical Intensity, etc.

Examples of AMPLITUDE quantities: Pressure (Pa), Voltage (V), Acceleration (m/s2),

temperature, etc. The amplitude of the field quantities should be in RMS.

As the decibel value depends entirely on the ratio between a measured value and
the reference value, it is therefore critical to select the proper reference for the
calculation. This is particularly important when comparing values between tests or
measurements.

In acoustics, dB are often used to report sound pressure level (SPL). The reference for
pressure in Pascals has been established as 20 micro Pascals (20e-6 Pa). This value
represents the average human hearing threshold at 1000 Hz, or the smallest pressure
fluctuation perceivable to the average human ear at 1000 Hz.

Let’s look at a sample calculation to see how it works.

Example
Using your Simcenter SCADAS hardware and Simcenter Testlab software, suppose you
set up a microphone to record an orchestra. While they are tuning their instruments,
you make a quick recording. The RMS amplitude of the sound reads 1.084
Pascals. What is the dB amplitude of the sound?

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Frequency spectrum shown in Amplitude format

Since sound pressure is an AMPLITUDE quantity, we will use the formulation below

Remember our dB Reference for Sound Pressure Level is 20 micro Pascals. Filling in our
equation we get the following:

You can check your work by having Simcenter Testlab display your recording in decibel
format.
Right-click on the Y-axis, select “Format”

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Then select “dB/Level”

Looks like we got the correct answer! Of course this is ONLY because we chose the same
reference value as Simcenter Testlab used.

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Double Check - Simcenter Units Editor

You can always check to see what the default reference value that Simcenter Testlab will
use by looking in the Simcenter Configuration & Units System editor and finding your
particular quantity.

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Sound Fields: Free versus

Diffuse Field, Near versus Far
Field
In the world of acoustics, there are many terms that are used to describe the acoustic
field around a sound emitting object. Four of the most important are listed below:

 Near Field
 Far Field
 Free Field
 Diffuse Field

This article explains the differences and usage of these acoustic sound field terms.

Near Field Versus Far Field

As one may suspect, the acoustic terms “near field” and “far field” have to do with the
physical distance from the sound source (Figure 1). Depending on how far away an
observer is from a sound emitting object, the acoustic energy produced by the sound
source will behave quite differently. It is therefore important to understand these
differences, and design measurements carefully.

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Figure 1: Sound waves behave differently in the near field (A) and far field (B).

Far field
The acoustic far field is defined as beginning at a distance of 2 wavelengths away from
the sound source and extends outward to infinity (Figure 2). As wavelength is a
function of frequency, the start of the far field is also a function of frequency.

Figure 2: The far field begins at 2 wavelengths away from the source.

In the far field, the source is far enough away to essentially appear as a point in the
distance, with no discernable dimension or size. At this distance, the spherical shape of
the sound waves has grown to a large enough radius that one can reasonably
approximate the wave front as a plane-wave, with no curvature (Point B in Figure 1).

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At this distance, sound pressure level is governed by the inverse square law, and a single
microphone sound recording will give reliable & predictable results. For each doubling
of distance away from the source, the sound pressure will drop 6 dB in the far field.

In many acoustic standards, measurements are often specified at a distance of at

least one meter from the sound emitting object to ensure that the measurement is taken
in the far field for the most critical frequencies.

Near Field
When close to a sound emitting object, the sound waves behave in a much more
complex fashion, and there is no fixed relationship between pressure and distance. Very
close to the source, the sound energy circulates back and forth with the vibrating surface
of the source, never escaping or propagating away.

These are sometimes called “evanescent” waves. As we move out away from the source,
some of the sound field continues to circulate, and some propagates away from the
object (Figure 3).

Figure 3: The near field is complex, with sound energy both circulating and propagating.

This transition from circulating to propagating continues in an unpredictable fashion

until we reach the threshold distance of 2 wavelengths, where the sound field strictly
propagates (the far field.)

