Acoustics
Acoustics
Acoustics
Application Manual
VERSION 2016
by
Gamma Technologies
GT SUPPORT
TELEPHONE: (630) 325-5848
E-MAIL: [email protected]
TABLE OF CONTENTS
CHAPTER 1: Acoustics.......................................................................................................................... 1
1.1 General Acoustics and Signal Processing........................................................................................ 1
1.1.1 Sampling Rate, Storage Rate, and Aliasing:........................................................................... 2
1.1.2 Order Tracking and Harmonics: ............................................................................................. 2
1.2 Objects used to Predict and Analyze Sound and Frequency Content .............................................. 3
1.2.1 'AcoustExtMicrophone'........................................................................................................... 3
1.2.2 Flow Noise.............................................................................................................................. 3
1.2.3 Drive-by and Doppler Effect .................................................................................................. 4
1.2.4 External Data Configurations / Data Only Analysis............................................................... 5
1.2.5 'AcoustToWAVFile' ............................................................................................................... 5
1.2.6 'AcoustTransLoss'................................................................................................................... 5
1.2.7 'AcoustInsLoss' ....................................................................................................................... 6
1.2.8 'AcoustTf2Mic' ....................................................................................................................... 6
1.2.9 'Data' ....................................................................................................................................... 6
1.2.10 'FastFourierTr'......................................................................................................................... 6
1.2.11 'OrderTracking' ....................................................................................................................... 6
1.2.12 Calculation of Natural Frequencies (Eigenfrequencies) ......................................................... 7
1.3 The High Frequency Filter............................................................................................................... 7
1.4 Window Types and Widths.............................................................................................................. 8
CHAPTER 1: Acoustics
GT-SUITE includes capabilities to analyze a signal in the time domain by transforming it to the frequency
domain. In addition, it is possible to calculate the frequency response of various systems in the frequency
domain. This capability is known as Linear Acoustics. The first part of this chapter will focus on the
frequency analysis of time-domain signals. The latter part will describe the capabilities and use of Linear
Acoustics.
GT-SUITE allows one to either place analysis objects into a model file or to create a post-processing file
that includes or points to data from a simulation or measurement. When these objects are in a model file,
the data will be processed automatically when the simulation has finished. This makes the analysis of the
simulation results more convenient. When these objects are in a post-processing file (a model with only
analysis components on the map), the data can be analyzed using the Pre-Processing mode, which does
not initiate a time-based simulation. This capability is especially useful for analyzing simulations that
take a long time to run or when studying the effect of analysis options on the results. If the same set of
data needs to be analyzed several times, then it will be more efficient to perform the simulation in one
model and analyze the results in a separate model.
A signal in the time domain is converted to the frequency domain using a Fourier Transform. A Fourier
series is a periodic function that may be represented by a sum of sines and cosines whose frequencies are
integer multiples of the base frequency (1/period) of the signal, or in equation form,
Ft Ao An cos nt Bn sin nt
n 1
A o = Mean of F(t)
A n , B n are known as the Fourier Coefficients
The computation of the Fourier Transform of a function is performed in order to compute the Fourier
coefficients. Once the Fourier coefficients are known, the amplitude of at each frequency component is
calculated simply as the square root of the sum of the squares of the Fourier coefficients, or in equation
form,
An algorithm for the fast, efficient computation of Fourier transforms, today known as the Fast Fourier
Transform (FFT) is used by GT-SUITE to decompose a time-domain signal into frequencies and their
corresponding amplitudes.
To perform an FFT of a signal most efficiently, the signal must consist of some number of points that is a
power of 2, for example 1024, 2048, etc. Generally, a GT-SUITE simulation will not take 2N time steps
in a cycle, so in order to be able to take an FFT within GT-SUITE the signal is first interpolated onto the
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next highest power of 2 points from the number of points stored for analysis. GT-SUITE incorporates a
high order interpolating polynomial for this interpolation, as linear interpolation will introduce spurious
high frequencies into the signal which will distort the FFT.
A typical 4-stroke engine simulation will have a minimum of 720 points per cycle. The idle speed of a
typical passenger car engine is 800 RPM, yielding a fundamental frequency of 6.67 Hz. Hence, the
sampling frequency is 4800 Hz (720 multiplied by 6.67). By the Nyquist theorem, all frequency
components above 2400 Hz in the signal will be aliased in the low frequency range. This is an acceptable
limit, since one dimensional flow analysis typically is not accurate up to such high frequencies (plane
wave theory). This is a very conservative analysis, since most low speed simulations require time steps
smaller than 1 crank-angle degree to maintain stability and it is strongly recommended to use smaller
discretization lengths in models where acoustic predictions are made. Both of these factors will increase
the number of points per cycle in the simulation. In fact, one typical model using the recommended
discretization lengths had 4000 time steps per cycle at 1000 RPM, yielding a Nyquist folding frequency
of 16,666 Hz.
Sampling rate is a concern when running a model that takes analysis data from an external file and
processes it. Specifically, data may be saved from a GT-POWER simulation and used as input in another
model that analyzes the data. Care must be taken to ensure that all points in the signal are stored. GT-
SUITE applications store plotting data using an algorithm that prevents storage of excess data. The
default maximum is typically not sufficient, so should be increased in Plot Setup to a value that is greater
than the number of time steps in the model.
Transmission loss simulation is one exception to the guidelines described above. The calculations
involved in the 'AcoustTransLoss' template are complicated. Experience has shown that the results of this
calculation are very sensitive to the number of points used. It is recommended to ensure that enough data,
typically 4096, is sampled in this type of simulation. Typically, a 360 degree driver is used, which
dictates a maximum time step of 0.088 degrees.