This mix of circulating and propagating waves means that there is no fixed relationship
between distance and sound pressure in the near field and making measurements with a
single microphone can be troublesome and unrepeatable. Typically, measuring in the
near field requires the use of more than one microphone (Figure 4) to accurately capture
the energy borne by the circulating and propagating waves.

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Figure 4: Acoustic arrays featuring many microphones can be used close to a source to accurately capture sound
energy in the near field.

Free Field versus Diffuse Field

When sound radiates from an object, it can reach an observer directly by traveling in a
straight line, or indirectly via reflections. Reflected sound waves can bounce off surfaces
such as walls, the floor, ceiling, as well as other objects in the area. Often when we
experience sound, we are receiving both direct and reflected sound waves. Under
carefully controlled circumstances, however, we can experience the extreme ends of this
continuum: 1) an acoustic field where zero reflections are present, and only the direct
sound is observed, and 2) the opposite acoustic field, where zero direct sound is
observed, and only reflected sound is present. The names given to these two extreme
acoustic environments are FREE FIELD and DIFFUSE FIELD respectively (Figure 5).

Figure 5: Illustrations of the free field (zero reflections) and diffuse field (only reflections).

Free Field
In an acoustic free field there are no reflections; sound waves reach an observer directly
from a sound emitting object. The sound wave passes the observer exactly once, and
never returns.

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Two common examples of acoustic free fields are:

 The sound source is far enough away that it appears as a single point source, far in the
distance. Visualize an airplane flying high overhead on a clear day.
 An anechoic chamber is a special facility constructed to approximate an acoustic free
field by using materials to absorb sound waves before they can be reflected (Figure 6).

Figure 6: An anechoic chamber is used to approximate a free field.

In an anechoic chamber, specially designed fiberglass wedges cover the walls, floor and
ceiling to absorb sound, so it is not reflected. To be effective (especially at low
frequencies,) these rooms need to be very large, with long wedges, and often feature
mechanical isolation from the surrounding building and foundation so no vibration is
transmitted to the chamber.
Diffuse Field
A diffuse field describes an acoustic field where sound waves reach the observer from all
directions. The reflected sound is of similar magnitude to the direct sound when it
reaches the observer, and as a result, does not appear to have a single source. A
microphone in a diffuse field measures the same magnitude regardless of orientation or

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location; the sound level is the same everywhere. A reverberant chamber for acoustic
material testing is shown in Figure 7.

7: A reverberant chamber has highly reflective walls to create a diffuse sound field.

A reverberation room is designed to have reflective walls built at oblique angles so no

walls are parallel to each other. This causes the sound waves to be reflected a maximum
number of times around the room to help create a diffuse field. Often, hemi-spherical
features are added to large walls to increase wave diffraction (spreading out), adding to
the diffusivity of the chamber. One of these hemispheres can be seen in Figure 7.

How to Know
It is difficult to determine by visual inspection what type of acoustic field is present.
Using acoustic measurements, the following can be observed:

 If in a free field, far field acoustic environment, there is a 6 decibel decrease in the
measured sound pressure level when doubling the distance from a sound emitting
object. This behavior is explained by the inverse square law.
 In a diffuse field, like a reverberant chamber, the sound level is the same, no matter
where the microphone measurement recording is made.
 In a perfectly diffuse sound field the sound intensity is zero.

Conclusion

Acoustic field behavior is an important consideration when measuring sound. It is important to

know the field type so sound measurement levels can be properly interpreted.

In some circumstances, it is possible to apply diffuse or free field corrections to adjust the
measurement levels. Diffuse and free field corrections are often provided for microphones,
headsets, and binaural measurement devices.

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Sound Pressure
If you are new to the field of noise & vibration, the terms sound pressure, sound power,
and sound intensity may be a bit confusing. This is understandable, as they are all
commonly used, as well as interrelated… Not to mention they are all often expressed
in decibels, which can be confusing in its own right. However, they each represent a
different, important aspect of sound and how it is transmitted and experienced. In a
three-part series of posts, we will look at these three quantities of sound and see how
they are all measured, calculated and used.