The base frequency is determined by the engine speed or driver object. In GT-POWER the base
frequency is associated with the engine cycle. This is the frequency given by the engine speed in cycles
per second. There is also a rotational frequency associated with the revolution of the engine. This
rotational frequency will be twice the base frequency for 4-stroke engines. For 2-stroke engines, the base
and rotational frequencies will be the same. For example, consider an engine running at 6000 RPM. The
base frequency for a 4-stroke engine would be 50 Hz while the rotational frequency would be 100 Hz.
For a 2-stroke engine both the base and rotational frequencies would be 100 Hz.
Harmonics are defined as multiples of the base frequency. Orders are defined as multiples of the
rotational frequency. Due to this definition, there are half-orders that account for multiples of the base
frequency. Take the engine example above (6000 RPM). The 1st, 2nd, and 4th harmonics will have
frequencies of 50, 100, and 200 Hz, respectively. The 1st, 2nd, and 4th orders will have frequencies of 100,
200, and 400 Hz, respectively. The 2nd harmonic will have the same frequency as the 1st order. Likewise,
the 4th harmonic will have a frequency equal to the 2nd order. There is also a ½ order with a frequency
equal to the 1st harmonic, or 50 Hz. For a 2-stroke engine, the orders and harmonics will be the same.
1.2.1 'AcoustExtMicrophone'
This component is used to predict the free field sound pressure level (SPL) generated by any intake and/or
exhaust system opening. It has options for filtering, "windowing" and order tracking. It also can include
ground and drive-by effects in the predictions, can predict the effect of multiple sources (two tailpipes, for
instance) and can model flow noise (GT-POWER only).
The object is placed on the system map by the user, and connects to the orifice that connects to the inlet or
outlet pipe (which connects to an EndEnvironment part) via a sensor connection. (It should not be
connected to any other orifice in the system as these results will not be valid.) At the orifice end, the
sensor must sense the velocity at the orifice. The pressure at the free-field microphone location is
calculated by treating the orifice as a simple pulsating monopole, for which the velocity at the monopole
can be transformed to pressure at any location in the free field by the following equation.
*S d r
P * u t
const * * r dt c
= density of the free field medium
S = Cross sectional area of the orifice
const
r = distance of microphone from orifice
t = time
u = instantaneous fluid velocity at the orifice
c = speed of sound in the free field
The above formula is valid only for SPL in the free field, which is generally accepted to be several
diameters distance away from the orifice.
cylinder motion will be called "pulsation noise".) It occurs in both steady and unsteady flows and is not
dependent upon the pulsations produced by the engine. The sound power level is predicted using the
following equation:
The sound power level is converted to a sound pressure level at a distance from the opening by use of
standard acoustic relations and added to the pulsating sound pressure level.
From the equation, one can conclude that flow noise is most dependent upon the average velocity and
flow rate. The diameter will be constant for a given configuration and the pressure and temperature will
not change by much over the range of engine speeds and loads. From experience, the flow noise will
overwhelm the pulsation noise at high engine speeds and loads. Similarly, the flow noise will be
overwhelmed by the pulsation noise at low engine speeds and loads.
This model requires the calibration of the efficiency to match measured results. The efficiency includes
the effects of flow noise generated by all parts in an exhaust system, including baffles, perforations,
expansions and contractions. Addition or removal of parts in the system may change the efficiency and
require additional calibration. As a result, this model should be used with caution. It is advised to build a
database of efficiencies for different configurations. Once this database is large enough and the users
have developed enough experience, the database can be used to make estimates of efficiency during
development of new and untested exhaust systems.
Because calibration of the efficiency is often needed, it is suggested to save the velocity of the exit orifice
(by requesting the plot) and to make a model that is run in "pre-processing" mode, so one can make the
acoustic calculations quickly while calibrating. Based on the knowledge that flow noise dominates at the
highest engine speed, one can calibrate most quickly by calibrating the efficiency at only the highest
engine speed. Then, test at the lower engine speeds to confirm the quality of the calibration or adjust the
calibration as needed.
Please note from the equation that the flow noise model will not predict noise at specific frequencies, but
over the total range of frequencies. For this reason, the influence of flow noise is not included in most
plots. However, flow noise is included in the Campbell diagrams (contours of dB versus engine speed
and frequency) by distributing the noise over the whole frequency range and in the Order plots by
showing a total with and without flow noise. Flow noise can be included in the generation of a sound
(*.WAV) file.
microphone and tailpipe will vary according to the vehicle motion. In the latter situation, the prediction
of the SPL at the microphone must include the effect of the relative motion of the vehicle with respect to
the microphone, resulting in the well-known Doppler phase shift effect. In GT-POWER, this effect is
computed by including the component of relative motion of the vehicle with respect to the stationary
microphone in the gas velocity at the orifice. The relative motion component is computed simply as the
dot product of a unit vector between the radiating orifice and microphone with a unit vector in the vehicle
motion direction, multiplied by the vehicle speed. As a result, when the vehicle is traveling towards the
microphone, the vehicle motion relative velocity adds to the orifice velocity, resulting in shorter
wavelengths at the microphone, and thus higher frequency content. Conversely, when the vehicle is
moving away from the microphone, the relative motion subtracts from the orifice velocity, resulting in
longer wavelengths and thus lower frequencies. Thus, the same orifice signal will sound higher pitched
as it approaches the microphone, and lower pitched as it travels away, just like the train whistle does as it
approaches and then departs from a railroad crossing.
This is a very powerful configuration as it eliminates the need to wait for the full physical simulation to
finish before the acoustic results are computed. Therefore, a data only analysis is orders of magnitude
faster than a full simulation. This is especially useful as a beginner or when needing to analyze the same
data set multiple times, as it allows for the microphone settings to be changes and the results quickly
analyzed, for example, moving the microphone position, experimenting with ground effects, or calibrating
flow noise.
Most acoustic analysis templates operate directly on calculated data, and can therefore utilize external
data configurations. Along with the microphone the transmission loss, insertion loss, transfer function,
and fast Fourier transform calculations can all be used in external data configurations. For complete
details on setting up these simulations, see the context help for each individual object.
1.2.5 'AcoustToWAVFile'
This component is used to create a *.wav sound file from a microphone's predicted sound. This template
can create a file that uses the final steady state cycle repeatedly to efficiently generate a steady state sound
that is longer than the final cycle (GT-POWER only). Flow noise can be included in the sound file.
1.2.6 'AcoustTransLoss'
This component is used to conduct a 4 microphone transmission loss (TL) analysis which provides a
measure of the attenuation a silencing element will provide. The technique used is that of Chung and
Blaser, described in their paper, "Transfer Function Method of Measuring In-Duct Acoustic Properties,
Part I, Theory", from the J. Acoust. Soc. Am., 68(3), Sept., 1980. The method involves decomposing a
pressure signal into its forward and backward (or incident and reflected) components at two locations in
the system: one upstream of the silencing component and one downstream. The microphone pairs should
be placed close together (1 or 2 subvolumes apart) and close to the silencing element inlet and outlet, so
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CHAPTER 1 - Acoustics
as to minimize effects of friction and heat transfer on the transmission loss. The transmission loss is then
calculated as the ratio of the incident waves at the upstream and downstream locations, effectively
providing a measure of how much of the noise entering a silencing element is still present at the exit of
the element. The user can average the transmission loss results data to reduce numerical "noise" in the
result data.
Since a TL result is the ratio of the waves, the source strength itself should not affect the results.
Therefore, speaker mean velocity and velocity amplitude should be chosen to get good convergence and
reduce numerical noise. This can be done by setting the mean velocity to a small positive value (for
example, 1 m/s) and the velocity amplitude to a value significantly smaller than the mean (for example,
0.1 m/s).
1.2.7 'AcoustInsLoss'
This component is used to calculate insertion loss between two microphones. Insertion loss is simply the
frequency-by-frequency decibel difference between two external microphones. Technically, an insertion
loss is measured by calculating the SPL of an intake or exhaust system with silencing elements installed
and then again with the silencing elements replaced by straight pipes of the same length as the silencing
elements they replace. Then the frequency SPL components of the silenced system are subtracted from
the corresponding straight pipe system components. The resulting difference over the frequency range is
the insertion loss (GT-POWER only).
1.2.8 'AcoustTf2Mic'
This component is used to conduct a two microphone transfer function analysis. It simply takes the ratio
of the FFT of each of the two signals at their corresponding frequencies. When the signal being analyzed
is a pressure signal and the output is presented in the form of dB versus frequency, the result is known as
noise reduction.
1.2.9 'Data'
This component simply contains explicit data, either from another simulation or from an experiment, to
which the user may point when performing Fourier analysis. The user can performs FFT's on this data,
for example, and plot it versus another FFT performed in the simulation. Similarly, if the data object
points to velocity profiles from another simulation, they can use an external microphone object to predict
the noise generated by this Data object and another orifice, and then use an Insertion Loss object to
compute the insertion loss between the two data sets. When using an object made from this template, the
number of data points that are saved should be increased by raising the number of "Default Maximum
Plot Points" in Plot Setup to a higher value, typically 4096. The reduction in the number of points that
may occur by the default algorithm for saving plot data will reduce the accuracy of the FFT's and could
add clicking to the output of an 'AcoustToWAVFile' object.
1.2.10 'FastFourierTr'
This component is used to perform a Fast Fourier Transform on a signal. It is not restricted to flow and
acoustic quantities and may be used to analyze any type of signal that may be sensed or plotted. An FFT
analysis can also be performed in the external data configuration as described in the microphone section
above.
1.2.11 'OrderTracking'
This reference object is used to define the orders to be tracked in an analysis. It lets the user define a
specific set of orders to be plotted, along with the total SPL, for order tracking plots which depict SPL
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CHAPTER 1 - Acoustics
versus speed for a specified set of orders. The user will usually select the orders that correspond with the
known orders of interest for their engine, say 2, 4, 6, and 8 for a I4 engine, or 1.5, 3, 4.5, and 6 for a V6
engine. This order tracking object will then be referred to in an external microphone or FFT. (GT-
POWER only)
Filter Order Normalized Wave Number at 50% Amplitude Band Width from 0.9 to 0.1 Amplitude
-------------- ------------------------------------------------------- -----------------------------------------------
10 0.72 0.27
24 0.35 0.15
54 0.17 0.06
The above table shows that the 10th order filter has the highest cutoff frequency and the largest rolloff
bandwidth, while the 54th order filter has the lowest cutoff frequency and smallest rolloff bandwidth. The
24th order filter is a good compromise between these qualities. Experience has shown that it should be
used for most standard data analyses. The 54th order filter should be used in the 'AcoustExtMicrophone'
and 'AcoustToWAV' objects when a WAV sound is created. This eliminates any clicks and pops that
might be heard.
0.5
0.25
0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalized wave number
The signal is adjusted by weighting the signal towards the middle of the window (the finite duration over
which the signal will be analyzed) and away from the ends. This is also known as tapering or
"windowing". The available window types and windowing procedure are shown below.
Examination of the window types shown below reveals that windowing a function will in fact distort the
function, resulting in a signal that is different from the original signal. This is an undesirable byproduct
of the windowing procedure, but may be outweighed by any side-lobe leakage errors that result from the
FFT of a highly non-periodic signal. Therefore, any window should be avoided if possible. All steady
state models should not use a window since they produce a periodic signal. Even most transient do not
require a window since only 1 cycle is analyzed at a time and the signal is nearly periodic. However, a
rapid transient may produce a highly non-periodic signal that will require a window due to the leakage
errors.
The Hanning window will generally result in a signal attenuation of about 3 dB across the frequency
range. However, for nearly periodic signal, the frequency content could be significantly affected as well.
The Tilting window is especially useful when no signal attenuation is desired.