Sound Pressure

Sound pressure is the foundation of most acoustic work not only because it is a quantity
analogous to our sense of hearing, but also because sound pressure measurements are
one of the only measurements for sound one can actually make! As we’ll see, sound
pressure measurements are the foundation of both sound power and sound intensity
calculations.

When an object makes sound, it does so by vibrating back and forth. This causes the air
molecules next to the object to vibrate as well. This vibrating chain reaction continues
outward (at the speed of sound) away from the object in the form of waves. These

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waves are analogous to the waves formed in water when a pebble is dropped into a
pond.

Sound propagates away from a source in all directions

As the name suggests, we use the unit of pressure (Pascals, or Newton/meter^2) to

quantify sound pressure. This value represents the summed amplitude of all the
different sine waves that make up a sound (also called the “Overall Level”). It should be
noted, however, that this pressure is just the alternating portion of the pressure our ears
(and microphones) are subject to. We also live under a huge amount of “static” pressure
due to the earth’s gravitational pull on our atmosphere. This is commonly referred to as
“atmospheric pressure.” Atmospheric pressure at sea level is about 101.3 kPa, or 194 dB!

However, because atmospheric pressure is constant for the most part, and since we are
really only interested in the alternating portion of the pressure signal, we generally
subtract atmospheric pressure and normalize sound pressure levels to be reported as the
difference above/below zero. As we see in Figure 1 below, a normalized sound wave
creates pressures that are both above and below zero, corresponding to the red and blue
shaded regions respectively. Even though the normalized sound pressure is both
positive and negative, we only report the amplitude of the pressure wave as being
positive. This amplitude can be described using Peak, Peak-to-Peak, or RMS
scaling. When we hear a sound, our brain acts as an integrator of these positive and
negative oscillations, and we perceive a steady positive amplitude, we do not perceive
the actual fluctuation of the individual sine waves.

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Figure 1: Sound pressure amplitude can be reported in a variety of formats.

Measuring Sound Pressure

A traditional engineering microphone detects sound waves in the air by measuring the
displacement of a thin metallic membrane inside the microphone. This is very similar to
how our ear works – our ears have a thin membrane called the ear drum that behaves
the same way as a microphone membrane. The membrane begins to vibrate as the
pressure waves reach the microphone - the larger the amplitude of the wave, the more
displacement in the microphone membrane, the larger the signal sent from the
microphone (Figure 2).

Figure 2: Microphones translate sound pressure into a voltage signal.

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In Figure 2-A, we see small amplitude sound waves hitting the microphone, causing the
microphone membrane to vibrate back and forth with a small amplitude. This relative
motion between the membrane and an electrically charged disc called the “back plate”
results in a capacitive difference. This difference generates a voltage output from the
microphone that is proportional to the membrane displacement. In Figure 2-B, we see
the same sound source outputting a higher amplitude sound wave, which causes the
microphone membrane to vibrate with a higher amplitude, and thus output a larger
voltage.

Pressure Propagation
Just like the waves in the pond, the sound waves propagate away from the source in all
directions. This causes the wave to spread out, and as a result the amplitude goes down
as a function of distance. This is because we put a fixed amount of energy into the
water with our pebble - for the amount of energy transferred to the water to remain
constant, the amplitude of the wave must decrease as the wave front gets larger. This is
also why things are louder up close: the amplitude of the sound waves is larger nearer to
the source. As we move farther away, the same amount of energy is spread out over a
larger area, so the amplitude gets smaller. In fact, when no reflections are present, the
amplitude goes down by exactly half when we double our distance from the source. We
are going to use this property of sound to our advantage for both sound power and
sound intensity. Stay tuned for more!

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What is A-weighting?
A-weighting is a frequency dependent curve (or filter) which is applied to sound pressure
microphone measurements to mimic the effects of human hearing.

Given the same sound pressure levels, microphone recordings can be very different than
the levels perceived by the human ear (Figure 1).

Figure 1: A microphone (left) and human ear (right) will record/perceive sound differently.