The equations used for the transient window types are listed below along with a figure showing the
window types. It is recommended to use the Hanning type since the derivatives with respect to time
remain continuous and smooth.
0.8
0.6
Weight
0.4
Hanning
0.2
Barlett
Welch
0
0 0.25 0.5 0.75 1
Time/Transient Window Width
The Tilting window subtracts from the signal itself F(t) a straight line that connects the signal endpoints.
Therefore, the signal to be analyzed F'(t), is given by
F (T ) F (0)
F ' (t ) F (t ) t F ( 0)
T
2.5
1.5
1
Before Tilting
0.5 S(t) Tilt
After Tilting
0
0 0.2 0.4 0.6 0.8 1
-0.5
-1
-1.5
Time/Transient Window Width
For models that will not be used to capture intake acoustic behavior, it is appropriate to use a relatively
simple representation of the air box. In this case, the primary concern is the pressure drop across the
assembly. For air boxes with very simple geometry (single cylindrical or rectangular chamber), a single
'Pipe*' or 'FlowSplit*' object can be used to model the volume. For more complex geometry (i.e. if the air
box "turns a corner" or has multiple connected chambers), it may be necessary to connect two or more
flowsplits to create the air box volume The most significant pressure drop occurs at the expansion into the
chamber and contraction at the chamber exit, and these pressure drops are captured even with a single
volume representation.
The pressure drop across the filter is usually small relative to the expansion/contraction pressure losses
and so the filter is generally ignored. Typically, the filter is folded, in order to increase the effective flow
area through the filter, thus minimizing (if not eliminating) the influence it has on the pressure loss in the
air box.
If the pressure loss through the air box of the model is not as large as the measured value, it is possible to
include additional pressure drop attributed to 3-D effects or the filter using an orifice connection. To use
this method, the air box should include at least two flowsplits connected by an 'OrificeConn'. This orifice
can be used to represent the filter and the diameter can be calibrated to provide the correct restriction. A
more detailed description of this is included in the "Basic Air Box" example in the Additional Flowsplit
Examples section of the Flow manual (GT-ISE File>Manuals>Modeling Theory>Flow.pdf)
Alternatively, a small diameter can be used, on the order of 0.1 mm, and the "Number of Holes" can be
calibrated to match the measured data. This is different because it will change the impedance and as a
result, the wave dynamic and acoustic behavior, of the air box. In addition, the "Hole Thickness" of the
'OrificeConn' (also in the Options tab) can be changed to model the thickness of the filter. This can also
change the behavior of the air box.
For models that will be used to predict intake acoustic behavior, it is usually necessary to include more
detail in the air box model. The sub-volumes that make up the air box volume should all be similar in
size to the sub-volumes in the remainder of the intake system. Because the typical air box has such a
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CHAPTER 2 - Modeling Guidelines
large cross section, this may make it necessary to connect these small sub-volumes in more than one
dimension. Even though most components solve the 1-D Navier-Stokes equations, the flowsplit solution
accounts for losses attributed to changes in momentum when flow changes direction (through an angle).
Because of this, it is possible to create a 3-dimensional grid of connected volumes. This provides a
pseudo-3D representation of the air box volume and allows pressure waves to "bounce" within the air box
in X, Y, and Z directions. An integrated tool named GEM3D has been provided that allows an air box to
be drawn in the same manner as a CAD program. The tool automatically creates the interconnected
flowsplits and pipes that make up the 3-D representation of the air box. This tool can be accessed from
the Tools menu in GT-ISE.
Pipe
Expansion Ø1 = Expansion Ø2
Expansion Ø3 = Expansion Ø2
The effect of expansion and contraction at the inlet and outlet of the air box,
has a large cross section, making it desirable to allow pressure waves to travel in more than one
dimension. Even though most components solve the 1-D Navier-Stokes equations, the flowsplit solution
accounts for losses attributed to changes in momentum when flow changes direction (through an angle).
Because of this, it is possible to create a 3-dimensional grid of connected volumes. This provides a quasi-
3D representation of the muffler volume and allows pressure waves to "bounce" within the muffler in X,
Y, and Z directions. An integrated tool called GEM3D has been provided that allows a muffler or
silencer to be drawn in the same manner as a CAD program. The tool automatically creates the
interconnected flowsplits and pipes that make up the 3-D representation of the muffler. This tool can be
accessed from the Tools menu in GT-ISE. More information about GEM3D can be found in the GEM3D
manual (File>Help>Manuals>Graphical Applications>GEM3D.pdf), tutorial
(File>Help>Tutorials>Graphical Applications>GEM3D>GEM3D-tutorials.pdf), and examples (File >
Examples).
Shell
Perforated Pipe
dx dx dx
Perforation Hole
3
1 2
dx
Expansion Ø1 = Pipe Ø1
Expansion Ø2 = Pipe Ø2
=2 Pipe Ø1 dx
1.5
2
p d wool base
CD
UL air wool 1 f droplet
2.4 Turbochargers
This section discusses the accuracy of acoustic predictions in a turbocharged or supercharged model. It is
really an answer to the question of "How accurate are the predictions of pressure waves that pass through
the compressor/turbine?".
The compressor/turbine models in GT-POWER are map-based (see previous sections), not physical
models of the blades, housing, etc. Therefore, the accuracy of the acoustic noise through the
compressor/turbine depends on the map's ability to correctly predict the transmission of the pressure
waves. With a map-based model, a pressure wave entering the compressor/turbine will result in a
pressure wave leaving. Unfortunately, there is no evidence to show this pressure wave is the same as the
real pressure wave. This is because the map data was determined from steady-state flow tests that do not
include pressure waves.
There is some evidence that the map-based model predicts acoustic behavior reasonably through turbines.
At the same time, there is evidence showing that the attenuation of noise through the compressor is too
low. This behavior is discussed in the SAE Paper "Acoustic One-Dimensional Compressor Model for
Integration in a Gas-Dynamic Code" (2012-01-0834) written by Antonio Torregrosa, Francisco Arnau,
Pedro Piqueras, Miguel Reyes-Belmonte from the Universitat Polizecnica de Valencia and Magnus
Knutsson, Johan Lennblad from the Volvo Car Corporation. This paper describes a physical approach
modeling the geometries inside the compressor housing and corrects pressure ratios. A measured
compressor map includes pressure losses of the compressor housing geometries. The pressures to
calculate the pressure ratio are typically measured upstream/downstream of the compressor housing.
Since additional flow elements with pressure losses are modeled in this approach, the compressor map
needs to be corrected.
Based on the SAE paper GT created a template called 'CompressorAcoustic'. It allows the input of the
compressor housing geometry and corrects the compressor map automatically for the pressure losses
introduced by the additional flow elements. The 'CompressorAcoustic' template inherits all features of
the 'Compressor' template and is intended to replace it on the model map. An example model using the
template can be found doing a right mouse click on the template and selecting Show Examples Using.
modeled. This can lead to modeling inaccuracies, particularly for conditions where the exhaust pressure
pulse amplitude is high (i.e. low engine speeds, integrated turbine exhaust manifolds, twin entry turbines,
etc.). Additionally, a small temporal offset is introduced due to the travelling time of the boundary
condition from volute inlet to impeller inlet.
These issues can be resolved by simply adding a pipe part before the turbine to represent the turbine
volute (scroll) volume. Since the flow losses in the volute are already considered in the turbine map data,
the pipe pressure loss coefficients and friction multiplier should be set to 0. If the volute volume is
known, the geometric input data can be calculated with the following equations1:
r r
l vol l 1
1
2
2
4 V
d vol
vol
l vol
For some cases, the volute volume may be unknown and the volute diameter must be assumed. The
diameter should be set to a value smaller than the inlet diameter of the volute to prevent overestimation of
the volute volume.
1. Aymanns, R., Scharf, J., Uhlmann, T., and Lückmann, D. "A Revision of Quasi Steady Modelling
of Turbocharger Turbines in the Simulation of Pulse Charged Engines," Motortechnische
Zeitschrift, Aug. 2012.
The image below shows a typical setup for such a device. The sound transfer device is connected to the
intake of the engine on one side and to the cabin side piping on the other. The 3rd port of the template
should be capped since a sound device will not have a bleed port. The properties of the diaphragm can be
modified in the pressure regulator object to match the desired characteristics. Once the model is setup,
the same microphone analysis can be done on the cabin piping to analyze the response of the sound
device.
>DPFAcoustic). Geometric inputs similar to the DieselParticFilter are needed to build a DPFAcoustic
component. Calibration of the DPFAcoustic follows steps outlined in the online help for this template.
The RLT contour map in GT-POST can be used to examine any RLT result and most input values from
the simulation. The variable to examine is chosen from the tree on the left side of the application. The
values are shown graphically on a map of the model on the right side. This provides a fast, easy, and
convenient way to find mistakes. Some of the variables that are useful when calibrating a model for
acoustics include, but are not limited to; diameters, lengths, pressures, temperatures, discharge
coefficients, wall temperatures, flow rates, and more.
Microphone pressure(s)
Microphone noise (dB, dB(A), order tracking, frequency contours, etc.)
It can be very valuable to have photos of the test-bench setup, especially showing the intake and exhaust
system setup. This will make it easier to spot differences between model and test-bench later-on, when
the engine is no longer on the test bench. It is common to have different intake and exhaust systems on
the test bench and in the vehicle.
3.3.2 Step 2) Calibrate the crank-angle resolved pressure near the cylinders
The next step is to compare the crank-angle resolved pressures near the cylinders. Things like effective
pipe lengths, characteristics lengths, and effective volumes can greatly affect the instantaneous pressure
(acoustics), but have little to no effect on performance quantities (average back pressure, VE, etc.). These
things will be more noticeable in engine tuning (VE) for locations near the cylinder (for example, the
intake runner length has a large impact on VE, while the zip tube length has a much smaller effect). If
you don't have measured crank-angle resolved pressure, this step will have to be skipped.
systems due to the hotter temperature and greater temperature variation along the system. This is
probably not significant for an intake system (unless a large temperature variation is expected, like a
turbocharged engine). If no measured temperatures exist, this step will have to be skipped.
4.1 Introduction
Linear acoustics refers to the analysis of an acoustic system using a linearized set of equations that
describe the flow and acoustic properties of the system. Linear analysis is a system characterization that
requires only the geometric dimensions, boundary conditions, and initial conditions. The analysis is
performed directly in the frequency domain, so a time based simulation is not performed. This is
different than the standard non-linear GT-POWER acoustic solver that operates in the time domain,
which must use mathematical tools (Fast Fourier Transforms) to transform results into the frequency
domain. This linear characterization is accomplished by representing the system with a transfer matrix.
The transfer matrix represents the relationship between the state variables at two points in a system. From
this representation the acoustic properties of the system can be calculated.
The linear analysis is designed for users who already have experience with acoustics and similar tools.
The linear analysis can be conducted with most of the same flow components that are used in a flow
simulation. This allows models to be built only once and reduces the number of tools used to conduct the
acoustic analysis.
The linear analysis captures the general behavior of a system. The main advantage to the linear analysis
is computation speed. A linear analysis can be done in a fraction of the time it takes for a GT-POWER
acoustic (non-linear) simulation to calculate the same result. Therefore, a good approximation of the
system response can be calculated using much less computation time than the standard solver.
The main disadvantages of the linear analysis are that it is not a physical flow simulation and it does not
capture some of the finer details. The analysis is a "black box" type of analysis and not a complete flow
simulation, so the details about the components in the system are not calculated. The linear analysis does
not calculate pressure, velocities, etc. for each component in the system. It only calculates the overall
system response.