There are several reasons why there can be difference in sound levels between a
microphone recording and human ear perception of sound.

The sound pressure on the microphone diaphragm versus the ear drum (Figure 2) can be
very different, even in the same sound field:

 The air volume in the ear canal has a resonance around 4000 Hz, causing higher
sound levels on the ear drum than the microphone diaphragm
 The presence of human head, torso and outer ear interfere with and alter the
sound field

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Other differences are due to the cochlea hearing organ (Figure 2) and psychological
effects:

 The hearing organ, or cochlea, has difficulty detecting sounds at low frequencies
and very high frequencies
 As humans age, damage to the cochlea causes high frequency hearing loss
 The cochlea has a logarithmic shape, causing humans to be better able to
distinguish changes in pitch at lower frequencies than higher frequencies

Figure 2: Human Ear: Pinna, Ear Canal, Middle Ear, and Cochlea

Unlike the human ear, a microphone is not surrounded by a torso, pinna, and ear
canal. There is no logarithmically shaped hearing organ in a microphone. It is no
wonder there are differences between sound perceived by the human ear and a
microphone recording!

The A-weighting Curve

A-weighting is an established, standard curve that attempts to alter the sound pressure
levels of recorded by a microphone measurement to more closely match the perception
of the human ear.

The A-weighting curve (Figure 3) shows decibels of attenuation or gain at every

frequency over the range of human hearing. This gain/attenuation is applied to the
microphone measurement.

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Figure 3: A-weighting curve - dB of accentuation vs frequency

Typically, this gain/attenuation can be applied via:

 Analog filter built into an electric circuit in the recording device

 Digitally applied after the recording is stored on a computer

Some key features of the A-weighting curve:

 Below 1000 Hz, the sound levels are attenuated

 At 1000 Hz, there is no gain or attenuation
 Between 1000 and about 6000 Hz, the levels are amplified a few decibels
 At about 6000 Hz and higher, the sound levels are attenuated

Sound calibrators often generate tones at 1000 Hz to take advantage of the

characteristics of the A-weighting filter. The calculated calibration values cannot be
affected by whether the A-weighting filter is present or not. Operators cannot
accidentally forget to turn on/off the A-weighting filter and unintentionally change the
calibration value.

How is the A-weighting curve used?

Suppose you measure 80 dB with a microphone at 100 Hz. If you look at the A-weighting
curve at 100 Hz, an attenuation value of -20 dB is indicated (Figure 4).

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Figure 4: An -20-dB attenuation at 100 Hz

Subtracting 20 dB from 80 dB yields 60 dB. To indicate that A-weighting was applied to

the signal, the recorded value would be annotated as 60 dB (A) or 60 dBA.

To remove any doubts, it is also customary to annotate the original, unweighted values
as “Linear”. For example, “80 dB Linear” or “80 dBL” or “80 dB (L)”.

This procedure is done at every frequency in the spectrum. It can also be applied to an
octave spectrum as well.

Is dB (A) always lower than dB?

The A-weighting curve attenuates a great deal of the frequency range. Intuitively, one
might guess that the overall dB (A) value will always be lower than the original overall
dB (or dBL) value. This is not always the case.

For example, observe the spectrum shown in Figure 5. The overall dB value is 84.2 dB
without A-weighting applied.

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Figure 5: Spectrum of human whistle, no weighting applied

Now observe the spectrum shown in Figure 6 which has A-weighting applied. The
overall dB value is 85.1 dB (A).

Figure 6: Spectrum of human whistle with A-weighting applied

The A-weighted dB value of 85.1 dB (A) is higher than the Linear weighted value of 84.2
dB. Why? This is because the whistle, with a frequency of approximately 1600 Hz, is in
the frequency range (1000 to 6000 Hz) of the A-weighting curve where sound levels are
amplified. The net effect of this amplification is greater than the reduction in the
spectrum at all other frequencies. For example, the levels below 1000 Hz are greatly
reduced by introducing A-weighting, but because they are lower in amplitude their net
effect on the overall level is less than the amplification of the 1600 Hz whistle.