The linear analysis is useful when a general system response is adequate and fast computation times are
desired. For example, a linear analysis is a good choice to compare different mufflers. Many different
muffler combinations can be analyzed quickly and the general response can be compared to determine
which muffler gives the desired response. The linear analysis may not be useful when finer details are
desired. For example, it may not be a good choice when studying the effect of intricate details inside the
muffler on the acoustic response of the muffler.
1.84
freq CS
D
freq = Frequency
C S = Speed of sound
D = Diameter
For typical air boxes, the equivalent diameter is on the order of 200mm and speed of sound is in the range
of 343 m/s. For typical mufflers, the equivalent diameter is on the order of 350mm and the speed of
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sound is between 500 and 600 m/s. Using these values, the range of valid frequencies can be estimated.
In general, the maximum valid frequency of a linear acoustic analysis is in the range of 1000 Hz.
The equations used in the linear acoustic analysis are linearized forms of the fluid dynamic equations
(which are non-linear). When these equations are linearized, it is assumed that the velocity perturbations
are small compared to the mean velocity. This means the velocity amplitude must be small compared to
the average velocity for the system that is being analyzed. Because of this assumption, the linear analysis
ignores second order terms that include the ratio of the perturbation velocity to the mean velocity. These
are assumed to be much less than first order terms. For this reason, non-linear analyses will be more
accurate than linear analyses when the velocity amplitude is not small compared the mean velocity.
2
P 2 jkM P k2
P 0
z2 1 M 2 dz 1 M2
P = pressure
z = coordinate of flow
j = -1) (imaginary number)
k = wave number
M0 = Mach number
The transfer matrix relates the state variables at different interfaces of one acoustic component thus
describing the response of the component. In using transfer matrix methods, a number of different
formulations are possible, depending on the choice of state variables. Most commonly the acoustic
pressure (p) and volumetric flow rate (u) are employed. Additionally, a formulation based on incident
(P+) and reflected (P-) waves and a formulation using reflection and transmission coefficients are
established.
The transfer matrix method is based on the representation of single elements in terms of a 2×2 matrix
(four coefficients) that describe how the state variables at one end of the element change towards another
end. For example, a transfer matrix for a component with two ends is shown below.
p1 t11 t12 p2
u1 t 21 t 22 u2
The state variables employed here are the acoustic pressure (p) and volumetric flow rate (u). The transfer
matrix is made up of the transfer matrix coefficients t 11 , t 12 , t 21 , t 22 . The transfer matrix relates the state
variables at 1 to the state variables at 2.
The transfer matrix of each individual component is calculated as a 2×2 matrix. Then, the matrix
coefficients for each component are combined into a large system matrix for the entire system. From this
system matrix, a 2×2 transfer matrix is created for the entire system.
TM conversion:
1 1 1/ 2 Y0 / 2
S M 0 1 T M1 where M 1 , M01
1 / Y1 1 / Y1 1/ 2 Y0 / 2
TM conversion:
1 S12 S 21 S 21
T1 , R2 , R1 , T2 S 22 S12
S11 S11 S11 S11
Conversion:
2 AS 1 Ksi c
PS ZS Y where Y
1 Ksi 1 Ksi A
PS = Source Pressure
ZS = Source Impedance
Y = Characteristic Impedance
= Density
c = Speed of Sound
A = Effective cross-sectional area
Conversion:
V 1
PS ZS
Adm Adm
PS = Source Pressure
ZS = Source Impedance
It is also possible to impose a variable mean flow as a function of engine speed. To do this a table of the
desired mean flow as a function of engine speed must be defined for the "Mean Flow Table" attribute on
the Main tab in the acoustic source object ('AcoustSource'). The mean flow must still be specified in
each orifice connection as these inputs will be used to scale the relative mass flow to be imposed at each
location in the system. The mass flow specified at the orifice connection adjacent to the source will get
the imposed mean flow from the table, while all other mass flows will be scaled accordingly based on
their relative value.
The linear analysis will store the mass flow rate used in the mean flow calculations to the "Average Mass
Flow Rate" RLT for all orifice connections.
4.6.2 Viscothermal
Viscous and thermal effects can be included in a linear acoustic analysis. Viscous effects are modeled by
including corrections for the viscosity of the fluid and thermal effects include corrections for thermal
conductivity, Prandtl number, and specific heat ratio as a function of temperature (details below). To turn
on these effects, the flag in the linear analysis object ('AcoustLinTransLoss', 'AcoustLinInsLoss',
or 'AcoustLinEigen') must be set to include viscothermal effects. The temperature used to determine the
fluid properties is taken from the wall temperature specified in each part.
Frequency analysis, the temperatures used for each part may be extracted from the results of a previous
simulation run (.gdx file). To do this, the previous simulation run must contain all the parts in the linear
model with the same names (that is how the method knows which temperature to use for each part). The
model may be either an engine model or flow model (no engine part). In the linear model, set
the Initialization State attribute in the Initialization folder of Run Setup to "user_imposed" and make sure
the Use RLTs to Initialize user_imposed Cases (Flow) attribute is checked. Specify the filename of the
previous simulation run that contains the desired temperatures for the Filename for RLT Initialization
attribute. Finally, the case number of the previous simulation run should be specified for the Case
number for RLT Initialization attribute.
For an RPM per order analysis, the temperatures can be given in the form of a table as a function of
engine speed. To do this an 'RLTDependenceXY' reference object is defined for the wall temperature of
each part. This object points to an 'XYTable' reference object that gives the temperature (Y values) in
units of K as a function of engine speed (X values) in units of RPM. The other attributes of
the 'RLTDependenceXY' object are not used and can be anything. When the model is set up like this, the
analysis will use the proper temperature correction at each engine speed. This allows for a variable
temperature effect with engine speed in the linear analysis. Typically the temperatures may be known
from a previous simulation or experiment.
The linear analysis will store the temperature used in the viscothermal calculations to the "Mass Averaged
Temperature" RLTs for all flow parts.
The details of the corrections used to calculate the mean flow and viscothermal effects can be seen in the
equations below. Directly below is a general transfer matrix equation as a function of the wave number
(k). This is a general form and does not contain the details about any particular component.
P0 T11 (k ) T12 ( k ) P1
u0 T21 (k ) T22 (k ) u1
The next equation below shows the same general transfer matrix with an exponential term as a correction
and the transfer matrix as a function of a modified wave number (k c ). If both the mean flow and
viscothermal effects are turned on, then this shows the general form of the transfer matrix that is used in
the linear analysis. If only the effects of mean flow are turned on, then the alpha ( ) term of the modified
wave number is not included (essentially set to 0). If only the viscothermal effects are turned on, then the
entire exponential term is not included and the denominator of the modified wave number is not included
(essentially both are set to 1).
P0 kM T ( k ) T12 ( k c ) P1
exp j 2
L 11 c
u0 1 M T21 ( k c ) T22 ( k c ) u1
k k 2 1
kc with (1 j) 1
1 M2 2R Pr
k = Wave number
kc = Modified wave number
M = Mach number
R = Geometric radius
= Dynamic viscosity
= Gamma
Pr = Prandtl number
= Frequency
4.7.1 'AcoustSource'
This template is used to provide the source characteristics in a linear model. These source characteristics
may be obtained from measured data, theoretical equations, an external code, or using the standard GT-
POWER non-linear solution. One of the following three formats for the source characteristics may be
used; pressure/impedance, amplitude/reflection, and volume/admittance. These can be specified versus
frequency or engine speed. The source characteristics may be generated by a 'MultiLoad' object (see
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below), measurements, analytical expressions, or even other codes. An existing source named "Speaker"
is provided in the template library. This source was generated in a non-linear simulation using
the 'MultiLoad' template and a white noise speaker. This source can be used when simulating a
broadband noise source (for example, transmission loss) or for general studies. This object must be on
the project map to start a linear acoustic simulation. The non-linear analog to this object is
the 'EndFlowSpeaker'.
4.7.2 'EndFlowRadiation'
This template is used to specify the radiation impedance boundary conditions at the downstream end.
This allows the boundary condition to be specified with spherical or hemispherical radiation, an anechoic
termination, or using a known impedance. This template provides standard conditions like pressure,
temperature, and composition, as well as the acoustic impedance of the boundary. It also can be
calculated in GT-POWER assuming an open end or a flange. The impedance boundary may be specified
as a function of frequency if the data is known. The 'OpenEnd' option will assume the flow ends in the
free field and the sound radiates spherically. The 'Flange' option will assume the sound radiates
hemispherically. There is also a flag to treat the boundary as anechoic. This template is the analog of
both the 'EndEnvironment' and 'EndFlowAnechoic' templates in the non-linear solver.
4.7.3 'AcoustLinTransLoss'
This template is used to conduct a linear 4-pole transmission loss of a system. It provides an effective
measure of the reduction of noise after it passes through the system. It will first calculate the transfer
matrix of the system, and then use this to determine the transmission loss. When performing a
transmission loss simulation the downstream boundary condition should be set to 'anechoic'. Therefore,
the anechoic option in the 'EndFlowRadiation' object should be 'on'. The non-linear analog to this object
is 'AcoustTransLoss'.
4.7.4 'AcoustLinInsLoss'
This template is used to conduct an insertion loss or a noise reduction calculation. This template will
simply calculate for each frequency the difference in sound pressure level sampled at two points. Noise
reduction is generally defined as the difference in sound pressure level at two different points. This
provides a measure of the level of noise that was reduced as it passed through a certain component or
system. Insertion loss is a special case for which the two points are at the same relative location on 2
different systems. The first point is on a reference system that typically contains a straight pipe while the
second system is on the system of interest. This provides a measure of the reduction of noise due to the
insertion of the particular system of interest. When calculating insertion loss the non-linear analog to this
object is the 'AcoustInsLoss'. When calculating noise reduction the non-linear analog to this object is
the 'AcoustTf2Mic' object.
4.7.5 'AcoustLinEigen'
This template calculates the natural frequencies (resonances) of a system. The resonances represent the
frequencies at which the system will naturally oscillate when excited. These resonances are identified by
conducting a frequency sweep with a constant excitation level applied. The response of the system to this
excitation is calculated at each frequency, and then normalized to the maximum response. This sets the
greatest amplitude response equal to 1 and scales all the other responses according to this. The natural
frequencies arise from a solution to an eigenfunction, so are also called eigenfrequencies or modes. All of
these eigenfrequencies are available in a plot or can be viewed in GT-POST in the same way as orders.
There is no non-linear analog to this template; however, the 'Order/Mode Storage Flag' attribute in Output
Setup will perform the same task (for more information see the output control help).
4.7.6 'AcoustLinExtMic'
This template is used to calculate the pressure at a microphone located in the free-field at some distance
from a system opening. The microphone pressure is calculated using the equation shown below. The
microphone pressure effectively gives the radiated noise that an observer would hear. Some of the
quantities calculated include sound pressure levels, carpet plots, order tracking plots, and frequency
contours. This object connects to the orifice via a standard sensor connection that senses the mass flow
rate. Either 'AcoustLinTransLoss' or 'AcoustLinInsLoss' must be present in the model to initiate the
calculation of the transfer matrix, which is used to calculate the SPL. The non-linear analog of this object
is the 'AcoustExtMicrophone'.