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Table of Values
Table 1 contains a list of the A-weighting attenuation and amplification values in dB for
1/3 octave center frequencies.

History
The A-weighting curve was derived from the Fletcher-Munson curves of equal
loudness. The American National Standards Institute (ANSI) was the first to implement
the curve in Sound Level Meter standard published in 1936.

It should be kept in mind that the curve is not a perfect representation of effects of
human hearing. To be able to develop cost effective analog circuits, the curve had to be
simplified. For example, the A-weighting curve does not change as a function of the
sound level like human hearing.

Other Types of Weighting

In addition to A-weighting, there are other acoustic weighting functions. They include B,
C and D weighting as shown in Figure 7.

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Figure 7: A, B, C, and D Weighting Curves

Used in various applications and industries, the B and C weighting curves are like A-
weighting, but do not have as much attenuation below 1000 Hz. The C weighting curve
is the flattest of the A, B and C curves. The D-weighting curve is typically used in very
high pressure aeronautical noise applications, like airplane flyover noise.

Simcenter Testlab: Applying A-Weighting

In any graph in Simcenter Testlab, A-weighting can be applied by right clicking on the Y-
axis as shown in Figure 8. Select “Processing -> Weighting -> A”.

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Figure 8: Right click on Y-axis and select “Processing -> Weighting -> A” to apply A-weighting.

Applying A-weighting can also be done in colormap and octave displays as well.
This works inside the standard Simcenter Testlab, as well as Active Pictures in
PowerPoint, Word, Excel, etc.

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All Articles - Siemens PLM

Community Testing
Knowledge Base
Siemens PLM Community

Digital Signal Processing

 What is Fourier Transform?

 Digital Signal Processing: Sampling Rates, Bandwidth, Spectral Lines, and more...
 Gain, Range, Quantization
 Aliasing
 Overloads
 Averaging Types: What's the difference?
 Kurtosis
 Time-Frequency Analysis: Wavelets
 Spectrum versus Autopower
 Autopower Function...Demystified!
 Power Spectral Density
 Shock Response Spectrum (SRS)
 Windows and Leakage
 Window Types
 Window correction factors
 Exponential Window Correction Factors
 RMS Calculations
 The Gibbs Phenomenon
 Introduction to Filters: FIR and IIR

Modal Data Acquisition

 Simcenter Testlab Impact Testing

 Modal Impact Testing: User Defined Impact Sequence
 What modal impact hammer tip should I use?
 Modal Tips: Roving hammer versus roving accelerometer
 Modal Testing: Driving Point Survey – What’s at stake?
 Attach Modal Shaker at Stiff or Flexible Area of Stucture?
 Using Pseudo-Random for high quality FRF measurements
 Multi Input Multi Output MIMO Testing
 Ground Vibration Testing and Flutter

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Modal Analysis and Operational Deflection Shapes

 Natural Frequency and Resonance

 What is Frequency Response Function (FRF)?
 The FRF and it Many Forms!
 How to calculate damping from a FRF?
 Getting Started with Modal Curvefitting
 Modal Assurance Criterion
 Simcenter Testlab Modal Analysis: Modification Prediction
 Import CAD into Simcenter Testlab
 Animate CAD Geometry
 Alias Table: Mapping Test Data to Geometry
 Geometry in Simcenter Testlab
 Maximum Likelihood estimation of a Modal Model (MLMM)
 Difference Between Residues and Residuals?
 What is an Operational Deflection Shape (ODS)?
 Making a Video of a Time Animation

Acoustics

 History of Acoustics
 Sound Pressure
 What is a decibel?
 What is A-weighting?
 Octaves and Human Hearing
 What is Sound Power?
 Decibel Math
 What is the Acoustic Quantity called Q?
 Sound Absorption
 Sound Transmission Loss
 Noise level certification, how to select the right standard?
 Sound pressure, sound power, and sound intensity: What is the difference?
 Sound fields: Free and diffuse field, near and far field