*S d
P * u TM
const * * r d
= density of the free field medium
S = Cross sectional area of the orifice
const
r = distance of microphone from orifice
= frequency
u(TM) = velocity calculated from the Transfer Matrix
4.7.7 'TMBlackBox'
This template is used to specify a 4-pole transfer matrix. This is used when the transfer matrix for a part
or system is already known. Rather than modeling the system, and to save time and complexity, this
template can be used to directly specify the transfer matrix. This template has no physical characteristics
or geometric dimensions. The transfer matrix is used in the linear analysis to calculate the quantity of
interest (ex. Transmission loss). This transfer matrix contains the real and imaginary parts of the 2x2
matrix. This data can come from a 'TMatrixGenerator' part (see below), experimental measurements,
analytical equations, other software, or specified in a User-model. There is no non-linear analog template
of this object.
4.7.8 'TMatrixGenerator'
This template is used to generate a transfer matrix from the results of a standard (non-linear) GT-POWER
acoustic simulation. The transfer matrix generated from this template can then be used in a
'TMBlackBox' template when conducting a linear analysis. This template is associated with linear
acoustic analysis but is used in a non-linear simulation. The transfer matrix is calculated by determining
the values that would give equivalent acoustic results for the system as the non-linear simulation. The
transfer matrix can then be used as an input to the 'TMBlackBox' object for use in a linear analysis. To
determine the entire transfer matrix the boundary conditions of the system must be reversed. To
accomplish this in a single simulation run, it is necessary to create a second circuit containing a copy of
the system of interest. Then the boundary conditions can be switched. This will allow for the calculation
of the transfer matrix.
4.7.9 'MultiLoad'
This template is used to characterize a source in the time-domain for use as input to linear acoustic
analysis. It uses the results of a standard (non-linear) GT-POWER acoustic simulation to generate the
equivalent source characteristics for use in a linear analysis. This template is associated with linear
acoustic analysis but is used in a non-linear simulation. The time-based source in the non-linear solver
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must be translated into the frequency domain for use with the linear analysis. The linear source
characteristics are calculated by determining the values that would give the equivalent acoustic results as
the non-linear simulation at the location of the 'MultiLoad' object. The source characteristics can then be
used in an 'AcoustSource' object to provide a source for a linear acoustic analysis. The source
characteristics can be calculated using one of three methods; internally calculated, externally calculated,
or externally measured. The internally calculated method will determine the source characteristics inside
of a system. When calculated externally, the source characteristics are obtained directly from the
volumetric flow rate at an opening to the environment. When measured externally the source
characteristics are obtained from a microphone object placed in the free-field.
Either 'AcoustLinTransLoss' or 'AcoustLinInsLoss' must also be present in the model to initiate the
calculation of the transfer matrix, which is used to calculate the source characteristics.
An 'AcousticSource' template is required on the project map for all linear acoustic simulations. When
the solver detects this template, it will conduct a linear acoustic analysis on the system connected to the
source.
It is necessary to modify the port numbers of certain linear acoustic parts when connecting them in a map.
The port numbers of the linear acoustic analysis parts begin at 0 and do not increment automatically.
When the linear acoustic objects ('AcoustLinTransLoss', 'AcoustLinEigen', and
'AcoustLinInsLoss') are connected to the existing model, all ports will be assigned a value of 0.
Therefore, it is necessary to renumber the ports accordingly when connecting these objects.
The direction of the link arrows is used by the linear analysis. The link direction must be from
the 'AcoustSource' object to the flow part. For intake systems, the acoustic source is generally placed at
the end nearest to the engine, which is the source of the noise. This results in link arrows pointing in the
direction opposite to the flow direction. For exhaust systems, the link arrows point in the same direction
as the flow.
The linear analysis must use 'Pipe*' parts at the source, downstream boundary (end environment or
anechoic), and any connection to a boundary. If the system has a 'FlowSplit*' part connected to any of
these boundaries, a 'Pipe*' part must be added in-between the flowsplit and boundary.
When using a linear microphone object it is necessary to place the 'AcoustLinInsLoss' object onto the
project map. This triggers the calculation of transfer matrix coefficients for the system, which are needed
to calculate the free-field sound pressure level.
The linear acoustic analysis is performed without predicting the flow in the system. The analysis requires
only geometric quantities, initial conditions, and boundary conditions. Therefore, the linear solution may
be done using a pre-process run.
A summary of each example has been included in each model to describe the purpose of the model, how
features and templates are being used, and what modeling or design function is being achieved.
1. Beranek, L., and Ver, I., eds., Noise and Vibration Control Engineering, John Wiley & Sons, Inc.:
New York, N.Y., 1992.
INDEX
A L
Absorbing material, 14 Linear Acoustics, 22
AcoustExtMicrophone, 3 Applicability, 22
Acoustic data, 19 Templates, 27
AcoustInsLoss, 6
AcoustLinEigen, 28 M
AcoustLinExtMic, 29
AcoustLinInsLoss, 28 Mean Flow effects, 26
AcoustLinTransLoss, 28 Measured data, 19
AcoustSource, 27 Muffler, 12
AcoustTf2Mic, 6 MultiLoad, 29
AcoustToWAVFile, 5
AcoustTransLoss, 5 O
Air box, 11 Orders, 2
Air filer, 11
Amplitude/Reflection, 25
P
B Pressure/Impedance, 25
Pressure/Volume Flow, 24
Bartlett, 9 Progressive/Reflective, 24
C R
Calibration, 19 Resonator, 12
Procedure, 20
Correlation, 19
Procedure, 20 S
Sound symposer, 17
D Source Characteristic formats, 25
Data
T
Measured, 19
Required, 19 Templates, 3, 27
DPFAcoustic, 17 TMatrixGenerator, 29
TMBlackBox, 29
E Transfer Matrix, 22
Formats, 24
EndFlowRadiation, 28 Theory, 23
Transmission/Reflection, 24
F Turbine
Filtering, 7 Volute, 15
FlowSplitAbsorbing, 14 Turbocharger, 15
H V
W Hanning, 9
Welch, 9
Welch, 9
Windows, 8
Window Types
Wool, 14
Bartlett, 9