Sound Quality

 History of Acoustics
 Sound Pressure
 What is a decibel?
 What is A-weighting?
 Octaves and Human Hearing
 Loudness and Sones
 Calculating a Specific Loudness Spectrum in Simcenter Testlab
 How to Calculate N10 Time Varying Loudness in Simcenter Testlab?
 Articulation Index
 Auditory Masking

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 Tone-to-Noise Ratio and Prominence Ratio

 Fluctuation Strength and Roughness
 Critical Bands in Human Hearing
 Kurtosis
 Time-Frequency Analysis: Wavelets
 Sound Quality Jury Testing
 Alter Speed of Rotating Sound for Listening
 Simcenter Testlab Sound Diagnosis

Durability

 History of Fatigue
 Stress and Strain
 Calculating Damage with Miner's Rule
 What is a SN-Curve?
 Rainflow Counting
 Difference between 'Range-Mean' and 'From-To' Counting
 Power Spectral Density
 Shock Response Spectrum (SRS)
 How to Calculate a Shock Response Spectrum with Testlab?
 Some Thoughts on Accelerated Durability Testing
 Goodman-Haigh Diagram for Infinite Life
 Measuring strain gauges in Simcenter Testlab
 Rosette Strain Gauges
 Calculating Damage in Simcenter Tecware Process Builder
 Strain Gauges: Selecting an Excitation Voltage
 Simcenter Testlab SCADAS and Long Strain Cables

Rotating Machinery

 What is an order?
 Torsional Vibration: What is it?
 Zebra Tape Butt Joint Correction for Torsional Vibrations
 Balancing: Static, Coupled, and Dynamic
 Removing Spikes from RPM Signals
 Using the Smoothing function to remove rpm spikes
 RPM Calculation problems and Laser Tachometers
 Simcenter Testlab Signature
 Fixed Sampling versus Synchronous Sampling
 Tips and tricks for acquiring torsional orders
 Harmonic Removal
 Interpreting Colormaps
 Cycle to Cycle Averaging in Simcenter Testlab
 Speed Sweep Data Processing: RPM Increment, Framesize, Sweep Rate, and Overlap
 Order Cuts: How to Get the Correct Amplitude?

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Vibration Control

 What is Vibration Control Testing?

 Vibration Control FAQs
 Power Spectral Density
 Shock Response Spectrum (SRS)
 Sine Control: Estimation Methods
 Sine Control: Closed Loop Control Parameters
 Sine Control: Notching
 Vibration Control: Understanding Selfcheck
 Direct Field Acoustic Noise Testing
 What is Total Harmonic Distortion (THD)?
 Kurtosis
 "Excessive DC Level" message
 ITF Warning in Random Vibration Control

Simcenter Testlab General

 Simcenter Testlab Overview

 Simcenter Testlab Active Picture
 Simcenter Testlab Active Picture Plug-in Download
 Simcenter Testlab Token Licensing: What is it? How does it work?
 Simcenter Testlab: Import/Export to Excel
 Simcenter Testlab: Documentation Tab
 Simcenter Testlab: Templates
 Simcenter Testlab: Keyboard and Mouse Shortcuts
 Simcenter Testlab: CTRL-PageUp and CTRL-PageDown to change worksheets
 Simcenter Testlab: How to create a custom workflow
 How to lockdown a test in Simcenter Testlab
 Simcenter Testlab: Format Based Printing with Powerpoint
 Simcenter Testlab: Replace Data Origin...
 Simcenter Testlab: Batch Reporting
 Nastran and Simcenter Testlab: Viewing Mode Shapes and FRFs
 Import ANSYS Geometry and Mode Shapes in Simcenter Testlab

Simcenter Testlab Display tips:

 Simcenter Testlab: Display Layouts

 Simcenter Testlab: Customized Legend Text
 Simcenter Testlab: Curve Linestyles
 Simcenter Testlab: Cursor Tips and Tricks
 Simcenter Testlab: Coupled Cursors and Coupled Limits
 Simcenter Testlab: Interactive order cursor!
 Simcenter Testlab: Order Cursor vs Time in Colormap
 Simcenter Testlab: The "Double X cursor"
 Simcenter Testlab: Automatic Peak-Valley Cursor

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 Simcenter Testlab: Harmonic Cursors

 Simcenter Testlab: XY Graph
 Simcenter Testlab: Keyboard and Mouse Shortcuts
 Simcenter Testlab: Matrix-Heatmap Display
 Simcenter Testlab: Replace a curve in a display directly?
 Simcenter Testlab: Overlay Mode Shapes
 Simcenter Testlab: Compare data to target quickly with preview mode

Simcenter SCADAS and other Acquisition Hardware

 Simcenter SCADAS Mobile and Recorder Hardware

 Simcenter SCADAS XS: Everything you need to know!
 Simcenter SCADAS Cable Guide
 Simcenter Sound Camera
 Simcenter Soundbrush
 Qsources: Acoustic and Structural Exciters
 Simcenter Testlab Smart App Favorites!
 Single Ended versus Differential Inputs
 AC versus DC Coupling
 How to use SCADAS T8 Thermocouple card?
 ICP versus IEPE versus Charge Accelerometers
 Long Cable Lengths and ICP Transducers
 Long cable lengths and strain gauges

Simcenter Testlab Acquisition Tips

 Simcenter Testlab Signature

 Hands-free acquisition (almost)
 Capturing Transient Events in Simcenter Testlab
 Cool Channel Setup tricks for Triaxial Accelerometers
 Copy Channel Setup from One Project to Another?
 View Channel Setup from Desktop Navigator!
 Data Acquisition Based on GPS Location
 How to calculate a VECTOR SUM
 Tips and tricks for acquiring torsional orders
 CAN Bus Measurements
 No more tach! Use OBDII to get engine and vehicle speed...
 Record Raw CANBUS Data and Decode Offline!
 Single Plane Balancing in Simcenter Testlab
 Measuring strain gauges in Simcenter Testlab
 Correcting a bad calibration factor
 Simcenter Testlab Calibration

Simcenter Testlab Time Signal Calculator

 Simcenter Testlab: Time Signal Calculator Tips

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 Simcenter Testlab: Calculating Statistics from Time Histories

 Simcenter Testlab: Appending Time Histories
 Simcenter Testlab: Create a new recording from a segment cut
 Simcenter Testlab: Applying an arbitrary filter shape to a time history
 Simcenter Testlab: Integrating a Time History and Avoiding Offsets
 Analyzing PWM Signals with Time Signal Calculator
 Introduction to Filters: FIR vs IIR
 Applying a Custom Filter Shape: FILTER_FRF command
 Filter an Order and Listen to It
 Generate Custom Signals in Time Signal Calculator
 Removing Spikes from RPM Signals
 Harmonic Removal
 Rosette Strain Gauges

Simcenter Testlab Processing Tips

 Simcenter Testlab Throughput Processing Tips

 RPM Extraction in Simcenter Testlab
 Run Comparison in Simcenter Testlab
 Order Sections: Setting Parameters
 Time Signal Calculator Tips
 Simcenter Testlab: Calculating Statistics from Time Histories
 Simcenter Testlab: Appending Time Histories
 Simcenter Testlab: Create a new recording from a segment cut
 Simcenter Testlab Sound Diagnosis
 Can I view data in Google Earth?
 Analyzing PWM Signals in Simcenter Testlab

Simcenter Tecware

 Running Simcenter Tecware with Simcenter Testlab tokens

 Simcenter Tecware: Open Time Files fast!
 Simcenter Tecware: Overlay multiple signals in X and Y display
 Simcenter Tecware: Access to different file formats
 Simcenter Tecware: Spike Filter
 Simcenter Tecware: Removing Multiple Segments
 Simcenter Tecware: Calculate Infinite Life with the Goodman Diagram
 Calculating damage with Simcenter Tecware Processbuilder
 Butterworth filter "regular" and "zero phase"

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